From william.king at quentustech.com Tue May 1 00:16:16 2012 From: william.king at quentustech.com (William King) Date: Mon, 30 Apr 2012 13:16:16 -0700 Subject: [Freeswitch-users] How to update a live system with very little downtime (How does 'make current' work?) In-Reply-To: References: <361E98F99D3CC3439EED59BC1924ED695ECFC2@SERVER2003.SecuReachSystems.local> Message-ID: <4F9EF310.8000706@quentustech.com> For minimum downtime just leave FS running, go to where you have the FS source and do: git pull make /etc/init.d/freeswitch stop make install /etc/init.d/freeswitch start That's assuming you have a freeswitch init script. That'll leave you with only the downtime that it takes to restart FS, and the length of the make install(usually very short). William King Senior Engineer Quentus Technologies, INC 1037 NE 65th St Suite 273 Seattle, WA 98115 Main: (877) 211-9337 Office: (206) 388-4772 Cell: (253) 686-5518 william.king at quentustech.com On 04/30/2012 11:52 AM, Kerem Erciyes wrote: > Or you could always build a debian-live image that includes > freeswitch-current and all the wanpipe driver etc, and swap them out > at will at 5 mins downtime maximum, store all you data on HDD of /etc > and /freeswitch/conf and you are good to go.... > > On Tue, Apr 17, 2012 at 12:02 AM, Michael Collins > wrote: > > > > On Mon, Apr 16, 2012 at 12:01 PM, Jason Moran > > > wrote: > > Sorry for such a basic question: > > 1. If I do a 'make current' while a system is running it > will not have any effect until I restart FS, correct? > > 2. What if 'make current' fails? What if it fails and > the server gets restarted, will FS work? > > 3. Is it more advisable to stop FS, do the 'make > current', then start FS even though it will have a longer down > time? > > > > I know it's best to keep FS up-to-date -- but it is difficult > when you only have 1 system available and it is expected to > have 100% uptime.. > > Thanks, > Jason > > > Since you don't have a spare system (which is totally naughty, > btw, especially if your corporate overlords expect 100% uptime) > then my next question would be this: how are you set on CPU power, > RAM, and disk space? The reason I ask is that it is entirely > possible to create a secondary git clone of the FS repo and use a > completely different install path when running the configure > script. You could "make current" in that secondary repo and get a > sort of preview of what would happen w/o touching your production > build env or install dirs. If the update goes well in your test > scenario then you could do the update in your production environment. > > It's kinda hackish but if you have spare disk space then it's a > really cheap option. It isn't perfect, but it does give you at > least some semblance of "try before you buy." > > -MC > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Kerem Erciyes - Sistem Danismani > http://keremerciyes.com > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120430/c68db023/attachment-0001.html From y.aydogan at gmail.com Tue May 1 00:08:41 2012 From: y.aydogan at gmail.com (Y. Aydogan) Date: Mon, 30 Apr 2012 16:08:41 -0400 Subject: [Freeswitch-users] No ringback tone for google voice calls Message-ID: Hi, I am having some issues with ringback tones. I configured google voice and I don't hear any ringback tone for outgoing calls through google voice. Attached is a capture which shows an outgoing call. I replaced my destination number with 5555555555 in the logs for privacy. Thanks in advance. Yucel freeswitch at internal> 2012-04-29 12:49:51.258713 [DEBUG] sofia.c:6288 IP 192.168.0.101 Rejected by acl "net_list_5". Falling back to Digest auth. 2012-04-29 12:49:51.284881 [DEBUG] sofia.c:6288 IP 192.168.0.101 Rejected by acl "net_list_5". Falling back to Digest auth. 2012-04-29 12:49:51.289720 [NOTICE] switch_channel.c:784 New Channel sofia/sipinterface_1/yu... at 192.168.0.112 [27a51939- dda7-42db-8210-3aa643ebc7e6] 2012-04-29 12:49:51.289720 [DEBUG] switch_core_state_machine.c:318 (sofia/sipinterface_1/yu... at 192.168.0.112) Running State Change CS_NEW 2012-04-29 12:49:51.289720 [DEBUG] switch_core_state_machine.c:324 (sofia/sipinterface_1/yu... at 192.168.0.112) State NEW 2012-04-29 12:49:51.305713 [DEBUG] sofia.c:4559 Channel sofia/ sipinterface_1/yu... at 192.168.0.112 entering state [received][100] 2012-04-29 12:49:51.305713 [DEBUG] sofia.c:4570 Remote SDP: v=0 o=- 22714590 22714590 IN IP4 192.168.0.101 s=- c=IN IP4 192.168.0.101 t=0 0 m=audio 16416 RTP/AVP 0 2 4 8 18 96 97 98 100 101 a=rtpmap:0 PCMU/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729a/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=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 2012-04-29 12:49:51.305713 [DEBUG] sofia_glue.c:4353 Audio Codec Compare [PCMU:0:8000:30:64000]/[G7221:115:32000:20:48000] 2012-04-29 12:49:51.305713 [DEBUG] sofia_glue.c:4353 Audio Codec Compare [PCMU:0:8000:30:64000]/[G7221:107:16000:20:32000] 2012-04-29 12:49:51.305713 [DEBUG] sofia_glue.c:4353 Audio Codec Compare [PCMU:0:8000:30:64000]/[G722:9:8000:20:64000] 2012-04-29 12:49:51.305713 [DEBUG] sofia_glue.c:4353 Audio Codec Compare [PCMU:0:8000:30:64000]/[PCMU:0:8000:20:64000] 2012-04-29 12:49:51.305713 [DEBUG] sofia_glue.c:4353 Audio Codec Compare [PCMU:0:8000:30:64000]/[PCMA:8:8000:20:64000] 2012-04-29 12:49:51.305713 [DEBUG] sofia_glue.c:4353 Audio Codec Compare [PCMU:0:8000:30:64000]/[GSM:3:8000:20:13200] 2012-04-29 12:49:51.305713 [DEBUG] sofia_glue.c:4406 Substituting codec PCMU at 30i@8000h 2012-04-29 12:49:51.305713 [DEBUG] sofia_glue.c:2721 Set Codec sofia/ sipinterface_1/yu... at 192.168.0.112 PCMU/8000 30 ms 240 samples 64000 bits 2012-04-29 12:49:51.307729 [DEBUG] sofia_glue.c:4457 Set 2833 dtmf send/recv payload to 101 2012-04-29 12:49:51.307729 [DEBUG] sofia.c:4732 (sofia/sipinterface_1/ yu... at 192.168.0.112) State Change CS_NEW -> CS_INIT 2012-04-29 12:49:51.307729 [DEBUG] switch_core_session.c:1057 Send signal sofia/sipinterface_1/yu... at 192.168.0.112 [BREAK] 2012-04-29 12:49:51.307729 [DEBUG] switch_core_state_machine.c:318 (sofia/sipinterface_1/yu... at 192.168.0.112) Running State Change CS_INIT 2012-04-29 12:49:51.307729 [DEBUG] switch_core_state_machine.c:342 (sofia/sipinterface_1/yu... at 192.168.0.112) State INIT 2012-04-29 12:49:51.307729 [DEBUG] mod_sofia.c:83 sofia/sipinterface_1/ yu... at 192.168.0.112 SOFIA INIT 2012-04-29 12:49:51.307729 [DEBUG] mod_sofia.c:123 (sofia/ sipinterface_1/yu... at 192.168.0.112) State Change CS_INIT -> CS_ROUTING 2012-04-29 12:49:51.307729 [DEBUG] switch_core_session.c:1057 Send signal sofia/sipinterface_1/yu... at 192.168.0.112 [BREAK] 2012-04-29 12:49:51.307729 [DEBUG] switch_core_state_machine.c:342 (sofia/sipinterface_1/yu... at 192.168.0.112) State INIT going to sleep 2012-04-29 12:49:51.307729 [DEBUG] switch_core_state_machine.c:318 (sofia/sipinterface_1/yu... at 192.168.0.112) Running State Change CS_ROUTING 2012-04-29 12:49:51.307729 [DEBUG] switch_channel.c:1615 (sofia/ sipinterface_1/yu... at 192.168.0.112) Callstate Change DOWN -> RINGING 2012-04-29 12:49:51.307729 [DEBUG] switch_core_state_machine.c:345 (sofia/sipinterface_1/yu... at 192.168.0.112) State ROUTING 2012-04-29 12:49:51.307729 [DEBUG] mod_sofia.c:146 sofia/ sipinterface_1/yu... at 192.168.0.112 SOFIA ROUTING 2012-04-29 12:49:51.307729 [DEBUG] switch_core_state_machine.c:77 sofia/sipinterface_1/yu... at 192.168.0.112 Standard ROUTING 2012-04-29 12:49:51.307729 [INFO] mod_dialplan_xml.c:331 Processing Yucel Aydogan ->5555555555 in context context_1 Dialplan: sofia/sipinterface_1/yu... at 192.168.0.112 parsing [context_1- >dingaling_2_pattern_1] continue=true Dialplan: sofia/sipinterface_1/yu... at 192.168.0.112 Regex (PASS) [dingaling_2_pattern_1] destination_number(5555555555) =~ /^1{0,1} ([2-9][0-8][0-9][2-9][0-9]{6})$/ break=on-false Dialplan: sofia/sipinterface_1/yu... at 192.168.0.112 Action set(prepend=1) Dialplan: sofia/sipinterface_1/yu... at 192.168.0.112 Action set(hangup_after_bridge=true) Dialplan: sofia/sipinterface_1/yu... at 192.168.0.112 Action bridge(dingaling/GVOICE/+${prepend}5555555... at voice.google.com) Dialplan: sofia/sipinterface_1/yu... at 192.168.0.112 parsing [context_1- >conditioning_callerid] continue=true Dialplan: sofia/sipinterface_1/yu... at 192.168.0.112 Regex (PASS) [conditioning_callerid] ${internal_caller_id_number}(8309) =~ /^.+$/ break=on-false Dialplan: sofia/sipinterface_1/yu... at 192.168.0.112 Action set(effective_caller_id_name=${internal_caller_id_name}) Dialplan: sofia/sipinterface_1/yu... at 192.168.0.112 Action set(effective_caller_id_number=${internal_caller_id_number}) Dialplan: sofia/sipinterface_1/yu... at 192.168.0.112 parsing [context_1- >postroute_global] continue=true Dialplan: sofia/sipinterface_1/yu... at 192.168.0.112 Absolute Condition [postroute_global] Dialplan: sofia/sipinterface_1/yu... at 192.168.0.112 Action hash(insert/$ {domain_name}-spymap/${caller_id_number}/${uuid}) Dialplan: sofia/sipinterface_1/yu... at 192.168.0.112 Action hash(insert/$ {domain_name}-last_dial/${caller_id_number}/${destination_number}) Dialplan: sofia/sipinterface_1/yu... at 192.168.0.112 Action hash(insert/$ {domain_name}-last_dial/global/${uuid}) Dialplan: sofia/sipinterface_1/yu... at 192.168.0.112 Action set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) Dialplan: sofia/sipinterface_1/yu... at 192.168.0.112 parsing [context_1- >preanswer_gtalk] continue=true Dialplan: sofia/sipinterface_1/yu... at 192.168.0.112 Regex (FAIL) [preanswer_gtalk] source(mod_sofia) =~ /^mod_dingaling$/ break=on- false Dialplan: sofia/sipinterface_1/yu... at 192.168.0.112 parsing [context_1- >main_number_9] continue=true Dialplan: sofia/sipinterface_1/yu... at 192.168.0.112 Regex (FAIL) [main_number_9] destination_number(5555555555) =~ /^978nxxxxxx$/ break=on-false Dialplan: sofia/sipinterface_1/yu... at 192.168.0.112 parsing [context_1- >main_number_8] continue=true Dialplan: sofia/sipinterface_1/yu... at 192.168.0.112 Regex (FAIL) [main_number_8] destination_number(5555555555) =~ /^6000$/ break=on- false Dialplan: sofia/sipinterface_1/yu... at 192.168.0.112 parsing [context_1- >main_number_7] continue=true Dialplan: sofia/sipinterface_1/yu... at 192.168.0.112 Regex (FAIL) [main_number_7] destination_number(5555555555) =~ /^1039$/ break=on- false Dialplan: sofia/sipinterface_1/yu... at 192.168.0.112 parsing [context_1- >main_number_2] continue=true Dialplan: sofia/sipinterface_1/yu... at 192.168.0.112 Regex (FAIL) [main_number_2] destination_number(5555555555) =~ /^8309$/ break=on- false Dialplan: sofia/sipinterface_1/yu... at 192.168.0.112 parsing [context_1- >main_number_3] continue=true Dialplan: sofia/sipinterface_1/yu... at 192.168.0.112 Regex (FAIL) [main_number_3] destination_number(5555555555) =~ /^8310$/ break=on- false 2012-04-29 12:49:51.309717 [DEBUG] switch_core_state_machine.c:119 (sofia/sipinterface_1/yu... at 192.168.0.112) State Change CS_ROUTING -> CS_EXECUTE 2012-04-29 12:49:51.309717 [DEBUG] switch_core_session.c:1057 Send signal sofia/sipinterface_1/yu... at 192.168.0.112 [BREAK] 2012-04-29 12:49:51.309717 [DEBUG] switch_core_state_machine.c:345 (sofia/sipinterface_1/yu... at 192.168.0.112) State ROUTING going to sleep 2012-04-29 12:49:51.309717 [DEBUG] switch_core_state_machine.c:318 (sofia/sipinterface_1/yu... at 192.168.0.112) Running State Change CS_EXECUTE 2012-04-29 12:49:51.309717 [DEBUG] switch_core_state_machine.c:352 (sofia/sipinterface_1/yu... at 192.168.0.112) State EXECUTE 2012-04-29 12:49:51.309717 [DEBUG] mod_sofia.c:239 sofia/ sipinterface_1/yu... at 192.168.0.112 SOFIA EXECUTE 2012-04-29 12:49:51.309717 [DEBUG] switch_core_state_machine.c:157 sofia/sipinterface_1/yu... at 192.168.0.112 Standard EXECUTE EXECUTE sofia/sipinterface_1/yu... at 192.168.0.112 set(prepend=1) 2012-04-29 12:49:51.309717 [DEBUG] mod_dptools.c:1028 sofia/ sipinterface_1/yu... at 192.168.0.112 SET [prepend]=[1] EXECUTE sofia/sipinterface_1/yu... at 192.168.0.112 set(hangup_after_bridge=true) 2012-04-29 12:49:51.309717 [DEBUG] mod_dptools.c:1028 sofia/ sipinterface_1/yu... at 192.168.0.112 SET [hangup_after_bridge]=[true] EXECUTE sofia/sipinterface_1/yu... at 192.168.0.112 bridge(dingaling/ GVOICE/+15555555... at voice.google.com) 2012-04-29 12:49:51.309717 [NOTICE] switch_channel.c:784 New Channel dingaling/GVOICE/+15555555... at voice.google.com [92032331-e330-4269- b1fe-062de435aa78] 2012-04-29 12:49:51.309717 [DEBUG] mod_dingaling.c:1818 (dingaling/ GVOICE/+15555555... at voice.google.com) State Change CS_NEW -> CS_INIT 2012-04-29 12:49:51.309717 [DEBUG] switch_core_session.c:1057 Send signal dingaling/GVOICE/+15555555... at voice.google.com [BREAK] 2012-04-29 12:49:51.309717 [DEBUG] mod_dingaling.c:1342 dingaling/ GVOICE/+15555555... at voice.google.com CHANNEL KILL 2012-04-29 12:49:51.311724 [DEBUG] switch_core_state_machine.c:318 (dingaling/GVOICE/+15555555... at voice.google.com) Running State Change CS_INIT 2012-04-29 12:49:51.311724 [DEBUG] switch_core_state_machine.c:342 (dingaling/GVOICE/+15555555... at voice.google.com) State INIT 2012-04-29 12:49:51.311724 [NOTICE] mod_dingaling.c:1104 Ring-Ready dingaling/GVOICE/+15555555... at voice.google.com! 2012-04-29 12:49:51.311724 [DEBUG] mod_dingaling.c:1057 Don't have my codec yet here's one 2012-04-29 12:49:51.311724 [DEBUG] mod_dingaling.c:1077 Send Describe [PCMU at 8000] 2012-04-29 12:49:51.359973 [DEBUG] mod_dingaling.c:2935 using Existing session for 6160946247 2012-04-29 12:49:51.359973 [DEBUG] mod_dingaling.c:1077 Send Describe [PCMU at 8000] 2012-04-29 12:49:51.544699 [DEBUG] mod_dingaling.c:2935 using Existing session for 6160946247 2012-04-29 12:49:51.544699 [DEBUG] mod_dingaling.c:1002 Send Candidate 192.168.0.112:28560 [3FFPxUKMcoG72pKD] 2012-04-29 12:49:51.777702 [DEBUG] mod_dingaling.c:2935 using Existing session for 6160946247 2012-04-29 12:49:51.777702 [DEBUG] mod_dingaling.c:3273 1 candidates 2012-04-29 12:49:51.777702 [DEBUG] mod_dingaling.c:3293 candidate 173.194.76.127:19305 PASS ACL wan.auto 2012-04-29 12:49:51.777702 [DEBUG] mod_dingaling.c:3345 Acceptable Candidate 173.194.76.127:19305 2012-04-29 12:50:04.139089 [DEBUG] mod_dingaling.c:2935 using Existing session for 6160946247 2012-04-29 12:50:04.139089 [DEBUG] mod_dingaling.c:3187 Already decided on a codec 2012-04-29 12:50:04.141096 [DEBUG] mod_dingaling.c:859 Set Read Codec to PCMU at 8000 2012-04-29 12:50:04.141096 [DEBUG] mod_dingaling.c:874 Set Write Codec to PCMU at 8000 2012-04-29 12:50:05.193038 [DEBUG] switch_nat.c:500 mapped public port 28560 protocol UDP to localport 28560 2012-04-29 12:50:05.193038 [DEBUG] mod_dingaling.c:886 SETUP RTP 192.168.0.112:28560 -> 173.194.76.127:19305 2012-04-29 12:50:05.193038 [DEBUG] switch_rtp.c:1426 Not using a timer 2012-04-29 12:50:05.195316 [DEBUG] switch_rtp.c:3570 Activate VAD codec PCMU 20ms 2012-04-29 12:50:05.195316 [DEBUG] switch_channel.c:2694 (dingaling/ GVOICE/+15555555... at voice.google.com) Callstate Change DOWN -> ACTIVE 2012-04-29 12:50:05.195316 [DEBUG] switch_channel.c:2706 Send signal sofia/sipinterface_1/yu... at 192.168.0.112 [BREAK] 2012-04-29 12:50:05.195316 [NOTICE] mod_dingaling.c:1196 Channel [dingaling/GVOICE/+15555555... at voice.google.com] has been answered 2012-04-29 12:50:05.195316 [DEBUG] mod_dingaling.c:1199 (dingaling/ GVOICE/+15555555... at voice.google.com) State Change CS_INIT -> CS_ROUTING 2012-04-29 12:50:05.195316 [DEBUG] switch_core_session.c:1057 Send signal dingaling/GVOICE/+15555555... at voice.google.com [BREAK] 2012-04-29 12:50:05.195316 [DEBUG] mod_dingaling.c:1342 dingaling/ GVOICE/+15555555... at voice.google.com CHANNEL KILL 2012-04-29 12:50:05.195316 [DEBUG] switch_core_state_machine.c:342 (dingaling/GVOICE/+15555555... at voice.google.com) State INIT going to sleep 2012-04-29 12:50:05.195316 [DEBUG] switch_core_state_machine.c:318 (dingaling/GVOICE/+15555555... at voice.google.com) Running State Change CS_ROUTING 2012-04-29 12:50:05.195316 [DEBUG] switch_channel.c:1615 (dingaling/ GVOICE/+15555555... at voice.google.com) Callstate Change ACTIVE -> RINGING 2012-04-29 12:50:05.195316 [DEBUG] switch_core_state_machine.c:345 (dingaling/GVOICE/+15555555... at voice.google.com) State ROUTING 2012-04-29 12:50:05.195316 [DEBUG] mod_dingaling.c:1213 dingaling/ GVOICE/+15555555... at voice.google.com CHANNEL ROUTING 2012-04-29 12:50:05.195316 [DEBUG] switch_ivr_originate.c:66 (dingaling/GVOICE/+15555555... at voice.google.com) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2012-04-29 12:50:05.195316 [DEBUG] switch_core_session.c:1057 Send signal dingaling/GVOICE/+15555555... at voice.google.com [BREAK] 2012-04-29 12:50:05.195316 [DEBUG] mod_dingaling.c:1342 dingaling/ GVOICE/+15555555... at voice.google.com CHANNEL KILL 2012-04-29 12:50:05.195316 [DEBUG] switch_core_state_machine.c:345 (dingaling/GVOICE/+15555555... at voice.google.com) State ROUTING going to sleep 2012-04-29 12:50:05.195316 [DEBUG] switch_core_state_machine.c:318 (dingaling/GVOICE/+15555555... at voice.google.com) Running State Change CS_CONSUME_MEDIA 2012-04-29 12:50:05.195316 [DEBUG] switch_channel.c:1617 (dingaling/ GVOICE/+15555555... at voice.google.com) Callstate Change RINGING -> ACTIVE 2012-04-29 12:50:05.195316 [DEBUG] switch_core_state_machine.c:364 (dingaling/GVOICE/+15555555... at voice.google.com) State CONSUME_MEDIA 2012-04-29 12:50:05.195316 [DEBUG] switch_core_state_machine.c:364 (dingaling/GVOICE/+15555555... at voice.google.com) State CONSUME_MEDIA going to sleep 2012-04-29 12:50:05.198143 [NOTICE] mod_sofia.c:2096 Ring-Ready sofia/ sipinterface_1/yu... at 192.168.0.112! 2012-04-29 12:50:05.198143 [DEBUG] switch_core_session.c:676 Send signal sofia/sipinterface_1/yu... at 192.168.0.112 [BREAK] 2012-04-29 12:50:05.198143 [NOTICE] switch_ivr_originate.c:472 Ring Ready sofia/sipinterface_1/yu... at 192.168.0.112! 2012-04-29 12:50:05.198143 [DEBUG] sofia.c:4559 Channel sofia/ sipinterface_1/yu... at 192.168.0.112 entering state [early][180] 2012-04-29 12:50:05.225311 [DEBUG] sofia_glue.c:2961 AUDIO RTP [sofia/ sipinterface_1/yu... at 192.168.0.112] 192.168.0.112 port 19702 -> 192.168.0.101 port 16416 codec: 0 ms: 30 2012-04-29 12:50:05.225311 [DEBUG] switch_rtp.c:1418 Starting timer [soft] 240 bytes per 30ms 2012-04-29 12:50:05.227308 [DEBUG] sofia_glue.c:3179 Set 2833 dtmf send payload to 101 2012-04-29 12:50:05.227308 [DEBUG] sofia_glue.c:3184 Set 2833 dtmf receive payload to 101 2012-04-29 12:50:05.227308 [DEBUG] mod_sofia.c:680 Local SDP sofia/ sipinterface_1/yu... at 192.168.0.112: v=0 o=FreeSWITCH 1335709303 1335709304 IN IP4 192.168.0.112 s=FreeSWITCH c=IN IP4 192.168.0.112 t=0 0 m=audio 19702 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:30 a=sendrecv 2012-04-29 12:50:05.227308 [DEBUG] switch_core_session.c:676 Send signal sofia/sipinterface_1/yu... at 192.168.0.112 [BREAK] 2012-04-29 12:50:05.227308 [DEBUG] switch_channel.c:2694 (sofia/ sipinterface_1/yu... at 192.168.0.112) Callstate Change RINGING -> ACTIVE 2012-04-29 12:50:05.227308 [NOTICE] switch_ivr_originate.c:3288 Channel [sofia/sipinterface_1/yu... at 192.168.0.112] has been answered 2012-04-29 12:50:05.227308 [DEBUG] switch_ivr_originate.c:3333 Originate Resulted in Success: [dingaling/GVOICE/ +15555555... at voice.google.com] 2012-04-29 12:50:05.227308 [DEBUG] switch_core_session.c:676 Send signal dingaling/GVOICE/+15555555... at voice.google.com [BREAK] 2012-04-29 12:50:05.227308 [DEBUG] mod_dingaling.c:1342 dingaling/ GVOICE/+15555555... at voice.google.com CHANNEL KILL 2012-04-29 12:50:05.227308 [DEBUG] switch_core_session.c:676 Send signal sofia/sipinterface_1/yu... at 192.168.0.112 [BREAK] 2012-04-29 12:50:05.227308 [DEBUG] switch_ivr_bridge.c:1226 (dingaling/ GVOICE/+15555555... at voice.google.com) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA 2012-04-29 12:50:05.227308 [DEBUG] switch_core_session.c:1057 Send signal dingaling/GVOICE/+15555555... at voice.google.com [BREAK] 2012-04-29 12:50:05.227308 [DEBUG] mod_dingaling.c:1342 dingaling/ GVOICE/+15555555... at voice.google.com CHANNEL KILL 2012-04-29 12:50:05.230264 [DEBUG] switch_core_state_machine.c:318 (dingaling/GVOICE/+15555555... at voice.google.com) Running State Change CS_EXCHANGE_MEDIA 2012-04-29 12:50:05.230264 [DEBUG] switch_core_state_machine.c:355 (dingaling/GVOICE/+15555555... at voice.google.com) State EXCHANGE_MEDIA 2012-04-29 12:50:05.230264 [DEBUG] mod_dingaling.c:1350 CHANNEL LOOPBACK 2012-04-29 12:50:05.230264 [DEBUG] sofia.c:4559 Channel sofia/ sipinterface_1/yu... at 192.168.0.112 entering state [completed][200] 2012-04-29 12:50:05.242302 [DEBUG] sofia.c:4559 Channel sofia/ sipinterface_1/yu... at 192.168.0.112 entering state [ready][200] 2012-04-29 12:50:05.244336 [DEBUG] switch_rtp.c:2544 Correct ip/port confirmed. 2012-04-29 12:50:05.258274 [DEBUG] switch_core_io.c:970 Engaging Write Buffer at 480 bytes to accommodate 320->480 2012-04-29 12:50:05.267313 [DEBUG] switch_core_session.c:738 Send signal dingaling/GVOICE/+15555555... at voice.google.com [BREAK] 2012-04-29 12:50:05.267313 [DEBUG] mod_dingaling.c:1342 dingaling/ GVOICE/+15555555... at voice.google.com CHANNEL KILL 2012-04-29 12:50:05.267313 [DEBUG] switch_core_session.c:738 Send signal sofia/sipinterface_1/yu... at 192.168.0.112 [BREAK] 2012-04-29 12:50:05.358039 [DEBUG] switch_rtp.c:2544 Correct ip/port confirmed. 2012-04-29 12:50:05.358039 [DEBUG] switch_core_io.c:970 Engaging Write Buffer at 320 bytes to accommodate 480->320 From bdfoster at endigotech.com Tue May 1 03:44:57 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Mon, 30 Apr 2012 19:44:57 -0400 Subject: [Freeswitch-users] No ringback tone for google voice calls In-Reply-To: References: Message-ID: You have to generate it on your own, search ringback on the wiki. Its probably on the same page as the Google voice stuff. -BDF On Apr 30, 2012 5:00 PM, "Y. Aydogan" wrote: > Hi, > > I am having some issues with ringback tones. I configured google voice > and I don't hear any ringback tone for outgoing calls through google > voice. > > Attached is a capture which shows an outgoing call. I replaced my > destination number with 5555555555 in the logs for privacy. > > Thanks in advance. > Yucel > > > freeswitch at internal> 2012-04-29 12:49:51.258713 [DEBUG] sofia.c:6288 > IP 192.168.0.101 Rejected by acl "net_list_5". Falling back to Digest > auth. > 2012-04-29 12:49:51.284881 [DEBUG] sofia.c:6288 IP 192.168.0.101 > Rejected by acl "net_list_5". Falling back to Digest auth. > 2012-04-29 12:49:51.289720 [NOTICE] switch_channel.c:784 New Channel > sofia/sipinterface_1/yu... at 192.168.0.112 [27a51939- > dda7-42db-8210-3aa643ebc7e6] > 2012-04-29 12:49:51.289720 [DEBUG] switch_core_state_machine.c:318 > (sofia/sipinterface_1/yu... at 192.168.0.112) Running State Change CS_NEW > 2012-04-29 12:49:51.289720 [DEBUG] switch_core_state_machine.c:324 > (sofia/sipinterface_1/yu... at 192.168.0.112) State NEW > 2012-04-29 12:49:51.305713 [DEBUG] sofia.c:4559 Channel sofia/ > sipinterface_1/yu... at 192.168.0.112 entering state [received][100] > 2012-04-29 12:49:51.305713 [DEBUG] sofia.c:4570 Remote SDP: > v=0 > o=- 22714590 22714590 IN IP4 192.168.0.101 > s=- > c=IN IP4 192.168.0.101 > t=0 0 > m=audio 16416 RTP/AVP 0 2 4 8 18 96 97 98 100 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:2 G726-32/8000 > a=rtpmap:4 G723/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:18 G729a/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=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=ptime:30 > > 2012-04-29 12:49:51.305713 [DEBUG] sofia_glue.c:4353 Audio Codec > Compare [PCMU:0:8000:30:64000]/[G7221:115:32000:20:48000] > 2012-04-29 12:49:51.305713 [DEBUG] sofia_glue.c:4353 Audio Codec > Compare [PCMU:0:8000:30:64000]/[G7221:107:16000:20:32000] > 2012-04-29 12:49:51.305713 [DEBUG] sofia_glue.c:4353 Audio Codec > Compare [PCMU:0:8000:30:64000]/[G722:9:8000:20:64000] > 2012-04-29 12:49:51.305713 [DEBUG] sofia_glue.c:4353 Audio Codec > Compare [PCMU:0:8000:30:64000]/[PCMU:0:8000:20:64000] > 2012-04-29 12:49:51.305713 [DEBUG] sofia_glue.c:4353 Audio Codec > Compare [PCMU:0:8000:30:64000]/[PCMA:8:8000:20:64000] > 2012-04-29 12:49:51.305713 [DEBUG] sofia_glue.c:4353 Audio Codec > Compare [PCMU:0:8000:30:64000]/[GSM:3:8000:20:13200] > 2012-04-29 12:49:51.305713 [DEBUG] sofia_glue.c:4406 Substituting > codec PCMU at 30i@8000h > 2012-04-29 12:49:51.305713 [DEBUG] sofia_glue.c:2721 Set Codec sofia/ > sipinterface_1/yu... at 192.168.0.112 PCMU/8000 30 ms 240 samples 64000 > bits > 2012-04-29 12:49:51.307729 [DEBUG] sofia_glue.c:4457 Set 2833 dtmf > send/recv payload to 101 > 2012-04-29 12:49:51.307729 [DEBUG] sofia.c:4732 (sofia/sipinterface_1/ > yu... at 192.168.0.112) State Change CS_NEW -> CS_INIT > 2012-04-29 12:49:51.307729 [DEBUG] switch_core_session.c:1057 Send > signal sofia/sipinterface_1/yu... at 192.168.0.112 [BREAK] > 2012-04-29 12:49:51.307729 [DEBUG] switch_core_state_machine.c:318 > (sofia/sipinterface_1/yu... at 192.168.0.112) Running State Change > CS_INIT > 2012-04-29 12:49:51.307729 [DEBUG] switch_core_state_machine.c:342 > (sofia/sipinterface_1/yu... at 192.168.0.112) State INIT > 2012-04-29 12:49:51.307729 [DEBUG] mod_sofia.c:83 sofia/sipinterface_1/ > yu... at 192.168.0.112 SOFIA INIT > 2012-04-29 12:49:51.307729 [DEBUG] mod_sofia.c:123 (sofia/ > sipinterface_1/yu... at 192.168.0.112) State Change CS_INIT -> CS_ROUTING > 2012-04-29 12:49:51.307729 [DEBUG] switch_core_session.c:1057 Send > signal sofia/sipinterface_1/yu... at 192.168.0.112 [BREAK] > 2012-04-29 12:49:51.307729 [DEBUG] switch_core_state_machine.c:342 > (sofia/sipinterface_1/yu... at 192.168.0.112) State INIT going to sleep > 2012-04-29 12:49:51.307729 [DEBUG] switch_core_state_machine.c:318 > (sofia/sipinterface_1/yu... at 192.168.0.112) Running State Change > CS_ROUTING > 2012-04-29 12:49:51.307729 [DEBUG] switch_channel.c:1615 (sofia/ > sipinterface_1/yu... at 192.168.0.112) Callstate Change DOWN -> RINGING > 2012-04-29 12:49:51.307729 [DEBUG] switch_core_state_machine.c:345 > (sofia/sipinterface_1/yu... at 192.168.0.112) State ROUTING > 2012-04-29 12:49:51.307729 [DEBUG] mod_sofia.c:146 sofia/ > sipinterface_1/yu... at 192.168.0.112 SOFIA ROUTING > 2012-04-29 12:49:51.307729 [DEBUG] switch_core_state_machine.c:77 > sofia/sipinterface_1/yu... at 192.168.0.112 Standard ROUTING > 2012-04-29 12:49:51.307729 [INFO] mod_dialplan_xml.c:331 Processing > Yucel Aydogan ->5555555555 in context context_1 > Dialplan: sofia/sipinterface_1/yu... at 192.168.0.112 parsing [context_1- > >dingaling_2_pattern_1] continue=true > > Dialplan: sofia/sipinterface_1/yu... at 192.168.0.112 Regex (PASS) > [dingaling_2_pattern_1] destination_number(5555555555) =~ /^1{0,1} > ([2-9][0-8][0-9][2-9][0-9]{6})$/ break=on-false > Dialplan: sofia/sipinterface_1/yu... at 192.168.0.112 Action > set(prepend=1) > Dialplan: sofia/sipinterface_1/yu... at 192.168.0.112 Action > set(hangup_after_bridge=true) > Dialplan: sofia/sipinterface_1/yu... at 192.168.0.112 Action > bridge(dingaling/GVOICE/+${prepend}5555555... at voice.google.com) > Dialplan: sofia/sipinterface_1/yu... at 192.168.0.112 parsing [context_1- > >conditioning_callerid] continue=true > > Dialplan: sofia/sipinterface_1/yu... at 192.168.0.112 Regex (PASS) > [conditioning_callerid] ${internal_caller_id_number}(8309) =~ /^.+$/ > break=on-false > Dialplan: sofia/sipinterface_1/yu... at 192.168.0.112 Action > set(effective_caller_id_name=${internal_caller_id_name}) > Dialplan: sofia/sipinterface_1/yu... at 192.168.0.112 Action > set(effective_caller_id_number=${internal_caller_id_number}) > Dialplan: sofia/sipinterface_1/yu... at 192.168.0.112 parsing [context_1- > >postroute_global] continue=true > > Dialplan: sofia/sipinterface_1/yu... at 192.168.0.112 Absolute Condition > [postroute_global] > Dialplan: sofia/sipinterface_1/yu... at 192.168.0.112 Action hash(insert/$ > {domain_name}-spymap/${caller_id_number}/${uuid}) > Dialplan: sofia/sipinterface_1/yu... at 192.168.0.112 Action hash(insert/$ > {domain_name}-last_dial/${caller_id_number}/${destination_number}) > Dialplan: sofia/sipinterface_1/yu... at 192.168.0.112 Action hash(insert/$ > {domain_name}-last_dial/global/${uuid}) > Dialplan: sofia/sipinterface_1/yu... at 192.168.0.112 Action > set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) > Dialplan: sofia/sipinterface_1/yu... at 192.168.0.112 parsing [context_1- > >preanswer_gtalk] continue=true > > Dialplan: sofia/sipinterface_1/yu... at 192.168.0.112 Regex (FAIL) > [preanswer_gtalk] source(mod_sofia) =~ /^mod_dingaling$/ break=on- > false > Dialplan: sofia/sipinterface_1/yu... at 192.168.0.112 parsing [context_1- > >main_number_9] continue=true > > Dialplan: sofia/sipinterface_1/yu... at 192.168.0.112 Regex (FAIL) > [main_number_9] destination_number(5555555555) =~ /^978nxxxxxx$/ > break=on-false > Dialplan: sofia/sipinterface_1/yu... at 192.168.0.112 parsing [context_1- > >main_number_8] continue=true > > Dialplan: sofia/sipinterface_1/yu... at 192.168.0.112 Regex (FAIL) > [main_number_8] destination_number(5555555555) =~ /^6000$/ break=on- > false > Dialplan: sofia/sipinterface_1/yu... at 192.168.0.112 parsing [context_1- > >main_number_7] continue=true > > Dialplan: sofia/sipinterface_1/yu... at 192.168.0.112 Regex (FAIL) > [main_number_7] destination_number(5555555555) =~ /^1039$/ break=on- > false > Dialplan: sofia/sipinterface_1/yu... at 192.168.0.112 parsing [context_1- > >main_number_2] continue=true > > Dialplan: sofia/sipinterface_1/yu... at 192.168.0.112 Regex (FAIL) > [main_number_2] destination_number(5555555555) =~ /^8309$/ break=on- > false > Dialplan: sofia/sipinterface_1/yu... at 192.168.0.112 parsing [context_1- > >main_number_3] continue=true > > Dialplan: sofia/sipinterface_1/yu... at 192.168.0.112 Regex (FAIL) > [main_number_3] destination_number(5555555555) =~ /^8310$/ break=on- > false > 2012-04-29 12:49:51.309717 [DEBUG] switch_core_state_machine.c:119 > (sofia/sipinterface_1/yu... at 192.168.0.112) State Change CS_ROUTING -> > CS_EXECUTE > 2012-04-29 12:49:51.309717 [DEBUG] switch_core_session.c:1057 Send > signal sofia/sipinterface_1/yu... at 192.168.0.112 [BREAK] > 2012-04-29 12:49:51.309717 [DEBUG] switch_core_state_machine.c:345 > (sofia/sipinterface_1/yu... at 192.168.0.112) State ROUTING going to > sleep > 2012-04-29 12:49:51.309717 [DEBUG] switch_core_state_machine.c:318 > (sofia/sipinterface_1/yu... at 192.168.0.112) Running State Change > CS_EXECUTE > 2012-04-29 12:49:51.309717 [DEBUG] switch_core_state_machine.c:352 > (sofia/sipinterface_1/yu... at 192.168.0.112) State EXECUTE > 2012-04-29 12:49:51.309717 [DEBUG] mod_sofia.c:239 sofia/ > sipinterface_1/yu... at 192.168.0.112 SOFIA EXECUTE > 2012-04-29 12:49:51.309717 [DEBUG] switch_core_state_machine.c:157 > sofia/sipinterface_1/yu... at 192.168.0.112 Standard EXECUTE > EXECUTE sofia/sipinterface_1/yu... at 192.168.0.112 set(prepend=1) > 2012-04-29 12:49:51.309717 [DEBUG] mod_dptools.c:1028 sofia/ > sipinterface_1/yu... at 192.168.0.112 SET [prepend]=[1] > EXECUTE sofia/sipinterface_1/yu... at 192.168.0.112 > set(hangup_after_bridge=true) > 2012-04-29 12:49:51.309717 [DEBUG] mod_dptools.c:1028 sofia/ > sipinterface_1/yu... at 192.168.0.112 SET [hangup_after_bridge]=[true] > EXECUTE sofia/sipinterface_1/yu... at 192.168.0.112 bridge(dingaling/ > GVOICE/+15555555... at voice.google.com) > 2012-04-29 12:49:51.309717 [NOTICE] switch_channel.c:784 New Channel > dingaling/GVOICE/+15555555... at voice.google.com [92032331-e330-4269- > b1fe-062de435aa78] > 2012-04-29 12:49:51.309717 [DEBUG] mod_dingaling.c:1818 (dingaling/ > GVOICE/+15555555... at voice.google.com) State Change CS_NEW -> CS_INIT > 2012-04-29 12:49:51.309717 [DEBUG] switch_core_session.c:1057 Send > signal dingaling/GVOICE/+15555555... at voice.google.com [BREAK] > 2012-04-29 12:49:51.309717 [DEBUG] mod_dingaling.c:1342 dingaling/ > GVOICE/+15555555... at voice.google.com CHANNEL KILL > 2012-04-29 12:49:51.311724 [DEBUG] switch_core_state_machine.c:318 > (dingaling/GVOICE/+15555555... at voice.google.com) Running State Change > CS_INIT > 2012-04-29 12:49:51.311724 [DEBUG] switch_core_state_machine.c:342 > (dingaling/GVOICE/+15555555... at voice.google.com) State INIT > 2012-04-29 12:49:51.311724 [NOTICE] mod_dingaling.c:1104 Ring-Ready > dingaling/GVOICE/+15555555... at voice.google.com! > 2012-04-29 12:49:51.311724 [DEBUG] mod_dingaling.c:1057 Don't have my > codec yet here's one > 2012-04-29 12:49:51.311724 [DEBUG] mod_dingaling.c:1077 Send Describe > [PCMU at 8000] > 2012-04-29 12:49:51.359973 [DEBUG] mod_dingaling.c:2935 using Existing > session for 6160946247 > 2012-04-29 12:49:51.359973 [DEBUG] mod_dingaling.c:1077 Send Describe > [PCMU at 8000] > 2012-04-29 12:49:51.544699 [DEBUG] mod_dingaling.c:2935 using Existing > session for 6160946247 > 2012-04-29 12:49:51.544699 [DEBUG] mod_dingaling.c:1002 Send Candidate > 192.168.0.112:28560 [3FFPxUKMcoG72pKD] > 2012-04-29 12:49:51.777702 [DEBUG] mod_dingaling.c:2935 using Existing > session for 6160946247 > 2012-04-29 12:49:51.777702 [DEBUG] mod_dingaling.c:3273 1 candidates > 2012-04-29 12:49:51.777702 [DEBUG] mod_dingaling.c:3293 candidate > 173.194.76.127:19305 PASS ACL wan.auto > 2012-04-29 12:49:51.777702 [DEBUG] mod_dingaling.c:3345 Acceptable > Candidate 173.194.76.127:19305 > 2012-04-29 12:50:04.139089 [DEBUG] mod_dingaling.c:2935 using Existing > session for 6160946247 > 2012-04-29 12:50:04.139089 [DEBUG] mod_dingaling.c:3187 Already > decided on a codec > 2012-04-29 12:50:04.141096 [DEBUG] mod_dingaling.c:859 Set Read Codec > to PCMU at 8000 > 2012-04-29 12:50:04.141096 [DEBUG] mod_dingaling.c:874 Set Write Codec > to PCMU at 8000 > 2012-04-29 12:50:05.193038 [DEBUG] switch_nat.c:500 mapped public port > 28560 protocol UDP to localport 28560 > 2012-04-29 12:50:05.193038 [DEBUG] mod_dingaling.c:886 SETUP RTP > 192.168.0.112:28560 -> 173.194.76.127:19305 > 2012-04-29 12:50:05.193038 [DEBUG] switch_rtp.c:1426 Not using a timer > 2012-04-29 12:50:05.195316 [DEBUG] switch_rtp.c:3570 Activate VAD > codec PCMU 20ms > 2012-04-29 12:50:05.195316 [DEBUG] switch_channel.c:2694 (dingaling/ > GVOICE/+15555555... at voice.google.com) Callstate Change DOWN -> ACTIVE > 2012-04-29 12:50:05.195316 [DEBUG] switch_channel.c:2706 Send signal > sofia/sipinterface_1/yu... at 192.168.0.112 [BREAK] > 2012-04-29 12:50:05.195316 [NOTICE] mod_dingaling.c:1196 Channel > [dingaling/GVOICE/+15555555... at voice.google.com] has been answered > 2012-04-29 12:50:05.195316 [DEBUG] mod_dingaling.c:1199 (dingaling/ > GVOICE/+15555555... at voice.google.com) State Change CS_INIT -> > CS_ROUTING > 2012-04-29 12:50:05.195316 [DEBUG] switch_core_session.c:1057 Send > signal dingaling/GVOICE/+15555555... at voice.google.com [BREAK] > 2012-04-29 12:50:05.195316 [DEBUG] mod_dingaling.c:1342 dingaling/ > GVOICE/+15555555... at voice.google.com CHANNEL KILL > 2012-04-29 12:50:05.195316 [DEBUG] switch_core_state_machine.c:342 > (dingaling/GVOICE/+15555555... at voice.google.com) State INIT going to > sleep > 2012-04-29 12:50:05.195316 [DEBUG] switch_core_state_machine.c:318 > (dingaling/GVOICE/+15555555... at voice.google.com) Running State Change > CS_ROUTING > 2012-04-29 12:50:05.195316 [DEBUG] switch_channel.c:1615 (dingaling/ > GVOICE/+15555555... at voice.google.com) Callstate Change ACTIVE -> > RINGING > 2012-04-29 12:50:05.195316 [DEBUG] switch_core_state_machine.c:345 > (dingaling/GVOICE/+15555555... at voice.google.com) State ROUTING > 2012-04-29 12:50:05.195316 [DEBUG] mod_dingaling.c:1213 dingaling/ > GVOICE/+15555555... at voice.google.com CHANNEL ROUTING > 2012-04-29 12:50:05.195316 [DEBUG] switch_ivr_originate.c:66 > (dingaling/GVOICE/+15555555... at voice.google.com) State Change > CS_ROUTING -> CS_CONSUME_MEDIA > 2012-04-29 12:50:05.195316 [DEBUG] switch_core_session.c:1057 Send > signal dingaling/GVOICE/+15555555... at voice.google.com [BREAK] > 2012-04-29 12:50:05.195316 [DEBUG] mod_dingaling.c:1342 dingaling/ > GVOICE/+15555555... at voice.google.com CHANNEL KILL > 2012-04-29 12:50:05.195316 [DEBUG] switch_core_state_machine.c:345 > (dingaling/GVOICE/+15555555... at voice.google.com) State ROUTING going > to sleep > 2012-04-29 12:50:05.195316 [DEBUG] switch_core_state_machine.c:318 > (dingaling/GVOICE/+15555555... at voice.google.com) Running State Change > CS_CONSUME_MEDIA > 2012-04-29 12:50:05.195316 [DEBUG] switch_channel.c:1617 (dingaling/ > GVOICE/+15555555... at voice.google.com) Callstate Change RINGING -> > ACTIVE > 2012-04-29 12:50:05.195316 [DEBUG] switch_core_state_machine.c:364 > (dingaling/GVOICE/+15555555... at voice.google.com) State CONSUME_MEDIA > 2012-04-29 12:50:05.195316 [DEBUG] switch_core_state_machine.c:364 > (dingaling/GVOICE/+15555555... at voice.google.com) State CONSUME_MEDIA > going to sleep > 2012-04-29 12:50:05.198143 [NOTICE] mod_sofia.c:2096 Ring-Ready sofia/ > sipinterface_1/yu... at 192.168.0.112! > 2012-04-29 12:50:05.198143 [DEBUG] switch_core_session.c:676 Send > signal sofia/sipinterface_1/yu... at 192.168.0.112 [BREAK] > 2012-04-29 12:50:05.198143 [NOTICE] switch_ivr_originate.c:472 Ring > Ready sofia/sipinterface_1/yu... at 192.168.0.112! > 2012-04-29 12:50:05.198143 [DEBUG] sofia.c:4559 Channel sofia/ > sipinterface_1/yu... at 192.168.0.112 entering state [early][180] > 2012-04-29 12:50:05.225311 [DEBUG] sofia_glue.c:2961 AUDIO RTP [sofia/ > sipinterface_1/yu... at 192.168.0.112] 192.168.0.112 port 19702 -> > 192.168.0.101 port 16416 codec: 0 ms: 30 > 2012-04-29 12:50:05.225311 [DEBUG] switch_rtp.c:1418 Starting timer > [soft] 240 bytes per 30ms > 2012-04-29 12:50:05.227308 [DEBUG] sofia_glue.c:3179 Set 2833 dtmf > send payload to 101 > 2012-04-29 12:50:05.227308 [DEBUG] sofia_glue.c:3184 Set 2833 dtmf > receive payload to 101 > 2012-04-29 12:50:05.227308 [DEBUG] mod_sofia.c:680 Local SDP sofia/ > sipinterface_1/yu... at 192.168.0.112: > v=0 > o=FreeSWITCH 1335709303 1335709304 IN IP4 192.168.0.112 > s=FreeSWITCH > c=IN IP4 192.168.0.112 > t=0 0 > m=audio 19702 RTP/AVP 0 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:30 > a=sendrecv > > 2012-04-29 12:50:05.227308 [DEBUG] switch_core_session.c:676 Send > signal sofia/sipinterface_1/yu... at 192.168.0.112 [BREAK] > 2012-04-29 12:50:05.227308 [DEBUG] switch_channel.c:2694 (sofia/ > sipinterface_1/yu... at 192.168.0.112) Callstate Change RINGING -> ACTIVE > 2012-04-29 12:50:05.227308 [NOTICE] switch_ivr_originate.c:3288 > Channel [sofia/sipinterface_1/yu... at 192.168.0.112] has been answered > 2012-04-29 12:50:05.227308 [DEBUG] switch_ivr_originate.c:3333 > Originate Resulted in Success: [dingaling/GVOICE/ > +15555555... at voice.google.com] > 2012-04-29 12:50:05.227308 [DEBUG] switch_core_session.c:676 Send > signal dingaling/GVOICE/+15555555... at voice.google.com [BREAK] > 2012-04-29 12:50:05.227308 [DEBUG] mod_dingaling.c:1342 dingaling/ > GVOICE/+15555555... at voice.google.com CHANNEL KILL > 2012-04-29 12:50:05.227308 [DEBUG] switch_core_session.c:676 Send > signal sofia/sipinterface_1/yu... at 192.168.0.112 [BREAK] > 2012-04-29 12:50:05.227308 [DEBUG] switch_ivr_bridge.c:1226 (dingaling/ > GVOICE/+15555555... at voice.google.com) State Change CS_CONSUME_MEDIA -> > CS_EXCHANGE_MEDIA > 2012-04-29 12:50:05.227308 [DEBUG] switch_core_session.c:1057 Send > signal dingaling/GVOICE/+15555555... at voice.google.com [BREAK] > 2012-04-29 12:50:05.227308 [DEBUG] mod_dingaling.c:1342 dingaling/ > GVOICE/+15555555... at voice.google.com CHANNEL KILL > 2012-04-29 12:50:05.230264 [DEBUG] switch_core_state_machine.c:318 > (dingaling/GVOICE/+15555555... at voice.google.com) Running State Change > CS_EXCHANGE_MEDIA > 2012-04-29 12:50:05.230264 [DEBUG] switch_core_state_machine.c:355 > (dingaling/GVOICE/+15555555... at voice.google.com) State EXCHANGE_MEDIA > 2012-04-29 12:50:05.230264 [DEBUG] mod_dingaling.c:1350 CHANNEL > LOOPBACK > 2012-04-29 12:50:05.230264 [DEBUG] sofia.c:4559 Channel sofia/ > sipinterface_1/yu... at 192.168.0.112 entering state [completed][200] > 2012-04-29 12:50:05.242302 [DEBUG] sofia.c:4559 Channel sofia/ > sipinterface_1/yu... at 192.168.0.112 entering state [ready][200] > 2012-04-29 12:50:05.244336 [DEBUG] switch_rtp.c:2544 Correct ip/port > confirmed. > 2012-04-29 12:50:05.258274 [DEBUG] switch_core_io.c:970 Engaging Write > Buffer at 480 bytes to accommodate 320->480 > 2012-04-29 12:50:05.267313 [DEBUG] switch_core_session.c:738 Send > signal dingaling/GVOICE/+15555555... at voice.google.com [BREAK] > 2012-04-29 12:50:05.267313 [DEBUG] mod_dingaling.c:1342 dingaling/ > GVOICE/+15555555... at voice.google.com CHANNEL KILL > 2012-04-29 12:50:05.267313 [DEBUG] switch_core_session.c:738 Send > signal sofia/sipinterface_1/yu... at 192.168.0.112 [BREAK] > 2012-04-29 12:50:05.358039 [DEBUG] switch_rtp.c:2544 Correct ip/port > confirmed. > 2012-04-29 12:50:05.358039 [DEBUG] switch_core_io.c:970 Engaging Write > Buffer at 320 bytes to accommodate 480->320 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120430/6b360c1b/attachment-0001.html From bdfoster at endigotech.com Tue May 1 04:55:51 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Mon, 30 Apr 2012 20:55:51 -0400 Subject: [Freeswitch-users] No ringback tone for google voice calls In-Reply-To: References: Message-ID: Actually here it is: http://wiki.freeswitch.org/wiki/Variable_ringback Set this before you bridge the call. -BDF On Apr 30, 2012 7:44 PM, "Brian Foster" wrote: > You have to generate it on your own, search ringback on the wiki. Its > probably on the same page as the Google voice stuff. > > -BDF > On Apr 30, 2012 5:00 PM, "Y. Aydogan" wrote: > >> Hi, >> >> I am having some issues with ringback tones. I configured google voice >> and I don't hear any ringback tone for outgoing calls through google >> voice. >> >> Attached is a capture which shows an outgoing call. I replaced my >> destination number with 5555555555 in the logs for privacy. >> >> Thanks in advance. >> Yucel >> >> >> freeswitch at internal> 2012-04-29 12:49:51.258713 [DEBUG] sofia.c:6288 >> IP 192.168.0.101 Rejected by acl "net_list_5". Falling back to Digest >> auth. >> 2012-04-29 12:49:51.284881 [DEBUG] sofia.c:6288 IP 192.168.0.101 >> Rejected by acl "net_list_5". Falling back to Digest auth. >> 2012-04-29 12:49:51.289720 [NOTICE] switch_channel.c:784 New Channel >> sofia/sipinterface_1/yu... at 192.168.0.112 [27a51939- >> dda7-42db-8210-3aa643ebc7e6] >> 2012-04-29 12:49:51.289720 [DEBUG] switch_core_state_machine.c:318 >> (sofia/sipinterface_1/yu... at 192.168.0.112) Running State Change CS_NEW >> 2012-04-29 12:49:51.289720 [DEBUG] switch_core_state_machine.c:324 >> (sofia/sipinterface_1/yu... at 192.168.0.112) State NEW >> 2012-04-29 12:49:51.305713 [DEBUG] sofia.c:4559 Channel sofia/ >> sipinterface_1/yu... at 192.168.0.112 entering state [received][100] >> 2012-04-29 12:49:51.305713 [DEBUG] sofia.c:4570 Remote SDP: >> v=0 >> o=- 22714590 22714590 IN IP4 192.168.0.101 >> s=- >> c=IN IP4 192.168.0.101 >> t=0 0 >> m=audio 16416 RTP/AVP 0 2 4 8 18 96 97 98 100 101 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:2 G726-32/8000 >> a=rtpmap:4 G723/8000 >> a=rtpmap:8 PCMA/8000 >> a=rtpmap:18 G729a/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=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-15 >> a=ptime:30 >> >> 2012-04-29 12:49:51.305713 [DEBUG] sofia_glue.c:4353 Audio Codec >> Compare [PCMU:0:8000:30:64000]/[G7221:115:32000:20:48000] >> 2012-04-29 12:49:51.305713 [DEBUG] sofia_glue.c:4353 Audio Codec >> Compare [PCMU:0:8000:30:64000]/[G7221:107:16000:20:32000] >> 2012-04-29 12:49:51.305713 [DEBUG] sofia_glue.c:4353 Audio Codec >> Compare [PCMU:0:8000:30:64000]/[G722:9:8000:20:64000] >> 2012-04-29 12:49:51.305713 [DEBUG] sofia_glue.c:4353 Audio Codec >> Compare [PCMU:0:8000:30:64000]/[PCMU:0:8000:20:64000] >> 2012-04-29 12:49:51.305713 [DEBUG] sofia_glue.c:4353 Audio Codec >> Compare [PCMU:0:8000:30:64000]/[PCMA:8:8000:20:64000] >> 2012-04-29 12:49:51.305713 [DEBUG] sofia_glue.c:4353 Audio Codec >> Compare [PCMU:0:8000:30:64000]/[GSM:3:8000:20:13200] >> 2012-04-29 12:49:51.305713 [DEBUG] sofia_glue.c:4406 Substituting >> codec PCMU at 30i@8000h >> 2012-04-29 12:49:51.305713 [DEBUG] sofia_glue.c:2721 Set Codec sofia/ >> sipinterface_1/yu... at 192.168.0.112 PCMU/8000 30 ms 240 samples 64000 >> bits >> 2012-04-29 12:49:51.307729 [DEBUG] sofia_glue.c:4457 Set 2833 dtmf >> send/recv payload to 101 >> 2012-04-29 12:49:51.307729 [DEBUG] sofia.c:4732 (sofia/sipinterface_1/ >> yu... at 192.168.0.112) State Change CS_NEW -> CS_INIT >> 2012-04-29 12:49:51.307729 [DEBUG] switch_core_session.c:1057 Send >> signal sofia/sipinterface_1/yu... at 192.168.0.112 [BREAK] >> 2012-04-29 12:49:51.307729 [DEBUG] switch_core_state_machine.c:318 >> (sofia/sipinterface_1/yu... at 192.168.0.112) Running State Change >> CS_INIT >> 2012-04-29 12:49:51.307729 [DEBUG] switch_core_state_machine.c:342 >> (sofia/sipinterface_1/yu... at 192.168.0.112) State INIT >> 2012-04-29 12:49:51.307729 [DEBUG] mod_sofia.c:83 sofia/sipinterface_1/ >> yu... at 192.168.0.112 SOFIA INIT >> 2012-04-29 12:49:51.307729 [DEBUG] mod_sofia.c:123 (sofia/ >> sipinterface_1/yu... at 192.168.0.112) State Change CS_INIT -> CS_ROUTING >> 2012-04-29 12:49:51.307729 [DEBUG] switch_core_session.c:1057 Send >> signal sofia/sipinterface_1/yu... at 192.168.0.112 [BREAK] >> 2012-04-29 12:49:51.307729 [DEBUG] switch_core_state_machine.c:342 >> (sofia/sipinterface_1/yu... at 192.168.0.112) State INIT going to sleep >> 2012-04-29 12:49:51.307729 [DEBUG] switch_core_state_machine.c:318 >> (sofia/sipinterface_1/yu... at 192.168.0.112) Running State Change >> CS_ROUTING >> 2012-04-29 12:49:51.307729 [DEBUG] switch_channel.c:1615 (sofia/ >> sipinterface_1/yu... at 192.168.0.112) Callstate Change DOWN -> RINGING >> 2012-04-29 12:49:51.307729 [DEBUG] switch_core_state_machine.c:345 >> (sofia/sipinterface_1/yu... at 192.168.0.112) State ROUTING >> 2012-04-29 12:49:51.307729 [DEBUG] mod_sofia.c:146 sofia/ >> sipinterface_1/yu... at 192.168.0.112 SOFIA ROUTING >> 2012-04-29 12:49:51.307729 [DEBUG] switch_core_state_machine.c:77 >> sofia/sipinterface_1/yu... at 192.168.0.112 Standard ROUTING >> 2012-04-29 12:49:51.307729 [INFO] mod_dialplan_xml.c:331 Processing >> Yucel Aydogan ->5555555555 in context context_1 >> Dialplan: sofia/sipinterface_1/yu... at 192.168.0.112 parsing [context_1- >> >dingaling_2_pattern_1] continue=true >> >> Dialplan: sofia/sipinterface_1/yu... at 192.168.0.112 Regex (PASS) >> [dingaling_2_pattern_1] destination_number(5555555555) =~ /^1{0,1} >> ([2-9][0-8][0-9][2-9][0-9]{6})$/ break=on-false >> Dialplan: sofia/sipinterface_1/yu... at 192.168.0.112 Action >> set(prepend=1) >> Dialplan: sofia/sipinterface_1/yu... at 192.168.0.112 Action >> set(hangup_after_bridge=true) >> Dialplan: sofia/sipinterface_1/yu... at 192.168.0.112 Action >> bridge(dingaling/GVOICE/+${prepend}5555555... at voice.google.com) >> Dialplan: sofia/sipinterface_1/yu... at 192.168.0.112 parsing [context_1- >> >conditioning_callerid] continue=true >> >> Dialplan: sofia/sipinterface_1/yu... at 192.168.0.112 Regex (PASS) >> [conditioning_callerid] ${internal_caller_id_number}(8309) =~ /^.+$/ >> break=on-false >> Dialplan: sofia/sipinterface_1/yu... at 192.168.0.112 Action >> set(effective_caller_id_name=${internal_caller_id_name}) >> Dialplan: sofia/sipinterface_1/yu... at 192.168.0.112 Action >> set(effective_caller_id_number=${internal_caller_id_number}) >> Dialplan: sofia/sipinterface_1/yu... at 192.168.0.112 parsing [context_1- >> >postroute_global] continue=true >> >> Dialplan: sofia/sipinterface_1/yu... at 192.168.0.112 Absolute Condition >> [postroute_global] >> Dialplan: sofia/sipinterface_1/yu... at 192.168.0.112 Action hash(insert/$ >> {domain_name}-spymap/${caller_id_number}/${uuid}) >> Dialplan: sofia/sipinterface_1/yu... at 192.168.0.112 Action hash(insert/$ >> {domain_name}-last_dial/${caller_id_number}/${destination_number}) >> Dialplan: sofia/sipinterface_1/yu... at 192.168.0.112 Action hash(insert/$ >> {domain_name}-last_dial/global/${uuid}) >> Dialplan: sofia/sipinterface_1/yu... at 192.168.0.112 Action >> set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) >> Dialplan: sofia/sipinterface_1/yu... at 192.168.0.112 parsing [context_1- >> >preanswer_gtalk] continue=true >> >> Dialplan: sofia/sipinterface_1/yu... at 192.168.0.112 Regex (FAIL) >> [preanswer_gtalk] source(mod_sofia) =~ /^mod_dingaling$/ break=on- >> false >> Dialplan: sofia/sipinterface_1/yu... at 192.168.0.112 parsing [context_1- >> >main_number_9] continue=true >> >> Dialplan: sofia/sipinterface_1/yu... at 192.168.0.112 Regex (FAIL) >> [main_number_9] destination_number(5555555555) =~ /^978nxxxxxx$/ >> break=on-false >> Dialplan: sofia/sipinterface_1/yu... at 192.168.0.112 parsing [context_1- >> >main_number_8] continue=true >> >> Dialplan: sofia/sipinterface_1/yu... at 192.168.0.112 Regex (FAIL) >> [main_number_8] destination_number(5555555555) =~ /^6000$/ break=on- >> false >> Dialplan: sofia/sipinterface_1/yu... at 192.168.0.112 parsing [context_1- >> >main_number_7] continue=true >> >> Dialplan: sofia/sipinterface_1/yu... at 192.168.0.112 Regex (FAIL) >> [main_number_7] destination_number(5555555555) =~ /^1039$/ break=on- >> false >> Dialplan: sofia/sipinterface_1/yu... at 192.168.0.112 parsing [context_1- >> >main_number_2] continue=true >> >> Dialplan: sofia/sipinterface_1/yu... at 192.168.0.112 Regex (FAIL) >> [main_number_2] destination_number(5555555555) =~ /^8309$/ break=on- >> false >> Dialplan: sofia/sipinterface_1/yu... at 192.168.0.112 parsing [context_1- >> >main_number_3] continue=true >> >> Dialplan: sofia/sipinterface_1/yu... at 192.168.0.112 Regex (FAIL) >> [main_number_3] destination_number(5555555555) =~ /^8310$/ break=on- >> false >> 2012-04-29 12:49:51.309717 [DEBUG] switch_core_state_machine.c:119 >> (sofia/sipinterface_1/yu... at 192.168.0.112) State Change CS_ROUTING -> >> CS_EXECUTE >> 2012-04-29 12:49:51.309717 [DEBUG] switch_core_session.c:1057 Send >> signal sofia/sipinterface_1/yu... at 192.168.0.112 [BREAK] >> 2012-04-29 12:49:51.309717 [DEBUG] switch_core_state_machine.c:345 >> (sofia/sipinterface_1/yu... at 192.168.0.112) State ROUTING going to >> sleep >> 2012-04-29 12:49:51.309717 [DEBUG] switch_core_state_machine.c:318 >> (sofia/sipinterface_1/yu... at 192.168.0.112) Running State Change >> CS_EXECUTE >> 2012-04-29 12:49:51.309717 [DEBUG] switch_core_state_machine.c:352 >> (sofia/sipinterface_1/yu... at 192.168.0.112) State EXECUTE >> 2012-04-29 12:49:51.309717 [DEBUG] mod_sofia.c:239 sofia/ >> sipinterface_1/yu... at 192.168.0.112 SOFIA EXECUTE >> 2012-04-29 12:49:51.309717 [DEBUG] switch_core_state_machine.c:157 >> sofia/sipinterface_1/yu... at 192.168.0.112 Standard EXECUTE >> EXECUTE sofia/sipinterface_1/yu... at 192.168.0.112 set(prepend=1) >> 2012-04-29 12:49:51.309717 [DEBUG] mod_dptools.c:1028 sofia/ >> sipinterface_1/yu... at 192.168.0.112 SET [prepend]=[1] >> EXECUTE sofia/sipinterface_1/yu... at 192.168.0.112 >> set(hangup_after_bridge=true) >> 2012-04-29 12:49:51.309717 [DEBUG] mod_dptools.c:1028 sofia/ >> sipinterface_1/yu... at 192.168.0.112 SET [hangup_after_bridge]=[true] >> EXECUTE sofia/sipinterface_1/yu... at 192.168.0.112 bridge(dingaling/ >> GVOICE/+15555555... at voice.google.com) >> 2012-04-29 12:49:51.309717 [NOTICE] switch_channel.c:784 New Channel >> dingaling/GVOICE/+15555555... at voice.google.com [92032331-e330-4269- >> b1fe-062de435aa78] >> 2012-04-29 12:49:51.309717 [DEBUG] mod_dingaling.c:1818 (dingaling/ >> GVOICE/+15555555... at voice.google.com) State Change CS_NEW -> CS_INIT >> 2012-04-29 12:49:51.309717 [DEBUG] switch_core_session.c:1057 Send >> signal dingaling/GVOICE/+15555555... at voice.google.com [BREAK] >> 2012-04-29 12:49:51.309717 [DEBUG] mod_dingaling.c:1342 dingaling/ >> GVOICE/+15555555... at voice.google.com CHANNEL KILL >> 2012-04-29 12:49:51.311724 [DEBUG] switch_core_state_machine.c:318 >> (dingaling/GVOICE/+15555555... at voice.google.com) Running State Change >> CS_INIT >> 2012-04-29 12:49:51.311724 [DEBUG] switch_core_state_machine.c:342 >> (dingaling/GVOICE/+15555555... at voice.google.com) State INIT >> 2012-04-29 12:49:51.311724 [NOTICE] mod_dingaling.c:1104 Ring-Ready >> dingaling/GVOICE/+15555555... at voice.google.com! >> 2012-04-29 12:49:51.311724 [DEBUG] mod_dingaling.c:1057 Don't have my >> codec yet here's one >> 2012-04-29 12:49:51.311724 [DEBUG] mod_dingaling.c:1077 Send Describe >> [PCMU at 8000] >> 2012-04-29 12:49:51.359973 [DEBUG] mod_dingaling.c:2935 using Existing >> session for 6160946247 >> 2012-04-29 12:49:51.359973 [DEBUG] mod_dingaling.c:1077 Send Describe >> [PCMU at 8000] >> 2012-04-29 12:49:51.544699 [DEBUG] mod_dingaling.c:2935 using Existing >> session for 6160946247 >> 2012-04-29 12:49:51.544699 [DEBUG] mod_dingaling.c:1002 Send Candidate >> 192.168.0.112:28560 [3FFPxUKMcoG72pKD] >> 2012-04-29 12:49:51.777702 [DEBUG] mod_dingaling.c:2935 using Existing >> session for 6160946247 >> 2012-04-29 12:49:51.777702 [DEBUG] mod_dingaling.c:3273 1 candidates >> 2012-04-29 12:49:51.777702 [DEBUG] mod_dingaling.c:3293 candidate >> 173.194.76.127:19305 PASS ACL wan.auto >> 2012-04-29 12:49:51.777702 [DEBUG] mod_dingaling.c:3345 Acceptable >> Candidate 173.194.76.127:19305 >> 2012-04-29 12:50:04.139089 [DEBUG] mod_dingaling.c:2935 using Existing >> session for 6160946247 >> 2012-04-29 12:50:04.139089 [DEBUG] mod_dingaling.c:3187 Already >> decided on a codec >> 2012-04-29 12:50:04.141096 [DEBUG] mod_dingaling.c:859 Set Read Codec >> to PCMU at 8000 >> 2012-04-29 12:50:04.141096 [DEBUG] mod_dingaling.c:874 Set Write Codec >> to PCMU at 8000 >> 2012-04-29 12:50:05.193038 [DEBUG] switch_nat.c:500 mapped public port >> 28560 protocol UDP to localport 28560 >> 2012-04-29 12:50:05.193038 [DEBUG] mod_dingaling.c:886 SETUP RTP >> 192.168.0.112:28560 -> 173.194.76.127:19305 >> 2012-04-29 12:50:05.193038 [DEBUG] switch_rtp.c:1426 Not using a timer >> 2012-04-29 12:50:05.195316 [DEBUG] switch_rtp.c:3570 Activate VAD >> codec PCMU 20ms >> 2012-04-29 12:50:05.195316 [DEBUG] switch_channel.c:2694 (dingaling/ >> GVOICE/+15555555... at voice.google.com) Callstate Change DOWN -> ACTIVE >> 2012-04-29 12:50:05.195316 [DEBUG] switch_channel.c:2706 Send signal >> sofia/sipinterface_1/yu... at 192.168.0.112 [BREAK] >> 2012-04-29 12:50:05.195316 [NOTICE] mod_dingaling.c:1196 Channel >> [dingaling/GVOICE/+15555555... at voice.google.com] has been answered >> 2012-04-29 12:50:05.195316 [DEBUG] mod_dingaling.c:1199 (dingaling/ >> GVOICE/+15555555... at voice.google.com) State Change CS_INIT -> >> CS_ROUTING >> 2012-04-29 12:50:05.195316 [DEBUG] switch_core_session.c:1057 Send >> signal dingaling/GVOICE/+15555555... at voice.google.com [BREAK] >> 2012-04-29 12:50:05.195316 [DEBUG] mod_dingaling.c:1342 dingaling/ >> GVOICE/+15555555... at voice.google.com CHANNEL KILL >> 2012-04-29 12:50:05.195316 [DEBUG] switch_core_state_machine.c:342 >> (dingaling/GVOICE/+15555555... at voice.google.com) State INIT going to >> sleep >> 2012-04-29 12:50:05.195316 [DEBUG] switch_core_state_machine.c:318 >> (dingaling/GVOICE/+15555555... at voice.google.com) Running State Change >> CS_ROUTING >> 2012-04-29 12:50:05.195316 [DEBUG] switch_channel.c:1615 (dingaling/ >> GVOICE/+15555555... at voice.google.com) Callstate Change ACTIVE -> >> RINGING >> 2012-04-29 12:50:05.195316 [DEBUG] switch_core_state_machine.c:345 >> (dingaling/GVOICE/+15555555... at voice.google.com) State ROUTING >> 2012-04-29 12:50:05.195316 [DEBUG] mod_dingaling.c:1213 dingaling/ >> GVOICE/+15555555... at voice.google.com CHANNEL ROUTING >> 2012-04-29 12:50:05.195316 [DEBUG] switch_ivr_originate.c:66 >> (dingaling/GVOICE/+15555555... at voice.google.com) State Change >> CS_ROUTING -> CS_CONSUME_MEDIA >> 2012-04-29 12:50:05.195316 [DEBUG] switch_core_session.c:1057 Send >> signal dingaling/GVOICE/+15555555... at voice.google.com [BREAK] >> 2012-04-29 12:50:05.195316 [DEBUG] mod_dingaling.c:1342 dingaling/ >> GVOICE/+15555555... at voice.google.com CHANNEL KILL >> 2012-04-29 12:50:05.195316 [DEBUG] switch_core_state_machine.c:345 >> (dingaling/GVOICE/+15555555... at voice.google.com) State ROUTING going >> to sleep >> 2012-04-29 12:50:05.195316 [DEBUG] switch_core_state_machine.c:318 >> (dingaling/GVOICE/+15555555... at voice.google.com) Running State Change >> CS_CONSUME_MEDIA >> 2012-04-29 12:50:05.195316 [DEBUG] switch_channel.c:1617 (dingaling/ >> GVOICE/+15555555... at voice.google.com) Callstate Change RINGING -> >> ACTIVE >> 2012-04-29 12:50:05.195316 [DEBUG] switch_core_state_machine.c:364 >> (dingaling/GVOICE/+15555555... at voice.google.com) State CONSUME_MEDIA >> 2012-04-29 12:50:05.195316 [DEBUG] switch_core_state_machine.c:364 >> (dingaling/GVOICE/+15555555... at voice.google.com) State CONSUME_MEDIA >> going to sleep >> 2012-04-29 12:50:05.198143 [NOTICE] mod_sofia.c:2096 Ring-Ready sofia/ >> sipinterface_1/yu... at 192.168.0.112! >> 2012-04-29 12:50:05.198143 [DEBUG] switch_core_session.c:676 Send >> signal sofia/sipinterface_1/yu... at 192.168.0.112 [BREAK] >> 2012-04-29 12:50:05.198143 [NOTICE] switch_ivr_originate.c:472 Ring >> Ready sofia/sipinterface_1/yu... at 192.168.0.112! >> 2012-04-29 12:50:05.198143 [DEBUG] sofia.c:4559 Channel sofia/ >> sipinterface_1/yu... at 192.168.0.112 entering state [early][180] >> 2012-04-29 12:50:05.225311 [DEBUG] sofia_glue.c:2961 AUDIO RTP [sofia/ >> sipinterface_1/yu... at 192.168.0.112] 192.168.0.112 port 19702 -> >> 192.168.0.101 port 16416 codec: 0 ms: 30 >> 2012-04-29 12:50:05.225311 [DEBUG] switch_rtp.c:1418 Starting timer >> [soft] 240 bytes per 30ms >> 2012-04-29 12:50:05.227308 [DEBUG] sofia_glue.c:3179 Set 2833 dtmf >> send payload to 101 >> 2012-04-29 12:50:05.227308 [DEBUG] sofia_glue.c:3184 Set 2833 dtmf >> receive payload to 101 >> 2012-04-29 12:50:05.227308 [DEBUG] mod_sofia.c:680 Local SDP sofia/ >> sipinterface_1/yu... at 192.168.0.112: >> v=0 >> o=FreeSWITCH 1335709303 1335709304 IN IP4 192.168.0.112 >> s=FreeSWITCH >> c=IN IP4 192.168.0.112 >> t=0 0 >> m=audio 19702 RTP/AVP 0 101 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=silenceSupp:off - - - - >> a=ptime:30 >> a=sendrecv >> >> 2012-04-29 12:50:05.227308 [DEBUG] switch_core_session.c:676 Send >> signal sofia/sipinterface_1/yu... at 192.168.0.112 [BREAK] >> 2012-04-29 12:50:05.227308 [DEBUG] switch_channel.c:2694 (sofia/ >> sipinterface_1/yu... at 192.168.0.112) Callstate Change RINGING -> ACTIVE >> 2012-04-29 12:50:05.227308 [NOTICE] switch_ivr_originate.c:3288 >> Channel [sofia/sipinterface_1/yu... at 192.168.0.112] has been answered >> 2012-04-29 12:50:05.227308 [DEBUG] switch_ivr_originate.c:3333 >> Originate Resulted in Success: [dingaling/GVOICE/ >> +15555555... at voice.google.com] >> 2012-04-29 12:50:05.227308 [DEBUG] switch_core_session.c:676 Send >> signal dingaling/GVOICE/+15555555... at voice.google.com [BREAK] >> 2012-04-29 12:50:05.227308 [DEBUG] mod_dingaling.c:1342 dingaling/ >> GVOICE/+15555555... at voice.google.com CHANNEL KILL >> 2012-04-29 12:50:05.227308 [DEBUG] switch_core_session.c:676 Send >> signal sofia/sipinterface_1/yu... at 192.168.0.112 [BREAK] >> 2012-04-29 12:50:05.227308 [DEBUG] switch_ivr_bridge.c:1226 (dingaling/ >> GVOICE/+15555555... at voice.google.com) State Change CS_CONSUME_MEDIA -> >> CS_EXCHANGE_MEDIA >> 2012-04-29 12:50:05.227308 [DEBUG] switch_core_session.c:1057 Send >> signal dingaling/GVOICE/+15555555... at voice.google.com [BREAK] >> 2012-04-29 12:50:05.227308 [DEBUG] mod_dingaling.c:1342 dingaling/ >> GVOICE/+15555555... at voice.google.com CHANNEL KILL >> 2012-04-29 12:50:05.230264 [DEBUG] switch_core_state_machine.c:318 >> (dingaling/GVOICE/+15555555... at voice.google.com) Running State Change >> CS_EXCHANGE_MEDIA >> 2012-04-29 12:50:05.230264 [DEBUG] switch_core_state_machine.c:355 >> (dingaling/GVOICE/+15555555... at voice.google.com) State EXCHANGE_MEDIA >> 2012-04-29 12:50:05.230264 [DEBUG] mod_dingaling.c:1350 CHANNEL >> LOOPBACK >> 2012-04-29 12:50:05.230264 [DEBUG] sofia.c:4559 Channel sofia/ >> sipinterface_1/yu... at 192.168.0.112 entering state [completed][200] >> 2012-04-29 12:50:05.242302 [DEBUG] sofia.c:4559 Channel sofia/ >> sipinterface_1/yu... at 192.168.0.112 entering state [ready][200] >> 2012-04-29 12:50:05.244336 [DEBUG] switch_rtp.c:2544 Correct ip/port >> confirmed. >> 2012-04-29 12:50:05.258274 [DEBUG] switch_core_io.c:970 Engaging Write >> Buffer at 480 bytes to accommodate 320->480 >> 2012-04-29 12:50:05.267313 [DEBUG] switch_core_session.c:738 Send >> signal dingaling/GVOICE/+15555555... at voice.google.com [BREAK] >> 2012-04-29 12:50:05.267313 [DEBUG] mod_dingaling.c:1342 dingaling/ >> GVOICE/+15555555... at voice.google.com CHANNEL KILL >> 2012-04-29 12:50:05.267313 [DEBUG] switch_core_session.c:738 Send >> signal sofia/sipinterface_1/yu... at 192.168.0.112 [BREAK] >> 2012-04-29 12:50:05.358039 [DEBUG] switch_rtp.c:2544 Correct ip/port >> confirmed. >> 2012-04-29 12:50:05.358039 [DEBUG] switch_core_io.c:970 Engaging Write >> Buffer at 320 bytes to accommodate 480->320 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120430/c77b1f9f/attachment-0001.html From mitch.capper at gmail.com Tue May 1 06:53:18 2012 From: mitch.capper at gmail.com (Mitch Capper) Date: Mon, 30 Apr 2012 19:53:18 -0700 Subject: [Freeswitch-users] How to update a live system with very little downtime (How does 'make current' work?) In-Reply-To: <4F9EF310.8000706@quentustech.com> References: <361E98F99D3CC3439EED59BC1924ED695ECFC2@SERVER2003.SecuReachSystems.local> <4F9EF310.8000706@quentustech.com> Message-ID: Personally I think all of the in place suggestions are suicide if you are looking for minimum downtime. Collins' suggestion is best given the lack of a second server. Install FS to /usr/local/freeswitch-dev/ and test there. You should be able to simply change your FS ports and test against the dev version to make sure everything is working. FreeSWITCH config changes and sometimes upgrading your config to support new settings can be a pain. Inplace requires downtime vs a second dev copy on the same server should be able to nearly eliminate it. Testing allows your normal FS to go fine, change configs and once dev is working just install to your primary install folder and copy the config over. I have been running FS for along time and there have been a few times the upgrade has not quite just instantly worked, having a nice test side to work on things on is a lot easier than having to figure out what broke while people are waiting. ~mitch From avi at avimarcus.net Tue May 1 10:24:18 2012 From: avi at avimarcus.net (Avi Marcus) Date: Tue, 1 May 2012 09:24:18 +0300 Subject: [Freeswitch-users] Syncing Voicemail Files Between Servers Message-ID: Any suggestions for sync'ing the voicemail file between multiple backend servers? I have only 2 servers - one is a hot-backup - so there's no dedicated storage device. I couldn't find an event for voicemail creation, and would similarly assume there isn't one on deletion. Did I just miss the events? I could monitor the voicemail table and filesystem but that seems pretty hacky considering the event system provides direct notification to most information. (Alternatively, a mounted master-master file-system to save the VM to. But these kind of things aren't simple...) Thanks, -Avi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120501/3571c8c7/attachment.html From gabe at gundy.org Tue May 1 10:28:04 2012 From: gabe at gundy.org (Gabriel Gunderson) Date: Tue, 1 May 2012 00:28:04 -0600 Subject: [Freeswitch-users] Syncing Voicemail Files Between Servers In-Reply-To: References: Message-ID: On Tue, May 1, 2012 at 12:24 AM, Avi Marcus wrote: > (Alternatively, a mounted master-master file-system to save the VM to. But > these kind of things aren't simple...) Why not just use a separate NFS server that they each pull from? Gabe From yehavi.bourvine at gmail.com Tue May 1 10:46:05 2012 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Tue, 1 May 2012 09:46:05 +0300 Subject: [Freeswitch-users] Syncing Voicemail Files Between Servers In-Reply-To: References: Message-ID: Hi, We use here an NFS server connected to both machines storing the voicemail files. We use MySQL for the database and use replication to keep both in sync. Regards, __Yehavi: 2012/5/1 Avi Marcus > Any suggestions for sync'ing the voicemail file between multiple backend > servers? I have only 2 servers - one is a hot-backup - so there's no > dedicated storage device. > > I couldn't find an event for voicemail creation, and would similarly > assume there isn't one on deletion. Did I just miss the events? > > I could monitor the voicemail table and filesystem but that seems pretty > hacky considering the event system provides direct notification to most > information. > > (Alternatively, a mounted master-master file-system to save the VM to. But > these kind of things aren't simple...) > > Thanks, > -Avi > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120501/21abf09d/attachment.html From webudo at gua.net Tue May 1 05:12:53 2012 From: webudo at gua.net (=?ISO-8859-1?Q?Alejandro_Mej=EDa?=) Date: Mon, 30 Apr 2012 19:12:53 -0600 Subject: [Freeswitch-users] Silence instead of spanish wav installed Message-ID: <4F9F3895.7090503@gua.net> Hello, I installed a spanish sounds package to my Freeswitch under /usr/local/freeswitch/sounds/es/mx/ directory I'm trying it by entering an empty conference, and I only hear some seconds of silence (where conf-alone.wav should be played), and then the music on hold. I know the files are not missing, and Freeswitch is finding them because when I remove "/es" directory, I get error on CLI when executing a conference, and getting directly to MOH. Any help will be appreciated. Thanks! Alex From b2m at a-cti.com Tue May 1 13:45:16 2012 From: b2m at a-cti.com (Balamurugan Mahendran) Date: Tue, 1 May 2012 15:15:16 +0530 Subject: [Freeswitch-users] Syncing Voicemail Files Between Servers In-Reply-To: References: Message-ID: Amazon S3 is very simple and can be mounted as file system on any number of server, also very cheap. Thanks, Bala On Tue, May 1, 2012 at 12:16 PM, Yehavi Bourvine wrote: > Hi, > > We use here an NFS server connected to both machines storing the > voicemail files. We use MySQL for the database and use replication to keep > both in sync. > > Regards, __Yehavi: > > 2012/5/1 Avi Marcus > >> Any suggestions for sync'ing the voicemail file between multiple backend >> servers? I have only 2 servers - one is a hot-backup - so there's no >> dedicated storage device. >> >> I couldn't find an event for voicemail creation, and would similarly >> assume there isn't one on deletion. Did I just miss the events? >> >> I could monitor the voicemail table and filesystem but that seems pretty >> hacky considering the event system provides direct notification to most >> information. >> >> (Alternatively, a mounted master-master file-system to save the VM to. >> But these kind of things aren't simple...) >> >> Thanks, >> -Avi >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120501/bbe64d68/attachment-0001.html From anita.hall at simmortel.com Tue May 1 14:01:55 2012 From: anita.hall at simmortel.com (Anita Hall) Date: Tue, 1 May 2012 15:31:55 +0530 Subject: [Freeswitch-users] spandsp consultant T.30 Message-ID: Hi One of our clients is getting around 70% results using mod_spandsp on FreeSWITCH. The protocol is T.30 and the client is India. They are looking for someone who could help them to increase the results to more than 95%. If you or someone you know could help, please let me know. Thanks! regards, Anita -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120501/e1f0b075/attachment.html From brian at freeswitch.org Tue May 1 14:35:10 2012 From: brian at freeswitch.org (Brian West) Date: Tue, 1 May 2012 05:35:10 -0500 Subject: [Freeswitch-users] spandsp consultant T.30 In-Reply-To: References: Message-ID: <-2375528179531511362@unknownmsgid> Are you doing t.38? Sent from my iPhone On May 1, 2012, at 5:03 AM, Anita Hall wrote: > Hi > > One of our clients is getting around 70% results using mod_spandsp on FreeSWITCH. The protocol is T.30 and the client is India. They are looking for someone who could help them to increase the results to more than 95%. > > If you or someone you know could help, please let me know. > > Thanks! > > regards, > Anita > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From markus.lindenberg at gmail.com Tue May 1 15:31:19 2012 From: markus.lindenberg at gmail.com (Markus Lindenberg) Date: Tue, 1 May 2012 13:31:19 +0200 Subject: [Freeswitch-users] mod_voicemail and non-numeric id In-Reply-To: References: Message-ID: It's in the wiki: Just set voicemail_alternate_greet_id before bridging to voicemail. I'm using non-numeric IDs as well, so i set connectedline manually before trying to bridge to the user: When the call hits voicemail, i set voicemail_alternate_greet_id and then that's what voicemail will use: On Fri, Apr 27, 2012 at 09:51, Anton Kvashenkin wrote: > Hello list. > > I've decided to use non-numeric id's at users configuration. > > > > But when the call goes to voicemail app > > > > switch_ivr_play_say.c:244 Handle say:[001565229c95] (ru:ru) > 2012-04-27 11:39:54.452208 [INFO] mod_say_ru.c:721 ru_say!!! ?001565229c95! > ? say_opt.gender=0 ? say_opt.cases=0 > 2012-04-27 11:39:54.532157 [DEBUG] switch_ivr_play_say.c:1309 Codec > Activated L16 at 16000hz 1 channels 20ms > > The mod_say pronounces user id. How to force to pronounce number-alias? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From barnyritchley at hotmail.com Tue May 1 15:58:09 2012 From: barnyritchley at hotmail.com (Barnaby Ritchley) Date: Tue, 1 May 2012 12:58:09 +0100 Subject: [Freeswitch-users] Freeswitch and RFC4028 In-Reply-To: References: <4F93AD01.7010707@quentustech.com> Message-ID: Hi All We have a situation where FS seems not to behaving correctly with session timers. Call originates from remote party with invite as per: INVITE sip:+1234 at 1.2.3.4;user=phone SIP/2.0 Max-Forwards: 69 Session-Expires: 3600;refresher=uac Min-SE: 600 Supported: timer, 100rel To: +1234 From: ;tag=381443 P-Asserted-Identity: Call-ID: 1333524918-9038500 at xxxxx CSeq: 1 INVITE Allow: CANCEL, ACK, INVITE, BYE, OPTIONS, INFO, PRACK, UPDATE Via: SIP/2.0/UDP 7.8.9.1:5060;branch=z9hG4bKab17954781a678af6dfd6916593dc954 Contact: Expires: 330 Content-Type: application/sdp Accept: application/sdp Content-Length: 387 Here you can see the UAC supporting timer, and specifying a session expires of 3600. FS responds with 100 trying (not relevant) then 180 runing (also not relevant) then 200OK as per: SIP/2.0 200 OK From: ;tag=381443 To: +1234 ;tag=UrcX58SSF2Nrm Call-ID: 1333524918-9038500 at xxxxx CSeq: 1 INVITE Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Session-Expires: 3600;refresher=uac Min-SE: 600 Content-Type: application/sdp Content-Disposition: session Content-Length: 249 So, here freeswitch is responding with the session-expires and refresher UAC. Then after 60 mins, the call is dropped as FS sends a Bye to the UAC because it does not receive a session refresh. Here is the Bye: BYE sip:+5678 at 5.6.7.8:5060 SIP/2.0 Via: SIP/2.0/UDP 178.255.58.28;rport;branch=z9hG4bK2vm55Zp2D6vHm Max-Forwards: 70 From: +1234 ;tag=UrcX58SSF2Nrm To: ;tag=381443 Call-ID: 1333524918-9038500 at xxxxx CSeq: 27563136 BYE Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, NOTIFY Supported: timer, precondition, path, replaces Reason: SIP;cause=408;text="Session timeout" Content-Length: 0 When we raised this with the UAC, their response is that the reason they are not refreshing the session is because FS should be sending a Require: timer; header in the response, so their UAC thinks that session timers are not in use. This is documented in para 9 of RFC4028: If the refresher parameter in the Session-Expires header field in the 2xx response has a value of 'uac', the UAS MUST place a Require header field into the response with the value 'timer'. This is because the uac is performing refreshes and the response has to be processed for the UAC to know this. It appears the FS is not doing this, but is hanging up the call anyway, so we are getting dropped calls that are around the 60 mins mark. Whats everyones view on this? It would seem logical that if FS is not sending requires in the return 200, that the UAC will think that the session timers are not being used in the session, and therefore will not refresh it. Brgds -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120501/691603ca/attachment.html From vipkilla at gmail.com Tue May 1 17:31:49 2012 From: vipkilla at gmail.com (Vik Killa) Date: Tue, 1 May 2012 09:31:49 -0400 Subject: [Freeswitch-users] VoIP job available in Western New York. Message-ID: Our company is looking to hire a VoIP person with knowledge of FreeSWITCH, OpenSIPS, MySQL, PHP, Perl and of course Linux. Is there any FreeSWITCHers in the Western New York area looking for a job? From steveayre at gmail.com Tue May 1 18:31:35 2012 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 1 May 2012 15:31:35 +0100 Subject: [Freeswitch-users] Syncing Voicemail Files Between Servers In-Reply-To: References: Message-ID: Running DRBD would also be an option which remove the point of failure if the NFS server goes down. -Steve On 1 May 2012 07:46, Yehavi Bourvine wrote: > Hi, > > ? We use here an NFS server connected to both machines storing the voicemail > files. We use MySQL for the database and use replication to keep both in > sync. > > ??????????????????????? Regards, __Yehavi: > > 2012/5/1 Avi Marcus >> >> Any suggestions for sync'ing the voicemail file between multiple backend >> servers? I have only 2 servers - one is a hot-backup - so there's no >> dedicated storage device. >> >> I couldn't find an event for voicemail creation, and would similarly >> assume there isn't one on deletion.?Did I just miss the events? >> >> I could monitor the voicemail table and filesystem but that seems pretty >> hacky considering the event system provides direct notification to most >> information. >> >> (Alternatively, a mounted master-master file-system to save the VM to. But >> these kind of things aren't simple...) >> >> Thanks, >> -Avi >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From steveayre at gmail.com Tue May 1 18:36:36 2012 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 1 May 2012 15:36:36 +0100 Subject: [Freeswitch-users] Freeswitch and RFC4028 In-Reply-To: References: <4F93AD01.7010707@quentustech.com> Message-ID: Please report bugs to JIRA. The mailing list is not the place for reporting bugs. http://jira.freeswitch.org/ Reporting it on Jira let's us easily have any conversation about it there and properly documented in a single place. -Steve On 1 May 2012 12:58, Barnaby Ritchley wrote: > Hi All > > We have a situation where FS seems not to behaving correctly with session > timers. > > Call originates from remote party with invite as per: > > INVITE?sip:+1234 at 1.2.3.4;user=phone SIP/2.0 > Max-Forwards: 69 > Session-Expires: 3600;refresher=uac > Min-SE: 600 > Supported: timer, 100rel > To: +1234 > From: ;tag=381443 > P-Asserted-Identity: > Call-ID:?1333524918-9038500 at xxxxx > CSeq: 1?INVITE > Allow: CANCEL, ACK,?INVITE, BYE, OPTIONS, INFO, PRACK, UPDATE > Via: SIP/2.0/UDP > 7.8.9.1:5060;branch=z9hG4bKab17954781a678af6dfd6916593dc954 > Contact: > Expires: 330 > Content-Type: application/sdp > Accept: application/sdp > Content-Length: 387 > > > Here you can see the UAC supporting timer, and specifying a session expires > of 3600. > > FS responds with 100 trying (not relevant) then 180 runing (also not > relevant) then 200OK as per: > > SIP/2.0?200?OK > From: ;tag=381443 > To:?+1234 ;tag=UrcX58SSF2Nrm > Call-ID:?1333524918-9038500 at xxxxx > CSeq: 1 INVITE > Contact: > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, NOTIFY > Supported: timer, precondition, path, replaces > Allow-Events: talk, hold, refer > Session-Expires: 3600;refresher=uac > Min-SE: 600 > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 249 > > > So, here freeswitch is responding with the session-expires and refresher > UAC. ?Then after 60 mins, the call is dropped as FS sends a Bye to the UAC > because it does not receive a session refresh. ?Here is the Bye: > > BYE?sip:+5678 at 5.6.7.8:5060 SIP/2.0 > Via: SIP/2.0/UDP 178.255.58.28;rport;branch=z9hG4bK2vm55Zp2D6vHm > Max-Forwards: 70 > From: +1234 ;tag=UrcX58SSF2Nrm > To: ;tag=381443 > Call-ID:?1333524918-9038500 at xxxxx > CSeq: 27563136?BYE > Contact: > Allow: INVITE, ACK,?BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, NOTIFY > Supported: timer, precondition, path, replaces > Reason: SIP;cause=408;text="Session timeout" > Content-Length: 0 > > When we raised this with the UAC, their response is that the reason they are > not refreshing the session is because FS should be sending a Require: timer; > header in the response, so their UAC thinks that session timers are not in > use. > > This is documented in para 9 of RFC4028: > > If the refresher parameter in the Session-Expires header field in the > 2xx response has a value of 'uac', the UAS MUST place a Require > header field into the response with the value 'timer'. This is > because the uac is performing refreshes and the response has to be > processed for the UAC to know this. > > > It appears the FS is not doing this, but is hanging up the call anyway, so > we are getting dropped calls that are around the 60 mins mark. > > Whats everyones view on this? ?It would seem logical that if FS is not > sending requires in the return 200, that the UAC will think that the session > timers are not being used in the session, and therefore will not refresh it. > > > Brgds > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From vipkilla at gmail.com Tue May 1 18:39:12 2012 From: vipkilla at gmail.com (Vik Killa) Date: Tue, 1 May 2012 10:39:12 -0400 Subject: [Freeswitch-users] Syncing Voicemail Files Between Servers In-Reply-To: References: Message-ID: Anybody use glusterFS? It looks cool and easy to setup. On Tue, May 1, 2012 at 10:31 AM, Steven Ayre wrote: > Running DRBD would also be an option which remove the point of failure > if the NFS server goes down. > > -Steve > > > > > On 1 May 2012 07:46, Yehavi Bourvine wrote: >> Hi, >> >> ? We use here an NFS server connected to both machines storing the voicemail >> files. We use MySQL for the database and use replication to keep both in >> sync. >> >> ??????????????????????? Regards, __Yehavi: >> >> 2012/5/1 Avi Marcus >>> >>> Any suggestions for sync'ing the voicemail file between multiple backend >>> servers? I have only 2 servers - one is a hot-backup - so there's no >>> dedicated storage device. >>> >>> I couldn't find an event for voicemail creation, and would similarly >>> assume there isn't one on deletion.?Did I just miss the events? >>> >>> I could monitor the voicemail table and filesystem but that seems pretty >>> hacky considering the event system provides direct notification to most >>> information. >>> >>> (Alternatively, a mounted master-master file-system to save the VM to. But >>> these kind of things aren't simple...) >>> >>> Thanks, >>> -Avi >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From boris at tagnet.ru Tue May 1 20:58:06 2012 From: boris at tagnet.ru (Boris Kovalenko) Date: Tue, 01 May 2012 22:58:06 +0600 Subject: [Freeswitch-users] bind_digit_action and dtmf target leg Message-ID: <4FA0161E.8040605@tagnet.ru> Hello! What is dtmf target leg used for? Found it in docs but without explanation? -- Regards, Boris From djbinter at gmail.com Tue May 1 21:01:08 2012 From: djbinter at gmail.com (DJB International) Date: Tue, 1 May 2012 10:01:08 -0700 Subject: [Freeswitch-users] Freeswitch and RFC4028 In-Reply-To: References: <4F93AD01.7010707@quentustech.com> Message-ID: I believe your remote gateway INVITE that include refresher=uac in the INVITE was incorrect. From what I understand, the refresher should be in 200 OK message. Here is a link that you can clearly see SIP session timer call flows in different scenerio: http://www.dialogic.com/webhelp/IMG1010/10.5.3/WebHelp/Description/SIP/SIP_Session_Timer_Call_Flows.htm As per Steve, if you think it's a bug, please open a ticket in Jira. -djbinter On Tue, May 1, 2012 at 7:36 AM, Steven Ayre wrote: > Please report bugs to JIRA. The mailing list is not the place for > reporting bugs. > > http://jira.freeswitch.org/ > > Reporting it on Jira let's us easily have any conversation about it > there and properly documented in a single place. > > -Steve > > > > On 1 May 2012 12:58, Barnaby Ritchley wrote: > > Hi All > > > > We have a situation where FS seems not to behaving correctly with session > > timers. > > > > Call originates from remote party with invite as per: > > > > INVITE sip:+1234 at 1.2.3.4;user=phone SIP/2.0 > > Max-Forwards: 69 > > Session-Expires: 3600;refresher=uac > > Min-SE: 600 > > Supported: timer, 100rel > > To: +1234 > > From: ;tag=381443 > > P-Asserted-Identity: > > Call-ID: 1333524918-9038500 at xxxxx > > CSeq: 1 INVITE > > Allow: CANCEL, ACK, INVITE, BYE, OPTIONS, INFO, PRACK, UPDATE > > Via: SIP/2.0/UDP > > 7.8.9.1:5060;branch=z9hG4bKab17954781a678af6dfd6916593dc954 > > Contact: > > Expires: 330 > > Content-Type: application/sdp > > Accept: application/sdp > > Content-Length: 387 > > > > > > Here you can see the UAC supporting timer, and specifying a session > expires > > of 3600. > > > > FS responds with 100 trying (not relevant) then 180 runing (also not > > relevant) then 200OK as per: > > > > SIP/2.0 200 OK > > From: ;tag=381443 > > To: +1234 ;tag=UrcX58SSF2Nrm > > Call-ID: 1333524918-9038500 at xxxxx > > CSeq: 1 INVITE > > Contact: > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, NOTIFY > > Supported: timer, precondition, path, replaces > > Allow-Events: talk, hold, refer > > Session-Expires: 3600;refresher=uac > > Min-SE: 600 > > Content-Type: application/sdp > > Content-Disposition: session > > Content-Length: 249 > > > > > > So, here freeswitch is responding with the session-expires and refresher > > UAC. Then after 60 mins, the call is dropped as FS sends a Bye to the > UAC > > because it does not receive a session refresh. Here is the Bye: > > > > BYE sip:+5678 at 5.6.7.8:5060 SIP/2.0 > > Via: SIP/2.0/UDP 178.255.58.28;rport;branch=z9hG4bK2vm55Zp2D6vHm > > Max-Forwards: 70 > > From: +1234 ;tag=UrcX58SSF2Nrm > > To: ;tag=381443 > > Call-ID: 1333524918-9038500 at xxxxx > > CSeq: 27563136 BYE > > Contact: > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, NOTIFY > > Supported: timer, precondition, path, replaces > > Reason: SIP;cause=408;text="Session timeout" > > Content-Length: 0 > > > > When we raised this with the UAC, their response is that the reason they > are > > not refreshing the session is because FS should be sending a Require: > timer; > > header in the response, so their UAC thinks that session timers are not > in > > use. > > > > This is documented in para 9 of RFC4028: > > > > If the refresher parameter in the Session-Expires header field in the > > 2xx response has a value of 'uac', the UAS MUST place a Require > > header field into the response with the value 'timer'. This is > > because the uac is performing refreshes and the response has to be > > processed for the UAC to know this. > > > > > > It appears the FS is not doing this, but is hanging up the call anyway, > so > > we are getting dropped calls that are around the 60 mins mark. > > > > Whats everyones view on this? It would seem logical that if FS is not > > sending requires in the return 200, that the UAC will think that the > session > > timers are not being used in the session, and therefore will not refresh > it. > > > > > > Brgds > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120501/f0be26d7/attachment-0001.html From jmesquita at freeswitch.org Tue May 1 21:10:03 2012 From: jmesquita at freeswitch.org (=?utf-8?Q?Jo=C3=A3o_Mesquita?=) Date: Tue, 1 May 2012 14:10:03 -0300 Subject: [Freeswitch-users] bind_digit_action and dtmf target leg In-Reply-To: <4FA0161E.8040605@tagnet.ru> References: <4FA0161E.8040605@tagnet.ru> Message-ID: <8E932BDEC1D841D6A3D4909B7548DC44@freeswitch.org> I think it is the leg in which you want the DTMFs to be listened in. Hope that makes sense. Regards, -- Jo?o Mesquita Sent with Sparrow (http://www.sparrowmailapp.com/?sig) On Tuesday, May 1, 2012 at 1:58 PM, Boris Kovalenko wrote: > Hello! > > What is dtmf target leg used for? Found it in docs but without > explanation? > > -- > Regards, > Boris > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120501/5a5d42df/attachment.html From sat at calgaryit.com Tue May 1 21:11:44 2012 From: sat at calgaryit.com (George Sapak) Date: Tue, 1 May 2012 11:11:44 -0600 (MDT) Subject: [Freeswitch-users] Voicemail hangups 38 sec into recording In-Reply-To: <798770331.209417.1335892161279.JavaMail.root@server3> Message-ID: <1586891651.209426.1335892304229.JavaMail.root@server3> Still having issue with hangup: FreeSWITCH Version 1.1.beta1 (git-5e5a2ff 2012-04-23 07-50-57 -0500) I have set the waste resources: Thank You, George. From hawkeye06 at gmail.com Tue May 1 18:14:18 2012 From: hawkeye06 at gmail.com (hawkeye06) Date: Tue, 1 May 2012 09:14:18 -0500 Subject: [Freeswitch-users] freeswitch with google voice Message-ID: Hi I am trying to setup freeswitch with google voice. I don't really understand how to tell if it is working properly when viewing the debug messages in freeswitch console (dl_debug on) Here is what I am seeing for debug messages: http://pastebin.freeswitch.org/18963 It doesn't appear that there are any error messages. However, shouldn't I be getting further detail when I call the google voice number? Here is the log for freeswitch starting up (mod dingaling appears to successfully load up as well.) http://pastebin.freeswitch.org/18964 Thank you! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120501/bb968f8a/attachment.html From chris at gonumina.com Tue May 1 18:50:22 2012 From: chris at gonumina.com (Chris Ferreira) Date: Tue, 1 May 2012 10:50:22 -0400 Subject: [Freeswitch-users] VoIP job available in Western New York. In-Reply-To: References: Message-ID: <-8812328422832950994@unknownmsgid> I'd be interested to hear about this as well, but more for the NY/NJ Metro Area. Thanks, Chris ___________________ Mobile Reply On May 1, 2012, at 9:36 AM, Vik Killa wrote: > Our company is looking to hire a VoIP person with knowledge of > FreeSWITCH, OpenSIPS, MySQL, PHP, Perl and of course Linux. Is there > any FreeSWITCHers in the Western New York area looking for a job? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From msc at freeswitch.org Tue May 1 21:46:40 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 1 May 2012 10:46:40 -0700 Subject: [Freeswitch-users] VoIP job available in Western New York. In-Reply-To: References: Message-ID: On Tue, May 1, 2012 at 6:31 AM, Vik Killa wrote: > Our company is looking to hire a VoIP person with knowledge of > FreeSWITCH, OpenSIPS, MySQL, PHP, Perl and of course Linux. Is there > any FreeSWITCHers in the Western New York area looking for a job? > It's good to hear that enterprises are hiring people with FreeSWITCH skills! Thanks for letting the community at large know. For those of you wanting to send these kinds of messages to the list I'd like to make a few suggestions: - Try the freeswitch-biz list first since it is geared for this sort of thing. Only if there's no response should you post here. - Be sure to put "OT:" at the beginning of the subject line. It helps us keep things organized. :) - Ask the respondents to contact you off-list and be sure to give any necessary additional information (like a different email address, web address, phone number, etc.) - For those responding, be sure to respond off-list Following these suggestions will help keep the list clean and will prevent any inadvertent posting of confidential information. Thanks! -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120501/232407c9/attachment.html From anton.jugatsu at gmail.com Tue May 1 21:47:26 2012 From: anton.jugatsu at gmail.com (Anton Kvashenkin) Date: Tue, 1 May 2012 21:47:26 +0400 Subject: [Freeswitch-users] Voicemail hangups 38 sec into recording In-Reply-To: <1586891651.209426.1335892304229.JavaMail.root@server3> References: <798770331.209417.1335892161279.JavaMail.root@server3> <1586891651.209426.1335892304229.JavaMail.root@server3> Message-ID: It doesn't make any sense. At all. Try to provide more logs with a help of fs_logger.pl (search wiki). 2012/5/1 George Sapak > Still having issue with hangup: > > FreeSWITCH Version 1.1.beta1 (git-5e5a2ff 2012-04-23 07-50-57 -0500) > > I have set the waste resources: > > data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/> > > > > > > > Thank You, George. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120501/6037c255/attachment.html From msc at freeswitch.org Tue May 1 21:54:36 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 1 May 2012 10:54:36 -0700 Subject: [Freeswitch-users] Silence instead of spanish wav installed In-Reply-To: <4F9F3895.7090503@gua.net> References: <4F9F3895.7090503@gua.net> Message-ID: Can you manually play any of the sound files? Also, what sampling rate is the conference and what sampling rate are the files? -MC On Mon, Apr 30, 2012 at 6:12 PM, Alejandro Mej?a wrote: > Hello, > > I installed a spanish sounds package to my Freeswitch under > /usr/local/freeswitch/sounds/es/mx/ directory > I'm trying it by entering an empty conference, and I only hear some > seconds of silence (where conf-alone.wav should be played), and then the > music on hold. > I know the files are not missing, and Freeswitch is finding them because > when I remove "/es" directory, I get error on CLI when executing a > conference, and getting directly to MOH. > > Any help will be appreciated. > > Thanks! > > Alex > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120501/97a70db5/attachment-0001.html From msc at freeswitch.org Tue May 1 21:52:47 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 1 May 2012 10:52:47 -0700 Subject: [Freeswitch-users] bind_digit_action and dtmf target leg In-Reply-To: <8E932BDEC1D841D6A3D4909B7548DC44@freeswitch.org> References: <4FA0161E.8040605@tagnet.ru> <8E932BDEC1D841D6A3D4909B7548DC44@freeswitch.org> Message-ID: On Tue, May 1, 2012 at 10:10 AM, Jo?o Mesquita wrote: > I think it is the leg in which you want the DTMFs to be listened in. > > Correct. You have "dtmf target leg" and "event target leg" - these determine which leg on which to listen for DTMFs and on which leg the action should take place, respectively. (Note that these are optional.) In other words the DTMFs received on one leg could trigger an action on the other leg. You could even have an action triggered on both legs. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120501/714c5a21/attachment.html From boris at tagnet.ru Tue May 1 22:04:07 2012 From: boris at tagnet.ru (Boris Kovalenko) Date: Wed, 02 May 2012 00:04:07 +0600 Subject: [Freeswitch-users] bind_digit_action - another question Message-ID: <4FA02597.1090506@tagnet.ru> Hello! The simple setup: The call is made from number 102 to 101. No matter on what side I enter *1 - the ext_a is called. Now I change 'both' to 'peer'. The call is the same - from 102 to 101. And now ext_a is called only if I enter it on 101 side! I'm confused - docs describing another logic? Or I need another params for dtmf_targer_leg and event_target_leg? -- Regards, Boris From msc at freeswitch.org Tue May 1 22:08:06 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 1 May 2012 11:08:06 -0700 Subject: [Freeswitch-users] bind_digit_action - another question In-Reply-To: <4FA02597.1090506@tagnet.ru> References: <4FA02597.1090506@tagnet.ru> Message-ID: For kicks, change 'peer' to 'self' and retest. Is the behavior the same or no? It may be that you have your legs flip-flopped some how. -MC On Tue, May 1, 2012 at 11:04 AM, Boris Kovalenko wrote: > Hello! > > The simple setup: > > > > > data="rlm_a,*1,exec:execute_extension,ext_a XML test.tagnet.cc > ,both,self"/> > > > > > > > > > > > The call is made from number 102 to 101. No matter on what side I enter > *1 - the ext_a is called. > Now I change 'both' to 'peer'. The call is the same - from 102 to 101. > And now ext_a is called only if I enter it on 101 side! I'm confused - > docs describing another logic? Or I need another params for > dtmf_targer_leg and event_target_leg? > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120501/f483605f/attachment.html From msc at freeswitch.org Tue May 1 22:13:59 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 1 May 2012 11:13:59 -0700 Subject: [Freeswitch-users] Voicemail hangups 38 sec into recording In-Reply-To: <1586891651.209426.1335892304229.JavaMail.root@server3> References: <798770331.209417.1335892161279.JavaMail.root@server3> <1586891651.209426.1335892304229.JavaMail.root@server3> Message-ID: You may want to capture the who call with media as well, just in case. I recommend one of the packet capture tools listed here: http://wiki.freeswitch.org/wiki/Packet_Capture Use wireshark to analyze the call. It will show you the whole call broken down into a nice call flow graph and you can see if one side or the other sent a BYE. Usually there's a reason listed in the BYE as to why the client is hanging up. If not you can go back to the console log and look for clues there. If you need help then go ahead and put your logs into a pastebin and post the URL in this thread. -MC On Tue, May 1, 2012 at 10:11 AM, George Sapak wrote: > Still having issue with hangup: > > FreeSWITCH Version 1.1.beta1 (git-5e5a2ff 2012-04-23 07-50-57 -0500) > > I have set the waste resources: > > data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/> > > > > > > > Thank You, George. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120501/1d293422/attachment.html From jmesquita at freeswitch.org Tue May 1 22:47:19 2012 From: jmesquita at freeswitch.org (=?utf-8?Q?Jo=C3=A3o_Mesquita?=) Date: Tue, 1 May 2012 15:47:19 -0300 Subject: [Freeswitch-users] bind_digit_action and dtmf target leg In-Reply-To: References: <4FA0161E.8040605@tagnet.ru> <8E932BDEC1D841D6A3D4909B7548DC44@freeswitch.org> Message-ID: <538CC24C49854F83A592CEF01FD9330B@freeswitch.org> And Boris, mind paying the wiki tax on that one? Regards, -- Jo?o Mesquita Sent with Sparrow (http://www.sparrowmailapp.com/?sig) On Tuesday, May 1, 2012 at 2:52 PM, Michael Collins wrote: > > > On Tue, May 1, 2012 at 10:10 AM, Jo?o Mesquita wrote: > > I think it is the leg in which you want the DTMFs to be listened in. > > > > Correct. You have "dtmf target leg" and "event target leg" - these determine which leg on which to listen for DTMFs and on which leg the action should take place, respectively. (Note that these are optional.) > > In other words the DTMFs received on one leg could trigger an action on the other leg. You could even have an action triggered on both legs. > > -MC > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120501/f2fab3ab/attachment.html From darcyp at voice2net.ca Tue May 1 23:56:40 2012 From: darcyp at voice2net.ca (Darcy Primrose) Date: Tue, 1 May 2012 15:56:40 -0400 Subject: [Freeswitch-users] unsched Message-ID: <428CE889D2654FDB99DCAFB875EFAC92@owner397fa27d2> Hi, I am trying to get a call to drop back to voice mail on timeout after I originate it to the pstn via voip. I cannot use call_timeout since all outbound calls use early media, if I disable early media, then I get not ring back tone. So I am thinking something like sched_hangup, then use an execute on answer to disable the hangup, unsched_api or sched_del. I do not know if this is the best way to do this. I am having trouble getting the task_id from the sched_hangup so I can later delete it. Any suggestions would be most welcome. For now, I would like to accomplish this in the dial plan. Darcy Primrose -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120501/5ab3efab/attachment-0001.html From krice at freeswitch.org Wed May 2 00:22:36 2012 From: krice at freeswitch.org (Ken Rice) Date: Tue, 01 May 2012 15:22:36 -0500 Subject: [Freeswitch-users] unsched In-Reply-To: <428CE889D2654FDB99DCAFB875EFAC92@owner397fa27d2> Message-ID: Then what about http://wiki.freeswitch.org/wiki/Variable_leg_timeout K On 5/1/12 2:56 PM, "Darcy Primrose" wrote: > Hi, I am trying to get a call to drop back to voice mail on timeout after I > originate it to the pstn via voip. I cannot use call_timeout since all > outbound calls use early media, if I disable early media, then I get not ring > back tone. So I am thinking something like sched_hangup, then use an execute > on answer to disable the hangup, unsched_api or sched_del. I do not know if > this is the best way to do this. I am having trouble getting the task_id from > the sched_hangup so I can later delete it. Any suggestions would be most > welcome. For now, I would like to accomplish this in the dial plan. > > > > > Darcy Primrose > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120501/b87ab0c7/attachment.html From anton.jugatsu at gmail.com Wed May 2 09:39:22 2012 From: anton.jugatsu at gmail.com (Anton Kvashenkin) Date: Wed, 2 May 2012 09:39:22 +0400 Subject: [Freeswitch-users] mod_voicemail and non-numeric id In-Reply-To: References: Message-ID: Thanks for response. It was very helpful. I just export data="voicemail_alternate_greet_id=${dialed_extension}"/> in Local_Extension extension. 1 ??? 2012 ?. 15:31 ???????????? Markus Lindenberg < markus.lindenberg at gmail.com> ???????: > It's in the wiki: Just set voicemail_alternate_greet_id before > bridging to voicemail. > > I'm using non-numeric IDs as well, so i set connectedline manually > before trying to bridge to the user: > > data="effective_callee_id_number=${user_data(${user_id}@${domain_name} > var effective_caller_id_number)}"/> > > When the call hits voicemail, i set voicemail_alternate_greet_id and > then that's what voicemail will use: > > data="voicemail_alternate_greet_id=${effective_callee_id_number}"/> > > > data="loopback/app=voicemail:default ${domain_name} ${user_id}"/> > > > > On Fri, Apr 27, 2012 at 09:51, Anton Kvashenkin > wrote: > > Hello list. > > > > I've decided to use non-numeric id's at users configuration. > > > > > > > > But when the call goes to voicemail app > > > > > > > > switch_ivr_play_say.c:244 Handle say:[001565229c95] (ru:ru) > > 2012-04-27 11:39:54.452208 [INFO] mod_say_ru.c:721 ru_say!!! > 001565229c95! > > say_opt.gender=0 say_opt.cases=0 > > 2012-04-27 11:39:54.532157 [DEBUG] switch_ivr_play_say.c:1309 Codec > > Activated L16 at 16000hz 1 channels 20ms > > > > The mod_say pronounces user id. How to force to pronounce number-alias? > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120502/01849ebb/attachment.html From webudo at gua.net Wed May 2 06:59:33 2012 From: webudo at gua.net (=?ISO-8859-1?Q?Alejandro_Mej=EDa?=) Date: Tue, 01 May 2012 20:59:33 -0600 Subject: [Freeswitch-users] Silence instead of spanish wav installed In-Reply-To: References: <4F9F3895.7090503@gua.net> Message-ID: <4FA0A315.5020605@gua.net> Thanks for your reply Michael. Yes, I can play the files manually on any player on my computer. I'm a freeswitch newbie, so I don't know how to change conference sampling rate, so I guess it's on default 8khz, same as the sound files. On 01/05/2012 11:54 a.m., Michael Collins wrote: > Can you manually play any of the sound files? Also, what sampling rate > is the conference and what sampling rate are the files? > -MC > > On Mon, Apr 30, 2012 at 6:12 PM, Alejandro Mej?a > wrote: > > Hello, > > I installed a spanish sounds package to my Freeswitch under > /usr/local/freeswitch/sounds/es/mx/ directory > I'm trying it by entering an empty conference, and I only hear some > seconds of silence (where conf-alone.wav should be played), and > then the > music on hold. > I know the files are not missing, and Freeswitch is finding them > because > when I remove "/es" directory, I get error on CLI when executing a > conference, and getting directly to MOH. > > Any help will be appreciated. > > Thanks! > > Alex > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120501/4927b1e9/attachment-0001.html From nickolayr at gmail.com Wed May 2 07:17:54 2012 From: nickolayr at gmail.com (Nikolay Rogoshchenkov) Date: Tue, 1 May 2012 23:17:54 -0400 Subject: [Freeswitch-users] preAnswer call at lua script and then bridge it to another leg Message-ID: Hello, I have a dialplan where I get the incoming call to lua script . . . and then I have answering in answer.lua: =========================== session:preAnswer(); [...accessing to DB...] if (session:ready()) then repeat index = index + 1; session_call = string.format("[leg_timeout=30,ignore_early_media=true,hangup_after_bridge=true,origination_caller_id_number=".. cid .."]sofia/gateway/%s/%s",gwlist[index],dn); legB = freeswitch.Session(session_call); until ((hcause == 'USER_BUSY') or (hcause == 'SUCCESS') or (hcause == 'NO_ANSWER') or (index == #gwlist) or(session:ready() == false)) if (legB:ready()) then. freeswitch.bridge(session, legB) end =========================== But I have a one problem, when the initial legA was disconnected before legB was answered. It it still continued to call legB (during the leg_timeout). Could you please explain, how I can cancel legB right after I got disconnect on legA in this case? Thank you. -- Nikolay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120501/46ffa510/attachment-0001.html From chrisbware at yahoo.it Wed May 2 11:21:20 2012 From: chrisbware at yahoo.it (Chris B. Ware) Date: Wed, 2 May 2012 08:21:20 +0100 (BST) Subject: [Freeswitch-users] Freeswitch speaks italian Message-ID: <1335943280.43187.YahooMailNeo@web132301.mail.ird.yahoo.com> Michael, is there a place where I can upload italian sound files? Just to add a link to the wiki page, once I've updated it. Thanks, Chris -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120502/219c999b/attachment.html From sat at calgaryit.com Wed May 2 17:41:54 2012 From: sat at calgaryit.com (George Sapak) Date: Wed, 2 May 2012 07:41:54 -0600 (MDT) Subject: [Freeswitch-users] Voicemail hangups 38 sec into recording In-Reply-To: Message-ID: <1653450234.212833.1335966114148.JavaMail.root@server3> Here is the pastebin: http://pastebin.freeswitch.org/18969 Thank You, George ----- Original Message ----- From: "Michael Collins" To: "FreeSWITCH Users Help" Sent: Tuesday, May 1, 2012 12:13:59 PM Subject: Re: [Freeswitch-users] Voicemail hangups 38 sec into recording You may want to capture the who call with media as well, just in case. I recommend one of the packet capture tools listed here: http://wiki.freeswitch.org/wiki/Packet_Capture Use wireshark to analyze the call. It will show you the whole call broken down into a nice call flow graph and you can see if one side or the other sent a BYE. Usually there's a reason listed in the BYE as to why the client is hanging up. If not you can go back to the console log and look for clues there. If you need help then go ahead and put your logs into a pastebin and post the URL in this thread. -MC On Tue, May 1, 2012 at 10:11 AM, George Sapak < sat at calgaryit.com > wrote: Still having issue with hangup: FreeSWITCH Version 1.1.beta1 (git-5e5a2ff 2012-04-23 07-50-57 -0500) I have set the waste resources: Thank You, George. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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 at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From zedek_m6 at hotmail.com Wed May 2 13:36:20 2012 From: zedek_m6 at hotmail.com (kader) Date: Wed, 2 May 2012 02:36:20 -0700 (PDT) Subject: [Freeswitch-users] =?utf-8?q?_R=C3=A9cuperer_le_digit_taper_ou_mo?= =?utf-8?q?ment_de_l=27appel=28mod=5F_HTTAPI=29?= Message-ID: <33763365.post@talk.nabble.com> Bonjour je serai tr?s reconnaissant si vous pouvez me donner me d?bloquer dans ce probl?me. j'appel une extension dans dialplan, dans laquel je demande un fichier .jsp ou serveur web, ma demande arrive le plus normal ou serveur et ce dernier renvoie le fichier jsp ou freeswitch. mon probl?me c'est que j'arrive pas ? r?cuperer les digits que je tape sur le softphone sur le serveur. voila mon fichier jsp: <% response.setContentType("text/xml"); %> ~\d+ -- View this message in context: http://old.nabble.com/R%C3%A9cuperer-le-digit-taper-ou-moment-de-l%27appel%28mod_-HTTAPI%29-tp33763365p33763365.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From daggelinckxmichel at gmail.com Wed May 2 19:13:38 2012 From: daggelinckxmichel at gmail.com (Michel Daggelinckx) Date: Wed, 02 May 2012 17:13:38 +0200 Subject: [Freeswitch-users] =?utf-8?q?R=C3=A9cuperer_le_digit_taper_ou_mom?= =?utf-8?q?ent_de_l=27appel=28mod=5F_HTTAPI=29?= In-Reply-To: <33763365.post@talk.nabble.com> References: <33763365.post@talk.nabble.com> Message-ID: <4FA14F22.5030801@gmail.com> Bonjour Nous allons vous aider ? mieux si vous d?finissez la question en anglais. La plupart des d?veloppeurs ne parlent que l'anglais. Nous vous remercions de votre compr?hension. On 05/02/2012 11:36 AM, kader wrote: > Bonjour > je serai tr?s reconnaissant si vous pouvez me donner me d?bloquer dans ce > probl?me. > j'appel une extension dans dialplan, dans laquel je demande un fichier .jsp > ou serveur web, ma demande arrive le plus normal ou serveur et ce dernier > renvoie le fichier jsp ou freeswitch. > mon probl?me c'est que j'arrive pas ? r?cuperer les digits que je tape sur > le softphone sur le serveur. > voila mon fichier jsp: > <% > response.setContentType("text/xml"); > %> > > > input-timeout="5000" digit-timeout="2000"> > ~\d+ > > > -- Michel From msc at freeswitch.org Wed May 2 19:16:06 2012 From: msc at freeswitch.org (Michael Collins) Date: Wed, 2 May 2012 08:16:06 -0700 Subject: [Freeswitch-users] Freeswitch speaks italian In-Reply-To: <1335943280.43187.YahooMailNeo@web132301.mail.ird.yahoo.com> References: <1335943280.43187.YahooMailNeo@web132301.mail.ird.yahoo.com> Message-ID: If you put them on a web server or something where I can grab them then I will put them up on files.freeswitch.org. -MC On Wed, May 2, 2012 at 12:21 AM, Chris B. Ware wrote: > Michael, > > is there a place where I can upload italian sound files? Just to add a > link to the wiki page, once I've updated it. > > Thanks, > Chris > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120502/31332836/attachment.html From msc at freeswitch.org Wed May 2 19:27:50 2012 From: msc at freeswitch.org (Michael Collins) Date: Wed, 2 May 2012 08:27:50 -0700 Subject: [Freeswitch-users] =?iso-8859-1?q?R=E9cuperer_le_digit_taper_ou_m?= =?iso-8859-1?q?oment_de_l=27appel=28mod=5F_HTTAPI=29?= In-Reply-To: <33763365.post@talk.nabble.com> References: <33763365.post@talk.nabble.com> Message-ID: (English translation below) Bienvenue ? FreeSWITCH! J'esp?re que vous pouvez comprendre ce que j'?cris. (Je ne parle pas beaucoup de fran?ais ...) Nous avons besoin de plus d'informations. Nous ne savons pas pourquoi les chiffres ne sont pas re?us. Pouvez-vous nous donner un fichier de log ? partir FreeSWITCH? S'il vous pla?t utiliser notre pastebin ? http://pastebin.freeswitch.org. -MC (For those who prefer English, the issue is that he cannot receive DTMF digits that he types on his soft phone. I responded that we need more information - preferably a FreeSWITCH log put into pastebin.) 2012/5/2 kader > > Bonjour > je serai tr?s reconnaissant si vous pouvez me donner me d?bloquer dans ce > probl?me. > j'appel une extension dans dialplan, dans laquel je demande un fichier > .jsp > ou serveur web, ma demande arrive le plus normal ou serveur et ce dernier > renvoie le fichier jsp ou freeswitch. > mon probl?me c'est que j'arrive pas ? r?cuperer les digits que je tape sur > le softphone sur le serveur. > voila mon fichier jsp: > <% > response.setContentType("text/xml"); > %> > > > input-timeout="5000" digit-timeout="2000"> > ~\d+ > > > > -- > View this message in context: > http://old.nabble.com/R%C3%A9cuperer-le-digit-taper-ou-moment-de-l%27appel%28mod_-HTTAPI%29-tp33763365p33763365.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120502/10253a2a/attachment.html From msc at freeswitch.org Wed May 2 19:34:22 2012 From: msc at freeswitch.org (Michael Collins) Date: Wed, 2 May 2012 08:34:22 -0700 Subject: [Freeswitch-users] Voicemail hangups 38 sec into recording In-Reply-To: <1653450234.212833.1335966114148.JavaMail.root@server3> References: <1653450234.212833.1335966114148.JavaMail.root@server3> Message-ID: It looks like the Nortel is still hanging up on you. For kicks, try using export instead of set on this line before the bridge: I'm wonder if maybe the flag isn't getting set on the right leg. -MC On Wed, May 2, 2012 at 6:41 AM, George Sapak wrote: > Here is the pastebin: > > http://pastebin.freeswitch.org/18969 > > Thank You, > George > > ----- Original Message ----- > From: "Michael Collins" > To: "FreeSWITCH Users Help" > Sent: Tuesday, May 1, 2012 12:13:59 PM > Subject: Re: [Freeswitch-users] Voicemail hangups 38 sec into recording > > > You may want to capture the who call with media as well, just in case. I > recommend one of the packet capture tools listed here: > http://wiki.freeswitch.org/wiki/Packet_Capture > > Use wireshark to analyze the call. It will show you the whole call broken > down into a nice call flow graph and you can see if one side or the other > sent a BYE. Usually there's a reason listed in the BYE as to why the client > is hanging up. If not you can go back to the console log and look for clues > there. If you need help then go ahead and put your logs into a pastebin and > post the URL in this thread. > > -MC > > > On Tue, May 1, 2012 at 10:11 AM, George Sapak < sat at calgaryit.com > wrote: > > > Still having issue with hangup: > > FreeSWITCH Version 1.1.beta1 (git-5e5a2ff 2012-04-23 07-50-57 -0500) > > I have set the waste resources: > > data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/> > > > > > > > Thank You, George. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120502/e35e00c2/attachment-0001.html From msc at freeswitch.org Wed May 2 19:36:30 2012 From: msc at freeswitch.org (Michael Collins) Date: Wed, 2 May 2012 08:36:30 -0700 Subject: [Freeswitch-users] Silence instead of spanish wav installed In-Reply-To: <4FA0A315.5020605@gua.net> References: <4F9F3895.7090503@gua.net> <4FA0A315.5020605@gua.net> Message-ID: Okay, next step is log output. Get a console debug log and drop it onto pastebin. Lots of useful information available here: http://wiki.freeswitch.org/wiki/Reporting_Bugs -MC On Tue, May 1, 2012 at 7:59 PM, Alejandro Mej?a wrote: > Thanks for your reply Michael. > > Yes, I can play the files manually on any player on my computer. > I'm a freeswitch newbie, so I don't know how to change conference sampling > rate, so I guess it's on default 8khz, same as the sound files. > > On 01/05/2012 11:54 a.m., Michael Collins wrote: > > Can you manually play any of the sound files? Also, what sampling rate is > the conference and what sampling rate are the files? > -MC > > On Mon, Apr 30, 2012 at 6:12 PM, Alejandro Mej?a wrote: > >> Hello, >> >> I installed a spanish sounds package to my Freeswitch under >> /usr/local/freeswitch/sounds/es/mx/ directory >> I'm trying it by entering an empty conference, and I only hear some >> seconds of silence (where conf-alone.wav should be played), and then the >> music on hold. >> I know the files are not missing, and Freeswitch is finding them because >> when I remove "/es" directory, I get error on CLI when executing a >> conference, and getting directly to MOH. >> >> Any help will be appreciated. >> >> Thanks! >> >> Alex >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120502/4aeac916/attachment.html From msc at freeswitch.org Wed May 2 19:51:38 2012 From: msc at freeswitch.org (Michael Collins) Date: Wed, 2 May 2012 08:51:38 -0700 Subject: [Freeswitch-users] FreeSWITCH Conference Call: Vestec ASR Message-ID: Hello all! Just a reminder that the FreeSWITCH community conference call will be starting in just over an hour. The agenda page is here. Come join us for an interesting talk with the Vestec team about automatic speech recognition (ASR) in a FreeSWITCH environment. -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120502/3ee6276c/attachment.html From sat at calgaryit.com Wed May 2 19:58:59 2012 From: sat at calgaryit.com (George Sapak) Date: Wed, 2 May 2012 09:58:59 -0600 (MDT) Subject: [Freeswitch-users] Voicemail hangups 38 sec into recording In-Reply-To: Message-ID: <761178507.213797.1335974339900.JavaMail.root@server3> that worked what gives, on another note anyway to strip ;phone-context=national after the phone number from the To: line? I have all Aastra phones and some how the pick this up. To: Thank You, George. ----- Original Message ----- From: "Michael Collins" To: "FreeSWITCH Users Help" Sent: Wednesday, May 2, 2012 9:34:22 AM Subject: Re: [Freeswitch-users] Voicemail hangups 38 sec into recording It looks like the Nortel is still hanging up on you. For kicks, try using export instead of set on this line before the bridge: I'm wonder if maybe the flag isn't getting set on the right leg. -MC On Wed, May 2, 2012 at 6:41 AM, George Sapak < sat at calgaryit.com > wrote: Here is the pastebin: http://pastebin.freeswitch.org/18969 Thank You, George ----- Original Message ----- From: "Michael Collins" < msc at freeswitch.org > To: "FreeSWITCH Users Help" < freeswitch-users at lists.freeswitch.org > Sent: Tuesday, May 1, 2012 12:13:59 PM Subject: Re: [Freeswitch-users] Voicemail hangups 38 sec into recording You may want to capture the who call with media as well, just in case. I recommend one of the packet capture tools listed here: http://wiki.freeswitch.org/wiki/Packet_Capture Use wireshark to analyze the call. It will show you the whole call broken down into a nice call flow graph and you can see if one side or the other sent a BYE. Usually there's a reason listed in the BYE as to why the client is hanging up. If not you can go back to the console log and look for clues there. If you need help then go ahead and put your logs into a pastebin and post the URL in this thread. -MC On Tue, May 1, 2012 at 10:11 AM, George Sapak < sat at calgaryit.com > wrote: Still having issue with hangup: FreeSWITCH Version 1.1.beta1 (git-5e5a2ff 2012-04-23 07-50-57 -0500) I have set the waste resources: Thank You, George. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From msc at freeswitch.org Wed May 2 20:05:23 2012 From: msc at freeswitch.org (Michael Collins) Date: Wed, 2 May 2012 09:05:23 -0700 Subject: [Freeswitch-users] Voicemail hangups 38 sec into recording In-Reply-To: <761178507.213797.1335974339900.JavaMail.root@server3> References: <761178507.213797.1335974339900.JavaMail.root@server3> Message-ID: I'll have to defer to those who know more about the subject... -MC On Wed, May 2, 2012 at 8:58 AM, George Sapak wrote: > that worked what gives, on another note anyway to strip > ;phone-context=national after the phone number from the To: line? I have > all Aastra phones and some how the pick this up. > > To: > > Thank You, George. > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120502/e3a5e79c/attachment.html From bhegades at gmail.com Wed May 2 20:19:06 2012 From: bhegades at gmail.com (Mahendra Bhegade) Date: Wed, 2 May 2012 09:19:06 -0700 Subject: [Freeswitch-users] Help - SIP gateway issue with fs_cli but not with freeswtich console. In-Reply-To: References: Message-ID: Hi All, I had Freeswitch service running and freeswtich console too and that is where the problem was. FS_CLI was trying to connect to the Console instance where the gateway was not available. Thanks for the prompt response. Mahendra On Sun, Apr 29, 2012 at 3:36 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > You would need to supply some details like issue sofia global siptrace on > and console loglevel debug and capture the logs > On Apr 29, 2012 4:30 PM, "Mahendra Bhegade" wrote: > >> Hi , >> >> I was wondering if you can help me with the issue that I am facing. I >> have a lua script which dials out using a gateway and works fine from >> freeswitch console. If I execute the same script from fs_cli I get an error >> identifiying an error with gateway and the complains about the numbe being >> invalued >> >> I am using a flowroute gateway and sip end point is >> sofia/gateway/flowroute/15104496650 my entry for flowroute.xml is under >> conf/sip_profiles/external >> >> If some one can shed some light on the matter I would appreciate it. >> >> >> >> Mahendra >> >> 510-449-6650 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120502/015c3941/attachment.html From jmoran at secureachsystems.com Wed May 2 20:55:45 2012 From: jmoran at secureachsystems.com (Jason Moran) Date: Wed, 2 May 2012 12:55:45 -0400 Subject: [Freeswitch-users] Installing FS to a non-standard directory Message-ID: <361E98F99D3CC3439EED59BC1924ED695ED1CD@SERVER2003.SecuReachSystems.local> I do not want to install FS to the normal /usr/local/freeswitch/ Where do I make a change such that the install goes elsewhere? After running ./bootstrap.sh then ./configure it lists the install directory as /usr/local/freeswitch/ but I cannot find any inputs to change these values. Thanks, Jason -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120502/463ee061/attachment-0001.html From jmoran at secureachsystems.com Wed May 2 21:00:40 2012 From: jmoran at secureachsystems.com (Jason Moran) Date: Wed, 2 May 2012 13:00:40 -0400 Subject: [Freeswitch-users] Installing FS to a non-standard directory References: <361E98F99D3CC3439EED59BC1924ED695ED1CD@SERVER2003.SecuReachSystems.local> Message-ID: <361E98F99D3CC3439EED59BC1924ED695ED1CE@SERVER2003.SecuReachSystems.local> I think I've figured it out... ./configure --prefix=YOUR_NEW_PATH From: Jason Moran Sent: Wednesday, May 02, 2012 12:56 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Installing FS to a non-standard directory I do not want to install FS to the normal /usr/local/freeswitch/ Where do I make a change such that the install goes elsewhere? After running ./bootstrap.sh then ./configure it lists the install directory as /usr/local/freeswitch/ but I cannot find any inputs to change these values. Thanks, Jason -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120502/e7adb237/attachment.html From msc at freeswitch.org Wed May 2 23:57:36 2012 From: msc at freeswitch.org (Michael Collins) Date: Wed, 2 May 2012 12:57:36 -0700 Subject: [Freeswitch-users] Installing FS to a non-standard directory In-Reply-To: <361E98F99D3CC3439EED59BC1924ED695ED1CE@SERVER2003.SecuReachSystems.local> References: <361E98F99D3CC3439EED59BC1924ED695ED1CD@SERVER2003.SecuReachSystems.local> <361E98F99D3CC3439EED59BC1924ED695ED1CE@SERVER2003.SecuReachSystems.local> Message-ID: On Wed, May 2, 2012 at 10:00 AM, Jason Moran wrote: > I think I?ve figured it out?**** > > ./configure --prefix=YOUR_NEW_PATH > This is correct. Let us know if you found this information on the wiki or if you had to figure it out by some other means. (This is a perfect item that should be on the wiki.) -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120502/2b922827/attachment.html From webudo at gua.net Wed May 2 23:59:55 2012 From: webudo at gua.net (=?ISO-8859-1?Q?Alejandro_Mej=EDa?=) Date: Wed, 02 May 2012 13:59:55 -0600 Subject: [Freeswitch-users] Silence instead of spanish wav installed In-Reply-To: References: <4F9F3895.7090503@gua.net> <4FA0A315.5020605@gua.net> Message-ID: <4FA1923B.3080807@gua.net> Log's output here: http://pastebin.freeswitch.org/pastebin.php?dl=18970 Audio remains silent during .wav file's length when reaching the line: 2012-05-02 13:13:22.728301 [INFO] switch_rtp.c:3164 Auto Changing port from 172.16.1.97:5016 to 190.149.242.46:15046 Then it plays MOH normally. I'm using fusionpbx's canned version of FS, and I'm updating now to see if there was a bug that's been fixed. I'll let you know after the process completes. Thanks! Alex On 02/05/2012 09:36 a.m., Michael Collins wrote: > Okay, next step is log output. Get a console debug log and drop it > onto pastebin. Lots of useful information available here: > > http://wiki.freeswitch.org/wiki/Reporting_Bugs > > -MC > > On Tue, May 1, 2012 at 7:59 PM, Alejandro Mej?a > wrote: > > Thanks for your reply Michael. > > Yes, I can play the files manually on any player on my computer. > I'm a freeswitch newbie, so I don't know how to change conference > sampling rate, so I guess it's on default 8khz, same as the sound > files. > > On 01/05/2012 11:54 a.m., Michael Collins wrote: >> Can you manually play any of the sound files? Also, what sampling >> rate is the conference and what sampling rate are the files? >> -MC >> >> On Mon, Apr 30, 2012 at 6:12 PM, Alejandro Mej?a > > wrote: >> >> Hello, >> >> I installed a spanish sounds package to my Freeswitch under >> /usr/local/freeswitch/sounds/es/mx/ directory >> I'm trying it by entering an empty conference, and I only >> hear some >> seconds of silence (where conf-alone.wav should be played), >> and then the >> music on hold. >> I know the files are not missing, and Freeswitch is finding >> them because >> when I remove "/es" directory, I get error on CLI when >> executing a >> conference, and getting directly to MOH. >> >> Any help will be appreciated. >> >> Thanks! >> >> Alex >> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120502/b62035fa/attachment.html From webudo at gua.net Thu May 3 01:09:06 2012 From: webudo at gua.net (=?ISO-8859-1?Q?Alejandro_Mej=EDa?=) Date: Wed, 02 May 2012 15:09:06 -0600 Subject: [Freeswitch-users] Silence instead of spanish wav installed In-Reply-To: <4FA1923B.3080807@gua.net> References: <4F9F3895.7090503@gua.net> <4FA0A315.5020605@gua.net> <4FA1923B.3080807@gua.net> Message-ID: <4FA1A272.8030700@gua.net> Update finished, but the problem persists. Perhaps it helps you to have the audio file being used? (let me know and I can send it) On 02/05/2012 01:59 p.m., Alejandro Mej?a wrote: > Log's output here: http://pastebin.freeswitch.org/pastebin.php?dl=18970 > > Audio remains silent during .wav file's length when reaching the line: > 2012-05-02 13:13:22.728301 [INFO] switch_rtp.c:3164 Auto Changing port > from 172.16.1.97:5016 to 190.149.242.46:15046 > > Then it plays MOH normally. > > I'm using fusionpbx's canned version of FS, and I'm updating now to > see if there was a bug that's been fixed. > I'll let you know after the process completes. > > Thanks! > > Alex > > On 02/05/2012 09:36 a.m., Michael Collins wrote: >> Okay, next step is log output. Get a console debug log and drop it >> onto pastebin. Lots of useful information available here: >> >> http://wiki.freeswitch.org/wiki/Reporting_Bugs >> >> -MC >> >> On Tue, May 1, 2012 at 7:59 PM, Alejandro Mej?a > > wrote: >> >> Thanks for your reply Michael. >> >> Yes, I can play the files manually on any player on my computer. >> I'm a freeswitch newbie, so I don't know how to change conference >> sampling rate, so I guess it's on default 8khz, same as the sound >> files. >> >> On 01/05/2012 11:54 a.m., Michael Collins wrote: >>> Can you manually play any of the sound files? Also, what >>> sampling rate is the conference and what sampling rate are the >>> files? >>> -MC >>> >>> On Mon, Apr 30, 2012 at 6:12 PM, Alejandro Mej?a >> > wrote: >>> >>> Hello, >>> >>> I installed a spanish sounds package to my Freeswitch under >>> /usr/local/freeswitch/sounds/es/mx/ directory >>> I'm trying it by entering an empty conference, and I only >>> hear some >>> seconds of silence (where conf-alone.wav should be played), >>> and then the >>> music on hold. >>> I know the files are not missing, and Freeswitch is finding >>> them because >>> when I remove "/es" directory, I get error on CLI when >>> executing a >>> conference, and getting directly to MOH. >>> >>> Any help will be appreciated. >>> >>> Thanks! >>> >>> Alex >>> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120502/94b87435/attachment-0001.html From msc at freeswitch.org Thu May 3 01:19:46 2012 From: msc at freeswitch.org (Michael Collins) Date: Wed, 2 May 2012 14:19:46 -0700 Subject: [Freeswitch-users] Silence instead of spanish wav installed In-Reply-To: <4FA1A272.8030700@gua.net> References: <4F9F3895.7090503@gua.net> <4FA0A315.5020605@gua.net> <4FA1923B.3080807@gua.net> <4FA1A272.8030700@gua.net> Message-ID: Where did you download these files? I would like to download them myself and do some testing. -MC On Wed, May 2, 2012 at 2:09 PM, Alejandro Mej?a wrote: > Update finished, but the problem persists. > Perhaps it helps you to have the audio file being used? (let me know and I > can send it) > > On 02/05/2012 01:59 p.m., Alejandro Mej?a wrote: > > Log's output here: http://pastebin.freeswitch.org/pastebin.php?dl=18970 > > Audio remains silent during .wav file's length when reaching the line: > 2012-05-02 13:13:22.728301 [INFO] switch_rtp.c:3164 Auto Changing port > from 172.16.1.97:5016 to 190.149.242.46:15046 > > Then it plays MOH normally. > > I'm using fusionpbx's canned version of FS, and I'm updating now to see if > there was a bug that's been fixed. > I'll let you know after the process completes. > > Thanks! > > Alex > > On 02/05/2012 09:36 a.m., Michael Collins wrote: > > Okay, next step is log output. Get a console debug log and drop it onto > pastebin. Lots of useful information available here: > > http://wiki.freeswitch.org/wiki/Reporting_Bugs > > -MC > > On Tue, May 1, 2012 at 7:59 PM, Alejandro Mej?a wrote: > >> Thanks for your reply Michael. >> >> Yes, I can play the files manually on any player on my computer. >> I'm a freeswitch newbie, so I don't know how to change conference >> sampling rate, so I guess it's on default 8khz, same as the sound files. >> >> On 01/05/2012 11:54 a.m., Michael Collins wrote: >> >> Can you manually play any of the sound files? Also, what sampling rate is >> the conference and what sampling rate are the files? >> -MC >> >> On Mon, Apr 30, 2012 at 6:12 PM, Alejandro Mej?a wrote: >> >>> Hello, >>> >>> I installed a spanish sounds package to my Freeswitch under >>> /usr/local/freeswitch/sounds/es/mx/ directory >>> I'm trying it by entering an empty conference, and I only hear some >>> seconds of silence (where conf-alone.wav should be played), and then the >>> music on hold. >>> I know the files are not missing, and Freeswitch is finding them because >>> when I remove "/es" directory, I get error on CLI when executing a >>> conference, and getting directly to MOH. >>> >>> Any help will be appreciated. >>> >>> Thanks! >>> >>> Alex >>> >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120502/2378d94f/attachment.html From webudo at gua.net Thu May 3 01:26:03 2012 From: webudo at gua.net (=?ISO-8859-1?Q?Alejandro_Mej=EDa?=) Date: Wed, 02 May 2012 15:26:03 -0600 Subject: [Freeswitch-users] Silence instead of spanish wav installed In-Reply-To: References: <4F9F3895.7090503@gua.net> <4FA0A315.5020605@gua.net> <4FA1923B.3080807@gua.net> <4FA1A272.8030700@gua.net> Message-ID: <4FA1A66B.4080208@gua.net> It's a commercial pack of voices bought from westany.com That's why I didn't include the file here, or a link to it, because I have not read their terms of usage and I don't want to get in trouble. You may get some samples here: http://www.westany.com/mexican_spanish_freeswitch_voice_prompts On 02/05/2012 03:19 p.m., Michael Collins wrote: > Where did you download these files? I would like to download them > myself and do some testing. > -MC > > On Wed, May 2, 2012 at 2:09 PM, Alejandro Mej?a > wrote: > > Update finished, but the problem persists. > Perhaps it helps you to have the audio file being used? (let me > know and I can send it) > > On 02/05/2012 01:59 p.m., Alejandro Mej?a wrote: >> Log's output here: >> http://pastebin.freeswitch.org/pastebin.php?dl=18970 >> >> Audio remains silent during .wav file's length when reaching the >> line: >> 2012-05-02 13:13:22.728301 [INFO] switch_rtp.c:3164 Auto Changing >> port from 172.16.1.97:5016 to >> 190.149.242.46 :15046 >> >> Then it plays MOH normally. >> >> I'm using fusionpbx's canned version of FS, and I'm updating now >> to see if there was a bug that's been fixed. >> I'll let you know after the process completes. >> >> Thanks! >> >> Alex >> >> On 02/05/2012 09:36 a.m., Michael Collins wrote: >>> Okay, next step is log output. Get a console debug log and drop >>> it onto pastebin. Lots of useful information available here: >>> >>> http://wiki.freeswitch.org/wiki/Reporting_Bugs >>> >>> -MC >>> >>> On Tue, May 1, 2012 at 7:59 PM, Alejandro Mej?a >> > wrote: >>> >>> Thanks for your reply Michael. >>> >>> Yes, I can play the files manually on any player on my computer. >>> I'm a freeswitch newbie, so I don't know how to change >>> conference sampling rate, so I guess it's on default 8khz, >>> same as the sound files. >>> >>> On 01/05/2012 11:54 a.m., Michael Collins wrote: >>>> Can you manually play any of the sound files? Also, what >>>> sampling rate is the conference and what sampling rate are >>>> the files? >>>> -MC >>>> >>>> On Mon, Apr 30, 2012 at 6:12 PM, Alejandro Mej?a >>>> > wrote: >>>> >>>> Hello, >>>> >>>> I installed a spanish sounds package to my Freeswitch under >>>> /usr/local/freeswitch/sounds/es/mx/ directory >>>> I'm trying it by entering an empty conference, and I >>>> only hear some >>>> seconds of silence (where conf-alone.wav should be >>>> played), and then the >>>> music on hold. >>>> I know the files are not missing, and Freeswitch is >>>> finding them because >>>> when I remove "/es" directory, I get error on CLI when >>>> executing a >>>> conference, and getting directly to MOH. >>>> >>>> Any help will be appreciated. >>>> >>>> Thanks! >>>> >>>> Alex >>>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120502/c9793003/attachment-0001.html From msc at freeswitch.org Thu May 3 01:26:55 2012 From: msc at freeswitch.org (Michael Collins) Date: Wed, 2 May 2012 14:26:55 -0700 Subject: [Freeswitch-users] preAnswer call at lua script and then bridge it to another leg In-Reply-To: References: Message-ID: The best way to handle this would be to use the bridge application outside of the Lua script. Go ahead and do the Lua script stuff w/ your database and then construct your dialstring. Set that to a channel variable, like "my_dialstring": session:setVariable('my_dialstring',session_call) And then let the Lua script exit. Then just add a bridge to your dialplan: As a rule of thumb: if you *can* do it in the dialplan then you probably *should* do it in the dialplan. Use the scripting language for what it's good at and use the dialplan for what it's good at. Lua is good for logic, db lookups, and more complicated processing. The dialplan is awesome at connecting call legs. The bridge app will do all the work for you so you can focus on more important pursuits. -MC On Tue, May 1, 2012 at 8:17 PM, Nikolay Rogoshchenkov wrote: > Hello, > > I have a dialplan where I get the incoming call to lua script > > > > > . > . > . > > > > and then I have answering in answer.lua: > =========================== > session:preAnswer(); > > [...accessing to DB...] > > if (session:ready()) then > repeat > index = index + 1; > session_call = > string.format("[leg_timeout=30,ignore_early_media=true,hangup_after_bridge=true,origination_caller_id_number=".. > cid .."]sofia/gateway/%s/%s",gwlist[index],dn); > legB = freeswitch.Session(session_call); > until ((hcause == 'USER_BUSY') or (hcause == 'SUCCESS') or (hcause == > 'NO_ANSWER') or (index == #gwlist) or(session:ready() == false)) > > if (legB:ready()) then. > freeswitch.bridge(session, legB) > end > =========================== > > But I have a one problem, when the initial legA was disconnected before > legB was answered. It it still continued to call legB (during the > leg_timeout). > Could you please explain, how I can cancel legB right after I got > disconnect on legA in this case? Thank you. > -- > Nikolay > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120502/3400e691/attachment.html From msc at freeswitch.org Thu May 3 01:30:50 2012 From: msc at freeswitch.org (Michael Collins) Date: Wed, 2 May 2012 14:30:50 -0700 Subject: [Freeswitch-users] Silence instead of spanish wav installed In-Reply-To: <4FA1A66B.4080208@gua.net> References: <4F9F3895.7090503@gua.net> <4FA0A315.5020605@gua.net> <4FA1923B.3080807@gua.net> <4FA1A272.8030700@gua.net> <4FA1A66B.4080208@gua.net> Message-ID: Boo! Hiss! :P I'm not a fan of proprietary. Oh well. Tell you what - maybe since you paid them some $$ (or pesos or whatever) maybe they can provide you some support. :) Alternatively, you could contact them and let them know that the FS devs would like to troubleshoot an issue with the files and would like permission to receive a copy of the files *gratis*. Let me know what they say. -MC On Wed, May 2, 2012 at 2:26 PM, Alejandro Mej?a wrote: > It's a commercial pack of voices bought from westany.com > That's why I didn't include the file here, or a link to it, because I have > not read their terms of usage and I don't want to get in trouble. > You may get some samples here: > http://www.westany.com/mexican_spanish_freeswitch_voice_prompts > > > On 02/05/2012 03:19 p.m., Michael Collins wrote: > > Where did you download these files? I would like to download them myself > and do some testing. > -MC > > On Wed, May 2, 2012 at 2:09 PM, Alejandro Mej?a wrote: > >> Update finished, but the problem persists. >> Perhaps it helps you to have the audio file being used? (let me know and >> I can send it) >> >> On 02/05/2012 01:59 p.m., Alejandro Mej?a wrote: >> >> Log's output here: http://pastebin.freeswitch.org/pastebin.php?dl=18970 >> >> Audio remains silent during .wav file's length when reaching the line: >> 2012-05-02 13:13:22.728301 [INFO] switch_rtp.c:3164 Auto Changing port >> from 172.16.1.97:5016 to 190.149.242.46:15046 >> >> Then it plays MOH normally. >> >> I'm using fusionpbx's canned version of FS, and I'm updating now to see >> if there was a bug that's been fixed. >> I'll let you know after the process completes. >> >> Thanks! >> >> Alex >> >> On 02/05/2012 09:36 a.m., Michael Collins wrote: >> >> Okay, next step is log output. Get a console debug log and drop it onto >> pastebin. Lots of useful information available here: >> >> http://wiki.freeswitch.org/wiki/Reporting_Bugs >> >> -MC >> >> On Tue, May 1, 2012 at 7:59 PM, Alejandro Mej?a wrote: >> >>> Thanks for your reply Michael. >>> >>> Yes, I can play the files manually on any player on my computer. >>> I'm a freeswitch newbie, so I don't know how to change conference >>> sampling rate, so I guess it's on default 8khz, same as the sound files. >>> >>> On 01/05/2012 11:54 a.m., Michael Collins wrote: >>> >>> Can you manually play any of the sound files? Also, what sampling rate >>> is the conference and what sampling rate are the files? >>> -MC >>> >>> On Mon, Apr 30, 2012 at 6:12 PM, Alejandro Mej?a wrote: >>> >>>> Hello, >>>> >>>> I installed a spanish sounds package to my Freeswitch under >>>> /usr/local/freeswitch/sounds/es/mx/ directory >>>> I'm trying it by entering an empty conference, and I only hear some >>>> seconds of silence (where conf-alone.wav should be played), and then the >>>> music on hold. >>>> I know the files are not missing, and Freeswitch is finding them because >>>> when I remove "/es" directory, I get error on CLI when executing a >>>> conference, and getting directly to MOH. >>>> >>>> Any help will be appreciated. >>>> >>>> Thanks! >>>> >>>> Alex >>>> >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com >> >> >> >> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com >> >> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120502/801bec73/attachment-0001.html From webudo at gua.net Thu May 3 01:41:34 2012 From: webudo at gua.net (=?ISO-8859-1?Q?Alejandro_Mej=EDa?=) Date: Wed, 02 May 2012 15:41:34 -0600 Subject: [Freeswitch-users] Silence instead of spanish wav installed In-Reply-To: References: <4F9F3895.7090503@gua.net> <4FA0A315.5020605@gua.net> <4FA1923B.3080807@gua.net> <4FA1A272.8030700@gua.net> <4FA1A66B.4080208@gua.net> Message-ID: <4FA1AA0E.9090409@gua.net> I know. I'm also not a fan of propietary, but I had to place spanish recordings for one of my customers urgently, and I thought that buying those would save me some trouble finding missing wav files from open packages. Maybe you can recommend me an open package for mexican or latin american spanish that I can use instead of this one, ask for a refund with this guys, and make a donation to the FS project instead ;) On 02/05/2012 03:30 p.m., Michael Collins wrote: > Boo! Hiss! :P > > I'm not a fan of proprietary. Oh well. Tell you what - maybe since you > paid them some $$ (or pesos or whatever) maybe they can provide you > some support. :) Alternatively, you could contact them and let them > know that the FS devs would like to troubleshoot an issue with the > files and would like permission to receive a copy of the files /gratis/. > > Let me know what they say. > -MC > > On Wed, May 2, 2012 at 2:26 PM, Alejandro Mej?a > wrote: > > It's a commercial pack of voices bought from westany.com > > That's why I didn't include the file here, or a link to it, > because I have not read their terms of usage and I don't want to > get in trouble. > You may get some samples here: > http://www.westany.com/mexican_spanish_freeswitch_voice_prompts > > > On 02/05/2012 03:19 p.m., Michael Collins wrote: >> Where did you download these files? I would like to download them >> myself and do some testing. >> -MC >> >> On Wed, May 2, 2012 at 2:09 PM, Alejandro Mej?a > > wrote: >> >> Update finished, but the problem persists. >> Perhaps it helps you to have the audio file being used? (let >> me know and I can send it) >> >> On 02/05/2012 01:59 p.m., Alejandro Mej?a wrote: >>> Log's output here: >>> http://pastebin.freeswitch.org/pastebin.php?dl=18970 >>> >>> Audio remains silent during .wav file's length when reaching >>> the line: >>> 2012-05-02 13:13:22.728301 [INFO] switch_rtp.c:3164 Auto >>> Changing port from 172.16.1.97:5016 >>> to 190.149.242.46 >>> :15046 >>> >>> Then it plays MOH normally. >>> >>> I'm using fusionpbx's canned version of FS, and I'm updating >>> now to see if there was a bug that's been fixed. >>> I'll let you know after the process completes. >>> >>> Thanks! >>> >>> Alex >>> >>> On 02/05/2012 09:36 a.m., Michael Collins wrote: >>>> Okay, next step is log output. Get a console debug log and >>>> drop it onto pastebin. Lots of useful information available >>>> here: >>>> >>>> http://wiki.freeswitch.org/wiki/Reporting_Bugs >>>> >>>> -MC >>>> >>>> On Tue, May 1, 2012 at 7:59 PM, Alejandro Mej?a >>>> > wrote: >>>> >>>> Thanks for your reply Michael. >>>> >>>> Yes, I can play the files manually on any player on my >>>> computer. >>>> I'm a freeswitch newbie, so I don't know how to change >>>> conference sampling rate, so I guess it's on default >>>> 8khz, same as the sound files. >>>> >>>> On 01/05/2012 11:54 a.m., Michael Collins wrote: >>>>> Can you manually play any of the sound files? Also, >>>>> what sampling rate is the conference and what sampling >>>>> rate are the files? >>>>> -MC >>>>> >>>>> On Mon, Apr 30, 2012 at 6:12 PM, Alejandro Mej?a >>>>> > wrote: >>>>> >>>>> Hello, >>>>> >>>>> I installed a spanish sounds package to my >>>>> Freeswitch under >>>>> /usr/local/freeswitch/sounds/es/mx/ directory >>>>> I'm trying it by entering an empty conference, and >>>>> I only hear some >>>>> seconds of silence (where conf-alone.wav should be >>>>> played), and then the >>>>> music on hold. >>>>> I know the files are not missing, and Freeswitch >>>>> is finding them because >>>>> when I remove "/es" directory, I get error on CLI >>>>> when executing a >>>>> conference, and getting directly to MOH. >>>>> >>>>> Any help will be appreciated. >>>>> >>>>> Thanks! >>>>> >>>>> Alex >>>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at 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 at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120502/b62d57fb/attachment-0001.html From msc at freeswitch.org Thu May 3 01:55:51 2012 From: msc at freeswitch.org (Michael Collins) Date: Wed, 2 May 2012 14:55:51 -0700 Subject: [Freeswitch-users] Silence instead of spanish wav installed In-Reply-To: <4FA1AA0E.9090409@gua.net> References: <4F9F3895.7090503@gua.net> <4FA0A315.5020605@gua.net> <4FA1923B.3080807@gua.net> <4FA1A272.8030700@gua.net> <4FA1A66B.4080208@gua.net> <4FA1AA0E.9090409@gua.net> Message-ID: Well, since you paid these guys to give you the sounds, how about you pay me for the answer! :P If you do a "soxi" on the sounds you'll see something like this: racserv01:/usr/local/freeswitch/sounds# soxi conf-alone.wav Input File : 'conf-alone.wav' Channels : 1 Sample Rate : 8000 Precision : 24-bit Duration : 00:00:03.88 = 31009 samples ~ 290.709 CDDA sectors File Size : 124k Bit Rate : 256k *Sample Encoding: 32-bit Floating Point PCM* The sample encoding is 32-bit FP. If you resample the file with sox it will work. (I just tested it.): sox conf-alone.wav -r 8000 -e signed-integer -b 16 conf-alone-working.wav If you "soxi" it you should see something like this: racserv01:/usr/local/freeswitch/sounds# soxi conf-alone-working.wav Input File : 'conf-alone-working.wav' Channels : 1 Sample Rate : 8000 Precision : 16-bit Duration : 00:00:03.88 = 31009 samples ~ 290.709 CDDA sectors File Size : 62.1k Bit Rate : 128k *Sample Encoding: 16-bit Signed Integer PCM* Give it a try. If it works ask them if they supply the files in 16 bit signed integer format instead of 32 bit floating point. If not then you can just sox-ify them all with a shell script. Ping me offline if you run into any other trouble. -MC On Wed, May 2, 2012 at 2:41 PM, Alejandro Mej?a wrote: > I know. I'm also not a fan of propietary, but I had to place spanish > recordings for one of my customers urgently, and I thought that buying > those would save me some trouble finding missing wav files from open > packages. > Maybe you can recommend me an open package for mexican or latin american > spanish that I can use instead of this one, ask for a refund with this > guys, and make a donation to the FS project instead ;) > > > On 02/05/2012 03:30 p.m., Michael Collins wrote: > > Boo! Hiss! :P > > I'm not a fan of proprietary. Oh well. Tell you what - maybe since you > paid them some $$ (or pesos or whatever) maybe they can provide you some > support. :) Alternatively, you could contact them and let them know that > the FS devs would like to troubleshoot an issue with the files and would > like permission to receive a copy of the files *gratis*. > > Let me know what they say. > -MC > > On Wed, May 2, 2012 at 2:26 PM, Alejandro Mej?a wrote: > >> It's a commercial pack of voices bought from westany.com >> That's why I didn't include the file here, or a link to it, because I >> have not read their terms of usage and I don't want to get in trouble. >> You may get some samples here: >> http://www.westany.com/mexican_spanish_freeswitch_voice_prompts >> >> >> On 02/05/2012 03:19 p.m., Michael Collins wrote: >> >> Where did you download these files? I would like to download them myself >> and do some testing. >> -MC >> >> On Wed, May 2, 2012 at 2:09 PM, Alejandro Mej?a wrote: >> >>> Update finished, but the problem persists. >>> Perhaps it helps you to have the audio file being used? (let me know and >>> I can send it) >>> >>> On 02/05/2012 01:59 p.m., Alejandro Mej?a wrote: >>> >>> Log's output here: http://pastebin.freeswitch.org/pastebin.php?dl=18970 >>> >>> Audio remains silent during .wav file's length when reaching the line: >>> 2012-05-02 13:13:22.728301 [INFO] switch_rtp.c:3164 Auto Changing port >>> from 172.16.1.97:5016 to 190.149.242.46:15046 >>> >>> Then it plays MOH normally. >>> >>> I'm using fusionpbx's canned version of FS, and I'm updating now to see >>> if there was a bug that's been fixed. >>> I'll let you know after the process completes. >>> >>> Thanks! >>> >>> Alex >>> >>> On 02/05/2012 09:36 a.m., Michael Collins wrote: >>> >>> Okay, next step is log output. Get a console debug log and drop it onto >>> pastebin. Lots of useful information available here: >>> >>> http://wiki.freeswitch.org/wiki/Reporting_Bugs >>> >>> -MC >>> >>> On Tue, May 1, 2012 at 7:59 PM, Alejandro Mej?a wrote: >>> >>>> Thanks for your reply Michael. >>>> >>>> Yes, I can play the files manually on any player on my computer. >>>> I'm a freeswitch newbie, so I don't know how to change conference >>>> sampling rate, so I guess it's on default 8khz, same as the sound files. >>>> >>>> On 01/05/2012 11:54 a.m., Michael Collins wrote: >>>> >>>> Can you manually play any of the sound files? Also, what sampling rate >>>> is the conference and what sampling rate are the files? >>>> -MC >>>> >>>> On Mon, Apr 30, 2012 at 6:12 PM, Alejandro Mej?a wrote: >>>> >>>>> Hello, >>>>> >>>>> I installed a spanish sounds package to my Freeswitch under >>>>> /usr/local/freeswitch/sounds/es/mx/ directory >>>>> I'm trying it by entering an empty conference, and I only hear some >>>>> seconds of silence (where conf-alone.wav should be played), and then >>>>> the >>>>> music on hold. >>>>> I know the files are not missing, and Freeswitch is finding them >>>>> because >>>>> when I remove "/es" directory, I get error on CLI when executing a >>>>> conference, and getting directly to MOH. >>>>> >>>>> Any help will be appreciated. >>>>> >>>>> Thanks! >>>>> >>>>> Alex >>>>> >>>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com >>> >>> >>> >>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com >>> >>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 at freeswitch.orghttp://www.freeswitchsolutions.com >> >> >> >> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com >> >> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120502/0206f14d/attachment-0001.html From webudo at gua.net Thu May 3 02:13:43 2012 From: webudo at gua.net (=?ISO-8859-1?Q?Alejandro_Mej=EDa?=) Date: Wed, 02 May 2012 16:13:43 -0600 Subject: [Freeswitch-users] Silence instead of spanish wav installed In-Reply-To: References: <4F9F3895.7090503@gua.net> <4FA0A315.5020605@gua.net> <4FA1923B.3080807@gua.net> <4FA1A272.8030700@gua.net> <4FA1A66B.4080208@gua.net> <4FA1AA0E.9090409@gua.net> Message-ID: <4FA1B197.8080803@gua.net> Thanks man!!! :) You rock! I'll soxi all the files using a script and let you know. On 02/05/2012 03:55 p.m., Michael Collins wrote: > Well, since you paid these guys to give you the sounds, how about you > pay me for the answer! :P > > If you do a "soxi" on the sounds you'll see something like this: > racserv01:/usr/local/freeswitch/sounds# soxi conf-alone.wav > > Input File : 'conf-alone.wav' > Channels : 1 > Sample Rate : 8000 > Precision : 24-bit > Duration : 00:00:03.88 = 31009 samples ~ 290.709 CDDA sectors > File Size : 124k > Bit Rate : 256k > *Sample Encoding: 32-bit Floating Point PCM* > > The sample encoding is 32-bit FP. If you resample the file with sox it > will work. (I just tested it.): > > sox conf-alone.wav -r 8000 -e signed-integer -b 16 conf-alone-working.wav > > If you "soxi" it you should see something like this: > > racserv01:/usr/local/freeswitch/sounds# soxi conf-alone-working.wav > > Input File : 'conf-alone-working.wav' > Channels : 1 > Sample Rate : 8000 > Precision : 16-bit > Duration : 00:00:03.88 = 31009 samples ~ 290.709 CDDA sectors > File Size : 62.1k > Bit Rate : 128k > *Sample Encoding: 16-bit Signed Integer PCM* > > Give it a try. If it works ask them if they supply the files in 16 bit > signed integer format instead of 32 bit floating point. If not then > you can just sox-ify them all with a shell script. Ping me offline if > you run into any other trouble. > > -MC > > > On Wed, May 2, 2012 at 2:41 PM, Alejandro Mej?a > wrote: > > I know. I'm also not a fan of propietary, but I had to place > spanish recordings for one of my customers urgently, and I thought > that buying those would save me some trouble finding missing wav > files from open packages. > Maybe you can recommend me an open package for mexican or latin > american spanish that I can use instead of this one, ask for a > refund with this guys, and make a donation to the FS project > instead ;) > > > On 02/05/2012 03:30 p.m., Michael Collins wrote: >> Boo! Hiss! :P >> >> I'm not a fan of proprietary. Oh well. Tell you what - maybe >> since you paid them some $$ (or pesos or whatever) maybe they can >> provide you some support. :) Alternatively, you could contact >> them and let them know that the FS devs would like to >> troubleshoot an issue with the files and would like permission to >> receive a copy of the files /gratis/. >> >> Let me know what they say. >> -MC >> >> On Wed, May 2, 2012 at 2:26 PM, Alejandro Mej?a > > wrote: >> >> It's a commercial pack of voices bought from westany.com >> >> That's why I didn't include the file here, or a link to it, >> because I have not read their terms of usage and I don't want >> to get in trouble. >> You may get some samples here: >> http://www.westany.com/mexican_spanish_freeswitch_voice_prompts >> >> >> On 02/05/2012 03:19 p.m., Michael Collins wrote: >>> Where did you download these files? I would like to download >>> them myself and do some testing. >>> -MC >>> >>> On Wed, May 2, 2012 at 2:09 PM, Alejandro Mej?a >>> > wrote: >>> >>> Update finished, but the problem persists. >>> Perhaps it helps you to have the audio file being used? >>> (let me know and I can send it) >>> >>> On 02/05/2012 01:59 p.m., Alejandro Mej?a wrote: >>>> Log's output here: >>>> http://pastebin.freeswitch.org/pastebin.php?dl=18970 >>>> >>>> Audio remains silent during .wav file's length when >>>> reaching the line: >>>> 2012-05-02 13:13:22.728301 [INFO] switch_rtp.c:3164 >>>> Auto Changing port from 172.16.1.97:5016 >>>> to 190.149.242.46 >>>> :15046 >>>> >>>> Then it plays MOH normally. >>>> >>>> I'm using fusionpbx's canned version of FS, and I'm >>>> updating now to see if there was a bug that's been fixed. >>>> I'll let you know after the process completes. >>>> >>>> Thanks! >>>> >>>> Alex >>>> >>>> On 02/05/2012 09:36 a.m., Michael Collins wrote: >>>>> Okay, next step is log output. Get a console debug log >>>>> and drop it onto pastebin. Lots of useful information >>>>> available here: >>>>> >>>>> http://wiki.freeswitch.org/wiki/Reporting_Bugs >>>>> >>>>> -MC >>>>> >>>>> On Tue, May 1, 2012 at 7:59 PM, Alejandro Mej?a >>>>> > wrote: >>>>> >>>>> Thanks for your reply Michael. >>>>> >>>>> Yes, I can play the files manually on any player >>>>> on my computer. >>>>> I'm a freeswitch newbie, so I don't know how to >>>>> change conference sampling rate, so I guess it's >>>>> on default 8khz, same as the sound files. >>>>> >>>>> On 01/05/2012 11:54 a.m., Michael Collins wrote: >>>>>> Can you manually play any of the sound files? >>>>>> Also, what sampling rate is the conference and >>>>>> what sampling rate are the files? >>>>>> -MC >>>>>> >>>>>> On Mon, Apr 30, 2012 at 6:12 PM, Alejandro Mej?a >>>>>> > wrote: >>>>>> >>>>>> Hello, >>>>>> >>>>>> I installed a spanish sounds package to my >>>>>> Freeswitch under >>>>>> /usr/local/freeswitch/sounds/es/mx/ directory >>>>>> I'm trying it by entering an empty >>>>>> conference, and I only hear some >>>>>> seconds of silence (where conf-alone.wav >>>>>> should be played), and then the >>>>>> music on hold. >>>>>> I know the files are not missing, and >>>>>> Freeswitch is finding them because >>>>>> when I remove "/es" directory, I get error on >>>>>> CLI when executing a >>>>>> conference, and getting directly to MOH. >>>>>> >>>>>> Any help will be appreciated. >>>>>> >>>>>> Thanks! >>>>>> >>>>>> Alex >>>>>> >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at 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 at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120502/db86416e/attachment-0001.html From curriegrad2004 at gmail.com Thu May 3 02:35:51 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Wed, 2 May 2012 15:35:51 -0700 Subject: [Freeswitch-users] Installing FS to a non-standard directory In-Reply-To: References: <361E98F99D3CC3439EED59BC1924ED695ED1CD@SERVER2003.SecuReachSystems.local> <361E98F99D3CC3439EED59BC1924ED695ED1CE@SERVER2003.SecuReachSystems.local> Message-ID: Done, added on the wiki :P On Wed, May 2, 2012 at 12:57 PM, Michael Collins wrote: > > > On Wed, May 2, 2012 at 10:00 AM, Jason Moran > wrote: >> >> I think I?ve figured it out? >> >> ./configure --prefix=YOUR_NEW_PATH > > > This is correct. Let us know if you found this information on the wiki or if > you had to figure it out by some other means. (This is a perfect item that > should be on the wiki.) > > -MC > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From anton.jugatsu at gmail.com Thu May 3 08:38:22 2012 From: anton.jugatsu at gmail.com (Anton Kvashenkin) Date: Thu, 3 May 2012 08:38:22 +0400 Subject: [Freeswitch-users] preAnswer call at lua script and then bridge it to another leg In-Reply-To: References: Message-ID: Excellent answer, as always :) 3 ??? 2012 ?. 1:26 ???????????? Michael Collins ???????: > The best way to handle this would be to use the bridge application outside > of the Lua script. Go ahead and do the Lua script stuff w/ your database > and then construct your dialstring. Set that to a channel variable, like > "my_dialstring": > > session:setVariable('my_dialstring',session_call) > > And then let the Lua script exit. Then just add a bridge to your dialplan: > > > > As a rule of thumb: if you *can* do it in the dialplan then you probably > *should* do it in the dialplan. Use the scripting language for what it's > good at and use the dialplan for what it's good at. Lua is good for logic, > db lookups, and more complicated processing. The dialplan is awesome at > connecting call legs. The bridge app will do all the work for you so you > can focus on more important pursuits. > > -MC > > On Tue, May 1, 2012 at 8:17 PM, Nikolay Rogoshchenkov > wrote: > >> Hello, >> >> I have a dialplan where I get the incoming call to lua script >> >> >> >> >> . >> . >> . >> >> >> >> and then I have answering in answer.lua: >> =========================== >> session:preAnswer(); >> >> [...accessing to DB...] >> >> if (session:ready()) then >> repeat >> index = index + 1; >> session_call = >> string.format("[leg_timeout=30,ignore_early_media=true,hangup_after_bridge=true,origination_caller_id_number=".. >> cid .."]sofia/gateway/%s/%s",gwlist[index],dn); >> legB = freeswitch.Session(session_call); >> until ((hcause == 'USER_BUSY') or (hcause == 'SUCCESS') or (hcause == >> 'NO_ANSWER') or (index == #gwlist) or(session:ready() == false)) >> >> if (legB:ready()) then. >> freeswitch.bridge(session, legB) >> end >> =========================== >> >> But I have a one problem, when the initial legA was disconnected before >> legB was answered. It it still continued to call legB (during the >> leg_timeout). >> Could you please explain, how I can cancel legB right after I got >> disconnect on legA in this case? Thank you. >> -- >> Nikolay >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120503/7b49a7c4/attachment.html From nickolayr at gmail.com Thu May 3 05:25:05 2012 From: nickolayr at gmail.com (Nikolay Rogoshchenkov) Date: Wed, 2 May 2012 21:25:05 -0400 Subject: [Freeswitch-users] preAnswer call at lua script and then bridge it to another leg In-Reply-To: References: Message-ID: Thank you so much for your explanations! I will try this. -- Nikolay On Wed, May 2, 2012 at 5:26 PM, Michael Collins wrote: > The best way to handle this would be to use the bridge application outside > of the Lua script. Go ahead and do the Lua script stuff w/ your database > and then construct your dialstring. Set that to a channel variable, like > "my_dialstring": > > session:setVariable('my_dialstring',session_call) > > And then let the Lua script exit. Then just add a bridge to your dialplan: > > > > As a rule of thumb: if you *can* do it in the dialplan then you probably > *should* do it in the dialplan. Use the scripting language for what it's > good at and use the dialplan for what it's good at. Lua is good for logic, > db lookups, and more complicated processing. The dialplan is awesome at > connecting call legs. The bridge app will do all the work for you so you > can focus on more important pursuits. > > -MC > > On Tue, May 1, 2012 at 8:17 PM, Nikolay Rogoshchenkov > wrote: > >> Hello, >> >> I have a dialplan where I get the incoming call to lua script >> >> >> >> >> . >> . >> . >> >> >> >> and then I have answering in answer.lua: >> =========================== >> session:preAnswer(); >> >> [...accessing to DB...] >> >> if (session:ready()) then >> repeat >> index = index + 1; >> session_call = >> string.format("[leg_timeout=30,ignore_early_media=true,hangup_after_bridge=true,origination_caller_id_number=".. >> cid .."]sofia/gateway/%s/%s",gwlist[index],dn); >> legB = freeswitch.Session(session_call); >> until ((hcause == 'USER_BUSY') or (hcause == 'SUCCESS') or (hcause == >> 'NO_ANSWER') or (index == #gwlist) or(session:ready() == false)) >> >> if (legB:ready()) then. >> freeswitch.bridge(session, legB) >> end >> =========================== >> >> But I have a one problem, when the initial legA was disconnected before >> legB was answered. It it still continued to call legB (during the >> leg_timeout). >> Could you please explain, how I can cancel legB right after I got >> disconnect on legA in this case? Thank you. >> -- >> Nikolay >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120502/e6ed5e42/attachment.html From nestor at tiendalinux.com Thu May 3 10:13:48 2012 From: nestor at tiendalinux.com (Nestor A Diaz) Date: Thu, 03 May 2012 01:13:48 -0500 Subject: [Freeswitch-users] Question about digit-len and inter-digit-timeout on ivr Message-ID: <4FA2221C.50009@tiendalinux.com> Hi Guys. Just by curiosity: Let suppose i have an ivr with just two options: * digit = '1' * digit = '/^2[0-9]$/ with digit-len = 2 and inter-digit-timeout = 5000 If i press 1 i have to wait inter-digit-timeout to perform the action, however there will be no 1[0-9] so the system can perform the action without having to wait inter-digit-timeout time. Is there any way to change digit-len to something like 'dynamic' ? so this way everytime the ivr get a digit it will compare against possible choices and if there will be just one that match then it will perform the action inmediately. Slds. -- Nestor A. Diaz nestor at tiendalinux.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120503/2d98536f/attachment-0001.html From anita.hall at simmortel.com Thu May 3 11:05:33 2012 From: anita.hall at simmortel.com (Anita Hall) Date: Thu, 3 May 2012 12:35:33 +0530 Subject: [Freeswitch-users] spandsp consultant T.30 In-Reply-To: <-2375528179531511362@unknownmsgid> References: <-2375528179531511362@unknownmsgid> Message-ID: No Brian, we are doing T.30 over E1/PRI. We are getting good results around 70% but not good enough :) On 5/1/12, Brian West wrote: > Are you doing t.38? > > Sent from my iPhone > > On May 1, 2012, at 5:03 AM, Anita Hall wrote: > >> Hi >> >> One of our clients is getting around 70% results using mod_spandsp on >> FreeSWITCH. The protocol is T.30 and the client is India. They are looking >> for someone who could help them to increase the results to more than 95%. >> >> If you or someone you know could help, please let me know. >> >> Thanks! >> >> regards, >> Anita >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- regards, Anita From anita.hall at simmortel.com Thu May 3 11:08:52 2012 From: anita.hall at simmortel.com (Anita Hall) Date: Thu, 3 May 2012 12:38:52 +0530 Subject: [Freeswitch-users] Commercial Fax over T.30 for FreeSWITCH Message-ID: Hi I am looking for a commercial product for Incoming Fax over T.30 for FreeSWITCH. Like Commetrex is available for Asterisk, is there something for freeswitch ? regards, Anita -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120503/e92cd218/attachment.html From acrow at integrafin.co.uk Thu May 3 12:25:36 2012 From: acrow at integrafin.co.uk (Alex Crow) Date: Thu, 03 May 2012 09:25:36 +0100 Subject: [Freeswitch-users] Commercial Fax over T.30 for FreeSWITCH In-Reply-To: References: Message-ID: <4FA24100.7040908@integrafin.co.uk> On 03/05/12 08:08, Anita Hall wrote: > Hi > > I am looking for a commercial product for Incoming Fax over T.30 for > FreeSWITCH. > > Like Commetrex is available for Asterisk, is there something for > freeswitch ? > > regards, > Anita > Hi, I use Hylafax with T38Modem (and gateway to T.30 in Freeswitch). Hylafax is as good as any commercial product, and I get at least 99.5% success using it with onboard analog modems. I've not sent or received hundreds of faxes over T38modem but I don't see why it should be any worse. Just one question - are you running Freeswitch on a VM? Fax is very timing critical and I found running on a VM caused an awful lot of failures for T.30. For outbound: For inbound: Cheers Alex From anita.hall at simmortel.com Thu May 3 16:23:01 2012 From: anita.hall at simmortel.com (Anita Hall) Date: Thu, 3 May 2012 17:53:01 +0530 Subject: [Freeswitch-users] Commercial Fax over T.30 for FreeSWITCH In-Reply-To: <4FA24100.7040908@integrafin.co.uk> References: <4FA24100.7040908@integrafin.co.uk> Message-ID: Hi Alex Thanks for the tip :) No, I am running FreeSWITCH on an Ubuntu 64-bit Quad core machine. There is no VM. Is it possible to use Hylafax instead of or in combination to FreeSWITCH if I am using Sangoma E1 Cards ? What all things should I check to ensure that the timing on my system is good? Like, may be, my motherboard is bad or my kernel could be tuned better ? Cheers! regards, Anita On Thu, May 3, 2012 at 1:55 PM, Alex Crow wrote: > On 03/05/12 08:08, Anita Hall wrote: > > Hi > > > > I am looking for a commercial product for Incoming Fax over T.30 for > > FreeSWITCH. > > > > Like Commetrex is available for Asterisk, is there something for > > freeswitch ? > > > > regards, > > Anita > > > > Hi, > > I use Hylafax with T38Modem (and gateway to T.30 in Freeswitch). Hylafax > is as good as any commercial product, and I get at least 99.5% success > using it with onboard analog modems. I've not sent or received hundreds > of faxes over T38modem but I don't see why it should be any worse. > > Just one question - are you running Freeswitch on a VM? Fax is very > timing critical and I found running on a VM caused an awful lot of > failures for T.30. > > > For outbound: > > > > > > > > > > > > For inbound: > > > > > > > > > > Cheers > > Alex > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120503/91563b8e/attachment.html From avi at avimarcus.net Thu May 3 17:16:31 2012 From: avi at avimarcus.net (Avi Marcus) Date: Thu, 3 May 2012 16:16:31 +0300 Subject: [Freeswitch-users] Javascript - new V8 Engine? Message-ID: I know that lua has better performance and "embeddability" than any other language for IVRs and other scripts. If mod_javascript would be upgraded to use V8, would that make using javascript on par with Lua? If so.. what size bounty would push it along? Thanks! -Avi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120503/2a50f779/attachment.html From gmaruzz at gmail.com Thu May 3 18:38:05 2012 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Thu, 3 May 2012 16:38:05 +0200 Subject: [Freeswitch-users] can't build mod_gsmopen ?! In-Reply-To: References: <4F9C2F0F.9090008@googlemail.com> Message-ID: On Sat, Apr 28, 2012 at 8:21 PM, Anton Kvashenkin wrote: > Giovanni, hi. Pardon me, but can you checkout a few issues that I reported > at jira about gsmopen. > Hi Anton, sorry for taking so much time, but I fixed it almost immediately, then I bashed my head in the wal for one week trying to understand what was wrong now, that I was having no audio in my snom320, only when the flow was gsmopen->FS->snom. I was tracing all things, no clue. Today I testing with X-lite, with hope to maybe get a hint, and it worked perfectly. To make a loooong story short, was a bug in the snom320 firmware (unrelated with the problems you reported. Was not correctly advertising its codecs when in a transfer). Updated, works flawlessly. :) Anyway, the bugs you reported are fixed, please report more bugs -giovanni -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From steveu at coppice.org Thu May 3 18:56:35 2012 From: steveu at coppice.org (Steve Underwood) Date: Thu, 03 May 2012 22:56:35 +0800 Subject: [Freeswitch-users] Commercial Fax over T.30 for FreeSWITCH In-Reply-To: References: <4FA24100.7040908@integrafin.co.uk> Message-ID: <4FA29CA3.1020000@coppice.org> Hi Anita, What are you hoping to get from a commercial solution? You won't get improved reliability. If you go with the Digium solution support seems to be limited to an offer of a refund. I have helped a number of FAX for Asterisk users move to spandsp to improve their performance. What are you actually doing? Are you simply sending and receiving FAXes on a Sangoma E1 card? If that gives you anything less than 99.5% success, something is wrong in your setup. A timing problem with the E1 is the most likely cause. Not the motherboard or the kernel. The E1 interface itself. You MUST make the PSTN the source of the clock for the E1. If you make your card the source you WILL have serious problems with reliability. Using T.38 gateway operation to interwork between the PSTN and VoIP can give additional problems, as some system do strange things with the signaling, or don't open the right ports when the call switches from RTP to UDPTL. If those things are sorted out, and your IP interface is not losing lots of packets, you should have few failures. Steve On 05/03/2012 08:23 PM, Anita Hall wrote: > Hi Alex > > Thanks for the tip :) > > No, I am running FreeSWITCH on an Ubuntu 64-bit Quad core machine. > There is no VM. > > Is it possible to use Hylafax instead of or in combination to > FreeSWITCH if I am using Sangoma E1 Cards ? > > What all things should I check to ensure that the timing on my system > is good? Like, may be, my motherboard is bad or my kernel could be > tuned better ? > > Cheers! > > regards, > Anita > > > > On Thu, May 3, 2012 at 1:55 PM, Alex Crow > wrote: > > On 03/05/12 08:08, Anita Hall wrote: > > Hi > > > > I am looking for a commercial product for Incoming Fax over T.30 for > > FreeSWITCH. > > > > Like Commetrex is available for Asterisk, is there something for > > freeswitch ? > > > > regards, > > Anita > > > > Hi, > > I use Hylafax with T38Modem (and gateway to T.30 in Freeswitch). > Hylafax > is as good as any commercial product, and I get at least 99.5% success > using it with onboard analog modems. I've not sent or received > hundreds > of faxes over T38modem but I don't see why it should be any worse. > > Just one question - are you running Freeswitch on a VM? Fax is very > timing critical and I found running on a VM caused an awful lot of > failures for T.30. > > > For outbound: > > > > > > > > > > > > For inbound: > > > > > > > > > > Cheers > > Alex > From paul at cupis.co.uk Thu May 3 19:19:46 2012 From: paul at cupis.co.uk (Paul Cupis) Date: Thu, 3 May 2012 16:19:46 +0100 Subject: [Freeswitch-users] Commercial Fax over T.30 for FreeSWITCH In-Reply-To: References: <4FA24100.7040908@integrafin.co.uk> Message-ID: <20120503151946.GA824@eagle.cupis.co.uk> On Thu, May 03, 2012 at 05:53:01PM +0530, Anita Hall wrote: > No, I am running FreeSWITCH on an Ubuntu 64-bit Quad core machine. > There is no VM. > What all things should I check to ensure that the timing on my system > is good? Like, may be, my motherboard is bad or my kernel could be > tuned better ? The most common suggestion is to use a kernel with 1000Hz timer. I think that the default timer for Ubuntu is 250, so this would be first thing to check/change. You can check by running: grep CONFIG_HZ= /boot/config-`uname -r` and if it is not 1000, it would be worth installing a 1000Hz kernel and re-testing. Ubuntu may have a packages -lowlatency kernel which has a 1000Hz timer but this will also have other changes. I think a lot of people will compile a custom kernel based on the standard Ubuntu one, just changing that one configuration option. Regards, From anthony.minessale at gmail.com Fri May 4 00:29:09 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 3 May 2012 15:29:09 -0500 Subject: [Freeswitch-users] Commercial Fax over T.30 for FreeSWITCH In-Reply-To: <20120503151946.GA824@eagle.cupis.co.uk> References: <4FA24100.7040908@integrafin.co.uk> <20120503151946.GA824@eagle.cupis.co.uk> Message-ID: Once you get to a version of FS that supports timerfd there is less issue with kernel freq. Also, more people should try the new t30 modem emulation built right into FS to work with hlyafax see spandsp.conf.xml one word of caution is don't leave (FS off and the faxgettys on) very long or the system will start using the ttty's for ssh sessions etc and faxgetty will start sending at commands to your stdin. On Thu, May 3, 2012 at 10:19 AM, Paul Cupis wrote: > On Thu, May 03, 2012 at 05:53:01PM +0530, Anita Hall wrote: >> ? ?No, I am running FreeSWITCH on an Ubuntu 64-bit Quad core machine. >> ? ?There is no VM. > >> ? ?What all things should I check to ensure that the timing on my system >> ? ?is good? Like, may be, my motherboard is bad or my kernel could be >> ? ?tuned better ? > > The most common suggestion is to use a kernel with 1000Hz timer. I think > that the default timer for Ubuntu is 250, so this would be first thing > to check/change. > > You can check by running: > > ? ? grep CONFIG_HZ= /boot/config-`uname -r` > > and if it is not 1000, it would be worth installing a 1000Hz kernel > and re-testing. > > Ubuntu may have a packages -lowlatency kernel which has a 1000Hz timer > but this will also have other changes. I think a lot of people will > compile a custom kernel based on the standard Ubuntu one, just changing > that one configuration option. > > Regards, > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From koralu at gmail.com Fri May 4 00:55:35 2012 From: koralu at gmail.com (Cucu Florian) Date: Thu, 3 May 2012 23:55:35 +0300 Subject: [Freeswitch-users] Early media issue Message-ID: Hello, I have the following problem regarding early media. When I call from Xlite(extension 1000) to one of my SIP endpoint(extension 1001) I hear a fake long ring instead of media. Both endpoints accept G711ulaw and alaw codec. I think that 1001 send an 183 SIP response that is misinterpret by FS or Xlite. Is any solution to hear in xlite the correct media that 1001 send it? Thanks in advance. My Freeswitch logs are: 2012-05-03 23:40:25.027591 [INFO] mod_dialplan_xml.c:485 Processing 1000 <1>1001 in context default 2012-05-03 23:40:25.050838 [INFO] switch_ivr_async.c:3195 Bound B-Leg: *1 ee_extension::dx XML features 2012-05-03 23:40:25.050838 [INFO] switch_ivr_async.c:3195 Bound B-Leg: *2 r_session::/usr/local/freeswitch/recordings/1000.2012-05-03-23-40-25.wav 2012-05-03 23:40:25.067389 [INFO] switch_ivr_async.c:3195 Bound B-Leg: *3 ee_extension::cf XML features 2012-05-03 23:40:25.067389 [INFO] switch_ivr_async.c:3195 Bound B-Leg: *4 ee_extension::att_xfer XML features 2012-05-03 23:40:25.087371 [NOTICE] switch_channel.c:926 New Channel sofia/nal/sip:1001 at 192.168.1.69:5060 [35ddd1a6-9560-11e1-8f20-dd25b25b02e6] 2012-05-03 23:40:25.187367 [NOTICE] sofia.c:5633 Pre-Answer sofia/internal/ 001 at 192.168.1.69:5060! *2012-05-03 23:40:25.210095 [INFO] switch_ivr.c:725 Sending early media* *2012-05-03 23:40:25.210095 [NOTICE] mod_sofia.c:2578 Pre-Answer sofia/ inter000 at 192.168.1.10!* 2012-05-03 23:40:56.610515 [NOTICE] sofia.c:6293 Hangup sofia/internal/ sip:1001 at 192.168.1.69:5060 [CS_HIBERNATE] [NORMAL_TEMPORARY_FAILURE] 2012-05-03 23:40:56.610515 [NOTICE] switch_core_session.c:1398 Session 22 (sofia/internal/sip:1001 at 192.168.1.69:5060) Ended 2012-05-03 23:40:56.610515 [NOTICE] switch_core_session.c:1400 Close Channel sofia/internal/sip:1001 at 192.168.1.69:5060 [CS_DESTROY] 2012-05-03 23:40:56.610515 [NOTICE] mod_dptools.c:1135 Channel [sofia/internal/1000 at 192.168.1.10] has been answered 2012-05-03 23:40:57.627386 [NOTICE] switch_channel.c:926 New Channel loopback/app=voicemail:default 192.168.1.10 1001-a [494485e6-9560-11e1-8f2c-dd25b25b02e6] 2012-05-03 23:40:57.627386 [NOTICE] switch_channel.c:924 Rename Channel loopback/app=voicemail:default 192.168.1.10 1001-a->loopback/voicemail-a [494485e6-9560-11e1-8f2c-dd25b25b02e6] 2012-05-03 23:40:57.647627 [NOTICE] switch_channel.c:926 New Channel loopback/voicemail-b [49452e10-9560-11e1-8f30-dd25b25b02e6] 2012-05-03 23:40:57.647627 [NOTICE] mod_loopback.c:760 Pre-Answer loopback/voicemail-a! 2012-05-03 23:40:57.647627 [NOTICE] mod_dptools.c:1161 Pre-Answer loopback/voicemail-b! *2012-05-03 23:40:57.647627 [WARNING] switch_ivr_bridge.c:246 Cannot bypass media while bridged to a loopback address.* *2012-05-03 23:40:57.647627 [ERR] switch_core_io.c:131 sofia/internal/ 1000 at 192.168.1.10 has no read codec.* 2012-05-03 23:40:57.647627 [NOTICE] switch_core_io.c:132 Hangup sofia/internal/1000 at 192.168.1.10 [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] 2012-05-03 23:40:57.667365 [NOTICE] switch_ivr_bridge.c:667 Hangup loopback/voicemail-a [CS_EXCHANGE_MEDIA] [ORIGINATOR_CANCEL] 2012-05-03 23:40:57.667365 [NOTICE] mod_loopback.c:438 Hangup loopback/voicemail-b [CS_EXECUTE] [ORIGINATOR_CANCEL] 2012-05-03 23:40:57.667365 [NOTICE] switch_core_session.c:1398 Session 21 (sofia/internal/1000 at 192.168.1.10) Ended 2012-05-03 23:40:57.667365 [NOTICE] switch_core_session.c:1400 Close Channel sofia/internal/1000 at 192.168.1.10 [CS_DESTROY] 2012-05-03 23:40:57.719451 [WARNING] switch_ivr_play_say.c:339 Macro [voicemail_play_greeting]: '1001' did not match any patterns 2012-05-03 23:40:57.719451 [NOTICE] switch_core_session.c:1398 Session 23 (loopback/voicemail-a) Ended 2012-05-03 23:40:57.719451 [NOTICE] switch_core_session.c:1400 Close Channel loopback/voicemail-a [CS_DESTROY] 2012-05-03 23:40:57.719451 [NOTICE] switch_core_session.c:1398 Session 24 (loopback/voicemail-b) Ended 2012-05-03 23:40:57.719451 [NOTICE] switch_core_session.c:1400 Close Channel loopback/voicemail-b [CS_DESTROY] -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120503/48bae182/attachment.html From neil.stirton at ikiji.com Fri May 4 01:33:35 2012 From: neil.stirton at ikiji.com (Neil Stirton) Date: Thu, 03 May 2012 22:33:35 +0100 Subject: [Freeswitch-users] Call forwarding to mobile showing our trunk CLI instead of caller CLI Message-ID: Hi, Apologies for the noob question here. Setup Freeswitch on CentOS and using FusionPBX for 'easy' management. I've defined a hunt group (ext 7001) which simultaneously rings 3 internal SIP extensions + my mobile. All the internal extensions see the caller ID no problem but because I am also sending the call back out through our ITSP's trunk, the call always shows as our office calling my mobile phone rather than the actual caller's ID. I have the hunt group calling: Sip uri sofia/gateway// I'm sure it's a case of a variable setting something but if someone could point me in the right direction that would be very much appreciated. Many thanks Neil -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120503/d0d428d9/attachment.html From avi at avimarcus.net Fri May 4 02:20:44 2012 From: avi at avimarcus.net (Avi Marcus) Date: Fri, 4 May 2012 01:20:44 +0300 Subject: [Freeswitch-users] Call forwarding to mobile showing our trunk CLI instead of caller CLI In-Reply-To: References: Message-ID: Depending on your trunk provider, they might not allow you to change your outbound caller ID. It might be hard-coded for your account. If not.. it could be you need to set "caller-id-in-from" to be true within the gateway and reload the gateway. -Avi On Fri, May 4, 2012 at 12:33 AM, Neil Stirton wrote: > Hi, > > Apologies for the noob question here. > > Setup Freeswitch on CentOS and using FusionPBX for 'easy' management. > > I've defined a hunt group (ext 7001) which simultaneously rings 3 internal > SIP extensions + my mobile. > > All the internal extensions see the caller ID no problem but because I am > also sending the call back out through our ITSP's trunk, the call always > shows as our office calling my mobile phone rather than the actual caller's > ID. > I have the hunt group calling: > > Sip uri sofia/gateway// > > I'm sure it's a case of a variable setting something but if someone could > point me in the right direction that would be very much appreciated. > > Many thanks > Neil > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120504/362d8254/attachment-0001.html From msc at freeswitch.org Fri May 4 02:49:30 2012 From: msc at freeswitch.org (Michael Collins) Date: Thu, 3 May 2012 15:49:30 -0700 Subject: [Freeswitch-users] Early media issue In-Reply-To: References: Message-ID: I think we need to see more information, like the entire call log from start to finish. Use pastebin.freeswitch.org and use "FreeSWITCH Log" for syntax highlighting. -MC On Thu, May 3, 2012 at 1:55 PM, Cucu Florian wrote: > Hello, > > I have the following problem regarding early media. > When I call from Xlite(extension 1000) to one of my SIP endpoint(extension > 1001) I hear a fake long ring instead of media. > Both endpoints accept G711ulaw and alaw codec. I think that 1001 send an > 183 SIP response that is misinterpret by FS or Xlite. Is any solution to > hear in xlite the correct media that 1001 send it? > > Thanks in advance. > > > My Freeswitch logs are: > > 2012-05-03 23:40:25.027591 [INFO] mod_dialplan_xml.c:485 Processing 1000 > <1>1001 in context default > 2012-05-03 23:40:25.050838 [INFO] switch_ivr_async.c:3195 Bound B-Leg: *1 > ee_extension::dx XML features > 2012-05-03 23:40:25.050838 [INFO] switch_ivr_async.c:3195 Bound B-Leg: *2 > r_session::/usr/local/freeswitch/recordings/1000.2012-05-03-23-40-25.wav > 2012-05-03 23:40:25.067389 [INFO] switch_ivr_async.c:3195 Bound B-Leg: *3 > ee_extension::cf XML features > 2012-05-03 23:40:25.067389 [INFO] switch_ivr_async.c:3195 Bound B-Leg: *4 > ee_extension::att_xfer XML features > 2012-05-03 23:40:25.087371 [NOTICE] switch_channel.c:926 New Channel > sofia/nal/sip:1001 at 192.168.1.69:5060[35ddd1a6-9560-11e1-8f20-dd25b25b02e6] > 2012-05-03 23:40:25.187367 [NOTICE] sofia.c:5633 Pre-Answer sofia/internal/ > 001 at 192.168.1.69:5060! > *2012-05-03 23:40:25.210095 [INFO] switch_ivr.c:725 Sending early media* > *2012-05-03 23:40:25.210095 [NOTICE] mod_sofia.c:2578 Pre-Answer sofia/ > inter000 at 192.168.1.10!* > 2012-05-03 23:40:56.610515 [NOTICE] sofia.c:6293 Hangup sofia/internal/ > sip:1001 at 192.168.1.69:5060 [CS_HIBERNATE] [NORMAL_TEMPORARY_FAILURE] > 2012-05-03 23:40:56.610515 [NOTICE] switch_core_session.c:1398 Session 22 > (sofia/internal/sip:1001 at 192.168.1.69:5060) Ended > 2012-05-03 23:40:56.610515 [NOTICE] switch_core_session.c:1400 Close > Channel sofia/internal/sip:1001 at 192.168.1.69:5060 [CS_DESTROY] > 2012-05-03 23:40:56.610515 [NOTICE] mod_dptools.c:1135 Channel > [sofia/internal/1000 at 192.168.1.10] has been answered > 2012-05-03 23:40:57.627386 [NOTICE] switch_channel.c:926 New Channel > loopback/app=voicemail:default 192.168.1.10 1001-a > [494485e6-9560-11e1-8f2c-dd25b25b02e6] > 2012-05-03 23:40:57.627386 [NOTICE] switch_channel.c:924 Rename Channel > loopback/app=voicemail:default 192.168.1.10 1001-a->loopback/voicemail-a > [494485e6-9560-11e1-8f2c-dd25b25b02e6] > 2012-05-03 23:40:57.647627 [NOTICE] switch_channel.c:926 New Channel > loopback/voicemail-b [49452e10-9560-11e1-8f30-dd25b25b02e6] > 2012-05-03 23:40:57.647627 [NOTICE] mod_loopback.c:760 Pre-Answer > loopback/voicemail-a! > 2012-05-03 23:40:57.647627 [NOTICE] mod_dptools.c:1161 Pre-Answer > loopback/voicemail-b! > *2012-05-03 23:40:57.647627 [WARNING] switch_ivr_bridge.c:246 Cannot > bypass media while bridged to a loopback address.* > *2012-05-03 23:40:57.647627 [ERR] switch_core_io.c:131 sofia/internal/ > 1000 at 192.168.1.10 has no read codec.* > 2012-05-03 23:40:57.647627 [NOTICE] switch_core_io.c:132 Hangup > sofia/internal/1000 at 192.168.1.10 [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] > 2012-05-03 23:40:57.667365 [NOTICE] switch_ivr_bridge.c:667 Hangup > loopback/voicemail-a [CS_EXCHANGE_MEDIA] [ORIGINATOR_CANCEL] > 2012-05-03 23:40:57.667365 [NOTICE] mod_loopback.c:438 Hangup > loopback/voicemail-b [CS_EXECUTE] [ORIGINATOR_CANCEL] > 2012-05-03 23:40:57.667365 [NOTICE] switch_core_session.c:1398 Session 21 > (sofia/internal/1000 at 192.168.1.10) Ended > 2012-05-03 23:40:57.667365 [NOTICE] switch_core_session.c:1400 Close > Channel sofia/internal/1000 at 192.168.1.10 [CS_DESTROY] > 2012-05-03 23:40:57.719451 [WARNING] switch_ivr_play_say.c:339 Macro > [voicemail_play_greeting]: '1001' did not match any patterns > 2012-05-03 23:40:57.719451 [NOTICE] switch_core_session.c:1398 Session 23 > (loopback/voicemail-a) Ended > 2012-05-03 23:40:57.719451 [NOTICE] switch_core_session.c:1400 Close > Channel loopback/voicemail-a [CS_DESTROY] > 2012-05-03 23:40:57.719451 [NOTICE] switch_core_session.c:1398 Session 24 > (loopback/voicemail-b) Ended > 2012-05-03 23:40:57.719451 [NOTICE] switch_core_session.c:1400 Close > Channel loopback/voicemail-b [CS_DESTROY] > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120503/7bedb5b3/attachment.html From freeswitch at earthspike.net Fri May 4 03:04:14 2012 From: freeswitch at earthspike.net (John) Date: Fri, 04 May 2012 00:04:14 +0100 Subject: [Freeswitch-users] Commercial Fax over T.30 for FreeSWITCH In-Reply-To: <20120503151946.GA824@eagle.cupis.co.uk> References: <4FA24100.7040908@integrafin.co.uk> <20120503151946.GA824@eagle.cupis.co.uk> Message-ID: <4FA30EEE.7090702@earthspike.net> Ubuntu default timer for the server kernel is 100Hz (although I haven't checked 12.04). You need the -preempt kernel for low latency and 1000Hz. It's easy to apt-get install; you then need to tweak the GRUB config to make it your default kernel on (re)boot. John On 03/05/12 16:19, Paul Cupis wrote: > On Thu, May 03, 2012 at 05:53:01PM +0530, Anita Hall wrote: >> No, I am running FreeSWITCH on an Ubuntu 64-bit Quad core machine. >> There is no VM. >> What all things should I check to ensure that the timing on my system >> is good? Like, may be, my motherboard is bad or my kernel could be >> tuned better ? > The most common suggestion is to use a kernel with 1000Hz timer. I think > that the default timer for Ubuntu is 250, so this would be first thing > to check/change. > > You can check by running: > > grep CONFIG_HZ= /boot/config-`uname -r` > > and if it is not 1000, it would be worth installing a 1000Hz kernel > and re-testing. > > Ubuntu may have a packages -lowlatency kernel which has a 1000Hz timer > but this will also have other changes. I think a lot of people will > compile a custom kernel based on the standard Ubuntu one, just changing > that one configuration option. > > Regards, > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From anton.jugatsu at gmail.com Fri May 4 10:53:11 2012 From: anton.jugatsu at gmail.com (Anton Kvashenkin) Date: Fri, 4 May 2012 10:53:11 +0400 Subject: [Freeswitch-users] Early media issue In-Reply-To: References: Message-ID: Also don't forget to play with fs_logger.pl (search wiki). 2012/5/4 Michael Collins > I think we need to see more information, like the entire call log from > start to finish. Use pastebin.freeswitch.org and use "FreeSWITCH Log" for > syntax highlighting. > > -MC > > On Thu, May 3, 2012 at 1:55 PM, Cucu Florian wrote: > >> Hello, >> >> I have the following problem regarding early media. >> When I call from Xlite(extension 1000) to one of my SIP >> endpoint(extension 1001) I hear a fake long ring instead of media. >> Both endpoints accept G711ulaw and alaw codec. I think that 1001 send an >> 183 SIP response that is misinterpret by FS or Xlite. Is any solution to >> hear in xlite the correct media that 1001 send it? >> >> Thanks in advance. >> >> >> My Freeswitch logs are: >> >> 2012-05-03 23:40:25.027591 [INFO] mod_dialplan_xml.c:485 Processing 1000 >> <1>1001 in context default >> 2012-05-03 23:40:25.050838 [INFO] switch_ivr_async.c:3195 Bound B-Leg: *1 >> ee_extension::dx XML features >> 2012-05-03 23:40:25.050838 [INFO] switch_ivr_async.c:3195 Bound B-Leg: *2 >> r_session::/usr/local/freeswitch/recordings/1000.2012-05-03-23-40-25.wav >> 2012-05-03 23:40:25.067389 [INFO] switch_ivr_async.c:3195 Bound B-Leg: *3 >> ee_extension::cf XML features >> 2012-05-03 23:40:25.067389 [INFO] switch_ivr_async.c:3195 Bound B-Leg: *4 >> ee_extension::att_xfer XML features >> 2012-05-03 23:40:25.087371 [NOTICE] switch_channel.c:926 New Channel >> sofia/nal/sip:1001 at 192.168.1.69:5060[35ddd1a6-9560-11e1-8f20-dd25b25b02e6] >> 2012-05-03 23:40:25.187367 [NOTICE] sofia.c:5633 Pre-Answer >> sofia/internal/001 at 192.168.1.69:5060! >> *2012-05-03 23:40:25.210095 [INFO] switch_ivr.c:725 Sending early media* >> *2012-05-03 23:40:25.210095 [NOTICE] mod_sofia.c:2578 Pre-Answer sofia/ >> inter000 at 192.168.1.10!* >> 2012-05-03 23:40:56.610515 [NOTICE] sofia.c:6293 Hangup sofia/internal/ >> sip:1001 at 192.168.1.69:5060 [CS_HIBERNATE] [NORMAL_TEMPORARY_FAILURE] >> 2012-05-03 23:40:56.610515 [NOTICE] switch_core_session.c:1398 Session 22 >> (sofia/internal/sip:1001 at 192.168.1.69:5060) Ended >> 2012-05-03 23:40:56.610515 [NOTICE] switch_core_session.c:1400 Close >> Channel sofia/internal/sip:1001 at 192.168.1.69:5060 [CS_DESTROY] >> 2012-05-03 23:40:56.610515 [NOTICE] mod_dptools.c:1135 Channel >> [sofia/internal/1000 at 192.168.1.10] has been answered >> 2012-05-03 23:40:57.627386 [NOTICE] switch_channel.c:926 New Channel >> loopback/app=voicemail:default 192.168.1.10 1001-a >> [494485e6-9560-11e1-8f2c-dd25b25b02e6] >> 2012-05-03 23:40:57.627386 [NOTICE] switch_channel.c:924 Rename Channel >> loopback/app=voicemail:default 192.168.1.10 1001-a->loopback/voicemail-a >> [494485e6-9560-11e1-8f2c-dd25b25b02e6] >> 2012-05-03 23:40:57.647627 [NOTICE] switch_channel.c:926 New Channel >> loopback/voicemail-b [49452e10-9560-11e1-8f30-dd25b25b02e6] >> 2012-05-03 23:40:57.647627 [NOTICE] mod_loopback.c:760 Pre-Answer >> loopback/voicemail-a! >> 2012-05-03 23:40:57.647627 [NOTICE] mod_dptools.c:1161 Pre-Answer >> loopback/voicemail-b! >> *2012-05-03 23:40:57.647627 [WARNING] switch_ivr_bridge.c:246 Cannot >> bypass media while bridged to a loopback address.* >> *2012-05-03 23:40:57.647627 [ERR] switch_core_io.c:131 sofia/internal/ >> 1000 at 192.168.1.10 has no read codec.* >> 2012-05-03 23:40:57.647627 [NOTICE] switch_core_io.c:132 Hangup >> sofia/internal/1000 at 192.168.1.10 [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] >> 2012-05-03 23:40:57.667365 [NOTICE] switch_ivr_bridge.c:667 Hangup >> loopback/voicemail-a [CS_EXCHANGE_MEDIA] [ORIGINATOR_CANCEL] >> 2012-05-03 23:40:57.667365 [NOTICE] mod_loopback.c:438 Hangup >> loopback/voicemail-b [CS_EXECUTE] [ORIGINATOR_CANCEL] >> 2012-05-03 23:40:57.667365 [NOTICE] switch_core_session.c:1398 Session 21 >> (sofia/internal/1000 at 192.168.1.10) Ended >> 2012-05-03 23:40:57.667365 [NOTICE] switch_core_session.c:1400 Close >> Channel sofia/internal/1000 at 192.168.1.10 [CS_DESTROY] >> 2012-05-03 23:40:57.719451 [WARNING] switch_ivr_play_say.c:339 Macro >> [voicemail_play_greeting]: '1001' did not match any patterns >> 2012-05-03 23:40:57.719451 [NOTICE] switch_core_session.c:1398 Session 23 >> (loopback/voicemail-a) Ended >> 2012-05-03 23:40:57.719451 [NOTICE] switch_core_session.c:1400 Close >> Channel loopback/voicemail-a [CS_DESTROY] >> 2012-05-03 23:40:57.719451 [NOTICE] switch_core_session.c:1398 Session 24 >> (loopback/voicemail-b) Ended >> 2012-05-03 23:40:57.719451 [NOTICE] switch_core_session.c:1400 Close >> Channel loopback/voicemail-b [CS_DESTROY] >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120504/a382ff58/attachment-0001.html From neil.stirton at ikiji.com Fri May 4 04:04:10 2012 From: neil.stirton at ikiji.com (Neil Stirton) Date: Fri, 04 May 2012 01:04:10 +0100 Subject: [Freeswitch-users] Call forwarding to mobile showing our trunk CLI instead of caller CLI In-Reply-To: Message-ID: Hi Avi, Set that to true but feel like I need to be passing through some more info when bridging the call? You are most likely right that the trunk is controls what number is displayed on outbound calls ? this I will check also. Many thanks, Neil From: Avi Marcus Reply-To: FreeSWITCH Users Help Date: Thursday, 3 May 2012 23:20 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Call forwarding to mobile showing our trunk CLI instead of caller CLI Depending on your trunk provider, they might not allow you to change your outbound caller ID. It might be hard-coded for your account. If not.. it could be you need to set "caller-id-in-from" to be true within the gateway and reload the gateway. -Avi On Fri, May 4, 2012 at 12:33 AM, Neil Stirton wrote: > Hi, > > Apologies for the noob question here. > > Setup Freeswitch on CentOS and using FusionPBX for 'easy' management. > > I've defined a hunt group (ext 7001) which simultaneously rings 3 internal SIP > extensions + my mobile. > > All the internal extensions see the caller ID no problem but because I am also > sending the call back out through our ITSP's trunk, the call always shows as > our office calling my mobile phone rather than the actual caller's ID. > I have the hunt group calling: > > Sip uri sofia/gateway// > > I'm sure it's a case of a variable setting something but if someone could > point me in the right direction that would be very much appreciated. > > Many thanks > Neil > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120504/9b00a853/attachment.html From avi at avimarcus.net Fri May 4 11:42:36 2012 From: avi at avimarcus.net (Avi Marcus) Date: Fri, 4 May 2012 10:42:36 +0300 Subject: [Freeswitch-users] Call forwarding to mobile showing our trunk CLI instead of caller CLI In-Reply-To: References: Message-ID: It's basically set automatically -- the caller id from the inbound route is used when you bridge. As you said: > All the internal extensions see the caller ID no problem So therefore, either you are sending the caller ID in the wrong format to your carrier - RPID, pid, "From:", OR as I said, they are fixing your outbound caller ID. Are you in the UK? I know outbound calls via numbergroup only allows you to use CIDs registered on my account. I usually use xconnect to terminate to UK and afaik, they pass along the CID. Perhaps you need a backup carrier, and one that you use especially for these calls. -Avi On Fri, May 4, 2012 at 3:04 AM, Neil Stirton wrote: > Hi Avi, > > Set that to true but feel like I need to be passing through some more info > when bridging the call? > > You are most likely right that the trunk is controls what number is > displayed on outbound calls ? this I will check also. > > Many thanks, > Neil > > From: Avi Marcus > Reply-To: FreeSWITCH Users Help > Date: Thursday, 3 May 2012 23:20 > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Call forwarding to mobile showing our > trunk CLI instead of caller CLI > > Depending on your trunk provider, they might not allow you to change your > outbound caller ID. It might be hard-coded for your account. > > If not.. it could be you need to set "caller-id-in-from" to be true > within the gateway and reload the gateway. > > -Avi > > > On Fri, May 4, 2012 at 12:33 AM, Neil Stirton wrote: > >> Hi, >> >> Apologies for the noob question here. >> >> Setup Freeswitch on CentOS and using FusionPBX for 'easy' management. >> >> I've defined a hunt group (ext 7001) which simultaneously rings 3 >> internal SIP extensions + my mobile. >> >> All the internal extensions see the caller ID no problem but because I am >> also sending the call back out through our ITSP's trunk, the call always >> shows as our office calling my mobile phone rather than the actual caller's >> ID. >> I have the hunt group calling: >> >> Sip uri sofia/gateway// >> >> I'm sure it's a case of a variable setting something but if someone could >> point me in the right direction that would be very much appreciated. >> >> Many thanks >> Neil >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120504/23e92448/attachment.html From anita.hall at simmortel.com Fri May 4 11:48:38 2012 From: anita.hall at simmortel.com (Anita Hall) Date: Fri, 4 May 2012 13:18:38 +0530 Subject: [Freeswitch-users] Commercial Fax over T.30 for FreeSWITCH In-Reply-To: <4FA30EEE.7090702@earthspike.net> References: <4FA24100.7040908@integrafin.co.uk> <20120503151946.GA824@eagle.cupis.co.uk> <4FA30EEE.7090702@earthspike.net> Message-ID: Thanks a lot Steve, Paul, Anthm, John ! I will try both timerfd and the kernel freq thing :) Btw, I am not using any T.38 or IP. Here is the set-up and config E1 <-> Sangoma A108D <-> libPRI <-> FreeTDM <-> mod_spandsp The Sangoma cards take their timing source from the E1, echo cancellation is disabled. There is no T.38 or any other IP. We are using G.711 only Kernel timing related info $grep CONFIG_HZ= /boot/config-`uname -r` CONFIG_HZ=100 $ cat freetdm.conf [span wanpipe wp1] trunk_type => e1 group=1 b-channel => 1:1-15 d-channel => 1:16 b-channel => 1:17-31 etc. $ cat /etc/wanpipe/wanpipe1.conf [devices] wanpipe1 = WAN_AFT_TE1, Comment [interfaces] w1g1 = wanpipe1, , TDM_VOICE_API, Comment [wanpipe1] CARD_TYPE = AFT S514CPU = A CommPort = PRI AUTO_PCISLOT = NO PCISLOT = 6 PCIBUS = 1 FE_MEDIA = E1 FE_LCODE = HDB3 FE_FRAME = CRC4 FE_LINE = 1 TE_CLOCK = NORMAL TE_REF_CLOCK = 0 TE_SIG_MODE = CCS TE_HIGHIMPEDANCE = NO TE_RX_SLEVEL = 430 HW_RJ45_PORT_MAP = DEFAULT LBO = 120OH FE_TXTRISTATE = NO MTU = 1500 UDPPORT = 9000 TTL = 255 IGNORE_FRONT_END = NO TDMV_SPAN = 1 TDMV_DCHAN = 16 TE_AIS_MAINTENANCE = NO #NO: defualt YES: Start port in AIS Blue Alarm and keep line down #wanpipemon -i w1g1 -c Ttx_ais_off to disable AIS maintenance mode #wanpipemon -i w1g1 -c Ttx_ais_on to enable AIS maintenance mode TDMV_HW_DTMF = YES # YES: receive dtmf events from hardware TDMV_HW_FAX_DETECT = NO # YES: receive fax 1100hz events from hardware HWEC_OPERATION_MODE = OCT_NORMAL # OCT_NORMAL: echo cancelation enabled with nlp (default) # OCT_SPEECH: improves software tone detection by disabling NLP (echo possible) # OCT_NO_ECHO:disables echo cancelation but allows VQE/tone functions. HWEC_DTMF_REMOVAL = NO # NO: default YES: remove dtmf out of incoming media (must have hwdtmf enabled) HWEC_NOISE_REDUCTION = NO # NO: default YES: reduces noise on the line - could break fax HWEC_ACUSTIC_ECHO = NO # NO: default YES: enables acoustic echo cancellation HWEC_NLP_DISABLE = NO # NO: default YES: guarantees software tone detection (possible echo) HWEC_TX_AUTO_GAIN = 0 # 0: disable -40-0: default tx audio level to be maintained (-20 default) HWEC_RX_AUTO_GAIN = 0 # 0: disable -40-0: default tx audio level to be maintained (-20 default) HWEC_TX_GAIN = 0 # 0: disable -24-24: db values to be applied to tx signal HWEC_RX_GAIN = 0 # 0: disable -24-24: db values to be applied to tx signal [w1g1] ACTIVE_CH = ALL TDMV_HWEC = YES MTU = 80 $ cat autoload_configs/freetdm.conf.xml etc. regards, Anita On Fri, May 4, 2012 at 4:34 AM, John wrote: > Ubuntu default timer for the server kernel is 100Hz (although I haven't > checked 12.04). You need the -preempt kernel for low latency and > 1000Hz. It's easy to apt-get install; you then need to tweak the GRUB > config to make it your default kernel on (re)boot. > > John > > On 03/05/12 16:19, Paul Cupis wrote: > > On Thu, May 03, 2012 at 05:53:01PM +0530, Anita Hall wrote: > >> No, I am running FreeSWITCH on an Ubuntu 64-bit Quad core machine. > >> There is no VM. > >> What all things should I check to ensure that the timing on my system > >> is good? Like, may be, my motherboard is bad or my kernel could be > >> tuned better ? > > The most common suggestion is to use a kernel with 1000Hz timer. I think > > that the default timer for Ubuntu is 250, so this would be first thing > > to check/change. > > > > You can check by running: > > > > grep CONFIG_HZ= /boot/config-`uname -r` > > > > and if it is not 1000, it would be worth installing a 1000Hz kernel > > and re-testing. > > > > Ubuntu may have a packages -lowlatency kernel which has a 1000Hz timer > > but this will also have other changes. I think a lot of people will > > compile a custom kernel based on the standard Ubuntu one, just changing > > that one configuration option. > > > > Regards, > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120504/211d4fe2/attachment-0001.html From anita.hall at simmortel.com Fri May 4 12:41:15 2012 From: anita.hall at simmortel.com (Anita Hall) Date: Fri, 4 May 2012 14:11:15 +0530 Subject: [Freeswitch-users] odbc with FreeTDM Message-ID: Hi Could anyone share a sample config of odbc with FreeTDM ? Thanks. regards, Anita -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120504/e3e3f88c/attachment.html From peter.olsson at visionutveckling.se Fri May 4 13:53:12 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Fri, 4 May 2012 09:53:12 +0000 Subject: [Freeswitch-users] odbc with FreeTDM Message-ID: <1FFF97C269757C458224B7C895F35F150A163A@cantor.std.visionutv.se> AFAIK there is not ODBC in FreeTDM - what are you trying to do? /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Anita Hall Skickat: den 4 maj 2012 10:41 Till: FreeSWITCH Users Help ?mne: [Freeswitch-users] odbc with FreeTDM Hi Could anyone share a sample config of odbc with FreeTDM ? Thanks. regards, Anita !DSPAM:4fa3951932765625512396! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120504/301fd50d/attachment.html From anita.hall at simmortel.com Fri May 4 15:12:59 2012 From: anita.hall at simmortel.com (Anita Hall) Date: Fri, 4 May 2012 16:42:59 +0530 Subject: [Freeswitch-users] FreeSWITCH Conference Call: Vestec ASR In-Reply-To: References: Message-ID: Hi Michael I missed this conf call. Where can I find the recordings of this conf call and others ? regards, Anita On Wed, May 2, 2012 at 9:21 PM, Michael Collins wrote: > Hello all! > > Just a reminder that the FreeSWITCH community conference call will be > starting in just over an hour. The agenda page is here. > Come join us for an interesting talk with the Vestec team about automatic > speech recognition (ASR) in a FreeSWITCH environment. > > -Michael > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120504/98af6305/attachment.html From anton.jugatsu at gmail.com Fri May 4 15:16:50 2012 From: anton.jugatsu at gmail.com (Anton Kvashenkin) Date: Fri, 4 May 2012 15:16:50 +0400 Subject: [Freeswitch-users] FreeSWITCH Conference Call: Vestec ASR In-Reply-To: References: Message-ID: Here http://wiki.freeswitch.org/wiki/Weekly_Conference_Call 2012/5/4 Anita Hall > Hi Michael > > I missed this conf call. Where can I find the recordings of this conf call > and others ? > > regards, > Anita > > > > On Wed, May 2, 2012 at 9:21 PM, Michael Collins wrote: > >> Hello all! >> >> Just a reminder that the FreeSWITCH community conference call will be >> starting in just over an hour. The agenda page is here. >> Come join us for an interesting talk with the Vestec team about automatic >> speech recognition (ASR) in a FreeSWITCH environment. >> >> -Michael >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120504/f6801374/attachment.html From acrow at integrafin.co.uk Fri May 4 18:48:38 2012 From: acrow at integrafin.co.uk (Alex Crow) Date: Fri, 04 May 2012 15:48:38 +0100 Subject: [Freeswitch-users] Commercial Fax over T.30 for FreeSWITCH In-Reply-To: References: <4FA24100.7040908@integrafin.co.uk> Message-ID: <4FA3EC46.2090109@integrafin.co.uk> On 03/05/12 13:23, Anita Hall wrote: > Hi Alex > > Thanks for the tip :) > > No, I am running FreeSWITCH on an Ubuntu 64-bit Quad core machine. > There is no VM. > > Is it possible to use Hylafax instead of or in combination to > FreeSWITCH if I am using Sangoma E1 Cards ? > > What all things should I check to ensure that the timing on my system > is good? Like, may be, my motherboard is bad or my kernel could be > tuned better ? > > Cheers! > > regards, > Anita > > Yes, you can use it with the E1 cards in exactly the same way, instead of bridging to a gateway you'd bridge to FreeTDM channel(s) for outbound. Inbound it's exactly the same. However I've just noticed a later post by Anthony in which he describes soft-modems built into Freeswitch directly. I would try that first as it was a right royal pain compiling T38modem! Cheers Alex -- This message is intended only for the addressee and may contain confidential information. Unless you are that person, you may not disclose its contents or use it in any way and are requested to delete the message along with any attachments and notify us immediately. "Transact" is operated by Integrated Financial Arrangements plc Domain House, 5-7 Singer Street, London EC2A 4BQ Tel: (020) 7608 4900 Fax: (020) 7608 5300 (Registered office: as above; Registered in England and Wales under number: 3727592) Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856) From acrow at integrafin.co.uk Fri May 4 18:58:31 2012 From: acrow at integrafin.co.uk (Alex Crow) Date: Fri, 04 May 2012 15:58:31 +0100 Subject: [Freeswitch-users] Commercial Fax over T.30 for FreeSWITCH In-Reply-To: References: <4FA24100.7040908@integrafin.co.uk> <20120503151946.GA824@eagle.cupis.co.uk> Message-ID: <4FA3EE97.1090908@integrafin.co.uk> On 03/05/12 21:29, Anthony Minessale wrote: > Once you get to a version of FS that supports timerfd there is less > issue with kernel freq. > > Also, more people should try the new t30 modem emulation built right > into FS to work with hlyafax > > see spandsp.conf.xml > one word of caution is don't leave (FS off and the faxgettys on) very > long or the system will start using the ttty's for ssh sessions etc > and faxgetty will start sending at commands to your stdin. > > > > > > > > Anthony, That is fantastic - seems I have a recent enough build for this. One question - how would you route inbound fax calls to these emulated modems so that Hylafax can deal with them? Cheers Alex From rrodolfos at gmail.com Fri May 4 19:03:52 2012 From: rrodolfos at gmail.com (RrodolfoS .) Date: Fri, 4 May 2012 10:33:52 -0430 Subject: [Freeswitch-users] Openvox tdma400p fail to Disconnect/Hangup detect (Dahdi/Freetdm) Message-ID: Hi, Freeswitch can recive calls, but the call is end, openvox tdma400p, can't detect disconnection or hangup. Any idea? RrodolfoS From mario_fs at mgtech.com Fri May 4 19:33:25 2012 From: mario_fs at mgtech.com (Mario G) Date: Fri, 4 May 2012 08:33:25 -0700 Subject: [Freeswitch-users] Todays Git update has errors Message-ID: FYI - Been updating fine until today. Got the message below and update stopped. making all mod_event_socket Compiling /usr/local/src/freeswitch/src/mod/event_handlers/mod_event_socket/mod_event_socket.c... quiet_libtool: compile: gcc -I/usr/local/src/freeswitch/libs/curl/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DMACOSX -DHAVE_OPENSSL -g -O2 -pipe -no-cpp-precomp -Wall -std=c99 -pedantic -Wdeclaration-after-statement -D_GNU_SOURCE -DHAVE_CONFIG_H -c /usr/local/src/freeswitch/src/mod/event_handlers/mod_event_socket/mod_event_socket.c -fno-common -DPIC -o .libs/mod_event_socket.o cc1: warnings being treated as errors /usr/local/src/freeswitch/src/mod/event_handlers/mod_event_socket/mod_event_socket.c: In function 'read_packet': /usr/local/src/freeswitch/src/mod/event_handlers/mod_event_socket/mod_event_socket.c:1360: warning: format '%lld' expects type 'long long int', but argument 9 has type 'time_t' make[6]: *** [mod_event_socket.lo] Error 1 make[5]: *** [all] Error 1 make[4]: *** [mod_event_socket-all] Error 1 make[3]: *** [all-recursive] Error 1 make[2]: *** [all-recursive] Error 1 make[1]: *** [all] Error 2 make: *** [current] Error 2 From msc at freeswitch.org Fri May 4 20:08:14 2012 From: msc at freeswitch.org (Michael Collins) Date: Fri, 4 May 2012 09:08:14 -0700 Subject: [Freeswitch-users] FreeSWITCH-biz Mailing List Message-ID: Hello all, I'd like to encourage everyone to read this poston the main FreeSWITCH.org site. In short: we are finding that there are more and more business opportunities for members of the FreeSWITCH community. Those with marketable FreeSWITCH skills are increasingly in demand. One way to keep everyone informed about these opportunities is to have a place where people can post and discuss these opportunities. The ideal place for this is the FreeSWITCH-biz mailing list. We encourage everyone to subscribe to this list so that we can keep business discussions separate from user and developer questions. -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120504/73d8ab48/attachment.html From steveu at coppice.org Fri May 4 20:17:53 2012 From: steveu at coppice.org (Steve Underwood) Date: Sat, 05 May 2012 00:17:53 +0800 Subject: [Freeswitch-users] Commercial Fax over T.30 for FreeSWITCH In-Reply-To: References: <4FA24100.7040908@integrafin.co.uk> Message-ID: <4FA40131.8090009@coppice.org> On 05/03/2012 08:23 PM, Anita Hall wrote: > Hi Alex > > Thanks for the tip :) > > No, I am running FreeSWITCH on an Ubuntu 64-bit Quad core machine. > There is no VM. > > Is it possible to use Hylafax instead of or in combination to > FreeSWITCH if I am using Sangoma E1 Cards ? > > What all things should I check to ensure that the timing on my system > is good? Like, may be, my motherboard is bad or my kernel could be > tuned better ? Are you really interested in fixing your problems? It seems you have greater interest in tinkering with every possible solution. Many many people handle hundreds of thousands of FAXes per day with low failures rates using: asterisk + iaxmodem/spandsp + hylafax, or asterisk + spandsp, or freeswitch + spandsp and some people are probably now carrying significant traffic with freeswitch + spandsp + hylafax, which is a recent addition to the options. They all work well when set up properly. Steve From modesto at isimples.com.br Fri May 4 17:05:40 2012 From: modesto at isimples.com.br (Antonio Modesto) Date: Fri, 04 May 2012 10:05:40 -0300 Subject: [Freeswitch-users] Does Freeswitch replace Asterisk? Message-ID: <1336136740.3092.31.camel@modesto.localdomain.net> Hi, I Work at an ISP and we have an Asterisk PBX. Our PBX doesn't have anything special, it just do the common things like auto attendant, CDR, call transfer and such things. We have 2 digium FXO cards to connect to the PSTN. The problem is that asterisk is not working very well, sometimes some of the dahdi channels get stuck and we need to release it through the asterisk console, the detection of callerid doesn't work (We live in brazil, dtmf signaling), and other detais that if I list all of them here it's going to take some time. I read a lot about asterisk and its bad design, I think that it added a lot a features without worrying about making stable implementations. My question is, can freeswitch fully replace our Asterisk PBX? Or it's not its purpose? Regards. From dgarcia at anew.com.ve Fri May 4 20:55:35 2012 From: dgarcia at anew.com.ve (Saugort Dario Garcia Tovar) Date: Fri, 04 May 2012 12:25:35 -0430 Subject: [Freeswitch-users] Does Freeswitch replace Asterisk? In-Reply-To: <1336136740.3092.31.camel@modesto.localdomain.net> References: <1336136740.3092.31.camel@modesto.localdomain.net> Message-ID: <4FA40A07.3090308@anew.com.ve> Hi Antonio, Yes, If you are not using a special feauture or module devolped for asterisk, You can replace your Asterisk with Freeswitch. As you said, you are in Brazil, FS support digium cards, also if I am not wrong also support khomp cards (Brazilian supplier). All you need is a small machine to give it a FS a try. Only two points: first,about fxo channels gettings stuck, this happens in several manufacture hardware when pstn lines are analog, it is better digital lines. You should take time in see if your digium hardware need some tunning like adjust tone detection, or line voltage. Second, abour your callerid, you need check with your pstn provider to validate. On 5/4/2012 8:35 AM, Antonio Modesto wrote: > Hi, > > I Work at an ISP and we have an Asterisk PBX. Our PBX doesn't have > anything special, it just do the common things like auto attendant, CDR, > call transfer and such things. We have 2 digium FXO cards to connect to > the PSTN. The problem is that asterisk is not working very well, > sometimes some of the dahdi channels get stuck and we need to release it > through the asterisk console, the detection of callerid doesn't work (We > live in brazil, dtmf signaling), and other detais that if I list all of > them here it's going to take some time. I read a lot about asterisk and > its bad design, I think that it added a lot a features without worrying > about making stable implementations. My question is, can freeswitch > fully replace our Asterisk PBX? Or it's not its purpose? > > > Regards. > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ----- > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.1913 / Virus Database: 2425/4977 - Release Date: 05/04/12 > > -- Atentamente, *Dario Garc?a* Consultor. CCCT, Nivel C2, Sector Yarey, Mz, Ofc. MZ03a. Caracas-Venezuela. Tel?fono: +58 212 9081842 Cel: +58 412 2221515 dgarcia at anew.com.ve http://www.anew.com.ve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120504/d4856d69/attachment.html From bdfoster at endigotech.com Fri May 4 20:55:53 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Fri, 4 May 2012 12:55:53 -0400 Subject: [Freeswitch-users] Todays Git update has errors In-Reply-To: References: Message-ID: Bug reports go to JIRA, not on the mailing list. This bug has already been reported, I don't have the link handy but search for it on JIRA and add your notes there. This should only be affecting 32 bit arch's. -BDF On May 4, 2012 11:34 AM, "Mario G" wrote: > FYI - Been updating fine until today. Got the message below and update > stopped. > > making all mod_event_socket > Compiling > /usr/local/src/freeswitch/src/mod/event_handlers/mod_event_socket/mod_event_socket.c... > quiet_libtool: compile: gcc -I/usr/local/src/freeswitch/libs/curl/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/libs/libteletone/src -Werror > -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb > -DMACOSX -DHAVE_OPENSSL -g -O2 -pipe -no-cpp-precomp -Wall -std=c99 > -pedantic -Wdeclaration-after-statement -D_GNU_SOURCE -DHAVE_CONFIG_H -c > /usr/local/src/freeswitch/src/mod/event_handlers/mod_event_socket/mod_event_socket.c > -fno-common -DPIC -o .libs/mod_event_socket.o > cc1: warnings being treated as errors > /usr/local/src/freeswitch/src/mod/event_handlers/mod_event_socket/mod_event_socket.c: > In function 'read_packet': > /usr/local/src/freeswitch/src/mod/event_handlers/mod_event_socket/mod_event_socket.c:1360: > warning: format '%lld' expects type 'long long int', but argument 9 has > type 'time_t' > make[6]: *** [mod_event_socket.lo] Error 1 > make[5]: *** [all] Error 1 > make[4]: *** [mod_event_socket-all] Error 1 > make[3]: *** [all-recursive] Error 1 > make[2]: *** [all-recursive] Error 1 > make[1]: *** [all] Error 2 > make: *** [current] Error 2 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120504/c35147d6/attachment-0001.html From msc at freeswitch.org Fri May 4 20:57:49 2012 From: msc at freeswitch.org (Michael Collins) Date: Fri, 4 May 2012 09:57:49 -0700 Subject: [Freeswitch-users] Does Freeswitch replace Asterisk? In-Reply-To: <1336136740.3092.31.camel@modesto.localdomain.net> References: <1336136740.3092.31.camel@modesto.localdomain.net> Message-ID: Hello Antonio, Your scenario is not uncommon. It may be that you have a particularly bad version of Asterisk. (A few releases were particularly problematic.) It could be that your hardware has issues. Or it could be that Asterisk may not be a good fit for your scenario. FreeSWITCH can do basically all of what you are doing now. The only thing I would do is contact Sangoma and ask them about Digium FXO cards in Brazil. (Sangoma wrote the FreeTDM stack for FreeSWITCH, so they're the experts on the subject.) If you are looking for a FreeSWITCH + GUI solution then you might want to check out FusionPBX or blue.box. -MC On Fri, May 4, 2012 at 6:05 AM, Antonio Modesto wrote: > Hi, > > I Work at an ISP and we have an Asterisk PBX. Our PBX doesn't have > anything special, it just do the common things like auto attendant, CDR, > call transfer and such things. We have 2 digium FXO cards to connect to > the PSTN. The problem is that asterisk is not working very well, > sometimes some of the dahdi channels get stuck and we need to release it > through the asterisk console, the detection of callerid doesn't work (We > live in brazil, dtmf signaling), and other detais that if I list all of > them here it's going to take some time. I read a lot about asterisk and > its bad design, I think that it added a lot a features without worrying > about making stable implementations. My question is, can freeswitch > fully replace our Asterisk PBX? Or it's not its purpose? > > > Regards. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120504/84f0abd6/attachment.html From jeff at jefflenk.com Fri May 4 21:01:17 2012 From: jeff at jefflenk.com (Jeff Lenk) Date: Fri, 4 May 2012 10:01:17 -0700 (PDT) Subject: [Freeswitch-users] Todays Git update has errors In-Reply-To: References: Message-ID: <1336150877276-7527811.post@n2.nabble.com> git pull - this problem should already be fixed. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Todays-Git-update-has-errors-tp7527470p7527811.html Sent from the freeswitch-users mailing list archive at Nabble.com. From bdfoster at endigotech.com Fri May 4 21:08:33 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Fri, 4 May 2012 13:08:33 -0400 Subject: [Freeswitch-users] Todays Git update has errors In-Reply-To: <1336150877276-7527811.post@n2.nabble.com> References: <1336150877276-7527811.post@n2.nabble.com> Message-ID: Jeff, I know you did a commit, but that one didn't change anything on my end. Error message is the same. I haven't done anything since 10am EST, was there another commit? -BDF On May 4, 2012 1:01 PM, "Jeff Lenk" wrote: > git pull - this problem should already be fixed. > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Todays-Git-update-has-errors-tp7527470p7527811.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120504/bf68ebb3/attachment.html From thaddeus at thogan.com Fri May 4 21:10:00 2012 From: thaddeus at thogan.com (thaddeus at thogan.com) Date: Fri, 04 May 2012 12:10:00 -0500 Subject: [Freeswitch-users] Dialplan condition to test if SIP client is registered Message-ID: <6628-4fa40d80-1-39363f40@8181014> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120504/f0070f85/attachment.html From jeff at jefflenk.com Fri May 4 21:34:47 2012 From: jeff at jefflenk.com (Jeff Lenk) Date: Fri, 4 May 2012 10:34:47 -0700 (PDT) Subject: [Freeswitch-users] Todays Git update has errors In-Reply-To: References: <1336150877276-7527811.post@n2.nabble.com> Message-ID: <1336152887225-7527954.post@n2.nabble.com> Commit:e8098c060d62240a9e266b8de88e5e060432fe9d * FS-4184 --resolve anthm committed that one. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Todays-Git-update-has-errors-tp7527470p7527954.html Sent from the freeswitch-users mailing list archive at Nabble.com. From msc at freeswitch.org Fri May 4 21:34:31 2012 From: msc at freeswitch.org (Michael Collins) Date: Fri, 4 May 2012 10:34:31 -0700 Subject: [Freeswitch-users] Dialplan condition to test if SIP client is registered In-Reply-To: <6628-4fa40d80-1-39363f40@8181014> References: <6628-4fa40d80-1-39363f40@8181014> Message-ID: Hi Thaddeus, You definitely have several options here. While you could use the sofia_contact API, I'm wondering if perhaps you could let the dialplan and the bridge app do the work for you. You could bridge directly to the Android user and then if it fails it could then continue in the dialplan and ring the mobile number. This is covered briefly on the wiki, however it is covered very nicely in the new FreeSWITCH Cookbook in chapter 1. Try out the basic failover (pipe-separated list of endpoints) and if that doesn't work let us know. Maybe you could hop on IRC and talk live with other community members. Of course, it can't hurt to raise your karma by getting the FreeSWITCH books. :) (See freeswitch.org, upper left corner.) Thanks, Michael On Fri, May 4, 2012 at 10:10 AM, thaddeus at thogan.com wrote: > I was wondering if there is a way to tell if a SIP client is registered > with a dialplan condition? Or maybe a better way to accomplish the > following?: > > The users' cell phones (Android) are running a SIP client and connect to > freeswitch via Wifi when in the office. I want calls destined for a given > user to ring only their extension if they are connected via SIP, and ring > only their mobile number when they are not connected. > > Currently I just take every incoming call and bridge it back out to their > mobile number, but 95% of the time these users are on the Wifi network, and > I could just pass the call to their phones via SIP. > > I have tested hunt groups but the callers hear ringing for far too long. > Ring groups don't work because it is confusing when the user simultaneously > receiving the same call via SIP and mobile on the same phone. > > Thanks in advance for any insights and help! > > -- Thaddeus > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120504/ee24a8ff/attachment.html From acrow at integrafin.co.uk Fri May 4 21:39:19 2012 From: acrow at integrafin.co.uk (Alex Crow) Date: Fri, 04 May 2012 18:39:19 +0100 Subject: [Freeswitch-users] Commercial Fax over T.30 for FreeSWITCH In-Reply-To: <4FA40131.8090009@coppice.org> References: <4FA24100.7040908@integrafin.co.uk> <4FA40131.8090009@coppice.org> Message-ID: <4FA41447.1000700@integrafin.co.uk> On 04/05/12 17:17, Steve Underwood wrote: > On 05/03/2012 08:23 PM, Anita Hall wrote: >> Hi Alex >> >> Thanks for the tip :) >> >> No, I am running FreeSWITCH on an Ubuntu 64-bit Quad core machine. >> There is no VM. >> >> Is it possible to use Hylafax instead of or in combination to >> FreeSWITCH if I am using Sangoma E1 Cards ? >> >> What all things should I check to ensure that the timing on my system >> is good? Like, may be, my motherboard is bad or my kernel could be >> tuned better ? > Are you really interested in fixing your problems? It seems you have > greater interest in tinkering with every possible solution. Many many > people handle hundreds of thousands of FAXes per day with low failures > rates using: > asterisk + iaxmodem/spandsp + hylafax, or > asterisk + spandsp, or > freeswitch + spandsp > and some people are probably now carrying significant traffic with > freeswitch + spandsp + hylafax, which is a recent addition to the > options. They all work well when set up properly. > > Steve > > Steve, Forgive me, but to be honest I think that opening sentence is a bit harsh. Timing is important for fax, and I think the OP just wants to eliminate all the possible causes, which is commendable. I have had problems in the past with bad timing in the kernel on Ubuntu (although it may not apply here, I was on BRI) and it was a really difficult time with lots of complaints from users until I pinned it down. I think we just need to point out the great capabilities of the Free/OSS software that is available. I have been stunned with the quality and stability of both FreeSwitch and HylaFAX, and having used the latter for more than a decade in a commercial environment cannot fault it. It also has great Windows clients such as WHFC, great reporting of fax outcomes built-in, and just seems to do its job without ever needing attention (We had one box with 5+ years of uptime). Moving to to a new softswitch/PBX can be very daunting for a new user, especially one that comes from the Asterisk environment (lots of baggage) or one not accustomed to telephony in general. Cheers Alex From thaddeus at thogan.com Fri May 4 22:34:07 2012 From: thaddeus at thogan.com (thaddeus at thogan.com) Date: Fri, 04 May 2012 13:34:07 -0500 Subject: [Freeswitch-users] Dialplan condition to test if SIP client is registered In-Reply-To: Message-ID: <6629-4fa42100-f-13182400@24628931> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120504/1f31bddc/attachment.html From freeswitch at peely.com Fri May 4 22:37:34 2012 From: freeswitch at peely.com (peely) Date: Fri, 4 May 2012 11:37:34 -0700 (PDT) Subject: [Freeswitch-users] Kernel frequency Message-ID: <1336156654702-7528206.post@n2.nabble.com> Hi, I run my production stuff on Ubuntu Server and have been using the pre-emptive kernel which runs at 1000hz, this was because previous build of FreeSWITCH seemed to need to run at this frequency. In testing the latest Ubuntu LTS release, it seems they're not producing recent kernels for anything other than their generic Kernel, which runs at 250hz I believe. Is it still a requirement / recommendation that FreeSWITCH runs on 1000hz boxes? I know the preferred distro is Centos, but would appreciate some guidance on what may occur if I were to move to 250hz kernels on production boxes, obviously this would still be a "softly softly" move. Thanks, Neil. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Kernel-frequency-tp7528206.html Sent from the freeswitch-users mailing list archive at Nabble.com. From steveu at coppice.org Fri May 4 22:48:56 2012 From: steveu at coppice.org (Steve Underwood) Date: Sat, 05 May 2012 02:48:56 +0800 Subject: [Freeswitch-users] Commercial Fax over T.30 for FreeSWITCH In-Reply-To: <4FA41447.1000700@integrafin.co.uk> References: <4FA24100.7040908@integrafin.co.uk> <4FA40131.8090009@coppice.org> <4FA41447.1000700@integrafin.co.uk> Message-ID: <4FA42498.9070504@coppice.org> On 05/05/2012 01:39 AM, Alex Crow wrote: > On 04/05/12 17:17, Steve Underwood wrote: >> On 05/03/2012 08:23 PM, Anita Hall wrote: >>> Hi Alex >>> >>> Thanks for the tip :) >>> >>> No, I am running FreeSWITCH on an Ubuntu 64-bit Quad core machine. >>> There is no VM. >>> >>> Is it possible to use Hylafax instead of or in combination to >>> FreeSWITCH if I am using Sangoma E1 Cards ? >>> >>> What all things should I check to ensure that the timing on my system >>> is good? Like, may be, my motherboard is bad or my kernel could be >>> tuned better ? >> Are you really interested in fixing your problems? It seems you have >> greater interest in tinkering with every possible solution. Many many >> people handle hundreds of thousands of FAXes per day with low failures >> rates using: >> asterisk + iaxmodem/spandsp + hylafax, or >> asterisk + spandsp, or >> freeswitch + spandsp >> and some people are probably now carrying significant traffic with >> freeswitch + spandsp + hylafax, which is a recent addition to the >> options. They all work well when set up properly. >> >> Steve >> >> > Steve, > > Forgive me, but to be honest I think that opening sentence is a bit > harsh. Timing is important for fax, and I think the OP just wants to > eliminate all the possible causes, which is commendable. I have had > problems in the past with bad timing in the kernel on Ubuntu (although > it may not apply here, I was on BRI) and it was a really difficult time > with lots of complaints from users until I pinned it down. > > I think we just need to point out the great capabilities of the Free/OSS > software that is available. I have been stunned with the quality and > stability of both FreeSwitch and HylaFAX, and having used the latter for > more than a decade in a commercial environment cannot fault it. It also > has great Windows clients such as WHFC, great reporting of fax outcomes > built-in, and just seems to do its job without ever needing attention > (We had one box with 5+ years of uptime). Moving to to a new > softswitch/PBX can be very daunting for a new user, especially one that > comes from the Asterisk environment (lots of baggage) or one not > accustomed to telephony in general. > Have you seen what and where she has been posting? She's taking a scattergun approach, and its getting kind of annoying. She seems to assume every piece of software around is thoroughly broken, and the fact people are using it successfully in high volume applications is an illusion. Nice people get a lot of help from me, if I'm not too busy. She won't. Steve From drk at drkngs.net Fri May 4 22:24:45 2012 From: drk at drkngs.net (Dave R. Kompel) Date: Fri, 04 May 2012 11:24:45 -0700 Subject: [Freeswitch-users] =?iso-8859-1?q?Dialplan_condition_to_test_if_S?= =?iso-8859-1?q?IP_client_is=09registered?= In-Reply-To: Message-ID: <20120504182445.9e296b96@mail.tritonwest.net> Thaddeus & Michael: As Michael stated there are many ways to fail over if a phone/user isn't registered. However sometimes you may want to actually check the registration yourself. Or even do something based on a contact being registered, that has nothing to do with the actual one you are calling. I normally use the API: sofia_contact([profile/]@domain). It returns the channel string you would pass to bridge. If the user isn't registered it returns "error/" so it would use the logical "error" endpoint to handle the right treatment. If the regex "^error" match is true, then the user isn't registered. The extra thing you get from that API call is that you can use a "*" for the profile, and find out which SIP profile the user is registered in. For example in the case of an Android client, that may be using WIFI or 3G, and you have an external and internal profile they can register, you can tell if the user is "on site" or "off site" by the profile it returns, and can be more restrictive of the calls you send to someone when they're out of the office. --Dave _____ From: Michael Collins [mailto:msc at freeswitch.org] To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org] Sent: Fri, 04 May 2012 10:34:31 -0700 Subject: Re: [Freeswitch-users] Dialplan condition to test if SIP client is registered Hi Thaddeus, You definitely have several options here. While you could use the sofia_contact API, I'm wondering if perhaps you could let the dialplan and the bridge app do the work for you. You could bridge directly to the Android user and then if it fails it could then continue in the dialplan and ring the mobile number. This is covered briefly on the wiki, however it is covered very nicely in the new FreeSWITCH Cookbook in chapter 1. Try out the basic failover (pipe-separated list of endpoints) and if that doesn't work let us know. Maybe you could hop on IRC and talk live with other community members. Of course, it can't hurt to raise your karma by getting the FreeSWITCH books. :) (See freeswitch.org, upper left corner.) Thanks, Michael On Fri, May 4, 2012 at 10:10 AM, thaddeus at thogan.com wrote: I was wondering if there is a way to tell if a SIP client is registered with a dialplan condition? Or maybe a better way to accomplish the following?: The users' cell phones (Android) are running a SIP client and connect to freeswitch via Wifi when in the office. I want calls destined for a given user to ring only their extension if they are connected via SIP, and ring only their mobile number when they are not connected. Currently I just take every incoming call and bridge it back out to their mobile number, but 95% of the time these users are on the Wifi network, and I could just pass the call to their phones via SIP. I have tested hunt groups but the callers hear ringing for far too long. Ring groups don't work because it is confusing when the user simultaneously receiving the same call via SIP and mobile on the same phone. Thanks in advance for any insights and help! -- Thaddeus _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120504/ecf7126e/attachment.html From krice at freeswitch.org Fri May 4 23:20:16 2012 From: krice at freeswitch.org (Ken Rice) Date: Fri, 04 May 2012 14:20:16 -0500 Subject: [Freeswitch-users] FreeSWITCH 1.2 and packaging update Message-ID: As many of you know we have been working hot and heavy on getting 1.2 ready to drop. In the next 24 hours we will be pushing some further changes into tree that will affect a small number of users. These Changes are leading up to the 1.2 release and specifically address Packaging for Debian (and Debian based) system. If you arent currently using the debian package building bits of the FreeSWITCH source tree, you will not be affected. However if you are expect to see some big changes... These changes are to move us more inline with the Centos packages and more inline with what is required to get FreeSWITCH in the normal Debian repositories. Also, do we have a brave soul that might want to help with the FreeSWITCH Change log? Contact me off list or via IRC (SwK in #freeswitch on irc.freenode.net) if this is something you can help with. K -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120504/6575f5d9/attachment-0001.html From jmesquita at freeswitch.org Sat May 5 00:08:00 2012 From: jmesquita at freeswitch.org (=?utf-8?Q?Jo=C3=A3o_Mesquita?=) Date: Fri, 4 May 2012 17:08:00 -0300 Subject: [Freeswitch-users] New proof-of-concept module (mod_pickupgroup) Message-ID: Hello folks, I've been working lately with pickup groups and I found some situations where the way the intercept application is presented along with the hash application on the default configs will not suffice the expected behavior on a PBX environment. Of course there are many solutions to this problem and I bet that there are even possible solutions without coding and just dial plan tricks. Nonetheless, a beginner could probably overlook some of those situations and this is the reason why I have started this module. The first problem I am trying to solve is that hash will only hold 1 UUID per group. If you have 2 consecutive calls and want to pick both up, you won't be able to. Simple way to test it. A calls B and leaves B ringing. C calls D and leaves D ringing. E intercepts (in which case it will intercept C). F intercepts same group. Expected behavior is to intercept A, but hash has forgotten about A and therefore you will get reorder tone. By "borrowing" (nicer than stealing) the idea from `eavesdrop all`, I have created mod_pickupgroup as a proof of concept and I would like the community not only to test the module but also to provide input as far as the code is concerned (I am not that proficient with C) and as far as functionality is concerned. The code is currently hosted on Github with the intent to move it somewhere more definitive if the module ends up being useful. The link is here: http://github.com/jmesquita/mod_pickupgroup To compile it, you need to adjust the BASE variable inside the Makefile by pointing it to your latest FreeSWITCH? source. -- Jo?o Mesquita Sent with Sparrow (http://www.sparrowmailapp.com/?sig) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120504/af4933fe/attachment.html From vipkilla at gmail.com Sat May 5 00:27:05 2012 From: vipkilla at gmail.com (Vik Killa) Date: Fri, 4 May 2012 16:27:05 -0400 Subject: [Freeswitch-users] New proof-of-concept module (mod_pickupgroup) In-Reply-To: References: Message-ID: +1 From jmesquita at freeswitch.org Sat May 5 00:49:05 2012 From: jmesquita at freeswitch.org (=?utf-8?Q?Jo=C3=A3o_Mesquita?=) Date: Fri, 4 May 2012 17:49:05 -0300 Subject: [Freeswitch-users] Does Freeswitch replace Asterisk? In-Reply-To: References: <1336136740.3092.31.camel@modesto.localdomain.net> Message-ID: <308D166EF6444F968D136C820993582D@freeswitch.org> Dar?o, thank you for remembering Khomp! Antonio, I might be able to chime in here. The CallerID is probably not working in Brazil because we use a specific protocol called DTMF. Most US hard card developers have a hard time with that because the standard in the US is FSK (a different protocol). Khomp has DTMF signaling support for both FXO and FXS cards (meaning your regular analog phones with display can use "bina" as well). Send me an email if you would like for more info. We support FreeSWITCH and Asterisk, for that matter. Regards (Abra?os), -- Jo?o Mesquita Sent with Sparrow (http://www.sparrowmailapp.com/?sig) On Friday, May 4, 2012 at 1:57 PM, Michael Collins wrote: > Hello Antonio, > > Your scenario is not uncommon. It may be that you have a particularly bad version of Asterisk. (A few releases were particularly problematic.) It could be that your hardware has issues. Or it could be that Asterisk may not be a good fit for your scenario. > > FreeSWITCH can do basically all of what you are doing now. The only thing I would do is contact Sangoma and ask them about Digium FXO cards in Brazil. (Sangoma wrote the FreeTDM stack for FreeSWITCH, so they're the experts on the subject.) > > If you are looking for a FreeSWITCH + GUI solution then you might want to check out FusionPBX or blue.box. > > -MC > > On Fri, May 4, 2012 at 6:05 AM, Antonio Modesto wrote: > > Hi, > > > > I Work at an ISP and we have an Asterisk PBX. Our PBX doesn't have > > anything special, it just do the common things like auto attendant, CDR, > > call transfer and such things. We have 2 digium FXO cards to connect to > > the PSTN. The problem is that asterisk is not working very well, > > sometimes some of the dahdi channels get stuck and we need to release it > > through the asterisk console, the detection of callerid doesn't work (We > > live in brazil, dtmf signaling), and other detais that if I list all of > > them here it's going to take some time. I read a lot about asterisk and > > its bad design, I think that it added a lot a features without worrying > > about making stable implementations. My question is, can freeswitch > > fully replace our Asterisk PBX? Or it's not its purpose? > > > > > > Regards. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120504/bd34a189/attachment.html From msc at freeswitch.org Sat May 5 01:25:47 2012 From: msc at freeswitch.org (Michael Collins) Date: Fri, 4 May 2012 14:25:47 -0700 Subject: [Freeswitch-users] Dialplan condition to test if SIP client is registered In-Reply-To: <20120504182445.9e296b96@mail.tritonwest.net> References: <20120504182445.9e296b96@mail.tritonwest.net> Message-ID: Dave, Thanks for the extra info. I had forgotten that we added the sofia_contact(*/user at domain) syntax to let you figure out which profile they're on. -MC On Fri, May 4, 2012 at 11:24 AM, Dave R. Kompel wrote: > ** > Thaddeus & Michael: > > As Michael stated there are many ways to fail over if a phone/user isn't > registered. However sometimes you may want to actually check the > registration yourself. Or even do something based on a contact being > registered, that has nothing to do with the actual one you are calling. > > I normally use the API: sofia_contact([profile/]@domain). It returns > the channel string you would pass to bridge. If the user isn't registered > it returns "error/" so it would use the logical "error" endpoint > to handle the right treatment. If the regex "^error" match is true, then > the user isn't registered. > > The extra thing you get from that API call is that you can use a "*" for > the profile, and find out which SIP profile the user is registered in. For > example in the case of an Android client, that may be using WIFI or 3G, and > you have an external and internal profile they can register, you can tell > if the user is "on site" or "off site" by the profile it returns, and can > be more restrictive of the calls you send to someone when they're out of > the office. > > --Dave > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120504/42130004/attachment.html From mario_fs at mgtech.com Sat May 5 01:30:21 2012 From: mario_fs at mgtech.com (Mario G) Date: Fri, 4 May 2012 14:30:21 -0700 Subject: [Freeswitch-users] Todays Git update has errors In-Reply-To: <1336152887225-7527954.post@n2.nabble.com> References: <1336150877276-7527811.post@n2.nabble.com> <1336152887225-7527954.post@n2.nabble.com> Message-ID: FYI, sorry about not doing the Jira, I am on 64 bit osX, I saw a change 30 minutes later and it works now. Mario G On May 4, 2012, at 10:34 AM, Jeff Lenk wrote: > Commit:e8098c060d62240a9e266b8de88e5e060432fe9d > > * FS-4184 --resolve > > anthm committed that one. > > -- > View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Todays-Git-update-has-errors-tp7527470p7527954.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From msc at freeswitch.org Sat May 5 01:46:29 2012 From: msc at freeswitch.org (Michael Collins) Date: Fri, 4 May 2012 14:46:29 -0700 Subject: [Freeswitch-users] mod_fifo how temporary to disable call dispatch to an offline agent In-Reply-To: References: Message-ID: Hi Afshin, Sorry for the delay but I wanted to lab this up and test it before I responded. I do not see the behavior you report. When I use the sample FIFO stuff on the wiki I am able to have an on-hook agent make an outbound call and not receive an inbound call. As soon as the agent completes his outbound call he is available to receive an inbound call. Check this page: http://wiki.freeswitch.org/wiki/Mod_fifo#Simple_On-hook_Agent_Login.2FLogout_Example It works like a champ! -MC On Sat, Apr 28, 2012 at 10:05 PM, afshin afzali wrote: > Dear FreeSWITCHers, > > The offline agents need to make outgoing calls while waiting for call > arrivals. To prevent an influence outgoing by incoming call, they need > to logout, making call and then login to the queue ! also doing logout & > login automatically alters the actual login time :( > I'm looking for a solution which can disable call dispatch temporary for a > specific agent. > Appreciate all comments, > -- afshin > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120504/66c9dced/attachment.html From msc at freeswitch.org Sat May 5 03:01:30 2012 From: msc at freeswitch.org (Michael Collins) Date: Fri, 4 May 2012 16:01:30 -0700 Subject: [Freeswitch-users] Attended transfer via ESL In-Reply-To: <1335539693783-7506468.post@n2.nabble.com> References: <1335539693783-7506468.post@n2.nabble.com> Message-ID: Apologies for the late reply... If you didn't already fix your issue then perhaps you could use this: http://wiki.freeswitch.org/wiki/Mod_commands#uuid_broadcast The uuid_broadcast API lets you execute an arbitrary dp app on a channel. You could execute the att_xfer app on the channel in question. It *should* work just as if it had been called from the dialplan. Let us know how it goes. -MC On Fri, Apr 27, 2012 at 8:14 AM, Limit wrote: > I'm trying to make an attended transfer through EventSocket, using sendmsg > with att_xfer command, but having problems with it. > > ESL function: > execute("att_xfer", args, b_uuid); > > It creates new calls as normal dialplan att_xfer, but problem occured when > one of three sides hangs up. > Sometimes it goes as planned, but more often all channels hangup > automatically. > > att_xfer called from dialplan work as needed, looks like there is some > problem with EventSocket call specifically . Maybe I must set some > variables > before to use it properly? > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Attended-transfer-via-ESL-tp7506468.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120504/c07442fe/attachment.html From msc at freeswitch.org Sat May 5 03:04:54 2012 From: msc at freeswitch.org (Michael Collins) Date: Fri, 4 May 2012 16:04:54 -0700 Subject: [Freeswitch-users] Re ceiving only RTP packets In-Reply-To: <33758451.post@talk.nabble.com> References: <33758451.post@talk.nabble.com> Message-ID: Did you ever resolve this? If not, get a debug console log and put it up on pastebin and then put the pb url here in this thread. -MC On Fri, Apr 27, 2012 at 4:44 AM, rajprithiv89 < rajkumar.kanniappan at sasken.com> wrote: > > Hi, > > we are using csipsimple android voip application which uses Freeswitch as > SIP/Media server. > > We established voip calls between two android clients(A, B) by enabling > SRTP > in both the sides. > > Now the problem observed here is, when we captured the packet flow in > wireshark, the client which initiates the call(A) is sending SRTP packets > and the client which receives the call sends only RTP packets and the > FreeSwitch sends SRTP packets to both the clients. > > We have tested FreeSwitch with other application, there also we ended up in > the same problem(one sided SRTP packets flow). > Also we have tested the same CSipSimple application with Asterisk server. > It > works fine by sending SRTP packets from both the clients. > > > When i captured the Wireshark packets SDP contains: > > > >From A to Server: > > m= audio 4006 RTP/SAVP 98 97 99 > > > > >From Server to B > > m = audio 31562 RTP/SAVP 3 98 99 9 > m = audio 31562 RTP/AVP 3 98 99 9 > (sends two media line, one for SRTP and one for RTP) > > > > >From B to Server(Negotiation done) > > m = audio 4014 RTP/SAVP 3 96 > m = audio 0 RTP/AVP 3 98 99 9 (This is to inform, B is not going to use > RTP by putting the port number as 0) > > > >From Server to A(Negotiation done) > > m = audio 32338 RTP/SAVP 3 96 > > > > But in the end, > A is sending SRTP packets > Server is sending SRTP packets > B is sending only RTP packets (But in the negotiation RTP/AVP port was 0) > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120504/203342ba/attachment.html From msc at freeswitch.org Sat May 5 03:06:31 2012 From: msc at freeswitch.org (Michael Collins) Date: Fri, 4 May 2012 16:06:31 -0700 Subject: [Freeswitch-users] Receiving "early media" during call In-Reply-To: References: Message-ID: Console log w/ siptrace would be extremely helpful. Use pastebin.freeswitch.org and put the pb url in this thread and the gang here will take a look. -MC On Thu, Apr 26, 2012 at 4:06 AM, Pete Kelly wrote: > Hi > > I have a scenario where there is an established call (originated from a > freeswitch bridge to a downstream UAC) > > The downstream UA then initiates a re-invite and freeswitch accepts and > keeps sending the 200OK with SDP as normal. > > However the downstream UA is also sending a 183 with sdp to freeswitch at > this time, and it looks like freeswitch is ignoring this media. > > Is it possible to get freeswitch to accept this 183 media during a > reinvite transaction? > > Pete > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120504/5efbd399/attachment.html From msc at freeswitch.org Sat May 5 03:12:32 2012 From: msc at freeswitch.org (Michael Collins) Date: Fri, 4 May 2012 16:12:32 -0700 Subject: [Freeswitch-users] Meaning of originate_timeout In-Reply-To: References: Message-ID: On Wed, Apr 25, 2012 at 1:15 PM, Malay Thakershi wrote: > Hello, > > When doing an originate, I set originate_timeout=45 > > Does it mean the called phone will ring for maximum 45 seconds? > > Essentially yes. The called phone may appear to ring for fewer than 45 seconds if it takes some time to establish the call. Keep in mind that originate_timeout is for the entire originate operation. You could have different leg timeout values if you're calling more than one endpoint simultaneously. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120504/ac19e662/attachment.html From msc at freeswitch.org Sat May 5 03:18:28 2012 From: msc at freeswitch.org (Michael Collins) Date: Fri, 4 May 2012 16:18:28 -0700 Subject: [Freeswitch-users] Park and Page use case In-Reply-To: References: Message-ID: Ian, First off, a belated welcome to the FreeSWITCH community! Apologies for the tardiness of my reply - it's been a bit nuts around here. :) Regarding your calls to multiple Polycom phones, I recommend that you look at the "mad boss" example in the default.xml dialplan file. I've used it with Polycoms without issue. I must confess, though, that I've never tried it on a phone w/ multiple line keys. I'd love to have you try it out and give us some feedback. Thanks! -Michael On Mon, Apr 23, 2012 at 7:12 PM, Ian McMaster wrote: > As this is my first question, please let me first praise the FreeSWITCH > team for a fantastic product and all the efforts of the community. The two > books really helped convert all my Asterisk AGI / AMI java code to > FreeSWITCH. All IVR play/record/dtmf, etc is working perfectly with > FreeSwitch now (and the IVR application is quite large). > I came to FreeSwitch after limitations/frustrations with Asterisk. Happy > to be aboard, and hope to contribute in time. > > Use case: Controlling an IVR caller using the event socket; caller > chooses an option which will park the caller, and page out to a group of > phones for someone to pick up the parked caller. If the park is answered > the script exits, or if the park times out then an alternate action is > taken. > > The valet_park command is working fine, and I can decipher through the > ChannelExecuteComplete event whether the park was retrieved or not. > > I am struggling a bit with the paging to multiple polycom phones. > > - using something like 'originate > {sip_auto_answer=true}user/1001,user/1002,... > &playback(announce_page.PCMU) has limited success - my understanding is > that the first phone to answer 'wins' and the other phones do not receive > the playback. > - I have better success with conference > (a) Invite phones into the conference: conference page_group dial > {sip_auto_answer=true}user/1001 ... etc > (b) Play file: conference page_group play announce_page.PCMU (although I > am having a codec problem here!) > (c) End the conference: conference page_group hup all > While the conference by nature has certain pre-stocked files (i.e. you are > the first caller, and music on hold, etc), I believe I can silence them all > with a custom conference. > > My only show stopper is that if a page goes out to a phone with two line > buttons, and is already in use (on one line), then things go bad. The > incoming conference request is knocking the existing call on hold, the page > is playing out, and the user must resume the call with a soft button. > > Questions: > 1) Is there a better approach to paging multiple phones that I have yet > to try? > 2) Is there a way to avoid paging a phone that already has a caller on it? > 3) Is there conference variable to set the codec to allow the PCMU file > to play cleanly (it does play cleanly using originate without setting any > explicit codec). > > Thank you in advance, > Ian. > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120504/89cd0fe7/attachment-0001.html From xyangni at gmail.com Sat May 5 03:23:01 2012 From: xyangni at gmail.com (Yihui Li) Date: Sat, 5 May 2012 00:23:01 +0100 Subject: [Freeswitch-users] gsmopen error message Message-ID: Hi, I am excited by new mod_gsmopen with dongle and trying it with my E1550. The function is OK. But I got some error message when sending SMS Quote freeswitch at ubuntu> chat SMS|gsm01|07575565692|test 2012-05-04 23:40:11.102053 [ERR] gsmopen_protocol.cpp:574 rev 2431e0f|de019ab[(nil)|37 ][ERRORA 574 ][gsm01 ][-1, 0, 0] TIMEOUT=1100 2012-05-04 23:40:11.182283 [ERR] gsmopen_protocol.cpp:574 rev 2431e0f|de019ab[(nil)|37 ][ERRORA 574 ][gsm01 ][-1, 0, 0] TIMEOUT=5500 Sent Unquote It is under the default configuration. Does anyone know the meaning and how to avoid it? Thanks Regards Eric -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120505/34e026b5/attachment.html From gmaruzz at gmail.com Sat May 5 04:40:17 2012 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Sat, 5 May 2012 02:40:17 +0200 Subject: [Freeswitch-users] gsmopen error message In-Reply-To: References: Message-ID: On Sat, May 5, 2012 at 1:23 AM, Yihui Li wrote: > Hi, > > I am excited by new mod_gsmopen with dongle and trying it with my E1550. The > function is OK. But I got some error message when sending SMS > > Quote > freeswitch at ubuntu> chat SMS|gsm01|07575565692|test > 2012-05-04 23:40:11.102053 [ERR] gsmopen_protocol.cpp:574 rev > 2431e0f|de019ab[(nil)|37???? ][ERRORA? 574? ][gsm01???? ][-1, 0, 0] > TIMEOUT=1100 > 2012-05-04 23:40:11.182283 [ERR] gsmopen_protocol.cpp:574 rev > 2431e0f|de019ab[(nil)|37???? ][ERRORA? 574? ][gsm01???? ][-1, 0, 0] > TIMEOUT=5500 Nothing wrong. Sorry, it was a leftover of a debug setup I was using. Since some commit ago it's fixed (it just show itself as "DEBUG"). -giovanni -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From gmaruzz at gmail.com Sat May 5 04:41:12 2012 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Sat, 5 May 2012 02:41:12 +0200 Subject: [Freeswitch-users] gsmopen error message In-Reply-To: References: Message-ID: On Sat, May 5, 2012 at 2:40 AM, Giovanni Maruzzelli wrote: > On Sat, May 5, 2012 at 1:23 AM, Yihui Li wrote: >> Hi, >> >> I am excited by new mod_gsmopen with dongle and trying it with my E1550. The >> function is OK. But I got some error message when sending SMS >> >> Quote >> freeswitch at ubuntu> chat SMS|gsm01|07575565692|test >> 2012-05-04 23:40:11.102053 [ERR] gsmopen_protocol.cpp:574 rev >> 2431e0f|de019ab[(nil)|37???? ][ERRORA? 574? ][gsm01???? ][-1, 0, 0] >> TIMEOUT=1100 >> 2012-05-04 23:40:11.182283 [ERR] gsmopen_protocol.cpp:574 rev >> 2431e0f|de019ab[(nil)|37???? ][ERRORA? 574? ][gsm01???? ][-1, 0, 0] >> TIMEOUT=5500 > > Nothing wrong. > > Sorry, it was a leftover of a debug setup I was using. > > Since some commit ago it's fixed (it just show itself as "DEBUG"). Please update to latest git. Thanks for reporting! -giovanni -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From anthony.minessale at gmail.com Sat May 5 06:11:37 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 4 May 2012 21:11:37 -0500 Subject: [Freeswitch-users] New proof-of-concept module (mod_pickupgroup) In-Reply-To: References: Message-ID: Jo?o, After talking to you about this earlier in the week and having a second person making the same assumptions with intercept, I decided to implement the pickup endpoint/app pair. Basically the pickup endpoint is a dummy channel that never answers that you can originate to from anywhere you can place calls, call it alone or place it in a list for a forked dial. originate sofia/internal/100 at test.com,pickup/mygroup now the pickup channel will be created alongside the sip channel. >From another call route the call to the application pickup with data "mygroup" When you place that call, your session will be returned from the originate in place of the pickup/mygroup call being placed. It also works with presence so you could assign blf buttons to it. pickup+mygroup The group names can also have an @domain.com for multihoming etc. On Fri, May 4, 2012 at 3:27 PM, Vik Killa wrote: > +1 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From zhulizhong at live.com Sat May 5 04:56:56 2012 From: zhulizhong at live.com (James zhu) Date: Sat, 5 May 2012 00:56:56 +0000 Subject: [Freeswitch-users] Openvox tdma400p fail to Disconnect/Hangup detect (Dahdi/Freetdm) In-Reply-To: References: Message-ID: please show the conf files. Best regards, James.zhu Doing asterisk/PRI/ss7/dahdi, linux, asterisk/sangoma cards, recording device, VOIP gateway. website: www.hiastar.com > Date: Fri, 4 May 2012 10:33:52 -0430 > From: rrodolfos at gmail.com > To: FreeSWITCH-users at lists.freeswitch.org > Subject: [Freeswitch-users] Openvox tdma400p fail to Disconnect/Hangup detect (Dahdi/Freetdm) > > Hi, > > Freeswitch can recive calls, but the call is end, openvox tdma400p, > can't detect disconnection or hangup. > > Any idea? > > RrodolfoS > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120505/62f27663/attachment.html From anton.jugatsu at gmail.com Sat May 5 09:01:23 2012 From: anton.jugatsu at gmail.com (Anton Kvashenkin) Date: Sat, 5 May 2012 09:01:23 +0400 Subject: [Freeswitch-users] gsmopen error message In-Reply-To: References: Message-ID: For better tracking the bugs it's better to use JIRA. 2012/5/5 Giovanni Maruzzelli > On Sat, May 5, 2012 at 2:40 AM, Giovanni Maruzzelli > wrote: > > On Sat, May 5, 2012 at 1:23 AM, Yihui Li wrote: > >> Hi, > >> > >> I am excited by new mod_gsmopen with dongle and trying it with my > E1550. The > >> function is OK. But I got some error message when sending SMS > >> > >> Quote > >> freeswitch at ubuntu> chat SMS|gsm01|07575565692|test > >> 2012-05-04 23:40:11.102053 [ERR] gsmopen_protocol.cpp:574 rev > >> 2431e0f|de019ab[(nil)|37 ][ERRORA 574 ][gsm01 ][-1, 0, 0] > >> TIMEOUT=1100 > >> 2012-05-04 23:40:11.182283 [ERR] gsmopen_protocol.cpp:574 rev > >> 2431e0f|de019ab[(nil)|37 ][ERRORA 574 ][gsm01 ][-1, 0, 0] > >> TIMEOUT=5500 > > > > Nothing wrong. > > > > Sorry, it was a leftover of a debug setup I was using. > > > > Since some commit ago it's fixed (it just show itself as "DEBUG"). > > Please update to latest git. > > Thanks for reporting! > > -giovanni > > -- > Sincerely, > > Giovanni Maruzzelli > Cell : +39-347-2665618 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120505/b5cd50f2/attachment.html From anton.jugatsu at gmail.com Sat May 5 09:06:27 2012 From: anton.jugatsu at gmail.com (Anton Kvashenkin) Date: Sat, 5 May 2012 09:06:27 +0400 Subject: [Freeswitch-users] Re ceiving only RTP packets In-Reply-To: References: <33758451.post@talk.nabble.com> Message-ID: Very handy tool indeed fs_logger.pl. 2012/5/5 Michael Collins > Did you ever resolve this? If not, get a debug console log and put it up > on pastebin and then put the pb url here in this thread. > -MC > > > On Fri, Apr 27, 2012 at 4:44 AM, rajprithiv89 < > rajkumar.kanniappan at sasken.com> wrote: > >> >> Hi, >> >> we are using csipsimple android voip application which uses Freeswitch as >> SIP/Media server. >> >> We established voip calls between two android clients(A, B) by enabling >> SRTP >> in both the sides. >> >> Now the problem observed here is, when we captured the packet flow in >> wireshark, the client which initiates the call(A) is sending SRTP packets >> and the client which receives the call sends only RTP packets and the >> FreeSwitch sends SRTP packets to both the clients. >> >> We have tested FreeSwitch with other application, there also we ended up >> in >> the same problem(one sided SRTP packets flow). >> Also we have tested the same CSipSimple application with Asterisk server. >> It >> works fine by sending SRTP packets from both the clients. >> >> >> When i captured the Wireshark packets SDP contains: >> >> >> >From A to Server: >> >> m= audio 4006 RTP/SAVP 98 97 99 >> >> >> >> >From Server to B >> >> m = audio 31562 RTP/SAVP 3 98 99 9 >> m = audio 31562 RTP/AVP 3 98 99 9 >> (sends two media line, one for SRTP and one for RTP) >> >> >> >> >From B to Server(Negotiation done) >> >> m = audio 4014 RTP/SAVP 3 96 >> m = audio 0 RTP/AVP 3 98 99 9 (This is to inform, B is not going to use >> RTP by putting the port number as 0) >> >> >> >From Server to A(Negotiation done) >> >> m = audio 32338 RTP/SAVP 3 96 >> >> >> >> But in the end, >> A is sending SRTP packets >> Server is sending SRTP packets >> B is sending only RTP packets (But in the negotiation RTP/AVP port was 0) >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120505/c0b08249/attachment-0001.html From thaddeus at thogan.com Sat May 5 09:12:32 2012 From: thaddeus at thogan.com (Thaddeus Hogan) Date: Sat, 05 May 2012 00:12:32 -0500 Subject: [Freeswitch-users] Dialplan condition to test if SIP client is registered In-Reply-To: <20120504182445.9e296b96@mail.tritonwest.net> References: <20120504182445.9e296b96@mail.tritonwest.net> Message-ID: <4FA4B6C0.1070700@thogan.com> Thanks for the information. I got something working today using sofia_contact, but am having issues with SIP registrations surviving for a good deal of time after the user has disconnected from the network. Basically, the SIP client is no longer available when the user leaves the building and is disconnected from Wifi. When this happens, sofia_contact continues to return a value indicating the client is registered for quite some time. I found the command "sofia profile check_sync", which I used successfully to invalidate registrations for users that have disconnected from the network. But I could not figure out how to make this execute before my "sofia_contact" condition in the dialplan. I am assuming this is because I need it to run before the ROUTING stage, and I can only get it to run in the EXECUTE stage. So I have a couple questions. Is check_sync quick enough to run on every inbound call prior to making routing decisions? Also, according to the comments here http://wiki.freeswitch.org/wiki/FS_weekly_2011_06_29 I should be able to give check_sync a user at domain argument, but this does not seem to work. Are there other ways to determine if a bridge failed because a user's SIP client was not reachable, versus a timeout or call rejection? Thanks for helping with this! -- Thaddeus On 5/4/2012 1:24 PM, Dave R. Kompel wrote: > Thaddeus & Michael: > As Michael stated there are many ways to fail over if a phone/user > isn't registered. However sometimes you may want to actually check the > registration yourself. Or even do something based on a contact being > registered, that has nothing to do with the actual one you are calling. > I normally use the API: sofia_contact([profile/]@domain). It > returns the channel string you would pass to bridge. If the user isn't > registered it returns "error/" so it would use the logical > "error" endpoint to handle the right treatment. If the regex "^error" > match is true, then the user isn't registered. > The extra thing you get from that API call is that you can use a "*" > for the profile, and find out which SIP profile the user is registered > in. For example in the case of an Android client, that may be using > WIFI or 3G, and you have an external and internal profile they can > register, you can tell if the user is "on site" or "off site" by the > profile it returns, and can be more restrictive of the calls you send > to someone when they're out of the office. > --Dave > > ------------------------------------------------------------------------ > *From:* Michael Collins [mailto:msc at freeswitch.org] > *To:* FreeSWITCH Users Help > [mailto:freeswitch-users at lists.freeswitch.org] > *Sent:* Fri, 04 May 2012 10:34:31 -0700 > *Subject:* Re: [Freeswitch-users] Dialplan condition to test if > SIP client is registered > > Hi Thaddeus, > > You definitely have several options here. While you could use the > sofia_contact API, I'm wondering if perhaps you could let the > dialplan and the bridge app do the work for you. > > You could bridge directly to the Android user and then if it fails > it could then continue in the dialplan and ring the mobile number. > This is covered briefly on the wiki > , > however it is covered very nicely in the new > FreeSWITCH Cookbook in chapter 1. > > Try out the basic failover (pipe-separated list of endpoints) and > if that doesn't work let us know. Maybe you could hop on IRC and > talk live with other community members. Of course, it can't hurt > to raise your karma by getting the FreeSWITCH books. :) (See > freeswitch.org , upper left corner.) > > Thanks, > Michael > > On Fri, May 4, 2012 at 10:10 AM, thaddeus at thogan.com > > wrote: > > I was wondering if there is a way to tell if a SIP client is > registered with a dialplan condition? Or maybe a better way to > accomplish the following?: > > The users' cell phones (Android) are running a SIP client and > connect to freeswitch via Wifi when in the office. I want > calls destined for a given user to ring only their extension > if they are connected via SIP, and ring only their mobile > number when they are not connected. > > Currently I just take every incoming call and bridge it back > out to their mobile number, but 95% of the time these users > are on the Wifi network, and I could just pass the call to > their phones via SIP. > > I have tested hunt groups but the callers hear ringing for far > too long. Ring groups don't work because it is confusing when > the user simultaneously receiving the same call via SIP and > mobile on the same phone. > > Thanks in advance for any insights and help! > > -- Thaddeus > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120505/39a7c764/attachment.html From anton.jugatsu at gmail.com Sat May 5 09:29:49 2012 From: anton.jugatsu at gmail.com (Anton Kvashenkin) Date: Sat, 5 May 2012 09:29:49 +0400 Subject: [Freeswitch-users] Dialplan condition to test if SIP client is registered In-Reply-To: <4FA4B6C0.1070700@thogan.com> References: <20120504182445.9e296b96@mail.tritonwest.net> <4FA4B6C0.1070700@thogan.com> Message-ID: What about continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION 2012/5/5 Thaddeus Hogan > Thanks for the information. I got something working today using > sofia_contact, but am having issues with SIP registrations surviving for a > good deal of time after the user has disconnected from the network. > > Basically, the SIP client is no longer available when the user leaves the > building and is disconnected from Wifi. When this happens, sofia_contact > continues to return a value indicating the client is registered for quite > some time. > > I found the command "sofia profile check_sync", which I > used successfully to invalidate registrations for users that have > disconnected from the network. But I could not figure out how to make this > execute before my "sofia_contact" condition in the dialplan. I am assuming > this is because I need it to run before the ROUTING stage, and I can only > get it to run in the EXECUTE stage. > > So I have a couple questions. Is check_sync quick enough to run on every > inbound call prior to making routing decisions? Also, according to the > comments here http://wiki.freeswitch.org/wiki/FS_weekly_2011_06_29 I > should be able to give check_sync a user at domain argument, but this does > not seem to work. > > Are there other ways to determine if a bridge failed because a user's SIP > client was not reachable, versus a timeout or call rejection? > > Thanks for helping with this! > > -- Thaddeus > > > On 5/4/2012 1:24 PM, Dave R. Kompel wrote: > > Thaddeus & Michael: > > As Michael stated there are many ways to fail over if a phone/user isn't > registered. However sometimes you may want to actually check the > registration yourself. Or even do something based on a contact being > registered, that has nothing to do with the actual one you are calling. > > I normally use the API: sofia_contact([profile/]@domain). It returns > the channel string you would pass to bridge. If the user isn't registered > it returns "error/" so it would use the logical "error" endpoint > to handle the right treatment. If the regex "^error" match is true, then > the user isn't registered. > > The extra thing you get from that API call is that you can use a "*" for > the profile, and find out which SIP profile the user is registered in. For > example in the case of an Android client, that may be using WIFI or 3G, and > you have an external and internal profile they can register, you can tell > if the user is "on site" or "off site" by the profile it returns, and can > be more restrictive of the calls you send to someone when they're out of > the office. > > --Dave > > ------------------------------ > *From:* Michael Collins [mailto:msc at freeswitch.org ] > *To:* FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org > ] > *Sent:* Fri, 04 May 2012 10:34:31 -0700 > *Subject:* Re: [Freeswitch-users] Dialplan condition to test if SIP > client is registered > > Hi Thaddeus, > > You definitely have several options here. While you could use the > sofia_contact API, I'm wondering if perhaps you could let the dialplan and > the bridge app do the work for you. > > You could bridge directly to the Android user and then if it fails it > could then continue in the dialplan and ring the mobile number. This is > covered briefly on the wiki, > however it is covered very nicely in the new FreeSWITCH > Cookbook in chapter 1. > > Try out the basic failover (pipe-separated list of endpoints) and if that > doesn't work let us know. Maybe you could hop on IRC and talk live with > other community members. Of course, it can't hurt to raise your karma by > getting the FreeSWITCH books. :) (See freeswitch.org, upper left corner.) > > Thanks, > Michael > > On Fri, May 4, 2012 at 10:10 AM, thaddeus at thogan.com wrote: > >> I was wondering if there is a way to tell if a SIP client is registered >> with a dialplan condition? Or maybe a better way to accomplish the >> following?: >> >> The users' cell phones (Android) are running a SIP client and connect to >> freeswitch via Wifi when in the office. I want calls destined for a given >> user to ring only their extension if they are connected via SIP, and ring >> only their mobile number when they are not connected. >> >> Currently I just take every incoming call and bridge it back out to their >> mobile number, but 95% of the time these users are on the Wifi network, and >> I could just pass the call to their phones via SIP. >> >> I have tested hunt groups but the callers hear ringing for far too long. >> Ring groups don't work because it is confusing when the user simultaneously >> receiving the same call via SIP and mobile on the same phone. >> >> Thanks in advance for any insights and help! >> >> -- Thaddeus >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120505/219761ef/attachment-0001.html From gabe at gundy.org Sat May 5 09:45:39 2012 From: gabe at gundy.org (Gabriel Gunderson) Date: Fri, 4 May 2012 23:45:39 -0600 Subject: [Freeswitch-users] FreeSWITCH 1.2 and packaging update In-Reply-To: References: Message-ID: On Fri, May 4, 2012 at 1:20 PM, Ken Rice wrote: > If you arent currently using the debian package building bits of the > FreeSWITCH source tree, you will not be affected. However if you are expect > to see some big changes... > > These changes are to move us more inline with the Centos packages and more > inline with what is required to get FreeSWITCH in the normal Debian > repositories. Would it be helpful to build and report on different versions of Ubuntu and Debian? BTW, thanks for the excellent work in this area! Gabe From vitaliy.davudov at vts24.ru Sat May 5 11:12:22 2012 From: vitaliy.davudov at vts24.ru (=?UTF-8?B?0JLQuNGC0LDQu9C40Lkg0JTQsNCy0YPQtNC+0LI=?=) Date: Sat, 05 May 2012 11:12:22 +0400 Subject: [Freeswitch-users] Variable continue on fail In-Reply-To: <4F8822A5.9090402@vts24.ru> References: <4F87EF60.3070105@vts24.ru> <4F8822A5.9090402@vts24.ru> Message-ID: <4FA4D2D6.1010908@vts24.ru> Hi! Is there any news on this issue? 13.04.2012 16:57, ??????? ??????? ???????: > Yes, you are right! > > I did it: http://pastebin.freeswitch.org/18863 > > Additionally: > I've included in this extension new line: > > > > > ** > data="continue_on_fail=1,2,3,6,25,34,38,41,42,44,47,63,66,500,501"/> > > > > > > > Without that line a similar situation occurs if FS recieve > /NORMAL_CLEARING./ > > 13.04.2012 13:44, Anton Kvashenkin ???????: >> Ok, i got it. Even that there is no USER_BUSY at continue_on_fail >> variable, FS still tries to reach the second action, am i right? >> >> So, for better debugging, i suggest to paste full call log with >> enabled siptrace and /log 7 to pastebin.freeswitch.org >> . >> >> 13 ?????? 2012 ?. 13:18 ???????????? ??????? ??????? >> > ???????: >> >> Hi all! >> >> In my dialplan I've included variable continue on fail: >> >> >> >> >> > data="continue_on_fail=1,2,3,6,25,34,38,41,42,44,47,63,66,500,501"/> >> >> >> >> >> >> >> And if FS recieve from first gateway USER_BUSY, then FS try to >> bridge >> this call to another gateway. Although in line > application="set" >> data="continue_on_fail=1,2,3,6,25,34,38,41,42,44,47,63,66,500,501"/> >> there is no code Q.850 = 17. >> How resolve this issue? >> >> -- >> Best regards, >> Vitaly Davudov >> "VIP-TELECOM-SERVICE" Ltd. >> ("ETERIA" Group of companies) >> http://www.vts24.ru >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > -- > ? ?????????? ???????????, > ??????? ??????? ????????? > ??? "???-???????-??????" > (?????? ???????? "ETERIA") > http://www.vts24.ru > ???: (495) 989-47-00 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- ? ?????????? ???????????, ??????? ??????? ????????? ??? "???-???????-??????" (?????? ???????? "ETERIA") http://www.vts24.ru ???: (495) 989-47-00 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120505/5734139a/attachment.html From anton.jugatsu at gmail.com Sat May 5 11:19:35 2012 From: anton.jugatsu at gmail.com (Anton Kvashenkin) Date: Sat, 5 May 2012 11:19:35 +0400 Subject: [Freeswitch-users] Variable continue on fail In-Reply-To: <4FA4D2D6.1010908@vts24.ru> References: <4F87EF60.3070105@vts24.ru> <4F8822A5.9090402@vts24.ru> <4FA4D2D6.1010908@vts24.ru> Message-ID: Which version of FreeSWITCH do you use? I suggest to update to lattest git and try to reproduce this issue. 2012/5/5 ??????? ??????? > Hi! > Is there any news on this issue? > > 13.04.2012 16:57, ??????? ??????? ???????: > > Yes, you are right! > > I did it: http://pastebin.freeswitch.org/18863 > > Additionally: > I've included in this extension new line: > > > > > ** > data="continue_on_fail=1,2,3,6,25,34,38,41,42,44,47,63,66,500,501"/> > > > > > > > Without that line a similar situation occurs if FS recieve * > NORMAL_CLEARING.* > > 13.04.2012 13:44, Anton Kvashenkin ???????: > > Ok, i got it. Even that there is no USER_BUSY at continue_on_fail > variable, FS still tries to reach the second action, am i right? > > So, for better debugging, i suggest to paste full call log with enabled > siptrace and /log 7 to pastebin.freeswitch.org. > > 13 ?????? 2012 ?. 13:18 ???????????? ??????? ??????? < > vitaliy.davudov at vts24.ru> ???????: > >> Hi all! >> >> In my dialplan I've included variable continue on fail: >> >> >> >> >> > data="continue_on_fail=1,2,3,6,25,34,38,41,42,44,47,63,66,500,501"/> >> >> >> >> >> >> >> And if FS recieve from first gateway USER_BUSY, then FS try to bridge >> this call to another gateway. Although in line > data="continue_on_fail=1,2,3,6,25,34,38,41,42,44,47,63,66,500,501"/> >> there is no code Q.850 = 17. >> How resolve this issue? >> >> -- >> Best regards, >> Vitaly Davudov >> "VIP-TELECOM-SERVICE" Ltd. >> ("ETERIA" Group of companies) >> http://www.vts24.ru >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > -- > ? ?????????? ???????????, > ??????? ??????? ????????? > ??? "???-???????-??????" > (?????? ???????? "ETERIA")http://www.vts24.ru > ???: (495) 989-47-00 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > -- > ? ?????????? ???????????, > ??????? ??????? ????????? > ??? "???-???????-??????" > (?????? ???????? "ETERIA")http://www.vts24.ru > ???: (495) 989-47-00 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120505/b14442e3/attachment-0001.html From vitaliy.davudov at vts24.ru Sat May 5 16:47:22 2012 From: vitaliy.davudov at vts24.ru (=?UTF-8?B?0JLQuNGC0LDQu9C40Lkg0JTQsNCy0YPQtNC+0LI=?=) Date: Sat, 05 May 2012 16:47:22 +0400 Subject: [Freeswitch-users] Variable continue on fail In-Reply-To: References: <4F87EF60.3070105@vts24.ru> <4F8822A5.9090402@vts24.ru> <4FA4D2D6.1010908@vts24.ru> Message-ID: <4FA5215A.1020400@vts24.ru> Today I updated FS. Version is: freeswitch at internal> version FreeSWITCH Version 1.1.beta1 (git-073e405 2012-05-04 19-48-31 -0500) But nothing changed... 05.05.2012 11:19, Anton Kvashenkin ???????: > Which version of FreeSWITCH do you use? I suggest to update to lattest > git and try to reproduce this issue. > > 2012/5/5 ??????? ??????? > > > Hi! > Is there any news on this issue? > > 13.04.2012 16:57, ??????? ??????? ???????: >> Yes, you are right! >> >> I did it: http://pastebin.freeswitch.org/18863 >> >> Additionally: >> I've included in this extension new line: >> >> >> >> >> ** >> > data="continue_on_fail=1,2,3,6,25,34,38,41,42,44,47,63,66,500,501"/> >> >> >> >> >> >> >> Without that line a similar situation occurs if FS recieve >> /NORMAL_CLEARING./ >> >> 13.04.2012 13:44, Anton Kvashenkin ???????: >>> Ok, i got it. Even that there is no USER_BUSY at >>> continue_on_fail variable, FS still tries to reach the second >>> action, am i right? >>> >>> So, for better debugging, i suggest to paste full call log with >>> enabled siptrace and /log 7 to pastebin.freeswitch.org >>> . >>> >>> 13 ?????? 2012 ?. 13:18 ???????????? ??????? ??????? >>> > >>> ???????: >>> >>> Hi all! >>> >>> In my dialplan I've included variable continue on fail: >>> >>> >>> >>> >>> >> data="continue_on_fail=1,2,3,6,25,34,38,41,42,44,47,63,66,500,501"/> >>> >>> >>> >>> >>> >>> >>> And if FS recieve from first gateway USER_BUSY, then FS try >>> to bridge >>> this call to another gateway. Although in line >> application="set" >>> data="continue_on_fail=1,2,3,6,25,34,38,41,42,44,47,63,66,500,501"/> >>> there is no code Q.850 = 17. >>> How resolve this issue? >>> >>> -- >>> Best regards, >>> Vitaly Davudov >>> "VIP-TELECOM-SERVICE" Ltd. >>> ("ETERIA" Group of companies) >>> http://www.vts24.ru >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> -- >> ? ?????????? ???????????, >> ??????? ??????? ????????? >> ??? "???-???????-??????" >> (?????? ???????? "ETERIA") >> http://www.vts24.ru >> ???: (495) 989-47-00 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > -- > ? ?????????? ???????????, > ??????? ??????? ????????? > ??? "???-???????-??????" > (?????? ???????? "ETERIA") > http://www.vts24.ru > ???: (495) 989-47-00 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- ? ?????????? ???????????, ??????? ??????? ????????? ??? "???-???????-??????" (?????? ???????? "ETERIA") http://www.vts24.ru ???: (495) 989-47-00 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120505/0ca1af95/attachment-0001.html From dujinfang at gmail.com Sat May 5 18:48:36 2012 From: dujinfang at gmail.com (Seven Du) Date: Sat, 5 May 2012 22:48:36 +0800 Subject: [Freeswitch-users] mod_fifo how temporary to disable call dispatch to an offline agent In-Reply-To: References: Message-ID: <2740A3FE1B6E4D288416E9E1160B0C7A@gmail.com> Hi Michael, I tested that example and it did call the agent(Bria or any phone accept multi-lines) no matter it is busy or not, and I also looked the code looks there's no code to prevent this from happening. Michael I guess you used a phone only support 1 sip line, so any new incoming call will be rejected with USER_BUSY and fifo will try every seconds but only it will success when you hangup the current call. We once used a custom build sip ua with this solution, but leaves unsuccessful calls every second in logs if agents are "busy" calling out. I also tried to find a way to automatically "seize" an agent, but, technically, there's race conditions anyway. Say, agent send INVITE and it might receive INVITE at the same time because mod_fifo has no way to know the agent is start making a call. I currently use a temporary "call-back" solution: when client want to make a call, it send a request to my ESL app, and I update the fifo_outbound table to "seize" the agent line, if success(1 record line updated), then I originate the agent... It is a pain to make sure the "seized" line is cleared whenever the call is hangup or fail. It is kind of works in our lab, but I'm not sure how reliable it is. I had thought to add some code to mod_fifo, so in addition to "fifo out" add something like "fifo dial from to" to automatically resolve this. It seems no solution in mod_callcenter either it might be because most callcenters are either inbound or outbound, but in-out mixed agents are also seen for efficiency so it would be helpful to find a solution. Seven. On Saturday, May 5, 2012 at 5:46 AM, Michael Collins wrote: > Hi Afshin, > > Sorry for the delay but I wanted to lab this up and test it before I responded. I do not see the behavior you report. When I use the sample FIFO stuff on the wiki I am able to have an on-hook agent make an outbound call and not receive an inbound call. As soon as the agent completes his outbound call he is available to receive an inbound call. > > Check this page: > http://wiki.freeswitch.org/wiki/Mod_fifo#Simple_On-hook_Agent_Login.2FLogout_Example > > It works like a champ! > > -MC > > On Sat, Apr 28, 2012 at 10:05 PM, afshin afzali wrote: > > Dear FreeSWITCHers, > > > > The offline agents need to make outgoing calls while waiting for call arrivals. To prevent an influence outgoing by incoming call, they need to logout, making call and then login to the queue ! also doing logout & login automatically alters the actual login time :( > > I'm looking for a solution which can disable call dispatch temporary for a specific agent. > > Appreciate all comments, > > -- afshin > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120505/46fc5eaf/attachment.html From dujinfang at gmail.com Sat May 5 19:00:07 2012 From: dujinfang at gmail.com (Seven Du) Date: Sat, 5 May 2012 23:00:07 +0800 Subject: [Freeswitch-users] Attended transfer via ESL In-Reply-To: References: <1335539693783-7506468.post@n2.nabble.com> Message-ID: Cool I tried with pa call 1000 uuid_broadcast att_xfer:user/1001 and looks like it works well (because all my clients on the same computer so not 100% sure the sound was right) looking back to the Limit's ESL issue, it seems the sendmsg breaks some calling flow, here's a jira you may want to update http://jira.freeswitch.org/browse/FS-4018 On Saturday, May 5, 2012 at 7:01 AM, Michael Collins wrote: > Apologies for the late reply... > > If you didn't already fix your issue then perhaps you could use this: > http://wiki.freeswitch.org/wiki/Mod_commands#uuid_broadcast > > The uuid_broadcast API lets you execute an arbitrary dp app on a channel. You could execute the att_xfer app on the channel in question. It *should* work just as if it had been called from the dialplan. > > Let us know how it goes. > > -MC > > On Fri, Apr 27, 2012 at 8:14 AM, Limit wrote: > > I'm trying to make an attended transfer through EventSocket, using sendmsg > > with att_xfer command, but having problems with it. > > > > ESL function: > > execute("att_xfer", args, b_uuid); > > > > It creates new calls as normal dialplan att_xfer, but problem occured when > > one of three sides hangs up. > > Sometimes it goes as planned, but more often all channels hangup > > automatically. > > > > att_xfer called from dialplan work as needed, looks like there is some > > problem with EventSocket call specifically . Maybe I must set some variables > > before to use it properly? > > > > -- > > View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Attended-transfer-via-ESL-tp7506468.html > > Sent from the freeswitch-users mailing list archive at Nabble.com (http://Nabble.com). > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at 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 at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120505/567ba211/attachment.html From rrodolfos at gmail.com Sat May 5 19:46:31 2012 From: rrodolfos at gmail.com (RrodolfoS .) Date: Sat, 5 May 2012 11:16:31 -0430 Subject: [Freeswitch-users] Openvox tdma400p fail to Disconnect/Hangup detect (Dahdi/Freetdm) In-Reply-To: References: Message-ID: James, In freeswitch console (Cli) show this: 2012-05-05 10:10:33.464451 [ERR] ftmod_zt.c:1137 [s1c1][1:1] Got polarity reverse (ZT_EVENT_POLARITY) 2012-05-05 10:10:33.464451 [WARNING] ftdm_io.c:1786 [s1c1][1:1] Cannot open channel when is alarmed 2012-05-05 10:10:33.464451 [ERR] ftmod_analog.c:444 [s1c1][1:1] OPEN ERROR [Channel is alarmed ] 2012-05-05 10:10:33.464451 [WARNING] ftdm_io.c:2765 [s1c1][1:1] Channel not opened, proceeding anyway 2012-05-05 10:10:34.404449 [NOTICE] mod_freetdm.c:1887 Alarm cleared on channel 1:1 . . . 2012-05-05 10:13:06.704451 [ERR] ftmod_zt.c:1137 [s1c1][1:1] Got polarity reverse (ZT_EVENT_POLARITY) 2012-05-05 10:13:06.724453 [WARNING] ftmod_analog.c:621 [s1c1][1:1] Not hanging up on polarity reverse, too close to Answer reverse Details: FreeSWITCH Version 1.1.beta1 (git-4283408 2012-04-29 11-33-24 -0400) dahdi-linux-complete-2.6.1+2.6.1 downloaded from asterisk /etc/dahdi/system.conf fxsks=1 echocanceller=mg2,1 loadzone = ve defaultzone = ve /usr/local/freeswitch/conf/freetdm.conf [general] cpu_monitor => no cpu_monitoring_interval => 1000 cpu_set_alarm_threshold => 80 cpu_reset_alarm_threshold => 70 cpu_alarm_action => warn [span zt FXO1] fxo-channel => 1 /usr/local/freeswitch/conf/autoload_configs/freetdm.conf.xml RrodolfoS On Fri, May 4, 2012 at 8:26 PM, James zhu wrote: > please show the conf files. > Best regards, > James.zhu > Doing asterisk/PRI/ss7/dahdi, linux, asterisk/sangoma cards, recording > device, VOIP gateway. > website: www.hiastar.com > > >> Date: Fri, 4 May 2012 10:33:52 -0430 >> From: rrodolfos at gmail.com >> To: FreeSWITCH-users at lists.freeswitch.org >> Subject: [Freeswitch-users] Openvox tdma400p fail to Disconnect/Hangup >> detect (Dahdi/Freetdm) > >> >> Hi, >> >> Freeswitch can recive calls, but the call is end, openvox tdma400p, >> can't detect disconnection or hangup. >> >> Any idea? >> >> RrodolfoS >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From bernard.david.murphy at gmail.com Sat May 5 20:15:20 2012 From: bernard.david.murphy at gmail.com (Bernard Murphy) Date: Sat, 5 May 2012 17:15:20 +0100 Subject: [Freeswitch-users] SIP NOTIFY for MWI with AVAYA 9630 Message-ID: On anything newer than 1.06 of freeswitch the SIP NOTIFY sent to an Avaya 9630 phone does not get accepted and trigger the MWI light. Works absolutely fine on 1.0.6. The only difference in the messages i can see is in the contact parameter, possibly to do with the angle bracket. Can anyone advise if the difference shown in the two messages below is acceptable SIP or if something is broken in freeswitch. Also, I'm happy to plough through code and try to fix if you can point me in the right direction as to which src file constructs the SIP Notify message ? WORKING VERSION 1.0.6 No. Time Source Destination Protocol Length Info 1169 19:32:47 192.168.2.41 192.168.2.30 SIP 996 Request: NOTIFY sip:1001 at 192.168.2.30;transport=udp Frame 1169: 996 bytes on wire (7968 bits), 996 bytes captured (7968 bits) Ethernet II, Src: Vmware_1b:7d:13 (00:0c:29:1b:7d:13), Dst: Tenovis_b6:ec:4f (00:07:3b:b6:ec:4f) Internet Protocol Version 4, Src: 192.168.2.41 (192.168.2.41), Dst: 192.168.2.30 (192.168.2.30) User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) Session Initiation Protocol Request-Line: NOTIFY sip:1001 at 192.168.2.30;transport=udp SIP/2.0 Message Header Via: SIP/2.0/UDP 192.168.2.41;rport;branch=z9hG4bKQ656yDjS3mavS Transport: UDP Sent-by Address: 192.168.2.41 RPort: rport Branch: z9hG4bKQ656yDjS3mavS Max-Forwards: 70 From: ;tag=eQH9S8pjtvp6N To: ;tag=286229214fa041cb4fa80fb2_F1001192.168.2.30 Call-ID: 1_7c8e554c7b5cb4fa80abf_S at 192.168.2.30 CSeq: 27613481 NOTIFY Sequence Number: 27613481 Method: NOTIFY * Contact: * User-Agent: FreeSWITCH-mod_sofia/1.0.6-hacked-20120317T221523Z 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, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=21 Content-Type: application/simple-message-summary Content-Length: 91 Message Body Messages-Waiting: yes\r\n Message-Account: sip:1001 at 192.168.2.41\r\n Voice-Message: 1/0 (0/0)\r\n \r\n BROKEN VERSION LATEST GIT No. Time Source Destination Protocol Length Info 6824 18:33:12 192.168.2.41 192.168.2.30 SIP 1034 Request: NOTIFY sip:1001 at 192.168.2.30;transport=udp Frame 6824: 1034 bytes on wire (8272 bits), 1034 bytes captured (8272 bits) Ethernet II, Src: Vmware_1b:7d:13 (00:0c:29:1b:7d:13), Dst: Tenovis_b6:ec:4f (00:07:3b:b6:ec:4f) Internet Protocol Version 4, Src: 192.168.2.41 (192.168.2.41), Dst: 192.168.2.30 (192.168.2.30) User Datagram Protocol, Src Port: sip (5060), Dst Port: iad2 (1031) Session Initiation Protocol Request-Line: NOTIFY sip:1001 at 192.168.2.30;transport=udp SIP/2.0 Message Header Via: SIP/2.0/UDP 192.168.2.41;rport;branch=z9hG4bKyyreB17m4eS1c Transport: UDP Sent-by Address: 192.168.2.41 RPort: rport Branch: z9hG4bKyyreB17m4eS1c Max-Forwards: 70 From: ;tag=Oa5sgbB1o7sl SIP from address: sip:1001 at 192.168.2.41 SIP tag: Oa5sgbB1o7sl To: ;tag=2b269bdd4fa034ed4fa2ffe4_F1001192.168.2.30 SIP to address: sip:1001 at 192.168.2.41 SIP to address User Part: 1001 SIP to address Host Part: 192.168.2.41 SIP tag: 2b269bdd4fa034ed4fa2ffe4_F1001192.168.2.30 Call-ID: 7_2c27465e595004fa2f77c_S at 192.168.2.30 CSeq: 231978492 NOTIFY Sequence Number: 231978492 Method: NOTIFY * Contact: Contact-URI: sip:1001 at 192.168.2.41:5060;transport=udp Contactt-URI User Part: 1001 Contact-URI Host Part: 192.168.2.41 Contact-URI Host Port: 5060 Contact parameter: transport=udp>* User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-de019ab 2012-05-03 15-23-57 +0200 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: active;expires=60 Content-Type: application/simple-message-summary Content-Length: 91 Message Body Messages-Waiting: yes\r\n Message-Account: sip:1001 at 192.168.2.41\r\n Voice-Message: 2/0 (0/0)\r\n \r\n thanks as ever ! Bernie -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120505/73dbcc56/attachment-0001.html From a.afzali2003 at gmail.com Sat May 5 22:06:17 2012 From: a.afzali2003 at gmail.com (afshin afzali) Date: Sat, 5 May 2012 22:36:17 +0430 Subject: [Freeswitch-users] mod_fifo how temporary to disable call dispatch to an offline agent In-Reply-To: <2740A3FE1B6E4D288416E9E1160B0C7A@gmail.com> References: <2740A3FE1B6E4D288416E9E1160B0C7A@gmail.com> Message-ID: Hi Michael & Seven, Thank you guys for help ! -- afshin On Sat, May 5, 2012 at 7:18 PM, Seven Du wrote: > Hi Michael, > > I tested that example and it did call the agent(Bria or any phone accept > multi-lines) no matter it is busy or not, and I also looked the code looks > there's no code to prevent this from happening. > > Michael I guess you used a phone only support 1 sip line, so any new > incoming call will be rejected with USER_BUSY and fifo will try every > seconds but only it will success when you hangup the current call. We once > used a custom build sip ua with this solution, but leaves unsuccessful > calls every second in logs if agents are "busy" calling out. > > I also tried to find a way to automatically "seize" an agent, but, > technically, there's race conditions anyway. Say, agent send INVITE and it > might receive INVITE at the same time because mod_fifo has no way to know > the agent is start making a call. > > I currently use a temporary "call-back" solution: when client want to > make a call, it send a request to my ESL app, and I update the > fifo_outbound table to "seize" the agent line, if success(1 record line > updated), then I originate the agent... It is a pain to make sure the > "seized" line is cleared whenever the call is hangup or fail. It is kind of > works in our lab, but I'm not sure how reliable it is. > > I had thought to add some code to mod_fifo, so in addition to "fifo out" > add something like "fifo dial from to" to automatically resolve this. > > It seems no solution in mod_callcenter either it might be because most > callcenters are either inbound or outbound, but in-out mixed agents are > also seen for efficiency so it would be helpful to find a solution. > > Seven. > > On Saturday, May 5, 2012 at 5:46 AM, Michael Collins wrote: > > Hi Afshin, > > Sorry for the delay but I wanted to lab this up and test it before I > responded. I do not see the behavior you report. When I use the sample FIFO > stuff on the wiki I am able to have an on-hook agent make an outbound call > and not receive an inbound call. As soon as the agent completes his > outbound call he is available to receive an inbound call. > > Check this page: > > http://wiki.freeswitch.org/wiki/Mod_fifo#Simple_On-hook_Agent_Login.2FLogout_Example > > It works like a champ! > > -MC > > On Sat, Apr 28, 2012 at 10:05 PM, afshin afzali wrote: > > Dear FreeSWITCHers, > > The offline agents need to make outgoing calls while waiting for call > arrivals. To prevent an influence outgoing by incoming call, they need > to logout, making call and then login to the queue ! also doing logout & > login automatically alters the actual login time :( > I'm looking for a solution which can disable call dispatch temporary for a > specific agent. > Appreciate all comments, > -- afshin > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120505/84f33e03/attachment.html From sarahig1985 at gmail.com Sat May 5 23:54:19 2012 From: sarahig1985 at gmail.com (Sara Higfler) Date: Sat, 5 May 2012 20:54:19 +0100 Subject: [Freeswitch-users] Javascript Outbound Event Socket - Linger Command Message-ID: Hi, I'm a newbie to developing an outbound event handler for Freeswitch. I'm looking to use the Javascript ESL implemention (using node.js) and have managed to get basic scenarios working, including digit collection and checking against a MySQL database. One problem I have is the capture of call termination events in my script. Having read around a lot, I know that I'm meant to use the Linger command, but cannot find any examples of how to do this with a Javascript outbound handler. I've included the rough structure of my code below (details removed for brevity) - I would really appreciate if someone could help show me how I would implement the linger command to ensure I capture all call termination events. Kind regards. (function() { var server, esl; esl = require('esl'); util = require('util'); server = esl.createCallServer(); server.on('CONNECT', function(req, res) { var uri, channel_data, unique_id; channel_data = req.body; unique_id = channel_data['Unique-ID']; req.execute('answer'); req.execute('playback', 'hello.wav'); req.on('DTMF', function(req) { var digit; var channel_data; channel_data = req.body; digit = channel_data['DTMF-Digit']; console.log('DTMF Received=' + digit); return util.log('DTMF Received'); }); req.on('CHANNEL_ANSWER', function(req) { return util.log('Call was answered'); }); req.on('CHANNEL_HANGUP', function(req) { console.log('CHANNEL_HANGUP'); return util.log('CHANNEL_HANGUP'); }); req.on('CHANNEL_HANGUP_COMPLETE', function(req) { console.log('CHANNEL_HANGUP_COMPLETE'); return util.log('CHANNEL_HANGUP_COMPLETE'); }); req.on('DISCONNECT', function(req) { console.log('DISCONNECT'); return util.log('DISCONNECT'); }) }); return util.log('CONNECT received'); }); server.listen(9123); }).call(this); -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120505/a239b483/attachment.html From acrow at integrafin.co.uk Sun May 6 01:08:15 2012 From: acrow at integrafin.co.uk (Alex Crow) Date: Sat, 05 May 2012 22:08:15 +0100 Subject: [Freeswitch-users] Spandsp mulated modems inbound/no /dev/FS* device? In-Reply-To: <4FA3EE97.1090908@integrafin.co.uk> References: <4FA24100.7040908@integrafin.co.uk> <20120503151946.GA824@eagle.cupis.co.uk> <4FA3EE97.1090908@integrafin.co.uk> Message-ID: <4FA596BF.4090703@integrafin.co.uk> Hi, 1. Is it possible to route inbound fax calls to the emuiated modems set up in autoload.configs/spandsp.xml? 2. I set total-modems=1 but I don't have any /dev/FS* device. FS is not running as root - is that why? Do I need a udev rule to allow the device node to be created by FreeSWITCH? Cheers Alex From lists at telefaks.de Sun May 6 01:10:12 2012 From: lists at telefaks.de (Peter Steinbach) Date: Sat, 05 May 2012 23:10:12 +0200 Subject: [Freeswitch-users] Setting channel variables in FIFO recording template Message-ID: <4FA59734.2010307@telefaks.de> There was already a thread for this in 2010 ("Setting channel variables for FIFO-originated calls" http://freeswitch-users.2379917.n2.nabble.com/Setting-channel-variables-for-FIFO-originated-calls-td4874089.html) but my requirements are a bit different so I am asking this again: What am I going to do? I want to record a file in a fifo (agent is not a consumer) with the agent's id and caller_id_number and date. * if I set fifo_record_template in the dialplan, I do not have the agent's id available, because this (external) id shall not be the phone number and at the time of evaluation of the dialplan we cannot predict the agent's numer and it's id and the call date * If we enter the caller_id-number in the fifo config we can set the agent's number and id but we cannot set the caller_id_number and the date as Freeswith will not accept it 2012-05-05 23:03:53.517552 [CRIT] switch_channel.c:1183 Invalid data (${fifo_record_template} contains a variable) My current template is as follows: fifo_record_template=/usr/local/freeswitch/recordings/PROJ1/20120505/${caller_caller_id_number}__12345__202__20120505_224740__Sales.wav How can I overcome this problem? -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120505/3a568ab2/attachment.html From djbinter at gmail.com Sun May 6 01:19:29 2012 From: djbinter at gmail.com (DJB International) Date: Sat, 5 May 2012 14:19:29 -0700 Subject: [Freeswitch-users] SIP NOTIFY for MWI with AVAYA 9630 In-Reply-To: References: Message-ID: What do you have it set for contact-params in your sip profile? -djbinter On Sat, May 5, 2012 at 9:15 AM, Bernard Murphy < bernard.david.murphy at gmail.com> wrote: > On anything newer than 1.06 of freeswitch the SIP NOTIFY sent to an Avaya > 9630 phone does not get accepted and trigger the MWI light. Works > absolutely fine on 1.0.6. > > The only difference in the messages i can see is in the contact parameter, > possibly to do with the angle bracket. Can anyone advise if the difference > shown in the two messages below is acceptable SIP or if something is broken > in freeswitch. Also, I'm happy to plough through code and try to fix if you > can point me in the right direction as to which src file constructs the SIP > Notify message ? > > WORKING VERSION 1.0.6 > No. Time Source Destination Protocol > Length Info > 1169 19:32:47 192.168.2.41 192.168.2.30 SIP > 996 Request: NOTIFY sip:1001 at 192.168.2.30;transport=udp > > Frame 1169: 996 bytes on wire (7968 bits), 996 bytes captured (7968 bits) > Ethernet II, Src: Vmware_1b:7d:13 (00:0c:29:1b:7d:13), Dst: > Tenovis_b6:ec:4f (00:07:3b:b6:ec:4f) > Internet Protocol Version 4, Src: 192.168.2.41 (192.168.2.41), Dst: > 192.168.2.30 (192.168.2.30) > User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) > Session Initiation Protocol > Request-Line: NOTIFY sip:1001 at 192.168.2.30;transport=udp SIP/2.0 > Message Header > Via: SIP/2.0/UDP 192.168.2.41;rport;branch=z9hG4bKQ656yDjS3mavS > Transport: UDP > Sent-by Address: 192.168.2.41 > RPort: rport > Branch: z9hG4bKQ656yDjS3mavS > Max-Forwards: 70 > From: ;tag=eQH9S8pjtvp6N > To: >;tag=286229214fa041cb4fa80fb2_F1001192.168.2.30 > Call-ID: 1_7c8e554c7b5cb4fa80abf_S at 192.168.2.30 > CSeq: 27613481 NOTIFY > Sequence Number: 27613481 > Method: NOTIFY > * Contact: * > User-Agent: FreeSWITCH-mod_sofia/1.0.6-hacked-20120317T221523Z > 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, presence, dialog, line-seize, call-info, sla, > include-session-description, presence.winfo, message-summary, refer > Subscription-State: active;expires=21 > Content-Type: application/simple-message-summary > Content-Length: 91 > Message Body > Messages-Waiting: yes\r\n > Message-Account: sip:1001 at 192.168.2.41\r\n > Voice-Message: 1/0 (0/0)\r\n > \r\n > > BROKEN VERSION LATEST GIT > No. Time Source Destination Protocol > Length Info > 6824 18:33:12 192.168.2.41 192.168.2.30 SIP > 1034 Request: NOTIFY sip:1001 at 192.168.2.30;transport=udp > > Frame 6824: 1034 bytes on wire (8272 bits), 1034 bytes captured (8272 bits) > Ethernet II, Src: Vmware_1b:7d:13 (00:0c:29:1b:7d:13), Dst: > Tenovis_b6:ec:4f (00:07:3b:b6:ec:4f) > Internet Protocol Version 4, Src: 192.168.2.41 (192.168.2.41), Dst: > 192.168.2.30 (192.168.2.30) > User Datagram Protocol, Src Port: sip (5060), Dst Port: iad2 (1031) > Session Initiation Protocol > Request-Line: NOTIFY sip:1001 at 192.168.2.30;transport=udp SIP/2.0 > Message Header > Via: SIP/2.0/UDP 192.168.2.41;rport;branch=z9hG4bKyyreB17m4eS1c > Transport: UDP > Sent-by Address: 192.168.2.41 > RPort: rport > Branch: z9hG4bKyyreB17m4eS1c > Max-Forwards: 70 > From: ;tag=Oa5sgbB1o7sl > SIP from address: sip:1001 at 192.168.2.41 > SIP tag: Oa5sgbB1o7sl > To: >;tag=2b269bdd4fa034ed4fa2ffe4_F1001192.168.2.30 > SIP to address: sip:1001 at 192.168.2.41 > SIP to address User Part: 1001 > SIP to address Host Part: 192.168.2.41 > SIP tag: 2b269bdd4fa034ed4fa2ffe4_F1001192.168.2.30 > Call-ID: 7_2c27465e595004fa2f77c_S at 192.168.2.30 > CSeq: 231978492 NOTIFY > Sequence Number: 231978492 > Method: NOTIFY > * Contact: > Contact-URI: sip:1001 at 192.168.2.41:5060;transport=udp > Contactt-URI User Part: 1001 > Contact-URI Host Part: 192.168.2.41 > Contact-URI Host Port: 5060 > Contact parameter: transport=udp>* > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-de019ab 2012-05-03 > 15-23-57 +0200 > 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: active;expires=60 > Content-Type: application/simple-message-summary > Content-Length: 91 > Message Body > Messages-Waiting: yes\r\n > Message-Account: sip:1001 at 192.168.2.41\r\n > Voice-Message: 2/0 (0/0)\r\n > \r\n > > thanks as ever ! > Bernie > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120505/11e1f506/attachment-0001.html From jmesquita at freeswitch.org Sun May 6 04:24:27 2012 From: jmesquita at freeswitch.org (=?utf-8?Q?Jo=C3=A3o_Mesquita?=) Date: Sat, 5 May 2012 21:24:27 -0300 Subject: [Freeswitch-users] New proof-of-concept module (mod_pickupgroup) In-Reply-To: References: Message-ID: Tony, I have no words to express how amazed I am for how fast this came down the pipe and how easy it is to understand the code (except for the originate part since that requires a bit more than what I can handle right now). Like usual, thank you for the help. The first thing I tried doing when I saw the module is wikify it, but someone (djbinter, thanks) got to it before I had the chance. The next thing was to make it work on the default configs so I can propose a patch so that the feature can be properly demonstrated. While at it, I've found a couple of things details that I am reporting to Jira. I will try to come up with a patch for those as well, but it might be a bit over my head. Just let me know how I can help as usual and I will get it done ASAP. Once again, thanks. Regards, -- Jo?o Mesquita Sent with Sparrow (http://www.sparrowmailapp.com/?sig) On Friday, May 4, 2012 at 11:11 PM, Anthony Minessale wrote: > Jo?o, > > After talking to you about this earlier in the week and having a > second person making the same assumptions with intercept, I decided to > implement the pickup endpoint/app pair. > > Basically the pickup endpoint is a dummy channel that never answers > that you can originate to from anywhere you can place calls, call it > alone or place it in a list for a forked dial. > > originate sofia/internal/100 at test.com (mailto:100 at test.com),pickup/mygroup > > now the pickup channel will be created alongside the sip channel. > > > From another call route the call to the application pickup with data "mygroup" > > > > When you place that call, your session will be returned from the > originate in place of the pickup/mygroup call being placed. > > It also works with presence so you could assign blf buttons to it. > pickup+mygroup > > The group names can also have an @domain.com (http://domain.com) for multihoming etc. > > > > > > On Fri, May 4, 2012 at 3:27 PM, Vik Killa wrote: > > +1 > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com (mailto:anthony_minessale at hotmail.com) > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com (mailto:anthony.minessale at gmail.com) > IRC: irc.freenode.net (http://irc.freenode.net) #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org (mailto:888 at conference.freeswitch.org) > googletalk:conf+888 at conference.freeswitch.org (mailto:conf+888 at conference.freeswitch.org) > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120505/93e408ed/attachment.html From krice at freeswitch.org Sun May 6 04:39:05 2012 From: krice at freeswitch.org (Ken Rice) Date: Sat, 05 May 2012 19:39:05 -0500 Subject: [Freeswitch-users] New proof-of-concept module (mod_pickupgroup) In-Reply-To: Message-ID: YAY! Thanks Tony! Jo?o, can you handle the WikiTax on this one? Thanks Ken On 5/5/12 7:24 PM, "Jo?o Mesquita" wrote: > > Tony, > > I have no words to express how amazed I am for how fast this came down the > pipe and how easy it is to understand the code (except for the originate part > since that requires a bit more than what I can handle right now). Like usual, > thank you for the help. > > The first thing I tried doing when I saw the module is wikify it, but someone > (djbinter, thanks) got to it before I had the chance. > > The next thing was to make it work on the default configs so I can propose a > patch so that the feature can be properly demonstrated. While at it, I've > found a couple of things details that I am reporting to Jira. I will try to > come up with a patch for those as well, but it might be a bit over my head. > Just let me know how I can help as usual and I will get it done ASAP. > > Once again, thanks. > > Regards, > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120505/1791efb5/attachment.html From bdfoster at endigotech.com Sun May 6 06:13:53 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Sat, 5 May 2012 22:13:53 -0400 Subject: [Freeswitch-users] Spandsp mulated modems inbound/no /dev/FS* device? In-Reply-To: <4FA596BF.4090703@integrafin.co.uk> References: <4FA24100.7040908@integrafin.co.uk> <20120503151946.GA824@eagle.cupis.co.uk> <4FA3EE97.1090908@integrafin.co.uk> <4FA596BF.4090703@integrafin.co.uk> Message-ID: Quick question, why would you need that in the first place? Spandsp works fine sending/receiving faxes without that. The only reason would be something like hylafax integration. If that's what your looking for, there isn't any docs I can find on that. Maybe someone who knows how to can jump in and wikify that. -BDF On May 5, 2012 5:10 PM, "Alex Crow" wrote: > Hi, > > 1. Is it possible to route inbound fax calls to the emuiated modems set > up in autoload.configs/spandsp.xml? > > 2. I set total-modems=1 but I don't have any /dev/FS* device. FS is not > running as root - is that why? Do I need a udev rule to allow the device > node to be created by FreeSWITCH? > > Cheers > > Alex > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120505/c0affeff/attachment.html From bernard.david.murphy at gmail.com Sun May 6 12:18:23 2012 From: bernard.david.murphy at gmail.com (Bernard Murphy) Date: Sun, 6 May 2012 09:18:23 +0100 Subject: [Freeswitch-users] SIP NOTIFY for MWI with AVAYA 9630 In-Reply-To: References: Message-ID: thanks for the suggestion, but contact-params is not set at all On Sat, May 5, 2012 at 10:19 PM, DJB International wrote: > What do you have it set for contact-params in your sip profile? > > -djbinter > > On Sat, May 5, 2012 at 9:15 AM, Bernard Murphy < > bernard.david.murphy at gmail.com> wrote: > >> On anything newer than 1.06 of freeswitch the SIP NOTIFY sent to an Avaya >> 9630 phone does not get accepted and trigger the MWI light. Works >> absolutely fine on 1.0.6. >> >> The only difference in the messages i can see is in the contact >> parameter, possibly to do with the angle bracket. Can anyone advise if the >> difference shown in the two messages below is acceptable SIP or if >> something is broken in freeswitch. Also, I'm happy to plough through code >> and try to fix if you can point me in the right direction as to which src >> file constructs the SIP Notify message ? >> >> WORKING VERSION 1.0.6 >> No. Time Source Destination Protocol >> Length Info >> 1169 19:32:47 192.168.2.41 192.168.2.30 SIP >> 996 Request: NOTIFY sip:1001 at 192.168.2.30;transport=udp >> >> Frame 1169: 996 bytes on wire (7968 bits), 996 bytes captured (7968 bits) >> Ethernet II, Src: Vmware_1b:7d:13 (00:0c:29:1b:7d:13), Dst: >> Tenovis_b6:ec:4f (00:07:3b:b6:ec:4f) >> Internet Protocol Version 4, Src: 192.168.2.41 (192.168.2.41), Dst: >> 192.168.2.30 (192.168.2.30) >> User Datagram Protocol, Src Port: sip (5060), Dst Port: sip (5060) >> Session Initiation Protocol >> Request-Line: NOTIFY sip:1001 at 192.168.2.30;transport=udp SIP/2.0 >> Message Header >> Via: SIP/2.0/UDP 192.168.2.41;rport;branch=z9hG4bKQ656yDjS3mavS >> Transport: UDP >> Sent-by Address: 192.168.2.41 >> RPort: rport >> Branch: z9hG4bKQ656yDjS3mavS >> Max-Forwards: 70 >> From: ;tag=eQH9S8pjtvp6N >> To: > >;tag=286229214fa041cb4fa80fb2_F1001192.168.2.30 >> Call-ID: 1_7c8e554c7b5cb4fa80abf_S at 192.168.2.30 >> CSeq: 27613481 NOTIFY >> Sequence Number: 27613481 >> Method: NOTIFY >> * Contact: * >> User-Agent: FreeSWITCH-mod_sofia/1.0.6-hacked-20120317T221523Z >> 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, presence, dialog, line-seize, call-info, sla, >> include-session-description, presence.winfo, message-summary, refer >> Subscription-State: active;expires=21 >> Content-Type: application/simple-message-summary >> Content-Length: 91 >> Message Body >> Messages-Waiting: yes\r\n >> Message-Account: sip:1001 at 192.168.2.41\r\n >> Voice-Message: 1/0 (0/0)\r\n >> \r\n >> >> BROKEN VERSION LATEST GIT >> No. Time Source Destination Protocol >> Length Info >> 6824 18:33:12 192.168.2.41 192.168.2.30 SIP >> 1034 Request: NOTIFY sip:1001 at 192.168.2.30;transport=udp >> >> Frame 6824: 1034 bytes on wire (8272 bits), 1034 bytes captured (8272 >> bits) >> Ethernet II, Src: Vmware_1b:7d:13 (00:0c:29:1b:7d:13), Dst: >> Tenovis_b6:ec:4f (00:07:3b:b6:ec:4f) >> Internet Protocol Version 4, Src: 192.168.2.41 (192.168.2.41), Dst: >> 192.168.2.30 (192.168.2.30) >> User Datagram Protocol, Src Port: sip (5060), Dst Port: iad2 (1031) >> Session Initiation Protocol >> Request-Line: NOTIFY sip:1001 at 192.168.2.30;transport=udp SIP/2.0 >> Message Header >> Via: SIP/2.0/UDP 192.168.2.41;rport;branch=z9hG4bKyyreB17m4eS1c >> Transport: UDP >> Sent-by Address: 192.168.2.41 >> RPort: rport >> Branch: z9hG4bKyyreB17m4eS1c >> Max-Forwards: 70 >> From: ;tag=Oa5sgbB1o7sl >> SIP from address: sip:1001 at 192.168.2.41 >> SIP tag: Oa5sgbB1o7sl >> To: > >;tag=2b269bdd4fa034ed4fa2ffe4_F1001192.168.2.30 >> SIP to address: sip:1001 at 192.168.2.41 >> SIP to address User Part: 1001 >> SIP to address Host Part: 192.168.2.41 >> SIP tag: 2b269bdd4fa034ed4fa2ffe4_F1001192.168.2.30 >> Call-ID: 7_2c27465e595004fa2f77c_S at 192.168.2.30 >> CSeq: 231978492 NOTIFY >> Sequence Number: 231978492 >> Method: NOTIFY >> * Contact: >> Contact-URI: sip:1001 at 192.168.2.41:5060;transport=udp >> Contactt-URI User Part: 1001 >> Contact-URI Host Part: 192.168.2.41 >> Contact-URI Host Port: 5060 >> Contact parameter: transport=udp>* >> User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-de019ab 2012-05-03 >> 15-23-57 +0200 >> 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: active;expires=60 >> Content-Type: application/simple-message-summary >> Content-Length: 91 >> Message Body >> Messages-Waiting: yes\r\n >> Message-Account: sip:1001 at 192.168.2.41\r\n >> Voice-Message: 2/0 (0/0)\r\n >> \r\n >> >> thanks as ever ! >> Bernie >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120506/56821696/attachment-0001.html From acrow at integrafin.co.uk Sun May 6 12:26:22 2012 From: acrow at integrafin.co.uk (Alex Crow) Date: Sun, 06 May 2012 09:26:22 +0100 Subject: [Freeswitch-users] Spandsp mulated modems inbound/no /dev/FS* device? In-Reply-To: References: <4FA24100.7040908@integrafin.co.uk> <20120503151946.GA824@eagle.cupis.co.uk> <4FA3EE97.1090908@integrafin.co.uk> <4FA596BF.4090703@integrafin.co.uk> Message-ID: <4FA635AE.5050006@integrafin.co.uk> On 06/05/12 03:13, Brian Foster wrote: > > Quick question, why would you need that in the first place? Spandsp > works fine sending/receiving faxes without that. > > The only reason would be something like hylafax integration. If that's > what your looking for, there isn't any docs I can find on that. Maybe > someone who knows how to can jump in and wikify that. > > -BDF > Brian, Yes, it's for hylafax integration. I really like it, so do the staff at my organisation, and I don't see a need to change something that "just works". I was hoping that I might see a registration corresponding to the modem, but for some reason I don't think it is working as it should anyway (eg no /dev/FS* device node). Cheers Alex From devel at omninet.eu Sun May 6 15:13:25 2012 From: devel at omninet.eu (Anestis Mavro) Date: Sun, 6 May 2012 14:13:25 +0300 Subject: [Freeswitch-users] sip response "101 dialing" - problems Message-ID: <80AD7F99C2E24533ACD412C2A2A2A68A@omni1.local> After updating to git (adfe462) this Friday I noticed problems with outgoing calls from Linksys PAP2T and old 1001 devices. It looks like these devices can't handle the new SIP response "101 dialing" which is new (I don't know when it was introduced). As soon as you dial you get a "busy tone" but the call continues and after a while it rings the other end. Does anybody know how to disable this SIP response? (It comes after "100 trying") Thank you -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120506/dfba64ba/attachment.html From steveu at coppice.org Sun May 6 16:02:42 2012 From: steveu at coppice.org (Steve Underwood) Date: Sun, 06 May 2012 20:02:42 +0800 Subject: [Freeswitch-users] Spandsp mulated modems inbound/no /dev/FS* device? In-Reply-To: <4FA635AE.5050006@integrafin.co.uk> References: <4FA24100.7040908@integrafin.co.uk> <20120503151946.GA824@eagle.cupis.co.uk> <4FA3EE97.1090908@integrafin.co.uk> <4FA596BF.4090703@integrafin.co.uk> <4FA635AE.5050006@integrafin.co.uk> Message-ID: <4FA66862.1090300@coppice.org> Hi Alex, Its the user ID which is causing your problem. If you just add the following to your configuration you will get five /dev/FS* devices if you run as root, but none if you run as freeswitch. This ought to be fixed. I think the device names are a bit questionable, too. Calling the /dev/FS/modem* or even /dev/FS/t31-* might increase flexibility for the future where more device types might be needed. The use of the T.31 modems needs adding to the wiki, with sections on Linux use and on Windows use. Windows use is far from obvious, as you need to install com0com to provide the loopback virtual serial port devices. Steve On 05/06/2012 04:26 PM, Alex Crow wrote: > On 06/05/12 03:13, Brian Foster wrote: >> Quick question, why would you need that in the first place? Spandsp >> works fine sending/receiving faxes without that. >> >> The only reason would be something like hylafax integration. If that's >> what your looking for, there isn't any docs I can find on that. Maybe >> someone who knows how to can jump in and wikify that. >> >> -BDF >> > Brian, > > Yes, it's for hylafax integration. I really like it, so do the staff at > my organisation, and I don't see a need to change something that "just > works". > > I was hoping that I might see a registration corresponding to the modem, > but for some reason I don't think it is working as it should anyway (eg > no /dev/FS* device node). > > Cheers > > Alex From freeswitch-list at puzzled.xs4all.nl Sun May 6 18:13:51 2012 From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists) Date: Sun, 06 May 2012 16:13:51 +0200 Subject: [Freeswitch-users] Spandsp mulated modems inbound/no /dev/FS* device? In-Reply-To: <4FA66862.1090300@coppice.org> References: <4FA24100.7040908@integrafin.co.uk> <20120503151946.GA824@eagle.cupis.co.uk> <4FA3EE97.1090908@integrafin.co.uk> <4FA596BF.4090703@integrafin.co.uk> <4FA635AE.5050006@integrafin.co.uk> <4FA66862.1090300@coppice.org> Message-ID: <4FA6871F.5050909@puzzled.xs4all.nl> On 06-05-12 14:02, Steve Underwood wrote: > Hi Alex, > > Its the user ID which is causing your problem. If you just add the following > > > > to your configuration you will get five /dev/FS* devices if you run as > root, but none if you run as freeswitch. This ought to be fixed. I think Could a udev rule with user=... (and group=...) perhaps fix that? > the device names are a bit questionable, too. Calling the /dev/FS/modem* > or even /dev/FS/t31-* might increase flexibility for the future where > more device types might be needed. +1 Regards, Patrick From brandon.mcginty at gmail.com Sun May 6 18:46:22 2012 From: brandon.mcginty at gmail.com (brandon.mcginty at gmail.com) Date: Sun, 6 May 2012 10:46:22 -0400 Subject: [Freeswitch-users] Recording all calls with good quality. Message-ID: <20120506144621.GA6722@bmcginty.hopto.org> Good day. I'm trying to record all calls sent through my Freeswitch PBX. The problem is that the quality is horrible. I've tried wav and GSM formats, and both are les than satisfactory. I'm looking for something around the quality of the recent recordings of the Freeswitch conference. (I checked on the wiki, and couldn't find anything through search.) Any information on the FS conference format or my configuration would be appreciated. I've got: Sincerely, Brandon McGinty-Carroll From darcyp at voice2net.ca Sun May 6 19:04:43 2012 From: darcyp at voice2net.ca (Darcy Primrose) Date: Sun, 6 May 2012 11:04:43 -0400 Subject: [Freeswitch-users] execute_on_fax_success/failure Message-ID: I am using excute_on_fax_success to call a script when a fax is received, the script converts the fax to pdf and emails it, Is there a way to pass the number of pages received, ie, a variable created by spandsp. Everything works, but I do not seem to be able to pass the variables. Thanks Darcy Primrose -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120506/16853457/attachment.html From steveu at coppice.org Sun May 6 19:20:17 2012 From: steveu at coppice.org (Steve Underwood) Date: Sun, 06 May 2012 23:20:17 +0800 Subject: [Freeswitch-users] Spandsp mulated modems inbound/no /dev/FS* device? In-Reply-To: <4FA6871F.5050909@puzzled.xs4all.nl> References: <4FA24100.7040908@integrafin.co.uk> <20120503151946.GA824@eagle.cupis.co.uk> <4FA3EE97.1090908@integrafin.co.uk> <4FA596BF.4090703@integrafin.co.uk> <4FA635AE.5050006@integrafin.co.uk> <4FA66862.1090300@coppice.org> <4FA6871F.5050909@puzzled.xs4all.nl> Message-ID: <4FA696B1.8050706@coppice.org> On 05/06/2012 10:13 PM, Patrick Lists wrote: > On 06-05-12 14:02, Steve Underwood wrote: >> Hi Alex, >> >> Its the user ID which is causing your problem. If you just add the following >> >> >> >> to your configuration you will get five /dev/FS* devices if you run as >> root, but none if you run as freeswitch. This ought to be fixed. I think > Could a udev rule with user=... (and group=...) perhaps fix that? > >> the device names are a bit questionable, too. Calling the /dev/FS/modem* >> or even /dev/FS/t31-* might increase flexibility for the future where >> more device types might be needed. > +1 Its not the actual pts devices which cause problems. Those are happily created with any user name that's tried. It is the /dev/FS* links which won't create unless you are root. I'm not clear how udev can be used to make those creatable by someone like the freeswitch user. If you create a directory /dev/FS, make that writable by all, change FS to create the link as /dev/FS/*, and run as user freeswitch things work OK. However, /dev/FS is lost on reboot, because of the dynamic nature of /dev. Steve From efs.tdw at recursor.net Sun May 6 17:37:25 2012 From: efs.tdw at recursor.net (efs.tdw at recursor.net) Date: Sun, 06 May 2012 23:37:25 +1000 Subject: [Freeswitch-users] FreeTDM and FXO Call Disconnect Supervision in Australia Message-ID: <4FA67E95.5040509@internode.on.net> I have searched the archives (and indeed the Internet at large) and everything I found pointed me to the perfect solution to all my problems: get ROIC (Reverse On Idle Condition) enabled on my phone line. Sadly further investigation revealed that Telstra (who provides most PSTN services in Australia, including mine) has decided to discontinue offering ROIC as of 2008: > Line reversals were withdrawn from new sales June/July this year [2008] and will be removed from existing services June/July next > year [2009]. (http://forums.whirlpool.net.au/archive/1076769). Let me describe the kinds of problem I am observing: 1. Remote party calls, waits for 5 seconds, then hangs up before I have answered. My phones go on ringing for another 25 seconds until voicemail kicks in and answers the call at the 30 second mark. I get a voicemail message containing nothing but a dial tone (there is a problem with using tone_detect to avoid this, but that is a topic for another post). 2. First remote party calls, waits for 15 seconds, then hangs up. Second remote party calls 10 seconds later. 5 seconds after the second call starts, voicemail kicks in and answers, believing that there has only been one caller and that the phone has been ringing for 30 seconds. Now I accept that to some extent these kinds of problems are unavoidable without the assistance of something like ROIC. There is no immediate way to tell the difference between a gap between two RINGs and the remote caller having hung up. The difference only becomes obvious at the time when the next RING ought to have started (I suppose a RING that was cut short could also be detected, but since the gaps are longer than the RINGs it is more likely that the hangup will occur during a gap). Likewise, if a second caller comes along just perfectly starting at the instant that the next RING was due for the first caller then I can't see how FreeTDM can avoid being tricked. However, at present, FreeTDM totally ignores FTDM_OOB_RING_STOP events, and only seems to take notice of the first FTDM_OOB_RING_START event (well it counts rings, but doesn't seem to notice that the count has stopped increasing, at least not until more than 25 seconds after the ringing has stopped). This creates an unnecessarily long window of opportunity for problems. Given the Australian ring cadence (0.4s RING, 0.2s gap, 0.4s RING, 2s gap) it is possible to detect the caller having hung up within at most 2 seconds - simply reset a timer on each FTDM_OOB_RING_START and FTDM_OOB_RING_STOP event, and reset the line state if the timer gets over 2 seconds. I have implemented this (rather awkwardly), and it works well for me, but my timeout is hard-coded and therefore it would not work with a US cadence (2s RING, 4s gap). What I would ideally like to see is for FreeTDM to support RING cadence specification (both for detection on the FXO lines and for distinctive ring on the FXS lines) but that would be a big job and way out of my league. Failing that, there would need to be some config setting to control how long to wait for a RING_START/RING_STOP event before concluding the caller had hung up. Which config file would be the most logical place? freetdm.conf just in case different channels need different timeouts? Or is there some better approach to the whole problem? [I am a little surprised there has not been more complaint about these issues. If nobody else suffers from the same problems then perhaps I just have something misconfigured??] -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120506/238a890a/attachment.html From avi at avimarcus.net Sun May 6 19:50:44 2012 From: avi at avimarcus.net (Avi Marcus) Date: Sun, 6 May 2012 18:50:44 +0300 Subject: [Freeswitch-users] Recording all calls with good quality. In-Reply-To: <20120506144621.GA6722@bmcginty.hopto.org> References: <20120506144621.GA6722@bmcginty.hopto.org> Message-ID: AFAIK, gsm is lossy, which is why it doesn't sound good. wav, however, should sound great. What codec are your calls in? Are you recording many calls? Check the load average while you are recording. If it's greater than the number of CPUs then probably your hard drives can't keep up. There are several options then, first let see if we can diagnose the issue. -Avi On Sun, May 6, 2012 at 5:46 PM, wrote: > Good day. > I'm trying to record all calls sent through my Freeswitch PBX. > The problem is that the quality is horrible. I've tried wav and GSM > formats, and both are les than satisfactory. > I'm looking for something around the quality of the recent recordings of > the Freeswitch conference. (I checked on the wiki, and couldn't find > anything through search.) > Any information on the FS conference format or my configuration would be > appreciated. > I've got: > > > > > > data="$${base_dir}/recordings/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/> > > > Sincerely, > Brandon McGinty-Carroll > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120506/9f2c0507/attachment.html From bdfoster at endigotech.com Sun May 6 21:39:33 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Sun, 6 May 2012 13:39:33 -0400 Subject: [Freeswitch-users] execute_on_fax_success/failure In-Reply-To: References: Message-ID: You news to pass fax_document_total_pages to your script. Please read the spandsp page on the wiki (search spandsp). Also by looking at your script you might be able to polish up your dialplan so that its not so complicated. On May 6, 2012 11:06 AM, "Darcy Primrose" wrote: > ** > I am using excute_on_fax_success to call a script when a fax is received, > the script converts the fax to pdf and emails it, Is there a way to pass > the number of pages received, ie, a variable created by spandsp. > Everything works, but I do not seem to be able to pass the variables. > > expression="^1(\d{10})$|(\d{10})$"> > data="faxtempdirectory=/faxfiles/${caller_id_number}"/> > > > > data="faxFile=$1$2-${caller_id_number}-${strftime(%Y-%m-%d-%H-%M-%S)-Success}"/> > > > > > > > > > Thanks > Darcy Primrose > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120506/919e8409/attachment.html From djbinter at gmail.com Sun May 6 23:48:07 2012 From: djbinter at gmail.com (DJB International) Date: Sun, 6 May 2012 12:48:07 -0700 Subject: [Freeswitch-users] sip response "101 dialing" - problems In-Reply-To: <80AD7F99C2E24533ACD412C2A2A2A68A@omni1.local> References: <80AD7F99C2E24533ACD412C2A2A2A68A@omni1.local> Message-ID: It was added on May3 in git-8664dc6: If it broke your device and you want to get your device going, you can roll up to the prior git revision for the time being, and please open a bug in Jira. -djbinter On Sun, May 6, 2012 at 4:13 AM, Anestis Mavro wrote: > ** ** > > ** ** > > After updating to git (adfe462) this Friday I noticed problems with > outgoing calls from Linksys PAP2T and old 1001 devices.**** > > It looks like these devices can?t handle the new SIP response ?101 > dialing? which is new (I don?t know when it was introduced).**** > > As soon as you dial you get a ?busy tone? but the call continues and after > a while it rings the other end.**** > > ** ** > > Does anybody know how to disable this SIP response? (It comes after ?100 > trying?)**** > > ** ** > > Thank you**** > > ** ** > > ** ** > > > __________ Information from ESET NOD32 Antivirus, version of virus > signature database 5054 (20100423) __________ > > The message was checked by ESET NOD32 Antivirus. > > http://www.eset.com > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120506/06267cf1/attachment-0001.html From darcyp at voice2net.ca Mon May 7 03:01:42 2012 From: darcyp at voice2net.ca (Darcy Primrose) Date: Sun, 6 May 2012 19:01:42 -0400 Subject: [Freeswitch-users] execute_on_fax_success/failure References: Message-ID: <99B4785CBF5B4B5F9082207335342A9E@owner397fa27d2> I did do that, however, if lists fax_document_total_pages as NA, and when I try to pass that parameter, I get a blank field. The dialplan below is not our production, it is merely a test platfrom we use to allow us to ensure your data flow is correct. Darcy ----- Original Message ----- From: Brian Foster To: FreeSWITCH Users Help Sent: Sunday, May 06, 2012 1:39 PM Subject: Re: [Freeswitch-users] execute_on_fax_success/failure You news to pass fax_document_total_pages to your script. Please read the spandsp page on the wiki (search spandsp). Also by looking at your script you might be able to polish up your dialplan so that its not so complicated. On May 6, 2012 11:06 AM, "Darcy Primrose" wrote: I am using excute_on_fax_success to call a script when a fax is received, the script converts the fax to pdf and emails it, Is there a way to pass the number of pages received, ie, a variable created by spandsp. Everything works, but I do not seem to be able to pass the variables. Thanks Darcy Primrose _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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 at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120506/fa0cd258/attachment.html From bdfoster at endigotech.com Mon May 7 03:37:13 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Sun, 6 May 2012 19:37:13 -0400 Subject: [Freeswitch-users] execute_on_fax_success/failure In-Reply-To: <99B4785CBF5B4B5F9082207335342A9E@owner397fa27d2> References: <99B4785CBF5B4B5F9082207335342A9E@owner397fa27d2> Message-ID: Make sure you're escaping the variable with ///. This variable works in my setup. Admittedly I'm using api_hangup_hook and letting my script decipher whether or not the fax was successful or not. You can also use fax_document_transferred_pages. The difference between that and ...total_pages is that transferred pages shows how many pages made it and total_pages shows how many should have made it. -BDF On May 6, 2012 7:02 PM, "Darcy Primrose" wrote: > ** > I did do that, however, if lists fax_document_total_pages as NA, and when > I try to pass that parameter, I get a blank field. The dialplan below is > not our production, it is merely a test platfrom we use to allow us to > ensure your data flow is correct. > > Darcy > > ----- Original Message ----- > *From:* Brian Foster > *To:* FreeSWITCH Users Help > *Sent:* Sunday, May 06, 2012 1:39 PM > *Subject:* Re: [Freeswitch-users] execute_on_fax_success/failure > > You news to pass fax_document_total_pages to your script. Please read the > spandsp page on the wiki (search spandsp). > > Also by looking at your script you might be able to polish up your > dialplan so that its not so complicated. > On May 6, 2012 11:06 AM, "Darcy Primrose" wrote: > >> ** >> I am using excute_on_fax_success to call a script when a fax is received, >> the script converts the fax to pdf and emails it, Is there a way to pass >> the number of pages received, ie, a variable created by spandsp. >> Everything works, but I do not seem to be able to pass the variables. >> >> > expression="^1(\d{10})$|(\d{10})$"> >> > data="faxtempdirectory=/faxfiles/${caller_id_number}"/> >> >> >> >> > data="faxFile=$1$2-${caller_id_number}-${strftime(%Y-%m-%d-%H-%M-%S)-Success}"/> >> >> >> >> >> >> >> >> >> Thanks >> Darcy Primrose >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120506/c14b7f84/attachment-0001.html From bdfoster at endigotech.com Mon May 7 03:40:49 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Sun, 6 May 2012 19:40:49 -0400 Subject: [Freeswitch-users] execute_on_fax_success/failure In-Reply-To: References: <99B4785CBF5B4B5F9082207335342A9E@owner397fa27d2> Message-ID: Sorry you should be escaping with \\\ not ///. All vars from spandsp need to be escaped with this. The others you shouldn't have to do that. -BDF On May 6, 2012 7:37 PM, "Brian Foster" wrote: > Make sure you're escaping the variable with ///. This variable works in my > setup. Admittedly I'm using api_hangup_hook and letting my script decipher > whether or not the fax was successful or not. You can also use > fax_document_transferred_pages. The difference between that and > ...total_pages is that transferred pages shows how many pages made it and > total_pages shows how many should have made it. > > -BDF > On May 6, 2012 7:02 PM, "Darcy Primrose" wrote: > >> ** >> I did do that, however, if lists fax_document_total_pages as NA, and when >> I try to pass that parameter, I get a blank field. The dialplan below is >> not our production, it is merely a test platfrom we use to allow us to >> ensure your data flow is correct. >> >> Darcy >> >> ----- Original Message ----- >> *From:* Brian Foster >> *To:* FreeSWITCH Users Help >> *Sent:* Sunday, May 06, 2012 1:39 PM >> *Subject:* Re: [Freeswitch-users] execute_on_fax_success/failure >> >> You news to pass fax_document_total_pages to your script. Please read the >> spandsp page on the wiki (search spandsp). >> >> Also by looking at your script you might be able to polish up your >> dialplan so that its not so complicated. >> On May 6, 2012 11:06 AM, "Darcy Primrose" wrote: >> >>> ** >>> I am using excute_on_fax_success to call a script when a fax is >>> received, the script converts the fax to pdf and emails it, Is there a way >>> to pass the number of pages received, ie, a variable created by spandsp. >>> Everything works, but I do not seem to be able to pass the variables. >>> >>> >> expression="^1(\d{10})$|(\d{10})$"> >>> >> data="faxtempdirectory=/faxfiles/${caller_id_number}"/> >>> >>> >>> >>> >> data="faxFile=$1$2-${caller_id_number}-${strftime(%Y-%m-%d-%H-%M-%S)-Success}"/> >>> >>> >>> >>> >>> >>> >>> >>> >>> Thanks >>> Darcy Primrose >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120506/a3c72ad6/attachment.html From darcyp at voice2net.ca Mon May 7 04:29:02 2012 From: darcyp at voice2net.ca (Darcy Primrose) Date: Sun, 6 May 2012 20:29:02 -0400 Subject: [Freeswitch-users] execute_on_fax_success/failure References: <99B4785CBF5B4B5F9082207335342A9E@owner397fa27d2> Message-ID: <68A9174972294D3DB4061464B242508A@owner397fa27d2> Brian, thank you so much for taking time to go thru this and explain it, works like a charm, so many little things the freeswitch does, I understand why the wiki documentation is so important. I do see in spandsp where it tells you to escape with \\\, it just never lit the light bulb. Thanks Again Darcy ----- Original Message ----- From: Brian Foster To: FreeSWITCH Users Help Sent: Sunday, May 06, 2012 7:40 PM Subject: Re: [Freeswitch-users] execute_on_fax_success/failure Sorry you should be escaping with \\\ not ///. All vars from spandsp need to be escaped with this. The others you shouldn't have to do that. -BDF On May 6, 2012 7:37 PM, "Brian Foster" wrote: Make sure you're escaping the variable with ///. This variable works in my setup. Admittedly I'm using api_hangup_hook and letting my script decipher whether or not the fax was successful or not. You can also use fax_document_transferred_pages. The difference between that and ...total_pages is that transferred pages shows how many pages made it and total_pages shows how many should have made it. -BDF On May 6, 2012 7:02 PM, "Darcy Primrose" wrote: I did do that, however, if lists fax_document_total_pages as NA, and when I try to pass that parameter, I get a blank field. The dialplan below is not our production, it is merely a test platfrom we use to allow us to ensure your data flow is correct. Darcy ----- Original Message ----- From: Brian Foster To: FreeSWITCH Users Help Sent: Sunday, May 06, 2012 1:39 PM Subject: Re: [Freeswitch-users] execute_on_fax_success/failure You news to pass fax_document_total_pages to your script. Please read the spandsp page on the wiki (search spandsp). Also by looking at your script you might be able to polish up your dialplan so that its not so complicated. On May 6, 2012 11:06 AM, "Darcy Primrose" wrote: I am using excute_on_fax_success to call a script when a fax is received, the script converts the fax to pdf and emails it, Is there a way to pass the number of pages received, ie, a variable created by spandsp. Everything works, but I do not seem to be able to pass the variables. Thanks Darcy Primrose _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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 at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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 at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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 at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120506/68352638/attachment-0001.html From kevin at lethe.com Mon May 7 03:01:41 2012 From: kevin at lethe.com (Kevin Olinger) Date: Sun, 6 May 2012 16:01:41 -0700 Subject: [Freeswitch-users] freetdm ring cadence Message-ID: Is it possible to specify a ring cadence in the dialplan with freetdm? I have an analog extension (TDM400) that I would like to ring with a different cadence for internal and external calls. Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120506/43a7a174/attachment.html From zhulizhong at live.com Mon May 7 04:30:44 2012 From: zhulizhong at live.com (James zhu) Date: Mon, 7 May 2012 00:30:44 +0000 Subject: [Freeswitch-users] Openvox tdma400p fail to Disconnect/Hangup detect (Dahdi/Freetdm) In-Reply-To: References: , , Message-ID: please check the the channels with alarm. what is the result of dahdi_tool? Best regards, James.zhu Doing asterisk/PRI/ss7/dahdi, linux, asterisk/sangoma cards, recording device, VOIP gateway. website: www.hiastar.com > Date: Sat, 5 May 2012 11:16:31 -0430 > From: rrodolfos at gmail.com > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Openvox tdma400p fail to Disconnect/Hangup detect (Dahdi/Freetdm) > > James, > > In freeswitch console (Cli) show this: > > 2012-05-05 10:10:33.464451 [ERR] ftmod_zt.c:1137 [s1c1][1:1] Got > polarity reverse (ZT_EVENT_POLARITY) > 2012-05-05 10:10:33.464451 [WARNING] ftdm_io.c:1786 [s1c1][1:1] Cannot > open channel when is alarmed > 2012-05-05 10:10:33.464451 [ERR] ftmod_analog.c:444 [s1c1][1:1] OPEN > ERROR [Channel is alarmed > ] > 2012-05-05 10:10:33.464451 [WARNING] ftdm_io.c:2765 [s1c1][1:1] > Channel not opened, proceeding anyway > 2012-05-05 10:10:34.404449 [NOTICE] mod_freetdm.c:1887 Alarm cleared > on channel 1:1 > . > . > . > 2012-05-05 10:13:06.704451 [ERR] ftmod_zt.c:1137 [s1c1][1:1] Got > polarity reverse (ZT_EVENT_POLARITY) > 2012-05-05 10:13:06.724453 [WARNING] ftmod_analog.c:621 [s1c1][1:1] > Not hanging up on polarity reverse, too close to Answer reverse > Details: > FreeSWITCH Version 1.1.beta1 (git-4283408 2012-04-29 11-33-24 -0400) > dahdi-linux-complete-2.6.1+2.6.1 downloaded from asterisk > > /etc/dahdi/system.conf > fxsks=1 > echocanceller=mg2,1 > loadzone = ve > defaultzone = ve > > /usr/local/freeswitch/conf/freetdm.conf > [general] > cpu_monitor => no > cpu_monitoring_interval => 1000 > cpu_set_alarm_threshold => 80 > cpu_reset_alarm_threshold => 70 > cpu_alarm_action => warn > [span zt FXO1] > fxo-channel => 1 > > /usr/local/freeswitch/conf/autoload_configs/freetdm.conf.xml > > > > > > > > > > > > > > > > > > > > > RrodolfoS > > On Fri, May 4, 2012 at 8:26 PM, James zhu wrote: > > please show the conf files. > > Best regards, > > James.zhu > > Doing asterisk/PRI/ss7/dahdi, linux, asterisk/sangoma cards, recording > > device, VOIP gateway. > > website: www.hiastar.com > > > > > >> Date: Fri, 4 May 2012 10:33:52 -0430 > >> From: rrodolfos at gmail.com > >> To: FreeSWITCH-users at lists.freeswitch.org > >> Subject: [Freeswitch-users] Openvox tdma400p fail to Disconnect/Hangup > >> detect (Dahdi/Freetdm) > > > >> > >> Hi, > >> > >> Freeswitch can recive calls, but the call is end, openvox tdma400p, > >> can't detect disconnection or hangup. > >> > >> Any idea? > >> > >> RrodolfoS > >> > >> _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at 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 at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120507/766a510f/attachment.html From msc at freeswitch.org Mon May 7 05:10:57 2012 From: msc at freeswitch.org (Michael Collins) Date: Sun, 6 May 2012 18:10:57 -0700 Subject: [Freeswitch-users] New proof-of-concept module (mod_pickupgroup) In-Reply-To: References: Message-ID: Awesome! Tony: thanks again for adding a cool new feature so quickly Jo?o: thanks for writing the proof-of-concept code that jumpstarted the process Dorn B: thanks for jumping all over the wiki stuff so quickly! It is much appreciated. -MC On Sat, May 5, 2012 at 5:24 PM, Jo?o Mesquita wrote: > Tony, > > I have no words to express how amazed I am for how fast this came down the > pipe and how easy it is to understand the code (except for the originate > part since that requires a bit more than what I can handle right now). Like > usual, thank you for the help. > > The first thing I tried doing when I saw the module is wikify it, but > someone (djbinter, thanks) got to it before I had the chance. > > The next thing was to make it work on the default configs so I can propose > a patch so that the feature can be properly demonstrated. While at it, I've > found a couple of things details that I am reporting to Jira. I will try to > come up with a patch for those as well, but it might be a bit over my head. > Just let me know how I can help as usual and I will get it done ASAP. > > Once again, thanks. > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120506/c16f322e/attachment.html From msc at freeswitch.org Mon May 7 05:14:10 2012 From: msc at freeswitch.org (Michael Collins) Date: Sun, 6 May 2012 18:14:10 -0700 Subject: [Freeswitch-users] Variable continue on fail In-Reply-To: <4FA4D2D6.1010908@vts24.ru> References: <4F87EF60.3070105@vts24.ru> <4F8822A5.9090402@vts24.ru> <4FA4D2D6.1010908@vts24.ru> Message-ID: I don't recall you ever posting a console debug log to pastebin.freeswitch.org. Could you please do that? It will help us to know exactly what is happening. -MC On Sat, May 5, 2012 at 12:12 AM, ??????? ??????? wrote: > Hi! > Is there any news on this issue? > > 13.04.2012 16:57, ??????? ??????? ???????: > > Yes, you are right! > > I did it: http://pastebin.freeswitch.org/18863 > > Additionally: > I've included in this extension new line: > > > > > ** > data="continue_on_fail=1,2,3,6,25,34,38,41,42,44,47,63,66,500,501"/> > > > > > > > Without that line a similar situation occurs if FS recieve * > NORMAL_CLEARING.* > > 13.04.2012 13:44, Anton Kvashenkin ???????: > > Ok, i got it. Even that there is no USER_BUSY at continue_on_fail > variable, FS still tries to reach the second action, am i right? > > So, for better debugging, i suggest to paste full call log with enabled > siptrace and /log 7 to pastebin.freeswitch.org. > > 13 ?????? 2012 ?. 13:18 ???????????? ??????? ??????? < > vitaliy.davudov at vts24.ru> ???????: > >> Hi all! >> >> In my dialplan I've included variable continue on fail: >> >> >> >> >> > data="continue_on_fail=1,2,3,6,25,34,38,41,42,44,47,63,66,500,501"/> >> >> >> >> >> >> >> And if FS recieve from first gateway USER_BUSY, then FS try to bridge >> this call to another gateway. Although in line > data="continue_on_fail=1,2,3,6,25,34,38,41,42,44,47,63,66,500,501"/> >> there is no code Q.850 = 17. >> How resolve this issue? >> >> -- >> Best regards, >> Vitaly Davudov >> "VIP-TELECOM-SERVICE" Ltd. >> ("ETERIA" Group of companies) >> http://www.vts24.ru >> >> >> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120506/a9ef1773/attachment-0001.html From msc at freeswitch.org Mon May 7 05:57:19 2012 From: msc at freeswitch.org (Michael Collins) Date: Sun, 6 May 2012 18:57:19 -0700 Subject: [Freeswitch-users] mod_fifo how temporary to disable call dispatch to an offline agent In-Reply-To: <2740A3FE1B6E4D288416E9E1160B0C7A@gmail.com> References: <2740A3FE1B6E4D288416E9E1160B0C7A@gmail.com> Message-ID: Seven, Can you do a "fifo list_verbose " for these conditions: * Agent not logged in * Agent logged in and waiting for a call * Agent logged in and is currently on inbound call * Agent logged in and is currently on outbound call I want to make sure that we are testing the same thing. Thanks, MC On Sat, May 5, 2012 at 7:48 AM, Seven Du wrote: > Hi Michael, > > I tested that example and it did call the agent(Bria or any phone accept > multi-lines) no matter it is busy or not, and I also looked the code looks > there's no code to prevent this from happening. > > Michael I guess you used a phone only support 1 sip line, so any new > incoming call will be rejected with USER_BUSY and fifo will try every > seconds but only it will success when you hangup the current call. We once > used a custom build sip ua with this solution, but leaves unsuccessful > calls every second in logs if agents are "busy" calling out. > > I also tried to find a way to automatically "seize" an agent, but, > technically, there's race conditions anyway. Say, agent send INVITE and it > might receive INVITE at the same time because mod_fifo has no way to know > the agent is start making a call. > > I currently use a temporary "call-back" solution: when client want to > make a call, it send a request to my ESL app, and I update the > fifo_outbound table to "seize" the agent line, if success(1 record line > updated), then I originate the agent... It is a pain to make sure the > "seized" line is cleared whenever the call is hangup or fail. It is kind of > works in our lab, but I'm not sure how reliable it is. > > I had thought to add some code to mod_fifo, so in addition to "fifo out" > add something like "fifo dial from to" to automatically resolve this. > > It seems no solution in mod_callcenter either it might be because most > callcenters are either inbound or outbound, but in-out mixed agents are > also seen for efficiency so it would be helpful to find a solution. > > Seven. > > On Saturday, May 5, 2012 at 5:46 AM, Michael Collins wrote: > > Hi Afshin, > > Sorry for the delay but I wanted to lab this up and test it before I > responded. I do not see the behavior you report. When I use the sample FIFO > stuff on the wiki I am able to have an on-hook agent make an outbound call > and not receive an inbound call. As soon as the agent completes his > outbound call he is available to receive an inbound call. > > Check this page: > > http://wiki.freeswitch.org/wiki/Mod_fifo#Simple_On-hook_Agent_Login.2FLogout_Example > > It works like a champ! > > -MC > > On Sat, Apr 28, 2012 at 10:05 PM, afshin afzali wrote: > > Dear FreeSWITCHers, > > The offline agents need to make outgoing calls while waiting for call > arrivals. To prevent an influence outgoing by incoming call, they need > to logout, making call and then login to the queue ! also doing logout & > login automatically alters the actual login time :( > I'm looking for a solution which can disable call dispatch temporary for a > specific agent. > Appreciate all comments, > -- afshin > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120506/cf7be638/attachment.html From rrodolfos at gmail.com Mon May 7 05:58:32 2012 From: rrodolfos at gmail.com (RrodolfoS .) Date: Sun, 6 May 2012 21:28:32 -0430 Subject: [Freeswitch-users] Openvox tdma400p fail to Disconnect/Hangup detect (Dahdi/Freetdm) In-Reply-To: References: Message-ID: James, lsdahdi ### Span 1: WCTDM/4 "Wildcard TDM400P REV E/F Board 5" (MASTER) 1 FXO FXSKS (EC: MG2 - INACTIVE) 2 unknown Reserved 3 unknown Reserved 4 unknown Reserved dahdi_scan [1] active=yes alarms=OK description=Wildcard TDM400P REV E/F Board 5 name=WCTDM/4 manufacturer=Digium devicetype=Wildcard TDM400P REV E/F location=PCI Bus 02 Slot 03 basechan=1 totchans=4 irq=0 type=analog port=1,FXO port=2,none port=3,none port=4,none dahdi_speed Count: 988371 dahdi_test Opened pseudo dahdi interface, measuring accuracy... 99.995% 99.989% 99.994% 99.994% 99.994% 99.994% 99.994% 99.994% 99.995% 99.988% 99.994% 99.995% 99.994% 99.994% 99.994% 99.994% 99.994% 99.992% 99.994% 99.994% 99.995% 99.994% 99.994% 99.994% ftdm alarms 1 1 +OK No alarms ftdm iostats 1 1 -- IO statistics for channel 1:1 -- Rx errors: 0 Rx queue size: 0 Rx queue len: 0 Rx count: 0 Tx errors: 0 Tx queue size: 0 Tx queue len: 0 Tx count: 0 Tx idle: 0 +OK ftdm ioread 1 1 2012-05-06 21:16:55.380451 [INFO] ftmod_zt.c:656 Setting echo cancel to 64 taps for 1:1 ftdm dump 1 1 span_id: 1 chan_id: 1 physical_span_id: 1 physical_chan_id: 1 physical_status: ok physical_status_red: 0 physical_status_yellow: 0 physical_status_rai: 0 physical_status_blue: 0 physical_status_ais: 0 physical_status_general: 0 signaling_status: UP type: FXO state: DOWN last_state: UP txgain: 0.00 rxgain: 0.00 cid_date: cid_name: cid_num: ani: aniII: dnis: rdnis: cause: NONE session: (none) -- States -- -- Function -- -- Location -- -- Time Offset -- DOWN => GET_CALLERID [process_event] [ftmod_analog.c:1026] 0ms GET_CALLERID => RING [ftdm_analog_channel_run] [ftmod_analog.c:497] 4999ms RING => PROGRESS [channel_receive_message_fxo] [mod_freetdm.c:1029] 72ms PROGRESS => PROGRESS_MEDIA [channel_receive_message_fxo] [mod_freetdm.c:1029] 10ms PROGRESS_MEDIA => UP [channel_receive_message_fxo] [mod_freetdm.c:1029] 0ms UP => DOWN [process_event] [ftmod_analog.c:1051] 64980ms Time since last state change: 277022ms ftdm list +OK span: 1 (FXO1) type: analog physical_status: ok signaling_status: UP chan_count: 1 dialplan: XML context: public dial_regex: fail_dial_regex: hold_music: analog_options: none RrodolfoS P.D: Venezuelan PSTN On Sun, May 6, 2012 at 8:00 PM, James zhu wrote: > please check the the channels with alarm. what is the result of dahdi_tool? > > > Best regards, > James.zhu > Doing asterisk/PRI/ss7/dahdi, linux, asterisk/sangoma cards, recording > device, VOIP gateway. > website: www.hiastar.com > > >> Date: Sat, 5 May 2012 11:16:31 -0430 >> From: rrodolfos at gmail.com >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] Openvox tdma400p fail to Disconnect/Hangup >> detect (Dahdi/Freetdm) > >> >> James, >> >> In freeswitch console (Cli) show this: >> >> 2012-05-05 10:10:33.464451 [ERR] ftmod_zt.c:1137 [s1c1][1:1] Got >> polarity reverse (ZT_EVENT_POLARITY) >> 2012-05-05 10:10:33.464451 [WARNING] ftdm_io.c:1786 [s1c1][1:1] Cannot >> open channel when is alarmed >> 2012-05-05 10:10:33.464451 [ERR] ftmod_analog.c:444 [s1c1][1:1] OPEN >> ERROR [Channel is alarmed >> ] >> 2012-05-05 10:10:33.464451 [WARNING] ftdm_io.c:2765 [s1c1][1:1] >> Channel not opened, proceeding anyway >> 2012-05-05 10:10:34.404449 [NOTICE] mod_freetdm.c:1887 Alarm cleared >> on channel 1:1 >> . >> . >> . >> 2012-05-05 10:13:06.704451 [ERR] ftmod_zt.c:1137 [s1c1][1:1] Got >> polarity reverse (ZT_EVENT_POLARITY) >> 2012-05-05 10:13:06.724453 [WARNING] ftmod_analog.c:621 [s1c1][1:1] >> Not hanging up on polarity reverse, too close to Answer reverse >> Details: >> FreeSWITCH Version 1.1.beta1 (git-4283408 2012-04-29 11-33-24 -0400) >> dahdi-linux-complete-2.6.1+2.6.1 downloaded from asterisk >> >> /etc/dahdi/system.conf >> fxsks=1 >> echocanceller=mg2,1 >> loadzone = ve >> defaultzone = ve >> >> /usr/local/freeswitch/conf/freetdm.conf >> [general] >> cpu_monitor => no >> cpu_monitoring_interval => 1000 >> cpu_set_alarm_threshold => 80 >> cpu_reset_alarm_threshold => 70 >> cpu_alarm_action => warn >> [span zt FXO1] >> fxo-channel => 1 >> >> /usr/local/freeswitch/conf/autoload_configs/freetdm.conf.xml >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> RrodolfoS >> >> On Fri, May 4, 2012 at 8:26 PM, James zhu wrote: >> > please show the conf files. >> > Best regards, >> > James.zhu >> > Doing asterisk/PRI/ss7/dahdi, linux, asterisk/sangoma cards, recording >> > device, VOIP gateway. >> > website: www.hiastar.com >> > >> > >> >> Date: Fri, 4 May 2012 10:33:52 -0430 >> >> From: rrodolfos at gmail.com >> >> To: FreeSWITCH-users at lists.freeswitch.org >> >> Subject: [Freeswitch-users] Openvox tdma400p fail to Disconnect/Hangup >> >> detect (Dahdi/Freetdm) >> > >> >> >> >> Hi, >> >> >> >> Freeswitch can recive calls, but the call is end, openvox tdma400p, >> >> can't detect disconnection or hangup. >> >> >> >> Any idea? >> >> >> >> RrodolfoS >> >> >> >> >> >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> >> >> >> >> >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> http://wiki.freeswitch.org >> >> http://www.cluecon.com >> >> >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at 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 at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at 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 at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From msc at freeswitch.org Mon May 7 06:14:14 2012 From: msc at freeswitch.org (Michael Collins) Date: Sun, 6 May 2012 19:14:14 -0700 Subject: [Freeswitch-users] Javascript Outbound Event Socket - Linger Command In-Reply-To: References: Message-ID: Hi Sara, Welcome to FreeSWITCH. You are quite right - there is not documentation on how to send the linger command with an ESL application. I grep'd around the libs/esl directory and found that fs_cli.c and esl.c have a function called esl_send_recv and a few occurrences of it used like this: esl_send_recv(&handle, "linger"); I'm no Javascript person but if I understand the syntax correctly you'd need to do something like this as soon as the call is connected, i.e. right before the req.execute("answer") line: req.send_recv("linger"); Could you give that a try and let us know if it works? If it does we'd appreciate you updating the wiki. :) Thanks, MC On Sat, May 5, 2012 at 12:54 PM, Sara Higfler wrote: > Hi, > > I'm a newbie to developing an outbound event handler for Freeswitch. I'm > looking to use the Javascript ESL implemention (using node.js) and have > managed to get basic scenarios working, including digit collection and > checking against a MySQL database. One problem I have is the capture of > call termination events in my script. Having read around a lot, I know > that I'm meant to use the Linger command, but cannot find any examples of > how to do this with a Javascript outbound handler. I've included the rough > structure of my code below (details removed for brevity) - I would really > appreciate if someone could help show me how I would implement the linger > command to ensure I capture all call termination events. > > Kind regards. > > (function() > { > var server, esl; > > esl = require('esl'); > util = require('util'); > > server = esl.createCallServer(); > > server.on('CONNECT', function(req, res) > { > var uri, channel_data, unique_id; > > channel_data = req.body; > unique_id = channel_data['Unique-ID']; > > req.execute('answer'); > > req.execute('playback', 'hello.wav'); > > req.on('DTMF', function(req) > { > var digit; > var channel_data; > channel_data = req.body; > > digit = channel_data['DTMF-Digit']; > console.log('DTMF Received=' + digit); > return util.log('DTMF Received'); > }); > > req.on('CHANNEL_ANSWER', function(req) > { > return util.log('Call was answered'); > }); > > req.on('CHANNEL_HANGUP', function(req) > { > console.log('CHANNEL_HANGUP'); > return util.log('CHANNEL_HANGUP'); > }); > req.on('CHANNEL_HANGUP_COMPLETE', function(req) > { > console.log('CHANNEL_HANGUP_COMPLETE'); > return util.log('CHANNEL_HANGUP_COMPLETE'); > }); > > req.on('DISCONNECT', function(req) > { > console.log('DISCONNECT'); > return util.log('DISCONNECT'); > }) > > }); > return util.log('CONNECT received'); > }); > > server.listen(9123); > }).call(this); > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120506/7e535570/attachment-0001.html From andrew.paul85 at gmail.com Mon May 7 12:54:34 2012 From: andrew.paul85 at gmail.com (Andrew Paul) Date: Mon, 7 May 2012 14:24:34 +0530 Subject: [Freeswitch-users] Max Call Limit Message-ID: Hai , How can i make freeswitch gateway(trunk) to handle maximum 'n' number of calls simutaneiously . Once if it crosses the limit i should get the response "503" response . I can make this one globally in switch.conf.xml by setting max-session . Thanks And Regards ANDREW -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120507/3c76597e/attachment.html From jmesquita at freeswitch.org Mon May 7 15:22:23 2012 From: jmesquita at freeswitch.org (=?utf-8?Q?Jo=C3=A3o_Mesquita?=) Date: Mon, 7 May 2012 08:22:23 -0300 Subject: [Freeswitch-users] Max Call Limit In-Reply-To: References: Message-ID: <93E71F0FA3A1472B80AF5A886BBD3E4E@freeswitch.org> Andrew, Look at the limit application. You will have to do this at the dial plan level. http://wiki.freeswitch.org/wiki/Limit Regards, -- Jo?o Mesquita Sent with Sparrow (http://www.sparrowmailapp.com/?sig) On Monday, May 7, 2012 at 5:54 AM, Andrew Paul wrote: > Hai , > How can i make freeswitch gateway(trunk) to handle maximum 'n' number of calls simutaneiously . Once if it crosses the limit i should get the response "503" response . I can make this one globally in switch.conf.xml by setting max-session . > > > > Thanks And Regards > > ANDREW > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120507/5849c634/attachment.html From modesto at isimples.com.br Mon May 7 15:31:05 2012 From: modesto at isimples.com.br (Antonio Modesto) Date: Mon, 07 May 2012 08:31:05 -0300 Subject: [Freeswitch-users] Does Freeswitch replace Asterisk? In-Reply-To: <308D166EF6444F968D136C820993582D@freeswitch.org> References: <1336136740.3092.31.camel@modesto.localdomain.net> <308D166EF6444F968D136C820993582D@freeswitch.org> Message-ID: <1336390265.2939.26.camel@modesto.localdomain.net> Hi, It's good to know that the list is very active! Talking about the DTMF signaling, some guys on the Asterisk list told me that they got it running here in Brazil using DTMF. When I tried to configure it on my asterisk, I had a lot of erros, sometimes the caller id worked but it had some digits in inverted positions (It sounds very strange). As my asterisk is in production, I have a limited time to try this kind of thing on it, I will do some tests with FS in a free machine here. Jo?o, I liked to know that are active br guys in this list, sure I will contact you to exchange some ideas. Thank you very much. On Fri, 2012-05-04 at 17:49 -0300, Jo?o Mesquita wrote: > Dar?o, thank you for remembering Khomp! > > > > Antonio, I might be able to chime in here. The CallerID is probably > not working in Brazil because we use a specific protocol called DTMF. > Most US hard card developers have a hard time with that because the > standard in the US is FSK (a different protocol). Khomp has DTMF > signaling support for both FXO and FXS cards (meaning your regular > analog phones with display can use "bina" as well). > > > Send me an email if you would like for more info. We support > FreeSWITCH and Asterisk, for that matter. > > > Regards (Abra?os), > > > -- > Jo?o Mesquita > Sent with Sparrow > > > > On Friday, May 4, 2012 at 1:57 PM, Michael Collins wrote: > > > > Hello Antonio, > > > > Your scenario is not uncommon. It may be that you have a > > particularly bad version of Asterisk. (A few releases were > > particularly problematic.) It could be that your hardware has > > issues. Or it could be that Asterisk may not be a good fit for your > > scenario. > > > > FreeSWITCH can do basically all of what you are doing now. The only > > thing I would do is contact Sangoma and ask them about Digium FXO > > cards in Brazil. (Sangoma wrote the FreeTDM stack for FreeSWITCH, so > > they're the experts on the subject.) > > > > If you are looking for a FreeSWITCH + GUI solution then you might > > want to check out FusionPBX or blue.box. > > > > -MC > > > > > > On Fri, May 4, 2012 at 6:05 AM, Antonio Modesto > > wrote: > > > > > Hi, > > > > > > I Work at an ISP and we have an Asterisk PBX. Our PBX > > > doesn't have > > > anything special, it just do the common things like auto > > > attendant, CDR, > > > call transfer and such things. We have 2 digium FXO cards to > > > connect to > > > the PSTN. The problem is that asterisk is not working very well, > > > sometimes some of the dahdi channels get stuck and we need to > > > release it > > > through the asterisk console, the detection of callerid doesn't > > > work (We > > > live in brazil, dtmf signaling), and other detais that if I list > > > all of > > > them here it's going to take some time. I read a lot about > > > asterisk and > > > its bad design, I think that it added a lot a features without > > > worrying > > > about making stable implementations. My question is, can > > > freeswitch > > > fully replace our Asterisk PBX? Or it's not its purpose? > > > > > > > > > Regards. > > > > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Atenciosamente, Ant?nio Modesto Gerente de TI Pra?a Get?lio Vargas, 77 ? Sala 308 ? Centro Santo Ant?nio do Monte ? MG ? CEP: 35560-000 Tel:(37) 3281-2800 Contato: isimples at isimples.com.br http://www.isimples.com.br Aviso:Esta mensagem e quaisquer arquivos em anexo podem conter informa??es confidenciais e/ou privilegiadas. Se voc? n?o for o destinat?rio ou a pessoa autorizada a receber esta mensagem, por favor, n?o leia, copie, repasse, imprima, guarde, nem tome qualquer a??o baseada nessas informa??es. Notifique o remetente imediatamente por e-mail e apague a mensagem permanentemente. Aten??o: embora a Isimples Telecom, tome seus cuidados para garantir a aus?ncia de v?rus neste e-mail, a empresa n?o se responsabiliza por quaisquer perdas ou danos decorrentes do uso da mensagem e seus anexos. A seguran?a e aus?ncia de erros na transmiss?o do e-mail n?o podem ser garantidas, j? que as informa??es podem ser interceptadas, corrompidas, perdidas, destru?das, atrasadas, chegarem incompletas, ou, ainda, conter v?rus. Recomendamos checar se o e-mail e seus anexos cont?m v?rus, uma vez que nem a Isimples Telecom ou o remetente se responsabilizam pela transmiss?o destes. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120507/8fec9bb5/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: logo_isimples.png Type: image/png Size: 18197 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120507/8fec9bb5/attachment-0001.png From jerry.richards at teotech.com Mon May 7 18:37:25 2012 From: jerry.richards at teotech.com (Jerry Richards) Date: Mon, 7 May 2012 14:37:25 +0000 Subject: [Freeswitch-users] Freeswitch And 2 Network I/Fs Message-ID: <1545146083A72C4DB7B66584B7E5D98402BBC647@BY2PRD0410MB377.namprd04.prod.outlook.com> Is there an example of configuring Freeswitch to support both eth0 and eth1, when they are connected to different networks? I know I can create two sip_profiles, which works fine for registration, but in the past, I've had issues with Freeswitch not sending media to the right interface (or at least, not knowing how to control that). Thanks, Jerry -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120507/aaf8ee1c/attachment.html From saami_mh at ymail.com Mon May 7 16:15:11 2012 From: saami_mh at ymail.com (Samira Mh) Date: Mon, 7 May 2012 05:15:11 -0700 (PDT) Subject: [Freeswitch-users] error/NO_ROUTE_DESTINATION while define groupserror/NO_ROUTE_DESTINATION Message-ID: <1336392911.49162.YahooMailNeo@web120104.mail.ne1.yahoo.com> Hello, i have defined the group named: "custom" in the below path as follow : /usr/src/freeswitch-1.0.6/conf/directory/default.xml ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ??? ? ? ? then issue "reloadxml" ?and?run the comand:?group_call ?custom but i come accross with the following error: error/NO_ROUTE_DESTINATION please help,all configs are defined correctly , but i couldn't define the "custom" group on freeswitch -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120507/f37bee40/attachment.html From jmesquita at freeswitch.org Mon May 7 18:46:30 2012 From: jmesquita at freeswitch.org (=?utf-8?Q?Jo=C3=A3o_Mesquita?=) Date: Mon, 7 May 2012 11:46:30 -0300 Subject: [Freeswitch-users] Freeswitch And 2 Network I/Fs In-Reply-To: <1545146083A72C4DB7B66584B7E5D98402BBC647@BY2PRD0410MB377.namprd04.prod.outlook.com> References: <1545146083A72C4DB7B66584B7E5D98402BBC647@BY2PRD0410MB377.namprd04.prod.outlook.com> Message-ID: <6CB54AEE0E4348E0B66FF250DFDBE286@freeswitch.org> Just look at the default internal profile configuration. Special attention to the ext-rtp-ip and ext-sip-ip. Other than that, rtp-ip and sip-ip should be all you need. -- Jo?o Mesquita Sent with Sparrow (http://www.sparrowmailapp.com/?sig) On Monday, May 7, 2012 at 11:37 AM, Jerry Richards wrote: > > Is there an example of configuring Freeswitch to support both eth0 and eth1, when they are connected to different networks? > > > > > > I know I can create two sip_profiles, which works fine for registration, but in the past, I've had issues with Freeswitch not sending media to the right interface (or at least, not knowing how to control that). > > > > > > Thanks, > Jerry > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120507/bf5988f7/attachment.html From B.Tietz at pinguin.ag Mon May 7 18:50:44 2012 From: B.Tietz at pinguin.ag (B.Tietz at pinguin.ag) Date: Mon, 7 May 2012 16:50:44 +0200 Subject: [Freeswitch-users] Freeswitch And 2 Network I/Fs In-Reply-To: <1545146083A72C4DB7B66584B7E5D98402BBC647@BY2PRD0410MB377.namprd04.prod.outlook.com> References: <1545146083A72C4DB7B66584B7E5D98402BBC647@BY2PRD0410MB377.namprd04.prod.outlook.com> Message-ID: <07BF4904977CC645B485E970424193AD10E9088BA2@localhost> Hi, from ODBC in the core in the wiki we have: setup the IP in the following params in each sip-profile: rtp-ip, sip-ip, presence-hosts, ext-rtp-ip, ext-sip-ip regards, Benjamin Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Jerry Richards Gesendet: Montag, 7. Mai 2012 16:37 An: 'freeswitch-users at lists.freeswitch.org' Betreff: [Freeswitch-users] Freeswitch And 2 Network I/Fs Is there an example of configuring Freeswitch to support both eth0 and eth1, when they are connected to different networks? I know I can create two sip_profiles, which works fine for registration, but in the past, I've had issues with Freeswitch not sending media to the right interface (or at least, not knowing how to control that). Thanks, Jerry -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120507/578d6522/attachment.html From vipkilla at gmail.com Mon May 7 18:53:03 2012 From: vipkilla at gmail.com (Vik Killa) Date: Mon, 7 May 2012 10:53:03 -0400 Subject: [Freeswitch-users] FreeSWITCH not sending P-Asserted-Identity Message-ID: I cannot get FS to send the P-Asserted-Identity header instead it always sends the P-Preferred-Identity header. According to http://wiki.freeswitch.org/wiki/Channel_Variables#sip_cid_type I must set the privacy flag in order for P-Asserted-Identity to be used, so here is my dialplan entry and the SIP INVITE that is sent: ------------------------------------------------------------------------ INVITE sip:+17165555555 at YYY.YYY.YYY.YYY:5060 SIP/2.0 Via: SIP/2.0/UDP 72.43.207.154;rport;branch=z9hG4bKK0pepF2H0atSp Max-Forwards: 68 From: "1000" ;tag=8U2rQ1pmXXaDQ To: Call-ID: de66ec94-12f5-1230-cbb1-5e6ae73de10d CSeq: 27864844 INVITE Contact: User-Agent: FSwitch 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 Privacy: id Content-Type: application/sdp Content-Disposition: session Content-Length: 209 X-FS-Support: update_display,send_info P-Preferred-Identity: "1000" v=0 o=FreeSWITCH 1336383832 1336383833 IN IP4 XX.XX.XX.XX s=FreeSWITCH c=IN IP4 XX.XX.XX.XX t=0 0 m=audio 17984 RTP/AVP 0 9 8 3 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 From vipkilla at gmail.com Mon May 7 19:12:48 2012 From: vipkilla at gmail.com (Vik Killa) Date: Mon, 7 May 2012 11:12:48 -0400 Subject: [Freeswitch-users] FreeSWITCH not sending P-Asserted-Identity In-Reply-To: References: Message-ID: I've also tried setting with same results. On Mon, May 7, 2012 at 10:53 AM, Vik Killa wrote: > I cannot get FS to send the P-Asserted-Identity header instead it > always sends the P-Preferred-Identity header. > According to http://wiki.freeswitch.org/wiki/Channel_Variables#sip_cid_type > I must set the privacy flag in order for P-Asserted-Identity to be > used, so here is my dialplan entry and the SIP INVITE that is sent: > > > ? ? > ? ? ? ? > ? ? ? ? data="effective_caller_id_name=${outbound_caller_id_name}"/> > ? ? ? ? data="effective_caller_id_number=${outbound_caller_id_number}"/> > ? ? ? ? > ? ? ? ? > ? ? ? ? > ? ? ? ? data="{sip_cid_type=pid,origination_privacy=hide_namehide_number}${lcr_auto_route}"/> > ? ? > > > > ? ------------------------------------------------------------------------ > ? INVITE sip:+17165555555 at YYY.YYY.YYY.YYY:5060 SIP/2.0 > ? Via: SIP/2.0/UDP 72.43.207.154;rport;branch=z9hG4bKK0pepF2H0atSp > ? Max-Forwards: 68 > ? From: "1000" ;tag=8U2rQ1pmXXaDQ > ? To: > ? Call-ID: de66ec94-12f5-1230-cbb1-5e6ae73de10d > ? CSeq: 27864844 INVITE > ? Contact: > ? User-Agent: FSwitch > ? 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 > ? Privacy: id > ? Content-Type: application/sdp > ? Content-Disposition: session > ? Content-Length: 209 > ? X-FS-Support: update_display,send_info > ? P-Preferred-Identity: "1000" > > ? v=0 > ? o=FreeSWITCH 1336383832 1336383833 IN IP4 XX.XX.XX.XX > ? s=FreeSWITCH > ? c=IN IP4 XX.XX.XX.XX > ? t=0 0 > ? m=audio 17984 RTP/AVP 0 9 8 3 101 13 > ? a=rtpmap:101 telephone-event/8000 > ? a=fmtp:101 0-16 > ? a=ptime:20 From djbinter at gmail.com Mon May 7 19:35:16 2012 From: djbinter at gmail.com (Dorn DJBinter) Date: Mon, 7 May 2012 08:35:16 -0700 Subject: [Freeswitch-users] FreeSWITCH not sending P-Asserted-Identity In-Reply-To: References: Message-ID: <-2643950612768503375@unknownmsgid> I think you missed the delimiter (+) on origination_privacy. -djbinter Sent from my iPad On May 7, 2012, at 8:14 AM, Vik Killa wrote: > I've also tried setting > > with same results. > > On Mon, May 7, 2012 at 10:53 AM, Vik Killa wrote: >> I cannot get FS to send the P-Asserted-Identity header instead it >> always sends the P-Preferred-Identity header. >> According to http://wiki.freeswitch.org/wiki/Channel_Variables#sip_cid_type >> I must set the privacy flag in order for P-Asserted-Identity to be >> used, so here is my dialplan entry and the SIP INVITE that is sent: >> >> >> >> >> > data="effective_caller_id_name=${outbound_caller_id_name}"/> >> > data="effective_caller_id_number=${outbound_caller_id_number}"/> >> >> >> >> > data="{sip_cid_type=pid,origination_privacy=hide_namehide_number}${lcr_auto_route}"/> >> >> >> >> >> ------------------------------------------------------------------------ >> INVITE sip:+17165555555 at YYY.YYY.YYY.YYY:5060 SIP/2.0 >> Via: SIP/2.0/UDP 72.43.207.154;rport;branch=z9hG4bKK0pepF2H0atSp >> Max-Forwards: 68 >> From: "1000" ;tag=8U2rQ1pmXXaDQ >> To: >> Call-ID: de66ec94-12f5-1230-cbb1-5e6ae73de10d >> CSeq: 27864844 INVITE >> Contact: >> User-Agent: FSwitch >> 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 >> Privacy: id >> Content-Type: application/sdp >> Content-Disposition: session >> Content-Length: 209 >> X-FS-Support: update_display,send_info >> P-Preferred-Identity: "1000" >> >> v=0 >> o=FreeSWITCH 1336383832 1336383833 IN IP4 XX.XX.XX.XX >> s=FreeSWITCH >> c=IN IP4 XX.XX.XX.XX >> t=0 0 >> m=audio 17984 RTP/AVP 0 9 8 3 101 13 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=ptime:20 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From philq at qsystemsengineering.com Mon May 7 19:54:09 2012 From: philq at qsystemsengineering.com (Phil Quesinberry) Date: Mon, 07 May 2012 11:54:09 -0400 Subject: [Freeswitch-users] Bypass media succeeds from extension to gateway but fails from extension to extension Message-ID: <011601cd2c69$a7206a00$f5613e00$@com> I'm not sure if this is a bug or just a NAT-related configurational problem. If it's truly a bug, let me know and I'll be happy to file a Jira. I'm trying to get bypass media to work with extension to extension calls. Both endpoint extensions are behind NAT in two different locations. This looks like a possible bug because FS sends the internal IP address of one of the endpoints to the other for media, but when making a call from the same extension to an external gateway for PSTN termination, it sends the phone's external IP address as it should, and the call succeeds. Everything works fine, of course, when "proxy media" is used. The SIP traffic for the failed call is here, look for the word "WRONG!" to see the incorrect address being passed in the SDP: http://pastebin.freeswitch.org/19003 Regards, Phil Quesinberry Q Systems Engineering, Inc. Electronic Controls and Embedded Systems Development (410) 969-8002 http://www.qsystemsengineering.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120507/faad01b0/attachment.html From vipkilla at gmail.com Mon May 7 20:16:54 2012 From: vipkilla at gmail.com (Vik Killa) Date: Mon, 7 May 2012 12:16:54 -0400 Subject: [Freeswitch-users] FreeSWITCH not sending P-Asserted-Identity In-Reply-To: <-2643950612768503375@unknownmsgid> References: <-2643950612768503375@unknownmsgid> Message-ID: This says nothing about a delimiter: http://wiki.freeswitch.org/wiki/Variable_origination_privacy On Mon, May 7, 2012 at 11:35 AM, Dorn DJBinter wrote: > I think you missed the delimiter (+) on origination_privacy. > > -djbinter > > Sent from my iPad > > On May 7, 2012, at 8:14 AM, Vik Killa wrote: > >> I've also tried setting >> >> with same results. >> >> On Mon, May 7, 2012 at 10:53 AM, Vik Killa wrote: >>> I cannot get FS to send the P-Asserted-Identity header instead it >>> always sends the P-Preferred-Identity header. >>> According to http://wiki.freeswitch.org/wiki/Channel_Variables#sip_cid_type >>> I must set the privacy flag in order for P-Asserted-Identity to be >>> used, so here is my dialplan entry and the SIP INVITE that is sent: >>> >>> >>> ? ? >>> ? ? ? ? >>> ? ? ? ?>> data="effective_caller_id_name=${outbound_caller_id_name}"/> >>> ? ? ? ?>> data="effective_caller_id_number=${outbound_caller_id_number}"/> >>> ? ? ? ? >>> ? ? ? ? >>> ? ? ? ? >>> ? ? ? ?>> data="{sip_cid_type=pid,origination_privacy=hide_namehide_number}${lcr_auto_route}"/> >>> ? ? >>> >>> >>> >>> ? ------------------------------------------------------------------------ >>> ? INVITE sip:+17165555555 at YYY.YYY.YYY.YYY:5060 SIP/2.0 >>> ? Via: SIP/2.0/UDP 72.43.207.154;rport;branch=z9hG4bKK0pepF2H0atSp >>> ? Max-Forwards: 68 >>> ? From: "1000" ;tag=8U2rQ1pmXXaDQ >>> ? To: >>> ? Call-ID: de66ec94-12f5-1230-cbb1-5e6ae73de10d >>> ? CSeq: 27864844 INVITE >>> ? Contact: >>> ? User-Agent: FSwitch >>> ? 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 >>> ? Privacy: id >>> ? Content-Type: application/sdp >>> ? Content-Disposition: session >>> ? Content-Length: 209 >>> ? X-FS-Support: update_display,send_info >>> ? P-Preferred-Identity: "1000" >>> >>> ? v=0 >>> ? o=FreeSWITCH 1336383832 1336383833 IN IP4 XX.XX.XX.XX >>> ? s=FreeSWITCH >>> ? c=IN IP4 XX.XX.XX.XX >>> ? t=0 0 >>> ? m=audio 17984 RTP/AVP 0 9 8 3 101 13 >>> ? a=rtpmap:101 telephone-event/8000 >>> ? a=fmtp:101 0-16 >>> ? a=ptime:20 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From vipkilla at gmail.com Mon May 7 20:21:12 2012 From: vipkilla at gmail.com (Vik Killa) Date: Mon, 7 May 2012 12:21:12 -0400 Subject: [Freeswitch-users] FreeSWITCH not sending P-Asserted-Identity In-Reply-To: References: <-2643950612768503375@unknownmsgid> Message-ID: Also, I just tried it using the + delimiter, it's still sending P-Preferred-Identity On Mon, May 7, 2012 at 12:16 PM, Vik Killa wrote: > This says nothing about a delimiter: > http://wiki.freeswitch.org/wiki/Variable_origination_privacy > > > > On Mon, May 7, 2012 at 11:35 AM, Dorn DJBinter wrote: >> I think you missed the delimiter (+) on origination_privacy. >> >> -djbinter >> From marketing at cluecon.com Mon May 7 20:31:37 2012 From: marketing at cluecon.com (Michael Collins) Date: Mon, 7 May 2012 09:31:37 -0700 Subject: [Freeswitch-users] News and Notes: ClueCon Hotel Gets a New Name, FreeSWITCH Gets a New Feature Message-ID: Happy new week to all! We have some interesting news items to share. First off, we are happy to report that the Wyndham Hotel, where ClueCon 2012 is being held, has a new owner and will become a Hyatt property. As part of the transition, the Wyndham will be renamed in June. The new name will be Hyatt Chicago Miracle Mile. The ownership transfer and name change will not affect ClueCon attendees, and the ClueCon website will continue to have the latest hotel information. Please feel free to call the Wyndham and make your ClueCon hotel reservations at your earliest convenience. Also in ClueCon news we'd like to let everyone know that we have two new sponsors: Vitelity and OpenVox. We are pleased to welcome them both to this year's event. Each will be giving a technical presentation this year and we look forward to seeing them in person. Please visit our sitefor registration information. FreeSWITCH received a new feature last week: a much-improved group pickup implementation. The new pickup dialplanapplication works in conjunction with the new "pickup" pseudo-channel. Together these allow for simple yet elegant call pickup features to be added to the dialplan. A number of FreeSWITCH community members are testing this new feature, and soon there will be an example added to the "Vanilla" dialplan. This Wednesday, May 9, we will be welcoming FreeSWITCH Community member Giovanni Maruzzelli to the community conference call. He will be discussing how to implement a GSM channel in FreeSWITCH using mod_gsmopen and a GSM dongle. We invite everyone to join us this Wednesday at 1PM Eastern, 10AM Pacific (UTC 1700) for a very interesting discussion. Have a great week and we'll be in touch next Monday. -- Michael S Collins ClueCon Team www.ClueCon.com 877-7-4ACLUE -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120507/be2311a1/attachment-0001.html From sarahig1985 at gmail.com Mon May 7 21:10:57 2012 From: sarahig1985 at gmail.com (Sara Higfler) Date: Mon, 7 May 2012 18:10:57 +0100 Subject: [Freeswitch-users] Javascript Outbound Event Socket - Linger Command In-Reply-To: References: Message-ID: Hi Michael, Thanks for your feedback - I'm glad that I hadn't missed anything obvious via google. I think I've made good progress with this issue today. It seems I can use either of the following to fire the linger command: req.linger(); req.send("linger"); Within the node debug logs I see the following, which confirms the request was successful. 7 May 12:55:44 - on_data(Content-Type: command/reply Reply-Text: +OK Content-Type: command/reply Reply-Text: +OK will linger 600 seconds What has really had me confused was which callback event to be expecting in the case of linger being invoked. After further reviewing the partial examples I could find and looking at the debug traces, it seems I have to arm the callback on the "esl_disconnect_notice" event. Within this callback, I need to check the content-dispostion header to determine if it is a disconnect or a linger. I'm using the following code today, which seems to pick up on every call termination. It would be good if someone can confirm if I'm cleaning up appropriately in the two cases (disconnect and linger). req.on('esl_disconnect_notice', function(req) { switch (req.headers['Content-Disposition']) { case 'disconnect': req.end(); break; case 'linger': req.exit(); break; } console.log('esl_disconnect_notice:'+req.headers['Content-Disposition']); return util.log('esl_disconnect_notice'); }); Thanks again! On Mon, May 7, 2012 at 3:14 AM, Michael Collins wrote: > Hi Sara, > > Welcome to FreeSWITCH. You are quite right - there is not documentation on > how to send the linger command with an ESL application. I grep'd around the > libs/esl directory and found that fs_cli.c and esl.c have a function called > esl_send_recv and a few occurrences of it used like this: > > esl_send_recv(&handle, "linger"); > > I'm no Javascript person but if I understand the syntax correctly you'd > need to do something like this as soon as the call is connected, i.e. right > before the req.execute("answer") line: > > req.send_recv("linger"); > > Could you give that a try and let us know if it works? If it does we'd > appreciate you updating the wiki. :) > > Thanks, > MC > > On Sat, May 5, 2012 at 12:54 PM, Sara Higfler wrote: > >> Hi, >> >> I'm a newbie to developing an outbound event handler for Freeswitch. I'm >> looking to use the Javascript ESL implemention (using node.js) and have >> managed to get basic scenarios working, including digit collection and >> checking against a MySQL database. One problem I have is the capture of >> call termination events in my script. Having read around a lot, I know >> that I'm meant to use the Linger command, but cannot find any examples of >> how to do this with a Javascript outbound handler. I've included the rough >> structure of my code below (details removed for brevity) - I would really >> appreciate if someone could help show me how I would implement the linger >> command to ensure I capture all call termination events. >> >> Kind regards. >> >> (function() >> { >> var server, esl; >> >> esl = require('esl'); >> util = require('util'); >> >> server = esl.createCallServer(); >> >> server.on('CONNECT', function(req, res) >> { >> var uri, channel_data, unique_id; >> >> channel_data = req.body; >> unique_id = channel_data['Unique-ID']; >> >> req.execute('answer'); >> >> req.execute('playback', 'hello.wav'); >> >> req.on('DTMF', function(req) >> { >> var digit; >> var channel_data; >> channel_data = req.body; >> >> digit = channel_data['DTMF-Digit']; >> console.log('DTMF Received=' + digit); >> return util.log('DTMF Received'); >> }); >> >> req.on('CHANNEL_ANSWER', function(req) >> { >> return util.log('Call was answered'); >> }); >> >> req.on('CHANNEL_HANGUP', function(req) >> { >> console.log('CHANNEL_HANGUP'); >> return util.log('CHANNEL_HANGUP'); >> }); >> req.on('CHANNEL_HANGUP_COMPLETE', function(req) >> { >> console.log('CHANNEL_HANGUP_COMPLETE'); >> return util.log('CHANNEL_HANGUP_COMPLETE'); >> }); >> >> req.on('DISCONNECT', function(req) >> { >> console.log('DISCONNECT'); >> return util.log('DISCONNECT'); >> }) >> >> }); >> return util.log('CONNECT received'); >> }); >> >> server.listen(9123); >> }).call(this); >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120507/2d2b32bb/attachment.html From msc at freeswitch.org Mon May 7 21:14:52 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 7 May 2012 10:14:52 -0700 Subject: [Freeswitch-users] Javascript Outbound Event Socket - Linger Command In-Reply-To: References: Message-ID: Thanks for the update. If this does indeed work for you we would appreciate you putting your information on the wiki. Thanks, MC On Mon, May 7, 2012 at 10:10 AM, Sara Higfler wrote: > Hi Michael, > > Thanks for your feedback - I'm glad that I hadn't missed anything obvious > via google. I think I've made good progress with this issue today. It > seems I can use either of the following to fire the linger command: > req.linger(); > req.send("linger"); > > Within the node debug logs I see the following, which confirms the request > was successful. > > 7 May 12:55:44 - on_data(Content-Type: command/reply > Reply-Text: +OK > > Content-Type: command/reply > Reply-Text: +OK will linger 600 seconds > > What has really had me confused was which callback event to be expecting > in the case of linger being invoked. After further reviewing the partial > examples I could find and looking at the debug traces, it seems I have to > arm the callback on the "esl_disconnect_notice" event. Within this > callback, I need to check the content-dispostion header to determine if it > is a disconnect or a linger. I'm using the following code today, which > seems to pick up on every call termination. It would be good if someone > can confirm if I'm cleaning up appropriately in the two cases (disconnect > and linger). > > req.on('esl_disconnect_notice', function(req) > { > switch (req.headers['Content-Disposition']) > { > case 'disconnect': > req.end(); > break; > case 'linger': > req.exit(); > break; > } > > console.log('esl_disconnect_notice:'+req.headers['Content-Disposition']); > return util.log('esl_disconnect_notice'); > }); > > Thanks again! > > On Mon, May 7, 2012 at 3:14 AM, Michael Collins wrote: > >> Hi Sara, >> >> Welcome to FreeSWITCH. You are quite right - there is not documentation >> on how to send the linger command with an ESL application. I grep'd around >> the libs/esl directory and found that fs_cli.c and esl.c have a function >> called esl_send_recv and a few occurrences of it used like this: >> >> esl_send_recv(&handle, "linger"); >> >> I'm no Javascript person but if I understand the syntax correctly you'd >> need to do something like this as soon as the call is connected, i.e. right >> before the req.execute("answer") line: >> >> req.send_recv("linger"); >> >> Could you give that a try and let us know if it works? If it does we'd >> appreciate you updating the wiki. :) >> >> Thanks, >> MC >> >> On Sat, May 5, 2012 at 12:54 PM, Sara Higfler wrote: >> >>> Hi, >>> >>> I'm a newbie to developing an outbound event handler for Freeswitch. >>> I'm looking to use the Javascript ESL implemention (using node.js) and have >>> managed to get basic scenarios working, including digit collection and >>> checking against a MySQL database. One problem I have is the capture of >>> call termination events in my script. Having read around a lot, I know >>> that I'm meant to use the Linger command, but cannot find any examples of >>> how to do this with a Javascript outbound handler. I've included the rough >>> structure of my code below (details removed for brevity) - I would really >>> appreciate if someone could help show me how I would implement the linger >>> command to ensure I capture all call termination events. >>> >>> Kind regards. >>> >>> (function() >>> { >>> var server, esl; >>> >>> esl = require('esl'); >>> util = require('util'); >>> >>> server = esl.createCallServer(); >>> >>> server.on('CONNECT', function(req, res) >>> { >>> var uri, channel_data, unique_id; >>> >>> channel_data = req.body; >>> unique_id = channel_data['Unique-ID']; >>> >>> req.execute('answer'); >>> >>> req.execute('playback', 'hello.wav'); >>> >>> req.on('DTMF', function(req) >>> { >>> var digit; >>> var channel_data; >>> channel_data = req.body; >>> >>> digit = channel_data['DTMF-Digit']; >>> console.log('DTMF Received=' + digit); >>> return util.log('DTMF Received'); >>> }); >>> >>> req.on('CHANNEL_ANSWER', function(req) >>> { >>> return util.log('Call was answered'); >>> }); >>> >>> req.on('CHANNEL_HANGUP', function(req) >>> { >>> console.log('CHANNEL_HANGUP'); >>> return util.log('CHANNEL_HANGUP'); >>> }); >>> req.on('CHANNEL_HANGUP_COMPLETE', function(req) >>> { >>> console.log('CHANNEL_HANGUP_COMPLETE'); >>> return util.log('CHANNEL_HANGUP_COMPLETE'); >>> }); >>> >>> req.on('DISCONNECT', function(req) >>> { >>> console.log('DISCONNECT'); >>> return util.log('DISCONNECT'); >>> }) >>> >>> }); >>> return util.log('CONNECT received'); >>> }); >>> >>> server.listen(9123); >>> }).call(this); >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120507/eb2deb0f/attachment-0001.html From djbinter at gmail.com Mon May 7 23:05:14 2012 From: djbinter at gmail.com (DJB International) Date: Mon, 7 May 2012 12:05:14 -0700 Subject: [Freeswitch-users] FreeSWITCH not sending P-Asserted-Identity In-Reply-To: References: <-2643950612768503375@unknownmsgid> Message-ID: >From what I understand, if you use the privacy dialplan application, and set it to the proper flags, then it should do P-Asserted. -djbinter On Mon, May 7, 2012 at 9:21 AM, Vik Killa wrote: > Also, I just tried it using the + delimiter, it's still sending > P-Preferred-Identity > > On Mon, May 7, 2012 at 12:16 PM, Vik Killa wrote: > > This says nothing about a delimiter: > > http://wiki.freeswitch.org/wiki/Variable_origination_privacy > > > > > > > > On Mon, May 7, 2012 at 11:35 AM, Dorn DJBinter > wrote: > >> I think you missed the delimiter (+) on origination_privacy. > >> > >> -djbinter > >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120507/1be87137/attachment.html From vipkilla at gmail.com Mon May 7 23:10:27 2012 From: vipkilla at gmail.com (Vik Killa) Date: Mon, 7 May 2012 15:10:27 -0400 Subject: [Freeswitch-users] FreeSWITCH not sending P-Asserted-Identity In-Reply-To: References: <-2643950612768503375@unknownmsgid> Message-ID: That is what I thought but clearly it is not working.... On Mon, May 7, 2012 at 3:05 PM, DJB International wrote: > >From what I understand, if you use the privacy dialplan application, and > set it to the proper flags, then it should do P-Asserted. > > -djbinter > From nasida at live.ru Mon May 7 23:37:10 2012 From: nasida at live.ru (Yuriy Nasida) Date: Mon, 7 May 2012 23:37:10 +0400 Subject: [Freeswitch-users] productivity of FS Message-ID: Hello guys, I have FS box with 200-250 concurrent calls. FS git 05/03/2012. Intel(R) Xeon(R) CPU E5620 @ 2.40GHzMemTotal: 12187540 kB I use xml_mod_curl + php for dynamic directory\dialplan. Also I use lua scripts + mysql for some cases. I use sip options ping for some checking of productivity.Sometimes (when loading is high) I have big delay with sip ping (>2 sec). The box which sends sip ping in same network with main box. My question:1) Is it many concurrent calls for setup I have ?2) If not, what the best way for understanding where I lose productivity ? Please advise.Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120507/ce7ebc38/attachment.html From jmesquita at freeswitch.org Mon May 7 23:40:17 2012 From: jmesquita at freeswitch.org (=?utf-8?Q?Jo=C3=A3o_Mesquita?=) Date: Mon, 7 May 2012 16:40:17 -0300 Subject: [Freeswitch-users] productivity of FS In-Reply-To: References: Message-ID: <94F2FEE465A940E8A3172DED85F24A91@freeswitch.org> There is so much information missing on your problem description that I don't even know where to start. When you say load is high, how high? Do you have an exact number? Regards, -- Jo?o Mesquita Sent with Sparrow (http://www.sparrowmailapp.com/?sig) On Monday, May 7, 2012 at 4:37 PM, Yuriy Nasida wrote: > Hello guys, > > I have FS box with 200-250 concurrent calls. FS git 05/03/2012. > Intel(R) Xeon(R) CPU E5620 @ 2.40GHz > MemTotal: 12187540 kB > > I use xml_mod_curl + php for dynamic directory\dialplan. Also I use lua scripts + mysql for some cases. > > I use sip options ping for some checking of productivity. > Sometimes (when loading is high) I have big delay with sip ping (>2 sec). The box which sends sip ping in same network with main box. > > My question: > 1) Is it many concurrent calls for setup I have ? > 2) If not, what the best way for understanding where I lose productivity ? > > Please advise. > Thanks. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120507/5c2e8824/attachment.html From nasida at live.ru Tue May 8 00:05:11 2012 From: nasida at live.ru (Yuriy Nasida) Date: Tue, 8 May 2012 00:05:11 +0400 Subject: [Freeswitch-users] productivity of FS In-Reply-To: <94F2FEE465A940E8A3172DED85F24A91@freeswitch.org> References: , <94F2FEE465A940E8A3172DED85F24A91@freeswitch.org> Message-ID: I mean 200-250 concurrent calls. Please let me know what information from me can help as well. Thanks. Date: Mon, 7 May 2012 16:40:17 -0300 From: jmesquita at freeswitch.org To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] productivity of FS There is so much information missing on your problem description that I don't even know where to start. When you say load is high, how high? Do you have an exact number? Regards, -- Jo?o MesquitaSent with Sparrow On Monday, May 7, 2012 at 4:37 PM, Yuriy Nasida wrote: Hello guys, I have FS box with 200-250 concurrent calls. FS git 05/03/2012. Intel(R) Xeon(R) CPU E5620 @ 2.40GHzMemTotal: 12187540 kB I use xml_mod_curl + php for dynamic directory\dialplan. Also I use lua scripts + mysql for some cases. I use sip options ping for some checking of productivity.Sometimes (when loading is high) I have big delay with sip ping (>2 sec). The box which sends sip ping in same network with main box. My question:1) Is it many concurrent calls for setup I have ?2) If not, what the best way for understanding where I lose productivity ? Please advise.Thanks. _________________________________________________________________________Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120508/26930a94/attachment-0001.html From msc at freeswitch.org Tue May 8 00:11:59 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 7 May 2012 13:11:59 -0700 Subject: [Freeswitch-users] error/NO_ROUTE_DESTINATION while define groupserror/NO_ROUTE_DESTINATION In-Reply-To: <1336392911.49162.YahooMailNeo@web120104.mail.ne1.yahoo.com> References: <1336392911.49162.YahooMailNeo@web120104.mail.ne1.yahoo.com> Message-ID: Most likely the configuration did not make it into the right part of the XML tree. Can you put your entire default.xml file on pastebin.freeswitch.org? We should be able to help you figure it out. FYI, copied and pasted your stuff right into my default.xml file and it worked like a champ. All I did was paste it in, save the file, and hit F6 at fs_cli and it worked just fine: freeswitch at default> group_call custom [sip_invite_domain=192.168.1.79,presence_id=1005 at 192.168.1.79 ]error/user_not_registered,[sip_invite_domain=192.168.1.79,presence_id= 1006 at 192.168.1.79]error/user_not_registered freeswitch at default> -MC On Mon, May 7, 2012 at 5:15 AM, Samira Mh wrote: > Hello, > i have defined the group named: "custom" in the below path as follow : > > /usr/src/freeswitch-1.0.6/conf/directory/default.xml > > > > > > > > > > > then issue "reloadxml" and run the comand: group_call custom > but i come accross with the following error: > error/NO_ROUTE_DESTINATION > > please help,all configs are defined correctly , but i couldn't define the > "custom" group on freeswitch > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120507/f93b56fd/attachment.html From bdfoster at endigotech.com Tue May 8 00:29:08 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Mon, 7 May 2012 16:29:08 -0400 Subject: [Freeswitch-users] productivity of FS In-Reply-To: References: <94F2FEE465A940E8A3172DED85F24A91@freeswitch.org> Message-ID: What you've asked has a long and complex answer. It depends on several things. 200-250 concurrent calls isn't unheard of, people in this very community have done 1000-3000 calls easily. I'd re-read what Joao has asked. By the way having other things like MySQL or a webserver running on the server can severely restrict what you can handle as far as freeswitch is concerned. You have a finite amount of system resources available to you, no matter what you decide to use the server for. -BDF On May 7, 2012 4:06 PM, "Yuriy Nasida" wrote: > I mean 200-250 concurrent calls. Please let me know what information > from me can help as well. > > Thanks. > > ------------------------------ > Date: Mon, 7 May 2012 16:40:17 -0300 > From: jmesquita at freeswitch.org > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] productivity of FS > > There is so much information missing on your problem description that I > don't even know where to start. When you say load is high, how high? Do you > have an exact number? > > Regards, > > -- > Jo?o Mesquita > Sent with Sparrow > > On Monday, May 7, 2012 at 4:37 PM, Yuriy Nasida wrote: > > Hello guys, > > I have FS box with 200-250 concurrent calls. FS git 05/03/2012. > Intel(R) Xeon(R) CPU E5620 @ 2.40GHz > MemTotal: 12187540 kB > > I use xml_mod_curl + php for dynamic directory\dialplan. Also I use lua > scripts + mysql for some cases. > > I use sip options ping for some checking of productivity. > Sometimes (when loading is high) I have big delay with sip ping (>2 sec). > The box which sends sip ping in same network with main box. > > My question: > 1) Is it many concurrent calls for setup I have ? > 2) If not, what the best way for understanding where I lose productivity ? > > Please advise. > Thanks. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel > Communication Server Official FreeSWITCH Sites > http://www.freeswitch.org http://wiki.freeswitch.org > http://www.cluecon.com FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120507/dc85184c/attachment.html From brandon.mcginty at gmail.com Tue May 8 01:53:00 2012 From: brandon.mcginty at gmail.com (brandon.mcginty at gmail.com) Date: Mon, 7 May 2012 17:53:00 -0400 Subject: [Freeswitch-users] Recording all calls with good quality. In-Reply-To: References: <20120506144621.GA6722@bmcginty.hopto.org> Message-ID: <20120507215300.GC6722@bmcginty.hopto.org> I've tried the following codec settings, set in vars.xml. The first and third lines are the defaults upon install. My CPU load shows fine, even with 2 clients connected to an external (secondary) switch, through this one. I truly appreciate the help. Brandon McGinty-Carroll On Sun, May 06, 2012 at 06:50:44PM +0300, Avi Marcus wrote: > AFAIK, gsm is lossy, which is why it doesn't sound good. > wav, however, should sound great. What codec are your calls in? Are you > recording many calls? > Check the load average while you are recording. If it's greater than the > number of CPUs then probably your hard drives can't keep up. There are > several options then, first let see if we can diagnose the issue. > > -Avi > > > On Sun, May 6, 2012 at 5:46 PM, wrote: > > > Good day. > > I'm trying to record all calls sent through my Freeswitch PBX. > > The problem is that the quality is horrible. I've tried wav and GSM > > formats, and both are les than satisfactory. > > I'm looking for something around the quality of the recent recordings of > > the Freeswitch conference. (I checked on the wiki, and couldn't find > > anything through search.) > > Any information on the FS conference format or my configuration would be > > appreciated. > > I've got: > > > > > > > > > > > > > data="$${base_dir}/recordings/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/> > > > > > > Sincerely, > > Brandon McGinty-Carroll > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From philq at qsystemsengineering.com Tue May 8 02:01:10 2012 From: philq at qsystemsengineering.com (Phil Quesinberry) Date: Mon, 07 May 2012 18:01:10 -0400 Subject: [Freeswitch-users] productivity of FS Message-ID: <024c01cd2c9c$ec638d80$c52aa880$@com> I must echo the sentiments of the other more experienced users here. A couple of quick questions do come to mind though. Is FS proxying the media between the endpoints? If so, is there any transcoding going on? If that's the case, you may want to enable late negotiation and tweak the codecs to avoid transcoding whenever possible, that can potentially take up a lot of system resources. Phil Quesinberry Q Systems Engineering, Inc. Electronic Controls and Embedded Systems Development (410) 969-8002 http://www.qsystemsengineering.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120507/32228456/attachment-0001.html From gabe at gundy.org Tue May 8 05:15:57 2012 From: gabe at gundy.org (Gabriel Gunderson) Date: Mon, 7 May 2012 19:15:57 -0600 Subject: [Freeswitch-users] SIP NOTIFY for MWI with AVAYA 9630 In-Reply-To: References: Message-ID: On Sat, May 5, 2012 at 10:15 AM, Bernard Murphy wrote: > On anything newer than 1.06 of freeswitch the SIP NOTIFY sent to an Avaya > 9630 phone does not get accepted and trigger the MWI light. Works absolutely > fine on 1.0.6. Time to file a bug in Jira? Gabe From gabe at gundy.org Tue May 8 05:24:33 2012 From: gabe at gundy.org (Gabriel Gunderson) Date: Mon, 7 May 2012 19:24:33 -0600 Subject: [Freeswitch-users] execute_on_fax_success/failure In-Reply-To: <68A9174972294D3DB4061464B242508A@owner397fa27d2> References: <99B4785CBF5B4B5F9082207335342A9E@owner397fa27d2> <68A9174972294D3DB4061464B242508A@owner397fa27d2> Message-ID: On Sun, May 6, 2012 at 6:29 PM, Darcy Primrose wrote: > Brian, thank you so much for taking time to go thru this and explain it, > works like a charm, so many little things the freeswitch does, I understand > why the wiki documentation is so important.? I do see in spandsp where it > tells you to?escape with?\\\, it just never lit the light bulb. Does this need more clarification on the wiki? Does it need mentioned in other places? If so, you might consider updating it to help the next time you or someone else goes looking for it ;) Gabe From gabe at gundy.org Tue May 8 05:34:48 2012 From: gabe at gundy.org (Gabriel Gunderson) Date: Mon, 7 May 2012 19:34:48 -0600 Subject: [Freeswitch-users] Bypass media succeeds from extension to gateway but fails from extension to extension In-Reply-To: <011601cd2c69$a7206a00$f5613e00$@com> References: <011601cd2c69$a7206a00$f5613e00$@com> Message-ID: On Mon, May 7, 2012 at 9:54 AM, Phil Quesinberry wrote: > I?m trying to get bypass media to work with extension to extension calls. > Both endpoint extensions are behind NAT in two different locations.? This > looks like a possible bug because FS sends the internal IP address of one of > the endpoints to the other for media, but when making a call from the same > extension to an external gateway for PSTN termination, it sends the phone?s > external IP address as it should, and the call succeeds. My guess is that this is not a bug. The NAT stuff has been gone over and over and used in many different configurations. Usually, in my experience, NAT issues are cause by config errors or a broken device. What do you have for your sofia configs? Gabe From gabe at gundy.org Tue May 8 05:37:24 2012 From: gabe at gundy.org (Gabriel Gunderson) Date: Mon, 7 May 2012 19:37:24 -0600 Subject: [Freeswitch-users] FreeSWITCH not sending P-Asserted-Identity In-Reply-To: References: <-2643950612768503375@unknownmsgid> Message-ID: On Mon, May 7, 2012 at 1:10 PM, Vik Killa wrote: > That is what I thought but clearly it is not working.... So what are you thinking at this point? Need to update the wiki or open a Jira? Gabe From psossa.mobile at gmail.com Tue May 8 05:48:55 2012 From: psossa.mobile at gmail.com (Paulo Sossa) Date: Mon, 7 May 2012 19:48:55 -0600 Subject: [Freeswitch-users] DISA on EXTENSION - Got plivo installed ... now what? Message-ID: Hi Folks Reaching out to the community to see if I can get a tiny little help I'm trying to set up DISA on my extension 1005 But I all get is a busy signal with the error log below I can call out, no problem using my ITSP. I am planning to use PLIVO right after I get DISA going. am I missing something on the IVR? as it tells me "In Context, No Route, Abort..." I followed this instructions with no luck.... http://wiki.freeswitch.org/wiki/Dialplan_XML#Example_19:_DISA Thanks for the help in advance. MY LOG: 2012-05-07 18:36:01.596130 [CONSOLE] switch_core.c:1902 FreeSWITCH Version 1.1.beta1 (git-982cb1f 2012-04-04 09-04-12 +0200) Started. Max Sessions[1000] Session Rate[30] SQL [Enabled] 2012-05-07 18:36:04.974462 [NOTICE] switch_channel.c:926 New Channel sofia/internal/1001 at 216.70.70.79 [39e376c5-3a70-4355-825c-da97aca3a395] 2012-05-07 18:36:04.974462 [INFO] mod_dialplan_xml.c:485 Processing PSS 1001 <1001>->1005 in context outbound 2012-05-07 18:36:04.974462 [INFO] switch_core_state_machine.c:177 No Route, Aborting 2012-05-07 18:36:04.974462 [NOTICE] switch_core_state_machine.c:178 Hangup sofia/internal/1001 at 216.70.70.79 [CS_ROUTING] [NO_ROUTE_DESTINATION] 2012-05-07 18:36:04.974462 [NOTICE] switch_core_session.c:1400 Session 1 (sofia/internal/1001 at 216.70.70.79) Ended 2012-05-07 18:36:04.974462 [NOTICE] switch_core_session.c:1402 Close Channel sofia/internal/1001 at 216.70.70.79 [CS_DESTROY] From saami_mh at ymail.com Tue May 8 07:44:06 2012 From: saami_mh at ymail.com (Samira Mh) Date: Mon, 7 May 2012 20:44:06 -0700 (PDT) Subject: [Freeswitch-users] error/NO_ROUTE_DESTINATION while define groupserror/NO_ROUTE_DESTINATION In-Reply-To: References: <1336392911.49162.YahooMailNeo@web120104.mail.ne1.yahoo.com> Message-ID: <1336448646.54490.YahooMailNeo@web120101.mail.ne1.yahoo.com> hello, thanks alot for your help i put the entire of the default.xml on?http://pastebin.freeswitch.org/19008?, i am sure all of the settings are correctly.. sorry but i don't understand how to?copied and pasted my stuff into your?default.xml file? and hit F6 ? how can i do that? ________________________________ From: Michael Collins To: FreeSWITCH Users Help Sent: Tuesday, May 8, 2012 12:41 AM Subject: Re: [Freeswitch-users] error/NO_ROUTE_DESTINATION while define groupserror/NO_ROUTE_DESTINATION Most likely the configuration did not make it into the right part of the XML tree. Can you put your entire default.xml file on pastebin.freeswitch.org? We should be able to help you figure it out. FYI, copied and pasted your stuff right into my default.xml file and it worked like a champ. All I did was paste it in, save the file, and hit F6 at fs_cli and it worked just fine: freeswitch at default> group_call custom [sip_invite_domain=192.168.1.79,presence_id=1005 at 192.168.1.79]error/user_not_registered,[sip_invite_domain=192.168.1.79,presence_id=1006 at 192.168.1.79]error/user_not_registered freeswitch at default> -MC On Mon, May 7, 2012 at 5:15 AM, Samira Mh wrote: Hello, >i have defined the group named: "custom" in the below path as follow : > > >/usr/src/freeswitch-1.0.6/conf/directory/default.xml > > > > > > > >? >? ? ? ? >? ? ? ? ? >? ? ? ? ? >? ? ??? >? ? ? > > >then issue "reloadxml" ?and?run the comand:?group_call ?custom >but i come accross with the following error: >error/NO_ROUTE_DESTINATION > > >please help,all configs are defined correctly , but i couldn't define the "custom" group on freeswitch >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >FreeSWITCH-users mailing list >FreeSWITCH-users at 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 at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120507/649592c9/attachment.html From msc at freeswitch.org Tue May 8 07:55:19 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 7 May 2012 20:55:19 -0700 Subject: [Freeswitch-users] error/NO_ROUTE_DESTINATION while define groupserror/NO_ROUTE_DESTINATION In-Reply-To: <1336448646.54490.YahooMailNeo@web120101.mail.ne1.yahoo.com> References: <1336392911.49162.YahooMailNeo@web120104.mail.ne1.yahoo.com> <1336448646.54490.YahooMailNeo@web120101.mail.ne1.yahoo.com> Message-ID: Samira, When I say that I pressed F6 I mean that I went to fs_cli and pressed F6. You can also do "reloadxml". If you don't reload your xml after making a change then FreeSWITCH can't apply those changes. I copied your xml right out of the pastebin and put it into conf/directory/default.xml on my system and it worked just fine. I think that perhaps you need to issue the "reloadxml" command or that you maybe have edited the wrong file. Please confirm that you have edited /usr/local/freeswitch/conf/directory/default.xml and not some other file, perhaps the default.xml in the source tree. -MC On Mon, May 7, 2012 at 8:44 PM, Samira Mh wrote: > > hello, > thanks alot for your help > i put the entire of the default.xml on > http://pastebin.freeswitch.org/19008 , i am sure all of the settings are > correctly.. > sorry but i don't understand how to copied and pasted my stuff into your default.xml > file > and hit F6 ? how can i do that? > ------------------------------ > *From:* Michael Collins > *To:* FreeSWITCH Users Help > *Sent:* Tuesday, May 8, 2012 12:41 AM > *Subject:* Re: [Freeswitch-users] error/NO_ROUTE_DESTINATION while define > groupserror/NO_ROUTE_DESTINATION > > Most likely the configuration did not make it into the right part of the > XML tree. Can you put your entire default.xml file on > pastebin.freeswitch.org? We should be able to help you figure it out. > FYI, copied and pasted your stuff right into my default.xml file and it > worked like a champ. All I did was paste it in, save the file, and hit F6 > at fs_cli and it worked just fine: > > freeswitch at default> group_call custom > [sip_invite_domain=192.168.1.79,presence_id=1005 at 192.168.1.79 > ]error/user_not_registered,[sip_invite_domain=192.168.1.79,presence_id= > 1006 at 192.168.1.79]error/user_not_registered > freeswitch at default> > > -MC > > On Mon, May 7, 2012 at 5:15 AM, Samira Mh wrote: > > Hello, > i have defined the group named: "custom" in the below path as follow : > > /usr/src/freeswitch-1.0.6/conf/directory/default.xml > > > > > > > > > > > then issue "reloadxml" and run the comand: group_call custom > but i come accross with the following error: > error/NO_ROUTE_DESTINATION > > please help,all configs are defined correctly , but i couldn't define the > "custom" group on freeswitch > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120507/c2ec5c4b/attachment-0001.html From msc at freeswitch.org Tue May 8 08:01:11 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 7 May 2012 21:01:11 -0700 Subject: [Freeswitch-users] DISA on EXTENSION - Got plivo installed ... now what? In-Reply-To: References: Message-ID: Hi Paulo, Welcome to the FreeSWITCH-users mailing list! I'm going to recommend that you get a console debug log and put it in pastebin.freeswitch.org. If you need guidance on how to do that you'll find the data collection information on this wiki page to be extremely helpful: http://wiki.freeswitch.org/wiki/Reporting_Bugs#Debugging_Steps The log you pasted here is an info level debug. I suspect that you have FreeSWITCH running in the foreground and you copied the log lines right from the terminal window. I recommend using fs_cli and letting FreeSWITCH run in the background by using the "-nc" flag. We consider that a standard practice and I think you'll find that it works well. Thanks, MC On Mon, May 7, 2012 at 6:48 PM, Paulo Sossa wrote: > Hi Folks > Reaching out to the community to see if I can get a tiny little help > > I'm trying to set up DISA on my extension 1005 > > But I all get is a busy signal with the error log below > > I can call out, no problem using my ITSP. > > I am planning to use PLIVO right after I get DISA going. > > am I missing something on the IVR? as it tells me "In Context, No > Route, Abort..." > > I followed this instructions with no luck.... > http://wiki.freeswitch.org/wiki/Dialplan_XML#Example_19:_DISA > > Thanks for the help in advance. > > MY LOG: > > > 2012-05-07 18:36:01.596130 [CONSOLE] switch_core.c:1902 > FreeSWITCH Version 1.1.beta1 (git-982cb1f 2012-04-04 09-04-12 +0200) > Started. > Max Sessions[1000] > Session Rate[30] > SQL [Enabled] > 2012-05-07 18:36:04.974462 [NOTICE] switch_channel.c:926 New Channel > sofia/internal/1001 at 216.70.70.79 > [39e376c5-3a70-4355-825c-da97aca3a395] > 2012-05-07 18:36:04.974462 [INFO] mod_dialplan_xml.c:485 Processing > PSS 1001 <1001>->1005 in context outbound > 2012-05-07 18:36:04.974462 [INFO] switch_core_state_machine.c:177 No > Route, Aborting > 2012-05-07 18:36:04.974462 [NOTICE] switch_core_state_machine.c:178 > Hangup sofia/internal/1001 at 216.70.70.79 [CS_ROUTING] > [NO_ROUTE_DESTINATION] > 2012-05-07 18:36:04.974462 [NOTICE] switch_core_session.c:1400 Session > 1 (sofia/internal/1001 at 216.70.70.79) Ended > 2012-05-07 18:36:04.974462 [NOTICE] switch_core_session.c:1402 Close > Channel sofia/internal/1001 at 216.70.70.79 [CS_DESTROY] > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120507/f31057eb/attachment.html From saami_mh at ymail.com Tue May 8 08:07:12 2012 From: saami_mh at ymail.com (Samira Mh) Date: Mon, 7 May 2012 21:07:12 -0700 (PDT) Subject: [Freeswitch-users] error/NO_ROUTE_DESTINATION while define groupserror/NO_ROUTE_DESTINATION In-Reply-To: References: <1336392911.49162.YahooMailNeo@web120104.mail.ne1.yahoo.com> <1336448646.54490.YahooMailNeo@web120101.mail.ne1.yahoo.com> Message-ID: <1336450032.30944.YahooMailNeo@web120106.mail.ne1.yahoo.com> Hello, " that is my first email was send for you:: /usr/src/freeswitch-1.0.6/conf/directory/default.xml ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ??? ? ? ? then issue "reloadxml" ?and?run the comand:?group_call ?custom but i come accross with the following error: error/NO_ROUTE_DESTINATION ------------------------------------------------------------------------------------------------- as you see i ?issue the?reloadxml?i have defined the seetings in the ?path:?? ?/usr/src/freeswitch-1.0.6/conf/directory/default.xml ? ?instead of the path :?/usr/local/freeswitch/conf/directory/default.xml? now seeting are located in?/usr/local/freeswitch/conf/directory/default.xml ?and all of things worked correctly ;; thanks alot for your quick answer ,thanks soooo much , that was great help, it is kind of you to help me; ________________________________ From: Michael Collins To: FreeSWITCH Users Help Sent: Tuesday, May 8, 2012 8:25 AM Subject: Re: [Freeswitch-users] error/NO_ROUTE_DESTINATION while define groupserror/NO_ROUTE_DESTINATION Samira, When I say that I pressed F6 I mean that I went to fs_cli and pressed F6. You can also do "reloadxml". If you don't reload your xml after making a change then FreeSWITCH can't apply those changes. I copied your xml right out of the pastebin and put it into conf/directory/default.xml on my system and it worked just fine. I think that perhaps you need to issue the "reloadxml" command or that you maybe have edited the wrong file. Please confirm that you have edited /usr/local/freeswitch/conf/directory/default.xml and not some other file, perhaps the default.xml in the source tree. -MC On Mon, May 7, 2012 at 8:44 PM, Samira Mh wrote: > >hello, >thanks alot for your help >i put the entire of the default.xml on?http://pastebin.freeswitch.org/19008?, i am sure all of the settings are correctly.. >sorry but i don't understand how to?copied and pasted my stuff into your?default.xml file? >and hit F6 ? how can i do that? >________________________________ > From: Michael Collins >To: FreeSWITCH Users Help >Sent: Tuesday, May 8, 2012 12:41 AM >Subject: Re: [Freeswitch-users] error/NO_ROUTE_DESTINATION while define groupserror/NO_ROUTE_DESTINATION > > > >Most likely the configuration did not make it into the right part of the XML tree. Can you put your entire default.xml file on pastebin.freeswitch.org? We should be able to help you figure it out. FYI, copied and pasted your stuff right into my default.xml file and it worked like a champ. All I did was paste it in, save the file, and hit F6 at fs_cli and it worked just fine: > >freeswitch at default> group_call custom >[sip_invite_domain=192.168.1.79,presence_id=1005 at 192.168.1.79]error/user_not_registered,[sip_invite_domain=192.168.1.79,presence_id=1006 at 192.168.1.79]error/user_not_registered >freeswitch at default> > >-MC > > >On Mon, May 7, 2012 at 5:15 AM, Samira Mh wrote: > >Hello, >>i have defined the group named: "custom" in the below path as follow : >> >> >>/usr/src/freeswitch-1.0.6/conf/directory/default.xml >> >> >> >> >> >> >> >>? >>? ? ? ? >>? ? ? ? ? >>? ? ? ? ? >>? ? ??? >>? ? ? >> >> >>then issue "reloadxml" ?and?run the comand:?group_call ?custom >>but i come accross with the following error: >>error/NO_ROUTE_DESTINATION >> >> >>please help,all configs are defined correctly , but i couldn't define the "custom" group on freeswitch >>_________________________________________________________________________ >>Professional FreeSWITCH Consulting Services: >>consulting at freeswitch.org >>http://www.freeswitchsolutions.com >> >> >> >> >>Official FreeSWITCH Sites >>http://www.freeswitch.org >>http://wiki.freeswitch.org >>http://www.cluecon.com >> >>FreeSWITCH-users mailing list >>FreeSWITCH-users at 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 at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >FreeSWITCH-users mailing list >FreeSWITCH-users at 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 at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >FreeSWITCH-users mailing list >FreeSWITCH-users at 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 at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120507/993ba4bc/attachment-0001.html From bdfoster at endigotech.com Tue May 8 08:48:22 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Tue, 8 May 2012 00:48:22 -0400 Subject: [Freeswitch-users] error/NO_ROUTE_DESTINATION while define groupserror/NO_ROUTE_DESTINATION In-Reply-To: <1336450032.30944.YahooMailNeo@web120106.mail.ne1.yahoo.com> References: <1336392911.49162.YahooMailNeo@web120104.mail.ne1.yahoo.com> <1336448646.54490.YahooMailNeo@web120101.mail.ne1.yahoo.com> <1336450032.30944.YahooMailNeo@web120106.mail.ne1.yahoo.com> Message-ID: This may not have anything to do with your problem, but just to let you know git head is considered stable and is recommended. We've been running it here with no major issues and find it more reliable than 1.0.6. Just my two cents. As far as I know 1.0.6 isn't really updated. Join the rest of us, you'll be surprized. -BDF On May 8, 2012 12:08 AM, "Samira Mh" wrote: > > > Hello, > " that is my first email was send for you:: > > /usr/src/freeswitch-1.0.6/conf/directory/default.xml > > > > > > > > > > > then issue "reloadxml" and run the comand: group_call custom > but i come accross with the following error: > error/NO_ROUTE_DESTINATION > > > > ------------------------------------------------------------------------------------------------- > as you see i issue the reloadxml i have defined the seetings in the > path: /usr/src/freeswitch-1.0.6/conf/directory/default.xml instead > of the path : /usr/local/freeswitch/conf/directory/default.xml > now seeting are located > in /usr/local/freeswitch/conf/directory/default.xml and all of things > worked correctly ;; > thanks alot for your quick answer ,thanks soooo much , that was great help, > it is kind of you to help me; > > > > > > > > ------------------------------ > *From:* Michael Collins > *To:* FreeSWITCH Users Help > *Sent:* Tuesday, May 8, 2012 8:25 AM > *Subject:* Re: [Freeswitch-users] error/NO_ROUTE_DESTINATION while define > groupserror/NO_ROUTE_DESTINATION > > Samira, > > When I say that I pressed F6 I mean that I went to fs_cli and pressed F6. > You can also do "reloadxml". If you don't reload your xml after making a > change then FreeSWITCH can't apply those changes. > > I copied your xml right out of the pastebin and put it into > conf/directory/default.xml on my system and it worked just fine. I think > that perhaps you need to issue the "reloadxml" command or that you maybe > have edited the wrong file. Please confirm that you have edited > /usr/local/freeswitch/conf/directory/default.xml and not some other file, > perhaps the default.xml in the source tree. > > -MC > > On Mon, May 7, 2012 at 8:44 PM, Samira Mh wrote: > > > hello, > thanks alot for your help > i put the entire of the default.xml on > http://pastebin.freeswitch.org/19008 , i am sure all of the settings are > correctly.. > sorry but i don't understand how to copied and pasted my stuff into your default.xml > file > and hit F6 ? how can i do that? > ------------------------------ > *From:* Michael Collins > *To:* FreeSWITCH Users Help > *Sent:* Tuesday, May 8, 2012 12:41 AM > *Subject:* Re: [Freeswitch-users] error/NO_ROUTE_DESTINATION while define > groupserror/NO_ROUTE_DESTINATION > > Most likely the configuration did not make it into the right part of the > XML tree. Can you put your entire default.xml file on > pastebin.freeswitch.org? We should be able to help you figure it out. > FYI, copied and pasted your stuff right into my default.xml file and it > worked like a champ. All I did was paste it in, save the file, and hit F6 > at fs_cli and it worked just fine: > > freeswitch at default> group_call custom > [sip_invite_domain=192.168.1.79,presence_id=1005 at 192.168.1.79 > ]error/user_not_registered,[sip_invite_domain=192.168.1.79,presence_id= > 1006 at 192.168.1.79]error/user_not_registered > freeswitch at default> > > -MC > > On Mon, May 7, 2012 at 5:15 AM, Samira Mh wrote: > > Hello, > i have defined the group named: "custom" in the below path as follow : > > /usr/src/freeswitch-1.0.6/conf/directory/default.xml > > > > > > > > > > > then issue "reloadxml" and run the comand: group_call custom > but i come accross with the following error: > error/NO_ROUTE_DESTINATION > > please help,all configs are defined correctly , but i couldn't define the > "custom" group on freeswitch > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120508/993cf185/attachment.html From krice at freeswitch.org Tue May 8 11:38:48 2012 From: krice at freeswitch.org (Ken Rice) Date: Tue, 08 May 2012 02:38:48 -0500 Subject: [Freeswitch-users] Bypass media succeeds from extension to gateway but fails from extension to extension In-Reply-To: Message-ID: This is not a bug in FreeSWITCH... Bypass Media is a special mode to allow us to act as a psuedo proxy. Other then possibly filtering some codecs the rest of the SDPs are simply copied across the bridge. That means if your sip devices are sending RFC1918 IPs in their SDPs then FreeSWITCH will forward an RFC1918 SDP. Why is that? Because FreeSWITCH has no way to know where the media is actually coming from since FreeSWITCH is not in the media path. K On 5/7/12 8:34 PM, "Gabriel Gunderson" wrote: > On Mon, May 7, 2012 at 9:54 AM, Phil Quesinberry > wrote: >> I?m trying to get bypass media to work with extension to extension calls. >> Both endpoint extensions are behind NAT in two different locations.? This >> looks like a possible bug because FS sends the internal IP address of one of >> the endpoints to the other for media, but when making a call from the same >> extension to an external gateway for PSTN termination, it sends the phone?s >> external IP address as it should, and the call succeeds. > > My guess is that this is not a bug. The NAT stuff has been gone over > and over and used in many different configurations. Usually, in my > experience, NAT issues are cause by config errors or a broken device. > > What do you have for your sofia configs? > > Gabe > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From nasida at live.ru Tue May 8 13:32:03 2012 From: nasida at live.ru (Yuriy Nasida) Date: Tue, 8 May 2012 13:32:03 +0400 Subject: [Freeswitch-users] problems with adding one more file with licensed g7921 Message-ID: Hello guys. I bought additional licensed g7921 ports and have problems with adding them. I follow the instruction http://files-sync.freeswitch.org/g729/INSTALL.txtSo i have 2 conf file in /etc/freeswitch/. It is old (20 ports) and new additional conf file (10 ports). After restarting of FS i still have 20 ports only (not 30). FS works as root. Probably I have to restart licensed as well ? it will lead to restarting of FS ? Please advise.Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120508/e36f4e73/attachment.html From nasida at live.ru Tue May 8 13:45:41 2012 From: nasida at live.ru (Yuriy Nasida) Date: Tue, 8 May 2012 13:45:41 +0400 Subject: [Freeswitch-users] problems with adding one more file with licensed g7921 In-Reply-To: References: Message-ID: I meaned g729. It was just typing error in subject. From: nasida at live.ru To: freeswitch-users at lists.freeswitch.org Date: Tue, 8 May 2012 13:32:03 +0400 Subject: [Freeswitch-users] problems with adding one more file with licensed g7921 Hello guys. I bought additional licensed g7921 ports and have problems with adding them. I follow the instruction http://files-sync.freeswitch.org/g729/INSTALL.txtSo i have 2 conf file in /etc/freeswitch/. It is old (20 ports) and new additional conf file (10 ports). After restarting of FS i still have 20 ports only (not 30). FS works as root. Probably I have to restart licensed as well ? it will lead to restarting of FS ? Please advise.Thanks. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120508/d6e95535/attachment.html From B.Tietz at pinguin.ag Tue May 8 13:47:16 2012 From: B.Tietz at pinguin.ag (B.Tietz at pinguin.ag) Date: Tue, 8 May 2012 11:47:16 +0200 Subject: [Freeswitch-users] problems with adding one more file with licensed g7921 In-Reply-To: References: Message-ID: <07BF4904977CC645B485E970424193AD10E9088E9D@localhost> Hi, you have to restart the freeswitch_licence_server too regards Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Yuriy Nasida Gesendet: Dienstag, 8. Mai 2012 11:32 An: freeswitch-users at lists.freeswitch.org Betreff: [Freeswitch-users] problems with adding one more file with licensed g7921 Hello guys. I bought additional licensed g7921 ports and have problems with adding them. I follow the instruction http://files-sync.freeswitch.org/g729/INSTALL.txt So i have 2 conf file in /etc/freeswitch/. It is old (20 ports) and new additional conf file (10 ports). After restarting of FS i still have 20 ports only (not 30). FS works as root. Probably I have to restart licensed as well ? it will lead to restarting of FS ? Please advise. Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120508/39de5c3e/attachment.html From saami_mh at ymail.com Tue May 8 13:55:54 2012 From: saami_mh at ymail.com (Samira Mh) Date: Tue, 8 May 2012 02:55:54 -0700 (PDT) Subject: [Freeswitch-users] error/NO_ROUTE_DESTINATION while define groupserror/NO_ROUTE_DESTINATION In-Reply-To: References: <1336392911.49162.YahooMailNeo@web120104.mail.ne1.yahoo.com> <1336448646.54490.YahooMailNeo@web120101.mail.ne1.yahoo.com> <1336450032.30944.YahooMailNeo@web120106.mail.ne1.yahoo.com> Message-ID: <1336470954.60527.YahooMailNeo@web120106.mail.ne1.yahoo.com> my problem was solved by changing the path to?/usr/local/freeswitch/conf/directory/default.xml instead of?/usr/src/freeswitch-1.0.6/conf/directory/default.xml ________________________________ From: Brian Foster To: FreeSWITCH Users Help Sent: Tuesday, May 8, 2012 9:18 AM Subject: Re: [Freeswitch-users] error/NO_ROUTE_DESTINATION while define groupserror/NO_ROUTE_DESTINATION This may not have anything to do with your problem, but just to let you know git head is considered stable and is recommended. We've been running it here with no major issues and find it more reliable than 1.0.6. Just my two cents. As far as I know 1.0.6 isn't really updated. Join the rest of us, you'll be surprized. -BDF On May 8, 2012 12:08 AM, "Samira Mh" wrote: > > > >Hello, >" that is my first email was send for you:: > > >/usr/src/freeswitch-1.0.6/conf/directory/default.xml > > > > > > > >? >? ? ? ? >? ? ? ? ? >? ? ? ? ? >? ? ??? >? ? ? > > >then issue "reloadxml" ?and?run the comand:?group_call ?custom >but i come accross with the following error: >error/NO_ROUTE_DESTINATION > > > > >------------------------------------------------------------------------------------------------- >as you see i ?issue the?reloadxml?i have defined the seetings in the ?path:?? ?/usr/src/freeswitch-1.0.6/conf/directory/default.xml ? ?instead of the path :?/usr/local/freeswitch/conf/directory/default.xml? >now seeting are located in?/usr/local/freeswitch/conf/directory/default.xml ?and all of things worked correctly ;; >thanks alot for your quick answer ,thanks soooo much , that was great help, >it is kind of you to help me; > > > > > > > > > > > > > > > >________________________________ > From: Michael Collins >To: FreeSWITCH Users Help >Sent: Tuesday, May 8, 2012 8:25 AM >Subject: Re: [Freeswitch-users] error/NO_ROUTE_DESTINATION while define groupserror/NO_ROUTE_DESTINATION > > >Samira, > >When I say that I pressed F6 I mean that I went to fs_cli and pressed F6. You can also do "reloadxml". If you don't reload your xml after making a change then FreeSWITCH can't apply those changes. > >I copied your xml right out of the pastebin and put it into conf/directory/default.xml on my system and it worked just fine. I think that perhaps you need to issue the "reloadxml" command or that you maybe have edited the wrong file. Please confirm that you have edited /usr/local/freeswitch/conf/directory/default.xml and not some other file, perhaps the default.xml in the source tree. > >-MC > > >On Mon, May 7, 2012 at 8:44 PM, Samira Mh wrote: > > >> >>hello, >>thanks alot for your help >>i put the entire of the default.xml on?http://pastebin.freeswitch.org/19008?, i am sure all of the settings are correctly.. >>sorry but i don't understand how to?copied and pasted my stuff into your?default.xml file? >>and hit F6 ? how can i do that? >>________________________________ >> From: Michael Collins >>To: FreeSWITCH Users Help >>Sent: Tuesday, May 8, 2012 12:41 AM >>Subject: Re: [Freeswitch-users] error/NO_ROUTE_DESTINATION while define groupserror/NO_ROUTE_DESTINATION >> >> >> >>Most likely the configuration did not make it into the right part of the XML tree. Can you put your entire default.xml file on pastebin.freeswitch.org? We should be able to help you figure it out. FYI, copied and pasted your stuff right into my default.xml file and it worked like a champ. All I did was paste it in, save the file, and hit F6 at fs_cli and it worked just fine: >> >>freeswitch at default> group_call custom >>[sip_invite_domain=192.168.1.79,presence_id=1005 at 192.168.1.79]error/user_not_registered,[sip_invite_domain=192.168.1.79,presence_id=1006 at 192.168.1.79]error/user_not_registered >>freeswitch at default> >> >>-MC >> >> >>On Mon, May 7, 2012 at 5:15 AM, Samira Mh wrote: >> >>Hello, >>>i have defined the group named: "custom" in the below path as follow : >>> >>> >>>/usr/src/freeswitch-1.0.6/conf/directory/default.xml >>> >>> >>> >>> >>> >>> >>> >>>? >>>? ? ? ? >>>? ? ? ? ? >>>? ? ? ? ? >>>? ? ??? >>>? ? ? >>> >>> >>>then issue "reloadxml" ?and?run the comand:?group_call ?custom >>>but i come accross with the following error: >>>error/NO_ROUTE_DESTINATION >>> >>> >>>please help,all configs are defined correctly , but i couldn't define the "custom" group on freeswitch >>>_________________________________________________________________________ >>>Professional FreeSWITCH Consulting Services: >>>consulting at freeswitch.org >>>http://www.freeswitchsolutions.com >>> >>> >>> >>> >>>Official FreeSWITCH Sites >>>http://www.freeswitch.org >>>http://wiki.freeswitch.org >>>http://www.cluecon.com >>> >>>FreeSWITCH-users mailing list >>>FreeSWITCH-users at 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 at freeswitch.org >>http://www.freeswitchsolutions.com >> >> >> >> >>Official FreeSWITCH Sites >>http://www.freeswitch.org >>http://wiki.freeswitch.org >>http://www.cluecon.com >> >>FreeSWITCH-users mailing list >>FreeSWITCH-users at 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 at freeswitch.org >>http://www.freeswitchsolutions.com >> >> >> >> >>Official FreeSWITCH Sites >>http://www.freeswitch.org >>http://wiki.freeswitch.org >>http://www.cluecon.com >> >>FreeSWITCH-users mailing list >>FreeSWITCH-users at 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 at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >FreeSWITCH-users mailing list >FreeSWITCH-users at 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 at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >FreeSWITCH-users mailing list >FreeSWITCH-users at 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 at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120508/2221cab8/attachment-0001.html From nasida at live.ru Tue May 8 14:04:14 2012 From: nasida at live.ru (Yuriy Nasida) Date: Tue, 8 May 2012 14:04:14 +0400 Subject: [Freeswitch-users] problems with adding one more file with licensed g7921 In-Reply-To: <07BF4904977CC645B485E970424193AD10E9088E9D@localhost> References: , <07BF4904977CC645B485E970424193AD10E9088E9D@localhost> Message-ID: Done. Now it works. Thanks. From: B.Tietz at pinguin.ag To: freeswitch-users at lists.freeswitch.org Date: Tue, 8 May 2012 11:47:16 +0200 Subject: Re: [Freeswitch-users] problems with adding one more file with licensed g7921 Hi, you have to restart the freeswitch_licence_server too regards Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Yuriy Nasida Gesendet: Dienstag, 8. Mai 2012 11:32 An: freeswitch-users at lists.freeswitch.org Betreff: [Freeswitch-users] problems with adding one more file with licensed g7921 Hello guys. I bought additional licensed g7921 ports and have problems with adding them. I follow the instruction http://files-sync.freeswitch.org/g729/INSTALL.txtSo i have 2 conf file in /etc/freeswitch/. It is old (20 ports) and new additional conf file (10 ports). After restarting of FS i still have 20 ports only (not 30). FS works as root. Probably I have to restart licensed as well ? it will lead to restarting of FS ? Please advise.Thanks. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120508/a4aed015/attachment.html From anita.hall at simmortel.com Tue May 8 15:45:33 2012 From: anita.hall at simmortel.com (Anita Hall) Date: Tue, 8 May 2012 17:15:33 +0530 Subject: [Freeswitch-users] Cookbook: bridge multiple endpoints failover Message-ID: Hi I am following FreeSWITCH cookbook Chapter 1, Page 15: Ringing multiple endpoints sequentially. I am also using mod_lua to do a simple app that does the bridging. The problem scenario is as follows: The first endpoint rings. It is not answered. The second endpoint starts ringing. It is answered but the call is immediately disconnected. And then the first endpoint starts ringing again! Probably, the reason why the first endpoint starts ringing again is a repeat try in bridge application. Is there anyway to disable it ? Here is my dialplan snippet And here is bridge.lua # cat scripts/bridge.lua -- arguments from dialplan setAnswer = argv[1] num1 = argv[2] limiter1 = argv[3] num2 = argv[4] session:answer() welcome= "welcome.wav" session:streamFile(welcome) freeswitch.consoleLog("INFO","Prompt file is '" .. welcome .. "'\n") freeswitch.consoleLog("INFO","Arguments '" .. argv[1] .. " " .. argv[2] .. " " .. argv[3] .. " " .. argv[4] .. "'\n") globalChanVars = "{ignore_early_media=true,originate_continue_on_timeout=true,call_timout=60,monitor_early_media_fail=user_busy:2:480+620!destination_out_of_order:2:1776.7}" chan1 = "[leg_timeout=15]" .. "freetdm/1/a/" .. num1 chan2 = "[leg_timeout=55]" .. "freetdm/1/a/" .. num2 -- global channel variables are applicable to all channels in the session newSession = freeswitch.Session(globalChanVars .. chan1 .. limiter1 .. chan2) -- bridge new Session to the current session freeswitch.bridge(session, newSession) Much thanks! regards, Anita -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120508/209540b6/attachment.html From anita.hall at simmortel.com Tue May 8 15:51:58 2012 From: anita.hall at simmortel.com (Anita Hall) Date: Tue, 8 May 2012 17:21:58 +0530 Subject: [Freeswitch-users] MOH before bridge not working Message-ID: Hi I am using mod_lua to do the bridge but MOH is not working. I have tried both hold_music and temp_hold_music. Dialplan is When I do the bridge inside the lua script, there is no MOH being played. Some thing else is needed to play MOH ? regards, Anita -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120508/8ab4ab22/attachment.html From vipkilla at gmail.com Tue May 8 15:56:51 2012 From: vipkilla at gmail.com (Vik Killa) Date: Tue, 8 May 2012 07:56:51 -0400 Subject: [Freeswitch-users] FreeSWITCH not sending P-Asserted-Identity In-Reply-To: References: <-2643950612768503375@unknownmsgid> Message-ID: > So what are you thinking at this point? Need to update the wiki or open a Jira? I could open a ticket with Jira, I just assumed that I was doing something wrong but I guess if nobody can help me on here it could be a bug. From anita.hall at simmortel.com Tue May 8 16:35:12 2012 From: anita.hall at simmortel.com (Anita Hall) Date: Tue, 8 May 2012 18:05:12 +0530 Subject: [Freeswitch-users] Commercial Fax over T.30 for FreeSWITCH In-Reply-To: <4FA42498.9070504@coppice.org> References: <4FA24100.7040908@integrafin.co.uk> <4FA40131.8090009@coppice.org> <4FA41447.1000700@integrafin.co.uk> <4FA42498.9070504@coppice.org> Message-ID: Hi Steve I am sorry for the trouble. You are right, I should stay focused on FS + spandsp. I double checked, the clock is taken from the E1. I have in wanpipe1.conf TE_CLOCK = NORMAL TE_REF_CLOCK = 0 Thanks. regards, Anita On Sat, May 5, 2012 at 12:18 AM, Steve Underwood wrote: > On 05/05/2012 01:39 AM, Alex Crow wrote: > > On 04/05/12 17:17, Steve Underwood wrote: > >> On 05/03/2012 08:23 PM, Anita Hall wrote: > >>> Hi Alex > >>> > >>> Thanks for the tip :) > >>> > >>> No, I am running FreeSWITCH on an Ubuntu 64-bit Quad core machine. > >>> There is no VM. > >>> > >>> Is it possible to use Hylafax instead of or in combination to > >>> FreeSWITCH if I am using Sangoma E1 Cards ? > >>> > >>> What all things should I check to ensure that the timing on my system > >>> is good? Like, may be, my motherboard is bad or my kernel could be > >>> tuned better ? > >> Are you really interested in fixing your problems? It seems you have > >> greater interest in tinkering with every possible solution. Many many > >> people handle hundreds of thousands of FAXes per day with low failures > >> rates using: > >> asterisk + iaxmodem/spandsp + hylafax, or > >> asterisk + spandsp, or > >> freeswitch + spandsp > >> and some people are probably now carrying significant traffic with > >> freeswitch + spandsp + hylafax, which is a recent addition to the > >> options. They all work well when set up properly. > >> > >> Steve > >> > >> > > Steve, > > > > Forgive me, but to be honest I think that opening sentence is a bit > > harsh. Timing is important for fax, and I think the OP just wants to > > eliminate all the possible causes, which is commendable. I have had > > problems in the past with bad timing in the kernel on Ubuntu (although > > it may not apply here, I was on BRI) and it was a really difficult time > > with lots of complaints from users until I pinned it down. > > > > I think we just need to point out the great capabilities of the Free/OSS > > software that is available. I have been stunned with the quality and > > stability of both FreeSwitch and HylaFAX, and having used the latter for > > more than a decade in a commercial environment cannot fault it. It also > > has great Windows clients such as WHFC, great reporting of fax outcomes > > built-in, and just seems to do its job without ever needing attention > > (We had one box with 5+ years of uptime). Moving to to a new > > softswitch/PBX can be very daunting for a new user, especially one that > > comes from the Asterisk environment (lots of baggage) or one not > > accustomed to telephony in general. > > > Have you seen what and where she has been posting? She's taking a > scattergun approach, and its getting kind of annoying. She seems to > assume every piece of software around is thoroughly broken, and the fact > people are using it successfully in high volume applications is an > illusion. Nice people get a lot of help from me, if I'm not too busy. > She won't. > > Steve > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120508/981045ad/attachment-0001.html From miha at softnet.si Tue May 8 16:51:38 2012 From: miha at softnet.si (Miha) Date: Tue, 08 May 2012 14:51:38 +0200 Subject: [Freeswitch-users] Call intercept Message-ID: <4FA916DA.4020201@softnet.si> Hi, I need a little help about call intercept. I have read wiki (http://wiki.freeswitch.org/wiki/Callgroup_intercept) but still having few problems. 1. I have Opensips which works like load_balancer and registrar. Behind Opensips I have two FS server, which are for load_balacing. 2. I need to set a call intercept thing. I done it like it is written on FS wiki but this does not work as I have Opensips. I have put this in public dialplan: After I try to intercept the call this fails as in database is written different UUID. CAll goes like this: Phone-openips-FS-Openips-Phone Thank you for all your help! MIha -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120508/92534f7c/attachment.html From thaddeus at thogan.com Tue May 8 19:11:45 2012 From: thaddeus at thogan.com (thaddeus at thogan.com) Date: Tue, 08 May 2012 10:11:45 -0500 Subject: [Freeswitch-users] Dialplan condition to test if SIP client is registered In-Reply-To: Message-ID: <2d5-4fa93780-5-a26ad20@121818419> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120508/c1f28687/attachment.html From anthony.minessale at gmail.com Tue May 8 19:18:12 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 8 May 2012 10:18:12 -0500 Subject: [Freeswitch-users] FreeSWITCH not sending P-Asserted-Identity In-Reply-To: References: <-2643950612768503375@unknownmsgid> Message-ID: you need the screen param to trigger asserted, its on by default but as soon as you specify the privacy from the var it's up to you to put it back. On Tue, May 8, 2012 at 6:56 AM, Vik Killa wrote: >> So what are you thinking at this point? Need to update the wiki or open a Jira? > > I could open a ticket with Jira, I just assumed that I was doing > something wrong but I guess if nobody can help me on here it could be > a bug. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From shawn.yu at janmarini.com Tue May 8 21:33:57 2012 From: shawn.yu at janmarini.com (Shawn Yu) Date: Tue, 8 May 2012 10:33:57 -0700 Subject: [Freeswitch-users] Freeswitch + Lync Message-ID: We have Lync in our environment, and have successfully configured the following to work with Freeswitch: - Outbound call from Lync via Freeswitch - Inbound call rings both Lync and Freeswitch endpoint (Polycom phone) - Calls between Lync and Polycom phone We would also like to have Lync to remote call control the Polycom phone. Is this possible? Shawn -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120508/2ccf6c35/attachment.html From bdfoster at endigotech.com Tue May 8 21:45:46 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Tue, 8 May 2012 13:45:46 -0400 Subject: [Freeswitch-users] Freeswitch + Lync In-Reply-To: References: Message-ID: That's more than likely a Lync-specific question, and I'm not sure how many people use Lync on this mailing list. Hang tight though, someone may know. I'd be looking other places though. -BDF On Tue, May 8, 2012 at 1:33 PM, Shawn Yu wrote: > We have Lync in our environment, and have successfully configured the > following to work with Freeswitch:**** > > **- **Outbound call from Lync via Freeswitch**** > > **- **Inbound call rings both Lync and Freeswitch endpoint > (Polycom phone)**** > > **- **Calls between Lync and Polycom phone**** > > ** ** > > We would also like to have Lync to remote call control the Polycom phone. > Is this possible?**** > > ** ** > > ** ** > > Shawn**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-800-7876 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120508/694b76ae/attachment-0001.html From vipkilla at gmail.com Tue May 8 21:53:41 2012 From: vipkilla at gmail.com (Vik Killa) Date: Tue, 8 May 2012 13:53:41 -0400 Subject: [Freeswitch-users] FreeSWITCH not sending P-Asserted-Identity In-Reply-To: References: <-2643950612768503375@unknownmsgid> Message-ID: Ok thanks. the wiki is misleading, I'll update it. From jerry.richards at teotech.com Tue May 8 22:37:59 2012 From: jerry.richards at teotech.com (Jerry Richards) Date: Tue, 8 May 2012 18:37:59 +0000 Subject: [Freeswitch-users] Freeswitch And 2 Network I/Fs In-Reply-To: <07BF4904977CC645B485E970424193AD10E9088BA2@localhost> References: <1545146083A72C4DB7B66584B7E5D98402BBC647@BY2PRD0410MB377.namprd04.prod.outlook.com> <07BF4904977CC645B485E970424193AD10E9088BA2@localhost> Message-ID: <1545146083A72C4DB7B66584B7E5D98402BC0B3E@BY2PRD0410MB377.namprd04.prod.outlook.com> Hi, Okay, I tried setting the tags you suggest (rtp-ip, sip-ip, presence-hosts, ext-rtp-ip, ext-sip-ip). I am able to register, but I can't call from one network to the other (it just goes to voicemail). I posted a trace at http://pastebin.freeswitch.org/19016 Thanks, Jerry From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of B.Tietz at pinguin.ag Sent: Monday, May 07, 2012 7:51 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Freeswitch And 2 Network I/Fs Hi, from ODBC in the core in the wiki we have: setup the IP in the following params in each sip-profile: rtp-ip, sip-ip, presence-hosts, ext-rtp-ip, ext-sip-ip regards, Benjamin Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Jerry Richards Gesendet: Montag, 7. Mai 2012 16:37 An: 'freeswitch-users at lists.freeswitch.org' Betreff: [Freeswitch-users] Freeswitch And 2 Network I/Fs Is there an example of configuring Freeswitch to support both eth0 and eth1, when they are connected to different networks? I know I can create two sip_profiles, which works fine for registration, but in the past, I've had issues with Freeswitch not sending media to the right interface (or at least, not knowing how to control that). Thanks, Jerry -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120508/df6f327d/attachment.html From koralu at gmail.com Tue May 8 22:48:04 2012 From: koralu at gmail.com (Adrian Andrei) Date: Tue, 8 May 2012 21:48:04 +0300 Subject: [Freeswitch-users] mod_xml_cdr is not logging Message-ID: Hello I tried to set up mod_xml_cdr but it doesn't work at all. I made the following steps: - Uncomment mod_xml_cdr in modules.conf. - Ok - Edit conf/autoload_configs/modules.conf.xml. - OK - Load mod_xml_cdr from CLI - NO errors - My *xml_cdr.conf.xml* looks like: (The php files are from contrib/trixter/xml-cdr.) In some posts I read that if log-dir and err-log-dir are changed from "default" I should set also the log-http-and-disk. But it doesn't work. Wiki said that I should check the freeswitch.xml to see if it is included xml_cdr.conf.xml but I can't find any line with xml_cdr.conf.xml. The version of freeswitch is FreeSWITCH Version 1.0.head (git-2c52f23 2012-02-18 08-37-47 -0600) so I think is new enough to have this feature available. Could anyone help me to find out what I'm missing? Are any commands in FS in order to run a mod_cdr_xml debug? Ty in advance. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120508/6dfccbeb/attachment.html From rmorin at blie-ent.com Tue May 8 22:48:14 2012 From: rmorin at blie-ent.com (Rob Morin) Date: Tue, 8 May 2012 14:48:14 -0400 Subject: [Freeswitch-users] make current fails Message-ID: <02aa01cd2d4b$205b8bc0$6112a340$@blie-ent.com> I'm running on CentOS6, and did my initial installation with ./configure -without-libcurl as recommended in Jira FS-3384 and others. Now I'm not able to do a 'make current' - it fails on some libraries that don't have symbols. Is there a better way to do this, other than to manually update (git ., ./bootstrap.sh, ./configure -without-libcurl, make, make install) every time? Thank you, Rob -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120508/9c0e171b/attachment.html From krice at freeswitch.org Tue May 8 22:53:16 2012 From: krice at freeswitch.org (Ken Rice) Date: Tue, 08 May 2012 13:53:16 -0500 Subject: [Freeswitch-users] make current fails In-Reply-To: <02aa01cd2d4b$205b8bc0$6112a340$@blie-ent.com> Message-ID: You might need to rebootstrap... We are currently changing some things around getting ready to bump the versions an some changes were required to Makefile.am this morning.... This could cause the errors you are seeing or if you pulled at the wrong time it would have actually caused the build to just hang K On 5/8/12 1:48 PM, "Rob Morin" wrote: > I?m running on CentOS6, and did my initial installation with > ./configure ?without-libcurl > as recommended in Jira FS-3384 and others. Now I?m not able to do a ?make > current? ? it fails on some libraries that don?t have symbols. > > Is there a better way to do this, other than to manually update (git ?, > ./bootstrap.sh, ./configure ?without-libcurl, make, make install) every time? > > Thank you, > Rob > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120508/c5cf9d71/attachment-0001.html From nickolayr at gmail.com Tue May 8 22:59:58 2012 From: nickolayr at gmail.com (Nikolay Rogoshchenkov) Date: Tue, 8 May 2012 14:59:58 -0400 Subject: [Freeswitch-users] MOH before bridge not working In-Reply-To: References: Message-ID: Let do the bridge outside the lua, it is more right. -- Nikolay On Tue, May 8, 2012 at 7:51 AM, Anita Hall wrote: > Hi > > I am using mod_lua to do the bridge but MOH is not working. I have tried > both hold_music and temp_hold_music. > > Dialplan is > > data="hold_music=danza-espanola-op-37-h-142-xii-arabesca.wav" /> > data="temp_hold_music=danza-espanola-op-37-h-142-xii-arabesca.wav" /> > > > When I do the bridge inside the lua script, there is no MOH being played. > > Some thing else is needed to play MOH ? > > regards, > Anita > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120508/6450340b/attachment.html From krice at freeswitch.org Wed May 9 00:06:39 2012 From: krice at freeswitch.org (Ken Rice) Date: Tue, 08 May 2012 15:06:39 -0500 Subject: [Freeswitch-users] 1.2-rc2 Tagged and Tarball available on files.freeswitch.org Message-ID: Hey Guys we?ve reached another milestone in getting FreeSWITCH 1.2 out the door. http://files.freeswitch.org/freeswitch-1.2.rc2.tar.bz2 Yep that?s what you want to test... Look for debian packages and updated RPMs for these soon K -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120508/46f0844b/attachment.html From andrew at cassidywebservices.co.uk Wed May 9 01:12:13 2012 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Tue, 8 May 2012 22:12:13 +0100 Subject: [Freeswitch-users] Freeswitch + Lync In-Reply-To: References: Message-ID: It's all something I looked into once, but avaya charge ?lol for what I needed to do it at the time and that's all I had available. Remote Call Control USED to require TAPI drivers under OCS. I'm not sure if that's still true, but from memory it's done using SIP+CSTA which you may have to bridge to the event socket or process using python or something from the dialplan. With regards to interoperability, I've not tried it but see no reason for it to work. You may need a Lync Mediation server, but you might be able to get away with using the gateways options. It's SIP+TLS, but not sure what codecs are involved. The outbound call via lync should just be gateway configuration, which should also facilitiate ringing both a phone and lync using standard sofia dial strings. Don't take my word for any of this though, it's been a long time since I last looked. If they still do a free trial and I have some free time I might look into it again. On 8 May 2012 18:45, Brian Foster wrote: > That's more than likely a Lync-specific question, and I'm not sure how > many people use Lync on this mailing list. Hang tight though, someone may > know. I'd be looking other places though. > > -BDF > > On Tue, May 8, 2012 at 1:33 PM, Shawn Yu wrote: > >> We have Lync in our environment, and have successfully configured the >> following to work with Freeswitch:**** >> >> **- **Outbound call from Lync via Freeswitch**** >> >> **- **Inbound call rings both Lync and Freeswitch endpoint >> (Polycom phone)**** >> >> **- **Calls between Lync and Polycom phone**** >> >> ** ** >> >> We would also like to have Lync to remote call control the Polycom phone. >> Is this possible?**** >> >> ** ** >> >> ** ** >> >> Shawn**** >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Brian D. Foster > Endigo Computer LLC > Email: bdfoster at endigotech.com > Phone: 317-800-7876 > Indianapolis, Indiana, USA > > This message contains confidential information and is intended for those > listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If > you are not the intended recipient you are notified that disclosing, > copying, distributing or taking any action in reliance on the contents of > this information is strictly prohibited. E-mail transmission cannot be > guaranteed to be secure or error-free as information could be intercepted, > corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. > The sender therefore does not accept liability for any errors or omissions > in the contents of this message, which arise as a result of e-mail > transmission. If verification is required please request a hard-copy > version. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T *03300 100 960 *F *03300 100 961 *E *andrew at cassidywebservices.co.uk *W *www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120508/c2f51ea0/attachment.html From jerry.richards at teotech.com Wed May 9 01:34:31 2012 From: jerry.richards at teotech.com (Jerry Richards) Date: Tue, 8 May 2012 21:34:31 +0000 Subject: [Freeswitch-users] Freeswitch And 2 Network I/Fs In-Reply-To: <1545146083A72C4DB7B66584B7E5D98402BC0B3E@BY2PRD0410MB377.namprd04.prod.outlook.com> References: <1545146083A72C4DB7B66584B7E5D98402BBC647@BY2PRD0410MB377.namprd04.prod.outlook.com> <07BF4904977CC645B485E970424193AD10E9088BA2@localhost> <1545146083A72C4DB7B66584B7E5D98402BC0B3E@BY2PRD0410MB377.namprd04.prod.outlook.com> Message-ID: <1545146083A72C4DB7B66584B7E5D98402BC0BFF@BY2PRD0410MB377.namprd04.prod.outlook.com> Okay, I think I have it working. This is the bridge statement I'm using: If anyone recommends a better bridge, please let me know. Thanks, Jerry From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jerry Richards Sent: Tuesday, May 08, 2012 11:38 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Freeswitch And 2 Network I/Fs Hi, Okay, I tried setting the tags you suggest (rtp-ip, sip-ip, presence-hosts, ext-rtp-ip, ext-sip-ip). I am able to register, but I can't call from one network to the other (it just goes to voicemail). I posted a trace at http://pastebin.freeswitch.org/19016 Thanks, Jerry From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of B.Tietz at pinguin.ag Sent: Monday, May 07, 2012 7:51 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Freeswitch And 2 Network I/Fs Hi, from ODBC in the core in the wiki we have: setup the IP in the following params in each sip-profile: rtp-ip, sip-ip, presence-hosts, ext-rtp-ip, ext-sip-ip regards, Benjamin Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Jerry Richards Gesendet: Montag, 7. Mai 2012 16:37 An: 'freeswitch-users at lists.freeswitch.org' Betreff: [Freeswitch-users] Freeswitch And 2 Network I/Fs Is there an example of configuring Freeswitch to support both eth0 and eth1, when they are connected to different networks? I know I can create two sip_profiles, which works fine for registration, but in the past, I've had issues with Freeswitch not sending media to the right interface (or at least, not knowing how to control that). Thanks, Jerry -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120508/5ac9d6f4/attachment-0001.html From chris at opencsta.org Wed May 9 02:36:26 2012 From: chris at opencsta.org (Chris Mylonas) Date: Wed, 9 May 2012 08:36:26 +1000 Subject: [Freeswitch-users] Freeswitch + Lync In-Reply-To: References: Message-ID: <266513FC-67FF-4757-A45C-FBD3494C930E@opencsta.org> If anyone can forward me info on LYNC stuff I can take a look at it. The CSTA library on the opencsta website is very old and needs a clean up. It is functional and supports Siemens/Alcatel limited/Ericsson phone systems using ASN.1 notation rather than XML. Once XML is in place - then it's a matter of tying it to ESL and some other internals of FS I guess. It is written in java with an LGPL license - I'm new here, and I'm busy doing other stuff - but feel free to throw stuff at me to look at! Cheers Chris On 09/05/2012, at 7:12 AM, Andrew Cassidy wrote: > It's all something I looked into once, but avaya charge ?lol for what I needed to do it at the time and that's all I had available. Remote Call Control USED to require TAPI drivers under OCS. I'm not sure if that's still true, but from memory it's done using SIP+CSTA which you may have to bridge to the event socket or process using python or something from the dialplan. > > With regards to interoperability, I've not tried it but see no reason for it to work. You may need a Lync Mediation server, but you might be able to get away with using the gateways options. It's SIP+TLS, but not sure what codecs are involved. > > The outbound call via lync should just be gateway configuration, which should also facilitiate ringing both a phone and lync using standard sofia dial strings. > > Don't take my word for any of this though, it's been a long time since I last looked. > > If they still do a free trial and I have some free time I might look into it again. > > On 8 May 2012 18:45, Brian Foster wrote: > That's more than likely a Lync-specific question, and I'm not sure how many people use Lync on this mailing list. Hang tight though, someone may know. I'd be looking other places though. > > -BDF > > On Tue, May 8, 2012 at 1:33 PM, Shawn Yu wrote: > We have Lync in our environment, and have successfully configured the following to work with Freeswitch: > > - Outbound call from Lync via Freeswitch > > - Inbound call rings both Lync and Freeswitch endpoint (Polycom phone) > > - Calls between Lync and Polycom phone > > > > We would also like to have Lync to remote call control the Polycom phone. Is this possible? > > > > > > Shawn > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Brian D. Foster > Endigo Computer LLC > Email: bdfoster at endigotech.com > Phone: 317-800-7876 > Indianapolis, Indiana, USA > > This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Andrew Cassidy BSc (Hons) MBCS SSCA > Managing Director > > > T 03300 100 960 F 03300 100 961 > E andrew at cassidywebservices.co.uk > W www.cassidywebservices.co.uk > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120509/e0357dad/attachment.html From chris at opencsta.org Wed May 9 02:49:33 2012 From: chris at opencsta.org (Chris Mylonas) Date: Wed, 9 May 2012 08:49:33 +1000 Subject: [Freeswitch-users] 1.2-rc2 Tagged and Tarball available on files.freeswitch.org In-Reply-To: References: Message-ID: Congratulations!! I know it's been a long time coming - I've been lurking around the FS site/irc for months! (my FS plans keep getting interfered with though!) :) On 09/05/2012, at 6:06 AM, Ken Rice wrote: > Hey Guys we?ve reached another milestone in getting FreeSWITCH 1.2 out the door. > > http://files.freeswitch.org/freeswitch-1.2.rc2.tar.bz2 > > Yep that?s what you want to test... > > Look for debian packages and updated RPMs for these soon > > K > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120509/13373619/attachment.html From msc at freeswitch.org Wed May 9 03:43:11 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 8 May 2012 16:43:11 -0700 Subject: [Freeswitch-users] mod_xml_cdr is not logging In-Reply-To: References: Message-ID: Are you getting files on disk? -MC On Tue, May 8, 2012 at 11:48 AM, Adrian Andrei wrote: > Hello > > I tried to set up mod_xml_cdr but it doesn't work at all. I made the > following steps: > > - Uncomment mod_xml_cdr in modules.conf. - Ok > - Edit conf/autoload_configs/modules.conf.xml. - OK > - Load mod_xml_cdr from CLI - NO errors > - My *xml_cdr.conf.xml* looks like: > > > > > > > > > value="/usr/local/freeswitch/log/cdr/errors"/> > > > > (The php files are from contrib/trixter/xml-cdr.) > > In some posts I read that if log-dir and err-log-dir are changed from > "default" I should set also the log-http-and-disk. But it doesn't work. > > Wiki said that I should check the freeswitch.xml to see if it is included > xml_cdr.conf.xml but I can't find any line with xml_cdr.conf.xml. The > version of freeswitch is FreeSWITCH Version 1.0.head (git-2c52f23 > 2012-02-18 08-37-47 -0600) so I think is new enough to have this feature > available. > > Could anyone help me to find out what I'm missing? Are any commands in FS > in order to run a mod_cdr_xml debug? > > Ty in advance. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120508/7a162723/attachment-0001.html From msc at freeswitch.org Wed May 9 03:45:15 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 8 May 2012 16:45:15 -0700 Subject: [Freeswitch-users] Cookbook: bridge multiple endpoints failover In-Reply-To: References: Message-ID: This is a classic case of using the wrong tool for the job. Let the dialplan bridge app do the work for you. Only use Lua where you really need it. There is rarely a good reason to bridge from within a dialplan script. -MC On Tue, May 8, 2012 at 4:45 AM, Anita Hall wrote: > Hi > > I am following FreeSWITCH cookbook Chapter 1, Page 15: Ringing multiple > endpoints sequentially. > > I am also using mod_lua to do a simple app that does the bridging. > > The problem scenario is as follows: The first endpoint rings. It is not > answered. The second endpoint starts ringing. It is answered but the call > is immediately disconnected. And then the first endpoint starts ringing > again! > > Probably, the reason why the first endpoint starts ringing again is a > repeat try in bridge application. Is there anyway to disable it ? > > Here is my dialplan snippet > > > > And here is bridge.lua > > # cat scripts/bridge.lua > > -- arguments from dialplan > setAnswer = argv[1] > num1 = argv[2] > limiter1 = argv[3] > num2 = argv[4] > > > session:answer() > welcome= "welcome.wav" > session:streamFile(welcome) > freeswitch.consoleLog("INFO","Prompt file is '" .. welcome .. "'\n") > > freeswitch.consoleLog("INFO","Arguments '" .. argv[1] .. " " .. argv[2] .. > " " .. argv[3] .. " " .. argv[4] .. "'\n") > globalChanVars = > "{ignore_early_media=true,originate_continue_on_timeout=true,call_timout=60,monitor_early_media_fail=user_busy:2:480+620!destination_out_of_order:2:1776.7}" > > chan1 = "[leg_timeout=15]" .. "freetdm/1/a/" .. num1 > chan2 = "[leg_timeout=55]" .. "freetdm/1/a/" .. num2 > > -- global channel variables are applicable to all channels in the session > newSession = freeswitch.Session(globalChanVars .. chan1 .. limiter1 .. > chan2) > > -- bridge new Session to the current session > freeswitch.bridge(session, newSession) > > > > Much thanks! > > > regards, > Anita > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120508/ad525ff1/attachment.html From luis.daniel.lucio at gmail.com Wed May 9 03:47:09 2012 From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz) Date: Tue, 8 May 2012 19:47:09 -0400 Subject: [Freeswitch-users] What billing software do you recomend to use with FS? Message-ID: Can you gently recomend me a billing sofware for FW and why do you recomend me it. Regards. LD -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120508/fa6e7d47/attachment.html From msc at freeswitch.org Wed May 9 04:41:25 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 8 May 2012 17:41:25 -0700 Subject: [Freeswitch-users] Dialplan condition to test if SIP client is registered In-Reply-To: <2d5-4fa93780-5-a26ad20@121818419> References: <2d5-4fa93780-5-a26ad20@121818419> Message-ID: Thaddeus, FYI, doing an API from the dialplan is a piece of cake: http://wiki.freeswitch.org/wiki/Mod_commands#From_the_Dialplan In your case put this line in your dialplan before you do the sofia_contact check: That should solve the problem of people wandering off and their SIP clients not unregging. Give it a try and let us know how it works. -MC On Tue, May 8, 2012 at 8:11 AM, thaddeus at thogan.com wrote: > > Thanks for the information, I have explored the sofia_contact option and > it is promising, but I am unsure of how to make it work in my dialplan. > > I think I need to elaborate on the indended behavior some more. A user > will have three possible routes by which they can be reached. Two SIP > accounts and a POTS number. Each user has three routes but only two > devices, a desk phone and a cell phone. The cell phone is a target of the > POTS number and is also one of the SIP clients. I am afraid I did not > mention the desk phone initially, and I apologize for my question being > incomplete. > > Given this setup I want to select one of these two dialplans: > > 1. If SIP on cell phone is registered, ring && . > > 2. If SIP on cell is not registered, ring && > > sofia_contact looks promising, however it returns a string indicating that > a user is registered long after they have become unavailable. Example: User > walks out of building and cell phone loses the Wifi connection. The SIP > client did not un-register, it just went away. I have found that "sofia > check_sync" will check all registrations and correctly identify clients > that have gone off the network as no longer being registered. > > However, I cannot figure out how to run sofia_check sync before testing > the output of sofia_contact in my dialplan, any thoughts on how to do this? > Or is there a better way to get my above listed call routing to occur based > on the availability of a SIP client? > > Also, I did investigate using continue_on_fail to check for hangup causes > that did involve the SIP client being unreachable, but I could not make the > above logic occur using only that setting. That is, I always want to ring > , but only one of or , and never both > simultaneously or in sequence. > > Might I need to go into a Lua script for this kind of logic? > > Thanks again for your responses! > > -- Thaddeus > > > > On Friday, May 4, 2012 12:34 PM CDT, Michael Collins > wrote: > > Hi Thaddeus, > > You definitely have several options here. While you could use the > sofia_contact API, I'm wondering if perhaps you could let the dialplan and > the bridge app do the work for you. > > You could bridge directly to the Android user and then if it fails it could > then continue in the dialplan and ring the mobile number. This is covered > briefly on the wiki**, > however **it is covered very nicely in the new FreeSWITCH > Cookbook** in chapter 1. > > > Try out the basic failover (pipe-separated list of endpoints) and if that > doesn't work let us know. Maybe you could hop on IRC and talk live with > other community members. Of course, it can't hurt to raise your karma by > getting the FreeSWITCH books. :) (See freeswitch.org, upper left corner.) > > Thanks, > Michael > > On Fri, May 4, 2012 at 10:10 AM, thaddeus at thogan.com **wrote: > > > I was wondering if there is a way to tell if a SIP client is registered > > with a dialplan condition? Or maybe a better way to accomplish the > > following?: > > > > The users' cell phones (Android) are running a SIP client and connect to > > freeswitch via Wifi when in the office. I want calls destined for a given > > user to ring only their extension if they are connected via SIP, and ring > > only their mobile number when they are not connected. > > > > Currently I just take every incoming call and bridge it back out to their > > mobile number, but 95% of the time these users are on the Wifi network, > and > > I could just pass the call to their phones via SIP. > > > > I have tested hunt groups but the callers hear ringing for far too long. > > Ring groups don't work because it is confusing when the user > simultaneously > > receiving the same call via SIP and mobile on the same phone. > > > > Thanks in advance for any insights and help! > > > > -- Thaddeus > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120508/4983e2a9/attachment.html From dujinfang at gmail.com Wed May 9 07:19:09 2012 From: dujinfang at gmail.com (Seven Du) Date: Wed, 9 May 2012 11:19:09 +0800 Subject: [Freeswitch-users] mod_fifo how temporary to disable call dispatch to an offline agent In-Reply-To: References: <2740A3FE1B6E4D288416E9E1160B0C7A@gmail.com> Message-ID: <34092CE6BDDA402AB9D378E81579DA68@gmail.com> I use the example from the wiki, just I prefixed an f to aviod dialplan conflict, that means I have to call f6*1 to login f6#1 to logout f6101 to send a call to FIFO1 from the pastebin you can see agent 1000 will ring even it's on an outbound call: http://pastebin.freeswitch.org/19018 On Monday, May 7, 2012 at 9:57 AM, Michael Collins wrote: > Seven, > > Can you do a "fifo list_verbose " for these conditions: > * Agent not logged in > * Agent logged in and waiting for a call > * Agent logged in and is currently on inbound call > * Agent logged in and is currently on outbound call > > I want to make sure that we are testing the same thing. > > Thanks, > MC > > On Sat, May 5, 2012 at 7:48 AM, Seven Du wrote: > > Hi Michael, > > > > I tested that example and it did call the agent(Bria or any phone accept multi-lines) no matter it is busy or not, and I also looked the code looks there's no code to prevent this from happening. > > > > Michael I guess you used a phone only support 1 sip line, so any new incoming call will be rejected with USER_BUSY and fifo will try every seconds but only it will success when you hangup the current call. We once used a custom build sip ua with this solution, but leaves unsuccessful calls every second in logs if agents are "busy" calling out. > > > > I also tried to find a way to automatically "seize" an agent, but, technically, there's race conditions anyway. Say, agent send INVITE and it might receive INVITE at the same time because mod_fifo has no way to know the agent is start making a call. > > > > I currently use a temporary "call-back" solution: when client want to make a call, it send a request to my ESL app, and I update the fifo_outbound table to "seize" the agent line, if success(1 record line updated), then I originate the agent... It is a pain to make sure the "seized" line is cleared whenever the call is hangup or fail. It is kind of works in our lab, but I'm not sure how reliable it is. > > > > I had thought to add some code to mod_fifo, so in addition to "fifo out" add something like "fifo dial from to" to automatically resolve this. > > > > It seems no solution in mod_callcenter either it might be because most callcenters are either inbound or outbound, but in-out mixed agents are also seen for efficiency so it would be helpful to find a solution. > > > > Seven. > > > > > > On Saturday, May 5, 2012 at 5:46 AM, Michael Collins wrote: > > > > > Hi Afshin, > > > > > > Sorry for the delay but I wanted to lab this up and test it before I responded. I do not see the behavior you report. When I use the sample FIFO stuff on the wiki I am able to have an on-hook agent make an outbound call and not receive an inbound call. As soon as the agent completes his outbound call he is available to receive an inbound call. > > > > > > Check this page: > > > http://wiki.freeswitch.org/wiki/Mod_fifo#Simple_On-hook_Agent_Login.2FLogout_Example > > > > > > It works like a champ! > > > > > > -MC > > > > > > On Sat, Apr 28, 2012 at 10:05 PM, afshin afzali wrote: > > > > Dear FreeSWITCHers, > > > > > > > > The offline agents need to make outgoing calls while waiting for call arrivals. To prevent an influence outgoing by incoming call, they need to logout, making call and then login to the queue ! also doing logout & login automatically alters the actual login time :( > > > > I'm looking for a solution which can disable call dispatch temporary for a specific agent. > > > > Appreciate all comments, > > > > -- afshin > > > > > > > > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting Services: > > > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://wiki.freeswitch.org > > > http://www.cluecon.com > > > > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at 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 at freeswitch.org (mailto:consulting at freeswitch.org) > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at 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 at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120509/9d27ae3b/attachment-0001.html From nbhatti at gmail.com Wed May 9 09:47:33 2012 From: nbhatti at gmail.com (Muhammad Naseer Bhatti) Date: Wed, 9 May 2012 08:47:33 +0300 Subject: [Freeswitch-users] What billing software do you recomend to use with FS? In-Reply-To: References: Message-ID: Hi, take a look at http://wiki.freeswitch.org/wiki/Billing You will find a few options there. Thanks. On Wed, May 9, 2012 at 2:47 AM, Luis Daniel Lucio Quiroz < luis.daniel.lucio at gmail.com> wrote: > Can you gently recomend me a billing sofware for FW and why do you > recomend me it. > > Regards. > > LD > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120509/7a94281a/attachment.html From hmkias at gmail.com Wed May 9 09:51:59 2012 From: hmkias at gmail.com (HM Kias) Date: Wed, 9 May 2012 11:21:59 +0530 Subject: [Freeswitch-users] What billing software do you recomend to use with FS? In-Reply-To: References: Message-ID: Naseer, great job on the Billing sw, its really good. On Wed, May 9, 2012 at 11:17 AM, Muhammad Naseer Bhatti wrote: > Hi, take a look at http://wiki.freeswitch.org/wiki/Billing You will find > a few options there. > > Thanks. > > On Wed, May 9, 2012 at 2:47 AM, Luis Daniel Lucio Quiroz < > luis.daniel.lucio at gmail.com> wrote: > >> Can you gently recomend me a billing sofware for FW and why do you >> recomend me it. >> >> Regards. >> >> LD >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- HM Kias 91-9443467600 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120509/36426975/attachment.html From john.gathm at gmail.com Wed May 9 09:42:15 2012 From: john.gathm at gmail.com (John Gathm) Date: Wed, 9 May 2012 07:42:15 +0200 Subject: [Freeswitch-users] mod_sms send() not working as expected Message-ID: Hi, I am trying to make mod_sms work, it seems that with the configuration I give to freeswitch, it does not forward sms (though it might not be related to mod_sms but perhaps chatplan ?) works : -two extentions can call each other -the default example chatplan works with the autoreply ( Action reply(Hello, you said: ${_body} ), I receive the response SMS. however, when I just wan to use the "send" action ( ) it does not work, (*.*.* in ip's are only replaced by me in the mail for privacy, are full and correct in logs) 2012-05-08 23:15:38.784507 [INFO] mod_sms.c:300 Processing text message 1234->1005 in context default 2012-05-08 23:15:38.784507 [CONSOLE] mod_xml_curl.c:318 XML response is in /tmp/b953861e-9963-11e1-b83f-5d2cc73f3eee.tmp.xml Chatplan: 1005 parsing [default->demo] continue=false Chatplan: 1005 at 176.*.*.* Regex (PASS) [demo] to(1005 at 176.*.*.*) =~ /^(.*)$/ break=on-false Chatplan: 1005 at 176.*.*.* Action send() 2012-05-08 23:15:38.784507 [DEBUG] sofia_presence.c:208 Can't find registered user 1005 at 176.*.*.* 2012-05-08 23:15:38.784507 [WARNING] sofia_presence.c:4021 Not sending message to ourselves! although my clients are registered properly (and can call each other) 2012-05-08 22:44:20.944505 [WARNING] sofia_reg.c:1442 SIP auth challenge (REGISTER) on sofia profile 'internal' for [1234 at 176.*.*.*] from ip 78.*.*.* 2012-05-08 22:44:21.165051 [CONSOLE] mod_xml_curl.c:318 XML response is in /tmp/5a2cb204-995f-11e1-b7fb-5d2cc73f3eee.tmp.xml 2012-05-08 22:44:21.184514 [CONSOLE] mod_xml_curl.c:318 XML response is in /tmp/5a300bc0-995f-11e1-b7fc-5d2cc73f3eee.tmp.xml 2012-05-08 22:44:21.244503 [WARNING] sofia_reg.c:1442 SIP auth challenge (REGISTER) on sofia profile 'internal' for [1005 at 176.*.*.*] from ip 78.*.*.* 2012-05-08 22:44:21.324503 [CONSOLE] mod_xml_curl.c:318 XML response is in /tmp/5a463dd2-995f-11e1-b7fe-5d2cc73f3eee.tmp.xml 2012-05-08 22:44:21.384540 [CONSOLE] mod_xml_curl.c:318 XML response is in /tmp/5a4ff886-995f-11e1-b7ff-5d2cc73f3eee.tmp.xml trying to send a sms to myself does not work either, I also get "can't find registered user " even when I'm the sender and logged. Thanks in advance for any help. Regards, J. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120509/b60bde64/attachment.html From tomp at tomp.co.uk Wed May 9 12:04:21 2012 From: tomp at tomp.co.uk (Tom Parrott) Date: Wed, 9 May 2012 09:04:21 +0100 Subject: [Freeswitch-users] 1.2-rc2 Tagged and Tarball available on files.freeswitch.org In-Reply-To: References: Message-ID: <5b796df030c0eaae30804b38a2f578b8.squirrel@my.netvps.co.uk> Excellent, will try and build that tonight. The RPM SPEC file has seen a lot of improvements over the last few months, it's a lot more modular now. Cheers Tom > Congratulations!! I know it's been a long time coming - I've been lurking > around the FS site/irc for months! (my FS plans keep getting interfered > with though!) > > :) > > > On 09/05/2012, at 6:06 AM, Ken Rice wrote: > >> Hey Guys we?ve reached another milestone in getting FreeSWITCH 1.2 out >> the door. >> >> http://files.freeswitch.org/freeswitch-1.2.rc2.tar.bz2 >> >> Yep that?s what you want to test... >> >> Look for debian packages and updated RPMs for these soon >> >> K >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > From koralu at gmail.com Wed May 9 13:53:52 2012 From: koralu at gmail.com (Adrian Andrei) Date: Wed, 9 May 2012 12:53:52 +0300 Subject: [Freeswitch-users] mod_xml_cdr is not logging In-Reply-To: References: Message-ID: Hello, Thank you for answer but I am not getting any files on disk. When I disable mod_cdr_xml the logs are saved naturally in freeswitch/log. Ty From kochanowski.wojtek at gmail.com Wed May 9 14:37:40 2012 From: kochanowski.wojtek at gmail.com (Wojtek Kochanowski) Date: Wed, 9 May 2012 12:37:40 +0200 Subject: [Freeswitch-users] mod_xml_cdr is not logging In-Reply-To: References: Message-ID: Check permissions of log-dir and error-log-dir. Owner and group also. 2012/5/9 Adrian Andrei > Hello, > > Thank you for answer but I am not getting any files on disk. When I > disable mod_cdr_xml the logs are saved naturally in freeswitch/log. > > Ty > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120509/fa044bf3/attachment.html From miha at softnet.si Wed May 9 15:14:27 2012 From: miha at softnet.si (Miha) Date: Wed, 09 May 2012 13:14:27 +0200 Subject: [Freeswitch-users] Call intercept In-Reply-To: <4FA916DA.4020201@softnet.si> References: <4FA916DA.4020201@softnet.si> Message-ID: <4FAA5193.6020608@softnet.si> HI, just to let you know I figure it out:) thanks any way! Regards, Miha On 5/8/2012 2:51 PM, Miha wrote: > Hi, > > I need a little help about call intercept. I have read wiki > (http://wiki.freeswitch.org/wiki/Callgroup_intercept) but still having > few problems. > > 1. I have Opensips which works like load_balancer and registrar. > Behind Opensips I have two FS server, which are for load_balacing. > 2. I need to set a call intercept thing. I done it like it is written > on FS wiki but this does not work as I have Opensips. > > I have put this in public dialplan: > > > > > After I try to intercept the call this fails as in database is written different UUID. > > CAll goes like this: > > Phone-openips-FS-Openips-Phone > > Thank you for all your help! > > MIha > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120509/f6a0a3d1/attachment-0001.html From jmesquita at freeswitch.org Wed May 9 15:54:35 2012 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Wed, 9 May 2012 08:54:35 -0300 Subject: [Freeswitch-users] Call intercept In-Reply-To: <4FAA5193.6020608@softnet.si> References: <4FA916DA.4020201@softnet.si> <4FAA5193.6020608@softnet.si> Message-ID: Care to share the solution for future reference? The mailing list is always indexed by Google or whatnot ... On May 9, 2012 8:16 AM, "Miha" wrote: > HI, > > just to let you know I figure it out:) > > thanks any way! > > Regards, > Miha > > On 5/8/2012 2:51 PM, Miha wrote: > > Hi, > > I need a little help about call intercept. I have read wiki ( > http://wiki.freeswitch.org/wiki/Callgroup_intercept) but still having few > problems. > > 1. I have Opensips which works like load_balancer and registrar. Behind > Opensips I have two FS server, which are for load_balacing. > 2. I need to set a call intercept thing. I done it like it is written on > FS wiki but this does not work as I have Opensips. > > I have put this in public dialplan: > > > > > After I try to intercept the call this fails as in database is written different UUID. > > CAll goes like this: > > Phone-openips-FS-Openips-Phone > > Thank you for all your help! > > MIha > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120509/7a1fed25/attachment.html From abaci64 at gmail.com Wed May 9 18:28:56 2012 From: abaci64 at gmail.com (Abaci) Date: Wed, 09 May 2012 10:28:56 -0400 Subject: [Freeswitch-users] Freeswitch And 2 Network I/Fs In-Reply-To: <1545146083A72C4DB7B66584B7E5D98402BC0BFF@BY2PRD0410MB377.namprd04.prod.outlook.com> References: <1545146083A72C4DB7B66584B7E5D98402BBC647@BY2PRD0410MB377.namprd04.prod.outlook.com> <07BF4904977CC645B485E970424193AD10E9088BA2@localhost> <1545146083A72C4DB7B66584B7E5D98402BC0B3E@BY2PRD0410MB377.namprd04.prod.outlook.com> <1545146083A72C4DB7B66584B7E5D98402BC0BFF@BY2PRD0410MB377.namprd04.prod.outlook.com> Message-ID: <4FAA7F28.4080707@gmail.com> did you try On 5/8/2012 5:34 PM, Jerry Richards wrote: > > Okay, I think I have it working. This is the bridge statement I'm using: > > data="${sofia_contact(internal/${dialed_extension})},${sofia_contact(external/${dialed_extension})}"/> > > If anyone recommends a better bridge, please let me know. > > Thanks, > > Jerry > > *From:*freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of > *Jerry Richards > *Sent:* Tuesday, May 08, 2012 11:38 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Freeswitch And 2 Network I/Fs > > Hi, > > Okay, I tried setting the tags you suggest (rtp-ip, sip-ip, > presence-hosts, ext-rtp-ip, ext-sip-ip). I am able to register, but I > can't call from one network to the other (it just goes to voicemail). > I posted a trace at http://pastebin.freeswitch.org/19016 > > Thanks, > > Jerry > > *From:*freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of > *B.Tietz at pinguin.ag > *Sent:* Monday, May 07, 2012 7:51 AM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Freeswitch And 2 Network I/Fs > > Hi, > > from ODBC in the core in the wiki we have: > > setup the IP in the following params in each sip-profile: > > rtp-ip, sip-ip, presence-hosts, ext-rtp-ip, ext-sip-ip > > regards, > > Benjamin > > *Von:*freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] *Im Auftrag von > *Jerry Richards > *Gesendet:* Montag, 7. Mai 2012 16:37 > *An:* 'freeswitch-users at lists.freeswitch.org' > *Betreff:* [Freeswitch-users] Freeswitch And 2 Network I/Fs > > Is there an example of configuring Freeswitch to support both eth0 and > eth1, when they are connected to different networks? > > I know I can create two sip_profiles, which works fine for > registration, but in the past, I've had issues with Freeswitch not > sending media to the right interface (or at least, not knowing how to > control that). > > Thanks, > Jerry > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120509/de1ea382/attachment.html From f.dinucci at alice.it Wed May 9 17:12:21 2012 From: f.dinucci at alice.it (Fernando Di Nucci) Date: Wed, 09 May 2012 15:12:21 +0200 Subject: [Freeswitch-users] freeswitch and voipstunt Message-ID: <4FAA6D35.80302@alice.it> Hi all, I have an account at voipstunt. Everything's fine using their softphone, but in their website http://www.voipstunt.com/en/sipp.html they claim it is possible to use their service from a sip device. I tried this in freeswitch: but freeswitch can't register. Any help appreciated, thank you From anthony.minessale at gmail.com Wed May 9 19:56:00 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 9 May 2012 10:56:00 -0500 Subject: [Freeswitch-users] mod_sms send() not working as expected In-Reply-To: References: Message-ID: You can't call send again on an inbound message because it's already sent to you. What exactly are you looking to do? On Wed, May 9, 2012 at 12:42 AM, John Gathm wrote: > Hi, > > I am trying to make mod_sms work, it seems that with the configuration I > give to freeswitch, it does not forward sms (though it might not be related > to mod_sms but perhaps chatplan ?) > > works : > -two extentions can call each other > -the default example chatplan works with the autoreply ( Action reply(Hello, > you said: ${_body}? ), I receive the response SMS. > > however, when I just wan to use the "send" action ( application="send"/> ) it does not work, > > (*.*.* in ip's are only replaced by me in the mail for privacy, are full and > correct in logs) > > 2012-05-08 23:15:38.784507 [INFO] mod_sms.c:300 Processing text message > 1234->1005 in context default > 2012-05-08 23:15:38.784507 [CONSOLE] mod_xml_curl.c:318 XML response is in > /tmp/b953861e-9963-11e1-b83f-5d2cc73f3eee.tmp.xml > Chatplan: 1005 parsing [default->demo] continue=false > Chatplan: 1005 at 176.*.*.* Regex (PASS) [demo] to(1005 at 176.*.*.*) =~ /^(.*)$/ > break=on-false > Chatplan: 1005 at 176.*.*.* Action send() > 2012-05-08 23:15:38.784507 [DEBUG] sofia_presence.c:208 Can't find > registered user 1005 at 176.*.*.* > 2012-05-08 23:15:38.784507 [WARNING] sofia_presence.c:4021 Not sending > message to ourselves! > > although my clients are registered properly (and can call each other) > > 2012-05-08 22:44:20.944505 [WARNING] sofia_reg.c:1442 SIP auth challenge > (REGISTER) on sofia profile 'internal' for [1234 at 176.*.*.*] from ip 78.*.*.* > 2012-05-08 22:44:21.165051 [CONSOLE] mod_xml_curl.c:318 XML response is in > /tmp/5a2cb204-995f-11e1-b7fb-5d2cc73f3eee.tmp.xml > 2012-05-08 22:44:21.184514 [CONSOLE] mod_xml_curl.c:318 XML response is in > /tmp/5a300bc0-995f-11e1-b7fc-5d2cc73f3eee.tmp.xml > 2012-05-08 22:44:21.244503 [WARNING] sofia_reg.c:1442 SIP auth challenge > (REGISTER) on sofia profile 'internal' for [1005 at 176.*.*.*] from ip 78.*.*.* > 2012-05-08 22:44:21.324503 [CONSOLE] mod_xml_curl.c:318 XML response is in > /tmp/5a463dd2-995f-11e1-b7fe-5d2cc73f3eee.tmp.xml > 2012-05-08 22:44:21.384540 [CONSOLE] mod_xml_curl.c:318 XML response is in > /tmp/5a4ff886-995f-11e1-b7ff-5d2cc73f3eee.tmp.xml > > trying? to send a sms to myself does not work either, I also get "can't find > registered user " even when I'm the sender and logged. > > Thanks in advance for any help. > Regards, > > J. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From fluixab at bellsouth.net Wed May 9 19:56:16 2012 From: fluixab at bellsouth.net (Bernard Fluixa) Date: Wed, 9 May 2012 11:56:16 -0400 Subject: [Freeswitch-users] No ESL events received Message-ID: <916CC095-A5CA-4086-A1C9-23705396D5C1@bellsouth.net> Hi, I am running an ESL C program that collects Freeswitch events in a thread (esl_thread_create_detached(EventCollectorThread, switch_handle) ) following an "originate" command with the "return_ring_ready=true" option set, run by another process. When I call a soft phone, I receive all events (after the esl_events(switch_handle, ESL_EVENT_TYPE_PLAIN, "ALL") execution) and everything works as expected. However, that same program does NOT receive any Freeswitch events when I call a real phone number through my VOIP provider (flowroute) although Freeswitch console shows that the call has been answered (Channel [sofia/external/+19999999999] has been answered). What am I missing? Thank you Bernard From anton.jugatsu at gmail.com Wed May 9 20:22:19 2012 From: anton.jugatsu at gmail.com (Anton Kvashenkin) Date: Wed, 9 May 2012 20:22:19 +0400 Subject: [Freeswitch-users] freeswitch and voipstunt In-Reply-To: <4FAA6D35.80302@alice.it> References: <4FAA6D35.80302@alice.it> Message-ID: Without any siptraces (at least) to solve your issue would be extremely hard. 2012/5/9 Fernando Di Nucci > Hi all, > > I have an account at voipstunt. Everything's fine using their softphone, > but in their website > > http://www.voipstunt.com/en/sipp.html > > they claim it is possible to use their service from a sip device. > > I tried this in freeswitch: > > > > > > > > > > > > > > > > but freeswitch can't register. > > Any help appreciated, thank you > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120509/1ae133b6/attachment.html From nickolayr at gmail.com Wed May 9 20:34:14 2012 From: nickolayr at gmail.com (Nikolay Rogoshchenkov) Date: Wed, 9 May 2012 12:34:14 -0400 Subject: [Freeswitch-users] freeswitch and voipstunt In-Reply-To: <4FAA6D35.80302@alice.it> References: <4FAA6D35.80302@alice.it> Message-ID: The realm must be: "voipstunt.com" (see example SMC 7908 VoWBRA at http://www.voipstunt.com/en/sipp.html) -- Nikolay On Wed, May 9, 2012 at 9:12 AM, Fernando Di Nucci wrote: > Hi all, > > I have an account at voipstunt. Everything's fine using their softphone, > but in their website > > http://www.voipstunt.com/en/sipp.html > > they claim it is possible to use their service from a sip device. > > I tried this in freeswitch: > > > > > > > > > > > > > > > > but freeswitch can't register. > > Any help appreciated, thank you > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120509/e97b2908/attachment.html From jmesquita at freeswitch.org Wed May 9 21:09:20 2012 From: jmesquita at freeswitch.org (=?utf-8?Q?Jo=C3=A3o_Mesquita?=) Date: Wed, 9 May 2012 14:09:20 -0300 Subject: [Freeswitch-users] freeswitch and voipstunt In-Reply-To: <4FAA6D35.80302@alice.it> References: <4FAA6D35.80302@alice.it> Message-ID: <361EA393A12C4117ABC241D294B4EAFA@freeswitch.org> More info such as a console log and sip trace would be helpful? Help us help you -- Jo?o Mesquita Sent with Sparrow (http://www.sparrowmailapp.com/?sig) On Wednesday, May 9, 2012 at 10:12 AM, Fernando Di Nucci wrote: > Hi all, > > I have an account at voipstunt. Everything's fine using their softphone, > but in their website > > http://www.voipstunt.com/en/sipp.html > > they claim it is possible to use their service from a sip device. > > I tried this in freeswitch: > > > > > > > > > > > > > > > > but freeswitch can't register. > > Any help appreciated, thank you > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120509/e64ee9c1/attachment-0001.html From f.dinucci at alice.it Wed May 9 21:24:41 2012 From: f.dinucci at alice.it (Fernando Di Nucci) Date: Wed, 09 May 2012 19:24:41 +0200 Subject: [Freeswitch-users] freeswitch and voipstunt Message-ID: <4FAAA859.5060601@alice.it> First of all thank you for your time. According to your suggestions I tried: after issuing at fs_cli console: /debug 9 sofia profile external siptrace on I get ------------------------------------------------------------------------ 2012-05-09 19:18:39.041881 [WARNING] sofia_reg.c:405 Timeout Registering voipstunt 2012-05-09 19:18:40.041894 [WARNING] sofia_reg.c:429 voipstunt Failed Registration [0], setting retry to 60 seconds. send 645 bytes to udp/[77.72.169.134]:5060 at 17:18:40.065594: ------------------------------------------------------------------------ REGISTER sip:sip.voipstunt.com;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.30.30:5080;rport;branch=z9hG4bK2XryvegBXNejD Max-Forwards: 70 From: ;tag=2crj3HatDt40g To: Call-ID: 35f3320b-4bd0-414a-8324-0a98c411f6c6 CSeq: 27955864 REGISTER Contact: Expires: 800 User-Agent: FreeSWITCH-mod_sofia/1.0.head-svn-exported Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Content-Length: 0 ------------------------------------------------------------------------ repeated many times. I'm new to freeswitch, so if a have to do something different, please let me know. Thank you From msc at freeswitch.org Wed May 9 21:33:04 2012 From: msc at freeswitch.org (Michael Collins) Date: Wed, 9 May 2012 10:33:04 -0700 Subject: [Freeswitch-users] freeswitch and voipstunt In-Reply-To: <4FAAA859.5060601@alice.it> References: <4FAAA859.5060601@alice.it> Message-ID: So you are sending registration packets out but you aren't getting anything back? I would try a few different things. First, for testing, try moving your gateway over to the internal profile. I'm wondering if they are trying to send stuff back to port 5060. -MC On Wed, May 9, 2012 at 10:24 AM, Fernando Di Nucci wrote: > First of all thank you for your time. > > According to your suggestions I tried: > > > > > > > > > > > > > > > > after issuing at fs_cli console: > > /debug 9 > sofia profile external siptrace on > > I get > > ------------------------------------------------------------------------ > 2012-05-09 19:18:39.041881 [WARNING] sofia_reg.c:405 Timeout Registering > voipstunt > 2012-05-09 19:18:40.041894 [WARNING] sofia_reg.c:429 voipstunt Failed > Registration [0], setting retry to 60 seconds. > send 645 bytes to udp/[77.72.169.134]:5060 at 17:18:40.065594: > ------------------------------------------------------------------------ > REGISTER sip:sip.voipstunt.com;transport=udp SIP/2.0 > Via: SIP/2.0/UDP 192.168.30.30:5080;rport;branch=z9hG4bK2XryvegBXNejD > Max-Forwards: 70 > From: ;tag=2crj3HatDt40g > To: > Call-ID: 35f3320b-4bd0-414a-8324-0a98c411f6c6 > CSeq: 27955864 REGISTER > Contact: > > Expires: 800 > User-Agent: FreeSWITCH-mod_sofia/1.0.head-svn-exported > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY > Supported: timer, precondition, path, replaces > Content-Length: 0 > > ------------------------------------------------------------------------ > > > repeated many times. > > I'm new to freeswitch, so if a have to do something different, please > let me know. > > Thank you > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120509/cdcdb1a8/attachment.html From nickolayr at gmail.com Wed May 9 21:35:38 2012 From: nickolayr at gmail.com (Nikolay Rogoshchenkov) Date: Wed, 9 May 2012 13:35:38 -0400 Subject: [Freeswitch-users] freeswitch and voipstunt In-Reply-To: <4FAAA859.5060601@alice.it> References: <4FAAA859.5060601@alice.it> Message-ID: Do you have configured public interface? Show us "sofia status" output? -- Nikolay On Wed, May 9, 2012 at 1:24 PM, Fernando Di Nucci wrote: > First of all thank you for your time. > > According to your suggestions I tried: > > > > > > > > > > > > > > > > after issuing at fs_cli console: > > /debug 9 > sofia profile external siptrace on > > I get > > ------------------------------------------------------------------------ > 2012-05-09 19:18:39.041881 [WARNING] sofia_reg.c:405 Timeout Registering > voipstunt > 2012-05-09 19:18:40.041894 [WARNING] sofia_reg.c:429 voipstunt Failed > Registration [0], setting retry to 60 seconds. > send 645 bytes to udp/[77.72.169.134]:5060 at 17:18:40.065594: > ------------------------------------------------------------------------ > REGISTER sip:sip.voipstunt.com;transport=udp SIP/2.0 > Via: SIP/2.0/UDP 192.168.30.30:5080;rport;branch=z9hG4bK2XryvegBXNejD > Max-Forwards: 70 > From: ;tag=2crj3HatDt40g > To: > Call-ID: 35f3320b-4bd0-414a-8324-0a98c411f6c6 > CSeq: 27955864 REGISTER > Contact: > > Expires: 800 > User-Agent: FreeSWITCH-mod_sofia/1.0.head-svn-exported > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY > Supported: timer, precondition, path, replaces > Content-Length: 0 > > ------------------------------------------------------------------------ > > > repeated many times. > > I'm new to freeswitch, so if a have to do something different, please > let me know. > > Thank you > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120509/f6ca4ded/attachment.html From f.dinucci at alice.it Wed May 9 22:28:12 2012 From: f.dinucci at alice.it (Fernando Di Nucci) Date: Wed, 09 May 2012 20:28:12 +0200 Subject: [Freeswitch-users] freeswitch and voipstunt Message-ID: <4FAAB73C.1060705@alice.it> freeswitch at internal> sofia status Name Type Data State ================================================================================================= external profile sip:mod_sofia at 192.168.30.30:5080 RUNNING (0) external::voipstunt gateway sip:*********@sip.voipstunt.com TRYING (retry: 46s) external::pstn gateway sip:81 at 192.168.30.176:5062 NOREG external::Messagenet gateway sip:*******@sip.messagenet.it:5061 REGED 192.168.30.30 alias internal ALIASED internal profile sip:mod_sofia at 192.168.30.30:5060 RUNNING (0) ================================================================================================= 2 profiles 1 alias pstn is a ht-503 (working) and also messagenet is working. @Michael Collins: move the gateway to the internal profile means putting the file in internal folder ? Please be patient ! I tried to register X-lite 4 but no luck, the only registration that works seems their own softphone... From nickolayr at gmail.com Wed May 9 23:13:19 2012 From: nickolayr at gmail.com (Nikolay Rogoshchenkov) Date: Wed, 9 May 2012 15:13:19 -0400 Subject: [Freeswitch-users] freeswitch and voipstunt In-Reply-To: <4FAAB73C.1060705@alice.it> References: <4FAAB73C.1060705@alice.it> Message-ID: I have the same problem when I trying to register from FreeSWITCH: send 649 bytes to udp/[77.72.169.134]:5060 at 18:41:43.893908: ------------------------------------------------------------------------ REGISTER sip:sip.voipstunt.com;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.9.0.23;rport;branch=z9hG4bK113r8tDFpmgFe Max-Forwards: 70 From: ;tag=17DvgFXt5UHDm To: Call-ID: ea47e284-069a-e111-85db-0017a4453d75 CSeq: 27958366 REGISTER Contact: Expires: 3600 User-Agent: Core_InternalSWITCH-2.0/-2012 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Length: 0 But, I have answer (401...) when I trying to register from eyeBeam. [image: Inline image 1] -- Nikolay On Wed, May 9, 2012 at 2:28 PM, Fernando Di Nucci wrote: > freeswitch at internal> sofia status > Name Type > Data State > > ================================================================================================= > external profile > sip:mod_sofia at 192.168.30.30:5080 RUNNING (0) > external::voipstunt gateway > sip:*********@sip.voipstunt.com TRYING (retry: 46s) > external::pstn gateway > sip:81 at 192.168.30.176:5062 NOREG > external::Messagenet gateway > sip:*******@sip.messagenet.it:5061 REGED > 192.168.30.30 alias > internal ALIASED > internal profile > sip:mod_sofia at 192.168.30.30:5060 RUNNING (0) > > ================================================================================================= > 2 profiles 1 alias > > > pstn is a ht-503 (working) and also messagenet is working. > > > > > @Michael Collins: move the gateway to the internal profile means putting > the file in internal folder ? Please be patient ! > > > I tried to register X-lite 4 but no luck, the only registration that > works seems their own softphone... > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120509/04997895/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/png Size: 47960 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120509/04997895/attachment-0001.png From john.gathm at gmail.com Wed May 9 23:26:19 2012 From: john.gathm at gmail.com (John Gathm) Date: Wed, 9 May 2012 21:26:19 +0200 Subject: [Freeswitch-users] mod_sms send() not working as expected In-Reply-To: References: Message-ID: Actually I'm just trying to send a simple message between two endpoints extensions. I was mistaken and believed the chatplan was necessary to achieve this, so I disabled it to get the default behaviour of sending. But I still get the error 2012-05-09 19:17:42.224503 [INFO] mod_sms.c:300 Processing text message 1234->1234 in context default 2012-05-09 19:17:42.224503 [CONSOLE] mod_xml_curl.c:318 XML response is in /tmp/a64045dc-9a0b-11e1-852b-5d2cc73f3eee.tmp.xml Chatplan: 1234 parsing [default->demo] continue=false 2012-05-09 19:17:42.224503 [DEBUG] sofia_presence.c:208 Can't find registered user 1234 at 176.*.*.* 2012-05-09 19:17:42.224503 [WARNING] sofia_presence.c:4021 Not sending message to ourselves! FYI, I'm using freeswitch in Amazon Web Services, and it is NATted. Freeswitch does see the client registered with the internal private IP, not the public one. Could this be a lead ? Although this does not explain why it works with voice calls. freeswitch at internal> sofia_count_reg 1234 at 176.*.*.* -1 freeswitch at internal> sofia_count_reg 1234 at 10.*.*.* 1 Regards, J. On Wed, May 9, 2012 at 5:56 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > You can't call send again on an inbound message because it's already > sent to you. > What exactly are you looking to do? > > > On Wed, May 9, 2012 at 12:42 AM, John Gathm wrote: > > Hi, > > > > I am trying to make mod_sms work, it seems that with the configuration I > > give to freeswitch, it does not forward sms (though it might not be > related > > to mod_sms but perhaps chatplan ?) > > > > works : > > -two extentions can call each other > > -the default example chatplan works with the autoreply ( Action > reply(Hello, > > you said: ${_body} ), I receive the response SMS. > > > > however, when I just wan to use the "send" action ( > application="send"/> ) it does not work, > > > > (*.*.* in ip's are only replaced by me in the mail for privacy, are full > and > > correct in logs) > > > > 2012-05-08 23:15:38.784507 [INFO] mod_sms.c:300 Processing text message > > 1234->1005 in context default > > 2012-05-08 23:15:38.784507 [CONSOLE] mod_xml_curl.c:318 XML response is > in > > /tmp/b953861e-9963-11e1-b83f-5d2cc73f3eee.tmp.xml > > Chatplan: 1005 parsing [default->demo] continue=false > > Chatplan: 1005 at 176.*.*.* Regex (PASS) [demo] to(1005 at 176.*.*.*) =~ > /^(.*)$/ > > break=on-false > > Chatplan: 1005 at 176.*.*.* Action send() > > 2012-05-08 23:15:38.784507 [DEBUG] sofia_presence.c:208 Can't find > > registered user 1005 at 176.*.*.* > > 2012-05-08 23:15:38.784507 [WARNING] sofia_presence.c:4021 Not sending > > message to ourselves! > > > > although my clients are registered properly (and can call each other) > > > > 2012-05-08 22:44:20.944505 [WARNING] sofia_reg.c:1442 SIP auth challenge > > (REGISTER) on sofia profile 'internal' for [1234 at 176.*.*.*] from ip > 78.*.*.* > > 2012-05-08 22:44:21.165051 [CONSOLE] mod_xml_curl.c:318 XML response is > in > > /tmp/5a2cb204-995f-11e1-b7fb-5d2cc73f3eee.tmp.xml > > 2012-05-08 22:44:21.184514 [CONSOLE] mod_xml_curl.c:318 XML response is > in > > /tmp/5a300bc0-995f-11e1-b7fc-5d2cc73f3eee.tmp.xml > > 2012-05-08 22:44:21.244503 [WARNING] sofia_reg.c:1442 SIP auth challenge > > (REGISTER) on sofia profile 'internal' for [1005 at 176.*.*.*] from ip > 78.*.*.* > > 2012-05-08 22:44:21.324503 [CONSOLE] mod_xml_curl.c:318 XML response is > in > > /tmp/5a463dd2-995f-11e1-b7fe-5d2cc73f3eee.tmp.xml > > 2012-05-08 22:44:21.384540 [CONSOLE] mod_xml_curl.c:318 XML response is > in > > /tmp/5a4ff886-995f-11e1-b7ff-5d2cc73f3eee.tmp.xml > > > > trying to send a sms to myself does not work either, I also get "can't > find > > registered user " even when I'm the sender and logged. > > > > Thanks in advance for any help. > > Regards, > > > > J. > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120509/1a954e1d/attachment.html From lesley.pervis at gmail.com Wed May 9 23:33:59 2012 From: lesley.pervis at gmail.com (Lesley Pervis) Date: Wed, 9 May 2012 13:33:59 -0600 Subject: [Freeswitch-users] User-specific gateways Message-ID: Is this supposed to work? For me, when the user registers, the gateway does not register. There's no indication in either the FreeSWITCH or sofia logs that an attempt is even contemplated. http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide#User-Specific_Gateways -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120509/076051dc/attachment.html From lesley.pervis at gmail.com Wed May 9 23:52:46 2012 From: lesley.pervis at gmail.com (Lesley Pervis) Date: Wed, 9 May 2012 13:52:46 -0600 Subject: [Freeswitch-users] User-specific gateways In-Reply-To: References: Message-ID: Never mind. The user and the gateway can't be named the same thing. On Wed, May 9, 2012 at 1:33 PM, Lesley Pervis wrote: > Is this supposed to work? For me, when the user registers, the gateway > does not register. There's no indication in either the FreeSWITCH or sofia > logs that an attempt is even contemplated. > > > http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide#User-Specific_Gateways > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120509/33529c26/attachment.html From f.dinucci at alice.it Thu May 10 00:57:09 2012 From: f.dinucci at alice.it (Fernando Di Nucci) Date: Wed, 09 May 2012 22:57:09 +0200 Subject: [Freeswitch-users] freeswitch and voipstunt Message-ID: <4FAADA25.8050804@alice.it> That is to say I'm hopeless ? :-( From nickolayr at gmail.com Thu May 10 01:27:40 2012 From: nickolayr at gmail.com (Nikolay Rogoshchenkov) Date: Wed, 9 May 2012 17:27:40 -0400 Subject: [Freeswitch-users] freeswitch and voipstunt In-Reply-To: <4FAADA25.8050804@alice.it> References: <4FAADA25.8050804@alice.it> Message-ID: Since it's works with eyeBeam it must work with FS too. Try to play with examples: here http://wiki.freeswitch.org/wiki/SIP_Provider_Examples -- Nikolay On Wed, May 9, 2012 at 4:57 PM, Fernando Di Nucci wrote: > That is to say I'm hopeless ? :-( > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120509/f6c711ba/attachment.html From shane at longwhitecloud.com Thu May 10 00:46:22 2012 From: shane at longwhitecloud.com (Shane Harrison) Date: Thu, 10 May 2012 08:46:22 +1200 Subject: [Freeswitch-users] Freetdm DTMF In-Reply-To: References: Message-ID: Hi All, Have a situation where I have a call between a SIP phone and a FreeTDM channel. Pushing *3 on the analog FreeTDM phone is detected and this is bound to a dialplan extension (attended transfer) that does a read(): However pushing further digits on the analog phone ie. extension number of phone we wish to do an attended transfer to , doesn't result in the DTMF being detected. Note that this all works the other way around ie. using the SIP phone. When the *3 digits are pushed on the analog phone I see the logs report: ftdm_io.c:3463 [s2c1][1:4] Queuing DTMF * (debug = 0) mod_freetdm.c:702 Queuing DTMF [*] in channel FreeTDM/2:1/ ftdm_io.c:3463 [s2c1][1:4] Queuing DTMF 4 (debug = 0) mod_freetdm.c:702 Queuing DTMF [4] in channel FreeTDM/2:1/ When the further keys are pushed ftdm_io reports nothing. I have tried inserting a start_dtmf before the read() but it had no effect. Any thoughts as to why DTMF isn't being seen from the analog phone after the read()? Cheers Shane -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120510/b8074f43/attachment-0001.html From lesley.pervis at gmail.com Thu May 10 02:00:49 2012 From: lesley.pervis at gmail.com (Lesley Pervis) Date: Wed, 9 May 2012 16:00:49 -0600 Subject: [Freeswitch-users] User-specific gateways In-Reply-To: References: Message-ID: Actually, does anyone know how to set the sofia profile a user-specific-gateway is associated with? The one I've got working uses external, and I can't find any configuration to cause or affect this. On Wed, May 9, 2012 at 1:52 PM, Lesley Pervis wrote: > Never mind. The user and the gateway can't be named the same thing. > > > On Wed, May 9, 2012 at 1:33 PM, Lesley Pervis wrote: > >> Is this supposed to work? For me, when the user registers, the gateway >> does not register. There's no indication in either the FreeSWITCH or sofia >> logs that an attempt is even contemplated. >> >> >> http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide#User-Specific_Gateways >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120509/9b4007c9/attachment.html From jmesquita at freeswitch.org Thu May 10 02:03:40 2012 From: jmesquita at freeswitch.org (=?utf-8?Q?Jo=C3=A3o_Mesquita?=) Date: Wed, 9 May 2012 19:03:40 -0300 Subject: [Freeswitch-users] User-specific gateways In-Reply-To: References: Message-ID: Look at the profile config. If you set your profile to scan the user directory, it will own the gateways it find. Drawback is that you can only have 1 profile scan it at a time. -- Jo?o Mesquita Sent with Sparrow (http://www.sparrowmailapp.com/?sig) On Wednesday, May 9, 2012 at 7:00 PM, Lesley Pervis wrote: > Actually, does anyone know how to set the sofia profile a user-specific-gateway is associated with? The one I've got working uses external, and I can't find any configuration to cause or affect this. > > > On Wed, May 9, 2012 at 1:52 PM, Lesley Pervis wrote: > > Never mind. The user and the gateway can't be named the same thing. > > > > > > On Wed, May 9, 2012 at 1:33 PM, Lesley Pervis wrote: > > > Is this supposed to work? For me, when the user registers, the gateway does not register. There's no indication in either the FreeSWITCH or sofia logs that an attempt is even contemplated. > > > > > > http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide#User-Specific_Gateways > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120509/d39a9786/attachment.html From curriegrad2004 at gmail.com Thu May 10 02:22:21 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Wed, 9 May 2012 15:22:21 -0700 Subject: [Freeswitch-users] Freetdm DTMF In-Reply-To: References: Message-ID: I'm guessing the bind digits in your analog card was set to listen for this sequence on the a-leg given if the call was being routed from the IP side to the analog side. Try changing that to listen on the b-leg. On 5/9/12, Shane Harrison wrote: > Hi All, > > Have a situation where I have a call between a SIP phone and a FreeTDM > channel. Pushing *3 on the analog FreeTDM phone is detected and this is > bound to a dialplan extension (attended transfer) that does a read(): > > > However pushing further digits on the analog phone ie. extension number of > phone we wish to do an attended transfer to , doesn't result in the DTMF > being detected. Note that this all works the other way around ie. using > the SIP phone. > > When the *3 digits are pushed on the analog phone I see the logs report: > > ftdm_io.c:3463 [s2c1][1:4] Queuing DTMF * (debug = 0) > mod_freetdm.c:702 Queuing DTMF [*] in channel FreeTDM/2:1/ > > ftdm_io.c:3463 [s2c1][1:4] Queuing DTMF 4 (debug = 0) > > mod_freetdm.c:702 Queuing DTMF [4] in channel FreeTDM/2:1/ > > When the further keys are pushed ftdm_io reports nothing. > > I have tried inserting a start_dtmf before the read() but it had no effect. > Any thoughts as to why DTMF isn't being seen from the analog phone after > the read()? > > Cheers > Shane > From shane.harrison at paragon.co.nz Thu May 10 03:12:00 2012 From: shane.harrison at paragon.co.nz (Shane Harrison) Date: Thu, 10 May 2012 11:12:00 +1200 Subject: [Freeswitch-users] Freetdm DTMF In-Reply-To: References: Message-ID: Thanks. I am currently using bind_meta_app (set to both legs) already rather than bind_digits. I'll give bind_digits a shot and see if it behaves differently. Note that I do detect the initial *3 digits and because bind_meta_app is both legs, this is successful no matter which direction the call is setup from. However once the dialplan moves to the extension the *3 is bound to, digits are no longer received. The worrying thing for me is that ftdm_io.c doesn't even write to the log that it has received them (nor freetdm above that of course which is understandable) and I am surprised that the read() influences that since it works prior on the *3 digits. Cheers Shane On Thu, May 10, 2012 at 10:22 AM, curriegrad2004 wrote: > I'm guessing the bind digits in your analog card was set to listen for > this sequence on the a-leg given if the call was being routed from the > IP side to the analog side. > > Try changing that to listen on the b-leg. > > On 5/9/12, Shane Harrison wrote: > > Hi All, > > > > Have a situation where I have a call between a SIP phone and a FreeTDM > > channel. Pushing *3 on the analog FreeTDM phone is detected and this is > > bound to a dialplan extension (attended transfer) that does a read(): > > > > > > However pushing further digits on the analog phone ie. extension number > of > > phone we wish to do an attended transfer to , doesn't result in the DTMF > > being detected. Note that this all works the other way around ie. using > > the SIP phone. > > > > When the *3 digits are pushed on the analog phone I see the logs report: > > > > ftdm_io.c:3463 [s2c1][1:4] Queuing DTMF * (debug = 0) > > mod_freetdm.c:702 Queuing DTMF [*] in channel FreeTDM/2:1/ > > > > ftdm_io.c:3463 [s2c1][1:4] Queuing DTMF 4 (debug = 0) > > > > mod_freetdm.c:702 Queuing DTMF [4] in channel FreeTDM/2:1/ > > > > When the further keys are pushed ftdm_io reports nothing. > > > > I have tried inserting a start_dtmf before the read() but it had no > effect. > > Any thoughts as to why DTMF isn't being seen from the analog phone after > > the read()? > > > > Cheers > > Shane > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Paragon Electronic Design Ltd L6 Crest House 92 Queens Drive P0 Box 30449 Lower Hutt 5040 +64 4 5703870 Extn 875 +64 21 608919 (mobile) "Solving your problems with the right technology" -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120510/54c76205/attachment.html From curriegrad2004 at gmail.com Thu May 10 04:26:45 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Wed, 9 May 2012 17:26:45 -0700 Subject: [Freeswitch-users] Freetdm DTMF In-Reply-To: References: Message-ID: apologies for telling you the wrong thing. I was replying you from my phone btw :P Yeah, bind_meta_app is the app you would use, but try changing it to point to the b-leg, not the a-leg On Wed, May 9, 2012 at 4:12 PM, Shane Harrison wrote: > Thanks.? I am currently using bind_meta_app (set to both legs) already > rather than bind_digits.? I'll give bind_digits a shot and see if it behaves > differently. > > Note that I do detect the initial *3 digits and because bind_meta_app is > both legs, this is successful no matter which direction the call is setup > from.? However once the dialplan moves to the extension the *3 is bound to, > digits are no longer received. > > The worrying thing for me is that ftdm_io.c doesn't even write to the log > that it has received them (nor freetdm above that of course which is > understandable) and I am surprised that the read() influences that since it > works prior on the *3 digits. > > Cheers > Shane > > > > On Thu, May 10, 2012 at 10:22 AM, curriegrad2004 > wrote: >> >> I'm guessing the bind digits in your analog card was set to listen for >> this sequence on the a-leg given if the call was being routed from the >> IP side to the analog side. >> >> Try changing that to listen on the b-leg. >> >> On 5/9/12, Shane Harrison wrote: >> > Hi All, >> > >> > Have a situation where I have a call between a SIP phone and a FreeTDM >> > channel. ? Pushing *3 on the analog FreeTDM phone is detected and this >> > is >> > bound to a dialplan extension (attended transfer) that does a read(): >> > >> > >> > However pushing further digits on the analog phone ie. extension number >> > of >> > phone we wish to do an attended transfer to , doesn't result in the DTMF >> > being detected. ?Note that this all works the other way around ie. using >> > the SIP phone. >> > >> > When the *3 digits are pushed on the analog phone I see the logs report: >> > >> > ?ftdm_io.c:3463 [s2c1][1:4] Queuing DTMF * (debug = 0) >> > mod_freetdm.c:702 Queuing DTMF [*] in channel FreeTDM/2:1/ >> > >> > ftdm_io.c:3463 [s2c1][1:4] Queuing DTMF 4 (debug = 0) >> > >> > mod_freetdm.c:702 Queuing DTMF [4] in channel FreeTDM/2:1/ >> > >> > When the further keys are pushed ftdm_io reports nothing. >> > >> > I have tried inserting a start_dtmf before the read() but it had no >> > effect. >> > Any thoughts as to why DTMF isn't being seen from the analog phone after >> > the read()? >> > >> > Cheers >> > Shane >> > >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > > -- > Paragon Electronic Design Ltd > L6 Crest House > 92 Queens Drive > P0 Box 30449 > Lower Hutt 5040 > > +64 4 5703870 Extn 875 > +64 21 608919? (mobile) > > "Solving your problems with the right technology" > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From curriegrad2004 at gmail.com Thu May 10 04:28:57 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Wed, 9 May 2012 17:28:57 -0700 Subject: [Freeswitch-users] Freetdm DTMF In-Reply-To: References: Message-ID: and crap, since I wasn't even reading anything here, on the subsequent transfers from your FXO card, enable the in-band DTMF detector that FS has. The details on the in-band DTMF detector is here: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf But use this with caution, if there is a DTMF detector on the FXO card itself, make sure you disable it before using it. On Wed, May 9, 2012 at 5:26 PM, curriegrad2004 wrote: > apologies for telling you the wrong thing. I was replying you from my > phone btw :P > > Yeah, bind_meta_app is the app you would use, but try changing it to > point to the b-leg, not the a-leg > > On Wed, May 9, 2012 at 4:12 PM, Shane Harrison > wrote: >> Thanks.? I am currently using bind_meta_app (set to both legs) already >> rather than bind_digits.? I'll give bind_digits a shot and see if it behaves >> differently. >> >> Note that I do detect the initial *3 digits and because bind_meta_app is >> both legs, this is successful no matter which direction the call is setup >> from.? However once the dialplan moves to the extension the *3 is bound to, >> digits are no longer received. >> >> The worrying thing for me is that ftdm_io.c doesn't even write to the log >> that it has received them (nor freetdm above that of course which is >> understandable) and I am surprised that the read() influences that since it >> works prior on the *3 digits. >> >> Cheers >> Shane >> >> >> >> On Thu, May 10, 2012 at 10:22 AM, curriegrad2004 >> wrote: >>> >>> I'm guessing the bind digits in your analog card was set to listen for >>> this sequence on the a-leg given if the call was being routed from the >>> IP side to the analog side. >>> >>> Try changing that to listen on the b-leg. >>> >>> On 5/9/12, Shane Harrison wrote: >>> > Hi All, >>> > >>> > Have a situation where I have a call between a SIP phone and a FreeTDM >>> > channel. ? Pushing *3 on the analog FreeTDM phone is detected and this >>> > is >>> > bound to a dialplan extension (attended transfer) that does a read(): >>> > >>> > >>> > However pushing further digits on the analog phone ie. extension number >>> > of >>> > phone we wish to do an attended transfer to , doesn't result in the DTMF >>> > being detected. ?Note that this all works the other way around ie. using >>> > the SIP phone. >>> > >>> > When the *3 digits are pushed on the analog phone I see the logs report: >>> > >>> > ?ftdm_io.c:3463 [s2c1][1:4] Queuing DTMF * (debug = 0) >>> > mod_freetdm.c:702 Queuing DTMF [*] in channel FreeTDM/2:1/ >>> > >>> > ftdm_io.c:3463 [s2c1][1:4] Queuing DTMF 4 (debug = 0) >>> > >>> > mod_freetdm.c:702 Queuing DTMF [4] in channel FreeTDM/2:1/ >>> > >>> > When the further keys are pushed ftdm_io reports nothing. >>> > >>> > I have tried inserting a start_dtmf before the read() but it had no >>> > effect. >>> > Any thoughts as to why DTMF isn't being seen from the analog phone after >>> > the read()? >>> > >>> > Cheers >>> > Shane >>> > >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> >> >> -- >> Paragon Electronic Design Ltd >> L6 Crest House >> 92 Queens Drive >> P0 Box 30449 >> Lower Hutt 5040 >> >> +64 4 5703870 Extn 875 >> +64 21 608919? (mobile) >> >> "Solving your problems with the right technology" >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> From shane.harrison at paragon.co.nz Thu May 10 06:27:45 2012 From: shane.harrison at paragon.co.nz (Shane Harrison) Date: Thu, 10 May 2012 14:27:45 +1200 Subject: [Freeswitch-users] Freetdm DTMF In-Reply-To: References: Message-ID: Thanks for the thoughts. As I said, I am already setting it to both legs - I will try simply trying one leg but am sceptical :-) I also mentioned that I called the start_dtmf just before calling the read so unless I am doing something wrong here..... I'll try and post the XML tonight when I get home. Oh and it is an FXS card not an FXO of course since it has a phone plugged into it. The question still remains though, why is the in-band DTMF detection working for the bind_meta_app digit detection but not after that? Cheers Shane On Thu, May 10, 2012 at 12:28 PM, curriegrad2004 wrote: > and crap, since I wasn't even reading anything here, on the subsequent > transfers from your FXO card, enable the in-band DTMF detector that FS > has. The details on the in-band DTMF detector is here: > > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf > > But use this with caution, if there is a DTMF detector on the FXO card > itself, make sure you disable it before using it. > > On Wed, May 9, 2012 at 5:26 PM, curriegrad2004 > wrote: > > apologies for telling you the wrong thing. I was replying you from my > > phone btw :P > > > > Yeah, bind_meta_app is the app you would use, but try changing it to > > point to the b-leg, not the a-leg > > > > On Wed, May 9, 2012 at 4:12 PM, Shane Harrison > > wrote: > >> Thanks. I am currently using bind_meta_app (set to both legs) already > >> rather than bind_digits. I'll give bind_digits a shot and see if it > behaves > >> differently. > >> > >> Note that I do detect the initial *3 digits and because bind_meta_app is > >> both legs, this is successful no matter which direction the call is > setup > >> from. However once the dialplan moves to the extension the *3 is bound > to, > >> digits are no longer received. > >> > >> The worrying thing for me is that ftdm_io.c doesn't even write to the > log > >> that it has received them (nor freetdm above that of course which is > >> understandable) and I am surprised that the read() influences that > since it > >> works prior on the *3 digits. > >> > >> Cheers > >> Shane > >> > >> > >> > >> On Thu, May 10, 2012 at 10:22 AM, curriegrad2004 < > curriegrad2004 at gmail.com> > >> wrote: > >>> > >>> I'm guessing the bind digits in your analog card was set to listen for > >>> this sequence on the a-leg given if the call was being routed from the > >>> IP side to the analog side. > >>> > >>> Try changing that to listen on the b-leg. > >>> > >>> On 5/9/12, Shane Harrison wrote: > >>> > Hi All, > >>> > > >>> > Have a situation where I have a call between a SIP phone and a > FreeTDM > >>> > channel. Pushing *3 on the analog FreeTDM phone is detected and > this > >>> > is > >>> > bound to a dialplan extension (attended transfer) that does a read(): > >>> > > >>> > > >>> > However pushing further digits on the analog phone ie. extension > number > >>> > of > >>> > phone we wish to do an attended transfer to , doesn't result in the > DTMF > >>> > being detected. Note that this all works the other way around ie. > using > >>> > the SIP phone. > >>> > > >>> > When the *3 digits are pushed on the analog phone I see the logs > report: > >>> > > >>> > ftdm_io.c:3463 [s2c1][1:4] Queuing DTMF * (debug = 0) > >>> > mod_freetdm.c:702 Queuing DTMF [*] in channel FreeTDM/2:1/ > >>> > > >>> > ftdm_io.c:3463 [s2c1][1:4] Queuing DTMF 4 (debug = 0) > >>> > > >>> > mod_freetdm.c:702 Queuing DTMF [4] in channel FreeTDM/2:1/ > >>> > > >>> > When the further keys are pushed ftdm_io reports nothing. > >>> > > >>> > I have tried inserting a start_dtmf before the read() but it had no > >>> > effect. > >>> > Any thoughts as to why DTMF isn't being seen from the analog phone > after > >>> > the read()? > >>> > > >>> > Cheers > >>> > Shane > >>> > > >>> > >>> > _________________________________________________________________________ > >>> Professional FreeSWITCH Consulting Services: > >>> consulting at freeswitch.org > >>> http://www.freeswitchsolutions.com > >>> > >>> > >>> > >>> > >>> Official FreeSWITCH Sites > >>> http://www.freeswitch.org > >>> http://wiki.freeswitch.org > >>> http://www.cluecon.com > >>> > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >> > >> > >> > >> > >> -- > >> Paragon Electronic Design Ltd > >> L6 Crest House > >> 92 Queens Drive > >> P0 Box 30449 > >> Lower Hutt 5040 > >> > >> +64 4 5703870 Extn 875 > >> +64 21 608919 (mobile) > >> > >> "Solving your problems with the right technology" > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Paragon Electronic Design Ltd L6 Crest House 92 Queens Drive P0 Box 30449 Lower Hutt 5040 +64 4 5703870 Extn 875 +64 21 608919 (mobile) "Solving your problems with the right technology" -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120510/2f0cfbf3/attachment.html From curriegrad2004 at gmail.com Thu May 10 06:44:13 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Wed, 9 May 2012 19:44:13 -0700 Subject: [Freeswitch-users] Freetdm DTMF In-Reply-To: References: Message-ID: I just had to laugh at my self for mixing up the 2 again... bind_meta_app is only applicable to that extension that the inbound call was made to. Since you've transferred to another extension, the bind_meta_app won't work anymore because it's not defined in the extension you're transferring to. If you want this to happen, you'd have to manually define that bind_meta_app to those target extensions too. Remember, do this at your own peril - obvious misuse of bind_meta_app can open a huge security hole if you don't know what you're doing :) On Wed, May 9, 2012 at 7:27 PM, Shane Harrison wrote: > Thanks for the thoughts.? As I said, I am already setting it to both legs - > I will try simply trying one leg but am sceptical :-) > > I also mentioned that I called the start_dtmf just before calling the read > so unless I am doing something wrong here.....? I'll try and post the XML > tonight when I get home.? Oh and it is an FXS card not an FXO of course > since it has a phone plugged into it. > > The question still remains though, why is the in-band DTMF detection working > for the bind_meta_app digit detection but not after that? > > Cheers > Shane > > > On Thu, May 10, 2012 at 12:28 PM, curriegrad2004 > wrote: >> >> and crap, since I wasn't even reading anything here, on the subsequent >> transfers from your FXO card, enable the in-band DTMF detector that FS >> has. The details on the in-band DTMF detector is here: >> >> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf >> >> But use this with caution, if there is a DTMF detector on the FXO card >> itself, make sure you disable it before using it. >> >> On Wed, May 9, 2012 at 5:26 PM, curriegrad2004 >> wrote: >> > apologies for telling you the wrong thing. I was replying you from my >> > phone btw :P >> > >> > Yeah, bind_meta_app is the app you would use, but try changing it to >> > point to the b-leg, not the a-leg >> > >> > On Wed, May 9, 2012 at 4:12 PM, Shane Harrison >> > wrote: >> >> Thanks.? I am currently using bind_meta_app (set to both legs) already >> >> rather than bind_digits.? I'll give bind_digits a shot and see if it >> >> behaves >> >> differently. >> >> >> >> Note that I do detect the initial *3 digits and because bind_meta_app >> >> is >> >> both legs, this is successful no matter which direction the call is >> >> setup >> >> from.? However once the dialplan moves to the extension the *3 is bound >> >> to, >> >> digits are no longer received. >> >> >> >> The worrying thing for me is that ftdm_io.c doesn't even write to the >> >> log >> >> that it has received them (nor freetdm above that of course which is >> >> understandable) and I am surprised that the read() influences that >> >> since it >> >> works prior on the *3 digits. >> >> >> >> Cheers >> >> Shane >> >> >> >> >> >> >> >> On Thu, May 10, 2012 at 10:22 AM, curriegrad2004 >> >> >> >> wrote: >> >>> >> >>> I'm guessing the bind digits in your analog card was set to listen for >> >>> this sequence on the a-leg given if the call was being routed from the >> >>> IP side to the analog side. >> >>> >> >>> Try changing that to listen on the b-leg. >> >>> >> >>> On 5/9/12, Shane Harrison wrote: >> >>> > Hi All, >> >>> > >> >>> > Have a situation where I have a call between a SIP phone and a >> >>> > FreeTDM >> >>> > channel. ? Pushing *3 on the analog FreeTDM phone is detected and >> >>> > this >> >>> > is >> >>> > bound to a dialplan extension (attended transfer) that does a >> >>> > read(): >> >>> > >> >>> > >> >>> > However pushing further digits on the analog phone ie. extension >> >>> > number >> >>> > of >> >>> > phone we wish to do an attended transfer to , doesn't result in the >> >>> > DTMF >> >>> > being detected. ?Note that this all works the other way around ie. >> >>> > using >> >>> > the SIP phone. >> >>> > >> >>> > When the *3 digits are pushed on the analog phone I see the logs >> >>> > report: >> >>> > >> >>> > ?ftdm_io.c:3463 [s2c1][1:4] Queuing DTMF * (debug = 0) >> >>> > mod_freetdm.c:702 Queuing DTMF [*] in channel FreeTDM/2:1/ >> >>> > >> >>> > ftdm_io.c:3463 [s2c1][1:4] Queuing DTMF 4 (debug = 0) >> >>> > >> >>> > mod_freetdm.c:702 Queuing DTMF [4] in channel FreeTDM/2:1/ >> >>> > >> >>> > When the further keys are pushed ftdm_io reports nothing. >> >>> > >> >>> > I have tried inserting a start_dtmf before the read() but it had no >> >>> > effect. >> >>> > Any thoughts as to why DTMF isn't being seen from the analog phone >> >>> > after >> >>> > the read()? >> >>> > >> >>> > Cheers >> >>> > Shane >> >>> > >> >>> >> >>> >> >>> _________________________________________________________________________ >> >>> Professional FreeSWITCH Consulting Services: >> >>> consulting at freeswitch.org >> >>> http://www.freeswitchsolutions.com >> >>> >> >>> >> >>> >> >>> >> >>> Official FreeSWITCH Sites >> >>> http://www.freeswitch.org >> >>> http://wiki.freeswitch.org >> >>> http://www.cluecon.com >> >>> >> >>> FreeSWITCH-users mailing list >> >>> FreeSWITCH-users at lists.freeswitch.org >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >>> >> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >>> http://www.freeswitch.org >> >> >> >> >> >> >> >> >> >> -- >> >> Paragon Electronic Design Ltd >> >> L6 Crest House >> >> 92 Queens Drive >> >> P0 Box 30449 >> >> Lower Hutt 5040 >> >> >> >> +64 4 5703870 Extn 875 >> >> +64 21 608919? (mobile) >> >> >> >> "Solving your problems with the right technology" >> >> >> >> >> >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> >> >> >> >> >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> http://wiki.freeswitch.org >> >> http://www.cluecon.com >> >> >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at 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 at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > > -- > Paragon Electronic Design Ltd > L6 Crest House > 92 Queens Drive > P0 Box 30449 > Lower Hutt 5040 > > +64 4 5703870 Extn 875 > +64 21 608919? (mobile) > > "Solving your problems with the right technology" > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From shane.harrison at paragon.co.nz Thu May 10 07:39:00 2012 From: shane.harrison at paragon.co.nz (Shane Harrison) Date: Thu, 10 May 2012 15:39:00 +1200 Subject: [Freeswitch-users] Freetdm DTMF In-Reply-To: References: Message-ID: Happy for you to laugh at yourself, I'm just happy you are finding some time to take some interest in my problem. Much appreciated. I think some clarity is required here. I am simply trying to do an attended transfer as per the wiki (except I set the bind_meta_app to both legs) http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_att_xfer It works fine for SIP to SIP. For SIP to Freetdm it only works if the SIP is trying to do the transfer. If the FreeTDM is trying to do it ie. push *3 on the phone keypad, then the bind_meta_app works fine and detects the *3 and executes the appropriate extension ie. att_xfer, however the read() in the extension att_xfer does not see the subsequent DTMF pressed on the phone ie. the destination extension number. Cheers Shane On Thu, May 10, 2012 at 2:44 PM, curriegrad2004 wrote: > I just had to laugh at my self for mixing up the 2 again... > bind_meta_app is only applicable to that extension that the inbound > call was made to. Since you've transferred to another extension, the > bind_meta_app won't work anymore because it's not defined in the > extension you're transferring to. > > If you want this to happen, you'd have to manually define that > bind_meta_app to those target extensions too. Remember, do this at > your own peril - obvious misuse of bind_meta_app can open a huge > security hole if you don't know what you're doing :) > > On Wed, May 9, 2012 at 7:27 PM, Shane Harrison > wrote: > > Thanks for the thoughts. As I said, I am already setting it to both > legs - > > I will try simply trying one leg but am sceptical :-) > > > > I also mentioned that I called the start_dtmf just before calling the > read > > so unless I am doing something wrong here..... I'll try and post the XML > > tonight when I get home. Oh and it is an FXS card not an FXO of course > > since it has a phone plugged into it. > > > > The question still remains though, why is the in-band DTMF detection > working > > for the bind_meta_app digit detection but not after that? > > > > Cheers > > Shane > > > > > > On Thu, May 10, 2012 at 12:28 PM, curriegrad2004 < > curriegrad2004 at gmail.com> > > wrote: > >> > >> and crap, since I wasn't even reading anything here, on the subsequent > >> transfers from your FXO card, enable the in-band DTMF detector that FS > >> has. The details on the in-band DTMF detector is here: > >> > >> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf > >> > >> But use this with caution, if there is a DTMF detector on the FXO card > >> itself, make sure you disable it before using it. > >> > >> On Wed, May 9, 2012 at 5:26 PM, curriegrad2004 < > curriegrad2004 at gmail.com> > >> wrote: > >> > apologies for telling you the wrong thing. I was replying you from my > >> > phone btw :P > >> > > >> > Yeah, bind_meta_app is the app you would use, but try changing it to > >> > point to the b-leg, not the a-leg > >> > > >> > On Wed, May 9, 2012 at 4:12 PM, Shane Harrison > >> > wrote: > >> >> Thanks. I am currently using bind_meta_app (set to both legs) > already > >> >> rather than bind_digits. I'll give bind_digits a shot and see if it > >> >> behaves > >> >> differently. > >> >> > >> >> Note that I do detect the initial *3 digits and because bind_meta_app > >> >> is > >> >> both legs, this is successful no matter which direction the call is > >> >> setup > >> >> from. However once the dialplan moves to the extension the *3 is > bound > >> >> to, > >> >> digits are no longer received. > >> >> > >> >> The worrying thing for me is that ftdm_io.c doesn't even write to the > >> >> log > >> >> that it has received them (nor freetdm above that of course which is > >> >> understandable) and I am surprised that the read() influences that > >> >> since it > >> >> works prior on the *3 digits. > >> >> > >> >> Cheers > >> >> Shane > >> >> > >> >> > >> >> > >> >> On Thu, May 10, 2012 at 10:22 AM, curriegrad2004 > >> >> > >> >> wrote: > >> >>> > >> >>> I'm guessing the bind digits in your analog card was set to listen > for > >> >>> this sequence on the a-leg given if the call was being routed from > the > >> >>> IP side to the analog side. > >> >>> > >> >>> Try changing that to listen on the b-leg. > >> >>> > >> >>> On 5/9/12, Shane Harrison wrote: > >> >>> > Hi All, > >> >>> > > >> >>> > Have a situation where I have a call between a SIP phone and a > >> >>> > FreeTDM > >> >>> > channel. Pushing *3 on the analog FreeTDM phone is detected and > >> >>> > this > >> >>> > is > >> >>> > bound to a dialplan extension (attended transfer) that does a > >> >>> > read(): > >> >>> > > >> >>> > > >> >>> > However pushing further digits on the analog phone ie. extension > >> >>> > number > >> >>> > of > >> >>> > phone we wish to do an attended transfer to , doesn't result in > the > >> >>> > DTMF > >> >>> > being detected. Note that this all works the other way around ie. > >> >>> > using > >> >>> > the SIP phone. > >> >>> > > >> >>> > When the *3 digits are pushed on the analog phone I see the logs > >> >>> > report: > >> >>> > > >> >>> > ftdm_io.c:3463 [s2c1][1:4] Queuing DTMF * (debug = 0) > >> >>> > mod_freetdm.c:702 Queuing DTMF [*] in channel FreeTDM/2:1/ > >> >>> > > >> >>> > ftdm_io.c:3463 [s2c1][1:4] Queuing DTMF 4 (debug = 0) > >> >>> > > >> >>> > mod_freetdm.c:702 Queuing DTMF [4] in channel FreeTDM/2:1/ > >> >>> > > >> >>> > When the further keys are pushed ftdm_io reports nothing. > >> >>> > > >> >>> > I have tried inserting a start_dtmf before the read() but it had > no > >> >>> > effect. > >> >>> > Any thoughts as to why DTMF isn't being seen from the analog phone > >> >>> > after > >> >>> > the read()? > >> >>> > > >> >>> > Cheers > >> >>> > Shane > >> >>> > > >> >>> > >> >>> > >> >>> > _________________________________________________________________________ > >> >>> Professional FreeSWITCH Consulting Services: > >> >>> consulting at freeswitch.org > >> >>> http://www.freeswitchsolutions.com > >> >>> > >> >>> > >> >>> > >> >>> > >> >>> Official FreeSWITCH Sites > >> >>> http://www.freeswitch.org > >> >>> http://wiki.freeswitch.org > >> >>> http://www.cluecon.com > >> >>> > >> >>> FreeSWITCH-users mailing list > >> >>> FreeSWITCH-users at lists.freeswitch.org > >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >>> > >> >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> >>> http://www.freeswitch.org > >> >> > >> >> > >> >> > >> >> > >> >> -- > >> >> Paragon Electronic Design Ltd > >> >> L6 Crest House > >> >> 92 Queens Drive > >> >> P0 Box 30449 > >> >> Lower Hutt 5040 > >> >> > >> >> +64 4 5703870 Extn 875 > >> >> +64 21 608919 (mobile) > >> >> > >> >> "Solving your problems with the right technology" > >> >> > >> >> > >> >> > _________________________________________________________________________ > >> >> Professional FreeSWITCH Consulting Services: > >> >> consulting at freeswitch.org > >> >> http://www.freeswitchsolutions.com > >> >> > >> >> > >> >> > >> >> > >> >> Official FreeSWITCH Sites > >> >> http://www.freeswitch.org > >> >> http://wiki.freeswitch.org > >> >> http://www.cluecon.com > >> >> > >> >> FreeSWITCH-users mailing list > >> >> FreeSWITCH-users at 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 at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > > > > > -- > > Paragon Electronic Design Ltd > > L6 Crest House > > 92 Queens Drive > > P0 Box 30449 > > Lower Hutt 5040 > > > > +64 4 5703870 Extn 875 > > +64 21 608919 (mobile) > > > > "Solving your problems with the right technology" > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Paragon Electronic Design Ltd L6 Crest House 92 Queens Drive P0 Box 30449 Lower Hutt 5040 +64 4 5703870 Extn 875 +64 21 608919 (mobile) "Solving your problems with the right technology" -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120510/0182e13b/attachment-0001.html From freeswitch-list at necosec.de Thu May 10 09:50:36 2012 From: freeswitch-list at necosec.de (NeCoSec - Ulrich Schinz) Date: Thu, 10 May 2012 07:50:36 +0200 Subject: [Freeswitch-users] Howto? Migrating from Siemens HiPath4000 to FreeSWITCH Message-ID: <4FAB572C.20508@necosec.de> Hi FS-List, maybe there are some experts in here, that can help me to get some things clear. I'm working for an institution with about 500 extensions. At the moment there is a Siemens HiPath4000 used. It is an old version of the Software, we are using (3.x). We are paying a lot of money for that solution(rental fee/leasing), so I'm planning to move to another option. The building is an old monestary and the infrastructure (e.g. cables) is very old. So we are planning to get the Infrastructure up to cat7 cables. On the one hand, because we need to get most parts of the building online (internet for guests and conferences) and on the other hand it makes no sense to keep a 70-80 years old cabeling up. So with this in mind VoIP would be a good option.... To bring the cabeling up to date, we're planning to do it part by part. It's not a realistic option to get this done in a short time. The building is a very large one, so this has to be done in maybe 3-5 years... We are using a Telekom PRI/E1 (german Telekom). So my plan is: Setting up a FreeSWITCH server with 2 PRI cards (e.g. Sangoma A101D), one card connecting to the PRI of Telekom and the other card connecting to the HiPath4000. In this way I hope to get the calls connected through FreeSWITCH to the HiPath4000 and I can start getting ip-phones connected, where cat7-cables are available. In this way I have some advantages: - I can handle a smooth change to VoIP (don't have to have a hard cut) - I first can integrate open-minded people to the VoIP-technologie, so they can test and give feedback, what they like, and what they wanna have better... Older or not so open-minded people (we have some really old monks and padres) would then benfit from a tested environment. The switches are planed to have QoS and PoE. As phones I'm planning to use Snom-phones. (Snom300 and Snom320) Further I'd like to build a failover system, with a second server. Another option would have been to upgrade the HiPath to VoIP, but the costs for that are in no relation to what a FreeSWITCH System would cost... My experience with FreeSWITCH is limited to quite small enviroments and I never tried it with PRI (always using "normal" ISDN base, Sangoma a500...) My experience with HiPath4000 is limited to the available webfrontend. The "internal configuration" is done by the provider of that system. My experience with Linux and networksystems is about 12 years in small an large systems... In two years the HiPath is "payed" and it's ours. So we can use this system with no costs and have full access. My questions: - Do you think this is an good way to get it running? - Maybe someone with some experience in HiPath knows, whether it's complicated, to get the HiPath configured (getting extension-blocks handled by FreeSWITCH and on the other side keep other extensions on the HiPath)? - Is the routing with this 2 PRI-cards construct possible, or better easy done? - Is there a way to simulate a PRI-Provider. I.e. if I'm buying two PRI-cards I maybe could configure two test-boxes, each having one card. Connecting these two I maybe can play around with these PRI connections (just to get a feeling for it...)? - Do you have some hints, what to consider, if I'm doing this? I'm planning this for quite a while, and im thinking a lot about this, but I always have the feeling of forgetting some essential things and the whole construct blows up, if im starting it. This would be not very optimal ;) - Maybe someone has some hints, in which order to plan this setup. The plans become more and more concrete and I'm not sure, like said before, whether I'm doing and thinking things in right order. - Just a technical detail: the HiPath is in a room, that is not optimal for this kind of it-systems (aircondition, accessible for everybody). But the patchpanels are in that room (the old ones, old cables, 2 poles for HiPath-Protocol, parts of it with 4 poles for ISDN). And the PRI-connection to the Telekom is in this room. Would it be possible to place the servers in another room and get the PRI there? Would the cable-length be a problem (I think about 50 meters)? And for the failover system it would be optimal to have the system in another part of the building (in case of fire another place is better). There it would be nice to "patch" the PRI to another place... - As the building is very large, we have some fibrechanel connections to get the long distances bridged. To keep QoS up: Do I have to have special converters? (I don't think so, cause the switches should do that, but maybe I'm wrong). - Cascading switches (Phone -> Switch -> Fibre -> Switch -> Server): Does every switch have to do QoS, or would the first switch be enough. Or the other way around, would cascaded switches make problems, if both have QoS configuration...? Sorry for this very long post and so many questions.... Kind regards Uli From chris at opencsta.org Thu May 10 10:07:29 2012 From: chris at opencsta.org (Chris Mylonas) Date: Thu, 10 May 2012 16:07:29 +1000 Subject: [Freeswitch-users] Howto? Migrating from Siemens HiPath4000 to FreeSWITCH In-Reply-To: <4FAB572C.20508@necosec.de> References: <4FAB572C.20508@necosec.de> Message-ID: <01907D92-8A04-4388-B052-8DB5901F4F5C@opencsta.org> Hi Uli 1. The HP4k will always be ISDN slave - so on the FS side, you'll need to be the timing source i.e. pretend to be the telco from that ISDN port - it should simply be one signalling setting - I'm not familiar with FS ISDN stuff. On the other port, the German ISDN provider will no doubt be the timing source. 2. Depending on your indial configuration, but it sounds like you have a good plan. Setting up FS in front of the HP4k so that those areas that are cabled first can be handled by FS/IP-phones, whilst those that are not can be routed to HP4k. This is good. 3. Your experience with linux is handy - security is important. Keep your phones on a separate (V)LAN to mitigate risks. 4. Get hardware echo cancellation on the ISDN card, don't try and save $500 by leaving this out - it will be the first thing users complain about if you forego. 5. The management of your HP4k is costly and cumbersome, with FS or any other F/OSS system, you have a multitude of choices. Not sure about your QoS ethernet -> fibre -> ethernet query due to lack of experience. Similarly for ISDN cabling distances, but 50m should not be a problem. I'm sure there are settings to warn the card/software of how far you are from your next "hop". Sounds like a really good project! Good luck! Chris On 10/05/2012, at 3:50 PM, NeCoSec - Ulrich Schinz wrote: > Hi FS-List, > > maybe there are some experts in here, that can help me to get some > things clear. > > I'm working for an institution with about 500 extensions. At the moment > there is a Siemens HiPath4000 used. It is an old version > of the Software, we are using (3.x). We are paying a lot of money for > that solution(rental fee/leasing), so I'm planning to move > to another option. > > The building is an old monestary and the infrastructure (e.g. cables) is > very old. So we are planning to get the Infrastructure > up to cat7 cables. On the one hand, because we need to get most parts of > the building online (internet for guests and conferences) > and on the other hand it makes no sense to keep a 70-80 years old > cabeling up. So with this in mind VoIP would be a good option.... > > To bring the cabeling up to date, we're planning to do it part by part. > It's not a realistic option to get this done in a short time. > The building is a very large one, so this has to be done in maybe 3-5 > years... > > We are using a Telekom PRI/E1 (german Telekom). > > So my plan is: > Setting up a FreeSWITCH server with 2 PRI cards (e.g. Sangoma A101D), > one card connecting to the PRI of Telekom and the other card > connecting to the HiPath4000. In this way I hope to get the calls > connected through FreeSWITCH to the HiPath4000 and I can start > getting ip-phones connected, where cat7-cables are available. > In this way I have some advantages: > - I can handle a smooth change to VoIP (don't have to have a hard cut) > - I first can integrate open-minded people to the VoIP-technologie, so > they can test and give feedback, what they like, and what > they wanna have better... Older or not so open-minded people (we have > some really old monks and padres) would then benfit from > a tested environment. > > The switches are planed to have QoS and PoE. > > As phones I'm planning to use Snom-phones. (Snom300 and Snom320) > > Further I'd like to build a failover system, with a second server. > > Another option would have been to upgrade the HiPath to VoIP, but the > costs for that are in no relation to what a FreeSWITCH System > would cost... > > My experience with FreeSWITCH is limited to quite small enviroments and > I never tried it with PRI (always using "normal" ISDN base, > Sangoma a500...) > My experience with HiPath4000 is limited to the available webfrontend. > The "internal configuration" is done by the provider of that > system. > My experience with Linux and networksystems is about 12 years in small > an large systems... > > In two years the HiPath is "payed" and it's ours. So we can use this > system with no costs and have full access. > > My questions: > > - Do you think this is an good way to get it running? > > - Maybe someone with some experience in HiPath knows, whether it's > complicated, to get the HiPath configured (getting extension-blocks > handled by FreeSWITCH and on the other side keep other extensions on the > HiPath)? > > - Is the routing with this 2 PRI-cards construct possible, or better > easy done? > > - Is there a way to simulate a PRI-Provider. I.e. if I'm buying two > PRI-cards I maybe could configure two test-boxes, each having one > card. Connecting these two I maybe can play around with these PRI > connections (just to get a feeling for it...)? > > - Do you have some hints, what to consider, if I'm doing this? I'm > planning this for quite a while, and im thinking a lot about this, > but I always have the feeling of forgetting some essential things and > the whole construct blows up, if im starting it. This would be > not very optimal ;) > > - Maybe someone has some hints, in which order to plan this setup. The > plans become more and more concrete and I'm not sure, like > said before, whether I'm doing and thinking things in right order. > > - Just a technical detail: the HiPath is in a room, that is not optimal > for this kind of it-systems (aircondition, accessible for everybody). > But the patchpanels are in that room (the old ones, old cables, 2 poles > for HiPath-Protocol, parts of it with 4 poles for ISDN). And the > PRI-connection to the Telekom is in this room. Would it be possible to > place the servers in another room and get the PRI there? > Would the cable-length be a problem (I think about 50 meters)? > And for the failover system it would be optimal to have the system in > another part of the building (in case of fire another place is better). > There it would be nice to "patch" the PRI to another place... > > - As the building is very large, we have some fibrechanel connections to > get the long distances bridged. To keep QoS up: Do I have to > have special converters? (I don't think so, cause the switches should do > that, but maybe I'm wrong). > > - Cascading switches (Phone -> Switch -> Fibre -> Switch -> Server): > Does every switch have to do QoS, or would the first switch be > enough. Or the other way around, would cascaded switches make problems, > if both have QoS configuration...? > > Sorry for this very long post and so many questions.... > > Kind regards > Uli > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From miha at softnet.si Thu May 10 10:16:41 2012 From: miha at softnet.si (Miha) Date: Thu, 10 May 2012 08:16:41 +0200 Subject: [Freeswitch-users] Call intercept In-Reply-To: References: <4FA916DA.4020201@softnet.si> <4FAA5193.6020608@softnet.si> Message-ID: <4FAB5D49.9060101@softnet.si> I agree with you. I have just fallowed wiki steps (http://wiki.freeswitch.org/wiki/User:Agx#Call_Pickup). But this was made when only one FS server way up, after I start other FS server and do load_balacing between them with opensips this do not work. Does anyone know what is the best way to deal whit this (one call is active on fs1 and other on fs2, how call intercept can be made between to FS servers)? Thanks! Miha On 5/9/2012 1:54 PM, Jo?o Mesquita wrote: > > Care to share the solution for future reference? The mailing list is > always indexed by Google or whatnot ... > > On May 9, 2012 8:16 AM, "Miha" > wrote: > > HI, > > just to let you know I figure it out:) > > thanks any way! > > Regards, > Miha > > On 5/8/2012 2:51 PM, Miha wrote: >> Hi, >> >> I need a little help about call intercept. I have read wiki >> (http://wiki.freeswitch.org/wiki/Callgroup_intercept) but still >> having few problems. >> >> 1. I have Opensips which works like load_balancer and registrar. >> Behind Opensips I have two FS server, which are for load_balacing. >> 2. I need to set a call intercept thing. I done it like it is >> written on FS wiki but this does not work as I have Opensips. >> >> I have put this in public dialplan: >> >> >> >> >> After I try to intercept the call this fails as in database is written different UUID. >> >> CAll goes like this: >> >> Phone-openips-FS-Openips-Phone >> >> Thank you for all your help! >> >> MIha >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120510/3530ec67/attachment-0001.html From joohny at mail.ru Thu May 10 10:40:26 2012 From: joohny at mail.ru (=?UTF-8?B?0JXQstCz0LXQvdC40Lk=?=) Date: Thu, 10 May 2012 10:40:26 +0400 Subject: [Freeswitch-users] =?utf-8?q?Recording_all_calls_with_good_qualit?= =?utf-8?q?y=2E?= Message-ID: Hi. I see you are recording in stereo. Both channels has??horrible?quality? I had?horrible?quality in one channel when it was two interfaces looking in the same network. Evginey I've tried the following codec settings, set in vars.xml. The first and third lines are the defaults upon install. My CPU load shows fine, even with 2 clients connected to an external (secondary) switch, through this one. I truly appreciate the help. Brandon McGinty-Carroll On Sun, May 06, 2012 at 06:50:44PM +0300, Avi Marcus wrote: > AFAIK, gsm is lossy, which is why it doesn't sound good. > wav, however, should sound great. What codec are your calls in? Are you > recording many calls? > Check the load average while you are recording. If it's greater than the > number of CPUs then probably your hard drives can't keep up. There are > several options then, first let see if we can diagnose the issue. >? > -Avi >? >? > On Sun, May 6, 2012 at 5:46 PM, wrote: >? > > Good day. > > I'm trying to record all calls sent through my Freeswitch PBX. > > The problem is that the quality is horrible. I've tried wav and GSM > > formats, and both are les than satisfactory. > > I'm looking for something around the quality of the recent recordings of > > the Freeswitch conference. (I checked on the wiki, and couldn't find > > anything through search.) > > Any information on the FS conference format or my configuration would be > > appreciated. > > I've got: > > > > > > > > > > > > > data="$${base_dir}/recordings/${strftime(%Y-%m-%d-%H-%M-%S)}_${destination_number}_${caller_id_number}.wav"/> > > > > > > Sincerely, > > Brandon McGinty-Carroll > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > >?consulting at freeswitch.org > >?http://www.freeswitchsolutions.com > > > > > >? > > > > Official FreeSWITCH Sites > >?http://www.freeswitch.org > >?http://wiki.freeswitch.org > >?http://www.cluecon.com > > > > FreeSWITCH-users mailing list > >?FreeSWITCH-users at 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 at freeswitch.org >?http://www.freeswitchsolutions.com >? > >? >? > Official FreeSWITCH Sites >?http://www.freeswitch.org >?http://wiki.freeswitch.org >?http://www.cluecon.com >? > FreeSWITCH-users mailing list >?FreeSWITCH-users at lists.freeswitch.org >?http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >?http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120510/8cf13c06/attachment-0001.html From khorsmann at gmail.com Thu May 10 11:10:47 2012 From: khorsmann at gmail.com (Karsten Horsmann) Date: Thu, 10 May 2012 09:10:47 +0200 Subject: [Freeswitch-users] Howto? Migrating from Siemens HiPath4000 to FreeSWITCH In-Reply-To: <01907D92-8A04-4388-B052-8DB5901F4F5C@opencsta.org> References: <4FAB572C.20508@necosec.de> <01907D92-8A04-4388-B052-8DB5901F4F5C@opencsta.org> Message-ID: Hi Uli, yes of course you can configure Sangoma PRI Cards in NT Mode. Sangoma calls that CPE (TE) or NET (NT) Mode. Take a look at Btw. you can also buy an A102d with two PRI Ports. The "d" stands for echo-chancelation. Or you get an Mediagateway like the Patton Inalp SmartNode 4960. -- Mit freundlichen Gr??en *Karsten Horsmann* From markus.lindenberg at gmail.com Thu May 10 11:26:07 2012 From: markus.lindenberg at gmail.com (Markus Lindenberg) Date: Thu, 10 May 2012 09:26:07 +0200 Subject: [Freeswitch-users] Howto? Migrating from Siemens HiPath4000 to FreeSWITCH In-Reply-To: References: <4FAB572C.20508@necosec.de> <01907D92-8A04-4388-B052-8DB5901F4F5C@opencsta.org> Message-ID: Hi, On Thu, May 10, 2012 at 9:10 AM, Karsten Horsmann wrote: > > Btw. you can also buy an A102d with two PRI Ports. The "d" stands for > echo-chancelation. > > Or you get an Mediagateway like the Patton Inalp SmartNode 4960. I would also recommend to use a SmartNode or PRI board with two ports. This way you can sync the internal (HiPath facing) port to the external (Telekom) port, which will help avoid all kinds of timing trouble. Some single port boards have proprietary interconnects to sync multiple boards of the same vendor, but by buying one card / gateway with two PRIs you can skip the complexity. Using a standalone mediagateway like the Patton also makes it possible to put your server wherever you like and not worry about cabling the PRIs to the rack. You can route calls to multiple servers (e.g. send calls to server2 if server1 is down etc.) and don't depend on the FreeSWITCH machines to be up and running for the HiPath to work. We're using this kind of migration setup quite often, starting with a "transparent" SmartNode looped in between PRI and "old" PBX, then gradually routing numbers to the SIP world. Gru?, Markus From chris at opencsta.org Thu May 10 11:40:41 2012 From: chris at opencsta.org (Chris Mylonas) Date: Thu, 10 May 2012 17:40:41 +1000 Subject: [Freeswitch-users] Howto? Migrating from Siemens HiPath4000 to FreeSWITCH In-Reply-To: References: <4FAB572C.20508@necosec.de> <01907D92-8A04-4388-B052-8DB5901F4F5C@opencsta.org> Message-ID: <827E721D-9E3B-45C6-AA2E-B55541F03CFE@opencsta.org> Hi List, The only thing against an external device is that they can become EOL - i.e. SIP stack does not get further updates. Take a Linksys SPA400 for example (4 port FXO). This has now been superseeded by a Cisco 8800 (4xFXO + 4xFXS) An internal card may get longer software updates. Network security for the external device nearly makes this a null point. If the server that has the card dies, then it's a PITA to move the card to the back up box. Unless you have the backup box with an ISDN card as well. There are devices that do automatic switching on failure (I believe Xorcom had one that runs off USB). That's something that is often overlooked. I would go the external device because of convenience - and if the EOL thing is a problem, buy another one in 5 years or whenever. My 2 cents, CM On 10/05/2012, at 5:26 PM, Markus Lindenberg wrote: > Hi, > > On Thu, May 10, 2012 at 9:10 AM, Karsten Horsmann wrote: >> >> Btw. you can also buy an A102d with two PRI Ports. The "d" stands for >> echo-chancelation. >> >> Or you get an Mediagateway like the Patton Inalp SmartNode 4960. > > I would also recommend to use a SmartNode or PRI board with two ports. > This way you can sync the internal (HiPath facing) port to the > external (Telekom) port, which will help avoid all kinds of timing > trouble. Some single port boards have proprietary interconnects to > sync multiple boards of the same vendor, but by buying one card / > gateway with two PRIs you can skip the complexity. > > Using a standalone mediagateway like the Patton also makes it possible > to put your server wherever you like and not worry about cabling the > PRIs to the rack. You can route calls to multiple servers (e.g. send > calls to server2 if server1 is down etc.) and don't depend on the > FreeSWITCH machines to be up and running for the HiPath to work. > > We're using this kind of migration setup quite often, starting with a > "transparent" SmartNode looped in between PRI and "old" PBX, then > gradually routing numbers to the SIP world. > > Gru?, Markus > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From vitaliy.davudov at vts24.ru Thu May 10 11:43:26 2012 From: vitaliy.davudov at vts24.ru (=?UTF-8?B?0JLQuNGC0LDQu9C40Lkg0JTQsNCy0YPQtNC+0LI=?=) Date: Thu, 10 May 2012 11:43:26 +0400 Subject: [Freeswitch-users] Variable continue on fail In-Reply-To: References: <4F87EF60.3070105@vts24.ru> <4F8822A5.9090402@vts24.ru> <4FA4D2D6.1010908@vts24.ru> Message-ID: <4FAB719E.9010800@vts24.ru> Hi, Michael! I added a new post on pastebin: http://pastebin.freeswitch.org/19026 07.05.2012 5:14, Michael Collins ???????: > I don't recall you ever posting a console debug log to > pastebin.freeswitch.org . Could you > please do that? It will help us to know exactly what is happening. > -MC > > On Sat, May 5, 2012 at 12:12 AM, ??????? ??????? > > wrote: > > Hi! > Is there any news on this issue? > > 13.04.2012 16:57, ??????? ??????? ???????: >> Yes, you are right! >> >> I did it: http://pastebin.freeswitch.org/18863 >> >> Additionally: >> I've included in this extension new line: >> >> >> >> >> ** >> > data="continue_on_fail=1,2,3,6,25,34,38,41,42,44,47,63,66,500,501"/> >> >> >> >> >> >> >> Without that line a similar situation occurs if FS recieve >> /NORMAL_CLEARING./ >> >> 13.04.2012 13:44, Anton Kvashenkin ???????: >>> Ok, i got it. Even that there is no USER_BUSY at >>> continue_on_fail variable, FS still tries to reach the second >>> action, am i right? >>> >>> So, for better debugging, i suggest to paste full call log with >>> enabled siptrace and /log 7 to pastebin.freeswitch.org >>> . >>> >>> 13 ?????? 2012 ?. 13:18 ???????????? ??????? ??????? >>> > >>> ???????: >>> >>> Hi all! >>> >>> In my dialplan I've included variable continue on fail: >>> >>> >>> >>> >>> >> data="continue_on_fail=1,2,3,6,25,34,38,41,42,44,47,63,66,500,501"/> >>> >>> >>> >>> >>> >>> >>> And if FS recieve from first gateway USER_BUSY, then FS try >>> to bridge >>> this call to another gateway. Although in line >> application="set" >>> data="continue_on_fail=1,2,3,6,25,34,38,41,42,44,47,63,66,500,501"/> >>> there is no code Q.850 = 17. >>> How resolve this issue? >>> >>> -- >>> Best regards, >>> Vitaly Davudov >>> "VIP-TELECOM-SERVICE" Ltd. >>> ("ETERIA" Group of companies) >>> http://www.vts24.ru >>> >>> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- ? ?????????? ???????????, ??????? ??????? ????????? ??? "???-???????-??????" (?????? ???????? "ETERIA") http://www.vts24.ru ???: (495) 989-47-00 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120510/4083a6ec/attachment.html From markus.lindenberg at gmail.com Thu May 10 12:44:14 2012 From: markus.lindenberg at gmail.com (Markus Lindenberg) Date: Thu, 10 May 2012 10:44:14 +0200 Subject: [Freeswitch-users] Howto? Migrating from Siemens HiPath4000 to FreeSWITCH In-Reply-To: <827E721D-9E3B-45C6-AA2E-B55541F03CFE@opencsta.org> References: <4FAB572C.20508@necosec.de> <01907D92-8A04-4388-B052-8DB5901F4F5C@opencsta.org> <827E721D-9E3B-45C6-AA2E-B55541F03CFE@opencsta.org> Message-ID: Hi Chris, same here, convenience wins. I love the fact that the complete SmartNode configuration can be backed up as one readable configuration file. firmware up/downgrades and configuration backups are a matter of minutes and can always be safely rolled back or transferred. This is in contrast to Dialogic Diva BRI/PRI cards, which are pretty awesome but rely on a mess of packages, daemons and kernel modules compiled in place on the target system for the target kernel. So by just updating your server you could possibly break stuff and you also have to make sure that your OS is supported. Then there's things like the beroFix (never used them) which are self contained SIP/ISDN gateways on a PCI card. By buying "enterprise", you can migitate the possibility of your gateway/card going EOL. That's why Audiocodes' gateways are expensive compared to Linksys stuff. SmartNode development has always been active even for existing devices, i'm just a bit worried that there's no IPv6 support available or announced. On Thu, May 10, 2012 at 9:40 AM, Chris Mylonas wrote: > Hi List, > > The only thing against an external device is that they can become EOL - i.e. SIP stack does not get further updates. ?Take a Linksys SPA400 for example (4 port FXO). ?This has now been superseeded by a Cisco 8800 (4xFXO + 4xFXS) > An internal card may get longer software updates. > Network security for the external device nearly makes this a null point. > > If the server that has the card dies, then it's a PITA to move the card to the back up box. ?Unless you have the backup box with an ISDN card as well. ?There are devices that do automatic switching on failure (I believe Xorcom had one that runs off USB). > > > That's something that is often overlooked. > I would go the external device because of convenience - and if the EOL thing is a problem, buy another one in 5 years or whenever. > > My 2 cents, > CM > > > > > On 10/05/2012, at 5:26 PM, Markus Lindenberg wrote: > >> Hi, >> >> On Thu, May 10, 2012 at 9:10 AM, Karsten Horsmann wrote: >>> >>> Btw. you can also buy an A102d with two PRI Ports. The "d" stands for >>> echo-chancelation. >>> >>> Or you get an Mediagateway like the Patton Inalp SmartNode 4960. >> >> I would also recommend to use a SmartNode or PRI board with two ports. >> This way you can sync the internal (HiPath facing) port to the >> external (Telekom) port, which will help avoid all kinds of timing >> trouble. Some single port boards have ?proprietary interconnects to >> sync multiple boards of the same vendor, but by buying one card / >> gateway with two PRIs you can skip the complexity. >> >> Using a standalone mediagateway like the Patton also makes it possible >> to put your server wherever you like and not worry about cabling the >> PRIs to the rack. You can route calls to multiple servers (e.g. send >> calls to server2 if server1 is down etc.) and don't depend on the >> FreeSWITCH machines to be up and running for the HiPath to work. >> >> We're using this kind of migration setup quite often, starting with a >> "transparent" SmartNode looped in between PRI and "old" PBX, then >> gradually routing numbers to the SIP world. >> >> Gru?, Markus >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From nasida at live.ru Thu May 10 13:40:06 2012 From: nasida at live.ru (Yuriy Nasida) Date: Thu, 10 May 2012 13:40:06 +0400 Subject: [Freeswitch-users] productivity of FS In-Reply-To: References: , <94F2FEE465A940E8A3172DED85F24A91@freeswitch.org>, , Message-ID: Brian, Do you mean 1000-3000 calls with static xml directory\dialplan or with xml_mod_curl + php and lua + mysql ? In general I can move mysql + apache to the separate server but the highest cpu usage put the FS process not mysql+apache. Also do anybody know any tool for understanding which module of FS put highest cpu usage? Please advise.Thanks. Date: Mon, 7 May 2012 16:29:08 -0400 From: bdfoster at endigotech.com To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] productivity of FS What you've asked has a long and complex answer. It depends on several things. 200-250 concurrent calls isn't unheard of, people in this very community have done 1000-3000 calls easily. I'd re-read what Joao has asked. By the way having other things like MySQL or a webserver running on the server can severely restrict what you can handle as far as freeswitch is concerned. You have a finite amount of system resources available to you, no matter what you decide to use the server for. -BDF On May 7, 2012 4:06 PM, "Yuriy Nasida" wrote: I mean 200-250 concurrent calls. Please let me know what information from me can help as well. Thanks. Date: Mon, 7 May 2012 16:40:17 -0300 From: jmesquita at freeswitch.org To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] productivity of FS There is so much information missing on your problem description that I don't even know where to start. When you say load is high, how high? Do you have an exact number? Regards, -- Jo?o MesquitaSent with Sparrow On Monday, May 7, 2012 at 4:37 PM, Yuriy Nasida wrote: Hello guys, I have FS box with 200-250 concurrent calls. FS git 05/03/2012. Intel(R) Xeon(R) CPU E5620 @ 2.40GHzMemTotal: 12187540 kB I use xml_mod_curl + php for dynamic directory\dialplan. Also I use lua scripts + mysql for some cases. I use sip options ping for some checking of productivity.Sometimes (when loading is high) I have big delay with sip ping (>2 sec). The box which sends sip ping in same network with main box. My question:1) Is it many concurrent calls for setup I have ?2) If not, what the best way for understanding where I lose productivity ? Please advise. Thanks. _________________________________________________________________________Professional FreeSWITCH Consulting Services:consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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 at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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 at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120510/10a26865/attachment.html From freeswitch-list at puzzled.xs4all.nl Thu May 10 14:33:03 2012 From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists) Date: Thu, 10 May 2012 12:33:03 +0200 Subject: [Freeswitch-users] Howto? Migrating from Siemens HiPath4000 to FreeSWITCH In-Reply-To: References: <4FAB572C.20508@necosec.de> <01907D92-8A04-4388-B052-8DB5901F4F5C@opencsta.org> <827E721D-9E3B-45C6-AA2E-B55541F03CFE@opencsta.org> Message-ID: <4FAB995F.2060609@puzzled.xs4all.nl> On 10-05-12 10:44, Markus Lindenberg wrote: [snip] > This is in contrast to Dialogic Diva BRI/PRI cards, which are pretty > awesome but rely on a mess of packages, daemons and kernel modules > compiled in place on the target system for the target kernel. So by > just updating your server you could possibly break stuff and you also > have to make sure that your OS is supported. I agree that the Dialogic Diva cards are great. I've been using them for many years and they have always worked flawlessly. Like the Sangoma cards I might add. Afaik there is no channel driver for Diva cards in FreeSWITCH (Dialogic: create one!) so in relation to FreeSWITCH it would unfortunately not be an option. There is chan_capi in the Asterisk world which works great. Regarding the possibility of breaking stuff: Diva software works fine on both RHEL6 and CentOS6. What else do you need in the Enterprise? :) All Diva software is located in /usr/lib/eicon (also on 64bit...) and in my experience the kernel modules work fine on newer kernels due to ABI compatibility. I have never seen things break on target systems because if you create an RPM then the Diva software can easily be installed, updated or removed. I only install the bare minimum (scripts and firmware and an init script) so no daemons or multiple packages and the kernel modules are RPM packaged on a separate build system using the proper way instead of through some 10,000 line kitchen sink build script that requires to be run as root. If you don't want to worry about all the moving parts then a Smartnode, Audiocodes, BeroNet or Redfone box are definitely options to look at. Regards, Patrick From markus.lindenberg at gmail.com Thu May 10 14:53:56 2012 From: markus.lindenberg at gmail.com (Markus Lindenberg) Date: Thu, 10 May 2012 12:53:56 +0200 Subject: [Freeswitch-users] Howto? Migrating from Siemens HiPath4000 to FreeSWITCH In-Reply-To: <4FAB995F.2060609@puzzled.xs4all.nl> References: <4FAB572C.20508@necosec.de> <01907D92-8A04-4388-B052-8DB5901F4F5C@opencsta.org> <827E721D-9E3B-45C6-AA2E-B55541F03CFE@opencsta.org> <4FAB995F.2060609@puzzled.xs4all.nl> Message-ID: Hi Patrick, On Thu, May 10, 2012 at 12:33 PM, Patrick Lists wrote: > > I agree that the Dialogic Diva cards are great. I've been using them for > many years and they have always worked flawlessly. Like the Sangoma > cards I might add. Afaik there is no channel driver for Diva cards in > FreeSWITCH (Dialogic: create one!) so in relation to FreeSWITCH it would > unfortunately not be an option. There is chan_capi in the Asterisk world > which works great. There is one option: Diva SIPcontrol turns the Diva cards into a Mediagateway with SIP-Interface. Works great and is easy to set up. Afaik a two channel license suitable for a single BRI can be used free of charge. http://www.dialogic.com/Products/gateways/enterprise-media-gateways/sipcontrol.aspx From miha at softnet.si Thu May 10 16:30:23 2012 From: miha at softnet.si (Miha) Date: Thu, 10 May 2012 14:30:23 +0200 Subject: [Freeswitch-users] Phone not registering Message-ID: <4FABB4DF.5090200@softnet.si> Hi, here is pastebin of siptrace (http://pastebin.freeswitch.org/19029). Phone on local network are registered on FS. After I put between local network and Phone router, phones are unable to registered on FS. On other softswitch which is not FS phones are registering (same port, same scenario, etc.). Phones are SPA922. What could be causing the problem? Thanks! Miha From curriegrad2004 at gmail.com Thu May 10 18:07:39 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Thu, 10 May 2012 07:07:39 -0700 Subject: [Freeswitch-users] Freetdm DTMF In-Reply-To: References: Message-ID: It would be beneficial if you can post the dialplan on how FS handles the part where FreeTDM comes into play. I.e. the dialplan that does the transfer of the FreeTDM call into the default context. This seems to be opening up a can of worms here with FXS cards in FreeTDM... On Wed, May 9, 2012 at 8:39 PM, Shane Harrison wrote: > Happy for you to laugh at yourself, I'm just happy you are finding some time > to take some interest in my problem.? Much appreciated. > > I think some clarity is required here.? I am simply trying to do an attended > transfer as per the wiki (except I set the bind_meta_app to both legs) > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_att_xfer > > It works? fine for SIP to SIP.? For SIP to Freetdm it only works if the SIP > is trying to do the transfer.? If the FreeTDM is trying to do it ie. push *3 > on the phone keypad, then the bind_meta_app works fine and detects the *3 > and executes the appropriate extension ie. att_xfer, however the read() in > the extension att_xfer does not see the subsequent DTMF pressed on the phone > ie. the destination extension number. > > Cheers > Shane > > > On Thu, May 10, 2012 at 2:44 PM, curriegrad2004 > wrote: >> >> I just had to laugh at my self for mixing up the 2 again... >> bind_meta_app is only applicable to that extension that the inbound >> call was made to. Since you've transferred to another extension, the >> bind_meta_app won't work anymore because it's not defined in the >> extension you're transferring to. >> >> If you want this to happen, you'd have to manually define that >> bind_meta_app to those target extensions too. Remember, do this at >> your own peril - obvious misuse of bind_meta_app can open a huge >> security hole if you don't know what you're doing :) >> >> On Wed, May 9, 2012 at 7:27 PM, Shane Harrison >> wrote: >> > Thanks for the thoughts.? As I said, I am already setting it to both >> > legs - >> > I will try simply trying one leg but am sceptical :-) >> > >> > I also mentioned that I called the start_dtmf just before calling the >> > read >> > so unless I am doing something wrong here.....? I'll try and post the >> > XML >> > tonight when I get home.? Oh and it is an FXS card not an FXO of course >> > since it has a phone plugged into it. >> > >> > The question still remains though, why is the in-band DTMF detection >> > working >> > for the bind_meta_app digit detection but not after that? >> > >> > Cheers >> > Shane >> > >> > >> > On Thu, May 10, 2012 at 12:28 PM, curriegrad2004 >> > >> > wrote: >> >> >> >> and crap, since I wasn't even reading anything here, on the subsequent >> >> transfers from your FXO card, enable the in-band DTMF detector that FS >> >> has. The details on the in-band DTMF detector is here: >> >> >> >> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf >> >> >> >> But use this with caution, if there is a DTMF detector on the FXO card >> >> itself, make sure you disable it before using it. >> >> >> >> On Wed, May 9, 2012 at 5:26 PM, curriegrad2004 >> >> >> >> wrote: >> >> > apologies for telling you the wrong thing. I was replying you from my >> >> > phone btw :P >> >> > >> >> > Yeah, bind_meta_app is the app you would use, but try changing it to >> >> > point to the b-leg, not the a-leg >> >> > >> >> > On Wed, May 9, 2012 at 4:12 PM, Shane Harrison >> >> > wrote: >> >> >> Thanks.? I am currently using bind_meta_app (set to both legs) >> >> >> already >> >> >> rather than bind_digits.? I'll give bind_digits a shot and see if it >> >> >> behaves >> >> >> differently. >> >> >> >> >> >> Note that I do detect the initial *3 digits and because >> >> >> bind_meta_app >> >> >> is >> >> >> both legs, this is successful no matter which direction the call is >> >> >> setup >> >> >> from.? However once the dialplan moves to the extension the *3 is >> >> >> bound >> >> >> to, >> >> >> digits are no longer received. >> >> >> >> >> >> The worrying thing for me is that ftdm_io.c doesn't even write to >> >> >> the >> >> >> log >> >> >> that it has received them (nor freetdm above that of course which is >> >> >> understandable) and I am surprised that the read() influences that >> >> >> since it >> >> >> works prior on the *3 digits. >> >> >> >> >> >> Cheers >> >> >> Shane >> >> >> >> >> >> >> >> >> >> >> >> On Thu, May 10, 2012 at 10:22 AM, curriegrad2004 >> >> >> >> >> >> wrote: >> >> >>> >> >> >>> I'm guessing the bind digits in your analog card was set to listen >> >> >>> for >> >> >>> this sequence on the a-leg given if the call was being routed from >> >> >>> the >> >> >>> IP side to the analog side. >> >> >>> >> >> >>> Try changing that to listen on the b-leg. >> >> >>> >> >> >>> On 5/9/12, Shane Harrison wrote: >> >> >>> > Hi All, >> >> >>> > >> >> >>> > Have a situation where I have a call between a SIP phone and a >> >> >>> > FreeTDM >> >> >>> > channel. ? Pushing *3 on the analog FreeTDM phone is detected and >> >> >>> > this >> >> >>> > is >> >> >>> > bound to a dialplan extension (attended transfer) that does a >> >> >>> > read(): >> >> >>> > >> >> >>> > >> >> >>> > However pushing further digits on the analog phone ie. extension >> >> >>> > number >> >> >>> > of >> >> >>> > phone we wish to do an attended transfer to , doesn't result in >> >> >>> > the >> >> >>> > DTMF >> >> >>> > being detected. ?Note that this all works the other way around >> >> >>> > ie. >> >> >>> > using >> >> >>> > the SIP phone. >> >> >>> > >> >> >>> > When the *3 digits are pushed on the analog phone I see the logs >> >> >>> > report: >> >> >>> > >> >> >>> > ?ftdm_io.c:3463 [s2c1][1:4] Queuing DTMF * (debug = 0) >> >> >>> > mod_freetdm.c:702 Queuing DTMF [*] in channel FreeTDM/2:1/ >> >> >>> > >> >> >>> > ftdm_io.c:3463 [s2c1][1:4] Queuing DTMF 4 (debug = 0) >> >> >>> > >> >> >>> > mod_freetdm.c:702 Queuing DTMF [4] in channel FreeTDM/2:1/ >> >> >>> > >> >> >>> > When the further keys are pushed ftdm_io reports nothing. >> >> >>> > >> >> >>> > I have tried inserting a start_dtmf before the read() but it had >> >> >>> > no >> >> >>> > effect. >> >> >>> > Any thoughts as to why DTMF isn't being seen from the analog >> >> >>> > phone >> >> >>> > after >> >> >>> > the read()? >> >> >>> > >> >> >>> > Cheers >> >> >>> > Shane >> >> >>> > >> >> >>> >> >> >>> >> >> >>> >> >> >>> _________________________________________________________________________ >> >> >>> Professional FreeSWITCH Consulting Services: >> >> >>> consulting at freeswitch.org >> >> >>> http://www.freeswitchsolutions.com >> >> >>> >> >> >>> >> >> >>> >> >> >>> >> >> >>> Official FreeSWITCH Sites >> >> >>> http://www.freeswitch.org >> >> >>> http://wiki.freeswitch.org >> >> >>> http://www.cluecon.com >> >> >>> >> >> >>> FreeSWITCH-users mailing list >> >> >>> FreeSWITCH-users at lists.freeswitch.org >> >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >>> >> >> >>> >> >> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> >>> http://www.freeswitch.org >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> -- >> >> >> Paragon Electronic Design Ltd >> >> >> L6 Crest House >> >> >> 92 Queens Drive >> >> >> P0 Box 30449 >> >> >> Lower Hutt 5040 >> >> >> >> >> >> +64 4 5703870 Extn 875 >> >> >> +64 21 608919? (mobile) >> >> >> >> >> >> "Solving your problems with the right technology" >> >> >> >> >> >> >> >> >> >> >> >> _________________________________________________________________________ >> >> >> Professional FreeSWITCH Consulting Services: >> >> >> consulting at freeswitch.org >> >> >> http://www.freeswitchsolutions.com >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> Official FreeSWITCH Sites >> >> >> http://www.freeswitch.org >> >> >> http://wiki.freeswitch.org >> >> >> http://www.cluecon.com >> >> >> >> >> >> FreeSWITCH-users mailing list >> >> >> FreeSWITCH-users at 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 at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> >> >> >> >> >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> http://wiki.freeswitch.org >> >> http://www.cluecon.com >> >> >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> > >> > >> > >> > >> > -- >> > Paragon Electronic Design Ltd >> > L6 Crest House >> > 92 Queens Drive >> > P0 Box 30449 >> > Lower Hutt 5040 >> > >> > +64 4 5703870 Extn 875 >> > +64 21 608919? (mobile) >> > >> > "Solving your problems with the right technology" >> > >> > >> > _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at 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 at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > > -- > Paragon Electronic Design Ltd > L6 Crest House > 92 Queens Drive > P0 Box 30449 > Lower Hutt 5040 > > +64 4 5703870 Extn 875 > +64 21 608919? (mobile) > > "Solving your problems with the right technology" > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From jmesquita at freeswitch.org Thu May 10 18:16:23 2012 From: jmesquita at freeswitch.org (=?utf-8?Q?Jo=C3=A3o_Mesquita?=) Date: Thu, 10 May 2012 11:16:23 -0300 Subject: [Freeswitch-users] Howto? Migrating from Siemens HiPath4000 to FreeSWITCH In-Reply-To: References: <4FAB572C.20508@necosec.de> <01907D92-8A04-4388-B052-8DB5901F4F5C@opencsta.org> <827E721D-9E3B-45C6-AA2E-B55541F03CFE@opencsta.org> <4FAB995F.2060609@puzzled.xs4all.nl> Message-ID: <9E2035251CE84AC5935E062553E95D61@freeswitch.org> Ulrich, I work with a Brazilian company called Khomp. They have been replacing Dialogic cards on their traditional market for many years now and Dialogic barely exists in Brazil because of them. A few years ago, they've ported their product line to work with Asterisk and now they also support FreeSWITCH. The main advantage of this product (as well as Dialogic) is that all protocols are implemented on the hardware and therefore if something does not work on the protocol level, it will be fixed by the factory. Khomp in Brazil is known by their fast response technical support (you can't expect to have Dialogic traditional developers if you don't whereas the FreeSWITCH/Asterisk market is more tolerant). The other side effect and something that you are most certainly not considering is that you will need something more advanced then just PRI emulated as carrier (sending clock) to your Siemens. To depict that, let's imagine a scenario. Alice calls Bob from a Siemens based phone to a FreeSWITCH extension (at this instance, you have 1 channel occupied on the PRI). Bob decides that this calls is to Robert, who uses a Siemens phone and transfers that call. At this point, you will be using 2 channels of the PRI and if this scenario is common, you'll end up with a channel shortage for the other PSTN calls and such. The way to solve this, is to use a modified ISDN version supported by the Siemens called QSIG. QSIG allows you to execute a certain transfer and free the channels, leaving the call handled by the Siemens alone since FS has nothing to do with it anymore. Khomp does support this protocol and it is widely tested with Siemens since we do sell a lot of cards to Siemens system integrators around the world for their dialers, ivrs and whatnot. If you are interested, let me know. I don't know if Sangoma has a solution for that and I am not trying to make you buy us instead of them, but I would like to warn you of that as well. For those who like the external device, Khomp has a new "TDMoE" like interface that is called EBS (External Board Series). Since TDMoE does not scale well, they have implemented their own protocol of control called CTIoIP. It is quite interesting and it scales REALLY well with FreeSWITCH (I have personally tried). Just my 2 cents. Regards, -- Jo?o Mesquita Sent with Sparrow (http://www.sparrowmailapp.com/?sig) On Thursday, May 10, 2012 at 7:53 AM, Markus Lindenberg wrote: > Hi Patrick, > > On Thu, May 10, 2012 at 12:33 PM, Patrick Lists > wrote: > > > > I agree that the Dialogic Diva cards are great. I've been using them for > > many years and they have always worked flawlessly. Like the Sangoma > > cards I might add. Afaik there is no channel driver for Diva cards in > > FreeSWITCH (Dialogic: create one!) so in relation to FreeSWITCH it would > > unfortunately not be an option. There is chan_capi in the Asterisk world > > which works great. > > > > > There is one option: Diva SIPcontrol turns the Diva cards into a > Mediagateway with SIP-Interface. Works great and is easy to set up. > Afaik a two channel license suitable for a single BRI can be used free > of charge. > > http://www.dialogic.com/Products/gateways/enterprise-media-gateways/sipcontrol.aspx > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120510/e280f322/attachment.html From koralu at gmail.com Thu May 10 18:25:03 2012 From: koralu at gmail.com (Adrian Andrei) Date: Thu, 10 May 2012 17:25:03 +0300 Subject: [Freeswitch-users] mod_xml_cdr is not logging In-Reply-To: References: Message-ID: I check the permission of both directorys and is ok. I check the webserver logs and FS calls my php "http://localhost/xml-cdr/cdr.php" but the POST['cdr'] is empty. I tried to log manually the content of POST['cdr'] to local file and is blank. Also nothing is saved in log-dir on disk. On 5/9/12, Wojtek Kochanowski wrote: > Check permissions of log-dir and error-log-dir. Owner and group also. > > > 2012/5/9 Adrian Andrei > >> Hello, >> >> Thank you for answer but I am not getting any files on disk. When I >> disable mod_cdr_xml the logs are saved naturally in freeswitch/log. >> >> Ty >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > From kochanowski.wojtek at gmail.com Thu May 10 18:32:03 2012 From: kochanowski.wojtek at gmail.com (Wojtek Kochanowski) Date: Thu, 10 May 2012 16:32:03 +0200 Subject: [Freeswitch-users] mod_xml_cdr is not logging In-Reply-To: References: Message-ID: In my case there's an empty value, like this: which means /usr/local/freeswitch/log/cdr-csv. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120510/658ef552/attachment.html From philq at qsystemsengineering.com Thu May 10 18:38:22 2012 From: philq at qsystemsengineering.com (Phil Quesinberry) Date: Thu, 10 May 2012 10:38:22 -0400 Subject: [Freeswitch-users] Bypass media succeeds from extension to gateway but fails from extension to extension Message-ID: <003e01cd2eba$904e55a0$b0eb00e0$@com> Thanks Ken. I'm still learning here, so if that's the case then help me to understand why FS properly sends the external address when that same extension makes a call out through a gateway instead of to another extension. Same settings, same originating extension. I did try enabling ndlb-connectile-dysfunction to rewrite the contact IP on those calls but to no avail. Thanks! - Phil ---------- Ken Rice Tue May 8 11:38:48 MSD 2012 This is not a bug in FreeSWITCH... Bypass Media is a special mode to allow us to act as a psuedo proxy. Other then possibly filtering some codecs the rest of the SDPs are simply copied across the bridge. That means if your sip devices are sending RFC1918 IPs in their SDPs then FreeSWITCH will forward an RFC1918 SDP. Why is that? Because FreeSWITCH has no way to know where the media is actually coming from since FreeSWITCH is not in the media path. K _____________________________________________ From: Phil Quesinberry Sent: Monday, May 07, 2012 11:54 AM To: 'freeswitch-users at lists.freeswitch.org' Subject: Bypass media succeeds from extension to gateway but fails from extension to extension I'm not sure if this is a bug or just a NAT-related configurational problem. If it's truly a bug, let me know and I'll be happy to file a Jira. I'm trying to get bypass media to work with extension to extension calls. Both endpoint extensions are behind NAT in two different locations. This looks like a possible bug because FS sends the internal IP address of one of the endpoints to the other for media, but when making a call from the same extension to an external gateway for PSTN termination, it sends the phone's external IP address as it should, and the call succeeds. Everything works fine, of course, when "proxy media" is used. The SIP traffic for the failed call is here, look for the word "WRONG!" to see the incorrect address being passed in the SDP: http://pastebin.freeswitch.org/19003 Regards, Phil Quesinberry Q Systems Engineering, Inc. Electronic Controls and Embedded Systems Development (410) 969-8002 http://www.qsystemsengineering.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120510/12b9900e/attachment-0001.html From krice at freeswitch.org Thu May 10 19:08:53 2012 From: krice at freeswitch.org (Ken Rice) Date: Thu, 10 May 2012 10:08:53 -0500 Subject: [Freeswitch-users] Bypass media succeeds from extension to gateway but fails from extension to extension In-Reply-To: <003e01cd2eba$904e55a0$b0eb00e0$@com> Message-ID: The contact IP has nothing to do with where the media goes... That?s entirely defined in the SDP... Consider this Endpoint A (192.168.100.100) -> NAT A -> FreeSWITCH (4.2.2.2) -> NAT B -> Endpoint B (192.168.100.200) Now lets assume that NAT A and NAT B are 2 separate nat gateways and that Endpoint A and Endpoint B are on 2 different physical LANs... Telling Endpoint A to talk directly to Endpoint B without proxying media will never work since the endpoints think they are on the same LAN. There is no mechanism there to allow for the redirection and automagic adjustments of ports etc so that they can talk directly to each other... Now lets change this slightly so that endpoint B is 192.168.200.200. Unless NAT A knows how to get to 192.168.200.0/24 (assuming class C sized block) and NAT B knows how to get to 192.168.100.0/24 they are both going to use their default routing which is to NAT the outgoing RTP, and forward it to the next hop... Again, RTP will not make it to other side in either direction... FreeSWITCH cant compensate due to a number of factors... Your Endpoints have to be smart enough to actually compensate for the NAT in this situation OR your NAT boxes have to compensate for it... The simple answer, don?t use bypass media in this situation, the complex answer I wont get into here... Stop by IRC and ask around... There is a 3rd option here you might want to consider, contact consulting at freeswitch.org for some professional help... This may not be specifially what you need to get going as I have no clue what your skill level happens to be, and you did say you are still learning. Good Luck! On 5/10/12 9:38 AM, "Phil Quesinberry" wrote: > Thanks Ken. I?m still learning here, so if that?s the case then help me to > understand why FS properly sends the external address when that same extension > makes a call out through a gateway instead of to another extension. Same > settings, same originating extension. > > I did try enabling ndlb-connectile-dysfunction to rewrite the contact IP on > those calls but to no avail. > > Thanks! > > - Phil > > ---------- > > Ken Rice > Tue May 8 11:38:48 MSD 2012 > > This is not a bug in FreeSWITCH... > > Bypass Media is a special mode to allow us to act as a psuedo proxy. > > Other then possibly filtering some codecs the rest of the SDPs are simply > > copied across the bridge. That means if your sip devices are sending RFC1918 > > IPs in their SDPs then FreeSWITCH will forward an RFC1918 SDP. Why is that? > > Because FreeSWITCH has no way to know where the media is actually coming > > from since FreeSWITCH is not in the media path. > > K > > _____________________________________________ > From: Phil Quesinberry > > Sent: Monday, May 07, 2012 11:54 AM > To: 'freeswitch-users at lists.freeswitch.org' > Subject: Bypass media succeeds from extension to gateway but fails from > extension to extension > > I?m not sure if this is a bug or just a NAT-related configurational problem. > If it?s truly a bug, let me know and I?ll be happy to file a Jira. > > I?m trying to get bypass media to work with extension to extension calls. > Both endpoint extensions are behind NAT in two different locations. This > looks like a possible bug because FS sends the internal IP address of one of > the endpoints to the other for media, but when making a call from the same > extension to an external gateway for PSTN termination, it sends the phone?s > external IP address as it should, and the call succeeds. > > Everything works fine, of course, when ?proxy media? is used. > > The SIP traffic for the failed call is here, look for the word ?WRONG!? to see > the incorrect address being passed in the SDP: > > http://pastebin.freeswitch.org/19003 > > Regards, > > Phil Quesinberry > > Q Systems Engineering, Inc. > > Electronic Controls and Embedded Systems Development > > (410) 969-8002 > > http://www.qsystemsengineering.com > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120510/ac395c60/attachment.html From jerry.richards at teotech.com Thu May 10 19:23:20 2012 From: jerry.richards at teotech.com (Jerry Richards) Date: Thu, 10 May 2012 15:23:20 +0000 Subject: [Freeswitch-users] Gaps In RTP (wideband) Packets From mod_voicemail Message-ID: <1545146083A72C4DB7B66584B7E5D98402BC1FF4@BY2PRD0410MB377.namprd04.prod.outlook.com> Hello, I'm seeing a audio problem when listening to the operator when calling into voicemail to listen to messages and a wideband codec is used. Looking at a wireshark trace, I see gaps between transmitted RTP packets that exceed the jitter buffer depth in the endpoint softphone, causing underflow. This causes clipping of the front-end of words such as "Pressed" sounds like 'ess' and "Last name" sounds like 'ast name'. While this is generally not noticeable with the G.711 codec, it causes these clipped audio phrases with wideband codecs, such as BV32. It might be due to the processing delay in mod_voicemail between phrases? Has anyone seen this? Is there a tag to help improve this? Thanks, Jerry -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120510/4e29522e/attachment.html From koralu at gmail.com Thu May 10 19:27:55 2012 From: koralu at gmail.com (Adrian Andrei) Date: Thu, 10 May 2012 18:27:55 +0300 Subject: [Freeswitch-users] mod_xml_cdr is not logging In-Reply-To: References: Message-ID: Ok but the $_POST['cdr'] is empty. I tried diffrent methods to check this and is true. Please if you have some advice to debug I appriciate. On 5/8/12, Adrian Andrei wrote: > Hello > > I tried to set up mod_xml_cdr but it doesn't work at all. I made the > following steps: > > - Uncomment mod_xml_cdr in modules.conf. - Ok > - Edit conf/autoload_configs/modules.conf.xml. - OK > - Load mod_xml_cdr from CLI - NO errors > - My *xml_cdr.conf.xml* looks like: > > > > > > > > > value="/usr/local/freeswitch/log/cdr/errors"/> > > > > (The php files are from contrib/trixter/xml-cdr.) > > In some posts I read that if log-dir and err-log-dir are changed from > "default" I should set also the log-http-and-disk. But it doesn't work. > > Wiki said that I should check the freeswitch.xml to see if it is included > xml_cdr.conf.xml but I can't find any line with xml_cdr.conf.xml. The > version of freeswitch is FreeSWITCH Version 1.0.head (git-2c52f23 > 2012-02-18 08-37-47 -0600) so I think is new enough to have this feature > available. > > Could anyone help me to find out what I'm missing? Are any commands in FS > in order to run a mod_cdr_xml debug? > > Ty in advance. > From jerry.richards at teotech.com Thu May 10 19:35:41 2012 From: jerry.richards at teotech.com (Jerry Richards) Date: Thu, 10 May 2012 15:35:41 +0000 Subject: [Freeswitch-users] Gaps In RTP (wideband) Packets From mod_voicemail In-Reply-To: <1545146083A72C4DB7B66584B7E5D98402BC1FF4@BY2PRD0410MB377.namprd04.prod.outlook.com> References: <1545146083A72C4DB7B66584B7E5D98402BC1FF4@BY2PRD0410MB377.namprd04.prod.outlook.com> Message-ID: <1545146083A72C4DB7B66584B7E5D98402BC2015@BY2PRD0410MB377.namprd04.prod.outlook.com> By the way, in contrast, I created an XML block that plays the same messages as mod_voicemail (see block below). When this XML block is executed, there is no clipping of words when using wideband codec BV32. This is why I'm thinking it's related to delays caused by mod_voicemail processing. Thanks, Jerry From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jerry Richards Sent: Thursday, May 10, 2012 8:23 AM To: FreeSWITCH Users Help Subject: [Freeswitch-users] Gaps In RTP (wideband) Packets From mod_voicemail Hello, I'm seeing a audio problem when listening to the operator when calling into voicemail to listen to messages and a wideband codec is used. Looking at a wireshark trace, I see gaps between transmitted RTP packets that exceed the jitter buffer depth in the endpoint softphone, causing underflow. This causes clipping of the front-end of words such as "Pressed" sounds like 'ess' and "Last name" sounds like 'ast name'. While this is generally not noticeable with the G.711 codec, it causes these clipped audio phrases with wideband codecs, such as BV32. It might be due to the processing delay in mod_voicemail between phrases? Has anyone seen this? Is there a tag to help improve this? Thanks, Jerry -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120510/493e786c/attachment.html From freeswitch-list at puzzled.xs4all.nl Thu May 10 19:36:16 2012 From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists) Date: Thu, 10 May 2012 17:36:16 +0200 Subject: [Freeswitch-users] Howto? Migrating from Siemens HiPath4000 to FreeSWITCH In-Reply-To: References: <4FAB572C.20508@necosec.de> <01907D92-8A04-4388-B052-8DB5901F4F5C@opencsta.org> <827E721D-9E3B-45C6-AA2E-B55541F03CFE@opencsta.org> <4FAB995F.2060609@puzzled.xs4all.nl> Message-ID: <4FABE070.7000006@puzzled.xs4all.nl> Hi Markus, On 10-05-12 12:53, Markus Lindenberg wrote: [snip] > There is one option: Diva SIPcontrol turns the Diva cards into a > Mediagateway with SIP-Interface. Works great and is easy to set up. > Afaik a two channel license suitable for a single BRI can be used free > of charge. > > http://www.dialogic.com/Products/gateways/enterprise-media-gateways/sipcontrol.aspx Thanks for the tip! I had to do some digging but I think SIPcontrol is part of the Diva4Linux_installer archive. So it seems I can no longer avoid using that rather "creative" way of installing their stuff :) Thanks again, Patrick From nbhatti at gmail.com Thu May 10 19:36:30 2012 From: nbhatti at gmail.com (Muhammad Naseer Bhatti) Date: Thu, 10 May 2012 18:36:30 +0300 Subject: [Freeswitch-users] mod_xml_cdr is not logging In-Reply-To: References: Message-ID: Check /etc/hosts file. Sometimes the localhost entry is not present. Make sure it is there and pointing to 127.0.0.1 or use 127.0.0.1 explicitly in the config file. On Thu, May 10, 2012 at 6:27 PM, Adrian Andrei wrote: > Ok but the $_POST['cdr'] is empty. I tried diffrent methods to check > this and is true. > Please if you have some advice to debug I appriciate. > > On 5/8/12, Adrian Andrei wrote: > > Hello > > > > I tried to set up mod_xml_cdr but it doesn't work at all. I made the > > following steps: > > > > - Uncomment mod_xml_cdr in modules.conf. - Ok > > - Edit conf/autoload_configs/modules.conf.xml. - OK > > - Load mod_xml_cdr from CLI - NO errors > > - My *xml_cdr.conf.xml* looks like: > > > > > > > > > > > > > > > > > > > value="/usr/local/freeswitch/log/cdr/errors"/> > > > > > > > > (The php files are from contrib/trixter/xml-cdr.) > > > > In some posts I read that if log-dir and err-log-dir are changed from > > "default" I should set also the log-http-and-disk. But it doesn't work. > > > > Wiki said that I should check the freeswitch.xml to see if it is included > > xml_cdr.conf.xml but I can't find any line with xml_cdr.conf.xml. The > > version of freeswitch is FreeSWITCH Version 1.0.head (git-2c52f23 > > 2012-02-18 08-37-47 -0600) so I think is new enough to have this feature > > available. > > > > Could anyone help me to find out what I'm missing? Are any commands in FS > > in order to run a mod_cdr_xml debug? > > > > Ty in advance. > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120510/b9209bc3/attachment-0001.html From anthony.minessale at gmail.com Thu May 10 19:39:41 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 10 May 2012 10:39:41 -0500 Subject: [Freeswitch-users] Gaps In RTP (wideband) Packets From mod_voicemail In-Reply-To: <1545146083A72C4DB7B66584B7E5D98402BC2015@BY2PRD0410MB377.namprd04.prod.outlook.com> References: <1545146083A72C4DB7B66584B7E5D98402BC1FF4@BY2PRD0410MB377.namprd04.prod.outlook.com> <1545146083A72C4DB7B66584B7E5D98402BC2015@BY2PRD0410MB377.namprd04.prod.outlook.com> Message-ID: set send_silence_when_idle=true globally or per-leg On Thu, May 10, 2012 at 10:35 AM, Jerry Richards wrote: > By the way, in contrast, I created an XML block that plays the same messages > as mod_voicemail (see block below).? When this XML block is executed, there > is no clipping of words when using wideband codec BV32.? This is why I'm > thinking it's related to delays caused by mod_voicemail processing. > > > > ??? > > ????? > > ??????? > > ?????? > > ????????????? data="/opt/teoswitch/sounds/en/us/callie/voicemail/vm-hello.wav"/> > > ??????? data="/opt/teoswitch/sounds/en/us/callie/voicemail/vm-you_have.wav"/> > > ????????????? > > ??????? data="/opt/teoswitch/sounds/en/us/callie/voicemail/vm-new.wav"/> > > ??????? data="/opt/teoswitch/sounds/en/us/callie/voicemail/vm-messages.wav"/> > > ??????? data="/opt/teoswitch/sounds/en/us/callie/voicemail/vm-you_have.wav"/> > > ????????????? > > ??????? data="/opt/teoswitch/sounds/en/us/callie/voicemail/vm-saved.wav"/> > > ??????? data="/opt/teoswitch/sounds/en/us/callie/voicemail/vm-messages.wav"/> > > ??????? data="/opt/teoswitch/sounds/en/us/callie/voicemail/vm-listen_new.wav"/> > > ??????? data="/opt/teoswitch/sounds/en/us/callie/voicemail/vm-press.wav"/> > > ????????????? > > ??????? > > ????? > > ??? > > > > Thanks, > > Jerry > > > > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jerry > Richards > Sent: Thursday, May 10, 2012 8:23 AM > To: FreeSWITCH Users Help > Subject: [Freeswitch-users] Gaps In RTP (wideband) Packets From > mod_voicemail > > > > Hello, > > > > I'm seeing a audio problem when listening to the operator when calling into > voicemail to listen to messages and a wideband codec is used.? Looking at a > wireshark trace, I see gaps between transmitted RTP packets that exceed the > jitter buffer depth in the endpoint softphone, causing underflow.? ?This > causes clipping of the front-end of words such as "Pressed" sounds like > 'ess' and "Last name" sounds like 'ast name'.? While this is generally not > noticeable with the G.711 codec, it causes these clipped audio phrases with > wideband codecs, such as BV32.? It might be due to the processing delay in > mod_voicemail between phrases? > > > > Has anyone seen this?? Is there a tag to help improve this? > > > > Thanks, > > Jerry > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From koralu at gmail.com Thu May 10 19:45:55 2012 From: koralu at gmail.com (Adrian Andrei) Date: Thu, 10 May 2012 18:45:55 +0300 Subject: [Freeswitch-users] mod_xml_cdr is not logging In-Reply-To: References: Message-ID: Same result. I tried both 127.0.0.1 and localhost. etc/hosts is valid. On 5/10/12, Muhammad Naseer Bhatti wrote: > Check /etc/hosts file. Sometimes the localhost entry is not present. Make > sure it is there and pointing to 127.0.0.1 or use 127.0.0.1 explicitly in > the config file. > > On Thu, May 10, 2012 at 6:27 PM, Adrian Andrei wrote: > >> Ok but the $_POST['cdr'] is empty. I tried diffrent methods to check >> this and is true. >> Please if you have some advice to debug I appriciate. >> >> On 5/8/12, Adrian Andrei wrote: >> > Hello >> > >> > I tried to set up mod_xml_cdr but it doesn't work at all. I made the >> > following steps: >> > >> > - Uncomment mod_xml_cdr in modules.conf. - Ok >> > - Edit conf/autoload_configs/modules.conf.xml. - OK >> > - Load mod_xml_cdr from CLI - NO errors >> > - My *xml_cdr.conf.xml* looks like: >> > >> > >> > >> > >> > >> > >> > >> > >> > > > value="/usr/local/freeswitch/log/cdr/errors"/> >> > >> > >> > >> > (The php files are from contrib/trixter/xml-cdr.) >> > >> > In some posts I read that if log-dir and err-log-dir are changed from >> > "default" I should set also the log-http-and-disk. But it doesn't work. >> > >> > Wiki said that I should check the freeswitch.xml to see if it is >> > included >> > xml_cdr.conf.xml but I can't find any line with xml_cdr.conf.xml. The >> > version of freeswitch is FreeSWITCH Version 1.0.head (git-2c52f23 >> > 2012-02-18 08-37-47 -0600) so I think is new enough to have this >> > feature >> > available. >> > >> > Could anyone help me to find out what I'm missing? Are any commands in >> > FS >> > in order to run a mod_cdr_xml debug? >> > >> > Ty in advance. >> > >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > From jerry.richards at teotech.com Thu May 10 20:26:24 2012 From: jerry.richards at teotech.com (Jerry Richards) Date: Thu, 10 May 2012 16:26:24 +0000 Subject: [Freeswitch-users] Gaps In RTP (wideband) Packets From mod_voicemail In-Reply-To: References: <1545146083A72C4DB7B66584B7E5D98402BC1FF4@BY2PRD0410MB377.namprd04.prod.outlook.com> <1545146083A72C4DB7B66584B7E5D98402BC2015@BY2PRD0410MB377.namprd04.prod.outlook.com> Message-ID: <1545146083A72C4DB7B66584B7E5D98402BC206B@BY2PRD0410MB377.namprd04.prod.outlook.com> I'm setting send_silence_when_idle globally to a numerical value that fixes a loud white noise issue during sleeps. I originally had it set to true but still had the wideband clipping issue. Jerry -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Thursday, May 10, 2012 8:40 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Gaps In RTP (wideband) Packets From mod_voicemail set send_silence_when_idle=true globally or per-leg On Thu, May 10, 2012 at 10:35 AM, Jerry Richards wrote: > By the way, in contrast, I created an XML block that plays the same > messages as mod_voicemail (see block below).? When this XML block is > executed, there is no clipping of words when using wideband codec > BV32.? This is why I'm thinking it's related to delays caused by mod_voicemail processing. > > > > ??? > > ????? > > ??????? > > ?????? > > ????????????? data="/opt/teoswitch/sounds/en/us/callie/voicemail/vm-hello.wav"/> > > ??????? data="/opt/teoswitch/sounds/en/us/callie/voicemail/vm-you_have.wav"/> > > ????????????? > > ??????? data="/opt/teoswitch/sounds/en/us/callie/voicemail/vm-new.wav"/> > > ??????? data="/opt/teoswitch/sounds/en/us/callie/voicemail/vm-messages.wav"/> > > ??????? data="/opt/teoswitch/sounds/en/us/callie/voicemail/vm-you_have.wav"/> > > ????????????? > > ??????? data="/opt/teoswitch/sounds/en/us/callie/voicemail/vm-saved.wav"/> > > ??????? data="/opt/teoswitch/sounds/en/us/callie/voicemail/vm-messages.wav"/> > > ??????? data="/opt/teoswitch/sounds/en/us/callie/voicemail/vm-listen_new.wav"/ > > > > ??????? data="/opt/teoswitch/sounds/en/us/callie/voicemail/vm-press.wav"/> > > ????????????? > > ??????? > > ????? > > ??? > > > > Thanks, > > Jerry > > > > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Jerry Richards > Sent: Thursday, May 10, 2012 8:23 AM > To: FreeSWITCH Users Help > Subject: [Freeswitch-users] Gaps In RTP (wideband) Packets From > mod_voicemail > > > > Hello, > > > > I'm seeing a audio problem when listening to the operator when calling > into voicemail to listen to messages and a wideband codec is used.? > Looking at a wireshark trace, I see gaps between transmitted RTP > packets that exceed the jitter buffer depth in the endpoint softphone, > causing underflow.? ?This causes clipping of the front-end of words > such as "Pressed" sounds like 'ess' and "Last name" sounds like 'ast > name'.? While this is generally not noticeable with the G.711 codec, > it causes these clipped audio phrases with wideband codecs, such as > BV32.? It might be due to the processing delay in mod_voicemail between phrases? > > > > Has anyone seen this?? Is there a tag to help improve this? > > > > Thanks, > > Jerry > > > > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From anthony.minessale at gmail.com Thu May 10 20:56:04 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 10 May 2012 11:56:04 -0500 Subject: [Freeswitch-users] Gaps In RTP (wideband) Packets From mod_voicemail In-Reply-To: <1545146083A72C4DB7B66584B7E5D98402BC206B@BY2PRD0410MB377.namprd04.prod.outlook.com> References: <1545146083A72C4DB7B66584B7E5D98402BC1FF4@BY2PRD0410MB377.namprd04.prod.outlook.com> <1545146083A72C4DB7B66584B7E5D98402BC2015@BY2PRD0410MB377.namprd04.prod.outlook.com> <1545146083A72C4DB7B66584B7E5D98402BC206B@BY2PRD0410MB377.namprd04.prod.outlook.com> Message-ID: I found a restriction in the core limiting that functionality to sleeps > 100ms, the vm sleeps 100ms between files by default. update to latest and try again. On Thu, May 10, 2012 at 11:26 AM, Jerry Richards wrote: > I'm setting send_silence_when_idle globally to a numerical value that fixes a loud white noise issue during sleeps. ?I originally had it set to true but still had the wideband clipping issue. > > Jerry > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale > Sent: Thursday, May 10, 2012 8:40 AM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Gaps In RTP (wideband) Packets From mod_voicemail > > set send_silence_when_idle=true globally or per-leg > > On Thu, May 10, 2012 at 10:35 AM, Jerry Richards wrote: >> By the way, in contrast, I created an XML block that plays the same >> messages as mod_voicemail (see block below).? When this XML block is >> executed, there is no clipping of words when using wideband codec >> BV32.? This is why I'm thinking it's related to delays caused by mod_voicemail processing. >> >> >> >> ??? >> >> ????? >> >> ??????? >> >> ?????? >> >> ????????????? > data="/opt/teoswitch/sounds/en/us/callie/voicemail/vm-hello.wav"/> >> >> ??????? > data="/opt/teoswitch/sounds/en/us/callie/voicemail/vm-you_have.wav"/> >> >> ????????????? >> >> ??????? > data="/opt/teoswitch/sounds/en/us/callie/voicemail/vm-new.wav"/> >> >> ??????? > data="/opt/teoswitch/sounds/en/us/callie/voicemail/vm-messages.wav"/> >> >> ??????? > data="/opt/teoswitch/sounds/en/us/callie/voicemail/vm-you_have.wav"/> >> >> ????????????? >> >> ??????? > data="/opt/teoswitch/sounds/en/us/callie/voicemail/vm-saved.wav"/> >> >> ??????? > data="/opt/teoswitch/sounds/en/us/callie/voicemail/vm-messages.wav"/> >> >> ??????? > data="/opt/teoswitch/sounds/en/us/callie/voicemail/vm-listen_new.wav"/ >> > >> >> ??????? > data="/opt/teoswitch/sounds/en/us/callie/voicemail/vm-press.wav"/> >> >> ????????????? >> >> ??????? >> >> ????? >> >> ??? >> >> >> >> Thanks, >> >> Jerry >> >> >> >> From: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of >> Jerry Richards >> Sent: Thursday, May 10, 2012 8:23 AM >> To: FreeSWITCH Users Help >> Subject: [Freeswitch-users] Gaps In RTP (wideband) Packets From >> mod_voicemail >> >> >> >> Hello, >> >> >> >> I'm seeing a audio problem when listening to the operator when calling >> into voicemail to listen to messages and a wideband codec is used. >> Looking at a wireshark trace, I see gaps between transmitted RTP >> packets that exceed the jitter buffer depth in the endpoint softphone, >> causing underflow.? ?This causes clipping of the front-end of words >> such as "Pressed" sounds like 'ess' and "Last name" sounds like 'ast >> name'.? While this is generally not noticeable with the G.711 codec, >> it causes these clipped audio phrases with wideband codecs, such as >> BV32.? It might be due to the processing delay in mod_voicemail between phrases? >> >> >> >> Has anyone seen this?? Is there a tag to help improve this? >> >> >> >> Thanks, >> >> Jerry >> >> >> >> >> ______________________________________________________________________ >> ___ Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use >> rs >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From chris at gonumina.com Thu May 10 21:30:47 2012 From: chris at gonumina.com (Chris Ferreira) Date: Thu, 10 May 2012 13:30:47 -0400 Subject: [Freeswitch-users] Fax Issues with Cisco SPA112 and T.38 Message-ID: Hello All, I have a FreeSWITCH install running on CentOS 5.6 on a Linode VPS. I am trying to get my SPA112 (Version 1.1 Firmware) to send faxes (and eventually receive) successfully. I have the ATA registered as an extension and it's primary outgoing route is through Flowroute. I have T.38 enabled on the ATA and I have diabled ECM on the analog fax machine. All of the CDR's in Flowroute show that the test faxes are all ending their calls at 39 seconds. I can post screen shots of the ATA config or provide any other config info. Is there something I am missing, or should this setup work? I have poked around for a while for info, but unlike the PAP2T there is little info on these SPA112's. This is my first post asking for help as I usually try to resolve things on my own. But for this, I will defer to everyone's experience. Thanks, -Chris -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120510/8fe45096/attachment-0001.html From krice at freeswitch.org Thu May 10 21:49:06 2012 From: krice at freeswitch.org (Ken Rice) Date: Thu, 10 May 2012 12:49:06 -0500 Subject: [Freeswitch-users] 1.2-rc2 Tagged and Tarball available on files.freeswitch.org In-Reply-To: <5b796df030c0eaae30804b38a2f578b8.squirrel@my.netvps.co.uk> Message-ID: Hey Guys, Also, please keep in mind that this will be a stable supported branch... That means it'll be production quality... So please help us out by testing it... We've only gotta couple of bug reports so far, if you find an issue please double check to see if there is already a bug on jira, or open a bug tomake sure we address it. K On 5/9/12 3:04 AM, "Tom Parrott" wrote: > Excellent, will try and build that tonight. > > The RPM SPEC file has seen a lot of improvements over the last few months, > it's a lot more modular now. > > Cheers > Tom > >> Congratulations!! I know it's been a long time coming - I've been lurking >> around the FS site/irc for months! (my FS plans keep getting interfered >> with though!) >> >> :) >> >> >> On 09/05/2012, at 6:06 AM, Ken Rice wrote: >> >>> Hey Guys we?ve reached another milestone in getting FreeSWITCH 1.2 out >>> the door. >>> >>> http://files.freeswitch.org/freeswitch-1.2.rc2.tar.bz2 >>> >>> Yep that?s what you want to test... >>> >>> Look for debian packages and updated RPMs for these soon >>> >>> K >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From bdfoster at endigotech.com Thu May 10 22:03:06 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Thu, 10 May 2012 14:03:06 -0400 Subject: [Freeswitch-users] Fax Issues with Cisco SPA112 and T.38 In-Reply-To: References: Message-ID: 1. Debug tells all. We need the siptrace as well. 2. VPS's are usually bad at sending/receiving faxes even if the media is just being routed through it. 3. You could also look at session timers. Are other calls working? The only real way to know is if we have number 1 in our possession, otherwise we just start guessing. -BDF On Thu, May 10, 2012 at 1:30 PM, Chris Ferreira wrote: > Hello All, > > > > I have a FreeSWITCH install running on CentOS 5.6 on a Linode VPS. I am > trying to get my SPA112 (Version 1.1 Firmware) to send faxes (and > eventually receive) successfully. I have the ATA registered as an extension > and it's primary outgoing route is through Flowroute. I have T.38 enabled > on the ATA and I have diabled ECM on the analog fax machine. All of the > CDR's in Flowroute show that the test faxes are all ending their calls at > 39 seconds. > > > I can post screen shots of the ATA config or provide any other config info. > > > Is there something I am missing, or should this setup work? > > I have poked around for a while for info, but unlike the PAP2T there is > little info on these SPA112's. This is my first post asking for help as I > usually try to resolve things on my own. But for this, I will defer > to everyone's experience. > > > > Thanks, > > -Chris > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-800-7876 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120510/e4c9a381/attachment.html From nickolayr at gmail.com Thu May 10 22:07:06 2012 From: nickolayr at gmail.com (Nikolay Rogoshchenkov) Date: Thu, 10 May 2012 14:07:06 -0400 Subject: [Freeswitch-users] Can't compile with ODBC support under FreeBSD for current git Message-ID: Hello, I already have FreeSWITCH Version 1.1.beta1 (git-d730df7 2012-04-02 18-00-38 -0400) with ODBC support and my LUA scripts working properly with MySQL. But when I tried to *configure *the latest git (after git pull) I have got the following error: ================================================================================ [....] *** your path, or set the LIBGNUTLS_CONFIG environment variable to the *** full path to libgnutls-config. checking libtool major version... 2 using libtool library extension... la adding "-fPIC" to SWITCH_AM_CFLAGS adding "-fPIC" to SWITCH_AM_CXXFLAGS adding "-Werror" to SWITCH_AM_CFLAGS checking whether the compiler supports -fvisibility=hidden... yes adding "-fvisibility=hidden" to SWITCH_AM_CFLAGS adding "-DSWITCH_API_VISIBILITY=1" to SWITCH_AM_CFLAGS adding "-DHAVE_VISIBILITY=1" to SWITCH_AM_CFLAGS adding "-fvisibility=hidden" to SWITCH_AM_CXXFLAGS adding "-DSWITCH_API_VISIBILITY=1" to SWITCH_AM_CXXFLAGS adding "-DHAVE_VISIBILITY=1" to SWITCH_AM_CXXFLAGS checking CFLAGS for maximum ansi warnings... -Wall -std=c99 -pedantic adding "-g" to SWITCH_AM_CFLAGS adding "-ggdb" to SWITCH_AM_CFLAGS checking for jack... gnome-config: not found gnome-config: not found checking for snd_pcm_open in -lasound... no checking size of long... 8 checking what directory libraries are found in... lib checking for odbc header in /usr/include... no found checking for odbc header in /usr/local/include... found checking for SQLDisconnect in -lodbc... no checking for odbc library in /usr/lib... no found checking for odbc library in /usr/local/lib... found checking whether to include odbc... yes checking for SQLDisconnect in -lodbc... (cached) no configure: error: no usable libodbc; please install unixodbc devel package or equivalent # ================================================================================ but why? Thank you. -- Nikolay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120510/c5c7f551/attachment.html From arkaha at hotbox.ru Thu May 10 17:52:59 2012 From: arkaha at hotbox.ru (Limit) Date: Thu, 10 May 2012 06:52:59 -0700 (PDT) Subject: [Freeswitch-users] Attended transfer via ESL In-Reply-To: References: <1335539693783-7506468.post@n2.nabble.com> Message-ID: <1336657979551-7546785.post@n2.nabble.com> *mercutioviz*, thank you! Looks like it works for me! Transfer works just as called from dialplan. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Attended-transfer-via-ESL-tp7506468p7546785.html Sent from the freeswitch-users mailing list archive at Nabble.com. From shaik.bawajan at gmail.com Thu May 10 18:02:40 2012 From: shaik.bawajan at gmail.com (bawajan) Date: Thu, 10 May 2012 07:02:40 -0700 (PDT) Subject: [Freeswitch-users] Choopy one-way noise In-Reply-To: References: Message-ID: <1336658560567-7546814.post@n2.nabble.com> Hi Moises Silva, am new to freeswitch and am facing same noise issue when making call from ISDN (freetdm) to SIP UA. As per above mail, i have did ftdm trace and listen the clips. I found same noise in generated voice clip of line 2 (sip). am attached both audio files for your reference (ISDN line 1 and SIP line 2). Kindly let me know, what needs to do to rectify it. http://freeswitch-users.2379917.n2.nabble.com/file/n7546814/isdn_line1-in-s2c1.wav isdn_line1-in-s2c1.wav http://freeswitch-users.2379917.n2.nabble.com/file/n7546814/sip_line2-out-s2c1.wav sip_line2-out-s2c1.wav -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Choopy-one-way-noise-tp7332765p7546814.html Sent from the freeswitch-users mailing list archive at Nabble.com. From msc at freeswitch.org Thu May 10 22:17:37 2012 From: msc at freeswitch.org (Michael Collins) Date: Thu, 10 May 2012 11:17:37 -0700 Subject: [Freeswitch-users] mod_xml_cdr is not logging In-Reply-To: References: Message-ID: On Thu, May 10, 2012 at 8:45 AM, Adrian Andrei wrote: > Same result. I tried both 127.0.0.1 and localhost. etc/hosts is valid. > what happens when you go to fs_cli and type: reload mod_xml_cdr I'm curious. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120510/a012625a/attachment-0001.html From msc at freeswitch.org Thu May 10 22:18:18 2012 From: msc at freeswitch.org (Michael Collins) Date: Thu, 10 May 2012 11:18:18 -0700 Subject: [Freeswitch-users] Attended transfer via ESL In-Reply-To: <1336657979551-7546785.post@n2.nabble.com> References: <1335539693783-7506468.post@n2.nabble.com> <1336657979551-7546785.post@n2.nabble.com> Message-ID: Cool, thanks for letting us know it worked. -MC On Thu, May 10, 2012 at 6:52 AM, Limit wrote: > *mercutioviz*, thank you! > > Looks like it works for me! Transfer works just as called from dialplan. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120510/e6dfd60f/attachment.html From msc at freeswitch.org Thu May 10 22:19:16 2012 From: msc at freeswitch.org (Michael Collins) Date: Thu, 10 May 2012 11:19:16 -0700 Subject: [Freeswitch-users] Can't compile with ODBC support under FreeBSD for current git In-Reply-To: References: Message-ID: Could you help us out by filing a report on jira.freeswitch.org? It helps us keep bugs under control... Thanks, MC On Thu, May 10, 2012 at 11:07 AM, Nikolay Rogoshchenkov wrote: > Hello, > > I already have FreeSWITCH Version 1.1.beta1 (git-d730df7 2012-04-02 > 18-00-38 -0400) with ODBC support and my LUA scripts working properly with > MySQL. > But when I tried to *configure *the latest git (after git pull) I have > got the following error: > > ================================================================================ > > [....] > *** your path, or set the LIBGNUTLS_CONFIG environment variable to the > *** full path to libgnutls-config. > checking libtool major version... 2 > using libtool library extension... la > adding "-fPIC" to SWITCH_AM_CFLAGS > adding "-fPIC" to SWITCH_AM_CXXFLAGS > adding "-Werror" to SWITCH_AM_CFLAGS > checking whether the compiler supports -fvisibility=hidden... yes > adding "-fvisibility=hidden" to SWITCH_AM_CFLAGS > adding "-DSWITCH_API_VISIBILITY=1" to SWITCH_AM_CFLAGS > adding "-DHAVE_VISIBILITY=1" to SWITCH_AM_CFLAGS > adding "-fvisibility=hidden" to SWITCH_AM_CXXFLAGS > adding "-DSWITCH_API_VISIBILITY=1" to SWITCH_AM_CXXFLAGS > adding "-DHAVE_VISIBILITY=1" to SWITCH_AM_CXXFLAGS > checking CFLAGS for maximum ansi warnings... -Wall -std=c99 -pedantic > adding "-g" to SWITCH_AM_CFLAGS > adding "-ggdb" to SWITCH_AM_CFLAGS > checking for jack... gnome-config: not found > gnome-config: not found > checking for snd_pcm_open in -lasound... no > checking size of long... 8 > checking what directory libraries are found in... lib > checking for odbc header in /usr/include... no found > checking for odbc header in /usr/local/include... found > checking for SQLDisconnect in -lodbc... no > checking for odbc library in /usr/lib... no found > checking for odbc library in /usr/local/lib... found > checking whether to include odbc... yes > checking for SQLDisconnect in -lodbc... (cached) no > configure: error: no usable libodbc; please install unixodbc devel package > or equivalent > # > > ================================================================================ > > but why? > > Thank you. > -- > Nikolay > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120510/d521b490/attachment.html From msc at freeswitch.org Thu May 10 22:30:34 2012 From: msc at freeswitch.org (Michael Collins) Date: Thu, 10 May 2012 11:30:34 -0700 Subject: [Freeswitch-users] Phone not registering In-Reply-To: <4FABB4DF.5090200@softnet.si> References: <4FABB4DF.5090200@softnet.si> Message-ID: It's definitely a NAT issue. The phone is not responding to your 401 and is instead just sending another REGISTER packet. Notice that FS is responding on port 5060. Is that the port your phone is expecting to receive on? -MC On Thu, May 10, 2012 at 5:30 AM, Miha wrote: > Hi, > > here is pastebin of siptrace (http://pastebin.freeswitch.org/19029). > > Phone on local network are registered on FS. After I put between local > network and Phone router, phones are unable to registered on FS. > > On other softswitch which is not FS phones are registering (same port, > same scenario, etc.). > > Phones are SPA922. > > What could be causing the problem? > > Thanks! > > Miha > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120510/8f99a23d/attachment.html From chris at gonumina.com Thu May 10 22:33:48 2012 From: chris at gonumina.com (Chris Ferreira) Date: Thu, 10 May 2012 14:33:48 -0400 Subject: [Freeswitch-users] Fax Issues with Cisco SPA112 and T.38 In-Reply-To: References: Message-ID: Hi Bryan, Thanks for your response, much appreciated. 1. I have never had to do a debug other than analyze fs_cli output and iv'e never done a sip trace. I have started reading up on how to do these on the Wiki and will generate some files to take a look at. 2. I have heard this before about VPS's and even VMWare servers running on local physical hardware. I hope to be able to overcome these issues. 3. Other calls work fine throughout the switch and between carriers. I don't know if this is a good actual measure of anything, but VoIPSpear.com says that the servers MOS Score is a 4.0 . Other than the server running on a VPS, is my setup with the ATA registering to FreeSWITCH as an intermediary between Flowroute.com a correct practice? Or is there something else I could be doing? I know that if I was not needing to use an Analog Fax machine for sending/receiving that I would have other options. But unfortunately the Dinosaur must have it's Dial Tone. Thanks, -Chris On Thu, May 10, 2012 at 2:03 PM, Brian Foster wrote: > 1. Debug tells all. We need the siptrace as well. > > 2. VPS's are usually bad at sending/receiving faxes even if the media is > just being routed through it. > > 3. You could also look at session timers. Are other calls working? > > The only real way to know is if we have number 1 in our possession, > otherwise we just start guessing. > > -BDF > > On Thu, May 10, 2012 at 1:30 PM, Chris Ferreira wrote: > >> Hello All, >> >> >> >> I have a FreeSWITCH install running on CentOS 5.6 on a Linode VPS. I am >> trying to get my SPA112 (Version 1.1 Firmware) to send faxes (and >> eventually receive) successfully. I have the ATA registered as an extension >> and it's primary outgoing route is through Flowroute. I have T.38 enabled >> on the ATA and I have diabled ECM on the analog fax machine. All of the >> CDR's in Flowroute show that the test faxes are all ending their calls at >> 39 seconds. >> >> >> I can post screen shots of the ATA config or provide any other config >> info. >> >> >> Is there something I am missing, or should this setup work? >> >> I have poked around for a while for info, but unlike the PAP2T there is >> little info on these SPA112's. This is my first post asking for help as I >> usually try to resolve things on my own. But for this, I will defer >> to everyone's experience. >> >> >> >> Thanks, >> >> -Chris >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Brian D. Foster > Endigo Computer LLC > Email: bdfoster at endigotech.com > Phone: 317-800-7876 > Indianapolis, Indiana, USA > > This message contains confidential information and is intended for those > listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If > you are not the intended recipient you are notified that disclosing, > copying, distributing or taking any action in reliance on the contents of > this information is strictly prohibited. E-mail transmission cannot be > guaranteed to be secure or error-free as information could be intercepted, > corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. > The sender therefore does not accept liability for any errors or omissions > in the contents of this message, which arise as a result of e-mail > transmission. If verification is required please request a hard-copy > version. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120510/0b55a533/attachment-0001.html From nickolayr at gmail.com Thu May 10 22:35:16 2012 From: nickolayr at gmail.com (Nikolay Rogoshchenkov) Date: Thu, 10 May 2012 14:35:16 -0400 Subject: [Freeswitch-users] Can't compile with ODBC support under FreeBSD for current git In-Reply-To: References: Message-ID: Already done. Thank you. -- Nikolay On Thu, May 10, 2012 at 2:19 PM, Michael Collins wrote: > Could you help us out by filing a report on jira.freeswitch.org? It helps > us keep bugs under control... > Thanks, > MC > > On Thu, May 10, 2012 at 11:07 AM, Nikolay Rogoshchenkov < > nickolayr at gmail.com> wrote: > >> Hello, >> >> I already have FreeSWITCH Version 1.1.beta1 (git-d730df7 2012-04-02 >> 18-00-38 -0400) with ODBC support and my LUA scripts working properly with >> MySQL. >> But when I tried to *configure *the latest git (after git pull) I have >> got the following error: >> >> ================================================================================ >> >> [....] >> *** your path, or set the LIBGNUTLS_CONFIG environment variable to the >> *** full path to libgnutls-config. >> checking libtool major version... 2 >> using libtool library extension... la >> adding "-fPIC" to SWITCH_AM_CFLAGS >> adding "-fPIC" to SWITCH_AM_CXXFLAGS >> adding "-Werror" to SWITCH_AM_CFLAGS >> checking whether the compiler supports -fvisibility=hidden... yes >> adding "-fvisibility=hidden" to SWITCH_AM_CFLAGS >> adding "-DSWITCH_API_VISIBILITY=1" to SWITCH_AM_CFLAGS >> adding "-DHAVE_VISIBILITY=1" to SWITCH_AM_CFLAGS >> adding "-fvisibility=hidden" to SWITCH_AM_CXXFLAGS >> adding "-DSWITCH_API_VISIBILITY=1" to SWITCH_AM_CXXFLAGS >> adding "-DHAVE_VISIBILITY=1" to SWITCH_AM_CXXFLAGS >> checking CFLAGS for maximum ansi warnings... -Wall -std=c99 -pedantic >> adding "-g" to SWITCH_AM_CFLAGS >> adding "-ggdb" to SWITCH_AM_CFLAGS >> checking for jack... gnome-config: not found >> gnome-config: not found >> checking for snd_pcm_open in -lasound... no >> checking size of long... 8 >> checking what directory libraries are found in... lib >> checking for odbc header in /usr/include... no found >> checking for odbc header in /usr/local/include... found >> checking for SQLDisconnect in -lodbc... no >> checking for odbc library in /usr/lib... no found >> checking for odbc library in /usr/local/lib... found >> checking whether to include odbc... yes >> checking for SQLDisconnect in -lodbc... (cached) no >> configure: error: no usable libodbc; please install unixodbc devel >> package or equivalent >> # >> >> ================================================================================ >> >> but why? >> >> Thank you. >> -- >> Nikolay >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120510/ab9ecf76/attachment.html From koralu at gmail.com Thu May 10 22:39:53 2012 From: koralu at gmail.com (Adrian Andrei) Date: Thu, 10 May 2012 21:39:53 +0300 Subject: [Freeswitch-users] mod_xml_cdr is not logging In-Reply-To: References: Message-ID: The logs are: 2012-05-10 21:37:03.399431 [CONSOLE] switch_loadable_module.c:1765 Stopping: mod_xml_cdr 2012-05-10 21:37:03.399431 [NOTICE] switch_event.c:1935 Event Binding deleted for mod_xml_cdr:TRAP 2012-05-10 21:37:03.399431 [CONSOLE] switch_loadable_module.c:1785 mod_xml_cdr unloaded. 2012-05-10 21:37:05.459437 [NOTICE] mod_xml_cdr.c:99 Rotating log file paths +OK module unloaded +OK Reloading XML +OK module loaded 2012-05-10 21:37:05.459437 [NOTICE] mod_xml_cdr.c:126 Setting log file path to /usr/local/freeswitch/log/cdr 2012-05-10 21:37:05.459437 [NOTICE] mod_xml_cdr.c:164 Setting err log file path to /usr/local/freeswitch/log/cdr/errors 2012-05-10 21:37:05.459437 [CONSOLE] switch_loadable_module.c:1299 Successfully Loaded [mod_xml_cdr] freeswitch at peer1> 2012-05-10 21:37:05.459437 [INFO] mod_enum.c:812 ENUM Reloaded 2012-05-10 21:37:05.479431 [INFO] switch_time.c:1035 Timezone reloaded 530 definitions On Thu, May 10, 2012 at 9:17 PM, Michael Collins wrote: > > > On Thu, May 10, 2012 at 8:45 AM, Adrian Andrei wrote: > >> Same result. I tried both 127.0.0.1 and localhost. etc/hosts is valid. >> > what happens when you go to fs_cli and type: > reload mod_xml_cdr > > I'm curious. > -MC > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120510/2e22fef1/attachment.html From valernur at yahoo.com Thu May 10 23:13:52 2012 From: valernur at yahoo.com (Valer Nur) Date: Thu, 10 May 2012 12:13:52 -0700 (PDT) Subject: [Freeswitch-users] Choopy one-way noise In-Reply-To: <1336658560567-7546814.post@n2.nabble.com> References: <1336658560567-7546814.post@n2.nabble.com> Message-ID: <1336677232.8027.YahooMailNeo@web120406.mail.ne1.yahoo.com> Hi bawajan, I have listened to the files. What I can hear is a constant stationary noise. Usually the origin of such noise is hardware issues (mic/cables/etc.). If you are unable to find & fix the source of the problem, you can use noise cancellation software that will easily remove this noise. One software that you can try is PBXMate. A usage example is available on FreeSWITCH wiki: http://wiki.freeswitch.org/wiki/PBXMate-FreeSWITCH-integration Valer ________________________________ From: bawajan To: freeswitch-users at lists.freeswitch.org Sent: Thursday, May 10, 2012 5:02 PM Subject: Re: [Freeswitch-users] Choopy one-way noise Hi Moises Silva, am new to freeswitch and am facing same noise issue when making call from ISDN (freetdm) to SIP UA. As per above mail, i have did ftdm trace and listen the clips.? I found same noise in generated voice clip of line 2 (sip). am attached both audio files for your reference (ISDN line 1 and SIP line 2). Kindly let me know, what needs to do to rectify it. http://freeswitch-users.2379917.n2.nabble.com/file/n7546814/isdn_line1-in-s2c1.wav isdn_line1-in-s2c1.wav http://freeswitch-users.2379917.n2.nabble.com/file/n7546814/sip_line2-out-s2c1.wav sip_line2-out-s2c1.wav -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Choopy-one-way-noise-tp7332765p7546814.html Sent from the freeswitch-users mailing list archive at Nabble.com. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120510/8a896cd6/attachment.html From shane.harrison at paragon.co.nz Thu May 10 23:55:49 2012 From: shane.harrison at paragon.co.nz (Shane Harrison) Date: Fri, 11 May 2012 07:55:49 +1200 Subject: [Freeswitch-users] Freetdm DTMF In-Reply-To: References: Message-ID: Hi For calls originating from Freetdm extension I simply set up freetdm.conf.xml to use the XML dialplan and context fxs-ports like so: Then dialplan to get it into default context is: For calls into FreeTDM I have the following dialplan: Cheers Shane On Fri, May 11, 2012 at 2:07 AM, curriegrad2004 wrote: > It would be beneficial if you can post the dialplan on how FS handles > the part where FreeTDM comes into play. I.e. the dialplan that does > the transfer of the FreeTDM call into the default context. This seems > to be opening up a can of worms here with FXS cards in FreeTDM... > > On Wed, May 9, 2012 at 8:39 PM, Shane Harrison > wrote: > > Happy for you to laugh at yourself, I'm just happy you are finding some > time > > to take some interest in my problem. Much appreciated. > > > > I think some clarity is required here. I am simply trying to do an > attended > > transfer as per the wiki (except I set the bind_meta_app to both legs) > > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_att_xfer > > > > It works fine for SIP to SIP. For SIP to Freetdm it only works if the > SIP > > is trying to do the transfer. If the FreeTDM is trying to do it ie. > push *3 > > on the phone keypad, then the bind_meta_app works fine and detects the *3 > > and executes the appropriate extension ie. att_xfer, however the read() > in > > the extension att_xfer does not see the subsequent DTMF pressed on the > phone > > ie. the destination extension number. > > > > Cheers > > Shane > > > > > > On Thu, May 10, 2012 at 2:44 PM, curriegrad2004 < > curriegrad2004 at gmail.com> > > wrote: > >> > >> I just had to laugh at my self for mixing up the 2 again... > >> bind_meta_app is only applicable to that extension that the inbound > >> call was made to. Since you've transferred to another extension, the > >> bind_meta_app won't work anymore because it's not defined in the > >> extension you're transferring to. > >> > >> If you want this to happen, you'd have to manually define that > >> bind_meta_app to those target extensions too. Remember, do this at > >> your own peril - obvious misuse of bind_meta_app can open a huge > >> security hole if you don't know what you're doing :) > >> > >> On Wed, May 9, 2012 at 7:27 PM, Shane Harrison > >> wrote: > >> > Thanks for the thoughts. As I said, I am already setting it to both > >> > legs - > >> > I will try simply trying one leg but am sceptical :-) > >> > > >> > I also mentioned that I called the start_dtmf just before calling the > >> > read > >> > so unless I am doing something wrong here..... I'll try and post the > >> > XML > >> > tonight when I get home. Oh and it is an FXS card not an FXO of > course > >> > since it has a phone plugged into it. > >> > > >> > The question still remains though, why is the in-band DTMF detection > >> > working > >> > for the bind_meta_app digit detection but not after that? > >> > > >> > Cheers > >> > Shane > >> > > >> > > >> > On Thu, May 10, 2012 at 12:28 PM, curriegrad2004 > >> > > >> > wrote: > >> >> > >> >> and crap, since I wasn't even reading anything here, on the > subsequent > >> >> transfers from your FXO card, enable the in-band DTMF detector that > FS > >> >> has. The details on the in-band DTMF detector is here: > >> >> > >> >> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf > >> >> > >> >> But use this with caution, if there is a DTMF detector on the FXO > card > >> >> itself, make sure you disable it before using it. > >> >> > >> >> On Wed, May 9, 2012 at 5:26 PM, curriegrad2004 > >> >> > >> >> wrote: > >> >> > apologies for telling you the wrong thing. I was replying you from > my > >> >> > phone btw :P > >> >> > > >> >> > Yeah, bind_meta_app is the app you would use, but try changing it > to > >> >> > point to the b-leg, not the a-leg > >> >> > > >> >> > On Wed, May 9, 2012 at 4:12 PM, Shane Harrison > >> >> > wrote: > >> >> >> Thanks. I am currently using bind_meta_app (set to both legs) > >> >> >> already > >> >> >> rather than bind_digits. I'll give bind_digits a shot and see if > it > >> >> >> behaves > >> >> >> differently. > >> >> >> > >> >> >> Note that I do detect the initial *3 digits and because > >> >> >> bind_meta_app > >> >> >> is > >> >> >> both legs, this is successful no matter which direction the call > is > >> >> >> setup > >> >> >> from. However once the dialplan moves to the extension the *3 is > >> >> >> bound > >> >> >> to, > >> >> >> digits are no longer received. > >> >> >> > >> >> >> The worrying thing for me is that ftdm_io.c doesn't even write to > >> >> >> the > >> >> >> log > >> >> >> that it has received them (nor freetdm above that of course which > is > >> >> >> understandable) and I am surprised that the read() influences that > >> >> >> since it > >> >> >> works prior on the *3 digits. > >> >> >> > >> >> >> Cheers > >> >> >> Shane > >> >> >> > >> >> >> > >> >> >> > >> >> >> On Thu, May 10, 2012 at 10:22 AM, curriegrad2004 > >> >> >> > >> >> >> wrote: > >> >> >>> > >> >> >>> I'm guessing the bind digits in your analog card was set to > listen > >> >> >>> for > >> >> >>> this sequence on the a-leg given if the call was being routed > from > >> >> >>> the > >> >> >>> IP side to the analog side. > >> >> >>> > >> >> >>> Try changing that to listen on the b-leg. > >> >> >>> > >> >> >>> On 5/9/12, Shane Harrison wrote: > >> >> >>> > Hi All, > >> >> >>> > > >> >> >>> > Have a situation where I have a call between a SIP phone and a > >> >> >>> > FreeTDM > >> >> >>> > channel. Pushing *3 on the analog FreeTDM phone is detected > and > >> >> >>> > this > >> >> >>> > is > >> >> >>> > bound to a dialplan extension (attended transfer) that does a > >> >> >>> > read(): > >> >> >>> > > >> >> >>> > > >> >> >>> > However pushing further digits on the analog phone ie. > extension > >> >> >>> > number > >> >> >>> > of > >> >> >>> > phone we wish to do an attended transfer to , doesn't result in > >> >> >>> > the > >> >> >>> > DTMF > >> >> >>> > being detected. Note that this all works the other way around > >> >> >>> > ie. > >> >> >>> > using > >> >> >>> > the SIP phone. > >> >> >>> > > >> >> >>> > When the *3 digits are pushed on the analog phone I see the > logs > >> >> >>> > report: > >> >> >>> > > >> >> >>> > ftdm_io.c:3463 [s2c1][1:4] Queuing DTMF * (debug = 0) > >> >> >>> > mod_freetdm.c:702 Queuing DTMF [*] in channel FreeTDM/2:1/ > >> >> >>> > > >> >> >>> > ftdm_io.c:3463 [s2c1][1:4] Queuing DTMF 4 (debug = 0) > >> >> >>> > > >> >> >>> > mod_freetdm.c:702 Queuing DTMF [4] in channel FreeTDM/2:1/ > >> >> >>> > > >> >> >>> > When the further keys are pushed ftdm_io reports nothing. > >> >> >>> > > >> >> >>> > I have tried inserting a start_dtmf before the read() but it > had > >> >> >>> > no > >> >> >>> > effect. > >> >> >>> > Any thoughts as to why DTMF isn't being seen from the analog > >> >> >>> > phone > >> >> >>> > after > >> >> >>> > the read()? > >> >> >>> > > >> >> >>> > Cheers > >> >> >>> > Shane > >> >> >>> > > >> >> >>> > >> >> >>> > >> >> >>> > >> >> >>> > _________________________________________________________________________ > >> >> >>> Professional FreeSWITCH Consulting Services: > >> >> >>> consulting at freeswitch.org > >> >> >>> http://www.freeswitchsolutions.com > >> >> >>> > >> >> >>> > >> >> >>> > >> >> >>> > >> >> >>> Official FreeSWITCH Sites > >> >> >>> http://www.freeswitch.org > >> >> >>> http://wiki.freeswitch.org > >> >> >>> http://www.cluecon.com > >> >> >>> > >> >> >>> FreeSWITCH-users mailing list > >> >> >>> FreeSWITCH-users at lists.freeswitch.org > >> >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >> >>> > >> >> >>> > >> >> >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> >> >>> http://www.freeswitch.org > >> >> >> > >> >> >> > >> >> >> > >> >> >> > >> >> >> -- > >> >> >> Paragon Electronic Design Ltd > >> >> >> L6 Crest House > >> >> >> 92 Queens Drive > >> >> >> P0 Box 30449 > >> >> >> Lower Hutt 5040 > >> >> >> > >> >> >> +64 4 5703870 Extn 875 > >> >> >> +64 21 608919 (mobile) > >> >> >> > >> >> >> "Solving your problems with the right technology" > >> >> >> > >> >> >> > >> >> >> > >> >> >> > _________________________________________________________________________ > >> >> >> Professional FreeSWITCH Consulting Services: > >> >> >> consulting at freeswitch.org > >> >> >> http://www.freeswitchsolutions.com > >> >> >> > >> >> >> > >> >> >> > >> >> >> > >> >> >> Official FreeSWITCH Sites > >> >> >> http://www.freeswitch.org > >> >> >> http://wiki.freeswitch.org > >> >> >> http://www.cluecon.com > >> >> >> > >> >> >> FreeSWITCH-users mailing list > >> >> >> FreeSWITCH-users at 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 at freeswitch.org > >> >> http://www.freeswitchsolutions.com > >> >> > >> >> > >> >> > >> >> > >> >> Official FreeSWITCH Sites > >> >> http://www.freeswitch.org > >> >> http://wiki.freeswitch.org > >> >> http://www.cluecon.com > >> >> > >> >> FreeSWITCH-users mailing list > >> >> FreeSWITCH-users at lists.freeswitch.org > >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >> > >> >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> >> http://www.freeswitch.org > >> > > >> > > >> > > >> > > >> > -- > >> > Paragon Electronic Design Ltd > >> > L6 Crest House > >> > 92 Queens Drive > >> > P0 Box 30449 > >> > Lower Hutt 5040 > >> > > >> > +64 4 5703870 Extn 875 > >> > +64 21 608919 (mobile) > >> > > >> > "Solving your problems with the right technology" > >> > > >> > > >> > > _________________________________________________________________________ > >> > Professional FreeSWITCH Consulting Services: > >> > consulting at freeswitch.org > >> > http://www.freeswitchsolutions.com > >> > > >> > > >> > > >> > > >> > Official FreeSWITCH Sites > >> > http://www.freeswitch.org > >> > http://wiki.freeswitch.org > >> > http://www.cluecon.com > >> > > >> > FreeSWITCH-users mailing list > >> > FreeSWITCH-users at 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 at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > > > > > -- > > Paragon Electronic Design Ltd > > L6 Crest House > > 92 Queens Drive > > P0 Box 30449 > > Lower Hutt 5040 > > > > +64 4 5703870 Extn 875 > > +64 21 608919 (mobile) > > > > "Solving your problems with the right technology" > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Paragon Electronic Design Ltd L6 Crest House 92 Queens Drive P0 Box 30449 Lower Hutt 5040 +64 4 5703870 Extn 875 +64 21 608919 (mobile) "Solving your problems with the right technology" -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120511/fd602081/attachment-0001.html From tarik.bts.gi at gmail.com Thu May 10 23:43:17 2012 From: tarik.bts.gi at gmail.com (tarik.bts.gi at gmail.com) Date: Thu, 10 May 2012 19:43:17 +0000 Subject: [Freeswitch-users] Freeswitch hardware cofig Message-ID: <4fac1a56.02d6e00a.7161.ffff99e8@mx.google.com> What is the good hardware configuration that needs freeswitch to support more then 30 calls? We have now 1.6 GHz in the CPU and 2 GB in the mem but is not enough. From freeswitch at scottisheyes.com Fri May 11 00:53:03 2012 From: freeswitch at scottisheyes.com (James) Date: Thu, 10 May 2012 13:53:03 -0700 Subject: [Freeswitch-users] Freeswitch hardware cofig In-Reply-To: <4fac1a56.02d6e00a.7161.ffff99e8@mx.google.com> References: <4fac1a56.02d6e00a.7161.ffff99e8@mx.google.com> Message-ID: If you search, you'll probably find that this question has been asked several times - each time, the general response will be: It depends on what you're trying to accomplish. http://wiki.freeswitch.org/wiki/Specsheet#Minimum.2FRecommended_System_Requirements http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations#Recommended_hardware.2FOS There is no "real" answer, but if you supply more information as to what your FreeSWITCH installation will be doing, you might get a helpful response from one of the experts. On Thu, May 10, 2012 at 12:43 PM, wrote: > What is the good hardware configuration > that needs freeswitch to support more then 30 calls? We have now 1.6 GHz > in the CPU and 2 GB in the mem but is not enough. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120510/77836070/attachment.html From msc at freeswitch.org Fri May 11 00:59:00 2012 From: msc at freeswitch.org (Michael Collins) Date: Thu, 10 May 2012 13:59:00 -0700 Subject: [Freeswitch-users] mod_xml_cdr is not logging In-Reply-To: References: Message-ID: Anything in /usr/local/freeswitch/log/cdr/errors ? -MC On Thu, May 10, 2012 at 11:39 AM, Adrian Andrei wrote: > The logs are: > > 2012-05-10 21:37:03.399431 [CONSOLE] switch_loadable_module.c:1765 > Stopping: mod_xml_cdr > 2012-05-10 21:37:03.399431 [NOTICE] switch_event.c:1935 Event Binding > deleted for mod_xml_cdr:TRAP > 2012-05-10 21:37:03.399431 [CONSOLE] switch_loadable_module.c:1785 > mod_xml_cdr unloaded. > 2012-05-10 21:37:05.459437 [NOTICE] mod_xml_cdr.c:99 Rotating log file > paths > > +OK module unloaded > +OK Reloading XML > +OK module loaded > > 2012-05-10 21:37:05.459437 [NOTICE] mod_xml_cdr.c:126 Setting log file > path to /usr/local/freeswitch/log/cdr > 2012-05-10 21:37:05.459437 [NOTICE] mod_xml_cdr.c:164 Setting err log file > path to /usr/local/freeswitch/log/cdr/errors > 2012-05-10 21:37:05.459437 [CONSOLE] switch_loadable_module.c:1299 > Successfully Loaded [mod_xml_cdr] > freeswitch at peer1> 2012-05-10 21:37:05.459437 [INFO] mod_enum.c:812 ENUM > Reloaded > 2012-05-10 21:37:05.479431 [INFO] switch_time.c:1035 Timezone reloaded 530 > definitions > > > On Thu, May 10, 2012 at 9:17 PM, Michael Collins wrote: > >> >> >> On Thu, May 10, 2012 at 8:45 AM, Adrian Andrei wrote: >> >>> Same result. I tried both 127.0.0.1 and localhost. etc/hosts is valid. >>> >> what happens when you go to fs_cli and type: >> reload mod_xml_cdr >> >> I'm curious. >> -MC >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120510/95e4db94/attachment.html From chris at opencsta.org Fri May 11 02:12:24 2012 From: chris at opencsta.org (Chris Mylonas) Date: Fri, 11 May 2012 08:12:24 +1000 Subject: [Freeswitch-users] macports installed - does this create compilation problem Message-ID: <990069C5-9DBF-473C-B6ED-24D396EAF241@opencsta.org> Hi List, In the distant past I had FS running on my laptop (OSX 10.6.x) It hasn't compiled for months. I just found this JIRA http://jira.freeswitch.org/browse/FS-3473 There is a mention that the build environment is stuffed possibly because of macports. My build stuffs up on gsm..... and spandsp I have commented out, sic, applications/spandsp in modules.conf and now this is the error making all mod_spidermonkey cd config; make -j1 export cd pr; make -j1 export cd include; make export cd md; make export cd private; make export cd obsolete; make export cd src; make export cd io; make export cd malloc; make export cd md; make export cd unix; make export os_Darwin_ppc.s:49:no such instruction: `lwarx r4,0,r3' os_Darwin_ppc.s:50:no such instruction: `addi r0,r4,1' os_Darwin_ppc.s:51:invalid character '.' in mnemonic os_Darwin_ppc.s:52:invalid character '-' in mnemonic os_Darwin_ppc.s:53:no such instruction: `mr r3,r0' os_Darwin_ppc.s:54:no such instruction: `blr' os_Darwin_ppc.s:62:no such instruction: `lwarx r4,0,r3' os_Darwin_ppc.s:63:no such instruction: `addi r0,r4,-1' os_Darwin_ppc.s:64:invalid character '.' in mnemonic os_Darwin_ppc.s:65:invalid character '-' in mnemonic os_Darwin_ppc.s:66:no such instruction: `mr r3,r0' os_Darwin_ppc.s:67:no such instruction: `blr' os_Darwin_ppc.s:75:no such instruction: `lwarx r5,0,r3' os_Darwin_ppc.s:76:invalid character '.' in mnemonic os_Darwin_ppc.s:77:invalid character '-' in mnemonic os_Darwin_ppc.s:78:no such instruction: `mr r3,r5' os_Darwin_ppc.s:79:no such instruction: `blr' os_Darwin_ppc.s:87:no such instruction: `lwarx r5,0,r3' os_Darwin_ppc.s:88:too many memory references for `add' os_Darwin_ppc.s:89:invalid character '.' in mnemonic os_Darwin_ppc.s:90:invalid character '-' in mnemonic os_Darwin_ppc.s:91:no such instruction: `mr r3,r0' os_Darwin_ppc.s:92:no such instruction: `blr' make[10]: *** [os_Darwin_ppc.o] Error 1 make[9]: *** [export] Error 2 make[8]: *** [export] Error 2 make[7]: *** [export] Error 2 make[6]: *** [export] Error 2 make[5]: *** [/Users/chrismylonas/freeswitch-git/freeswitch/libs/js/libjs.la] Error 2 make[4]: *** [mod_spidermonkey-all] Error 1 make[3]: *** [all-recursive] Error 1 make[2]: *** [all-recursive] Error 1 make[1]: *** [all] Error 2 make: *** [current] Error 2 Can this be confirmed to be a macports problem? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120511/841b4dac/attachment.html From msc at freeswitch.org Fri May 11 02:18:13 2012 From: msc at freeswitch.org (Michael Collins) Date: Thu, 10 May 2012 15:18:13 -0700 Subject: [Freeswitch-users] Freeswitch hardware cofig In-Reply-To: <4fac1a56.02d6e00a.7161.ffff99e8@mx.google.com> References: <4fac1a56.02d6e00a.7161.ffff99e8@mx.google.com> Message-ID: On Thu, May 10, 2012 at 12:43 PM, wrote: > What is the good hardware configuration > that needs freeswitch to support more then 30 calls? We have now 1.6 GHz > in the CPU and 2 GB in the mem but is not enough. > What kind of processor? What operating system? Do you want 30 simultaneous calls or 30 new calls per second? Will you be doing transcoding on these calls? All those details will help you decide what to do next. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120510/14e50697/attachment.html From curriegrad2004 at gmail.com Fri May 11 03:48:36 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Thu, 10 May 2012 16:48:36 -0700 Subject: [Freeswitch-users] Freetdm DTMF In-Reply-To: References: Message-ID: I would try adding right before the bridge action for the dialplan for your FreeTDM FXS extensions. You may want to also apply the bind_meta_app actions on the dialplan for your FreeTDM outgoing calls for features like attended transfer to work. Hope this helps On Thu, May 10, 2012 at 12:55 PM, Shane Harrison wrote: > Hi > > For calls originating from Freetdm extension I simply set up > freetdm.conf.xml to use the XML dialplan and context fxs-ports like so: > > > > > > > > > > > > > > > Then dialplan to get it into default context is: > > > ?? ? > ?? ? ? > ?? ? ? ? data="toll_allow=local,domestic,international"/> > ?? ? ? ? > ?? ? ? > ?? ? > ?? > > > For calls into FreeTDM I have the following dialplan: > > > ?? ? ? ? ? > ?? ? ? ? > ?? ? ? ? > ?? ? ? ? > ?? ? ? ? > ?? ? ? ? > ?? ? ? ? > ?? ? ? ? > ?? ? ? ? > ?? ? ? ? > ?? ? ? ? > ?? ? ? ? > ?? ? ? ? > ?? ? ? ? > ?? ? ? ? > ?? ? ? ? data="insert/${domain_name}-call_return/300/${caller_id_number}"/> > ?? ? ? ? data="insert/${domain_name}-last_dial_ext/300/${uuid}"/> > ?? ? ? ? data="called_party_callgroup=${user_data(300@${domain_name} var > callgroup)}"/> > ?? ? ? ? > ?? ? ? ? data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/> > ?? ? ? ? > ?? ? ? ? > ?? ? ? ? > ?? ? ? ? > ?? ? ? ? > ?? ? ? > ?? ? > > > > Cheers > Shane > > > On Fri, May 11, 2012 at 2:07 AM, curriegrad2004 > wrote: >> >> It would be beneficial if you can post the dialplan on how FS handles >> the part where FreeTDM comes into play. I.e. the dialplan that does >> the transfer of the FreeTDM call into the default context. This seems >> to be opening up a can of worms here with FXS cards in FreeTDM... >> >> On Wed, May 9, 2012 at 8:39 PM, Shane Harrison >> wrote: >> > Happy for you to laugh at yourself, I'm just happy you are finding some >> > time >> > to take some interest in my problem.? Much appreciated. >> > >> > I think some clarity is required here.? I am simply trying to do an >> > attended >> > transfer as per the wiki (except I set the bind_meta_app to both legs) >> > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_att_xfer >> > >> > It works? fine for SIP to SIP.? For SIP to Freetdm it only works if the >> > SIP >> > is trying to do the transfer.? If the FreeTDM is trying to do it ie. >> > push *3 >> > on the phone keypad, then the bind_meta_app works fine and detects the >> > *3 >> > and executes the appropriate extension ie. att_xfer, however the read() >> > in >> > the extension att_xfer does not see the subsequent DTMF pressed on the >> > phone >> > ie. the destination extension number. >> > >> > Cheers >> > Shane >> > >> > >> > On Thu, May 10, 2012 at 2:44 PM, curriegrad2004 >> > >> > wrote: >> >> >> >> I just had to laugh at my self for mixing up the 2 again... >> >> bind_meta_app is only applicable to that extension that the inbound >> >> call was made to. Since you've transferred to another extension, the >> >> bind_meta_app won't work anymore because it's not defined in the >> >> extension you're transferring to. >> >> >> >> If you want this to happen, you'd have to manually define that >> >> bind_meta_app to those target extensions too. Remember, do this at >> >> your own peril - obvious misuse of bind_meta_app can open a huge >> >> security hole if you don't know what you're doing :) >> >> >> >> On Wed, May 9, 2012 at 7:27 PM, Shane Harrison >> >> wrote: >> >> > Thanks for the thoughts.? As I said, I am already setting it to both >> >> > legs - >> >> > I will try simply trying one leg but am sceptical :-) >> >> > >> >> > I also mentioned that I called the start_dtmf just before calling the >> >> > read >> >> > so unless I am doing something wrong here.....? I'll try and post the >> >> > XML >> >> > tonight when I get home.? Oh and it is an FXS card not an FXO of >> >> > course >> >> > since it has a phone plugged into it. >> >> > >> >> > The question still remains though, why is the in-band DTMF detection >> >> > working >> >> > for the bind_meta_app digit detection but not after that? >> >> > >> >> > Cheers >> >> > Shane >> >> > >> >> > >> >> > On Thu, May 10, 2012 at 12:28 PM, curriegrad2004 >> >> > >> >> > wrote: >> >> >> >> >> >> and crap, since I wasn't even reading anything here, on the >> >> >> subsequent >> >> >> transfers from your FXO card, enable the in-band DTMF detector that >> >> >> FS >> >> >> has. The details on the in-band DTMF detector is here: >> >> >> >> >> >> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf >> >> >> >> >> >> But use this with caution, if there is a DTMF detector on the FXO >> >> >> card >> >> >> itself, make sure you disable it before using it. >> >> >> >> >> >> On Wed, May 9, 2012 at 5:26 PM, curriegrad2004 >> >> >> >> >> >> wrote: >> >> >> > apologies for telling you the wrong thing. I was replying you from >> >> >> > my >> >> >> > phone btw :P >> >> >> > >> >> >> > Yeah, bind_meta_app is the app you would use, but try changing it >> >> >> > to >> >> >> > point to the b-leg, not the a-leg >> >> >> > >> >> >> > On Wed, May 9, 2012 at 4:12 PM, Shane Harrison >> >> >> > wrote: >> >> >> >> Thanks.? I am currently using bind_meta_app (set to both legs) >> >> >> >> already >> >> >> >> rather than bind_digits.? I'll give bind_digits a shot and see if >> >> >> >> it >> >> >> >> behaves >> >> >> >> differently. >> >> >> >> >> >> >> >> Note that I do detect the initial *3 digits and because >> >> >> >> bind_meta_app >> >> >> >> is >> >> >> >> both legs, this is successful no matter which direction the call >> >> >> >> is >> >> >> >> setup >> >> >> >> from.? However once the dialplan moves to the extension the *3 is >> >> >> >> bound >> >> >> >> to, >> >> >> >> digits are no longer received. >> >> >> >> >> >> >> >> The worrying thing for me is that ftdm_io.c doesn't even write to >> >> >> >> the >> >> >> >> log >> >> >> >> that it has received them (nor freetdm above that of course which >> >> >> >> is >> >> >> >> understandable) and I am surprised that the read() influences >> >> >> >> that >> >> >> >> since it >> >> >> >> works prior on the *3 digits. >> >> >> >> >> >> >> >> Cheers >> >> >> >> Shane >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> On Thu, May 10, 2012 at 10:22 AM, curriegrad2004 >> >> >> >> >> >> >> >> wrote: >> >> >> >>> >> >> >> >>> I'm guessing the bind digits in your analog card was set to >> >> >> >>> listen >> >> >> >>> for >> >> >> >>> this sequence on the a-leg given if the call was being routed >> >> >> >>> from >> >> >> >>> the >> >> >> >>> IP side to the analog side. >> >> >> >>> >> >> >> >>> Try changing that to listen on the b-leg. >> >> >> >>> >> >> >> >>> On 5/9/12, Shane Harrison wrote: >> >> >> >>> > Hi All, >> >> >> >>> > >> >> >> >>> > Have a situation where I have a call between a SIP phone and a >> >> >> >>> > FreeTDM >> >> >> >>> > channel. ? Pushing *3 on the analog FreeTDM phone is detected >> >> >> >>> > and >> >> >> >>> > this >> >> >> >>> > is >> >> >> >>> > bound to a dialplan extension (attended transfer) that does a >> >> >> >>> > read(): >> >> >> >>> > >> >> >> >>> > >> >> >> >>> > However pushing further digits on the analog phone ie. >> >> >> >>> > extension >> >> >> >>> > number >> >> >> >>> > of >> >> >> >>> > phone we wish to do an attended transfer to , doesn't result >> >> >> >>> > in >> >> >> >>> > the >> >> >> >>> > DTMF >> >> >> >>> > being detected. ?Note that this all works the other way around >> >> >> >>> > ie. >> >> >> >>> > using >> >> >> >>> > the SIP phone. >> >> >> >>> > >> >> >> >>> > When the *3 digits are pushed on the analog phone I see the >> >> >> >>> > logs >> >> >> >>> > report: >> >> >> >>> > >> >> >> >>> > ?ftdm_io.c:3463 [s2c1][1:4] Queuing DTMF * (debug = 0) >> >> >> >>> > mod_freetdm.c:702 Queuing DTMF [*] in channel FreeTDM/2:1/ >> >> >> >>> > >> >> >> >>> > ftdm_io.c:3463 [s2c1][1:4] Queuing DTMF 4 (debug = 0) >> >> >> >>> > >> >> >> >>> > mod_freetdm.c:702 Queuing DTMF [4] in channel FreeTDM/2:1/ >> >> >> >>> > >> >> >> >>> > When the further keys are pushed ftdm_io reports nothing. >> >> >> >>> > >> >> >> >>> > I have tried inserting a start_dtmf before the read() but it >> >> >> >>> > had >> >> >> >>> > no >> >> >> >>> > effect. >> >> >> >>> > Any thoughts as to why DTMF isn't being seen from the analog >> >> >> >>> > phone >> >> >> >>> > after >> >> >> >>> > the read()? >> >> >> >>> > >> >> >> >>> > Cheers >> >> >> >>> > Shane >> >> >> >>> > >> >> >> >>> >> >> >> >>> >> >> >> >>> >> >> >> >>> >> >> >> >>> _________________________________________________________________________ >> >> >> >>> Professional FreeSWITCH Consulting Services: >> >> >> >>> consulting at freeswitch.org >> >> >> >>> http://www.freeswitchsolutions.com >> >> >> >>> >> >> >> >>> >> >> >> >>> >> >> >> >>> >> >> >> >>> Official FreeSWITCH Sites >> >> >> >>> http://www.freeswitch.org >> >> >> >>> http://wiki.freeswitch.org >> >> >> >>> http://www.cluecon.com >> >> >> >>> >> >> >> >>> FreeSWITCH-users mailing list >> >> >> >>> FreeSWITCH-users at lists.freeswitch.org >> >> >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >>> >> >> >> >>> >> >> >> >>> >> >> >> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> >> >>> http://www.freeswitch.org >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> -- >> >> >> >> Paragon Electronic Design Ltd >> >> >> >> L6 Crest House >> >> >> >> 92 Queens Drive >> >> >> >> P0 Box 30449 >> >> >> >> Lower Hutt 5040 >> >> >> >> >> >> >> >> +64 4 5703870 Extn 875 >> >> >> >> +64 21 608919? (mobile) >> >> >> >> >> >> >> >> "Solving your problems with the right technology" >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> _________________________________________________________________________ >> >> >> >> Professional FreeSWITCH Consulting Services: >> >> >> >> consulting at freeswitch.org >> >> >> >> http://www.freeswitchsolutions.com >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> Official FreeSWITCH Sites >> >> >> >> http://www.freeswitch.org >> >> >> >> http://wiki.freeswitch.org >> >> >> >> http://www.cluecon.com >> >> >> >> >> >> >> >> FreeSWITCH-users mailing list >> >> >> >> FreeSWITCH-users at 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 at freeswitch.org >> >> >> http://www.freeswitchsolutions.com >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> Official FreeSWITCH Sites >> >> >> http://www.freeswitch.org >> >> >> http://wiki.freeswitch.org >> >> >> http://www.cluecon.com >> >> >> >> >> >> FreeSWITCH-users mailing list >> >> >> FreeSWITCH-users at lists.freeswitch.org >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> >> >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> >> http://www.freeswitch.org >> >> > >> >> > >> >> > >> >> > >> >> > -- >> >> > Paragon Electronic Design Ltd >> >> > L6 Crest House >> >> > 92 Queens Drive >> >> > P0 Box 30449 >> >> > Lower Hutt 5040 >> >> > >> >> > +64 4 5703870 Extn 875 >> >> > +64 21 608919? (mobile) >> >> > >> >> > "Solving your problems with the right technology" >> >> > >> >> > >> >> > >> >> > _________________________________________________________________________ >> >> > Professional FreeSWITCH Consulting Services: >> >> > consulting at freeswitch.org >> >> > http://www.freeswitchsolutions.com >> >> > >> >> > >> >> > >> >> > >> >> > Official FreeSWITCH Sites >> >> > http://www.freeswitch.org >> >> > http://wiki.freeswitch.org >> >> > http://www.cluecon.com >> >> > >> >> > FreeSWITCH-users mailing list >> >> > FreeSWITCH-users at 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 at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> >> >> >> >> >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> http://wiki.freeswitch.org >> >> http://www.cluecon.com >> >> >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> > >> > >> > >> > >> > -- >> > Paragon Electronic Design Ltd >> > L6 Crest House >> > 92 Queens Drive >> > P0 Box 30449 >> > Lower Hutt 5040 >> > >> > +64 4 5703870 Extn 875 >> > +64 21 608919? (mobile) >> > >> > "Solving your problems with the right technology" >> > >> > >> > _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at 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 at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > > -- > Paragon Electronic Design Ltd > L6 Crest House > 92 Queens Drive > P0 Box 30449 > Lower Hutt 5040 > > +64 4 5703870 Extn 875 > +64 21 608919? (mobile) > > "Solving your problems with the right technology" > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From curriegrad2004 at gmail.com Fri May 11 03:53:56 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Thu, 10 May 2012 16:53:56 -0700 Subject: [Freeswitch-users] Freeswitch hardware cofig In-Reply-To: References: <4fac1a56.02d6e00a.7161.ffff99e8@mx.google.com> Message-ID: If you want 30 concurrent calls, that setup *should* handle it just fine. As the others have said, your mileage may vary so, don't take my judgement as is. You'd probably want to do more research in this aspect. Besides, a 1.6GHz Intel Atom is going to be slightly slower than a 1.6GHz Sandy Bridge Celeron processor, so simply having a CPU clock cycle figure is meaningless unless you know how well your processor performs in your desired application. On Thu, May 10, 2012 at 3:18 PM, Michael Collins wrote: > > > On Thu, May 10, 2012 at 12:43 PM, wrote: >> >> What is the good hardware configuration >> that needs freeswitch to support more then 30 calls? We have now 1.6 GHz >> in the CPU and 2 GB in the mem but is not enough. > > What kind of processor? What operating system? Do you want 30 simultaneous > calls or 30 new calls per second? Will you be doing transcoding on these > calls? > > All those details will help you decide what to do next. > > -MC > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From chris at opencsta.org Fri May 11 04:01:20 2012 From: chris at opencsta.org (Chris Mylonas) Date: Fri, 11 May 2012 10:01:20 +1000 Subject: [Freeswitch-users] FreeSWITCH, OSX, Libtool, Macports Message-ID: <523D8618-21D9-442E-9D23-4321EAFA6558@opencsta.org> Hi FS Users, tl;dr; - removed macports, removed tree, pulled fresh tree = installed FS on OSX. Here is the longer version that had a bit of a pre-emptive whinge about libtool version mismatch. But we never got there - it worked! I'm in the process of re-installing FS on OSX (10.6.8). I have removed Macports to try and get this up and running. I re-bootstrapped and got this error: quiet_libtool: Version mismatch error. This is libtool 2.4, but the quiet_libtool: definition of this LT_INIT comes from libtool 2.2.4. quiet_libtool: You should recreate aclocal.m4 with macros from libtool 2.4 quiet_libtool: and run autoconf again. make[1]: *** [libfreeswitch_la-switch_apr.lo] Error 63 make: *** [all] Error 2 Removed the whole git tree just in case there was some left over junk. Pulled a fresh tree and noticed this remark during a fresh bootstrap arakis:freeswitch-git chrismylonas$ rm -Rf freeswitch/ arakis:freeswitch-git chrismylonas$ git clone git://git.freeswitch.org/freeswitch.git Cloning into freeswitch... remote: Counting objects: 185609, done. remote: Compressing objects: 100% (39208/39208), done. remote: Total 185609 (delta 143241), reused 181443 (delta 140035) Receiving objects: 100% (185609/185609), 77.85 MiB | 267 KiB/s, done. Resolving deltas: 100% (143241/143241), done. arakis:freeswitch-git chrismylonas$ cd freeswitch/ arakis:freeswitch chrismylonas$ ./bootstrap.sh bootstrap: checking installation... bootstrap: autoconf version 2.61 (ok) bootstrap: automake version 1.10 (ok) bootstrap: aclocal version 1.10 (ok) bootstrap: libtool version 2.2.4 (ok) Bootstrapping using: autoconf : /usr/bin/autoconf automake : /usr/bin/automake aclocal : /usr/bin/aclocal libtool : /usr/bin/glibtool (2.2.4.) libtoolize: /usr/bin/glibtoolize make : /usr/bin/make (GNU Make 3.81) awk : () It is still reporting a 2.2.4 version of libtool. ... ... ... In the end, it has been compiled and installed though -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120511/14fd930b/attachment.html From bdfoster at endigotech.com Fri May 11 04:09:53 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Thu, 10 May 2012 20:09:53 -0400 Subject: [Freeswitch-users] Freeswitch hardware cofig In-Reply-To: References: <4fac1a56.02d6e00a.7161.ffff99e8@mx.google.com> Message-ID: Agreed with other's assessment of the situation, you're leaving out too many details such as intended usage, OS, chipsets, etc. I can tell you that i have ran FS on an extremely old laptop (1.0 GHZ Single-Core AMD w/512MB RAM) and got 40 calls doing no transcoding. I could only do a couple of calls with transcoding, then it would just crap out. That was on Debian Sid. So, there's a baseline. Sadly, I don't remember any other details on the machine than those that I've listed. On the other hand, I've gotten 6000 concurrent calls up at 150-200 CPS, then the database crapped out. That one was tuned extensively, and was also running on an 8 core machine at 1.8 Ghz and about 4GB RAM. There are a few others in this community who have done much better than that (probably with better equipment). YMMV on all of this. There are so many factors involved that it's hard to paint a picture unless you can be as detailed as humanly possible, and even then sometimes that's not enough. -BDF On Thu, May 10, 2012 at 7:53 PM, curriegrad2004 wrote: > If you want 30 concurrent calls, that setup *should* handle it just > fine. As the others have said, your mileage may vary so, don't take my > judgement as is. You'd probably want to do more research in this > aspect. > > Besides, a 1.6GHz Intel Atom is going to be slightly slower than a > 1.6GHz Sandy Bridge Celeron processor, so simply having a CPU clock > cycle figure is meaningless unless you know how well your processor > performs in your desired application. > > On Thu, May 10, 2012 at 3:18 PM, Michael Collins > wrote: > > > > > > On Thu, May 10, 2012 at 12:43 PM, wrote: > >> > >> What is the good hardware configuration > >> that needs freeswitch to support more then 30 calls? We have now 1.6 GHz > >> in the CPU and 2 GB in the mem but is not enough. > > > > What kind of processor? What operating system? Do you want 30 > simultaneous > > calls or 30 new calls per second? Will you be doing transcoding on these > > calls? > > > > All those details will help you decide what to do next. > > > > -MC > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-800-7876 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120510/f7f631fa/attachment.html From mario_fs at mgtech.com Fri May 11 05:48:00 2012 From: mario_fs at mgtech.com (Mario G) Date: Thu, 10 May 2012 18:48:00 -0700 Subject: [Freeswitch-users] FreeSWITCH, OSX, Libtool, Macports In-Reply-To: <523D8618-21D9-442E-9D23-4321EAFA6558@opencsta.org> References: <523D8618-21D9-442E-9D23-4321EAFA6558@opencsta.org> Message-ID: <7FFFF20B-D26F-4EDC-A19F-6AE1C48D5048@mgtech.com> Chris, I have been running FS on 10.6,x since 2010 fine and have been updating a lot recently to help testing, I have had no problems. I don't use macport stuff. I wrote the original FS osX wiki but there are some differences since 2010 (I am working on fixing the wiki in the next few weeks). The wiki install instructions work if you add the following. Hope this helps: 1. Add this info: FreeSWITCH? has many functions that invoke PKG-CONFIG, so it must be downloaded and installed separately. # Got to [http://pkgconfig.freedesktop.org/releases/ here] and download the latest pkg-config, by default it is placed into your Downloads folder. MUST USE .25 for OSXdue to glib2 dependencies! # Open Downloads and click the file to uncompress it. # Launch the Terminal application if not already running and issue these commands to move the source to the src directory, build and install: cd ~/Downloads mv pkg-config-0.25 /usr/local/src cd /usr/local/src/pkg-config-0.25 ./configure make sudo make install Hi FS Users, > > tl;dr; - removed macports, removed tree, pulled fresh tree = installed FS on OSX. > > > Here is the longer version that had a bit of a pre-emptive whinge about libtool version mismatch. But we never got there - it worked! > > I'm in the process of re-installing FS on OSX (10.6.8). I have removed Macports to try and get this up and running. > I re-bootstrapped and got this error: > > quiet_libtool: Version mismatch error. This is libtool 2.4, but the > quiet_libtool: definition of this LT_INIT comes from libtool 2.2.4. > quiet_libtool: You should recreate aclocal.m4 with macros from libtool 2.4 > quiet_libtool: and run autoconf again. > make[1]: *** [libfreeswitch_la-switch_apr.lo] Error 63 > make: *** [all] Error 2 > > Removed the whole git tree just in case there was some left over junk. Pulled a fresh tree and noticed this remark during a fresh bootstrap > > arakis:freeswitch-git chrismylonas$ rm -Rf freeswitch/ > arakis:freeswitch-git chrismylonas$ git clone git://git.freeswitch.org/freeswitch.git > Cloning into freeswitch... > remote: Counting objects: 185609, done. > remote: Compressing objects: 100% (39208/39208), done. > remote: Total 185609 (delta 143241), reused 181443 (delta 140035) > Receiving objects: 100% (185609/185609), 77.85 MiB | 267 KiB/s, done. > Resolving deltas: 100% (143241/143241), done. > arakis:freeswitch-git chrismylonas$ cd freeswitch/ > arakis:freeswitch chrismylonas$ ./bootstrap.sh > bootstrap: checking installation... > bootstrap: autoconf version 2.61 (ok) > bootstrap: automake version 1.10 (ok) > bootstrap: aclocal version 1.10 (ok) > bootstrap: libtool version 2.2.4 (ok) > Bootstrapping using: > autoconf : /usr/bin/autoconf > automake : /usr/bin/automake > aclocal : /usr/bin/aclocal > libtool : /usr/bin/glibtool (2.2.4.) > libtoolize: /usr/bin/glibtoolize > make : /usr/bin/make (GNU Make 3.81) > awk : () > > It is still reporting a 2.2.4 version of libtool. > ... > ... > ... > In the end, it has been compiled and installed though > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120510/86fcaaa6/attachment-0001.html From bdfoster at endigotech.com Fri May 11 06:16:12 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Thu, 10 May 2012 22:16:12 -0400 Subject: [Freeswitch-users] Fax Issues with Cisco SPA112 and T.38 In-Reply-To: References: Message-ID: 1. Run fs_cli and do 'sofia global siptrace on' without quotes. Stay in the cli and run a call in an instance where it would fail. Make sure your scroll buffer on your SSH terminal (PuTTY or what ever you have) to unlimited or something bog like 20K lines. Copy the logs out and put them on http://pastebin.freeswitch.org. If prompted for a username/password, please read the dialog box. SpanDSP doesn't have a great ability to be degugged on the click but it will give is something to go on. 2. There usually isn't much you can do as far as ensuring the VPS is capable of keeping the timing right. VoIP is reliant heavily on a good timing source. Faxing is even more dependent. I'm not really sure what the MOS score is useful for, maybe someone else can chime in on that. Something you need to make sure you are doing (specifically with flowroute) is making sure you are doing a t.38 reinvite. Flowroute does not auto detect t.38. Also freeswitch might have to transcode from audio ulaw/alaw to t.38 if your FAX machine doesn't actually support t.38. I don't really know much about the new SPA 112/122's but usually the ATA has a t.38 passthru mode that is auto detected. That might be what you are referring to. -BDF On May 10, 2012 2:35 PM, "Chris Ferreira" wrote: > Hi Bryan, > > > Thanks for your response, much appreciated. > > > 1. I have never had to do a debug other than analyze fs_cli output and > iv'e never done a sip trace. I have started reading up on how to do these > on the Wiki and will generate some files to take a look at. > > > 2. I have heard this before about VPS's and even VMWare servers running on > local physical hardware. I hope to be able to overcome these issues. > > > 3. Other calls work fine throughout the switch and between carriers. > > > > I don't know if this is a good actual measure of anything, but > VoIPSpear.com says that the servers MOS Score is a 4.0 . > > > Other than the server running on a VPS, is my setup with the ATA > registering to FreeSWITCH as an intermediary between Flowroute.com a > correct practice? Or is there something else I could be doing? > > > I know that if I was not needing to use an Analog Fax machine for > sending/receiving that I would have other options. But unfortunately the > Dinosaur must have it's Dial Tone. > > > > Thanks, > > -Chris > > > > > On Thu, May 10, 2012 at 2:03 PM, Brian Foster wrote: > >> 1. Debug tells all. We need the siptrace as well. >> >> 2. VPS's are usually bad at sending/receiving faxes even if the media is >> just being routed through it. >> >> 3. You could also look at session timers. Are other calls working? >> >> The only real way to know is if we have number 1 in our possession, >> otherwise we just start guessing. >> >> -BDF >> >> On Thu, May 10, 2012 at 1:30 PM, Chris Ferreira wrote: >> >>> Hello All, >>> >>> >>> >>> I have a FreeSWITCH install running on CentOS 5.6 on a Linode VPS. I am >>> trying to get my SPA112 (Version 1.1 Firmware) to send faxes (and >>> eventually receive) successfully. I have the ATA registered as an extension >>> and it's primary outgoing route is through Flowroute. I have T.38 enabled >>> on the ATA and I have diabled ECM on the analog fax machine. All of the >>> CDR's in Flowroute show that the test faxes are all ending their calls at >>> 39 seconds. >>> >>> >>> I can post screen shots of the ATA config or provide any other config >>> info. >>> >>> >>> Is there something I am missing, or should this setup work? >>> >>> I have poked around for a while for info, but unlike the PAP2T there is >>> little info on these SPA112's. This is my first post asking for help as I >>> usually try to resolve things on my own. But for this, I will defer >>> to everyone's experience. >>> >>> >>> >>> Thanks, >>> >>> -Chris >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Brian D. Foster >> Endigo Computer LLC >> Email: bdfoster at endigotech.com >> Phone: 317-800-7876 >> Indianapolis, Indiana, USA >> >> This message contains confidential information and is intended for those >> listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If >> you are not the intended recipient you are notified that disclosing, >> copying, distributing or taking any action in reliance on the contents of >> this information is strictly prohibited. E-mail transmission cannot be >> guaranteed to be secure or error-free as information could be intercepted, >> corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. >> The sender therefore does not accept liability for any errors or omissions >> in the contents of this message, which arise as a result of e-mail >> transmission. If verification is required please request a hard-copy >> version. >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120510/0189c20d/attachment.html From chris at opencsta.org Fri May 11 06:23:18 2012 From: chris at opencsta.org (Chris Mylonas) Date: Fri, 11 May 2012 12:23:18 +1000 Subject: [Freeswitch-users] FreeSWITCH, OSX, Libtool, Macports In-Reply-To: <7FFFF20B-D26F-4EDC-A19F-6AE1C48D5048@mgtech.com> References: <523D8618-21D9-442E-9D23-4321EAFA6558@opencsta.org> <7FFFF20B-D26F-4EDC-A19F-6AE1C48D5048@mgtech.com> Message-ID: That's great Mario. I will update my blog accordingly where I posted some stuff (http://mrvoip.com.au/blog/install-freeswitch-osx-mysql-must-remove-macports) I just winged it because it used to work, so most of the prerequisites were installed. I did not put the --with-openssl flag when doing configure. What functionality do I lose? pkgconfig was already on my system from earlier efforts - so I guess I got lucky with that one. To be honest, once I resolved my macports issues (by removing it) it was just like doing it in linux. I'd like to give it another shot with a fresh OSX install - but I really don't want to go about setting up my machine again! I have an old imac that is not doing much, but I don't think it will run 10.6 :( I will try and give it a shot this afternoon seeing as my girlfriend is out taking photos :) Kind Regards, Chris On 11/05/2012, at 11:48 AM, Mario G wrote: > Chris, I have been running FS on 10.6,x since 2010 fine and have been updating a lot recently to help testing, I have had no problems. I don't use macport stuff. I wrote the original FS osX wiki but there are some differences since 2010 (I am working on fixing the wiki in the next few weeks). The wiki install instructions work if you add the following. Hope this helps: > > 1. Add this info: > FreeSWITCH? has many functions that invoke PKG-CONFIG, so it must be downloaded and installed separately. > # Got to [http://pkgconfig.freedesktop.org/releases/ here] and download the latest pkg-config, by default it is placed into your Downloads folder. MUST USE .25 for OSXdue to glib2 dependencies! > # Open Downloads and click the file to uncompress it. > # Launch the Terminal application if not already running and issue these commands to move the source to the src directory, build and install: > cd ~/Downloads > mv pkg-config-0.25 /usr/local/src > cd /usr/local/src/pkg-config-0.25 > ./configure > make > sudo make install > > 2. You MUST do this: > ./bootstrap.sh > ./configure --with-openssl > > 3. The FLITE fix is no longer needed. > > On May 10, 2012, at 5:01 PM, Chris Mylonas wrote: > >> Hi FS Users, >> >> tl;dr; - removed macports, removed tree, pulled fresh tree = installed FS on OSX. >> >> >> Here is the longer version that had a bit of a pre-emptive whinge about libtool version mismatch. But we never got there - it worked! >> >> I'm in the process of re-installing FS on OSX (10.6.8). I have removed Macports to try and get this up and running. >> I re-bootstrapped and got this error: >> >> quiet_libtool: Version mismatch error. This is libtool 2.4, but the >> quiet_libtool: definition of this LT_INIT comes from libtool 2.2.4. >> quiet_libtool: You should recreate aclocal.m4 with macros from libtool 2.4 >> quiet_libtool: and run autoconf again. >> make[1]: *** [libfreeswitch_la-switch_apr.lo] Error 63 >> make: *** [all] Error 2 >> >> Removed the whole git tree just in case there was some left over junk. Pulled a fresh tree and noticed this remark during a fresh bootstrap >> >> arakis:freeswitch-git chrismylonas$ rm -Rf freeswitch/ >> arakis:freeswitch-git chrismylonas$ git clone git://git.freeswitch.org/freeswitch.git >> Cloning into freeswitch... >> remote: Counting objects: 185609, done. >> remote: Compressing objects: 100% (39208/39208), done. >> remote: Total 185609 (delta 143241), reused 181443 (delta 140035) >> Receiving objects: 100% (185609/185609), 77.85 MiB | 267 KiB/s, done. >> Resolving deltas: 100% (143241/143241), done. >> arakis:freeswitch-git chrismylonas$ cd freeswitch/ >> arakis:freeswitch chrismylonas$ ./bootstrap.sh >> bootstrap: checking installation... >> bootstrap: autoconf version 2.61 (ok) >> bootstrap: automake version 1.10 (ok) >> bootstrap: aclocal version 1.10 (ok) >> bootstrap: libtool version 2.2.4 (ok) >> Bootstrapping using: >> autoconf : /usr/bin/autoconf >> automake : /usr/bin/automake >> aclocal : /usr/bin/aclocal >> libtool : /usr/bin/glibtool (2.2.4.) >> libtoolize: /usr/bin/glibtoolize >> make : /usr/bin/make (GNU Make 3.81) >> awk : () >> >> It is still reporting a 2.2.4 version of libtool. >> ... >> ... >> ... >> In the end, it has been compiled and installed though >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120511/2b0e897a/attachment-0001.html From anton.jugatsu at gmail.com Fri May 11 09:07:28 2012 From: anton.jugatsu at gmail.com (Anton Kvashenkin) Date: Fri, 11 May 2012 09:07:28 +0400 Subject: [Freeswitch-users] Fax Issues with Cisco SPA112 and T.38 In-Reply-To: References: Message-ID: Check this out http://wiki.freeswitch.org/wiki/Fs_logger.pl This is very useful tool. Very handy. Also, paste here your dialplan snippet. First, i will try passthru mode to catch re-invite from your ITSP try ngrep -d -qt -W byline -i t38 -o t38_debug.pcap 2012/5/11 Brian Foster > 1. Run fs_cli and do 'sofia global siptrace on' without quotes. Stay in > the cli and run a call in an instance where it would fail. Make sure your > scroll buffer on your SSH terminal (PuTTY or what ever you have) to > unlimited or something bog like 20K lines. Copy the logs out and put them > on http://pastebin.freeswitch.org. If prompted for a username/password, > please read the dialog box. SpanDSP doesn't have a great ability to be > degugged on the click but it will give is something to go on. > > 2. There usually isn't much you can do as far as ensuring the VPS is > capable of keeping the timing right. VoIP is reliant heavily on a good > timing source. Faxing is even more dependent. > > I'm not really sure what the MOS score is useful for, maybe someone else > can chime in on that. > > Something you need to make sure you are doing (specifically with > flowroute) is making sure you are doing a t.38 reinvite. Flowroute does not > auto detect t.38. Also freeswitch might have to transcode from audio > ulaw/alaw to t.38 if your FAX machine doesn't actually support t.38. > > I don't really know much about the new SPA 112/122's but usually the ATA > has a t.38 passthru mode that is auto detected. That might be what you are > referring to. > > -BDF > On May 10, 2012 2:35 PM, "Chris Ferreira" wrote: > >> Hi Bryan, >> >> >> Thanks for your response, much appreciated. >> >> >> 1. I have never had to do a debug other than analyze fs_cli output and >> iv'e never done a sip trace. I have started reading up on how to do these >> on the Wiki and will generate some files to take a look at. >> >> >> 2. I have heard this before about VPS's and even VMWare servers running >> on local physical hardware. I hope to be able to overcome these issues. >> >> >> 3. Other calls work fine throughout the switch and between carriers. >> >> >> >> I don't know if this is a good actual measure of anything, but >> VoIPSpear.com says that the servers MOS Score is a 4.0 . >> >> >> Other than the server running on a VPS, is my setup with the ATA >> registering to FreeSWITCH as an intermediary between Flowroute.com a >> correct practice? Or is there something else I could be doing? >> >> >> I know that if I was not needing to use an Analog Fax machine for >> sending/receiving that I would have other options. But unfortunately the >> Dinosaur must have it's Dial Tone. >> >> >> >> Thanks, >> >> -Chris >> >> >> >> >> On Thu, May 10, 2012 at 2:03 PM, Brian Foster wrote: >> >>> 1. Debug tells all. We need the siptrace as well. >>> >>> 2. VPS's are usually bad at sending/receiving faxes even if the media is >>> just being routed through it. >>> >>> 3. You could also look at session timers. Are other calls working? >>> >>> The only real way to know is if we have number 1 in our possession, >>> otherwise we just start guessing. >>> >>> -BDF >>> >>> On Thu, May 10, 2012 at 1:30 PM, Chris Ferreira wrote: >>> >>>> Hello All, >>>> >>>> >>>> >>>> I have a FreeSWITCH install running on CentOS 5.6 on a Linode VPS. I am >>>> trying to get my SPA112 (Version 1.1 Firmware) to send faxes (and >>>> eventually receive) successfully. I have the ATA registered as an extension >>>> and it's primary outgoing route is through Flowroute. I have T.38 enabled >>>> on the ATA and I have diabled ECM on the analog fax machine. All of the >>>> CDR's in Flowroute show that the test faxes are all ending their calls at >>>> 39 seconds. >>>> >>>> >>>> I can post screen shots of the ATA config or provide any other config >>>> info. >>>> >>>> >>>> Is there something I am missing, or should this setup work? >>>> >>>> I have poked around for a while for info, but unlike the PAP2T there is >>>> little info on these SPA112's. This is my first post asking for help as I >>>> usually try to resolve things on my own. But for this, I will defer >>>> to everyone's experience. >>>> >>>> >>>> >>>> Thanks, >>>> >>>> -Chris >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Brian D. Foster >>> Endigo Computer LLC >>> Email: bdfoster at endigotech.com >>> Phone: 317-800-7876 >>> Indianapolis, Indiana, USA >>> >>> This message contains confidential information and is intended for those >>> listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If >>> you are not the intended recipient you are notified that disclosing, >>> copying, distributing or taking any action in reliance on the contents of >>> this information is strictly prohibited. E-mail transmission cannot be >>> guaranteed to be secure or error-free as information could be intercepted, >>> corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. >>> The sender therefore does not accept liability for any errors or omissions >>> in the contents of this message, which arise as a result of e-mail >>> transmission. If verification is required please request a hard-copy >>> version. >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120511/463b13d5/attachment.html From shane at longwhitecloud.com Fri May 11 04:56:03 2012 From: shane at longwhitecloud.com (Shane Harrison) Date: Fri, 11 May 2012 12:56:03 +1200 Subject: [Freeswitch-users] Freetdm DTMF In-Reply-To: References: Message-ID: Thanks, I'll give that a go. But it doesn't make sense to me because after the bridge action I am receiving DTMF OK from the Freetdm module. As for the bind_meta_app actions for outgoing - yes probably a good idea if I want to transfer after making an outgoing call out a trunk. Cheers Shane On Fri, May 11, 2012 at 11:48 AM, curriegrad2004 wrote: > I would try adding right before > the bridge action for the dialplan for your FreeTDM FXS extensions. > You may want to also apply the bind_meta_app actions on the dialplan > for your FreeTDM outgoing calls for features like attended transfer to > work. > > Hope this helps > > On Thu, May 10, 2012 at 12:55 PM, Shane Harrison > wrote: > > Hi > > > > For calls originating from Freetdm extension I simply set up > > freetdm.conf.xml to use the XML dialplan and context fxs-ports like so: > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > Then dialplan to get it into default context is: > > > > > > > > > > > data="toll_allow=local,domestic,international"/> > > > > > > > > > > > > > > For calls into FreeTDM I have the following dialplan: > > > > > > > > > > > > > > > > > > > > > > > > data="transfer_ringback=$${hold_music}"/> > > > > > > > > > > > > > data="insert/${domain_name}-call_return/300/${caller_id_number}"/> > > > data="insert/${domain_name}-last_dial_ext/300/${uuid}"/> > > > data="called_party_callgroup=${user_data(300@${domain_name} var > > callgroup)}"/> > > > > > > data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/> > > > > > > > > > > > > > > > > > > > > > > Cheers > > Shane > > > > > > On Fri, May 11, 2012 at 2:07 AM, curriegrad2004 < > curriegrad2004 at gmail.com> > > wrote: > >> > >> It would be beneficial if you can post the dialplan on how FS handles > >> the part where FreeTDM comes into play. I.e. the dialplan that does > >> the transfer of the FreeTDM call into the default context. This seems > >> to be opening up a can of worms here with FXS cards in FreeTDM... > >> > >> On Wed, May 9, 2012 at 8:39 PM, Shane Harrison > >> wrote: > >> > Happy for you to laugh at yourself, I'm just happy you are finding > some > >> > time > >> > to take some interest in my problem. Much appreciated. > >> > > >> > I think some clarity is required here. I am simply trying to do an > >> > attended > >> > transfer as per the wiki (except I set the bind_meta_app to both legs) > >> > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_att_xfer > >> > > >> > It works fine for SIP to SIP. For SIP to Freetdm it only works if > the > >> > SIP > >> > is trying to do the transfer. If the FreeTDM is trying to do it ie. > >> > push *3 > >> > on the phone keypad, then the bind_meta_app works fine and detects the > >> > *3 > >> > and executes the appropriate extension ie. att_xfer, however the > read() > >> > in > >> > the extension att_xfer does not see the subsequent DTMF pressed on the > >> > phone > >> > ie. the destination extension number. > >> > > >> > Cheers > >> > Shane > >> > > >> > > >> > On Thu, May 10, 2012 at 2:44 PM, curriegrad2004 > >> > > >> > wrote: > >> >> > >> >> I just had to laugh at my self for mixing up the 2 again... > >> >> bind_meta_app is only applicable to that extension that the inbound > >> >> call was made to. Since you've transferred to another extension, the > >> >> bind_meta_app won't work anymore because it's not defined in the > >> >> extension you're transferring to. > >> >> > >> >> If you want this to happen, you'd have to manually define that > >> >> bind_meta_app to those target extensions too. Remember, do this at > >> >> your own peril - obvious misuse of bind_meta_app can open a huge > >> >> security hole if you don't know what you're doing :) > >> >> > >> >> On Wed, May 9, 2012 at 7:27 PM, Shane Harrison > >> >> wrote: > >> >> > Thanks for the thoughts. As I said, I am already setting it to > both > >> >> > legs - > >> >> > I will try simply trying one leg but am sceptical :-) > >> >> > > >> >> > I also mentioned that I called the start_dtmf just before calling > the > >> >> > read > >> >> > so unless I am doing something wrong here..... I'll try and post > the > >> >> > XML > >> >> > tonight when I get home. Oh and it is an FXS card not an FXO of > >> >> > course > >> >> > since it has a phone plugged into it. > >> >> > > >> >> > The question still remains though, why is the in-band DTMF > detection > >> >> > working > >> >> > for the bind_meta_app digit detection but not after that? > >> >> > > >> >> > Cheers > >> >> > Shane > >> >> > > >> >> > > >> >> > On Thu, May 10, 2012 at 12:28 PM, curriegrad2004 > >> >> > > >> >> > wrote: > >> >> >> > >> >> >> and crap, since I wasn't even reading anything here, on the > >> >> >> subsequent > >> >> >> transfers from your FXO card, enable the in-band DTMF detector > that > >> >> >> FS > >> >> >> has. The details on the in-band DTMF detector is here: > >> >> >> > >> >> >> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf > >> >> >> > >> >> >> But use this with caution, if there is a DTMF detector on the FXO > >> >> >> card > >> >> >> itself, make sure you disable it before using it. > >> >> >> > >> >> >> On Wed, May 9, 2012 at 5:26 PM, curriegrad2004 > >> >> >> > >> >> >> wrote: > >> >> >> > apologies for telling you the wrong thing. I was replying you > from > >> >> >> > my > >> >> >> > phone btw :P > >> >> >> > > >> >> >> > Yeah, bind_meta_app is the app you would use, but try changing > it > >> >> >> > to > >> >> >> > point to the b-leg, not the a-leg > >> >> >> > > >> >> >> > On Wed, May 9, 2012 at 4:12 PM, Shane Harrison > >> >> >> > wrote: > >> >> >> >> Thanks. I am currently using bind_meta_app (set to both legs) > >> >> >> >> already > >> >> >> >> rather than bind_digits. I'll give bind_digits a shot and see > if > >> >> >> >> it > >> >> >> >> behaves > >> >> >> >> differently. > >> >> >> >> > >> >> >> >> Note that I do detect the initial *3 digits and because > >> >> >> >> bind_meta_app > >> >> >> >> is > >> >> >> >> both legs, this is successful no matter which direction the > call > >> >> >> >> is > >> >> >> >> setup > >> >> >> >> from. However once the dialplan moves to the extension the *3 > is > >> >> >> >> bound > >> >> >> >> to, > >> >> >> >> digits are no longer received. > >> >> >> >> > >> >> >> >> The worrying thing for me is that ftdm_io.c doesn't even write > to > >> >> >> >> the > >> >> >> >> log > >> >> >> >> that it has received them (nor freetdm above that of course > which > >> >> >> >> is > >> >> >> >> understandable) and I am surprised that the read() influences > >> >> >> >> that > >> >> >> >> since it > >> >> >> >> works prior on the *3 digits. > >> >> >> >> > >> >> >> >> Cheers > >> >> >> >> Shane > >> >> >> >> > >> >> >> >> > >> >> >> >> > >> >> >> >> On Thu, May 10, 2012 at 10:22 AM, curriegrad2004 > >> >> >> >> > >> >> >> >> wrote: > >> >> >> >>> > >> >> >> >>> I'm guessing the bind digits in your analog card was set to > >> >> >> >>> listen > >> >> >> >>> for > >> >> >> >>> this sequence on the a-leg given if the call was being routed > >> >> >> >>> from > >> >> >> >>> the > >> >> >> >>> IP side to the analog side. > >> >> >> >>> > >> >> >> >>> Try changing that to listen on the b-leg. > >> >> >> >>> > >> >> >> >>> On 5/9/12, Shane Harrison wrote: > >> >> >> >>> > Hi All, > >> >> >> >>> > > >> >> >> >>> > Have a situation where I have a call between a SIP phone > and a > >> >> >> >>> > FreeTDM > >> >> >> >>> > channel. Pushing *3 on the analog FreeTDM phone is > detected > >> >> >> >>> > and > >> >> >> >>> > this > >> >> >> >>> > is > >> >> >> >>> > bound to a dialplan extension (attended transfer) that does > a > >> >> >> >>> > read(): > >> >> >> >>> > > >> >> >> >>> > > >> >> >> >>> > However pushing further digits on the analog phone ie. > >> >> >> >>> > extension > >> >> >> >>> > number > >> >> >> >>> > of > >> >> >> >>> > phone we wish to do an attended transfer to , doesn't result > >> >> >> >>> > in > >> >> >> >>> > the > >> >> >> >>> > DTMF > >> >> >> >>> > being detected. Note that this all works the other way > around > >> >> >> >>> > ie. > >> >> >> >>> > using > >> >> >> >>> > the SIP phone. > >> >> >> >>> > > >> >> >> >>> > When the *3 digits are pushed on the analog phone I see the > >> >> >> >>> > logs > >> >> >> >>> > report: > >> >> >> >>> > > >> >> >> >>> > ftdm_io.c:3463 [s2c1][1:4] Queuing DTMF * (debug = 0) > >> >> >> >>> > mod_freetdm.c:702 Queuing DTMF [*] in channel FreeTDM/2:1/ > >> >> >> >>> > > >> >> >> >>> > ftdm_io.c:3463 [s2c1][1:4] Queuing DTMF 4 (debug = 0) > >> >> >> >>> > > >> >> >> >>> > mod_freetdm.c:702 Queuing DTMF [4] in channel FreeTDM/2:1/ > >> >> >> >>> > > >> >> >> >>> > When the further keys are pushed ftdm_io reports nothing. > >> >> >> >>> > > >> >> >> >>> > I have tried inserting a start_dtmf before the read() but it > >> >> >> >>> > had > >> >> >> >>> > no > >> >> >> >>> > effect. > >> >> >> >>> > Any thoughts as to why DTMF isn't being seen from the analog > >> >> >> >>> > phone > >> >> >> >>> > after > >> >> >> >>> > the read()? > >> >> >> >>> > > >> >> >> >>> > Cheers > >> >> >> >>> > Shane > >> >> >> >>> > > >> >> >> >>> > >> >> >> >>> > >> >> >> >>> > >> >> >> >>> > >> >> >> >>> > _________________________________________________________________________ > >> >> >> >>> Professional FreeSWITCH Consulting Services: > >> >> >> >>> consulting at freeswitch.org > >> >> >> >>> http://www.freeswitchsolutions.com > >> >> >> >>> > >> >> >> >>> > >> >> >> >>> > >> >> >> >>> > >> >> >> >>> Official FreeSWITCH Sites > >> >> >> >>> http://www.freeswitch.org > >> >> >> >>> http://wiki.freeswitch.org > >> >> >> >>> http://www.cluecon.com > >> >> >> >>> > >> >> >> >>> FreeSWITCH-users mailing list > >> >> >> >>> FreeSWITCH-users at lists.freeswitch.org > >> >> >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >> >> >>> > >> >> >> >>> > >> >> >> >>> > >> >> >> >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> >> >> >>> http://www.freeswitch.org > >> >> >> >> > >> >> >> >> > >> >> >> >> > >> >> >> >> > >> >> >> >> -- > >> >> >> >> Paragon Electronic Design Ltd > >> >> >> >> L6 Crest House > >> >> >> >> 92 Queens Drive > >> >> >> >> P0 Box 30449 > >> >> >> >> Lower Hutt 5040 > >> >> >> >> > >> >> >> >> +64 4 5703870 Extn 875 > >> >> >> >> +64 21 608919 (mobile) > >> >> >> >> > >> >> >> >> "Solving your problems with the right technology" > >> >> >> >> > >> >> >> >> > >> >> >> >> > >> >> >> >> > >> >> >> >> > _________________________________________________________________________ > >> >> >> >> Professional FreeSWITCH Consulting Services: > >> >> >> >> consulting at freeswitch.org > >> >> >> >> http://www.freeswitchsolutions.com > >> >> >> >> > >> >> >> >> > >> >> >> >> > >> >> >> >> > >> >> >> >> Official FreeSWITCH Sites > >> >> >> >> http://www.freeswitch.org > >> >> >> >> http://wiki.freeswitch.org > >> >> >> >> http://www.cluecon.com > >> >> >> >> > >> >> >> >> FreeSWITCH-users mailing list > >> >> >> >> FreeSWITCH-users at 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 at freeswitch.org > >> >> >> http://www.freeswitchsolutions.com > >> >> >> > >> >> >> > >> >> >> > >> >> >> > >> >> >> Official FreeSWITCH Sites > >> >> >> http://www.freeswitch.org > >> >> >> http://wiki.freeswitch.org > >> >> >> http://www.cluecon.com > >> >> >> > >> >> >> FreeSWITCH-users mailing list > >> >> >> FreeSWITCH-users at lists.freeswitch.org > >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >> >> > >> >> >> > >> >> >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> >> >> http://www.freeswitch.org > >> >> > > >> >> > > >> >> > > >> >> > > >> >> > -- > >> >> > Paragon Electronic Design Ltd > >> >> > L6 Crest House > >> >> > 92 Queens Drive > >> >> > P0 Box 30449 > >> >> > Lower Hutt 5040 > >> >> > > >> >> > +64 4 5703870 Extn 875 > >> >> > +64 21 608919 (mobile) > >> >> > > >> >> > "Solving your problems with the right technology" > >> >> > > >> >> > > >> >> > > >> >> > > _________________________________________________________________________ > >> >> > Professional FreeSWITCH Consulting Services: > >> >> > consulting at freeswitch.org > >> >> > http://www.freeswitchsolutions.com > >> >> > > >> >> > > >> >> > > >> >> > > >> >> > Official FreeSWITCH Sites > >> >> > http://www.freeswitch.org > >> >> > http://wiki.freeswitch.org > >> >> > http://www.cluecon.com > >> >> > > >> >> > FreeSWITCH-users mailing list > >> >> > FreeSWITCH-users at 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 at freeswitch.org > >> >> http://www.freeswitchsolutions.com > >> >> > >> >> > >> >> > >> >> > >> >> Official FreeSWITCH Sites > >> >> http://www.freeswitch.org > >> >> http://wiki.freeswitch.org > >> >> http://www.cluecon.com > >> >> > >> >> FreeSWITCH-users mailing list > >> >> FreeSWITCH-users at lists.freeswitch.org > >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >> > >> >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> >> http://www.freeswitch.org > >> > > >> > > >> > > >> > > >> > -- > >> > Paragon Electronic Design Ltd > >> > L6 Crest House > >> > 92 Queens Drive > >> > P0 Box 30449 > >> > Lower Hutt 5040 > >> > > >> > +64 4 5703870 Extn 875 > >> > +64 21 608919 (mobile) > >> > > >> > "Solving your problems with the right technology" > >> > > >> > > >> > > _________________________________________________________________________ > >> > Professional FreeSWITCH Consulting Services: > >> > consulting at freeswitch.org > >> > http://www.freeswitchsolutions.com > >> > > >> > > >> > > >> > > >> > Official FreeSWITCH Sites > >> > http://www.freeswitch.org > >> > http://wiki.freeswitch.org > >> > http://www.cluecon.com > >> > > >> > FreeSWITCH-users mailing list > >> > FreeSWITCH-users at 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 at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > > > > > -- > > Paragon Electronic Design Ltd > > L6 Crest House > > 92 Queens Drive > > P0 Box 30449 > > Lower Hutt 5040 > > > > +64 4 5703870 Extn 875 > > +64 21 608919 (mobile) > > > > "Solving your problems with the right technology" > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120511/389cf43f/attachment-0001.html From shaik.bawajan at gmail.com Fri May 11 09:42:54 2012 From: shaik.bawajan at gmail.com (bawajan) Date: Thu, 10 May 2012 22:42:54 -0700 (PDT) Subject: [Freeswitch-users] Choopy one-way noise In-Reply-To: <1336677232.8027.YahooMailNeo@web120406.mail.ne1.yahoo.com> References: <1336658560567-7546814.post@n2.nabble.com> <1336677232.8027.YahooMailNeo@web120406.mail.ne1.yahoo.com> Message-ID: <1336714974876-7549294.post@n2.nabble.com> Hi Valer, Thanks, here am using a SIP UA (snom SI 120 IP Phone) and i have tested making calls from SIP UA to ISDN (freetdm). In this case the voice is very clear and good quality. Here am attaching both files for your reference. Kindly check it and help me to resolve. http://freeswitch-users.2379917.n2.nabble.com/file/n7549294/sip_line1-out-s2c1.wav sip_line1-out-s2c1.wav http://freeswitch-users.2379917.n2.nabble.com/file/n7549294/isdn_line2-in-s2c1.wav isdn_line2-in-s2c1.wav -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Choopy-one-way-noise-tp7332765p7549294.html Sent from the freeswitch-users mailing list archive at Nabble.com. From miha at softnet.si Fri May 11 10:25:51 2012 From: miha at softnet.si (Miha) Date: Fri, 11 May 2012 08:25:51 +0200 Subject: [Freeswitch-users] Phone not registering In-Reply-To: References: <4FABB4DF.5090200@softnet.si> Message-ID: <4FACB0EF.7070801@softnet.si> Hi @Michael, thanks for your reply. I also noticed after I send you an email that this is NAT issue. Phone send registration packet on port 5060 (src port is 50006), but FS do not reply back to port 50006 but instead reply on 5060 due to this phone does not receive 401 and and send another REGISTER packet. How can I deal whit this issue? Thank you very much for all your help! p.s.: I also send you separately wireshark trace that you can see this issue. Regards, Miha On 5/10/2012 8:30 PM, Michael Collins wrote: > It's definitely a NAT issue. The phone is not responding to your 401 > and is instead just sending another REGISTER packet. Notice that FS is > responding on port 5060. Is that the port your phone is expecting to > receive on? > > -MC > > On Thu, May 10, 2012 at 5:30 AM, Miha > wrote: > > Hi, > > here is pastebin of siptrace (http://pastebin.freeswitch.org/19029). > > Phone on local network are registered on FS. After I put between local > network and Phone router, phones are unable to registered on FS. > > On other softswitch which is not FS phones are registering (same port, > same scenario, etc.). > > Phones are SPA922. > > What could be causing the problem? > > Thanks! > > Miha > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120511/1171febf/attachment.html From shaik.bawajan at gmail.com Fri May 11 10:32:12 2012 From: shaik.bawajan at gmail.com (bawajan) Date: Thu, 10 May 2012 23:32:12 -0700 (PDT) Subject: [Freeswitch-users] Choopy one-way noise In-Reply-To: <1336714974876-7549294.post@n2.nabble.com> References: <1336658560567-7546814.post@n2.nabble.com> <1336677232.8027.YahooMailNeo@web120406.mail.ne1.yahoo.com> <1336714974876-7549294.post@n2.nabble.com> Message-ID: <1336717932250-7549383.post@n2.nabble.com> Hi, Am using Sangoma card AFT-A108-SH with PRI lines -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Choopy-one-way-noise-tp7332765p7549383.html Sent from the freeswitch-users mailing list archive at Nabble.com. From miha at softnet.si Fri May 11 11:18:36 2012 From: miha at softnet.si (Miha) Date: Fri, 11 May 2012 09:18:36 +0200 Subject: [Freeswitch-users] Phone not registering In-Reply-To: <4FACB0EF.7070801@softnet.si> References: <4FABB4DF.5090200@softnet.si> <4FACB0EF.7070801@softnet.si> Message-ID: <4FACBD4C.4000307@softnet.si> Hi @Michael, after I put in my internal sip profile FS send back to right dst_port. If I use this on live fs server, could this be causing any problems on registered phones or any easier abuse? Thanks! Miha On 5/11/2012 8:25 AM, Miha wrote: > Hi @Michael, > > thanks for your reply. > > I also noticed after I send you an email that this is NAT issue. Phone > send registration packet on port 5060 (src port is 50006), but FS do > not reply back to port 50006 but instead reply on 5060 due to this > phone does not receive 401 and and send another REGISTER packet. > > How can I deal whit this issue? > > Thank you very much for all your help! > > p.s.: I also send you separately wireshark trace that you can see this > issue. > > Regards, > Miha > > On 5/10/2012 8:30 PM, Michael Collins wrote: >> It's definitely a NAT issue. The phone is not responding to your 401 >> and is instead just sending another REGISTER packet. Notice that FS >> is responding on port 5060. Is that the port your phone is expecting >> to receive on? >> >> -MC >> >> On Thu, May 10, 2012 at 5:30 AM, Miha > > wrote: >> >> Hi, >> >> here is pastebin of siptrace (http://pastebin.freeswitch.org/19029). >> >> Phone on local network are registered on FS. After I put between >> local >> network and Phone router, phones are unable to registered on FS. >> >> On other softswitch which is not FS phones are registering (same >> port, >> same scenario, etc.). >> >> Phones are SPA922. >> >> What could be causing the problem? >> >> Thanks! >> >> Miha >> >> >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120511/c5b0a5d1/attachment.html From B.Tietz at pinguin.ag Fri May 11 12:02:49 2012 From: B.Tietz at pinguin.ag (B.Tietz at pinguin.ag) Date: Fri, 11 May 2012 10:02:49 +0200 Subject: [Freeswitch-users] Fax Issues with Cisco SPA112 and T.38 In-Reply-To: References: Message-ID: <07BF4904977CC645B485E970424193AD10EC17483A@localhost> Infos about MOS: http://en.wikipedia.org/wiki/Mean_opinion_score In short: MOS is seperated in classes which describe the user's view of the quality of the network or heard voice. It is calculated from RTCP-protocol VG, Benjamin T. I'm not really sure what the MOS score is useful for, maybe someone else can chime in on that. -BDF On May 10, 2012 2:35 PM, "Chris Ferreira" > wrote: Hi Bryan, Thanks for your response, much appreciated. 1. I have never had to do a debug other than analyze fs_cli output and iv'e never done a sip trace. I have started reading up on how to do these on the Wiki and will generate some files to take a look at. 2. I have heard this before about VPS's and even VMWare servers running on local physical hardware. I hope to be able to overcome these issues. 3. Other calls work fine throughout the switch and between carriers. I don't know if this is a good actual measure of anything, but VoIPSpear.com says that the servers MOS Score is a 4.0 . Other than the server running on a VPS, is my setup with the ATA registering to FreeSWITCH as an intermediary between Flowroute.com a correct practice? Or is there something else I could be doing? I know that if I was not needing to use an Analog Fax machine for sending/receiving that I would have other options. But unfortunately the Dinosaur must have it's Dial Tone. Thanks, -Chris On Thu, May 10, 2012 at 2:03 PM, Brian Foster > wrote: 1. Debug tells all. We need the siptrace as well. 2. VPS's are usually bad at sending/receiving faxes even if the media is just being routed through it. 3. You could also look at session timers. Are other calls working? The only real way to know is if we have number 1 in our possession, otherwise we just start guessing. -BDF On Thu, May 10, 2012 at 1:30 PM, Chris Ferreira > wrote: Hello All, I have a FreeSWITCH install running on CentOS 5.6 on a Linode VPS. I am trying to get my SPA112 (Version 1.1 Firmware) to send faxes (and eventually receive) successfully. I have the ATA registered as an extension and it's primary outgoing route is through Flowroute. I have T.38 enabled on the ATA and I have diabled ECM on the analog fax machine. All of the CDR's in Flowroute show that the test faxes are all ending their calls at 39 seconds. I can post screen shots of the ATA config or provide any other config info. Is there something I am missing, or should this setup work? I have poked around for a while for info, but unlike the PAP2T there is little info on these SPA112's. This is my first post asking for help as I usually try to resolve things on my own. But for this, I will defer to everyone's experience. Thanks, -Chris _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-800-7876 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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 at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120511/a4b57400/attachment-0001.html From koralu at gmail.com Fri May 11 12:41:23 2012 From: koralu at gmail.com (Adrian Andrei) Date: Fri, 11 May 2012 11:41:23 +0300 Subject: [Freeswitch-users] mod_xml_cdr is not logging In-Reply-To: References: Message-ID: When mod_xml_cdr is reloaded nothing apear in /usr/local/freeswitch/log/cdr/errors. When uncomment the cdr apears in /usr/local/freeswitch/log/cdr an I think that the process of logging cdr it works. But the problem is that the $_POST['cdr'] contain nothing. My php is as simple as possible: When FS calls cdr.php it creats dump.txt but is empty. Ty On 5/10/12, Michael Collins wrote: > Anything in /usr/local/freeswitch/log/cdr/errors ? > > -MC > > On Thu, May 10, 2012 at 11:39 AM, Adrian Andrei wrote: > >> The logs are: >> >> 2012-05-10 21:37:03.399431 [CONSOLE] switch_loadable_module.c:1765 >> Stopping: mod_xml_cdr >> 2012-05-10 21:37:03.399431 [NOTICE] switch_event.c:1935 Event Binding >> deleted for mod_xml_cdr:TRAP >> 2012-05-10 21:37:03.399431 [CONSOLE] switch_loadable_module.c:1785 >> mod_xml_cdr unloaded. >> 2012-05-10 21:37:05.459437 [NOTICE] mod_xml_cdr.c:99 Rotating log file >> paths >> >> +OK module unloaded >> +OK Reloading XML >> +OK module loaded >> >> 2012-05-10 21:37:05.459437 [NOTICE] mod_xml_cdr.c:126 Setting log file >> path to /usr/local/freeswitch/log/cdr >> 2012-05-10 21:37:05.459437 [NOTICE] mod_xml_cdr.c:164 Setting err log >> file >> path to /usr/local/freeswitch/log/cdr/errors >> 2012-05-10 21:37:05.459437 [CONSOLE] switch_loadable_module.c:1299 >> Successfully Loaded [mod_xml_cdr] >> freeswitch at peer1> 2012-05-10 21:37:05.459437 [INFO] mod_enum.c:812 ENUM >> Reloaded >> 2012-05-10 21:37:05.479431 [INFO] switch_time.c:1035 Timezone reloaded >> 530 >> definitions >> >> >> On Thu, May 10, 2012 at 9:17 PM, Michael Collins >> wrote: >> >>> >>> >>> On Thu, May 10, 2012 at 8:45 AM, Adrian Andrei wrote: >>> >>>> Same result. I tried both 127.0.0.1 and localhost. etc/hosts is valid. >>>> >>> what happens when you go to fs_cli and type: >>> reload mod_xml_cdr >>> >>> I'm curious. >>> -MC >>> >> > From hkalyoncu at gmail.com Fri May 11 13:29:39 2012 From: hkalyoncu at gmail.com (huseyin kalyoncu) Date: Fri, 11 May 2012 12:29:39 +0300 Subject: [Freeswitch-users] mod_xml_cdr is not logging In-Reply-To: References: Message-ID: /var/www/xml-cdr/dump.txt file must have write permission for web server user (if its other than root) On Fri, May 11, 2012 at 11:41 AM, Adrian Andrei wrote: > When mod_xml_cdr is reloaded nothing apear in > /usr/local/freeswitch/log/cdr/errors. > > When uncomment the cdr > apears in /usr/local/freeswitch/log/cdr an I think that the process of > logging cdr it works. > > But the problem is that the $_POST['cdr'] contain nothing. > > My php is as simple as possible: > $raw_cdr = $_POST['cdr']; > $writefile = fopen('/var/www/xml-cdr/dump.txt',"a+"); > fwrite($writefile, $raw_cdr); > fclose($writefile); > ?> > > When FS calls cdr.php it creats dump.txt but is empty. > > Ty > > On 5/10/12, Michael Collins wrote: > > Anything in /usr/local/freeswitch/log/cdr/errors ? > > > > -MC > > > > On Thu, May 10, 2012 at 11:39 AM, Adrian Andrei > wrote: > > > >> The logs are: > >> > >> 2012-05-10 21:37:03.399431 [CONSOLE] switch_loadable_module.c:1765 > >> Stopping: mod_xml_cdr > >> 2012-05-10 21:37:03.399431 [NOTICE] switch_event.c:1935 Event Binding > >> deleted for mod_xml_cdr:TRAP > >> 2012-05-10 21:37:03.399431 [CONSOLE] switch_loadable_module.c:1785 > >> mod_xml_cdr unloaded. > >> 2012-05-10 21:37:05.459437 [NOTICE] mod_xml_cdr.c:99 Rotating log file > >> paths > >> > >> +OK module unloaded > >> +OK Reloading XML > >> +OK module loaded > >> > >> 2012-05-10 21:37:05.459437 [NOTICE] mod_xml_cdr.c:126 Setting log file > >> path to /usr/local/freeswitch/log/cdr > >> 2012-05-10 21:37:05.459437 [NOTICE] mod_xml_cdr.c:164 Setting err log > >> file > >> path to /usr/local/freeswitch/log/cdr/errors > >> 2012-05-10 21:37:05.459437 [CONSOLE] switch_loadable_module.c:1299 > >> Successfully Loaded [mod_xml_cdr] > >> freeswitch at peer1> 2012-05-10 21:37:05.459437 [INFO] mod_enum.c:812 ENUM > >> Reloaded > >> 2012-05-10 21:37:05.479431 [INFO] switch_time.c:1035 Timezone reloaded > >> 530 > >> definitions > >> > >> > >> On Thu, May 10, 2012 at 9:17 PM, Michael Collins > >> wrote: > >> > >>> > >>> > >>> On Thu, May 10, 2012 at 8:45 AM, Adrian Andrei > wrote: > >>> > >>>> Same result. I tried both 127.0.0.1 and localhost. etc/hosts is > valid. > >>>> > >>> what happens when you go to fs_cli and type: > >>> reload mod_xml_cdr > >>> > >>> I'm curious. > >>> -MC > >>> > >> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120511/ffd74685/attachment.html From koralu at gmail.com Fri May 11 13:42:38 2012 From: koralu at gmail.com (Adrian Andrei) Date: Fri, 11 May 2012 12:42:38 +0300 Subject: [Freeswitch-users] mod_xml_cdr is not logging In-Reply-To: References: Message-ID: If I add another line in php file like fwrite($writefile, 'test') the output is "testtesttest" which is correct. So I don't think is a problem of write permission. I already put all the rights permission to cdr.php. ty for answer. On 5/11/12, huseyin kalyoncu wrote: > /var/www/xml-cdr/dump.txt file must have write permission for web server > user (if its other than root) > > > On Fri, May 11, 2012 at 11:41 AM, Adrian Andrei wrote: > >> When mod_xml_cdr is reloaded nothing apear in >> /usr/local/freeswitch/log/cdr/errors. >> >> When uncomment the cdr >> apears in /usr/local/freeswitch/log/cdr an I think that the process of >> logging cdr it works. >> >> But the problem is that the $_POST['cdr'] contain nothing. >> >> My php is as simple as possible: >> > $raw_cdr = $_POST['cdr']; >> $writefile = fopen('/var/www/xml-cdr/dump.txt',"a+"); >> fwrite($writefile, $raw_cdr); >> fclose($writefile); >> ?> >> >> When FS calls cdr.php it creats dump.txt but is empty. >> >> Ty >> >> On 5/10/12, Michael Collins wrote: >> > Anything in /usr/local/freeswitch/log/cdr/errors ? >> > >> > -MC >> > >> > On Thu, May 10, 2012 at 11:39 AM, Adrian Andrei >> wrote: >> > >> >> The logs are: >> >> >> >> 2012-05-10 21:37:03.399431 [CONSOLE] switch_loadable_module.c:1765 >> >> Stopping: mod_xml_cdr >> >> 2012-05-10 21:37:03.399431 [NOTICE] switch_event.c:1935 Event Binding >> >> deleted for mod_xml_cdr:TRAP >> >> 2012-05-10 21:37:03.399431 [CONSOLE] switch_loadable_module.c:1785 >> >> mod_xml_cdr unloaded. >> >> 2012-05-10 21:37:05.459437 [NOTICE] mod_xml_cdr.c:99 Rotating log file >> >> paths >> >> >> >> +OK module unloaded >> >> +OK Reloading XML >> >> +OK module loaded >> >> >> >> 2012-05-10 21:37:05.459437 [NOTICE] mod_xml_cdr.c:126 Setting log file >> >> path to /usr/local/freeswitch/log/cdr >> >> 2012-05-10 21:37:05.459437 [NOTICE] mod_xml_cdr.c:164 Setting err log >> >> file >> >> path to /usr/local/freeswitch/log/cdr/errors >> >> 2012-05-10 21:37:05.459437 [CONSOLE] switch_loadable_module.c:1299 >> >> Successfully Loaded [mod_xml_cdr] >> >> freeswitch at peer1> 2012-05-10 21:37:05.459437 [INFO] mod_enum.c:812 >> >> ENUM >> >> Reloaded >> >> 2012-05-10 21:37:05.479431 [INFO] switch_time.c:1035 Timezone reloaded >> >> 530 >> >> definitions >> >> >> >> >> >> On Thu, May 10, 2012 at 9:17 PM, Michael Collins >> >> wrote: >> >> >> >>> >> >>> >> >>> On Thu, May 10, 2012 at 8:45 AM, Adrian Andrei >> wrote: >> >>> >> >>>> Same result. I tried both 127.0.0.1 and localhost. etc/hosts is >> valid. >> >>>> >> >>> what happens when you go to fs_cli and type: >> >>> reload mod_xml_cdr >> >>> >> >>> I'm curious. >> >>> -MC >> >>> >> >> >> > >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > From tarik.bts.gi at gmail.com Fri May 11 13:49:45 2012 From: tarik.bts.gi at gmail.com (tarik.bts.gi at gmail.com) Date: Fri, 11 May 2012 09:49:45 +0000 Subject: [Freeswitch-users] Freeswitch hardware cofig Message-ID: <4face0b9.c8bde00a.05c1.03d0@mx.google.com> I have atom procressor, i want to have 30 simeltanous calls, and i dont have transcoding we use the same codec ------Original message------ From: Michael Collins To: "FreeSWITCH Users Help" Date: Thursday, May 10, 2012 3:18:13 PM GMT-0700 Subject: Re: [Freeswitch-users] Freeswitch hardware cofig On Thu, May 10, 2012 at 12:43 PM, wrote: > What is the good hardware configuration > that needs freeswitch to support more then 30 calls? We have now 1.6 GHz > in the CPU and 2 GB in the mem but is not enough. > What kind of processor? What operating system? Do you want 30 simultaneous calls or 30 new calls per second? Will you be doing transcoding on these calls? All those details will help you decide what to do next. -MC _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From sos at sokhapkin.dyndns.org Fri May 11 14:43:49 2012 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Fri, 11 May 2012 06:43:49 -0400 Subject: [Freeswitch-users] mod_xml_cdr is not logging In-Reply-To: References: Message-ID: <4456980.T5npWALvj0@sos> How do you run php on web server? I got a problem recently with fresh mod_fcgid, in default configuration POST request size is limited to 128M now. On Friday 11 May 2012 11:41:23 Adrian Andrei wrote: > When mod_xml_cdr is reloaded nothing apear in > /usr/local/freeswitch/log/cdr/errors. > > When uncomment the cdr > apears in /usr/local/freeswitch/log/cdr an I think that the process of > logging cdr it works. > > But the problem is that the $_POST['cdr'] contain nothing. > > My php is as simple as possible: > $raw_cdr = $_POST['cdr']; > $writefile = fopen('/var/www/xml-cdr/dump.txt',"a+"); > fwrite($writefile, $raw_cdr); > fclose($writefile); > ?> > > When FS calls cdr.php it creats dump.txt but is empty. > > Ty > > On 5/10/12, Michael Collins wrote: > > Anything in /usr/local/freeswitch/log/cdr/errors ? > > > > -MC > > > > On Thu, May 10, 2012 at 11:39 AM, Adrian Andrei wrote: > >> The logs are: > >> > >> 2012-05-10 21:37:03.399431 [CONSOLE] switch_loadable_module.c:1765 > >> Stopping: mod_xml_cdr > >> 2012-05-10 21:37:03.399431 [NOTICE] switch_event.c:1935 Event Binding > >> deleted for mod_xml_cdr:TRAP > >> 2012-05-10 21:37:03.399431 [CONSOLE] switch_loadable_module.c:1785 > >> mod_xml_cdr unloaded. > >> 2012-05-10 21:37:05.459437 [NOTICE] mod_xml_cdr.c:99 Rotating log file > >> paths > >> > >> +OK module unloaded > >> +OK Reloading XML > >> +OK module loaded > >> > >> 2012-05-10 21:37:05.459437 [NOTICE] mod_xml_cdr.c:126 Setting log file > >> path to /usr/local/freeswitch/log/cdr > >> 2012-05-10 21:37:05.459437 [NOTICE] mod_xml_cdr.c:164 Setting err log > >> file > >> path to /usr/local/freeswitch/log/cdr/errors > >> 2012-05-10 21:37:05.459437 [CONSOLE] switch_loadable_module.c:1299 > >> Successfully Loaded [mod_xml_cdr] > >> freeswitch at peer1> 2012-05-10 21:37:05.459437 [INFO] mod_enum.c:812 ENUM > >> Reloaded > >> 2012-05-10 21:37:05.479431 [INFO] switch_time.c:1035 Timezone reloaded > >> 530 > >> definitions > >> > >> > >> On Thu, May 10, 2012 at 9:17 PM, Michael Collins > >> > >> wrote: > >>> On Thu, May 10, 2012 at 8:45 AM, Adrian Andrei wrote: > >>>> Same result. I tried both 127.0.0.1 and localhost. etc/hosts is valid. > >>> > >>> what happens when you go to fs_cli and type: > >>> reload mod_xml_cdr > >>> > >>> I'm curious. > >>> -MC > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From azza.miled at gmail.com Fri May 11 14:47:42 2012 From: azza.miled at gmail.com (azza miled) Date: Fri, 11 May 2012 11:47:42 +0100 Subject: [Freeswitch-users] FS and Opensips Message-ID: Hi! I know that is a stupid question but can FS do automatic call ditribution for Opensips agent??? Thanks a lot -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120511/86e4417f/attachment-0001.html From koralu at gmail.com Fri May 11 15:34:37 2012 From: koralu at gmail.com (Adrian Andrei) Date: Fri, 11 May 2012 14:34:37 +0300 Subject: [Freeswitch-users] mod_xml_cdr is not logging In-Reply-To: <4456980.T5npWALvj0@sos> References: <4456980.T5npWALvj0@sos> Message-ID: I made a test that calls cdr.php and the $_POST['cdr'] is bigger(5Mb) than the cdr generated localy by FS and it works. I don't think is a problem of webserver. Ty On 5/11/12, Sergey Okhapkin wrote: > How do you run php on web server? I got a problem recently with fresh > mod_fcgid, in default configuration POST request size is limited to 128M > now. > > On Friday 11 May 2012 11:41:23 Adrian Andrei wrote: >> When mod_xml_cdr is reloaded nothing apear in >> /usr/local/freeswitch/log/cdr/errors. >> >> When uncomment the cdr >> apears in /usr/local/freeswitch/log/cdr an I think that the process of >> logging cdr it works. >> >> But the problem is that the $_POST['cdr'] contain nothing. >> >> My php is as simple as possible: >> > $raw_cdr = $_POST['cdr']; >> $writefile = fopen('/var/www/xml-cdr/dump.txt',"a+"); >> fwrite($writefile, $raw_cdr); >> fclose($writefile); >> ?> >> >> When FS calls cdr.php it creats dump.txt but is empty. >> >> Ty >> >> On 5/10/12, Michael Collins wrote: >> > Anything in /usr/local/freeswitch/log/cdr/errors ? >> > >> > -MC >> > >> > On Thu, May 10, 2012 at 11:39 AM, Adrian Andrei >> > wrote: >> >> The logs are: >> >> >> >> 2012-05-10 21:37:03.399431 [CONSOLE] switch_loadable_module.c:1765 >> >> Stopping: mod_xml_cdr >> >> 2012-05-10 21:37:03.399431 [NOTICE] switch_event.c:1935 Event Binding >> >> deleted for mod_xml_cdr:TRAP >> >> 2012-05-10 21:37:03.399431 [CONSOLE] switch_loadable_module.c:1785 >> >> mod_xml_cdr unloaded. >> >> 2012-05-10 21:37:05.459437 [NOTICE] mod_xml_cdr.c:99 Rotating log file >> >> paths >> >> >> >> +OK module unloaded >> >> +OK Reloading XML >> >> +OK module loaded >> >> >> >> 2012-05-10 21:37:05.459437 [NOTICE] mod_xml_cdr.c:126 Setting log file >> >> path to /usr/local/freeswitch/log/cdr >> >> 2012-05-10 21:37:05.459437 [NOTICE] mod_xml_cdr.c:164 Setting err log >> >> file >> >> path to /usr/local/freeswitch/log/cdr/errors >> >> 2012-05-10 21:37:05.459437 [CONSOLE] switch_loadable_module.c:1299 >> >> Successfully Loaded [mod_xml_cdr] >> >> freeswitch at peer1> 2012-05-10 21:37:05.459437 [INFO] mod_enum.c:812 >> >> ENUM >> >> Reloaded >> >> 2012-05-10 21:37:05.479431 [INFO] switch_time.c:1035 Timezone reloaded >> >> 530 >> >> definitions >> >> >> >> >> >> On Thu, May 10, 2012 at 9:17 PM, Michael Collins >> >> >> >> wrote: >> >>> On Thu, May 10, 2012 at 8:45 AM, Adrian Andrei >> >>> wrote: >> >>>> Same result. I tried both 127.0.0.1 and localhost. etc/hosts is >> >>>> valid. >> >>> >> >>> what happens when you go to fs_cli and type: >> >>> reload mod_xml_cdr >> >>> >> >>> I'm curious. >> >>> -MC >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From curriegrad2004 at gmail.com Fri May 11 18:18:13 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Fri, 11 May 2012 07:18:13 -0700 Subject: [Freeswitch-users] Freetdm DTMF In-Reply-To: References: Message-ID: To be quite honest, I wouldn't expect FreeTDM to collect any more digits once the call is in the dialplan. That's why I suggested to use in-band dtmf detection after the call gets bridged from the FreeTDM side to the SIP side of things. The same idea also applies to real TDM links (i.e. no analog stuff here)., meaning you have to do in-band detection on these type of links. I've seen a person on this ML thinking that RBS in a TDM link carries DTMF :P On Thu, May 10, 2012 at 5:56 PM, Shane Harrison wrote: > Thanks, I'll give that a go.? But it doesn't make sense to me because after > the bridge action I am receiving DTMF OK from the Freetdm module. > > As for the bind_meta_app actions for outgoing - yes probably a good idea if > I want to transfer after making an outgoing call out a trunk. > > Cheers > Shane > > > On Fri, May 11, 2012 at 11:48 AM, curriegrad2004 > wrote: >> >> I would try adding right before >> the bridge action for the dialplan for your FreeTDM FXS extensions. >> ?You may want to also apply the bind_meta_app actions on the dialplan >> for your FreeTDM outgoing calls for features like attended transfer to >> work. >> >> Hope this helps >> >> On Thu, May 10, 2012 at 12:55 PM, Shane Harrison >> wrote: >> > Hi >> > >> > For calls originating from Freetdm extension I simply set up >> > freetdm.conf.xml to use the XML dialplan and context fxs-ports like so: >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > Then dialplan to get it into default context is: >> > >> > >> > ?? ? >> > ?? ? ? >> > ?? ? ? ?> > data="toll_allow=local,domestic,international"/> >> > ?? ? ? ? >> > ?? ? ? >> > ?? ? >> > ?? >> > >> > >> > For calls into FreeTDM I have the following dialplan: >> > >> > >> > ?? ? ? ? ? >> > ?? ? ? ? >> > ?? ? ? ? >> > ?? ? ? ? >> > ?? ? ? ? >> > ?? ? ? ? >> > ?? ? ? ? >> > ?? ? ? ? >> > ?? ? ? ? >> > ?? ? ? ?> > data="transfer_ringback=$${hold_music}"/> >> > ?? ? ? ? >> > ?? ? ? ? >> > ?? ? ? ? >> > ?? ? ? ? >> > ?? ? ? ? >> > ?? ? ? ?> > data="insert/${domain_name}-call_return/300/${caller_id_number}"/> >> > ?? ? ? ?> > data="insert/${domain_name}-last_dial_ext/300/${uuid}"/> >> > ?? ? ? ?> > data="called_party_callgroup=${user_data(300@${domain_name} var >> > callgroup)}"/> >> > ?? ? ? ? >> > ?? ? ? ?> > >> > data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/> >> > ?? ? ? ? >> > ?? ? ? ? >> > ?? ? ? ? >> > ?? ? ? ? >> > ?? ? ? ? >> > ?? ? ? >> > ?? ? >> > >> > >> > >> > Cheers >> > Shane >> > >> > >> > On Fri, May 11, 2012 at 2:07 AM, curriegrad2004 >> > >> > wrote: >> >> >> >> It would be beneficial if you can post the dialplan on how FS handles >> >> the part where FreeTDM comes into play. I.e. the dialplan that does >> >> the transfer of the FreeTDM call into the default context. This seems >> >> to be opening up a can of worms here with FXS cards in FreeTDM... >> >> >> >> On Wed, May 9, 2012 at 8:39 PM, Shane Harrison >> >> wrote: >> >> > Happy for you to laugh at yourself, I'm just happy you are finding >> >> > some >> >> > time >> >> > to take some interest in my problem.? Much appreciated. >> >> > >> >> > I think some clarity is required here.? I am simply trying to do an >> >> > attended >> >> > transfer as per the wiki (except I set the bind_meta_app to both >> >> > legs) >> >> > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_att_xfer >> >> > >> >> > It works? fine for SIP to SIP.? For SIP to Freetdm it only works if >> >> > the >> >> > SIP >> >> > is trying to do the transfer.? If the FreeTDM is trying to do it ie. >> >> > push *3 >> >> > on the phone keypad, then the bind_meta_app works fine and detects >> >> > the >> >> > *3 >> >> > and executes the appropriate extension ie. att_xfer, however the >> >> > read() >> >> > in >> >> > the extension att_xfer does not see the subsequent DTMF pressed on >> >> > the >> >> > phone >> >> > ie. the destination extension number. >> >> > >> >> > Cheers >> >> > Shane >> >> > >> >> > >> >> > On Thu, May 10, 2012 at 2:44 PM, curriegrad2004 >> >> > >> >> > wrote: >> >> >> >> >> >> I just had to laugh at my self for mixing up the 2 again... >> >> >> bind_meta_app is only applicable to that extension that the inbound >> >> >> call was made to. Since you've transferred to another extension, the >> >> >> bind_meta_app won't work anymore because it's not defined in the >> >> >> extension you're transferring to. >> >> >> >> >> >> If you want this to happen, you'd have to manually define that >> >> >> bind_meta_app to those target extensions too. Remember, do this at >> >> >> your own peril - obvious misuse of bind_meta_app can open a huge >> >> >> security hole if you don't know what you're doing :) >> >> >> >> >> >> On Wed, May 9, 2012 at 7:27 PM, Shane Harrison >> >> >> wrote: >> >> >> > Thanks for the thoughts.? As I said, I am already setting it to >> >> >> > both >> >> >> > legs - >> >> >> > I will try simply trying one leg but am sceptical :-) >> >> >> > >> >> >> > I also mentioned that I called the start_dtmf just before calling >> >> >> > the >> >> >> > read >> >> >> > so unless I am doing something wrong here.....? I'll try and post >> >> >> > the >> >> >> > XML >> >> >> > tonight when I get home.? Oh and it is an FXS card not an FXO of >> >> >> > course >> >> >> > since it has a phone plugged into it. >> >> >> > >> >> >> > The question still remains though, why is the in-band DTMF >> >> >> > detection >> >> >> > working >> >> >> > for the bind_meta_app digit detection but not after that? >> >> >> > >> >> >> > Cheers >> >> >> > Shane >> >> >> > >> >> >> > >> >> >> > On Thu, May 10, 2012 at 12:28 PM, curriegrad2004 >> >> >> > >> >> >> > wrote: >> >> >> >> >> >> >> >> and crap, since I wasn't even reading anything here, on the >> >> >> >> subsequent >> >> >> >> transfers from your FXO card, enable the in-band DTMF detector >> >> >> >> that >> >> >> >> FS >> >> >> >> has. The details on the in-band DTMF detector is here: >> >> >> >> >> >> >> >> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf >> >> >> >> >> >> >> >> But use this with caution, if there is a DTMF detector on the FXO >> >> >> >> card >> >> >> >> itself, make sure you disable it before using it. >> >> >> >> >> >> >> >> On Wed, May 9, 2012 at 5:26 PM, curriegrad2004 >> >> >> >> >> >> >> >> wrote: >> >> >> >> > apologies for telling you the wrong thing. I was replying you >> >> >> >> > from >> >> >> >> > my >> >> >> >> > phone btw :P >> >> >> >> > >> >> >> >> > Yeah, bind_meta_app is the app you would use, but try changing >> >> >> >> > it >> >> >> >> > to >> >> >> >> > point to the b-leg, not the a-leg >> >> >> >> > >> >> >> >> > On Wed, May 9, 2012 at 4:12 PM, Shane Harrison >> >> >> >> > wrote: >> >> >> >> >> Thanks.? I am currently using bind_meta_app (set to both legs) >> >> >> >> >> already >> >> >> >> >> rather than bind_digits.? I'll give bind_digits a shot and see >> >> >> >> >> if >> >> >> >> >> it >> >> >> >> >> behaves >> >> >> >> >> differently. >> >> >> >> >> >> >> >> >> >> Note that I do detect the initial *3 digits and because >> >> >> >> >> bind_meta_app >> >> >> >> >> is >> >> >> >> >> both legs, this is successful no matter which direction the >> >> >> >> >> call >> >> >> >> >> is >> >> >> >> >> setup >> >> >> >> >> from.? However once the dialplan moves to the extension the *3 >> >> >> >> >> is >> >> >> >> >> bound >> >> >> >> >> to, >> >> >> >> >> digits are no longer received. >> >> >> >> >> >> >> >> >> >> The worrying thing for me is that ftdm_io.c doesn't even write >> >> >> >> >> to >> >> >> >> >> the >> >> >> >> >> log >> >> >> >> >> that it has received them (nor freetdm above that of course >> >> >> >> >> which >> >> >> >> >> is >> >> >> >> >> understandable) and I am surprised that the read() influences >> >> >> >> >> that >> >> >> >> >> since it >> >> >> >> >> works prior on the *3 digits. >> >> >> >> >> >> >> >> >> >> Cheers >> >> >> >> >> Shane >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> On Thu, May 10, 2012 at 10:22 AM, curriegrad2004 >> >> >> >> >> >> >> >> >> >> wrote: >> >> >> >> >>> >> >> >> >> >>> I'm guessing the bind digits in your analog card was set to >> >> >> >> >>> listen >> >> >> >> >>> for >> >> >> >> >>> this sequence on the a-leg given if the call was being routed >> >> >> >> >>> from >> >> >> >> >>> the >> >> >> >> >>> IP side to the analog side. >> >> >> >> >>> >> >> >> >> >>> Try changing that to listen on the b-leg. >> >> >> >> >>> >> >> >> >> >>> On 5/9/12, Shane Harrison wrote: >> >> >> >> >>> > Hi All, >> >> >> >> >>> > >> >> >> >> >>> > Have a situation where I have a call between a SIP phone >> >> >> >> >>> > and a >> >> >> >> >>> > FreeTDM >> >> >> >> >>> > channel. ? Pushing *3 on the analog FreeTDM phone is >> >> >> >> >>> > detected >> >> >> >> >>> > and >> >> >> >> >>> > this >> >> >> >> >>> > is >> >> >> >> >>> > bound to a dialplan extension (attended transfer) that does >> >> >> >> >>> > a >> >> >> >> >>> > read(): >> >> >> >> >>> > >> >> >> >> >>> > >> >> >> >> >>> > However pushing further digits on the analog phone ie. >> >> >> >> >>> > extension >> >> >> >> >>> > number >> >> >> >> >>> > of >> >> >> >> >>> > phone we wish to do an attended transfer to , doesn't >> >> >> >> >>> > result >> >> >> >> >>> > in >> >> >> >> >>> > the >> >> >> >> >>> > DTMF >> >> >> >> >>> > being detected. ?Note that this all works the other way >> >> >> >> >>> > around >> >> >> >> >>> > ie. >> >> >> >> >>> > using >> >> >> >> >>> > the SIP phone. >> >> >> >> >>> > >> >> >> >> >>> > When the *3 digits are pushed on the analog phone I see the >> >> >> >> >>> > logs >> >> >> >> >>> > report: >> >> >> >> >>> > >> >> >> >> >>> > ?ftdm_io.c:3463 [s2c1][1:4] Queuing DTMF * (debug = 0) >> >> >> >> >>> > mod_freetdm.c:702 Queuing DTMF [*] in channel FreeTDM/2:1/ >> >> >> >> >>> > >> >> >> >> >>> > ftdm_io.c:3463 [s2c1][1:4] Queuing DTMF 4 (debug = 0) >> >> >> >> >>> > >> >> >> >> >>> > mod_freetdm.c:702 Queuing DTMF [4] in channel FreeTDM/2:1/ >> >> >> >> >>> > >> >> >> >> >>> > When the further keys are pushed ftdm_io reports nothing. >> >> >> >> >>> > >> >> >> >> >>> > I have tried inserting a start_dtmf before the read() but >> >> >> >> >>> > it >> >> >> >> >>> > had >> >> >> >> >>> > no >> >> >> >> >>> > effect. >> >> >> >> >>> > Any thoughts as to why DTMF isn't being seen from the >> >> >> >> >>> > analog >> >> >> >> >>> > phone >> >> >> >> >>> > after >> >> >> >> >>> > the read()? >> >> >> >> >>> > >> >> >> >> >>> > Cheers >> >> >> >> >>> > Shane >> >> >> >> >>> > >> >> >> >> >>> >> >> >> >> >>> >> >> >> >> >>> >> >> >> >> >>> >> >> >> >> >>> >> >> >> >> >>> _________________________________________________________________________ >> >> >> >> >>> Professional FreeSWITCH Consulting Services: >> >> >> >> >>> consulting at freeswitch.org >> >> >> >> >>> http://www.freeswitchsolutions.com >> >> >> >> >>> >> >> >> >> >>> >> >> >> >> >>> >> >> >> >> >>> >> >> >> >> >>> Official FreeSWITCH Sites >> >> >> >> >>> http://www.freeswitch.org >> >> >> >> >>> http://wiki.freeswitch.org >> >> >> >> >>> http://www.cluecon.com >> >> >> >> >>> >> >> >> >> >>> FreeSWITCH-users mailing list >> >> >> >> >>> FreeSWITCH-users at lists.freeswitch.org >> >> >> >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> >>> >> >> >> >> >>> >> >> >> >> >>> >> >> >> >> >>> >> >> >> >> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> >> >> >>> http://www.freeswitch.org >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> -- >> >> >> >> >> Paragon Electronic Design Ltd >> >> >> >> >> L6 Crest House >> >> >> >> >> 92 Queens Drive >> >> >> >> >> P0 Box 30449 >> >> >> >> >> Lower Hutt 5040 >> >> >> >> >> >> >> >> >> >> +64 4 5703870 Extn 875 >> >> >> >> >> +64 21 608919? (mobile) >> >> >> >> >> >> >> >> >> >> "Solving your problems with the right technology" >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> _________________________________________________________________________ >> >> >> >> >> Professional FreeSWITCH Consulting Services: >> >> >> >> >> consulting at freeswitch.org >> >> >> >> >> http://www.freeswitchsolutions.com >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> Official FreeSWITCH Sites >> >> >> >> >> http://www.freeswitch.org >> >> >> >> >> http://wiki.freeswitch.org >> >> >> >> >> http://www.cluecon.com >> >> >> >> >> >> >> >> >> >> FreeSWITCH-users mailing list >> >> >> >> >> FreeSWITCH-users at 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 at freeswitch.org >> >> >> >> http://www.freeswitchsolutions.com >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> Official FreeSWITCH Sites >> >> >> >> http://www.freeswitch.org >> >> >> >> http://wiki.freeswitch.org >> >> >> >> http://www.cluecon.com >> >> >> >> >> >> >> >> FreeSWITCH-users mailing list >> >> >> >> FreeSWITCH-users at lists.freeswitch.org >> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> >> >> http://www.freeswitch.org >> >> >> > >> >> >> > >> >> >> > >> >> >> > >> >> >> > -- >> >> >> > Paragon Electronic Design Ltd >> >> >> > L6 Crest House >> >> >> > 92 Queens Drive >> >> >> > P0 Box 30449 >> >> >> > Lower Hutt 5040 >> >> >> > >> >> >> > +64 4 5703870 Extn 875 >> >> >> > +64 21 608919? (mobile) >> >> >> > >> >> >> > "Solving your problems with the right technology" >> >> >> > >> >> >> > >> >> >> > >> >> >> > >> >> >> > _________________________________________________________________________ >> >> >> > Professional FreeSWITCH Consulting Services: >> >> >> > consulting at freeswitch.org >> >> >> > http://www.freeswitchsolutions.com >> >> >> > >> >> >> > >> >> >> > >> >> >> > >> >> >> > Official FreeSWITCH Sites >> >> >> > http://www.freeswitch.org >> >> >> > http://wiki.freeswitch.org >> >> >> > http://www.cluecon.com >> >> >> > >> >> >> > FreeSWITCH-users mailing list >> >> >> > FreeSWITCH-users at 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 at freeswitch.org >> >> >> http://www.freeswitchsolutions.com >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> Official FreeSWITCH Sites >> >> >> http://www.freeswitch.org >> >> >> http://wiki.freeswitch.org >> >> >> http://www.cluecon.com >> >> >> >> >> >> FreeSWITCH-users mailing list >> >> >> FreeSWITCH-users at lists.freeswitch.org >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> >> >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> >> http://www.freeswitch.org >> >> > >> >> > >> >> > >> >> > >> >> > -- >> >> > Paragon Electronic Design Ltd >> >> > L6 Crest House >> >> > 92 Queens Drive >> >> > P0 Box 30449 >> >> > Lower Hutt 5040 >> >> > >> >> > +64 4 5703870 Extn 875 >> >> > +64 21 608919? (mobile) >> >> > >> >> > "Solving your problems with the right technology" >> >> > >> >> > >> >> > >> >> > _________________________________________________________________________ >> >> > Professional FreeSWITCH Consulting Services: >> >> > consulting at freeswitch.org >> >> > http://www.freeswitchsolutions.com >> >> > >> >> > >> >> > >> >> > >> >> > Official FreeSWITCH Sites >> >> > http://www.freeswitch.org >> >> > http://wiki.freeswitch.org >> >> > http://www.cluecon.com >> >> > >> >> > FreeSWITCH-users mailing list >> >> > FreeSWITCH-users at 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 at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> >> >> >> >> >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> http://wiki.freeswitch.org >> >> http://www.cluecon.com >> >> >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> > >> > >> > >> > >> > -- >> > Paragon Electronic Design Ltd >> > L6 Crest House >> > 92 Queens Drive >> > P0 Box 30449 >> > Lower Hutt 5040 >> > >> > +64 4 5703870 Extn 875 >> > +64 21 608919? (mobile) >> > >> > "Solving your problems with the right technology" >> > >> > >> > _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at 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 at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From michal.zubac at comgate.cz Fri May 11 18:31:44 2012 From: michal.zubac at comgate.cz (=?UTF-8?B?TWljaGFsIFp1YsOhxI0=?=) Date: Fri, 11 May 2012 16:31:44 +0200 Subject: [Freeswitch-users] disable SIP privacy Message-ID: <4FAD22D0.5020009@comgate.cz> Hi. I want to disable Privacy on outgoing SIP leg but I can't do it because incoming SIP leg has Privacy enabled. I want to set outgoing caller_id to some constant number for these cases. Can this even be done? I tried sip_Privacy variables, but with no effect. Thanks. Michal Zubac From paul at cupis.co.uk Fri May 11 19:07:19 2012 From: paul at cupis.co.uk (Paul Cupis) Date: Fri, 11 May 2012 16:07:19 +0100 Subject: [Freeswitch-users] disable SIP privacy In-Reply-To: <4FAD22D0.5020009@comgate.cz> References: <4FAD22D0.5020009@comgate.cz> Message-ID: <20120511150718.GA10459@eagle.cupis.co.uk> On Fri, May 11, 2012 at 04:31:44PM +0200, Michal Zub???? wrote: > I want to disable Privacy on outgoing SIP leg but I can't do it because > incoming SIP leg has Privacy enabled. I want to set outgoing caller_id > to some constant number for these cases. > Can this even be done? I tried sip_Privacy variables, but with no effect. Try something like this: Regards, From vipkilla at gmail.com Fri May 11 19:17:21 2012 From: vipkilla at gmail.com (Vik Killa) Date: Fri, 11 May 2012 11:17:21 -0400 Subject: [Freeswitch-users] XML_CURL directory configuration does not fail over to static XML Message-ID: I've noticed that when binding the FS directory configuration to XML_CURL, FS does not fail over to static XML when XML_CURL fails. I've verified this by loading identical directory configurations in XML_CURL and in static XML. When I run the following command, it only returns a result if XML_CURL is available: user_data 1000 at testdomain.com var mailbox if the XML_CURL query fails, it does not look in the static XML (which is loaded into memory), instead it returns: -ERR no reply Is this the expected behavior or is it a bug? This behavior is not the case for dialplan configuration. While binding XML_CURL to dialplan, if XML_CURL fails, it will use the static XML that is loaded into memory. From chaiyawut.so at gmail.com Fri May 11 18:14:09 2012 From: chaiyawut.so at gmail.com (Chaiyawut Sookplang) Date: Fri, 11 May 2012 21:14:09 +0700 Subject: [Freeswitch-users] SMS with GSMOpen not support Thai charecters encoded in utf-8 Message-ID: I used GSMOpen to send SMS and I got this problem. Seem like it doesn't support Thai alphabet even I encoded as UTF-8. The empty message was sent by the way. 2012-05-11 20:49:53.235742 [NOTICE] mod_gsmopen.cpp:2856 rev adae5e5|bbae68a[(nil)|37 ][NOTICA 2856 ][interface_1][-1, 0, 0] chat_send(proto=sms, from=interface_1, to=0860216060, subject=SIMPLE MESSAGE, body=???????, type=NULL, hint=interface_1) 2012-05-11 20:49:53.295735 [ERR] gsmopen_protocol.cpp:2405 rev adae5e5|bbae68a[(nil)|37 ][ERRORA 2405 ][interface_1][-1, 0, 0] error: Invalid or incomplete multibyte or wide character 84 From kovszilard at gmail.com Fri May 11 11:28:59 2012 From: kovszilard at gmail.com (=?UTF-8?B?U3ppbMOhcmQgS292w6Fjcw==?=) Date: Fri, 11 May 2012 09:28:59 +0200 Subject: [Freeswitch-users] Where can i find ESL.php? Message-ID: Hello I would like to try the event socket connection with php, but i don't know where can i download ESL.php. Anybody? Thanks, Szilard Kovacs -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120511/1c7f663b/attachment.html From kovszilard at gmail.com Fri May 11 12:04:02 2012 From: kovszilard at gmail.com (=?UTF-8?B?U3ppbMOhcmQgS292w6Fjcw==?=) Date: Fri, 11 May 2012 10:04:02 +0200 Subject: [Freeswitch-users] Where can i find ESL.php? In-Reply-To: References: Message-ID: Hello Sorry for the early question, i found the answer on the wiki. I have an other issue now. When i try make phpmod in /libs/esl, i'm getting the following error message on ubuntu 12.04: root at szilard-virtual-machine:/usr/local/src/freeswitch/libs/esl# make phpmod cc -I/usr/local/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2 -c src/esl.c -o src/esl.o cc -I/usr/local/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2 -c src/esl_event.c -o src/esl_event.o cc -I/usr/local/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2 -c src/esl_threadmutex.c -o src/esl_threadmutex.o cc -I/usr/local/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2 -c src/esl_config.c -o src/esl_config.o cc -I/usr/local/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2 -c src/esl_json.c -o src/esl_json.o cc -I/usr/local/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2 -c src/esl_buffer.c -o src/esl_buffer.o g++ -I/usr/local/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -c src/esl_oop.cpp -o src/esl_oop.o ar rcs libesl.a src/esl.o src/esl_event.o src/esl_threadmutex.o src/esl_config.o src/esl_json.o src/esl_buffer.o src/esl_oop.o ranlib libesl.a make MYLIB="../libesl.a" SOLINK="-shared -Xlinker -x" CFLAGS="-I/usr/local/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2" CXXFLAGS="-I/usr/local/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC" CXX_CFLAGS="" -C php make[1]: Entering directory `/usr/local/src/freeswitch/libs/esl/php' g++ -I/usr/local/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -I/usr/include/php5 -I/usr/include/php5/main -I/usr/include/php5/TSRM -I/usr/include/php5/Zend -I/usr/include/php5/ext -I/usr/include/php5/ext/date/lib -Wno-unused-label -Wno-unused-function -c esl_wrap.cpp -o esl_wrap.o esl_wrap.cpp: In function ?void _wrap_ESLevent_event_set(int, zval*, zval**, zval*, int)?: esl_wrap.cpp:1047:46: warning: format not a string literal and no format arguments [-Wformat-security] esl_wrap.cpp: In function ?void _wrap_ESLevent_event_get(int, zval*, zval**, zval*, int)?: esl_wrap.cpp:1073:46: warning: format not a string literal and no format arguments [-Wformat-security] esl_wrap.cpp: In function ?void _wrap_ESLevent_serialized_string_set(int, zval*, zval**, zval*, int)?: esl_wrap.cpp:1111:46: warning: format not a string literal and no format arguments [-Wformat-security] esl_wrap.cpp: In function ?void _wrap_ESLevent_serialized_string_get(int, zval*, zval**, zval*, int)?: esl_wrap.cpp:1141:46: warning: format not a string literal and no format arguments [-Wformat-security] esl_wrap.cpp: In function ?void _wrap_ESLevent_mine_set(int, zval*, zval**, zval*, int)?: esl_wrap.cpp:1172:46: warning: format not a string literal and no format arguments [-Wformat-security] esl_wrap.cpp: In function ?void _wrap_ESLevent_mine_get(int, zval*, zval**, zval*, int)?: esl_wrap.cpp:1198:46: warning: format not a string literal and no format arguments [-Wformat-security] esl_wrap.cpp: In function ?void _wrap_new_ESLevent__SWIG_0(int, zval*, zval**, zval*, int)?: esl_wrap.cpp:1234:46: warning: format not a string literal and no format arguments [-Wformat-security] esl_wrap.cpp: In function ?void _wrap_new_ESLevent__SWIG_1(int, zval*, zval**, zval*, int)?: esl_wrap.cpp:1269:46: warning: format not a string literal and no format arguments [-Wformat-security] esl_wrap.cpp: In function ?void _wrap_new_ESLevent__SWIG_2(int, zval*, zval**, zval*, int)?: esl_wrap.cpp:1294:46: warning: format not a string literal and no format arguments [-Wformat-security] esl_wrap.cpp: In function ?void _wrap_new_ESLevent(int, zval*, zval**, zval*, int)?: esl_wrap.cpp:1346:46: warning: format not a string literal and no format arguments [-Wformat-security] esl_wrap.cpp: In function ?void _wrap_ESLevent_serialize(int, zval*, zval**, zval*, int)?: esl_wrap.cpp:1403:46: warning: format not a string literal and no format arguments [-Wformat-security] esl_wrap.cpp: In function ?void _wrap_ESLevent_setPriority(int, zval*, zval**, zval*, int)?: esl_wrap.cpp:1441:46: warning: format not a string literal and no format arguments [-Wformat-security] esl_wrap.cpp: In function ?void _wrap_ESLevent_getHeader(int, zval*, zval**, zval*, int)?: esl_wrap.cpp:1488:46: warning: format not a string literal and no format arguments [-Wformat-security] esl_wrap.cpp: In function ?void _wrap_ESLevent_getBody(int, zval*, zval**, zval*, int)?: esl_wrap.cpp:1518:46: warning: format not a string literal and no format arguments [-Wformat-security] esl_wrap.cpp: In function ?void _wrap_ESLevent_getType(int, zval*, zval**, zval*, int)?: esl_wrap.cpp:1548:46: warning: format not a string literal and no format arguments [-Wformat-security] esl_wrap.cpp: In function ?void _wrap_ESLevent_addBody(int, zval*, zval**, zval*, int)?: esl_wrap.cpp:1581:46: warning: format not a string literal and no format arguments [-Wformat-security] esl_wrap.cpp: In function ?void _wrap_ESLevent_addHeader(int, zval*, zval**, zval*, int)?: esl_wrap.cpp:1621:46: warning: format not a string literal and no format arguments [-Wformat-security] esl_wrap.cpp: In function ?void _wrap_ESLevent_pushHeader(int, zval*, zval**, zval*, int)?: esl_wrap.cpp:1661:46: warning: format not a string literal and no format arguments [-Wformat-security] esl_wrap.cpp: In function ?void _wrap_ESLevent_unshiftHeader(int, zval*, zval**, zval*, int)?: esl_wrap.cpp:1701:46: warning: format not a string literal and no format arguments [-Wformat-security] esl_wrap.cpp: In function ?void _wrap_ESLevent_delHeader(int, zval*, zval**, zval*, int)?: esl_wrap.cpp:1734:46: warning: format not a string literal and no format arguments [-Wformat-security] esl_wrap.cpp: In function ?void _wrap_ESLevent_firstHeader(int, zval*, zval**, zval*, int)?: esl_wrap.cpp:1764:46: warning: format not a string literal and no format arguments [-Wformat-security] esl_wrap.cpp: In function ?void _wrap_ESLevent_nextHeader(int, zval*, zval**, zval*, int)?: esl_wrap.cpp:1794:46: warning: format not a string literal and no format arguments [-Wformat-security] esl_wrap.cpp: In function ?void _wrap_new_ESLconnection__SWIG_0(int, zval*, zval**, zval*, int)?: esl_wrap.cpp:1841:46: warning: format not a string literal and no format arguments [-Wformat-security] esl_wrap.cpp: In function ?void _wrap_new_ESLconnection__SWIG_1(int, zval*, zval**, zval*, int)?: esl_wrap.cpp:1881:46: warning: format not a string literal and no format arguments [-Wformat-security] esl_wrap.cpp: In function ?void _wrap_new_ESLconnection__SWIG_2(int, zval*, zval**, zval*, int)?: esl_wrap.cpp:1907:46: warning: format not a string literal and no format arguments [-Wformat-security] esl_wrap.cpp: In function ?void _wrap_new_ESLconnection(int, zval*, zval**, zval*, int)?: esl_wrap.cpp:1956:46: warning: format not a string literal and no format arguments [-Wformat-security] esl_wrap.cpp: In function ?void _wrap_ESLconnection_socketDescriptor(int, zval*, zval**, zval*, int)?: esl_wrap.cpp:1999:46: warning: format not a string literal and no format arguments [-Wformat-security] esl_wrap.cpp: In function ?void _wrap_ESLconnection_connected(int, zval*, zval**, zval*, int)?: esl_wrap.cpp:2025:46: warning: format not a string literal and no format arguments [-Wformat-security] esl_wrap.cpp: In function ?void _wrap_ESLconnection_getInfo(int, zval*, zval**, zval*, int)?: esl_wrap.cpp:2051:46: warning: format not a string literal and no format arguments [-Wformat-security] esl_wrap.cpp: In function ?void _wrap_ESLconnection_send(int, zval*, zval**, zval*, int)?: esl_wrap.cpp:2084:46: warning: format not a string literal and no format arguments [-Wformat-security] esl_wrap.cpp: In function ?void _wrap_ESLconnection_sendRecv(int, zval*, zval**, zval*, int)?: esl_wrap.cpp:2117:46: warning: format not a string literal and no format arguments [-Wformat-security] esl_wrap.cpp: In function ?void _wrap_ESLconnection_api(int, zval*, zval**, zval*, int)?: esl_wrap.cpp:2160:46: warning: format not a string literal and no format arguments [-Wformat-security] esl_wrap.cpp: In function ?void _wrap_ESLconnection_bgapi(int, zval*, zval**, zval*, int)?: esl_wrap.cpp:2211:46: warning: format not a string literal and no format arguments [-Wformat-security] esl_wrap.cpp: In function ?void _wrap_ESLconnection_sendEvent(int, zval*, zval**, zval*, int)?: esl_wrap.cpp:2243:46: warning: format not a string literal and no format arguments [-Wformat-security] esl_wrap.cpp: In function ?void _wrap_ESLconnection_sendMSG(int, zval*, zval**, zval*, int)?: esl_wrap.cpp:2285:46: warning: format not a string literal and no format arguments [-Wformat-security] esl_wrap.cpp: In function ?void _wrap_ESLconnection_recvEvent(int, zval*, zval**, zval*, int)?: esl_wrap.cpp:2311:46: warning: format not a string literal and no format arguments [-Wformat-security] esl_wrap.cpp: In function ?void _wrap_ESLconnection_recvEventTimed(int, zval*, zval**, zval*, int)?: esl_wrap.cpp:2344:46: warning: format not a string literal and no format arguments [-Wformat-security] esl_wrap.cpp: In function ?void _wrap_ESLconnection_filter(int, zval*, zval**, zval*, int)?: esl_wrap.cpp:2384:46: warning: format not a string literal and no format arguments [-Wformat-security] esl_wrap.cpp: In function ?void _wrap_ESLconnection_events(int, zval*, zval**, zval*, int)?: esl_wrap.cpp:2424:46: warning: format not a string literal and no format arguments [-Wformat-security] esl_wrap.cpp: In function ?void _wrap_ESLconnection_execute(int, zval*, zval**, zval*, int)?: esl_wrap.cpp:2475:46: warning: format not a string literal and no format arguments [-Wformat-security] esl_wrap.cpp: In function ?void _wrap_ESLconnection_executeAsync(int, zval*, zval**, zval*, int)?: esl_wrap.cpp:2526:46: warning: format not a string literal and no format arguments [-Wformat-security] esl_wrap.cpp: In function ?void _wrap_ESLconnection_setAsyncExecute(int, zval*, zval**, zval*, int)?: esl_wrap.cpp:2559:46: warning: format not a string literal and no format arguments [-Wformat-security] esl_wrap.cpp: In function ?void _wrap_ESLconnection_setEventLock(int, zval*, zval**, zval*, int)?: esl_wrap.cpp:2592:46: warning: format not a string literal and no format arguments [-Wformat-security] esl_wrap.cpp: In function ?void _wrap_ESLconnection_disconnect(int, zval*, zval**, zval*, int)?: esl_wrap.cpp:2618:46: warning: format not a string literal and no format arguments [-Wformat-security] esl_wrap.cpp: In function ?void _wrap_eslSetLogLevel(int, zval*, zval**, zval*, int)?: esl_wrap.cpp:2641:46: warning: format not a string literal and no format arguments [-Wformat-security] esl_wrap.cpp: At global scope: esl_wrap.cpp:2726:1: warning: deprecated conversion from string constant to ?char*? [-Wwrite-strings] g++ -shared -Xlinker -x esl_wrap.o ../libesl.a -L/usr/lib/x86_64-linux-gnu -lcrypt -lresolv -lcrypt -ldb -lrt -lm -ldl -lnsl -lcrypt -lcrypt -lpthread -o ESL.so -L. /usr/bin/ld: cannot find -ldb collect2: ld returned 1 exit status make[1]: *** [ESL.so] Error 1 make[1]: Leaving directory `/usr/local/src/freeswitch/libs/esl/php' make: *** [phpmod] Error 2 How can I fix this? On Fri, May 11, 2012 at 9:28 AM, Szil?rd Kov?cs wrote: > Hello > > I would like to try the event socket connection with php, but i don't know > where can i download ESL.php. Anybody? > > Thanks, > Szilard Kovacs > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120511/e1742049/attachment-0001.html From kovszilard at gmail.com Fri May 11 12:45:59 2012 From: kovszilard at gmail.com (=?UTF-8?B?U3ppbMOhcmQgS292w6Fjcw==?=) Date: Fri, 11 May 2012 10:45:59 +0200 Subject: [Freeswitch-users] Where can i find ESL.php? In-Reply-To: References: Message-ID: Finally i've got the solution: sudo apt-get install libdb-dev Thanks, and sorry for disturbing :) On Fri, May 11, 2012 at 10:04 AM, Szil?rd Kov?cs wrote: > Hello > > Sorry for the early question, i found the answer on the wiki. I have an > other issue now. When i try make phpmod in dir>/libs/esl, i'm getting the following error message on ubuntu 12.04: > > root at szilard-virtual-machine:/usr/local/src/freeswitch/libs/esl# make > phpmod > cc -I/usr/local/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE -g > -ggdb -I../../libs/libedit/src/ -fPIC -O2 -c src/esl.c -o src/esl.o > cc -I/usr/local/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE -g > -ggdb -I../../libs/libedit/src/ -fPIC -O2 -c src/esl_event.c -o > src/esl_event.o > cc -I/usr/local/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE -g > -ggdb -I../../libs/libedit/src/ -fPIC -O2 -c src/esl_threadmutex.c -o > src/esl_threadmutex.o > cc -I/usr/local/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE -g > -ggdb -I../../libs/libedit/src/ -fPIC -O2 -c src/esl_config.c -o > src/esl_config.o > cc -I/usr/local/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE -g > -ggdb -I../../libs/libedit/src/ -fPIC -O2 -c src/esl_json.c -o > src/esl_json.o > cc -I/usr/local/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE -g > -ggdb -I../../libs/libedit/src/ -fPIC -O2 -c src/esl_buffer.c -o > src/esl_buffer.o > g++ -I/usr/local/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE -g > -ggdb -I../../libs/libedit/src/ -fPIC -c src/esl_oop.cpp -o src/esl_oop.o > ar rcs libesl.a src/esl.o src/esl_event.o src/esl_threadmutex.o > src/esl_config.o src/esl_json.o src/esl_buffer.o src/esl_oop.o > ranlib libesl.a > make MYLIB="../libesl.a" SOLINK="-shared -Xlinker -x" > CFLAGS="-I/usr/local/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE -g > -ggdb -I../../libs/libedit/src/ -fPIC -O2" > CXXFLAGS="-I/usr/local/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE > -g -ggdb -I../../libs/libedit/src/ -fPIC" CXX_CFLAGS="" -C php > make[1]: Entering directory `/usr/local/src/freeswitch/libs/esl/php' > g++ -I/usr/local/src/freeswitch/libs/esl/src/include -DHAVE_EDITLINE -g > -ggdb -I../../libs/libedit/src/ -fPIC -I/usr/include/php5 > -I/usr/include/php5/main -I/usr/include/php5/TSRM -I/usr/include/php5/Zend > -I/usr/include/php5/ext -I/usr/include/php5/ext/date/lib -Wno-unused-label > -Wno-unused-function -c esl_wrap.cpp -o esl_wrap.o > esl_wrap.cpp: In function ?void _wrap_ESLevent_event_set(int, zval*, > zval**, zval*, int)?: > esl_wrap.cpp:1047:46: warning: format not a string literal and no format > arguments [-Wformat-security] > esl_wrap.cpp: In function ?void _wrap_ESLevent_event_get(int, zval*, > zval**, zval*, int)?: > esl_wrap.cpp:1073:46: warning: format not a string literal and no format > arguments [-Wformat-security] > esl_wrap.cpp: In function ?void _wrap_ESLevent_serialized_string_set(int, > zval*, zval**, zval*, int)?: > esl_wrap.cpp:1111:46: warning: format not a string literal and no format > arguments [-Wformat-security] > esl_wrap.cpp: In function ?void _wrap_ESLevent_serialized_string_get(int, > zval*, zval**, zval*, int)?: > esl_wrap.cpp:1141:46: warning: format not a string literal and no format > arguments [-Wformat-security] > esl_wrap.cpp: In function ?void _wrap_ESLevent_mine_set(int, zval*, > zval**, zval*, int)?: > esl_wrap.cpp:1172:46: warning: format not a string literal and no format > arguments [-Wformat-security] > esl_wrap.cpp: In function ?void _wrap_ESLevent_mine_get(int, zval*, > zval**, zval*, int)?: > esl_wrap.cpp:1198:46: warning: format not a string literal and no format > arguments [-Wformat-security] > esl_wrap.cpp: In function ?void _wrap_new_ESLevent__SWIG_0(int, zval*, > zval**, zval*, int)?: > esl_wrap.cpp:1234:46: warning: format not a string literal and no format > arguments [-Wformat-security] > esl_wrap.cpp: In function ?void _wrap_new_ESLevent__SWIG_1(int, zval*, > zval**, zval*, int)?: > esl_wrap.cpp:1269:46: warning: format not a string literal and no format > arguments [-Wformat-security] > esl_wrap.cpp: In function ?void _wrap_new_ESLevent__SWIG_2(int, zval*, > zval**, zval*, int)?: > esl_wrap.cpp:1294:46: warning: format not a string literal and no format > arguments [-Wformat-security] > esl_wrap.cpp: In function ?void _wrap_new_ESLevent(int, zval*, zval**, > zval*, int)?: > esl_wrap.cpp:1346:46: warning: format not a string literal and no format > arguments [-Wformat-security] > esl_wrap.cpp: In function ?void _wrap_ESLevent_serialize(int, zval*, > zval**, zval*, int)?: > esl_wrap.cpp:1403:46: warning: format not a string literal and no format > arguments [-Wformat-security] > esl_wrap.cpp: In function ?void _wrap_ESLevent_setPriority(int, zval*, > zval**, zval*, int)?: > esl_wrap.cpp:1441:46: warning: format not a string literal and no format > arguments [-Wformat-security] > esl_wrap.cpp: In function ?void _wrap_ESLevent_getHeader(int, zval*, > zval**, zval*, int)?: > esl_wrap.cpp:1488:46: warning: format not a string literal and no format > arguments [-Wformat-security] > esl_wrap.cpp: In function ?void _wrap_ESLevent_getBody(int, zval*, zval**, > zval*, int)?: > esl_wrap.cpp:1518:46: warning: format not a string literal and no format > arguments [-Wformat-security] > esl_wrap.cpp: In function ?void _wrap_ESLevent_getType(int, zval*, zval**, > zval*, int)?: > esl_wrap.cpp:1548:46: warning: format not a string literal and no format > arguments [-Wformat-security] > esl_wrap.cpp: In function ?void _wrap_ESLevent_addBody(int, zval*, zval**, > zval*, int)?: > esl_wrap.cpp:1581:46: warning: format not a string literal and no format > arguments [-Wformat-security] > esl_wrap.cpp: In function ?void _wrap_ESLevent_addHeader(int, zval*, > zval**, zval*, int)?: > esl_wrap.cpp:1621:46: warning: format not a string literal and no format > arguments [-Wformat-security] > esl_wrap.cpp: In function ?void _wrap_ESLevent_pushHeader(int, zval*, > zval**, zval*, int)?: > esl_wrap.cpp:1661:46: warning: format not a string literal and no format > arguments [-Wformat-security] > esl_wrap.cpp: In function ?void _wrap_ESLevent_unshiftHeader(int, zval*, > zval**, zval*, int)?: > esl_wrap.cpp:1701:46: warning: format not a string literal and no format > arguments [-Wformat-security] > esl_wrap.cpp: In function ?void _wrap_ESLevent_delHeader(int, zval*, > zval**, zval*, int)?: > esl_wrap.cpp:1734:46: warning: format not a string literal and no format > arguments [-Wformat-security] > esl_wrap.cpp: In function ?void _wrap_ESLevent_firstHeader(int, zval*, > zval**, zval*, int)?: > esl_wrap.cpp:1764:46: warning: format not a string literal and no format > arguments [-Wformat-security] > esl_wrap.cpp: In function ?void _wrap_ESLevent_nextHeader(int, zval*, > zval**, zval*, int)?: > esl_wrap.cpp:1794:46: warning: format not a string literal and no format > arguments [-Wformat-security] > esl_wrap.cpp: In function ?void _wrap_new_ESLconnection__SWIG_0(int, > zval*, zval**, zval*, int)?: > esl_wrap.cpp:1841:46: warning: format not a string literal and no format > arguments [-Wformat-security] > esl_wrap.cpp: In function ?void _wrap_new_ESLconnection__SWIG_1(int, > zval*, zval**, zval*, int)?: > esl_wrap.cpp:1881:46: warning: format not a string literal and no format > arguments [-Wformat-security] > esl_wrap.cpp: In function ?void _wrap_new_ESLconnection__SWIG_2(int, > zval*, zval**, zval*, int)?: > esl_wrap.cpp:1907:46: warning: format not a string literal and no format > arguments [-Wformat-security] > esl_wrap.cpp: In function ?void _wrap_new_ESLconnection(int, zval*, > zval**, zval*, int)?: > esl_wrap.cpp:1956:46: warning: format not a string literal and no format > arguments [-Wformat-security] > esl_wrap.cpp: In function ?void _wrap_ESLconnection_socketDescriptor(int, > zval*, zval**, zval*, int)?: > esl_wrap.cpp:1999:46: warning: format not a string literal and no format > arguments [-Wformat-security] > esl_wrap.cpp: In function ?void _wrap_ESLconnection_connected(int, zval*, > zval**, zval*, int)?: > esl_wrap.cpp:2025:46: warning: format not a string literal and no format > arguments [-Wformat-security] > esl_wrap.cpp: In function ?void _wrap_ESLconnection_getInfo(int, zval*, > zval**, zval*, int)?: > esl_wrap.cpp:2051:46: warning: format not a string literal and no format > arguments [-Wformat-security] > esl_wrap.cpp: In function ?void _wrap_ESLconnection_send(int, zval*, > zval**, zval*, int)?: > esl_wrap.cpp:2084:46: warning: format not a string literal and no format > arguments [-Wformat-security] > esl_wrap.cpp: In function ?void _wrap_ESLconnection_sendRecv(int, zval*, > zval**, zval*, int)?: > esl_wrap.cpp:2117:46: warning: format not a string literal and no format > arguments [-Wformat-security] > esl_wrap.cpp: In function ?void _wrap_ESLconnection_api(int, zval*, > zval**, zval*, int)?: > esl_wrap.cpp:2160:46: warning: format not a string literal and no format > arguments [-Wformat-security] > esl_wrap.cpp: In function ?void _wrap_ESLconnection_bgapi(int, zval*, > zval**, zval*, int)?: > esl_wrap.cpp:2211:46: warning: format not a string literal and no format > arguments [-Wformat-security] > esl_wrap.cpp: In function ?void _wrap_ESLconnection_sendEvent(int, zval*, > zval**, zval*, int)?: > esl_wrap.cpp:2243:46: warning: format not a string literal and no format > arguments [-Wformat-security] > esl_wrap.cpp: In function ?void _wrap_ESLconnection_sendMSG(int, zval*, > zval**, zval*, int)?: > esl_wrap.cpp:2285:46: warning: format not a string literal and no format > arguments [-Wformat-security] > esl_wrap.cpp: In function ?void _wrap_ESLconnection_recvEvent(int, zval*, > zval**, zval*, int)?: > esl_wrap.cpp:2311:46: warning: format not a string literal and no format > arguments [-Wformat-security] > esl_wrap.cpp: In function ?void _wrap_ESLconnection_recvEventTimed(int, > zval*, zval**, zval*, int)?: > esl_wrap.cpp:2344:46: warning: format not a string literal and no format > arguments [-Wformat-security] > esl_wrap.cpp: In function ?void _wrap_ESLconnection_filter(int, zval*, > zval**, zval*, int)?: > esl_wrap.cpp:2384:46: warning: format not a string literal and no format > arguments [-Wformat-security] > esl_wrap.cpp: In function ?void _wrap_ESLconnection_events(int, zval*, > zval**, zval*, int)?: > esl_wrap.cpp:2424:46: warning: format not a string literal and no format > arguments [-Wformat-security] > esl_wrap.cpp: In function ?void _wrap_ESLconnection_execute(int, zval*, > zval**, zval*, int)?: > esl_wrap.cpp:2475:46: warning: format not a string literal and no format > arguments [-Wformat-security] > esl_wrap.cpp: In function ?void _wrap_ESLconnection_executeAsync(int, > zval*, zval**, zval*, int)?: > esl_wrap.cpp:2526:46: warning: format not a string literal and no format > arguments [-Wformat-security] > esl_wrap.cpp: In function ?void _wrap_ESLconnection_setAsyncExecute(int, > zval*, zval**, zval*, int)?: > esl_wrap.cpp:2559:46: warning: format not a string literal and no format > arguments [-Wformat-security] > esl_wrap.cpp: In function ?void _wrap_ESLconnection_setEventLock(int, > zval*, zval**, zval*, int)?: > esl_wrap.cpp:2592:46: warning: format not a string literal and no format > arguments [-Wformat-security] > esl_wrap.cpp: In function ?void _wrap_ESLconnection_disconnect(int, zval*, > zval**, zval*, int)?: > esl_wrap.cpp:2618:46: warning: format not a string literal and no format > arguments [-Wformat-security] > esl_wrap.cpp: In function ?void _wrap_eslSetLogLevel(int, zval*, zval**, > zval*, int)?: > esl_wrap.cpp:2641:46: warning: format not a string literal and no format > arguments [-Wformat-security] > esl_wrap.cpp: At global scope: > esl_wrap.cpp:2726:1: warning: deprecated conversion from string constant > to ?char*? [-Wwrite-strings] > g++ -shared -Xlinker -x esl_wrap.o ../libesl.a -L/usr/lib/x86_64-linux-gnu > -lcrypt -lresolv -lcrypt -ldb -lrt -lm -ldl -lnsl -lcrypt -lcrypt -lpthread > -o ESL.so -L. > /usr/bin/ld: cannot find -ldb > collect2: ld returned 1 exit status > make[1]: *** [ESL.so] Error 1 > make[1]: Leaving directory `/usr/local/src/freeswitch/libs/esl/php' > make: *** [phpmod] Error 2 > > How can I fix this? > > On Fri, May 11, 2012 at 9:28 AM, Szil?rd Kov?cs wrote: > >> Hello >> >> I would like to try the event socket connection with php, but i don't >> know where can i download ESL.php. Anybody? >> >> Thanks, >> Szilard Kovacs >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120511/3f2684d3/attachment-0001.html From msc at freeswitch.org Fri May 11 19:24:01 2012 From: msc at freeswitch.org (Michael Collins) Date: Fri, 11 May 2012 08:24:01 -0700 Subject: [Freeswitch-users] Phone not registering In-Reply-To: <4FACBD4C.4000307@softnet.si> References: <4FABB4DF.5090200@softnet.si> <4FACB0EF.7070801@softnet.si> <4FACBD4C.4000307@softnet.si> Message-ID: You just have to be careful because the force rport can mess up phones that properly handle NAT. If you plan to use only this model of phone - or only phones that require force rport - then you'll be fine. If you find that you have a mix of phone models then it's best to create a separate profile to handle phones that are NAT friendly and then keep the current profile for phones that need the force rport. -MC On Fri, May 11, 2012 at 12:18 AM, Miha wrote: > Hi @Michael, > > after I put > > > > in my internal sip profile FS send back to right dst_port. > > If I use this on live fs server, could this be causing any problems on registered phones or any easier abuse? > > Thanks! > > Miha > > > On 5/11/2012 8:25 AM, Miha wrote: > > Hi @Michael, > > thanks for your reply. > > I also noticed after I send you an email that this is NAT issue. Phone > send registration packet on port 5060 (src port is 50006), but FS do not > reply back to port 50006 but instead reply on 5060 due to this phone does > not receive 401 and and send another REGISTER packet. > > How can I deal whit this issue? > > Thank you very much for all your help! > > p.s.: I also send you separately wireshark trace that you can see this > issue. > > Regards, > Miha > > On 5/10/2012 8:30 PM, Michael Collins wrote: > > It's definitely a NAT issue. The phone is not responding to your 401 and > is instead just sending another REGISTER packet. Notice that FS is > responding on port 5060. Is that the port your phone is expecting to > receive on? > > -MC > > On Thu, May 10, 2012 at 5:30 AM, Miha wrote: > >> Hi, >> >> here is pastebin of siptrace (http://pastebin.freeswitch.org/19029). >> >> Phone on local network are registered on FS. After I put between local >> network and Phone router, phones are unable to registered on FS. >> >> On other softswitch which is not FS phones are registering (same port, >> same scenario, etc.). >> >> Phones are SPA922. >> >> What could be causing the problem? >> >> Thanks! >> >> Miha >> >> >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120511/cc6cf04e/attachment.html From msc at freeswitch.org Fri May 11 19:28:05 2012 From: msc at freeswitch.org (Michael Collins) Date: Fri, 11 May 2012 08:28:05 -0700 Subject: [Freeswitch-users] mod_xml_cdr is not logging In-Reply-To: References: <4456980.T5npWALvj0@sos> Message-ID: At this point I would run a packet sniffer on port 80 on your FS machine and confirm whether or not it's even attempting to contact your web server. Also, check to make sure you don't have IP tables or some other external process/device interfering with your FS trying to contact the web server. Last resort is to get a different box, re-install FS and try posting to a web server running on the same box. -MC On Fri, May 11, 2012 at 4:34 AM, Adrian Andrei wrote: > I made a test that calls cdr.php and the $_POST['cdr'] is bigger(5Mb) > than the cdr generated localy by FS and it works. I don't think is a > problem of webserver. > > Ty > On 5/11/12, Sergey Okhapkin wrote: > > How do you run php on web server? I got a problem recently with fresh > > mod_fcgid, in default configuration POST request size is limited to 128M > > now. > > > > On Friday 11 May 2012 11:41:23 Adrian Andrei wrote: > >> When mod_xml_cdr is reloaded nothing apear in > >> /usr/local/freeswitch/log/cdr/errors. > >> > >> When uncomment the cdr > >> apears in /usr/local/freeswitch/log/cdr an I think that the process of > >> logging cdr it works. > >> > >> But the problem is that the $_POST['cdr'] contain nothing. > >> > >> My php is as simple as possible: > >> >> $raw_cdr = $_POST['cdr']; > >> $writefile = fopen('/var/www/xml-cdr/dump.txt',"a+"); > >> fwrite($writefile, $raw_cdr); > >> fclose($writefile); > >> ?> > >> > >> When FS calls cdr.php it creats dump.txt but is empty. > >> > >> Ty > >> > >> On 5/10/12, Michael Collins wrote: > >> > Anything in /usr/local/freeswitch/log/cdr/errors ? > >> > > >> > -MC > >> > > >> > On Thu, May 10, 2012 at 11:39 AM, Adrian Andrei > >> > wrote: > >> >> The logs are: > >> >> > >> >> 2012-05-10 21:37:03.399431 [CONSOLE] switch_loadable_module.c:1765 > >> >> Stopping: mod_xml_cdr > >> >> 2012-05-10 21:37:03.399431 [NOTICE] switch_event.c:1935 Event Binding > >> >> deleted for mod_xml_cdr:TRAP > >> >> 2012-05-10 21:37:03.399431 [CONSOLE] switch_loadable_module.c:1785 > >> >> mod_xml_cdr unloaded. > >> >> 2012-05-10 21:37:05.459437 [NOTICE] mod_xml_cdr.c:99 Rotating log > file > >> >> paths > >> >> > >> >> +OK module unloaded > >> >> +OK Reloading XML > >> >> +OK module loaded > >> >> > >> >> 2012-05-10 21:37:05.459437 [NOTICE] mod_xml_cdr.c:126 Setting log > file > >> >> path to /usr/local/freeswitch/log/cdr > >> >> 2012-05-10 21:37:05.459437 [NOTICE] mod_xml_cdr.c:164 Setting err log > >> >> file > >> >> path to /usr/local/freeswitch/log/cdr/errors > >> >> 2012-05-10 21:37:05.459437 [CONSOLE] switch_loadable_module.c:1299 > >> >> Successfully Loaded [mod_xml_cdr] > >> >> freeswitch at peer1> 2012-05-10 21:37:05.459437 [INFO] mod_enum.c:812 > >> >> ENUM > >> >> Reloaded > >> >> 2012-05-10 21:37:05.479431 [INFO] switch_time.c:1035 Timezone > reloaded > >> >> 530 > >> >> definitions > >> >> > >> >> > >> >> On Thu, May 10, 2012 at 9:17 PM, Michael Collins > >> >> > >> >> wrote: > >> >>> On Thu, May 10, 2012 at 8:45 AM, Adrian Andrei > >> >>> wrote: > >> >>>> Same result. I tried both 127.0.0.1 and localhost. etc/hosts is > >> >>>> valid. > >> >>> > >> >>> what happens when you go to fs_cli and type: > >> >>> reload mod_xml_cdr > >> >>> > >> >>> I'm curious. > >> >>> -MC > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at 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 at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120511/04efbb99/attachment-0001.html From vipkilla at gmail.com Fri May 11 19:44:20 2012 From: vipkilla at gmail.com (Vik Killa) Date: Fri, 11 May 2012 11:44:20 -0400 Subject: [Freeswitch-users] XML_CURL directory configuration does not fail over to static XML In-Reply-To: References: Message-ID: Upon further investigation, it seems all directory parameters and attributes from static XML are stored in memory and can be accessed if XML_CURL fails. This is not true for any directory variables though. Is there a specific reason why FS does not store directory variables in memory when using XML_CURL? From jerry.richards at teotech.com Fri May 11 20:15:26 2012 From: jerry.richards at teotech.com (Jerry Richards) Date: Fri, 11 May 2012 16:15:26 +0000 Subject: [Freeswitch-users] Gaps In RTP (wideband) Packets From mod_voicemail In-Reply-To: References: <1545146083A72C4DB7B66584B7E5D98402BC1FF4@BY2PRD0410MB377.namprd04.prod.outlook.com> <1545146083A72C4DB7B66584B7E5D98402BC2015@BY2PRD0410MB377.namprd04.prod.outlook.com> <1545146083A72C4DB7B66584B7E5D98402BC206B@BY2PRD0410MB377.namprd04.prod.outlook.com> Message-ID: <1545146083A72C4DB7B66584B7E5D98402BCDA3E@BY2PRD0410MB377.namprd04.prod.outlook.com> I updated to latest and still see some big gaps in the RTP stream (8 of them well over 100 mS), rather than the ptime rate of 20mS. I think the issue is that the audio engine should run independently and continue to transmit RTP packets at a consistent rate, regardless whether it's silence/comfort-noise or a playing a voicemail phrase. I think the reason it is noticeable in wideband (and not G.711) is because this the wideband codec needs to train (i.e. collect multiple RTP samples) for a time before it plays the audio. Thanks, Jerry -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Thursday, May 10, 2012 9:56 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Gaps In RTP (wideband) Packets From mod_voicemail I found a restriction in the core limiting that functionality to sleeps > 100ms, the vm sleeps 100ms between files by default. update to latest and try again. On Thu, May 10, 2012 at 11:26 AM, Jerry Richards wrote: > I'm setting send_silence_when_idle globally to a numerical value that fixes a loud white noise issue during sleeps. ?I originally had it set to true but still had the wideband clipping issue. > > Jerry > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Anthony Minessale > Sent: Thursday, May 10, 2012 8:40 AM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Gaps In RTP (wideband) Packets From > mod_voicemail > > set send_silence_when_idle=true globally or per-leg > > On Thu, May 10, 2012 at 10:35 AM, Jerry Richards wrote: >> By the way, in contrast, I created an XML block that plays the same >> messages as mod_voicemail (see block below).? When this XML block is >> executed, there is no clipping of words when using wideband codec >> BV32.? This is why I'm thinking it's related to delays caused by mod_voicemail processing. >> >> >> >> ??? >> >> ????? >> >> ??????? >> >> ?????? >> >> ????????????? > data="/opt/teoswitch/sounds/en/us/callie/voicemail/vm-hello.wav"/> >> >> ??????? > data="/opt/teoswitch/sounds/en/us/callie/voicemail/vm-you_have.wav"/> >> >> ????????????? >> >> ??????? > data="/opt/teoswitch/sounds/en/us/callie/voicemail/vm-new.wav"/> >> >> ??????? > data="/opt/teoswitch/sounds/en/us/callie/voicemail/vm-messages.wav"/> >> >> ??????? > data="/opt/teoswitch/sounds/en/us/callie/voicemail/vm-you_have.wav"/> >> >> ????????????? >> >> ??????? > data="/opt/teoswitch/sounds/en/us/callie/voicemail/vm-saved.wav"/> >> >> ??????? > data="/opt/teoswitch/sounds/en/us/callie/voicemail/vm-messages.wav"/> >> >> ??????? > data="/opt/teoswitch/sounds/en/us/callie/voicemail/vm-listen_new.wav" >> / >> > >> >> ??????? > data="/opt/teoswitch/sounds/en/us/callie/voicemail/vm-press.wav"/> >> >> ????????????? >> >> ??????? >> >> ????? >> >> ??? >> >> >> >> Thanks, >> >> Jerry >> >> >> >> From: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of >> Jerry Richards >> Sent: Thursday, May 10, 2012 8:23 AM >> To: FreeSWITCH Users Help >> Subject: [Freeswitch-users] Gaps In RTP (wideband) Packets From >> mod_voicemail >> >> >> >> Hello, >> >> >> >> I'm seeing a audio problem when listening to the operator when >> calling into voicemail to listen to messages and a wideband codec is used. >> Looking at a wireshark trace, I see gaps between transmitted RTP >> packets that exceed the jitter buffer depth in the endpoint >> softphone, causing underflow.? ?This causes clipping of the front-end >> of words such as "Pressed" sounds like 'ess' and "Last name" sounds >> like 'ast name'.? While this is generally not noticeable with the >> G.711 codec, it causes these clipped audio phrases with wideband >> codecs, such as BV32.? It might be due to the processing delay in mod_voicemail between phrases? >> >> >> >> Has anyone seen this?? Is there a tag to help improve this? >> >> >> >> Thanks, >> >> Jerry >> >> >> >> >> _____________________________________________________________________ >> _ ___ Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us >> e >> rs >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org > > > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From anthony.minessale at gmail.com Fri May 11 20:38:19 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 11 May 2012 11:38:19 -0500 Subject: [Freeswitch-users] Gaps In RTP (wideband) Packets From mod_voicemail In-Reply-To: <1545146083A72C4DB7B66584B7E5D98402BCDA3E@BY2PRD0410MB377.namprd04.prod.outlook.com> References: <1545146083A72C4DB7B66584B7E5D98402BC1FF4@BY2PRD0410MB377.namprd04.prod.outlook.com> <1545146083A72C4DB7B66584B7E5D98402BC2015@BY2PRD0410MB377.namprd04.prod.outlook.com> <1545146083A72C4DB7B66584B7E5D98402BC206B@BY2PRD0410MB377.namprd04.prod.outlook.com> <1545146083A72C4DB7B66584B7E5D98402BCDA3E@BY2PRD0410MB377.namprd04.prod.outlook.com> Message-ID: That is not possible and not efficient. The phrase macro system pauses 100ms between files it strings together. That last change when used together with send_silence_when_idle should transmit rtp during the sleep. You can test it possibly by doing a dp with sleep 4000 and see if you have 2 way rtp. Transmission of nonstop RTP is not a requirement in voip. another thing you can try is setting the rtp bug to turn off marker bit which resets remote jitter buffer.. set the variable rtp_manual_rtp_bugs=NEVER_SEND_MARKER On Fri, May 11, 2012 at 11:15 AM, Jerry Richards wrote: > I updated to latest and still see some big gaps in the RTP stream (8 of them well over 100 mS), rather than the ptime rate of 20mS. ?I think the issue is that the audio engine should run independently and continue to transmit RTP packets at a consistent rate, regardless whether it's silence/comfort-noise or a playing a voicemail phrase. ?I think the reason it is noticeable in wideband (and not G.711) is because this the wideband codec needs to train (i.e. collect multiple RTP samples) for a time before it plays the audio. > > Thanks, > Jerry > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale > Sent: Thursday, May 10, 2012 9:56 AM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Gaps In RTP (wideband) Packets From mod_voicemail > > I found a restriction in the core limiting that functionality to sleeps > 100ms, the vm sleeps 100ms between files by default. > update to latest and try again. > > > On Thu, May 10, 2012 at 11:26 AM, Jerry Richards wrote: >> I'm setting send_silence_when_idle globally to a numerical value that fixes a loud white noise issue during sleeps. ?I originally had it set to true but still had the wideband clipping issue. >> >> Jerry >> >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of >> Anthony Minessale >> Sent: Thursday, May 10, 2012 8:40 AM >> To: FreeSWITCH Users Help >> Subject: Re: [Freeswitch-users] Gaps In RTP (wideband) Packets From >> mod_voicemail >> >> set send_silence_when_idle=true globally or per-leg >> >> On Thu, May 10, 2012 at 10:35 AM, Jerry Richards wrote: >>> By the way, in contrast, I created an XML block that plays the same >>> messages as mod_voicemail (see block below).? When this XML block is >>> executed, there is no clipping of words when using wideband codec >>> BV32.? This is why I'm thinking it's related to delays caused by mod_voicemail processing. >>> >>> >>> >>> ??? >>> >>> ????? >>> >>> ??????? >>> >>> ?????? >>> >>> ????????????? >> data="/opt/teoswitch/sounds/en/us/callie/voicemail/vm-hello.wav"/> >>> >>> ??????? >> data="/opt/teoswitch/sounds/en/us/callie/voicemail/vm-you_have.wav"/> >>> >>> ????????????? >>> >>> ??????? >> data="/opt/teoswitch/sounds/en/us/callie/voicemail/vm-new.wav"/> >>> >>> ??????? >> data="/opt/teoswitch/sounds/en/us/callie/voicemail/vm-messages.wav"/> >>> >>> ??????? >> data="/opt/teoswitch/sounds/en/us/callie/voicemail/vm-you_have.wav"/> >>> >>> ????????????? >>> >>> ??????? >> data="/opt/teoswitch/sounds/en/us/callie/voicemail/vm-saved.wav"/> >>> >>> ??????? >> data="/opt/teoswitch/sounds/en/us/callie/voicemail/vm-messages.wav"/> >>> >>> ??????? >> data="/opt/teoswitch/sounds/en/us/callie/voicemail/vm-listen_new.wav" >>> / >>> > >>> >>> ??????? >> data="/opt/teoswitch/sounds/en/us/callie/voicemail/vm-press.wav"/> >>> >>> ????????????? >>> >>> ??????? >>> >>> ????? >>> >>> ??? >>> >>> >>> >>> Thanks, >>> >>> Jerry >>> >>> >>> >>> From: freeswitch-users-bounces at lists.freeswitch.org >>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of >>> Jerry Richards >>> Sent: Thursday, May 10, 2012 8:23 AM >>> To: FreeSWITCH Users Help >>> Subject: [Freeswitch-users] Gaps In RTP (wideband) Packets From >>> mod_voicemail >>> >>> >>> >>> Hello, >>> >>> >>> >>> I'm seeing a audio problem when listening to the operator when >>> calling into voicemail to listen to messages and a wideband codec is used. >>> Looking at a wireshark trace, I see gaps between transmitted RTP >>> packets that exceed the jitter buffer depth in the endpoint >>> softphone, causing underflow.? ?This causes clipping of the front-end >>> of words such as "Pressed" sounds like 'ess' and "Last name" sounds >>> like 'ast name'.? While this is generally not noticeable with the >>> G.711 codec, it causes these clipped audio phrases with wideband >>> codecs, such as BV32.? It might be due to the processing delay in mod_voicemail between phrases? >>> >>> >>> >>> Has anyone seen this?? Is there a tag to help improve this? >>> >>> >>> >>> Thanks, >>> >>> Jerry >>> >>> >>> >>> >>> _____________________________________________________________________ >>> _ ___ Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us >>> e >>> rs >>> http://www.freeswitch.org >>> >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 >> >> ______________________________________________________________________ >> ___ Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use >> rs >> http://www.freeswitch.org >> >> >> >> ______________________________________________________________________ >> ___ Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use >> rs >> http://www.freeswitch.org > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From mario_fs at mgtech.com Fri May 11 20:41:07 2012 From: mario_fs at mgtech.com (Mario G) Date: Fri, 11 May 2012 09:41:07 -0700 Subject: [Freeswitch-users] FreeSWITCH, OSX, Libtool, Macports In-Reply-To: References: <523D8618-21D9-442E-9D23-4321EAFA6558@opencsta.org> <7FFFF20B-D26F-4EDC-A19F-6AE1C48D5048@mgtech.com> Message-ID: <761EBAA8-6CE4-4A9B-BD8F-4A5EB064EEFA@mgtech.com> Some answers for you: On May 10, 2012, at 7:23 PM, Chris Mylonas wrote: > That's great Mario. I will update my blog accordingly where I posted some stuff (http://mrvoip.com.au/blog/install-freeswitch-osx-mysql-must-remove-macports) > > I just winged it because it used to work, so most of the prerequisites were installed. > I did not put the --with-openssl flag when doing configure. What functionality do I lose? From what I remember, it complained about gnutis or something like that, you have to check the messages but it is really important, used to not be required. It took several days to figure this one out. I think the message was: configure: error: --with-ssl was given, but GNUTLS is not available. > > > pkgconfig was already on my system from earlier efforts - so I guess I got lucky with that one. > To be honest, once I resolved my macports issues (by removing it) it was just like doing it in linux. The funny thing is I still see a few pkgconfig missing messages during the make but FS works fine for me, I think it's for things I am not using. I will be looking into this more before I update the wiki. > > I'd like to give it another shot with a fresh OSX install - but I really don't want to go about setting up my machine again! I have an old imac that is not doing much, but I don't think it will run 10.6 :( I will try and give it a shot this afternoon seeing as my girlfriend is out taking photos :) FYI, when I have not updated for a long time or make major changes like updating xCode: I backup the Mac Mini running FS as a bootable partition (I use SuperDuper), then I connect it to an iMac and boot using firewire target mode to boot from the backup. Now I can play round all I want. If all goes well, I just restore to the Mini, if not I can start over. For frequent updates I just backup the FS bin and source in the Mini with and do a git pull, etc.... Mario G > > Kind Regards, > Chris > > > > > On 11/05/2012, at 11:48 AM, Mario G wrote: > >> Chris, I have been running FS on 10.6,x since 2010 fine and have been updating a lot recently to help testing, I have had no problems. I don't use macport stuff. I wrote the original FS osX wiki but there are some differences since 2010 (I am working on fixing the wiki in the next few weeks). The wiki install instructions work if you add the following. Hope this helps: >> >> 1. Add this info: >> FreeSWITCH? has many functions that invoke PKG-CONFIG, so it must be downloaded and installed separately. >> # Got to [http://pkgconfig.freedesktop.org/releases/ here] and download the latest pkg-config, by default it is placed into your Downloads folder. MUST USE .25 for OSXdue to glib2 dependencies! >> # Open Downloads and click the file to uncompress it. >> # Launch the Terminal application if not already running and issue these commands to move the source to the src directory, build and install: >> cd ~/Downloads >> mv pkg-config-0.25 /usr/local/src >> cd /usr/local/src/pkg-config-0.25 >> ./configure >> make >> sudo make install > >> >> 2. You MUST do this: >> ./bootstrap.sh >> ./configure --with-openssl >> >> 3. The FLITE fix is no longer needed. >> >> On May 10, 2012, at 5:01 PM, Chris Mylonas wrote: >> >>> Hi FS Users, >>> >>> tl;dr; - removed macports, removed tree, pulled fresh tree = installed FS on OSX. >>> >>> >>> Here is the longer version that had a bit of a pre-emptive whinge about libtool version mismatch. But we never got there - it worked! >>> >>> I'm in the process of re-installing FS on OSX (10.6.8). I have removed Macports to try and get this up and running. >>> I re-bootstrapped and got this error: >>> >>> quiet_libtool: Version mismatch error. This is libtool 2.4, but the >>> quiet_libtool: definition of this LT_INIT comes from libtool 2.2.4. >>> quiet_libtool: You should recreate aclocal.m4 with macros from libtool 2.4 >>> quiet_libtool: and run autoconf again. >>> make[1]: *** [libfreeswitch_la-switch_apr.lo] Error 63 >>> make: *** [all] Error 2 >>> >>> Removed the whole git tree just in case there was some left over junk. Pulled a fresh tree and noticed this remark during a fresh bootstrap >>> >>> arakis:freeswitch-git chrismylonas$ rm -Rf freeswitch/ >>> arakis:freeswitch-git chrismylonas$ git clone git://git.freeswitch.org/freeswitch.git >>> Cloning into freeswitch... >>> remote: Counting objects: 185609, done. >>> remote: Compressing objects: 100% (39208/39208), done. >>> remote: Total 185609 (delta 143241), reused 181443 (delta 140035) >>> Receiving objects: 100% (185609/185609), 77.85 MiB | 267 KiB/s, done. >>> Resolving deltas: 100% (143241/143241), done. >>> arakis:freeswitch-git chrismylonas$ cd freeswitch/ >>> arakis:freeswitch chrismylonas$ ./bootstrap.sh >>> bootstrap: checking installation... >>> bootstrap: autoconf version 2.61 (ok) >>> bootstrap: automake version 1.10 (ok) >>> bootstrap: aclocal version 1.10 (ok) >>> bootstrap: libtool version 2.2.4 (ok) >>> Bootstrapping using: >>> autoconf : /usr/bin/autoconf >>> automake : /usr/bin/automake >>> aclocal : /usr/bin/aclocal >>> libtool : /usr/bin/glibtool (2.2.4.) >>> libtoolize: /usr/bin/glibtoolize >>> make : /usr/bin/make (GNU Make 3.81) >>> awk : () >>> >>> It is still reporting a 2.2.4 version of libtool. >>> ... >>> ... >>> ... >>> In the end, it has been compiled and installed though >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120511/997c5988/attachment-0001.html From freeswitch at scottisheyes.com Fri May 11 21:30:31 2012 From: freeswitch at scottisheyes.com (James) Date: Fri, 11 May 2012 10:30:31 -0700 Subject: [Freeswitch-users] XML_CURL directory configuration does not fail over to static XML In-Reply-To: References: Message-ID: I imagine it is so it can remain dynamic. When you change the static XML configs, you need to trigger a reload (you already know this most likely). XML curl doesn't need this at the moment, but would if it cached the results indefinitely in memory. It becomes a tradeoff between efficiency and dynamic behaviour. It might be an interesting XML CURL option to have, but personally, I value having dynamic results higher. You could, however, introduce caching in your XML curl web service layer to introduce efficiency and scalability. On Fri, May 11, 2012 at 8:44 AM, Vik Killa wrote: > Upon further investigation, it seems all directory parameters and > attributes from static XML are stored in memory and can be accessed if > XML_CURL fails. This is not true for any directory variables though. > Is there a specific reason why FS does not store directory variables > in memory when using XML_CURL? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120511/c19c6799/attachment.html From vipkilla at gmail.com Fri May 11 21:53:21 2012 From: vipkilla at gmail.com (Vik Killa) Date: Fri, 11 May 2012 13:53:21 -0400 Subject: [Freeswitch-users] XML_CURL directory configuration does not fail over to static XML In-Reply-To: References: Message-ID: > I imagine it is so it can remain dynamic. ?When you change the static XML > configs, you need to trigger a reload (you already know this most likely). After I modify the static XML and do reloadxml, if XML_CURL is still binding to directory and the request fails, the directory variables are still not read from the static XML. I understand the differences and trade-offs between the two. What I'm trying to accomplish is a HA solution that does not need to rely on XML_CURL 100%. In other words, if XML_CURL fails for some reason, all the XML will be close to up-to-date (maybe reloadxml every night?) static in memory. For some reason, the directory parameters and attributes as well as the entire dialplan will work from static XML if XML_CURL fails but not the directory variables. From freeswitch at scottisheyes.com Fri May 11 22:03:25 2012 From: freeswitch at scottisheyes.com (James) Date: Fri, 11 May 2012 11:03:25 -0700 Subject: [Freeswitch-users] XML_CURL directory configuration does not fail over to static XML In-Reply-To: References: Message-ID: Ah, I misunderstood - how, I don't know. :) This sounds like something appropriate to enter as a JIRA maybe? On Fri, May 11, 2012 at 10:53 AM, Vik Killa wrote: > > I imagine it is so it can remain dynamic. When you change the static XML > > configs, you need to trigger a reload (you already know this most > likely). > After I modify the static XML and do reloadxml, if XML_CURL is still > binding to directory and the request fails, the directory variables > are still not read from the static XML. I understand the differences > and trade-offs between the two. What I'm trying to accomplish is a HA > solution that does not need to rely on XML_CURL 100%. In other words, > if XML_CURL fails for some reason, all the XML will be close to > up-to-date (maybe reloadxml every night?) static in memory. For some > reason, the directory parameters and attributes as well as the entire > dialplan will work from static XML if XML_CURL fails but not the > directory variables. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120511/fd2287f0/attachment.html From gmaruzz at gmail.com Fri May 11 22:33:34 2012 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Fri, 11 May 2012 20:33:34 +0200 Subject: [Freeswitch-users] SMS with GSMOpen not support Thai charecters encoded in utf-8 In-Reply-To: References: Message-ID: Thanks Chaiyawut , no need to duplicate here what you correctly reported into Jira. I answered on Jira, that's the way to handle bugs :). -giovanni On Fri, May 11, 2012 at 4:14 PM, Chaiyawut Sookplang wrote: > I used GSMOpen to send SMS and I got this problem. Seem like it > doesn't support Thai alphabet even I encoded as UTF-8. The empty > message was sent by the way. > > 2012-05-11 20:49:53.235742 [NOTICE] mod_gsmopen.cpp:2856 rev > adae5e5|bbae68a[(nil)|37 ][NOTICA 2856 ][interface_1][-1, 0, 0] > chat_send(proto=sms, from=interface_1, to=0860216060, subject=SIMPLE > MESSAGE, body=???????, type=NULL, hint=interface_1) > 2012-05-11 20:49:53.295735 [ERR] gsmopen_protocol.cpp:2405 rev > adae5e5|bbae68a[(nil)|37 ][ERRORA 2405 ][interface_1][-1, 0, 0] error: > Invalid or incomplete multibyte or wide character 84 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From philq at qsystemsengineering.com Fri May 11 22:41:24 2012 From: philq at qsystemsengineering.com (Phil Quesinberry) Date: Fri, 11 May 2012 14:41:24 -0400 Subject: [Freeswitch-users] Bypass media succeeds from extension to gateway but fails from extension to extension Message-ID: <01b601cd2fa5$adc105b0$09431110$@com> Ken, Thanks for taking the time to write that out. I understand that and that makes perfect sense, although the endpoints in this case are configured to report their external IP addresses, either through STUN or a static NAT IP entry. What still is unclear to me is why this failed when FS was able to successfully bridge the same extension directly to the PSTN gateway. As for my skill level before I continue - I?ve got years and years of advanced networking experience but I?ve only been running FS for a few months and I still have a lot to learn, so please forgive me if I?m still missing the point. I feel like I?m really starting to make sense of it all now though, which probably only serves to make me a danger to myself and others. I don?t want to waste your time with dumb questions, but I get the impression that there aren?t that many people out there with the level of understanding to be able to answer questions like this who are actually willing to do so. I REALLY appreciate your time, I want to be sure that I fully understand this. The situation here, endpoint A is a local extension on the same LAN as FS: Endpoint A (192.168.1.4) -> FS (192.168.1.6) -> NAT A -> Internet -> NAT B -> Endpoint B (10.0.0.x/24) Endpoint B?s WAN address is 71.179.xx.xx >From the traffic I pastebinned, we can see that Endpoint A is sending its external WAN address info to FS, right before FS sends Endpoint A?s internal LAN address to Endpoint B, so I would think that FS should be passing Endpoint A?s WAN address along instead of its LAN address for media: recv 722 bytes from udp/[192.168.1.4]:5060 at 14:03:20.015051: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 71.179.xx.xx;rport=5060;branch=z9hG4bKgKQKBycpSp0XD;received=192.168.1.6 From: ;tag=6F6tm8F5v3D5j To: "Phil" ;tag=fce60a24bc Call-ID: cabfc783414db5c6 CSeq: 27863635 UPDATE Contact: "Phil" ;+sip.instance="" Server: Aastra 9480i/3.2.2.2044 Session-Expires: 120;refresher=uas Supported: path, timer Content-Type: application/sdp Content-Length: 178 v=0 o=MxSIP 0 2 IN IP4 192.168.1.4 s=SIP Call c=IN IP4 71.179.xx.xx t=0 0 m=audio 16384 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv ------------------------------------------------------------------------ 2012-05-07 10:03:20.011485 [DEBUG] switch_core_session.c:900 Send signal sofia/internal/102 at 192.168.1.6:5060 [BREAK] 2012-05-07 10:03:20.011485 [DEBUG] switch_core_session.c:900 Send signal sofia/internal/102 at 192.168.1.6:5060 [BREAK] 2012-05-07 10:03:20.011485 [DEBUG] switch_core_session.c:900 Send signal sofia/internal/102 at 192.168.1.6:5060 [BREAK] 2012-05-07 10:03:20.039490 [DEBUG] switch_core_session.c:754 Send signal sofia/internal/sip:225 at 74.93.xx.xx:1067 [BREAK] 2012-05-07 10:03:20.039490 [DEBUG] sofia.c:5628 Channel sofia/internal/102 at 192.168.1.6:5060 entering state [ready][200] 2012-05-07 10:03:20.039490 [DEBUG] sofia.c:5639 Remote SDP: v=0 o=MxSIP 0 2 IN IP4 192.168.1.4 s=SIP Call c=IN IP4 71.179.xx.xx t=0 0 m=audio 16384 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - a=ptime:20 send 1103 bytes to udp/[74.93.xx.xx]:1067 at 14:03:20.052233: ------------------------------------------------------------------------ INVITE sip:225 at 74.93.xx.xx:1067;transport=udp SIP/2.0 Via: SIP/2.0/UDP 71.179.xx.xx;rport;branch=z9hG4bKHvgcDSXSpZpgS Max-Forwards: 69 From: "Phil" ;tag=7rZKp308Sc4Qe To: ;tag=1672444754 Call-ID: 3682faf7-12f0-1230-4bae-feffffffffff CSeq: 27863631 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-883dd29 2012-05-06 11-27-00 +0000 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Require: timer Supported: timer, precondition, path, replaces Session-Expires: 120;refresher=uas Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 217 X-FS-Support: update_display,send_info Remote-Party-ID: "Phil" ;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1336379701 1336379703 IN IP4 71.179.xx.xx s=FreeSWITCH c=IN IP4 192.168.1.4 WRONG! t=0 0 m=audio 1638 RTP/AVP 0 8 98 3 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=30 a=silenceSupp:off - - - - a=ptime:20 ------------------------------------------------------------------------ 2012-05-07 10:03:20.039490 [DEBUG] sofia_glue.c:4817 Our existing sdp is still good [PCMU 71.179.xx.xx:16384], let's keep it. 2012-05-07 10:03:20.039490 [DEBUG] sofia_glue.c:5043 No 2833 in SDP. Disable 2833 dtmf and switch to INFO 2012-05-07 10:03:20.039490 [DEBUG] sofia.c:6212 Processing updated SDP 2012-05-07 10:03:20.039490 [DEBUG] sofia_glue.c:3214 Audio params are unchanged for sofia/internal/102 at 192.168.1.6:5060. 2012-05-07 10:03:20.039490 [DEBUG] sofia_glue.c:3224 sofia/internal/102 at 192.168.1.6:5060 Setting audio receive payload in Re-INVITE to 0 2012-05-07 10:03:20.039490 [DEBUG] switch_core_session.c:754 Send signal sofia/internal/102 at 192.168.1.6:5060 [BREAK] 2012-05-07 10:03:20.039490 [DEBUG] switch_core_session.c:900 Send signal sofia/internal/sip:225 at 74.93.xx.xx:1067 [BREAK] 2012-05-07 10:03:20.039490 [DEBUG] switch_ivr_bridge.c:1146 (sofia/internal/102 at 192.168.1.6:5060) State Change CS_EXECUTE -> CS_HIBERNATE 2012-05-07 10:03:20.039490 [DEBUG] switch_core_session.c:1205 Send signal sofia/internal/102 at 192.168.1.6:5060 [BREAK] 2012-05-07 10:03:20.039490 [DEBUG] switch_ivr_bridge.c:1147 (sofia/internal/sip:225 at 74.93.xx.xx:1067) State Change CS_PARK -> CS_HIBERNATE 2012-05-07 10:03:20.039490 [DEBUG] switch_core_session.c:1205 Send signal sofia/internal/sip:225 at 74.93.xx.xx:1067 [BREAK] 2012-05-07 10:03:20.039490 [DEBUG] switch_core_session.c:816 Send signal sofia/internal/sip:225 at 74.93.xx.xx:1067 [BREAK] 2012-05-07 10:03:20.039490 [DEBUG] switch_core_session.c:816 Send signal sofia/internal/102 at 192.168.1.6:5060 [BREAK] 2012-05-07 10:03:20.053837 [DEBUG] sofia.c:5628 Channel sofia/internal/sip:225 at 74.93.xx.xx:1067 entering state [calling][0] 2012-05-07 10:03:20.053837 [DEBUG] mod_sofia.c:2191 Not sending same id again "Phil" <102> send 1006 bytes to udp/[192.168.1.4]:5060 at 14:03:20.054056: ------------------------------------------------------------------------ INVITE sip:102 at 192.168.1.4:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 71.179.xx.xx;rport;branch=z9hG4bKj594emeXK8c3m Max-Forwards: 70 From: ;tag=6F6tm8F5v3D5j To: "Phil" ;tag=fce60a24bc Call-ID: cabfc783414db5c6 CSeq: 27863636 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-883dd29 2012-05-06 11-27-00 +0000 2012-05-07 10:03:20.053837 [DEBUG] switch_core_state_machine.c:426 (sofia/internal/sip:225 at 74.93.xx.xx:1067) State PARK going to sleep Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Require: timer Supported: timer, precondition, path, replaces 2012-05-07 10:03:20.053837 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/sip:225 at 74.93.xx.xx:1067) Running State Change CS_HIBERNATE Session-Expires: 120;refresher=uas Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 218 X-FS-Support: update_display,send_info v=0 o=FreeSWITCH 1336377747 1336377749 IN IP4 192.168.1.6 s=FreeSWITCH c=IN IP4 74.93.xx.xx t=0 0 m=audio 51720 RTP/AVP 0 8 98 3 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=30 a=silenceSupp:off - - - - a=ptime:20 ------------------------------------------------------------------------ Call fails with no audio. Now, if I try to make a call from that same endpoint to the PSTN in bypass media mode, FS sends the endpoint?s WAN address like I would expect it to, and the call works... see below. I'm getting rid of a bunch of ACKs and such for brevity and clarity but the transaction goes like this: Endpoint A sends its invite, complete with WAN address info: ------------------------------------------------------------------------ recv 1279 bytes from udp/[192.168.1.4]:5060 at 18:00:37.133916: ------------------------------------------------------------------------ INVITE sip:94109693xxx at 192.168.1.6:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.1.4;branch=z9hG4bK69b9ae7c0bd9ac684;rport Route: Proxy-Authorization: Digest username="102",realm="192.168.1.6",nonce="ecd26c6e-c2d9-4669-b842- 232e99abb7a2",uri="sip:94109693xxx at 192.168.1.6:5060;user=phone",response="6e 2edf5ebde2d6eaae7dd46e73cf5f88",algorithm=MD5,qop=auth,cnonce="6defb5b4",nc= 00000001 Max-Forwards: 70 From: "Phil" ;tag=6de7cac307 To: Call-ID: 8d9c7aa8f30bc1d3 CSeq: 30988 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "Phil" ;+sip.instance="" Min-SE: 120 Session-Expires: 120 Supported: path, 100rel, replaces, timer User-Agent: Aastra 9480i/3.2.2.2044 Content-Type: application/sdp Content-Length: 255 v=0 o=MxSIP 0 1 IN IP4 192.168.1.4 s=SIP Call c=IN IP4 71.179.xx.xx t=0 0 m=audio 16384 RTP/AVP 0 110 8 9 a=rtpmap:0 PCMU/8000 a=rtpmap:110 PCMU/16000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv Now FS sends the invite to the gateway, complete with proper info: send 1163 bytes to udp/[199.96.248.140]:5060 at 18:00:37.171788: ------------------------------------------------------------------------ INVITE sip:14109693xxx at 199.96.248.140 SIP/2.0 Via: SIP/2.0/UDP 71.179.xx.xx:5080;rport;branch=z9hG4bKK20cSjr649cta Max-Forwards: 69 From: "QSystems" ;tag=m2Djgjet89Ztj To: Call-ID: 0db1020c-1636-1230-a98d-feffffffffff CSeq: 28043554 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-883dd29 2012-05-06 11-27-00 +0000 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 279 X-accountcode: 102 X-FS-Support: update_display,send_info Remote-Party-ID: "QSystems" ;party=calling;screen=yes;privacy=off v=0 o=MxSIP 3719756330637231183 2523006830856012877 IN IP4 192.168.1.4 s=SIP Call c=IN IP4 71.179.xx.xx t=0 0 m=audio 16384 RTP/AVP 0 110 8 9 a=rtpmap:0 PCMU/8000 a=rtpmap:110 PCMU/16000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=silenceSupp:off - - - - a=ptime:20 ------------------------------------------------------------------------ In the meantime, FS receives and passes along 180 - Ringing PSTN destination answers the call, gateway comes back with upstream provider's media address: ------------------------------------------------------------------------ recv 886 bytes from udp/[199.96.248.140]:5060 at 18:00:43.320321: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 71.179.xx.xx:5080;rport=5080;branch=z9hG4bKK20cSjr649cta Record-Route: From: "QSystems" ;tag=m2Djgjet89Ztj To: ;tag=m3p9c1NF8UDyQ Call-ID: 0db1020c-1636-1230-a98d-feffffffffff CSeq: 28043554 INVITE Contact: User-Agent: AlcazarSBC 1.10 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 193 X-FS-Support: update_display,send_info v=0 o=FreeSWITCH 1336742245 1336742246 IN IP4 208.103.143.3 s=FreeSWITCH c=IN IP4 208.103.143.3 t=0 0 m=audio 16998 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - a=ptime:20 ------------------------------------------------------------------------ FS passes upstream info to Endpoint A, and then sends an update (for the display, I think): send 1045 bytes to udp/[192.168.1.4]:5060 at 18:00:43.330695: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.4;branch=z9hG4bK69b9ae7c0bd9ac684;rport=5060 From: "Phil" ;tag=6de7cac307 To: ;tag=0BaHerZS08taQ Call-ID: 8d9c7aa8f30bc1d3 CSeq: 30988 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-883dd29 2012-05-06 11-27-00 +0000 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Require: timer Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Session-Expires: 120;refresher=uac Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 193 v=0 o=FreeSWITCH 1336742245 1336742246 IN IP4 208.103.143.3 s=FreeSWITCH c=IN IP4 208.103.143.3 t=0 0 m=audio 16998 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - a=ptime:20 ------------------------------------------------------------------------ 2012-05-11 14:00:27.083254 [DEBUG] switch_core_session.c:900 Send signal sofia/internal/102 at 192.168.1.6:5060 [BREAK] send 973 bytes to udp/[192.168.1.4]:5060 at 18:00:43.331578: ------------------------------------------------------------------------ UPDATE sip:102 at 192.168.1.4:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 71.179.xx.xx;rport;branch=z9hG4bKN3ej0NXFj8ayj Max-Forwards: 70 From: ;tag=0BaHerZS08taQ To: "Phil" ;tag=6de7cac307 Call-ID: 8d9c7aa8f30bc1d3 CSeq: 28043557 UPDATE Contact: User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-883dd29 2012-05-06 11-27-00 +0000 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 193 P-Asserted-Identity: "Outbound Call" v=0 o=FreeSWITCH 1336742245 1336742246 IN IP4 208.103.143.3 s=FreeSWITCH c=IN IP4 208.103.143.3 t=0 0 m=audio 16998 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=silenceSupp:off - - - - a=ptime:20 ------------------------------------------------------------------------ Call completes normally, and everything works. Help me to understand why FS can't do the same thing for extension to extension calls. Thanks again, - Phil ---------- Ken Rice Thu May 10 19:08:53 MSD 2012 The contact IP has nothing to do with where the media goes... That?s entirely defined in the SDP... Consider this Endpoint A (192.168.100.100) -> NAT A -> FreeSWITCH (4.2.2.2) -> NAT B -> Endpoint B (192.168.100.200) Now lets assume that NAT A and NAT B are 2 separate nat gateways and that Endpoint A and Endpoint B are on 2 different physical LANs... Telling Endpoint A to talk directly to Endpoint B without proxying media will never work since the endpoints think they are on the same LAN. There is no mechanism there to allow for the redirection and automagic adjustments of ports etc so that they can talk directly to each other... Now lets change this slightly so that endpoint B is 192.168.200.200. Unless NAT A knows how to get to 192.168.200.0/24 (assuming class C sized block) and NAT B knows how to get to 192.168.100.0/24 they are both going to use their default routing which is to NAT the outgoing RTP, and forward it to the next hop... Again, RTP will not make it to other side in either direction... FreeSWITCH cant compensate due to a number of factors... Your Endpoints have to be smart enough to actually compensate for the NAT in this situation OR your NAT boxes have to compensate for it... The simple answer, don?t use bypass media in this situation, the complex answer I wont get into here... Stop by IRC and ask around... There is a 3rd option here you might want to consider, contact consulting at freeswitch.org for some professional help... This may not be specifially what you need to get going as I have no clue what your skill level happens to be, and you did say you are still learning. Good Luck! From koralu at gmail.com Fri May 11 23:36:35 2012 From: koralu at gmail.com (Adrian Andrei) Date: Fri, 11 May 2012 22:36:35 +0300 Subject: [Freeswitch-users] mod_xml_cdr is not logging In-Reply-To: References: <4456980.T5npWALvj0@sos> Message-ID: "interfering":This word points me to the right direction. I tried to call another webserver and it works. Thank you MC for the advice. On Fri, May 11, 2012 at 6:28 PM, Michael Collins wrote: > At this point I would run a packet sniffer on port 80 on your FS machine > and confirm whether or not it's even attempting to contact your web server. > Also, check to make sure you don't have IP tables or some other external > process/device interfering with your FS trying to contact the web server. > Last resort is to get a different box, re-install FS and try posting to a > web server running on the same box. > > -MC > > > On Fri, May 11, 2012 at 4:34 AM, Adrian Andrei wrote: > >> I made a test that calls cdr.php and the $_POST['cdr'] is bigger(5Mb) >> than the cdr generated localy by FS and it works. I don't think is a >> problem of webserver. >> >> Ty >> On 5/11/12, Sergey Okhapkin wrote: >> > How do you run php on web server? I got a problem recently with fresh >> > mod_fcgid, in default configuration POST request size is limited to >> 128M >> > now. >> > >> > On Friday 11 May 2012 11:41:23 Adrian Andrei wrote: >> >> When mod_xml_cdr is reloaded nothing apear in >> >> /usr/local/freeswitch/log/cdr/errors. >> >> >> >> When uncomment the cdr >> >> apears in /usr/local/freeswitch/log/cdr an I think that the process of >> >> logging cdr it works. >> >> >> >> But the problem is that the $_POST['cdr'] contain nothing. >> >> >> >> My php is as simple as possible: >> >> > >> $raw_cdr = $_POST['cdr']; >> >> $writefile = fopen('/var/www/xml-cdr/dump.txt',"a+"); >> >> fwrite($writefile, $raw_cdr); >> >> fclose($writefile); >> >> ?> >> >> >> >> When FS calls cdr.php it creats dump.txt but is empty. >> >> >> >> Ty >> >> >> >> On 5/10/12, Michael Collins wrote: >> >> > Anything in /usr/local/freeswitch/log/cdr/errors ? >> >> > >> >> > -MC >> >> > >> >> > On Thu, May 10, 2012 at 11:39 AM, Adrian Andrei >> >> > wrote: >> >> >> The logs are: >> >> >> >> >> >> 2012-05-10 21:37:03.399431 [CONSOLE] switch_loadable_module.c:1765 >> >> >> Stopping: mod_xml_cdr >> >> >> 2012-05-10 21:37:03.399431 [NOTICE] switch_event.c:1935 Event >> Binding >> >> >> deleted for mod_xml_cdr:TRAP >> >> >> 2012-05-10 21:37:03.399431 [CONSOLE] switch_loadable_module.c:1785 >> >> >> mod_xml_cdr unloaded. >> >> >> 2012-05-10 21:37:05.459437 [NOTICE] mod_xml_cdr.c:99 Rotating log >> file >> >> >> paths >> >> >> >> >> >> +OK module unloaded >> >> >> +OK Reloading XML >> >> >> +OK module loaded >> >> >> >> >> >> 2012-05-10 21:37:05.459437 [NOTICE] mod_xml_cdr.c:126 Setting log >> file >> >> >> path to /usr/local/freeswitch/log/cdr >> >> >> 2012-05-10 21:37:05.459437 [NOTICE] mod_xml_cdr.c:164 Setting err >> log >> >> >> file >> >> >> path to /usr/local/freeswitch/log/cdr/errors >> >> >> 2012-05-10 21:37:05.459437 [CONSOLE] switch_loadable_module.c:1299 >> >> >> Successfully Loaded [mod_xml_cdr] >> >> >> freeswitch at peer1> 2012-05-10 21:37:05.459437 [INFO] mod_enum.c:812 >> >> >> ENUM >> >> >> Reloaded >> >> >> 2012-05-10 21:37:05.479431 [INFO] switch_time.c:1035 Timezone >> reloaded >> >> >> 530 >> >> >> definitions >> >> >> >> >> >> >> >> >> On Thu, May 10, 2012 at 9:17 PM, Michael Collins >> >> >> >> >> >> wrote: >> >> >>> On Thu, May 10, 2012 at 8:45 AM, Adrian Andrei >> >> >>> wrote: >> >> >>>> Same result. I tried both 127.0.0.1 and localhost. etc/hosts is >> >> >>>> valid. >> >> >>> >> >> >>> what happens when you go to fs_cli and type: >> >> >>> reload mod_xml_cdr >> >> >>> >> >> >>> I'm curious. >> >> >>> -MC >> >> >> >> >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> >> >> >> >> >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> http://wiki.freeswitch.org >> >> http://www.cluecon.com >> >> >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at 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 at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at 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 at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120511/bb2edada/attachment.html From nasida at live.ru Fri May 11 23:49:25 2012 From: nasida at live.ru (Yuriy Nasida) Date: Fri, 11 May 2012 23:49:25 +0400 Subject: [Freeswitch-users] max-db-handles Message-ID: Hello guys. I have question about max-db-handles.Is it for the simultaneous connections to core DB (sqlite) only ?I use lua scripts + mysql via freeswitch.Dbh from dialplan. Does max-db-handles will affects on mysql as well ? Please advise.Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120511/cfc65a6a/attachment.html From msc at freeswitch.org Fri May 11 23:58:55 2012 From: msc at freeswitch.org (Michael Collins) Date: Fri, 11 May 2012 12:58:55 -0700 Subject: [Freeswitch-users] mod_xml_cdr is not logging In-Reply-To: References: <4456980.T5npWALvj0@sos> Message-ID: Good, because we were running out of things to try! :) -MC On Fri, May 11, 2012 at 12:36 PM, Adrian Andrei wrote: > "interfering":This word points me to the right direction. I tried to call > another webserver and it works. Thank you MC for the advice. > > On Fri, May 11, 2012 at 6:28 PM, Michael Collins wrote: > >> At this point I would run a packet sniffer on port 80 on your FS machine >> and confirm whether or not it's even attempting to contact your web server. >> Also, check to make sure you don't have IP tables or some other external >> process/device interfering with your FS trying to contact the web server. >> Last resort is to get a different box, re-install FS and try posting to a >> web server running on the same box. >> >> -MC >> >> >> On Fri, May 11, 2012 at 4:34 AM, Adrian Andrei wrote: >> >>> I made a test that calls cdr.php and the $_POST['cdr'] is bigger(5Mb) >>> than the cdr generated localy by FS and it works. I don't think is a >>> problem of webserver. >>> >>> Ty >>> On 5/11/12, Sergey Okhapkin wrote: >>> > How do you run php on web server? I got a problem recently with fresh >>> > mod_fcgid, in default configuration POST request size is limited to >>> 128M >>> > now. >>> > >>> > On Friday 11 May 2012 11:41:23 Adrian Andrei wrote: >>> >> When mod_xml_cdr is reloaded nothing apear in >>> >> /usr/local/freeswitch/log/cdr/errors. >>> >> >>> >> When uncomment the cdr >>> >> apears in /usr/local/freeswitch/log/cdr an I think that the process of >>> >> logging cdr it works. >>> >> >>> >> But the problem is that the $_POST['cdr'] contain nothing. >>> >> >>> >> My php is as simple as possible: >>> >> >> >> $raw_cdr = $_POST['cdr']; >>> >> $writefile = fopen('/var/www/xml-cdr/dump.txt',"a+"); >>> >> fwrite($writefile, $raw_cdr); >>> >> fclose($writefile); >>> >> ?> >>> >> >>> >> When FS calls cdr.php it creats dump.txt but is empty. >>> >> >>> >> Ty >>> >> >>> >> On 5/10/12, Michael Collins wrote: >>> >> > Anything in /usr/local/freeswitch/log/cdr/errors ? >>> >> > >>> >> > -MC >>> >> > >>> >> > On Thu, May 10, 2012 at 11:39 AM, Adrian Andrei >>> >> > wrote: >>> >> >> The logs are: >>> >> >> >>> >> >> 2012-05-10 21:37:03.399431 [CONSOLE] switch_loadable_module.c:1765 >>> >> >> Stopping: mod_xml_cdr >>> >> >> 2012-05-10 21:37:03.399431 [NOTICE] switch_event.c:1935 Event >>> Binding >>> >> >> deleted for mod_xml_cdr:TRAP >>> >> >> 2012-05-10 21:37:03.399431 [CONSOLE] switch_loadable_module.c:1785 >>> >> >> mod_xml_cdr unloaded. >>> >> >> 2012-05-10 21:37:05.459437 [NOTICE] mod_xml_cdr.c:99 Rotating log >>> file >>> >> >> paths >>> >> >> >>> >> >> +OK module unloaded >>> >> >> +OK Reloading XML >>> >> >> +OK module loaded >>> >> >> >>> >> >> 2012-05-10 21:37:05.459437 [NOTICE] mod_xml_cdr.c:126 Setting log >>> file >>> >> >> path to /usr/local/freeswitch/log/cdr >>> >> >> 2012-05-10 21:37:05.459437 [NOTICE] mod_xml_cdr.c:164 Setting err >>> log >>> >> >> file >>> >> >> path to /usr/local/freeswitch/log/cdr/errors >>> >> >> 2012-05-10 21:37:05.459437 [CONSOLE] switch_loadable_module.c:1299 >>> >> >> Successfully Loaded [mod_xml_cdr] >>> >> >> freeswitch at peer1> 2012-05-10 21:37:05.459437 [INFO] mod_enum.c:812 >>> >> >> ENUM >>> >> >> Reloaded >>> >> >> 2012-05-10 21:37:05.479431 [INFO] switch_time.c:1035 Timezone >>> reloaded >>> >> >> 530 >>> >> >> definitions >>> >> >> >>> >> >> >>> >> >> On Thu, May 10, 2012 at 9:17 PM, Michael Collins >>> >> >> >>> >> >> wrote: >>> >> >>> On Thu, May 10, 2012 at 8:45 AM, Adrian Andrei >>> >> >>> wrote: >>> >> >>>> Same result. I tried both 127.0.0.1 and localhost. etc/hosts is >>> >> >>>> valid. >>> >> >>> >>> >> >>> what happens when you go to fs_cli and type: >>> >> >>> reload mod_xml_cdr >>> >> >>> >>> >> >>> I'm curious. >>> >> >>> -MC >>> >> >>> >> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120511/a939254f/attachment-0001.html From tarik.bts.gi at gmail.com Sat May 12 01:03:36 2012 From: tarik.bts.gi at gmail.com (ghallab) Date: Fri, 11 May 2012 21:03:36 +0000 Subject: [Freeswitch-users] Freeswitch hardware cofig In-Reply-To: References: <4fac1a56.02d6e00a.7161.ffff99e8@mx.google.com> Message-ID: <4FAD7EA8.8090402@gmail.com> ok, I tried to collect the maximum information that I can: OS version: CentOS release 5.4 the cpu info is: http://pastebin.freeswitch.org/19039 the mem info is: http://pastebin.freeswitch.org/19040 the out put of the command top is http://pastebin.freeswitch.org/19038 the number of calls when I made the test was: 5 total. the out put of gcore and gdb: http://pastebin.freeswitch.org/19041 I want also to notice that I get all the time some warning messages like 2012-05-11 14:36:10.716722 [WARNING] sofia_reg.c:1442 SIP auth challenge (REGISTER) on sofia profile 'internal' for [7351 at domain] from ip @ip and we have 5 sofia profiles and 1 alias please give me your help as soon as possible On 05/11/2012 12:09 AM, Brian Foster wrote: > Agreed with other's assessment of the situation, you're leaving out > too many details such as intended usage, OS, chipsets, etc. I can tell > you that i have ran FS on an extremely old laptop (1.0 GHZ Single-Core > AMD w/512MB RAM) and got 40 calls doing no transcoding. I could only > do a couple of calls with transcoding, then it would just crap out. > That was on Debian Sid. > > So, there's a baseline. Sadly, I don't remember any other details on > the machine than those that I've listed. > > On the other hand, I've gotten 6000 concurrent calls up at 150-200 > CPS, then the database crapped out. That one was tuned extensively, > and was also running on an 8 core machine at 1.8 Ghz and about 4GB > RAM. There are a few others in this community who have done much > better than that (probably with better equipment). > > YMMV on all of this. There are so many factors involved that it's hard > to paint a picture unless you can be as detailed as humanly possible, > and even then sometimes that's not enough. > > -BDF > > On Thu, May 10, 2012 at 7:53 PM, curriegrad2004 > > wrote: > > If you want 30 concurrent calls, that setup *should* handle it just > fine. As the others have said, your mileage may vary so, don't take my > judgement as is. You'd probably want to do more research in this > aspect. > > Besides, a 1.6GHz Intel Atom is going to be slightly slower than a > 1.6GHz Sandy Bridge Celeron processor, so simply having a CPU clock > cycle figure is meaningless unless you know how well your processor > performs in your desired application. > > On Thu, May 10, 2012 at 3:18 PM, Michael Collins > > wrote: > > > > > > On Thu, May 10, 2012 at 12:43 PM, > wrote: > >> > >> What is the good hardware configuration > >> that needs freeswitch to support more then 30 calls? We have > now 1.6 GHz > >> in the CPU and 2 GB in the mem but is not enough. > > > > What kind of processor? What operating system? Do you want 30 > simultaneous > > calls or 30 new calls per second? Will you be doing transcoding > on these > > calls? > > > > All those details will help you decide what to do next. > > > > -MC > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Brian D. Foster > Endigo Computer LLC > Email: bdfoster at endigotech.com > Phone: 317-800-7876 > Indianapolis, Indiana, USA > > This message contains confidential information and is intended for > those listed in the "To:", "CC:", and/or "BCC:" fields of the message > header. If you are not the intended recipient you are notified that > disclosing, copying, distributing or taking any action in reliance on > the contents of this information is strictly prohibited. E-mail > transmission cannot be guaranteed to be secure or error-free as > information could be intercepted, corrupted, lost, destroyed, arrive > late or incomplete, or contain viruses. The sender therefore does not > accept liability for any errors or omissions in the contents of this > message, which arise as a result of e-mail transmission. If > verification is required please request a hard-copy version. > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120511/49b658cf/attachment.html From nathandownes at hotmail.com Sat May 12 01:58:02 2012 From: nathandownes at hotmail.com (Mr Nathan Downes) Date: Sat, 12 May 2012 07:58:02 +1000 Subject: [Freeswitch-users] RTP media Issue Message-ID: Hi First time trying this mailing list, I have my FS server in a data centre, which feeds site a, site b etc. over a fibre network. We are having really bad call quality issues at one of the sites. The residents report that every second or 2 there is a hiccup in the incoming voice and they miss a portion of the voice. It is consistent and affects everybody. I did some wireshark tracing of both sides of the calls (ext - fs and fs - provider) and to me it appears packets coming from my server have a wrong timestamp every second or so. The server is not over loaded and is pretty default Ubuntu 10.04 LTS, latest FS Head on there. I spent a week trying to diagnose with both the fibre provider and trunk provider but we couldn't figure out the cause. I then tried putting rtp-rewrite-timestamps in both the profiles (external one to provider, and internal one specifically for fibre company) and the problem appears to have gone away. I would like to figure out what is/was causing the issue and also find out if that option puts any extra overhead on the server. If there is almost no overhead I would probably leave it on and just assume I need I for this specific connection. I don't have the issue at another site over same fibre network and trunk provider, and people connecting direct over internet don't suffer same problem. I have about 2gb of wireshark traces I can put online if required. Thanks, Nathan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120512/68181ffa/attachment.html From rhuddleston at gmail.com Sat May 12 04:09:16 2012 From: rhuddleston at gmail.com (Robert-IPhone) Date: Fri, 11 May 2012 20:09:16 -0400 Subject: [Freeswitch-users] RTP media Issue In-Reply-To: References: Message-ID: <2CFE1B95-38B7-479E-B857-A726E0C8CCD8@gmail.com> Start with running an extended "mtr" between the locations. Maybe you are dealing with a latency issue. I've also seen issues with duplexing. So look at low level network issues Sent from my iPhone 4S On May 11, 2012, at 5:58 PM, Mr Nathan Downes wrote: > Hi > > First time trying this mailing list, I have my FS server in a data centre, which feeds site a, site b etc. over a fibre network. We are having really bad call quality issues at one of the sites. The residents report that every second or 2 there is a hiccup in the incoming voice and they miss a portion of the voice. It is consistent and affects everybody. I did some wireshark tracing of both sides of the calls (ext ? fs and fs ? provider) and to me it appears packets coming from my server have a wrong timestamp every second or so. The server is not over loaded and is pretty default Ubuntu 10.04 LTS, latest FS Head on there. I spent a week trying to diagnose with both the fibre provider and trunk provider but we couldn?t figure out the cause. I then tried putting rtp-rewrite-timestamps in both the profiles (external one to provider, and internal one specifically for fibre company) and the problem appears to have gone away. I would like to figure out what is/was causing the issue and also find out if that option puts any extra overhead on the server. If there is almost no overhead I would probably leave it on and just assume I need I for this specific connection. I don?t have the issue at another site over same fibre network and trunk provider, and people connecting direct over internet don?t suffer same problem. I have about 2gb of wireshark traces I can put online if required. > > Thanks, Nathan > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120511/a4f17ac9/attachment-0001.html From Dave.May at patlive.com Sat May 12 06:35:49 2012 From: Dave.May at patlive.com (Dave May) Date: Fri, 11 May 2012 22:35:49 -0400 Subject: [Freeswitch-users] Issues with RPM install Message-ID: <009DEE08474F5246848F1B355FA1C3800155E342@mail2.patlive.local> Just tried the RPM install -- great work at reducing the install time! Had some issues on CentOS 6.2 though, and wanted to post my fixes: 1. Default install doesn't include any sounds -- should probably have something basic there. 2. Sounds install via "yum install freeswitch-sounds-en-us-callie freeswitch-sounds-music" did not include sox dependency 3. Default init.d file points to incorrect freeswitch location Should be FS_FILE=${FS_FILE-/usr/bin/freeswitch} Instead of FS_FILE=${FS_FILE-/usr/freeswitch} Dave. From bdfoster at endigotech.com Sat May 12 07:00:04 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Fri, 11 May 2012 23:00:04 -0400 Subject: [Freeswitch-users] Issues with RPM install In-Reply-To: <009DEE08474F5246848F1B355FA1C3800155E342@mail2.patlive.local> References: <009DEE08474F5246848F1B355FA1C3800155E342@mail2.patlive.local> Message-ID: Thank you for your suggestions. The best place for these would be on http://jira.freeswitch.org. That way we can keep track of these suggestions and make sure they get implemented. Again, thank you for your feedback, much appreciated! -BDF On May 11, 2012 10:36 PM, "Dave May" wrote: > Just tried the RPM install -- great work at reducing the install time! > Had some issues on CentOS 6.2 though, and wanted to post my fixes: > > 1. Default install doesn't include any sounds -- should probably have > something basic there. > 2. Sounds install via "yum install freeswitch-sounds-en-us-callie > freeswitch-sounds-music" did not include sox dependency > 3. Default init.d file points to incorrect freeswitch location > Should be FS_FILE=${FS_FILE-/usr/bin/freeswitch} > Instead of FS_FILE=${FS_FILE-/usr/freeswitch} > > Dave. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120511/26a8ce51/attachment.html From nathandownes at hotmail.com Sat May 12 09:48:19 2012 From: nathandownes at hotmail.com (Mr Nathan Downes) Date: Sat, 12 May 2012 15:48:19 +1000 Subject: [Freeswitch-users] RTP media Issue In-Reply-To: <002901cd2fda$85568bb0$9003a310$@gmail.com> References: <002901cd2fda$85568bb0$9003a310$@gmail.com> Message-ID: Thanks for the response, Keys: Help Display mode Restart statistics Order of fields quit Packets Pings Host Loss% Snt Last Avg Best Wrst StDev 1. 10.77.0.254 0.0% 18911 0.2 0.2 0.2 0.2 0.0 2. 192.168.5.1 0.0% 18911 2.4 2.4 2.3 2.4 0.0 3. 10.77.4.22 0.0% 18911 2.7 3.1 2.4 4.1 0.5 1 is my switch at DC, 2 is unknown on fibre, 3 is endpoint ATA device at site about 140km away Duplex is set correctly on all interfaces, I don?t see anything on the fibre part of network it is all L2. I am not sure if it is a network issue, as the wireshark capture I was doing on the server that FS resides on, and it was reporting the stream going from FS to endpoint had the wrong timestamps. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Robert-IPhone Sent: Saturday, 12 May 2012 10:09 AM To: FreeSWITCH Users Help Cc: Subject: Re: [Freeswitch-users] RTP media Issue Start with running an extended "mtr" between the locations. Maybe you are dealing with a latency issue. I've also seen issues with duplexing. So look at low level network issues Sent from my iPhone 4S On May 11, 2012, at 5:58 PM, Mr Nathan Downes wrote: Hi First time trying this mailing list, I have my FS server in a data centre, which feeds site a, site b etc. over a fibre network. We are having really bad call quality issues at one of the sites. The residents report that every second or 2 there is a hiccup in the incoming voice and they miss a portion of the voice. It is consistent and affects everybody. I did some wireshark tracing of both sides of the calls (ext ? fs and fs ? provider) and to me it appears packets coming from my server have a wrong timestamp every second or so. The server is not over loaded and is pretty default Ubuntu 10.04 LTS, latest FS Head on there. I spent a week trying to diagnose with both the fibre provider and trunk provider but we couldn?t figure out the cause. I then tried putting rtp-rewrite-timestamps in both the profiles (external one to provider, and internal one specifically for fibre company) and the problem appears to have gone away. I would like to figure out what is/was causing the issue and also find out if that option puts any extra overhead on the server. If there is almost no overhead I would probably leave it on and just assume I need I for this specific connection. I don?t have the issue at another site over same fibre network and trunk provider, and people connecting direct over internet don?t suffer same problem. I have about 2gb of wireshark traces I can put online if required. Thanks, Nathan _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120512/fe20c93a/attachment.html From f.dinucci at alice.it Sat May 12 10:44:05 2012 From: f.dinucci at alice.it (Fernando Di Nucci) Date: Sat, 12 May 2012 08:44:05 +0200 Subject: [Freeswitch-users] freeswitch and voipstunt Message-ID: <4FAE06B5.1020908@alice.it> It is a routing issue. I have to fix my router-firewall. It lacks some iptables helper. Thank you all for your help From anthony.minessale at gmail.com Sat May 12 22:30:30 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 12 May 2012 13:30:30 -0500 Subject: [Freeswitch-users] RTP media Issue In-Reply-To: References: <002901cd2fda$85568bb0$9003a310$@gmail.com> Message-ID: With the option off the original timestamps from the inbound leg to Fs are passed out the outbound leg. With it on Fs creates new timestamps, there is no performance penalty but the disadvantage is that any jitter will he masked since the timestamps are changed . If the calls sound ok then its ok to leave it. On May 12, 2012 12:49 AM, "Mr Nathan Downes" wrote: > Thanks for the response,**** > > ** ** > > Keys: Help Display mode Restart statistics Order of fields quit** > ** > > Packets Pings**** > > Host Loss% Snt Last Avg Best Wrst > StDev**** > > 1. 10.77.0.254 0.0% 18911 0.2 0.2 0.2 > 0.2 0.0**** > > 2. 192.168.5.1 0.0% 18911 2.4 2.4 2.3 > 2.4 0.0**** > > 3. 10.77.4.22 0.0% 18911 2.7 3.1 2.4 > 4.1 0.5**** > > ** ** > > 1 is my switch at DC, 2 is unknown on fibre, 3 is endpoint ATA device at > site about 140km away**** > > ** ** > > Duplex is set correctly on all interfaces, I don?t see anything on the > fibre part of network it is all L2.**** > > ** ** > > I am not sure if it is a network issue, as the wireshark capture I was > doing on the server that FS resides on, and it was reporting the stream > going from FS to endpoint had the wrong timestamps.**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of * > Robert-IPhone > *Sent:* Saturday, 12 May 2012 10:09 AM > *To:* FreeSWITCH Users Help > *Cc:* > *Subject:* Re: [Freeswitch-users] RTP media Issue**** > > ** ** > > Start with running an extended "mtr" between the locations.**** > > ** ** > > Maybe you are dealing with a latency issue.**** > > ** ** > > I've also seen issues with duplexing.**** > > ** ** > > So look at low level network issues**** > > > > Sent from my iPhone 4S**** > > > On May 11, 2012, at 5:58 PM, Mr Nathan Downes > wrote:**** > > Hi**** > > **** > > First time trying this mailing list, I have my FS server in a data > centre, which feeds site a, site b etc. over a fibre network. We are > having really bad call quality issues at one of the sites. The residents > report that every second or 2 there is a hiccup in the incoming voice and > they miss a portion of the voice. It is consistent and affects everybody. > I did some wireshark tracing of both sides of the calls (ext ? fs and fs ? > provider) and to me it appears packets coming from my server have a wrong > timestamp every second or so. The server is not over loaded and is pretty > default Ubuntu 10.04 LTS, latest FS Head on there. I spent a week trying > to diagnose with both the fibre provider and trunk provider but we couldn?t > figure out the cause. I then tried putting rtp-rewrite-timestamps in both > the profiles (external one to provider, and internal one specifically for > fibre company) and the problem appears to have gone away. I would like to > figure out what is/was causing the issue and also find out if that option > puts any extra overhead on the server. If there is almost no overhead I > would probably leave it on and just assume I need I for this specific > connection. I don?t have the issue at another site over same fibre network > and trunk provider, and people connecting direct over internet don?t suffer > same problem. I have about 2gb of wireshark traces I can put online if > required.**** > > **** > > Thanks, Nathan**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120512/f7509592/attachment-0001.html From anthony.minessale at gmail.com Sun May 13 03:05:55 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 12 May 2012 18:05:55 -0500 Subject: [Freeswitch-users] max-db-handles In-Reply-To: References: Message-ID: Yes it applies to max amount of cached db handles spawned for Simon use. It applies more for odbc than sqlite. On May 11, 2012 2:50 PM, "Yuriy Nasida" wrote: > Hello guys. > > I have question about max-db-handles. > Is it for the simultaneous connections to core DB (sqlite) only ? > I use lua scripts + mysql via freeswitch.Dbh from dialplan. Does > max-db-handles will affects on mysql as well ? > > Please advise. > Thanks. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120512/de84a217/attachment.html From nathandownes at hotmail.com Sun May 13 03:33:05 2012 From: nathandownes at hotmail.com (Mr Nathan Downes) Date: Sun, 13 May 2012 09:33:05 +1000 Subject: [Freeswitch-users] RTP media Issue In-Reply-To: <007001cd3072$82416060$86c42120$@gmail.com> References: <002901cd2fda$85568bb0$9003a310$@gmail.com> <007001cd3072$82416060$86c42120$@gmail.com> Message-ID: Hi Anthony, Thanks for the input, I am happy to leave it on if there is no performance penalty, it's a quad core xeon so it isn't even scratching the surface yet. The weird thing is if I do a wireshark on the interface that goes from endpoint - fs and one on the fs - trunk at the same time I don't see the same issue coming in, that is what was confusing me. Maybe the endpoint isn't handling the timestamp issue as well as FS is, because the problem they report is worse than what I hear in the trace. We can consider this issue resolved unless I find out it is causing some other issue, I haven't found jitter too much problem yet as it is all over fibre and I have a 2-3ms end to end latency. Keep up the good work on a brilliant product! From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Sunday, 13 May 2012 4:31 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] RTP media Issue With the option off the original timestamps from the inbound leg to Fs are passed out the outbound leg. With it on Fs creates new timestamps, there is no performance penalty but the disadvantage is that any jitter will he masked since the timestamps are changed . If the calls sound ok then its ok to leave it. On May 12, 2012 12:49 AM, "Mr Nathan Downes" wrote: Thanks for the response, Keys: Help Display mode Restart statistics Order of fields quit Packets Pings Host Loss% Snt Last Avg Best Wrst StDev 1. 10.77.0.254 0.0% 18911 0.2 0.2 0.2 0.2 0.0 2. 192.168.5.1 0.0% 18911 2.4 2.4 2.3 2.4 0.0 3. 10.77.4.22 0.0% 18911 2.7 3.1 2.4 4.1 0.5 1 is my switch at DC, 2 is unknown on fibre, 3 is endpoint ATA device at site about 140km away Duplex is set correctly on all interfaces, I don't see anything on the fibre part of network it is all L2. I am not sure if it is a network issue, as the wireshark capture I was doing on the server that FS resides on, and it was reporting the stream going from FS to endpoint had the wrong timestamps. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Robert-IPhone Sent: Saturday, 12 May 2012 10:09 AM To: FreeSWITCH Users Help Cc: Subject: Re: [Freeswitch-users] RTP media Issue Start with running an extended "mtr" between the locations. Maybe you are dealing with a latency issue. I've also seen issues with duplexing. So look at low level network issues Sent from my iPhone 4S On May 11, 2012, at 5:58 PM, Mr Nathan Downes wrote: Hi First time trying this mailing list, I have my FS server in a data centre, which feeds site a, site b etc. over a fibre network. We are having really bad call quality issues at one of the sites. The residents report that every second or 2 there is a hiccup in the incoming voice and they miss a portion of the voice. It is consistent and affects everybody. I did some wireshark tracing of both sides of the calls (ext - fs and fs - provider) and to me it appears packets coming from my server have a wrong timestamp every second or so. The server is not over loaded and is pretty default Ubuntu 10.04 LTS, latest FS Head on there. I spent a week trying to diagnose with both the fibre provider and trunk provider but we couldn't figure out the cause. I then tried putting rtp-rewrite-timestamps in both the profiles (external one to provider, and internal one specifically for fibre company) and the problem appears to have gone away. I would like to figure out what is/was causing the issue and also find out if that option puts any extra overhead on the server. If there is almost no overhead I would probably leave it on and just assume I need I for this specific connection. I don't have the issue at another site over same fibre network and trunk provider, and people connecting direct over internet don't suffer same problem. I have about 2gb of wireshark traces I can put online if required. Thanks, Nathan _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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 at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120513/152a9d12/attachment.html From bdfoster at endigotech.com Sun May 13 05:49:34 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Sat, 12 May 2012 21:49:34 -0400 Subject: [Freeswitch-users] Huawei E169 Group Buy In-Reply-To: References: Message-ID: Fellow freeswitchers, I am sending this email out to let you all know that I am setting up a group buy for the Huawei E169. It should be compatable with mod_gsmopen, and that will be tested before we go and do a buy. The reason why there is interest in this particular model is the fact that it does have an external antenna jack. Otherwise they are fairly close to the E1550. If you are interested, make sure you do your research to make sure this single will work for you. No refunds. Send an email to gsmdongle at endigotech.com letting me know how many you want and the best way to get ahold of you. Deadline is May 30th to get your requests in. There will be no refunds. There will be no markup on price. Buyers are responsible for shipping. Price range should be $25-35. US/Canada residents only. Must be 18 years or older. Payment via PayPal only. Notice will be sent out a few days after May 30th to confirm we have enough interested and how much each single will be. At that time you will need to send a payment to the PayPal address. Payments will be in two stages, the first will be for the product itself, the second will be for the shipping to your address. There needs to be a minimum of 20 dongles ordered before I can go ahead and buy. Payments need to be collected prior to me going ahead and buying. If you have any questions, email me off list. -BDF -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120512/66df9a65/attachment-0001.html From steveu at coppice.org Sun May 13 09:41:57 2012 From: steveu at coppice.org (Steve Underwood) Date: Sun, 13 May 2012 13:41:57 +0800 Subject: [Freeswitch-users] Huawei E169 Group Buy In-Reply-To: References: Message-ID: <4FAF49A5.5000203@coppice.org> Hi, When looking at models of 3G dongles look carefully at whether they support the features you need. This basically comes down to supporting the voice features, and supporting the bands you need in your area. Most recent dongles with a Qualcomm chip set (which is most of them) support the voice features needed to work with FS. However, many SIM locked ones have the feature blocked. Unlocked ones should be OK. Most dongles now support quad band 2G operation, which covers most of the frequencies ever used for 2G GSM. However, most dongles only support one 3G band. Some support 2 bands. There are, however, several bands used for 3G around the planet. If the dongle you choose doesn't support your local 3G bands, you will only be able to use it for 2G communication. That might be OK. It might not. It depends on your local service offerings. Just beware. Steve On 05/13/2012 09:49 AM, Brian Foster wrote: > > Fellow freeswitchers, > > I am sending this email out to let you all know that I am setting up a > group buy for the Huawei E169. It should be compatable with > mod_gsmopen, and that will be tested before we go and do a buy. The > reason why there is interest in this particular model is the fact that > it does have an external antenna jack. Otherwise they are fairly close > to the E1550. > > If you are interested, make sure you do your research to make sure > this single will work for you. No refunds. Send an email to > gsmdongle at endigotech.com letting me > know how many you want and the best way to get ahold of you. Deadline > is May 30th to get your requests in. > > There will be no refunds. There will be no markup on price. Buyers are > responsible for shipping. Price range should be $25-35. US/Canada > residents only. Must be 18 years or older. Payment via PayPal only. > Notice will be sent out a few days after May 30th to confirm we have > enough interested and how much each single will be. At that time you > will need to send a payment to the PayPal address. Payments will be in > two stages, the first will be for the product itself, the second will > be for the shipping to your address. > > There needs to be a minimum of 20 dongles ordered before I can go > ahead and buy. Payments need to be collected prior to me going ahead > and buying. > > If you have any questions, email me off list. > > -BDF > > From bdfoster at endigotech.com Sun May 13 10:39:12 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Sun, 13 May 2012 02:39:12 -0400 Subject: [Freeswitch-users] Huawei E169 Group Buy In-Reply-To: <4FAF49A5.5000203@coppice.org> References: <4FAF49A5.5000203@coppice.org> Message-ID: The Huawei E169 is confirmed to work with mod_gsmopen. Special thanks to Marcus Brown (mzb- on #freedoh) for helping me figure all of this stuff out and for donating a E169, a SIM card, and a virtual machine to run tests. Steve, understood. As far as I can tell the bands that the E169 supports SHOULD cover most of the US carriers that support GSM. Like I said in my last email, it is up to you guys to make sure that it will work in your situation. If it doesn't I'd sell it on ebay, I don't think you'll have a problem getting rid of it. From: http://3g-modem.wetpaint.com/page/Huawei+E169+(E169G,+E169V,+K3520) specifications Huawei E169 manufacturer:Huaweimodel:E169interface:USB 2.0 (A plug)GSM frequency bands:850, 900, 1800, 1900UMTS frequency bands:900, 2100HSDPA:7,2 MBit/sHSUPA:-EDGE:236,8 KBit/sGPRS:57,6 KBit//sCSD:?connector for ext. antenna:+ (on some versions covered by housing)connector type:SMK CRS5001internal antenna diversity:+voice telephony:+NAND-flashmemory:+microSD-drive:+ (up to 8GB)rebranded versions:Vodafone K3520 Vodafone E169VAlso: http://www.alibaba.com/showroom/huawei-e169-modem.html -BDF On Sun, May 13, 2012 at 1:41 AM, Steve Underwood wrote: > Hi, > > When looking at models of 3G dongles look carefully at whether they > support the features you need. This basically comes down to supporting > the voice features, and supporting the bands you need in your area. > > Most recent dongles with a Qualcomm chip set (which is most of them) > support the voice features needed to work with FS. However, many SIM > locked ones have the feature blocked. Unlocked ones should be OK. > > Most dongles now support quad band 2G operation, which covers most of > the frequencies ever used for 2G GSM. However, most dongles only support > one 3G band. Some support 2 bands. There are, however, several bands > used for 3G around the planet. If the dongle you choose doesn't support > your local 3G bands, you will only be able to use it for 2G > communication. That might be OK. It might not. It depends on your local > service offerings. Just beware. > > Steve > > On 05/13/2012 09:49 AM, Brian Foster wrote: > > > > Fellow freeswitchers, > > > > I am sending this email out to let you all know that I am setting up a > > group buy for the Huawei E169. It should be compatable with > > mod_gsmopen, and that will be tested before we go and do a buy. The > > reason why there is interest in this particular model is the fact that > > it does have an external antenna jack. Otherwise they are fairly close > > to the E1550. > > > > If you are interested, make sure you do your research to make sure > > this single will work for you. No refunds. Send an email to > > gsmdongle at endigotech.com letting me > > know how many you want and the best way to get ahold of you. Deadline > > is May 30th to get your requests in. > > > > There will be no refunds. There will be no markup on price. Buyers are > > responsible for shipping. Price range should be $25-35. US/Canada > > residents only. Must be 18 years or older. Payment via PayPal only. > > Notice will be sent out a few days after May 30th to confirm we have > > enough interested and how much each single will be. At that time you > > will need to send a payment to the PayPal address. Payments will be in > > two stages, the first will be for the product itself, the second will > > be for the shipping to your address. > > > > There needs to be a minimum of 20 dongles ordered before I can go > > ahead and buy. Payments need to be collected prior to me going ahead > > and buying. > > > > If you have any questions, email me off list. > > > > -BDF > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-800-7876 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120513/8db510a8/attachment-0001.html From oseslija at gmail.com Sun May 13 10:41:29 2012 From: oseslija at gmail.com (Ognjen Seslija) Date: Sun, 13 May 2012 08:41:29 +0200 Subject: [Freeswitch-users] Phone not registering In-Reply-To: <4FACBD4C.4000307@softnet.si> References: <4FABB4DF.5090200@softnet.si> <4FACB0EF.7070801@softnet.si> <4FACBD4C.4000307@softnet.si> Message-ID: On 922s there are bunch of nat traversal capabilities. On 11 May 2012 09:22, "Miha" wrote: > Hi @Michael, > > after I put > > > > in my internal sip profile FS send back to right dst_port. > > If I use this on live fs server, could this be causing any problems on registered phones or any easier abuse? > > Thanks! > > Miha > > > On 5/11/2012 8:25 AM, Miha wrote: > > Hi @Michael, > > thanks for your reply. > > I also noticed after I send you an email that this is NAT issue. Phone > send registration packet on port 5060 (src port is 50006), but FS do not > reply back to port 50006 but instead reply on 5060 due to this phone does > not receive 401 and and send another REGISTER packet. > > How can I deal whit this issue? > > Thank you very much for all your help! > > p.s.: I also send you separately wireshark trace that you can see this > issue. > > Regards, > Miha > > On 5/10/2012 8:30 PM, Michael Collins wrote: > > It's definitely a NAT issue. The phone is not responding to your 401 and > is instead just sending another REGISTER packet. Notice that FS is > responding on port 5060. Is that the port your phone is expecting to > receive on? > > -MC > > On Thu, May 10, 2012 at 5:30 AM, Miha wrote: > >> Hi, >> >> here is pastebin of siptrace (http://pastebin.freeswitch.org/19029). >> >> Phone on local network are registered on FS. After I put between local >> network and Phone router, phones are unable to registered on FS. >> >> On other softswitch which is not FS phones are registering (same port, >> same scenario, etc.). >> >> Phones are SPA922. >> >> What could be causing the problem? >> >> Thanks! >> >> Miha >> >> >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120513/54eb585c/attachment.html From bdfoster at endigotech.com Sun May 13 10:59:59 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Sun, 13 May 2012 02:59:59 -0400 Subject: [Freeswitch-users] Huawei E169 Group Buy In-Reply-To: References: <4FAF49A5.5000203@coppice.org> Message-ID: One thing that you do want to keep in mind is that ATT/TMobile/etc might support a certain band on their network as a whole but might not have it in your area. If there is a better alternative dongle out there that would fit USA better, I'm all ears. Let me know and we'll look into it. Also a reminder, this isn't a sale. There will be no markup in any of this, and my company isn't involved whatsoever. I could just buy 20 or so of them and sell them at a markup, but I'm not really into that side of things. So, just because I'm emailing from a company domain DOES NOT mean that my company is involved. On Sun, May 13, 2012 at 2:39 AM, Brian Foster wrote: > The Huawei E169 is confirmed to work with mod_gsmopen. Special thanks to > Marcus Brown (mzb- on #freedoh) for helping me figure all of this stuff out > and for donating a E169, a SIM card, and a virtual machine to run tests. > > Steve, understood. As far as I can tell the bands that the E169 supports > SHOULD cover most of the US carriers that support GSM. Like I said in my > last email, it is up to you guys to make sure that it will work in your > situation. If it doesn't I'd sell it on ebay, I don't think you'll have a > problem getting rid of it. > > From: http://3g-modem.wetpaint.com/page/Huawei+E169+(E169G,+E169V,+K3520) > > specifications > > Huawei E169 > > manufacturer: Huawei model: E169 interface: USB 2.0 (A plug) GSM > frequency bands: 850, 900, 1800, 1900 UMTS frequency bands: 900, 2100HSDPA:7,2 MBit/sHSUPA:-EDGE:236,8 KBit/sGPRS:57,6 KBit//sCSD:?connector for ext. antenna:+ (on some versions covered by housing)connector type:SMK CRS5001internal antenna diversity:+voice telephony:+NAND-flashmemory:+microSD-drive:+ (up to 8GB)rebranded versions:Vodafone K3520 > Vodafone E169VAlso: http://www.alibaba.com/showroom/huawei-e169-modem.html > > -BDF > > > On Sun, May 13, 2012 at 1:41 AM, Steve Underwood wrote: > >> Hi, >> >> When looking at models of 3G dongles look carefully at whether they >> support the features you need. This basically comes down to supporting >> the voice features, and supporting the bands you need in your area. >> >> Most recent dongles with a Qualcomm chip set (which is most of them) >> support the voice features needed to work with FS. However, many SIM >> locked ones have the feature blocked. Unlocked ones should be OK. >> >> Most dongles now support quad band 2G operation, which covers most of >> the frequencies ever used for 2G GSM. However, most dongles only support >> one 3G band. Some support 2 bands. There are, however, several bands >> used for 3G around the planet. If the dongle you choose doesn't support >> your local 3G bands, you will only be able to use it for 2G >> communication. That might be OK. It might not. It depends on your local >> service offerings. Just beware. >> >> Steve >> >> On 05/13/2012 09:49 AM, Brian Foster wrote: >> > >> > Fellow freeswitchers, >> > >> > I am sending this email out to let you all know that I am setting up a >> > group buy for the Huawei E169. It should be compatable with >> > mod_gsmopen, and that will be tested before we go and do a buy. The >> > reason why there is interest in this particular model is the fact that >> > it does have an external antenna jack. Otherwise they are fairly close >> > to the E1550. >> > >> > If you are interested, make sure you do your research to make sure >> > this single will work for you. No refunds. Send an email to >> > gsmdongle at endigotech.com letting me >> > know how many you want and the best way to get ahold of you. Deadline >> > is May 30th to get your requests in. >> > >> > There will be no refunds. There will be no markup on price. Buyers are >> > responsible for shipping. Price range should be $25-35. US/Canada >> > residents only. Must be 18 years or older. Payment via PayPal only. >> > Notice will be sent out a few days after May 30th to confirm we have >> > enough interested and how much each single will be. At that time you >> > will need to send a payment to the PayPal address. Payments will be in >> > two stages, the first will be for the product itself, the second will >> > be for the shipping to your address. >> > >> > There needs to be a minimum of 20 dongles ordered before I can go >> > ahead and buy. Payments need to be collected prior to me going ahead >> > and buying. >> > >> > If you have any questions, email me off list. >> > >> > -BDF >> > >> > >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Brian D. Foster > Endigo Computer LLC > Email: bdfoster at endigotech.com > Phone: 317-800-7876 > Indianapolis, Indiana, USA > > This message contains confidential information and is intended for those > listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If > you are not the intended recipient you are notified that disclosing, > copying, distributing or taking any action in reliance on the contents of > this information is strictly prohibited. E-mail transmission cannot be > guaranteed to be secure or error-free as information could be intercepted, > corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. > The sender therefore does not accept liability for any errors or omissions > in the contents of this message, which arise as a result of e-mail > transmission. If verification is required please request a hard-copy > version. > > -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-800-7876 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120513/10794dd2/attachment-0001.html From steveu at coppice.org Sun May 13 12:07:03 2012 From: steveu at coppice.org (Steve Underwood) Date: Sun, 13 May 2012 16:07:03 +0800 Subject: [Freeswitch-users] Huawei E169 Group Buy In-Reply-To: References: <4FAF49A5.5000203@coppice.org> Message-ID: <4FAF6BA7.60503@coppice.org> Hi Brian, My understanding (as an outsider) is the US mostly uses 1700MHz and 1900MHz for UMTS. The E169 doesn't support either of those bands. I don't think 2100MHz is used at all in the US. I understand 900MHz support is pretty limited. Maybe none of this matters much if 2G support is available everywhere. However, it looks like there should be a better choice for people in the US. Hua Wei make a huge number of models to suit all tastes. Regards, Steve On 05/13/2012 02:39 PM, Brian Foster wrote: > The Huawei E169 is confirmed to work with mod_gsmopen. Special thanks > to Marcus Brown (mzb- on #freedoh) for helping me figure all of this > stuff out and for donating a E169, a SIM card, and a virtual machine > to run tests. > > Steve, understood. As far as I can tell the bands that the E169 > supports SHOULD cover most of the US carriers that support GSM. Like I > said in my last email, it is up to you guys to make sure that it will > work in your situation. If it doesn't I'd sell it on ebay, I don't > think you'll have a problem getting rid of it. > > From: > http://3g-modem.wetpaint.com/page/Huawei+E169+(E169G,+E169V,+K3520) > > > specifications > > Huawei E169 > > manufacturer: Huawei > model: E169 > interface: USB 2.0 (A plug) > GSM frequency bands: 850, 900, 1800, 1900 > UMTS frequency bands: 900, 2100 > HSDPA: 7,2 MBit/s > HSUPA: - > EDGE: 236,8 KBit/s > GPRS: 57,6 KBit//s > CSD: ? > connector for ext. antenna: + (on some versions covered by housing) > connector type: SMK CRS5001 > internal antenna diversity: + > voice telephony: + > NAND-flashmemory: + > microSD-drive: + (up to 8GB) > rebranded versions: Vodafone K3520 > Vodafone E169V > > Also: http://www.alibaba.com/showroom/huawei-e169-modem.html > > -BDF > > > On Sun, May 13, 2012 at 1:41 AM, Steve Underwood > wrote: > > Hi, > > When looking at models of 3G dongles look carefully at whether they > support the features you need. This basically comes down to supporting > the voice features, and supporting the bands you need in your area. > > Most recent dongles with a Qualcomm chip set (which is most of them) > support the voice features needed to work with FS. However, many SIM > locked ones have the feature blocked. Unlocked ones should be OK. > > Most dongles now support quad band 2G operation, which covers most of > the frequencies ever used for 2G GSM. However, most dongles only > support > one 3G band. Some support 2 bands. There are, however, several bands > used for 3G around the planet. If the dongle you choose doesn't > support > your local 3G bands, you will only be able to use it for 2G > communication. That might be OK. It might not. It depends on your > local > service offerings. Just beware. > > Steve > > On 05/13/2012 09:49 AM, Brian Foster wrote: > > > > Fellow freeswitchers, > > > > I am sending this email out to let you all know that I am > setting up a > > group buy for the Huawei E169. It should be compatable with > > mod_gsmopen, and that will be tested before we go and do a buy. The > > reason why there is interest in this particular model is the > fact that > > it does have an external antenna jack. Otherwise they are fairly > close > > to the E1550. > > > > If you are interested, make sure you do your research to make sure > > this single will work for you. No refunds. Send an email to > > gsmdongle at endigotech.com > > letting me > > know how many you want and the best way to get ahold of you. > Deadline > > is May 30th to get your requests in. > > > > There will be no refunds. There will be no markup on price. > Buyers are > > responsible for shipping. Price range should be $25-35. US/Canada > > residents only. Must be 18 years or older. Payment via PayPal only. > > Notice will be sent out a few days after May 30th to confirm we have > > enough interested and how much each single will be. At that time you > > will need to send a payment to the PayPal address. Payments will > be in > > two stages, the first will be for the product itself, the second > will > > be for the shipping to your address. > > > > There needs to be a minimum of 20 dongles ordered before I can go > > ahead and buy. Payments need to be collected prior to me going ahead > > and buying. > > > > If you have any questions, email me off list. > > > > -BDF > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Brian D. Foster > Endigo Computer LLC > Email: bdfoster at endigotech.com > Phone: 317-800-7876 > Indianapolis, Indiana, USA > > This message contains confidential information and is intended for > those listed in the "To:", "CC:", and/or "BCC:" fields of the message > header. If you are not the intended recipient you are notified that > disclosing, copying, distributing or taking any action in reliance on > the contents of this information is strictly prohibited. E-mail > transmission cannot be guaranteed to be secure or error-free as > information could be intercepted, corrupted, lost, destroyed, arrive > late or incomplete, or contain viruses. The sender therefore does not > accept liability for any errors or omissions in the contents of this > message, which arise as a result of e-mail transmission. If > verification is required please request a hard-copy version. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From gmaruzz at gmail.com Sun May 13 18:20:52 2012 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Sun, 13 May 2012 16:20:52 +0200 Subject: [Freeswitch-users] Huawei E169 Group Buy In-Reply-To: <4FAF6BA7.60503@coppice.org> References: <4FAF49A5.5000203@coppice.org> <4FAF6BA7.60503@coppice.org> Message-ID: Hi FreeSWITCHers thanks bdfoster for the initiative, the group buy would be a very good start. May I ask to be contacted in private, from who has access to one of those e169, so we can arrange the possibility for me to constantly checking development and documentation of mod_gsmopen with it? (eg. to have ssh access to a development machine with an e169 connected). Obviously, count on me for buying a couple of them, so I will not need remote access anymore. -giovanni On Sun, May 13, 2012 at 10:07 AM, Steve Underwood wrote: > Hi Brian, > > My understanding (as an outsider) is the US mostly uses 1700MHz and > 1900MHz for UMTS. The E169 doesn't support either of those bands. I > don't think 2100MHz is used at all in the US. I understand 900MHz > support is pretty limited. Maybe none of this matters much if 2G support > is available everywhere. However, it looks like there should be a better > choice for people in the US. Hua Wei make a huge number of models to > suit all tastes. > > Regards, > Steve > > > On 05/13/2012 02:39 PM, Brian Foster wrote: >> The Huawei E169 is confirmed to work with mod_gsmopen. Special thanks >> to Marcus Brown (mzb- on #freedoh) for helping me figure all of this >> stuff out and for donating a E169, a SIM card, and a virtual machine >> to run tests. >> >> Steve, understood. As far as I can tell the bands that the E169 >> supports SHOULD cover most of the US carriers that support GSM. Like I >> said in my last email, it is up to you guys to make sure that it will >> work in your situation. If it doesn't I'd sell it on ebay, I don't >> think you'll have a problem getting rid of it. >> >> From: >> http://3g-modem.wetpaint.com/page/Huawei+E169+(E169G,+E169V,+K3520) >> >> >> specifications >> >> Huawei E169 >> >> manufacturer: ? ? ? ? Huawei >> model: ? ? ? ?E169 >> interface: ? ?USB 2.0 (A plug) >> GSM frequency bands: ?850, 900, 1800, 1900 >> UMTS frequency bands: ? ? ? ? 900, 2100 >> HSDPA: ? ? ? ?7,2 MBit/s >> HSUPA: ? ? ? ?- >> EDGE: ? ? ? ? 236,8 KBit/s >> GPRS: ? ? ? ? 57,6 KBit//s >> CSD: ?? >> connector for ext. antenna: ? + (on some versions covered by housing) >> connector type: ? ? ? SMK CRS5001 >> internal antenna diversity: ? + >> voice telephony: ? ? ?+ >> NAND-flashmemory: ? ? + >> microSD-drive: ? ? ? ?+ (up to 8GB) >> rebranded versions: ? Vodafone K3520 >> Vodafone E169V >> >> Also: http://www.alibaba.com/showroom/huawei-e169-modem.html >> >> -BDF >> >> >> On Sun, May 13, 2012 at 1:41 AM, Steve Underwood > > wrote: >> >> ? ? Hi, >> >> ? ? When looking at models of 3G dongles look carefully at whether they >> ? ? support the features you need. This basically comes down to supporting >> ? ? the voice features, and supporting the bands you need in your area. >> >> ? ? Most recent dongles with a Qualcomm chip set (which is most of them) >> ? ? support the voice features needed to work with FS. However, many SIM >> ? ? locked ones have the feature blocked. Unlocked ones should be OK. >> >> ? ? Most dongles now support quad band 2G operation, which covers most of >> ? ? the frequencies ever used for 2G GSM. However, most dongles only >> ? ? support >> ? ? one 3G band. Some support 2 bands. There are, however, several bands >> ? ? used for 3G around the planet. If the dongle you choose doesn't >> ? ? support >> ? ? your local 3G bands, you will only be able to use it for 2G >> ? ? communication. That might be OK. It might not. It depends on your >> ? ? local >> ? ? service offerings. Just beware. >> >> ? ? Steve >> >> ? ? On 05/13/2012 09:49 AM, Brian Foster wrote: >> ? ? > >> ? ? > Fellow freeswitchers, >> ? ? > >> ? ? > I am sending this email out to let you all know that I am >> ? ? setting up a >> ? ? > group buy for the Huawei E169. It should be compatable with >> ? ? > mod_gsmopen, and that will be tested before we go and do a buy. The >> ? ? > reason why there is interest in this particular model is the >> ? ? fact that >> ? ? > it does have an external antenna jack. Otherwise they are fairly >> ? ? close >> ? ? > to the E1550. >> ? ? > >> ? ? > If you are interested, make sure you do your research to make sure >> ? ? > this single will work for you. No refunds. Send an email to >> ? ? > gsmdongle at endigotech.com >> ? ? > ? ? > letting me >> ? ? > know how many you want and the best way to get ahold of you. >> ? ? Deadline >> ? ? > is May 30th to get your requests in. >> ? ? > >> ? ? > There will be no refunds. There will be no markup on price. >> ? ? Buyers are >> ? ? > responsible for shipping. Price range should be $25-35. US/Canada >> ? ? > residents only. Must be 18 years or older. Payment via PayPal only. >> ? ? > Notice will be sent out a few days after May 30th to confirm we have >> ? ? > enough interested and how much each single will be. At that time you >> ? ? > will need to send a payment to the PayPal address. Payments will >> ? ? be in >> ? ? > two stages, the first will be for the product itself, the second >> ? ? will >> ? ? > be for the shipping to your address. >> ? ? > >> ? ? > There needs to be a minimum of 20 dongles ordered before I can go >> ? ? > ahead and buy. Payments need to be collected prior to me going ahead >> ? ? > and buying. >> ? ? > >> ? ? > If you have any questions, email me off list. >> ? ? > >> ? ? > -BDF >> ? ? > >> ? ? > >> >> >> ? ? _________________________________________________________________________ >> ? ? Professional FreeSWITCH Consulting Services: >> ? ? consulting at freeswitch.org >> ? ? http://www.freeswitchsolutions.com >> >> ? ? >> ? ? >> >> ? ? Official FreeSWITCH Sites >> ? ? http://www.freeswitch.org >> ? ? http://wiki.freeswitch.org >> ? ? http://www.cluecon.com >> >> ? ? FreeSWITCH-users mailing list >> ? ? FreeSWITCH-users at lists.freeswitch.org >> ? ? >> ? ? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> ? ? UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> ? ? http://www.freeswitch.org >> >> >> >> >> -- >> Brian D. Foster >> Endigo Computer LLC >> Email: bdfoster at endigotech.com >> Phone: 317-800-7876 >> Indianapolis, Indiana, USA >> >> This message contains confidential information and is intended for >> those listed in the "To:", "CC:", and/or "BCC:" fields of the message >> header. If you are not the intended recipient you are notified that >> disclosing, copying, distributing or taking any action in reliance on >> the contents of this information is strictly prohibited. E-mail >> transmission cannot be guaranteed to be secure or error-free as >> information could be intercepted, corrupted, lost, destroyed, arrive >> late or incomplete, or contain viruses. The sender therefore does not >> accept liability for any errors or omissions in the contents of this >> message, which arise as a result of e-mail transmission. If >> verification is required please request a hard-copy version. >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From bdfoster at endigotech.com Sun May 13 20:01:47 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Sun, 13 May 2012 12:01:47 -0400 Subject: [Freeswitch-users] Huawei E169 Group Buy In-Reply-To: <4FAF6BA7.60503@coppice.org> References: <4FAF49A5.5000203@coppice.org> <4FAF6BA7.60503@coppice.org> Message-ID: Steve, Ill do some more research when I get home tonight. If anyone has a suggestion please contact me off list. -BDF On May 13, 2012 4:11 AM, "Steve Underwood" wrote: > Hi Brian, > > My understanding (as an outsider) is the US mostly uses 1700MHz and > 1900MHz for UMTS. The E169 doesn't support either of those bands. I > don't think 2100MHz is used at all in the US. I understand 900MHz > support is pretty limited. Maybe none of this matters much if 2G support > is available everywhere. However, it looks like there should be a better > choice for people in the US. Hua Wei make a huge number of models to > suit all tastes. > > Regards, > Steve > > > On 05/13/2012 02:39 PM, Brian Foster wrote: > > The Huawei E169 is confirmed to work with mod_gsmopen. Special thanks > > to Marcus Brown (mzb- on #freedoh) for helping me figure all of this > > stuff out and for donating a E169, a SIM card, and a virtual machine > > to run tests. > > > > Steve, understood. As far as I can tell the bands that the E169 > > supports SHOULD cover most of the US carriers that support GSM. Like I > > said in my last email, it is up to you guys to make sure that it will > > work in your situation. If it doesn't I'd sell it on ebay, I don't > > think you'll have a problem getting rid of it. > > > > From: > > http://3g-modem.wetpaint.com/page/Huawei+E169+(E169G,+E169V,+K3520) > > > > > > > specifications > > > > Huawei E169 > > > > manufacturer: Huawei > > model: E169 > > interface: USB 2.0 (A plug) > > GSM frequency bands: 850, 900, 1800, 1900 > > UMTS frequency bands: 900, 2100 > > HSDPA: 7,2 MBit/s > > HSUPA: - > > EDGE: 236,8 KBit/s > > GPRS: 57,6 KBit//s > > CSD: ? > > connector for ext. antenna: + (on some versions covered by housing) > > connector type: SMK CRS5001 > > internal antenna diversity: + > > voice telephony: + > > NAND-flashmemory: + > > microSD-drive: + (up to 8GB) > > rebranded versions: Vodafone K3520 > > Vodafone E169V > > > > Also: http://www.alibaba.com/showroom/huawei-e169-modem.html > > > > -BDF > > > > > > On Sun, May 13, 2012 at 1:41 AM, Steve Underwood > > wrote: > > > > Hi, > > > > When looking at models of 3G dongles look carefully at whether they > > support the features you need. This basically comes down to > supporting > > the voice features, and supporting the bands you need in your area. > > > > Most recent dongles with a Qualcomm chip set (which is most of them) > > support the voice features needed to work with FS. However, many SIM > > locked ones have the feature blocked. Unlocked ones should be OK. > > > > Most dongles now support quad band 2G operation, which covers most of > > the frequencies ever used for 2G GSM. However, most dongles only > > support > > one 3G band. Some support 2 bands. There are, however, several bands > > used for 3G around the planet. If the dongle you choose doesn't > > support > > your local 3G bands, you will only be able to use it for 2G > > communication. That might be OK. It might not. It depends on your > > local > > service offerings. Just beware. > > > > Steve > > > > On 05/13/2012 09:49 AM, Brian Foster wrote: > > > > > > Fellow freeswitchers, > > > > > > I am sending this email out to let you all know that I am > > setting up a > > > group buy for the Huawei E169. It should be compatable with > > > mod_gsmopen, and that will be tested before we go and do a buy. The > > > reason why there is interest in this particular model is the > > fact that > > > it does have an external antenna jack. Otherwise they are fairly > > close > > > to the E1550. > > > > > > If you are interested, make sure you do your research to make sure > > > this single will work for you. No refunds. Send an email to > > > gsmdongle at endigotech.com > > > > letting me > > > know how many you want and the best way to get ahold of you. > > Deadline > > > is May 30th to get your requests in. > > > > > > There will be no refunds. There will be no markup on price. > > Buyers are > > > responsible for shipping. Price range should be $25-35. US/Canada > > > residents only. Must be 18 years or older. Payment via PayPal only. > > > Notice will be sent out a few days after May 30th to confirm we > have > > > enough interested and how much each single will be. At that time > you > > > will need to send a payment to the PayPal address. Payments will > > be in > > > two stages, the first will be for the product itself, the second > > will > > > be for the shipping to your address. > > > > > > There needs to be a minimum of 20 dongles ordered before I can go > > > ahead and buy. Payments need to be collected prior to me going > ahead > > > and buying. > > > > > > If you have any questions, email me off list. > > > > > > -BDF > > > > > > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > -- > > Brian D. Foster > > Endigo Computer LLC > > Email: bdfoster at endigotech.com > > Phone: 317-800-7876 > > Indianapolis, Indiana, USA > > > > This message contains confidential information and is intended for > > those listed in the "To:", "CC:", and/or "BCC:" fields of the message > > header. If you are not the intended recipient you are notified that > > disclosing, copying, distributing or taking any action in reliance on > > the contents of this information is strictly prohibited. E-mail > > transmission cannot be guaranteed to be secure or error-free as > > information could be intercepted, corrupted, lost, destroyed, arrive > > late or incomplete, or contain viruses. The sender therefore does not > > accept liability for any errors or omissions in the contents of this > > message, which arise as a result of e-mail transmission. If > > verification is required please request a hard-copy version. > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120513/bf62cb95/attachment-0001.html From william.suffill at gmail.com Sun May 13 21:23:50 2012 From: william.suffill at gmail.com (William Suffill) Date: Sun, 13 May 2012 13:23:50 -0400 Subject: [Freeswitch-users] Huawei E169 Group Buy In-Reply-To: References: <4FAF49A5.5000203@coppice.org> <4FAF6BA7.60503@coppice.org> Message-ID: http://en.wikipedia.org/wiki/Cellular_frequencies Was looking for a more definitive list for ATT and TMO bands. That should be the majority of the GSM in the US on a national level. Many MNVOs etc but underlying network tends to be ATT or TMO. Now the next question would be which MNVO gives cheap enough SIM only accounts to play with this with FreeSWITCH? -- W From curriegrad2004 at gmail.com Sun May 13 22:32:30 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Sun, 13 May 2012 11:32:30 -0700 Subject: [Freeswitch-users] Huawei E169 Group Buy In-Reply-To: References: <4FAF49A5.5000203@coppice.org> <4FAF6BA7.60503@coppice.org> Message-ID: >From what I've known, the 850/1900 bands are primarily used for UMTS up here in North America. Rogers, Telus and Bell use those UMTS frequencies up here in Canada. However, the UMTS AWS (1700) band that Steve was talking about, it's mainly used by Wind and Mobilicity in Canada. From my limited knowledge of how the mobile carriers use in the USA, T-Mobile uses the UMTS AWS band for their 3G service while AT&T uses the 850/1900 UMTS bands. Hope this clears up any confusion for people who are in North America and are willing to try these dongles out with FS. On Sun, May 13, 2012 at 10:23 AM, William Suffill wrote: > http://en.wikipedia.org/wiki/Cellular_frequencies > > Was looking for a more definitive list for ATT and TMO bands. That > should be the majority of the GSM in the US on a national level. Many > MNVOs etc but underlying network tends to be ATT or TMO. > > Now the next question would be which MNVO gives cheap enough SIM only > accounts to play with this with FreeSWITCH? > > -- W > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From david at styleflare.com Sun May 13 23:28:07 2012 From: david at styleflare.com (David J) Date: Sun, 13 May 2012 15:28:07 -0400 Subject: [Freeswitch-users] Huawei E169 Group Buy In-Reply-To: References: Message-ID: If the price is $25 I will take 2. On May 12, 2012 9:51 PM, "Brian Foster" wrote: > Fellow freeswitchers, > > I am sending this email out to let you all know that I am setting up a > group buy for the Huawei E169. It should be compatable with mod_gsmopen, > and that will be tested before we go and do a buy. The reason why there is > interest in this particular model is the fact that it does have an external > antenna jack. Otherwise they are fairly close to the E1550. > > If you are interested, make sure you do your research to make sure this > single will work for you. No refunds. Send an email to > gsmdongle at endigotech.com letting me know how many you want and the best > way to get ahold of you. Deadline is May 30th to get your requests in. > > There will be no refunds. There will be no markup on price. Buyers are > responsible for shipping. Price range should be $25-35. US/Canada residents > only. Must be 18 years or older. Payment via PayPal only. Notice will be > sent out a few days after May 30th to confirm we have enough interested and > how much each single will be. At that time you will need to send a payment > to the PayPal address. Payments will be in two stages, the first will be > for the product itself, the second will be for the shipping to your > address. > > There needs to be a minimum of 20 dongles ordered before I can go ahead > and buy. Payments need to be collected prior to me going ahead and buying. > > If you have any questions, email me off list. > > -BDF > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120513/4db9e2a2/attachment.html From bdfoster at endigotech.com Sun May 13 23:32:02 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Sun, 13 May 2012 15:32:02 -0400 Subject: [Freeswitch-users] Huawei E169 Group Buy In-Reply-To: References: Message-ID: Please send an email to gsmdongle at endigotech.com with how many you want and the best way to reach you. You aren't obligated to buy until we actually go and order them. Just getting a head count at this stage. don't know what the exact price will be yet, the more that are interested the lower the price is. -BDF On May 13, 2012 3:29 PM, "David J" wrote: > If the price is $25 I will take 2. > On May 12, 2012 9:51 PM, "Brian Foster" wrote: > >> Fellow freeswitchers, >> >> I am sending this email out to let you all know that I am setting up a >> group buy for the Huawei E169. It should be compatable with mod_gsmopen, >> and that will be tested before we go and do a buy. The reason why there is >> interest in this particular model is the fact that it does have an external >> antenna jack. Otherwise they are fairly close to the E1550. >> >> If you are interested, make sure you do your research to make sure this >> single will work for you. No refunds. Send an email to >> gsmdongle at endigotech.com letting me know how many you want and the best >> way to get ahold of you. Deadline is May 30th to get your requests in. >> >> There will be no refunds. There will be no markup on price. Buyers are >> responsible for shipping. Price range should be $25-35. US/Canada residents >> only. Must be 18 years or older. Payment via PayPal only. Notice will be >> sent out a few days after May 30th to confirm we have enough interested and >> how much each single will be. At that time you will need to send a payment >> to the PayPal address. Payments will be in two stages, the first will be >> for the product itself, the second will be for the shipping to your >> address. >> >> There needs to be a minimum of 20 dongles ordered before I can go ahead >> and buy. Payments need to be collected prior to me going ahead and buying. >> >> If you have any questions, email me off list. >> >> -BDF >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120513/1ee95ab3/attachment.html From tculjaga at gmail.com Mon May 14 02:51:17 2012 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Mon, 14 May 2012 00:51:17 +0200 Subject: [Freeswitch-users] joining a bridged call to a conference Message-ID: hello, i got an established call A calling B. A is a number outside FS (any number e.g. 16084191500), B is an extension registered to FS (e.g. 1002). after some time, i would like to join the call A => B to a conference roon e.g. 3001 (of course using ESL) any idea how to do it smooth ? whats the best practice for this ? i tried uuid_transfer but whatever i do it keeps killing one leg... thanks for help, T. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120514/8ac0235d/attachment.html From chris at opencsta.org Mon May 14 09:03:35 2012 From: chris at opencsta.org (Chris Mylonas) Date: Mon, 14 May 2012 15:03:35 +1000 Subject: [Freeswitch-users] mod_xml_scgi compilation error Message-ID: <36BA7D4D-12BC-4676-B60F-F1A61B9003D8@opencsta.org> Hi FS Users, I was building FS on a box to test out the mod_gsmopen stuff that has been mentioned on the list recently. Updated CentOS 5 this morning, checked out a new git head. Getting this message that says "warnings being treated as errors" Not sure if that is what is killing the build or if there genuinely is an error. Any help would be great, Cheers Chris making all mod_xml_scgi Compiling ../../../../libs/libscgi/src/scgi.c... cc1: warnings being treated as errors ../../../../libs/libscgi/src/scgi.c: In function ?scgi_build_message?: ../../../../libs/libscgi/src/scgi.c:94: warning: format ?%ld? expects type ?long int?, but argument 4 has type ?size_t? ../../../../libs/libscgi/src/scgi.c:94: warning: format ?%ld? expects type ?long int?, but argument 4 has type ?size_t? ../../../../libs/libscgi/src/scgi.c:103: warning: format ?%ld? expects type ?long int?, but argument 4 has type ?size_t? ../../../../libs/libscgi/src/scgi.c:103: warning: format ?%ld? expects type ?long int?, but argument 4 has type ?size_t? ../../../../libs/libscgi/src/scgi.c:110: warning: format ?%ld? expects type ?long int?, but argument 4 has type ?size_t? ../../../../libs/libscgi/src/scgi.c:110: warning: format ?%ld? expects type ?long int?, but argument 4 has type ?size_t? make[5]: *** [../../../../libs/libscgi/src/scgi.o] Error 1 make[4]: *** [all] Error 1 make[3]: *** [mod_xml_scgi-all] Error 1 make[2]: *** [all-recursive] Error 1 make[1]: *** [all-recursive] Error 1 make: *** [all] Error 2 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120514/f3a0164c/attachment-0001.html From peter.olsson at visionutveckling.se Mon May 14 09:40:29 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Mon, 14 May 2012 05:40:29 +0000 Subject: [Freeswitch-users] joining a bridged call to a conference In-Reply-To: References: Message-ID: I think "uuid_transfer -both" should work. If it's not working, there might be some kind of media negotiation issue (for instance if bypass media is used). /Peter 14 maj 2012 kl. 01:01 skrev "Tihomir Culjaga" : > hello, > > i got an established call A calling B. A is a number outside FS (any number e.g. 16084191500), B is an extension registered to FS (e.g. 1002). > > after some time, i would like to join the call A => B to a conference roon e.g. 3001 (of course using ESL) > > any idea how to do it smooth ? > whats the best practice for this ? > > i tried uuid_transfer but whatever i do it keeps killing one leg... > > > thanks for help, > T. > > > > !DSPAM:4fb03a3f32761890320954! > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > !DSPAM:4fb03a3f32761890320954! From chris at opencsta.org Mon May 14 11:30:53 2012 From: chris at opencsta.org (Chris Mylonas) Date: Mon, 14 May 2012 17:30:53 +1000 Subject: [Freeswitch-users] mod_gsmopen requires spandsp Message-ID: <1DE50890-EE01-4EEE-83A8-5B5F71BB3959@opencsta.org> Hi FS List, FYI - as a shortcut to building my freeswitch, I skip spandsp - but it looks like this mod_gsmopen wants it in there. [root at space build]# cd /usr/src/freeswitch/src/mod/endpoints/mod_gsmopen/ [root at space mod_gsmopen]# make clean [root at space mod_gsmopen]# make install Compiling gsmopen_protocol.cpp... Compiling /usr/src/freeswitch/src/mod/endpoints/mod_gsmopen/mod_gsmopen.cpp... mkdir .libs Compiling /usr/src/freeswitch/src/mod/endpoints/mod_gsmopen/mod_gsmopen.cpp ... Creating mod_gsmopen.so... /usr/bin/ld: cannot find -lspandsp collect2: ld returned 1 exit status g++ -I../../../../libs/spandsp/src -I../../../..//libs/tiff-3.8.2/libtiff -DGSMOPEN_C_VER=\"44fd0de\" -DMODGSMOPEN_C_VER=\"44fd0de\" -I/usr/src/freeswitch/libs/curl/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -shared -o .libs/mod_gsmopen.so -shared -Wl,-x .libs/mod_gsmopen.o gsmopen_protocol.o -lm /usr/src/freeswitch/.libs/libfreeswitch.so -L/usr/src/freeswitch/libs/apr-util/xml/expat/lib /usr/src/freeswitch/libs/apr-util/xml/expat/lib/.libs/libexpat.a /usr/src/freeswitch/libs/apr/.libs/libapr-1.a -luuid -lpthread -L/usr/src/freeswitch/libs/srtp -L/usr/kerberos/lib -lcrypt -lrt -lssl -lcrypto -ldl -lz -lncurses -ljpeg -L/usr/src/freeswitch/libs/spandsp/src -lspandsp -lctb-0.16 -lgsmme -Wl,--rpath -Wl,/usr/local/freeswitch/lib -Wl,--rpath -Wl,/usr/local/freeswitch/mod make[1]: *** [mod_gsmopen.so] Error 1 make: *** [install] Error 1 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120514/1be2a43a/attachment.html From chris at opencsta.org Mon May 14 11:32:14 2012 From: chris at opencsta.org (Chris Mylonas) Date: Mon, 14 May 2012 17:32:14 +1000 Subject: [Freeswitch-users] mod_xml_scgi compilation error In-Reply-To: <36BA7D4D-12BC-4676-B60F-F1A61B9003D8@opencsta.org> References: <36BA7D4D-12BC-4676-B60F-F1A61B9003D8@opencsta.org> Message-ID: <9B6666F4-AAA4-444B-8862-084C9EE32379@opencsta.org> I removed -Werror from the Makefile options line "SWITCH_AM_CFLAGS" and it build successfully. On 14/05/2012, at 3:03 PM, Chris Mylonas wrote: > Hi FS Users, > > I was building FS on a box to test out the mod_gsmopen stuff that has been mentioned on the list recently. > Updated CentOS 5 this morning, checked out a new git head. > > Getting this message that says "warnings being treated as errors" > Not sure if that is what is killing the build or if there genuinely is an error. > > Any help would be great, > Cheers > Chris > > > making all mod_xml_scgi > Compiling ../../../../libs/libscgi/src/scgi.c... > cc1: warnings being treated as errors > ../../../../libs/libscgi/src/scgi.c: In function ?scgi_build_message?: > ../../../../libs/libscgi/src/scgi.c:94: warning: format ?%ld? expects type ?long int?, but argument 4 has type ?size_t? > ../../../../libs/libscgi/src/scgi.c:94: warning: format ?%ld? expects type ?long int?, but argument 4 has type ?size_t? > ../../../../libs/libscgi/src/scgi.c:103: warning: format ?%ld? expects type ?long int?, but argument 4 has type ?size_t? > ../../../../libs/libscgi/src/scgi.c:103: warning: format ?%ld? expects type ?long int?, but argument 4 has type ?size_t? > ../../../../libs/libscgi/src/scgi.c:110: warning: format ?%ld? expects type ?long int?, but argument 4 has type ?size_t? > ../../../../libs/libscgi/src/scgi.c:110: warning: format ?%ld? expects type ?long int?, but argument 4 has type ?size_t? > make[5]: *** [../../../../libs/libscgi/src/scgi.o] Error 1 > make[4]: *** [all] Error 1 > make[3]: *** [mod_xml_scgi-all] Error 1 > make[2]: *** [all-recursive] Error 1 > make[1]: *** [all-recursive] Error 1 > make: *** [all] Error 2 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120514/31ba4745/attachment.html From andrew.paul85 at gmail.com Mon May 14 12:31:17 2012 From: andrew.paul85 at gmail.com (Andrew Paul) Date: Mon, 14 May 2012 14:01:17 +0530 Subject: [Freeswitch-users] RTP PROXY MEDIA Message-ID: Hai , I have connected asterisk with freeswitch. In freeswitch i enabled the rtp-proxy-media equals true . Whenever i am making call from asterisk to freeswitch and in freeswitch side i put on hold am able to hear MOH from asterisk. But after that when i try any transfers (REFER) it wont coming to asterisk and it is still processiong in freeswitch only. In freeswithch any method to rely sip messages directly to other end . Thanks And Regards Andrew Paul -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120514/b0a21972/attachment.html From sherifomran2000 at yahoo.com Mon May 14 12:33:53 2012 From: sherifomran2000 at yahoo.com (Sherif Omran) Date: Mon, 14 May 2012 01:33:53 -0700 (PDT) Subject: [Freeswitch-users] unallocated number error after integrating FS with Kamailio Message-ID: <1336984433.38446.YahooMailClassic@web110806.mail.gq1.yahoo.com> Hello Guys, I have Kamailio + FS running on the same server and integrated together according to this tutorial as SBC + Media Services http://kb.asipto.com/freeswitch:kamailio-3.1.x-freeswitch-1.0.6d-sbc Issues: 1- 2 Sip softphones are registered. I can see them online in kamailio. When I call FS Echo service, it works. However, when I call the other softphone number, it gives unallocated number. 2- If user is unreachable, I should get to his voice mail. But I get also unknown Invalid application voice mail. Any body figures this error? Thanks Sherif Omran -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120514/2789bcf6/attachment.html From gmaruzz at gmail.com Mon May 14 12:44:20 2012 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Mon, 14 May 2012 10:44:20 +0200 Subject: [Freeswitch-users] mod_gsmopen requires spandsp In-Reply-To: <1DE50890-EE01-4EEE-83A8-5B5F71BB3959@opencsta.org> References: <1DE50890-EE01-4EEE-83A8-5B5F71BB3959@opencsta.org> Message-ID: yes, it requires libspandsp, maybe the Makefile it's not yet tweaked to build the library automatically. So, please first build mod_spandsp, then mod_gsmopen. We'll fixx the Makefile soon, sorry for the inconvenience. -giovanni On Mon, May 14, 2012 at 9:30 AM, Chris Mylonas wrote: > Hi FS List, > > FYI - as a shortcut to building my freeswitch, I skip spandsp - but it looks > like this mod_gsmopen wants it in there. > > > [root at space build]# cd /usr/src/freeswitch/src/mod/endpoints/mod_gsmopen/ > [root at space mod_gsmopen]# make clean > [root at space mod_gsmopen]# make install > Compiling gsmopen_protocol.cpp... > Compiling > /usr/src/freeswitch/src/mod/endpoints/mod_gsmopen/mod_gsmopen.cpp... > mkdir .libs > Compiling /usr/src/freeswitch/src/mod/endpoints/mod_gsmopen/mod_gsmopen.cpp > ... > Creating mod_gsmopen.so... > /usr/bin/ld: cannot find -lspandsp > collect2: ld returned 1 exit status > g++ -I../../../../libs/spandsp/src -I../../../..//libs/tiff-3.8.2/libtiff > -DGSMOPEN_C_VER=\"44fd0de\" -DMODGSMOPEN_C_VER=\"44fd0de\" > -I/usr/src/freeswitch/libs/curl/include -I/usr/src/freeswitch/src/include > -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/libs/libteletone/src > -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g > -O2 -D_GNU_SOURCE -shared -o .libs/mod_gsmopen.so -shared -Wl,-x > .libs/mod_gsmopen.o gsmopen_protocol.o ?-lm > /usr/src/freeswitch/.libs/libfreeswitch.so > -L/usr/src/freeswitch/libs/apr-util/xml/expat/lib > /usr/src/freeswitch/libs/apr-util/xml/expat/lib/.libs/libexpat.a > /usr/src/freeswitch/libs/apr/.libs/libapr-1.a -luuid -lpthread > -L/usr/src/freeswitch/libs/srtp -L/usr/kerberos/lib -lcrypt -lrt -lssl > -lcrypto -ldl -lz -lncurses -ljpeg -L/usr/src/freeswitch/libs/spandsp/src > -lspandsp -lctb-0.16 -lgsmme ? -Wl,--rpath -Wl,/usr/local/freeswitch/lib > -Wl,--rpath -Wl,/usr/local/freeswitch/mod > make[1]: *** [mod_gsmopen.so] Error 1 > make: *** [install] Error 1 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From tarik.bts.gi at gmail.com Mon May 14 13:51:21 2012 From: tarik.bts.gi at gmail.com (ghallab) Date: Mon, 14 May 2012 09:51:21 +0000 Subject: [Freeswitch-users] load average is hight (Very urgent please) Message-ID: <4FB0D599.4000200@gmail.com> Hi all, This days am experiencing a problem with Freeswitch: it seems that it needs more hardware resources and voicemail don't work any more and it make a long delay to process call transfer. should I compile the last version to revolve the problem or try to increase hardware resources? I notice that I have anther instance of FS running fine in the same conditions. I tried to collect the maximum information that I can: OS version: CentOS release 5.4 the cpu info is: http://pastebin.freeswitch.org/19039 the mem info is: http://pastebin.freeswitch.org/19040 the out put of the command top is http://pastebin.freeswitch.org/19038 the number of calls when I made the test was: 5 total. the out put of gcore and gdb: http://pastebin.freeswitch.org/19041 number of sofia's profiles: 5 sofia profiles and 1 alias number of calls that I want: 30 simultaneous calls I want also to notice that I get all the time some warning messages like 2012-05-11 14:36:10.716722 [WARNING] sofia_reg.c:1442 SIP auth challenge (REGISTER) on sofia profile 'internal' for [7351 at domain] from ip @ip -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120514/888ce841/attachment.html From saami_mh at ymail.com Mon May 14 13:10:54 2012 From: saami_mh at ymail.com (Samira Mh) Date: Mon, 14 May 2012 02:10:54 -0700 (PDT) Subject: [Freeswitch-users] configuring gateway for making calls In-Reply-To: <1336392911.49162.YahooMailNeo@web120104.mail.ne1.yahoo.com> References: <1336392911.49162.YahooMailNeo@web120104.mail.ne1.yahoo.com> Message-ID: <1336986654.82652.YahooMailNeo@web120106.mail.ne1.yahoo.com> hi guys; i have configured gateway for making calls but it dosn't work; my config files are as follow: first create file that name " iptel.org.xml?" in the below path: ?/usr/local/freeswitch/conf/sip_profiles/external/ i have an account on?iptel.org like :(username:aimas;password:mypass) ?? --> ? ? ? ? ? ? ? the run : sofia profile external restart reloadxml reloxm and sofia status: ?Name ? ? ? ? ?Type ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? Data ? ? ?State ================================================================================================= ? ? ? ? ? ? ? ? ?external ? ? ? profile ? ? ? ? ? ? ? sip:mod_sofia at 10.0.3.15:5080 ? ? ?RUNNING (0) ? ? external::example.com ? ? ? gateway ? ? ? ? ? ? ? ? ? ?sip:joeuser at example.com ? ? ?NOREG ? ? ? ? ?external::custom ? ? ? gateway ? ? ? ? ? ? ? ? ? sip:arimas at sip.iptel.org ? ? ?REGED ? ? ? ? ? ? ? ? ?internal ? ? ? profile ? ? ? ? ? ? ? sip:mod_sofia at 10.0.3.15:5060 ? ? ?RUNNING (0) ? ? ? ? ? ? ? ? 10.0.3.15 ? ? ? ? alias ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? internal ? ? ?ALIASED ================================================================================================= then create file name "01_custom.xml " in the below path : /usr/local/freeswitch/conf/dialplan/default the content of the files ?01_custom.xml ? are: . now when i dial?9, 1-800-555-1212 on eyebeam that register on freeswitch server the call is failed? please help; thanks? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120514/b56f0b19/attachment.html From andrew.paul85 at gmail.com Mon May 14 13:21:32 2012 From: andrew.paul85 at gmail.com (Andrew Paul) Date: Mon, 14 May 2012 14:51:32 +0530 Subject: [Freeswitch-users] configuring gateway for making calls In-Reply-To: <1336986654.82652.YahooMailNeo@web120106.mail.ne1.yahoo.com> References: <1336392911.49162.YahooMailNeo@web120104.mail.ne1.yahoo.com> <1336986654.82652.YahooMailNeo@web120106.mail.ne1.yahoo.com> Message-ID: Hi samira, Try to give some freeswitch errors and give some idea about trunk details .If the proxy is different just try with outbound-proxy parameter . R u able to get incoming calls ? . Thanks And Regards Andrew On Mon, May 14, 2012 at 2:40 PM, Samira Mh wrote: > > hi guys; > i have configured gateway for making calls but it dosn't work; > my config files are as follow: > > first create file that name " iptel.org.xml " in the below path: > /usr/local/freeswitch/conf/sip_profiles/external/ > i have an account on iptel.org like :(username:aimas;password:mypass) > > > > > > > > > --> > > > the run : > sofia profile external restart reloadxml reloxm > and sofia status: > Name Type Data State > > ================================================================================================= > external profile > sip:mod_sofia at 10.0.3.15:5080 RUNNING (0) > external::example.com gateway > sip:joeuser at example.com NOREG > external::custom gateway > sip:arimas at sip.iptel.org REGED > internal profile > sip:mod_sofia at 10.0.3.15:5060 RUNNING (0) > 10.0.3.15 alias > internal ALIASED > > ================================================================================================= > then create file name "01_custom.xml " in the below path : > /usr/local/freeswitch/conf/dialplan/default > the content of the files 01_custom.xml are: > > > expression="^9(1\d{10})$"> > data="sofia/gateway/custom/$1"/> > > > > . > now when i dial 9, 1-800-555-1212 on eyebeam that register on freeswitch > server the call is failed > > please help; > thanks > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120514/4ccdb4ac/attachment.html From shaheryarkh at googlemail.com Mon May 14 13:21:50 2012 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Mon, 14 May 2012 11:21:50 +0200 Subject: [Freeswitch-users] load average is hight (Very urgent please) In-Reply-To: <4FB0D599.4000200@gmail.com> References: <4FB0D599.4000200@gmail.com> Message-ID: Seems like you system is under SIP brute-force attack. Somebody is trying to break into your freeswitch box. Just find that user's ip and block it in IPTables. If problem persists and hacker comes from another IP then you probably need to configure Fail2Ban service. Thank you. On Mon, May 14, 2012 at 11:51 AM, ghallab wrote: > Hi all, > > This days am experiencing a problem with Freeswitch: it seems that it > needs more hardware resources and voicemail don't work any more and it make > a long delay to process call transfer. should I compile the last version to > revolve the problem or try to increase hardware resources? I notice that I > have anther instance of FS running fine in the same conditions. > > I tried to collect the maximum information that I can: > OS version: CentOS release 5.4 > the cpu info is: > http://pastebin.freeswitch.org/19039 > the mem info is: > http://pastebin.freeswitch.org/19040 > > the out put of the command top is > http://pastebin.freeswitch.org/19038 > > the number of calls when I made the test was: > 5 total. > > the out put of gcore and gdb: > http://pastebin.freeswitch.org/19041 > > number of sofia's profiles: > 5 sofia profiles and 1 alias > number of calls that I want: > 30 simultaneous calls > > I want also to notice that I get all the time some warning messages like > > 2012-05-11 14:36:10.716722 [WARNING] sofia_reg.c:1442 SIP auth challenge > (REGISTER) on sofia profile 'internal' for [7351 at domain] from ip @ip > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120514/43e1bad9/attachment-0001.html From erkan at speedingtrade.com Mon May 14 13:31:35 2012 From: erkan at speedingtrade.com (=?iso-8859-1?Q?Erkan_=DCnl=FC_=28simalube=29?=) Date: Mon, 14 May 2012 11:31:35 +0200 Subject: [Freeswitch-users] configuring gateway for making calls In-Reply-To: <1336986654.82652.YahooMailNeo@web120106.mail.ne1.yahoo.com> References: <1336392911.49162.YahooMailNeo@web120104.mail.ne1.yahoo.com> <1336986654.82652.YahooMailNeo@web120106.mail.ne1.yahoo.com> Message-ID: <005c01cd31b4$5bae8250$130b86f0$@speedingtrade.com> Hi Samira, please build first of all them in the directory/default folder a XML file like this. Now you can register with your softphone to FS. See the user_context area in this section you must link it to your dialing plan The dialing plan is located in dialing plan folder. My simple dialing plan looks like this. The context name of the dialing plan must be the same in the directory XML user_context I color it in red. I hope that this info are usefully for you. Kind regards. Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Samira Mh Gesendet: Montag, 14. Mai 2012 11:11 An: Free SWITCH Users Help Betreff: [Freeswitch-users] configuring gateway for making calls hi guys; i have configured gateway for making calls but it dosn't work; my config files are as follow: first create file that name " iptel.org.xml " in the below path: /usr/local/freeswitch/conf/sip_profiles/external/ i have an account on iptel.org like :(username:aimas;password:mypass) --> the run : sofia profile external restart reloadxml reloxm and sofia status: Name Type Data State ============================================================================ ===================== external profile sip:mod_sofia at 10.0.3.15:5080 RUNNING (0) external::example.com gateway sip:joeuser at example.com NOREG external::custom gateway sip:arimas at sip.iptel.org REGED internal profile sip:mod_sofia at 10.0.3.15:5060 RUNNING (0) 10.0.3.15 alias internal ALIASED ============================================================================ ===================== then create file name "01_custom.xml " in the below path : /usr/local/freeswitch/conf/dialplan/default the content of the files 01_custom.xml are: . now when i dial 9, 1-800-555-1212 on eyebeam that register on freeswitch server the call is failed please help; thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120514/dd323e62/attachment.html From tarik.bts.gi at gmail.com Mon May 14 14:56:49 2012 From: tarik.bts.gi at gmail.com (ghallab) Date: Mon, 14 May 2012 10:56:49 +0000 Subject: [Freeswitch-users] load average is hight (Very urgent please) In-Reply-To: References: <4FB0D599.4000200@gmail.com> Message-ID: <4FB0E4F1.2040500@gmail.com> Thank you for your quick replay. Could you tell me please how do you come to this conclusion? and if this is a brute-force, it can make my system slow all the time? On 05/14/2012 09:21 AM, Muhammad Shahzad wrote: > Seems like you system is under SIP brute-force attack. Somebody is > trying to break into your freeswitch box. Just find that user's ip and > block it in IPTables. > > If problem persists and hacker comes from another IP then you probably > need to configure Fail2Ban service. > > Thank you. > > > On Mon, May 14, 2012 at 11:51 AM, ghallab > wrote: > > Hi all, > > This days am experiencing a problem with Freeswitch: it seems > that it needs more hardware resources and voicemail don't work > any more and it make a long delay to process call transfer. > should I compile the last version to revolve the problem or > try to increase hardware resources? I notice that I have > anther instance of FS running fine in the same conditions. > > I tried to collect the maximum information that I can: > OS version: CentOS release 5.4 > the cpu info is: > http://pastebin.freeswitch.org/19039 > the mem info is: > http://pastebin.freeswitch.org/19040 > > the out put of the command top is > http://pastebin.freeswitch.org/19038 > > the number of calls when I made the test was: > 5 total. > > the out put of gcore and gdb: > http://pastebin.freeswitch.org/19041 > > number of sofia's profiles: > 5 sofia profiles and 1 alias > number of calls that I want: > 30 simultaneous calls > > I want also to notice that I get all the time some warning > messages like > > 2012-05-11 14:36:10.716722 [WARNING] sofia_reg.c:1442 SIP auth > challenge (REGISTER) on sofia profile 'internal' for [7351 at domain] > from ip @ip > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +92 334 422 40 88 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.com > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120514/db411b10/attachment-0001.html From chris at opencsta.org Mon May 14 14:08:29 2012 From: chris at opencsta.org (Chris Mylonas) Date: Mon, 14 May 2012 20:08:29 +1000 Subject: [Freeswitch-users] mod_gsmopen requires spandsp In-Reply-To: References: <1DE50890-EE01-4EEE-83A8-5B5F71BB3959@opencsta.org> Message-ID: <3484676B-F4CA-47E3-A8B0-9743291C1056@opencsta.org> All good GM, no inconvenience, just a minor gotcha ;) I am unable to compile mod_gsmopen though. It complains about not being able to find ctb-0.16 The actual filename is libctbd-0.16.so in /usr/local/lib as you can see from the 2nd lot of stuff. How do I fix this? [root at space mod_gsmopen]# make install Creating mod_gsmopen.so... /usr/bin/ld: cannot find -lctb-0.16 collect2: ld returned 1 exit status g++ -I../../../../libs/spandsp/src -I../../../..//libs/tiff-3.8.2/libtiff -DGSMOPEN_C_VER=\"44fd0de\" -DMODGSMOPEN_C_VER=\"44fd0de\" -I/usr/src/freeswitch/libs/curl/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -shared -o .libs/mod_gsmopen.so -shared -Wl,-x .libs/mod_gsmopen.o gsmopen_protocol.o /usr/src/freeswitch/.libs/libfreeswitch.so -L/usr/src/freeswitch/libs/apr-util/xml/expat/lib /usr/src/freeswitch/libs/apr-util/xml/expat/lib/.libs/libexpat.a /usr/src/freeswitch/libs/apr/.libs/libapr-1.a -luuid -lpthread -L/usr/src/freeswitch/libs/srtp -L/usr/kerberos/lib -lcrypt -lrt -lssl -lcrypto -ldl -lncurses -L/usr/src/freeswitch/libs/spandsp/src /usr/src/freeswitch/libs/spandsp/src/.libs/libspandsp.a -L/usr/src/freeswitch/libs/tiff-3.8.2/libtiff /usr/src/freeswitch/libs/tiff-3.8.2/libtiff/.libs/libtiff.a -ljpeg -lz -lm -lc -lctb-0.16 -lgsmme -Wl,--rpath -Wl,/usr/local/freeswitch/lib -Wl,--rpath -Wl,/usr/local/freeswitch/mod make[1]: *** [mod_gsmopen.so] Error 1 make: *** [install] Error 1 [root at space mod_gsmopen]# ldd /usr/local/lib/libctbd-0.16.so linux-gate.so.1 => (0x00754000) libpthread.so.0 => /lib/libpthread.so.0 (0x00e83000) libstdc++.so.6 => /usr/lib/libstdc++.so.6 (0x00d28000) libm.so.6 => /lib/libm.so.6 (0x00964000) libgcc_s.so.1 => /lib/libgcc_s.so.1 (0x00e1f000) libc.so.6 => /lib/libc.so.6 (0x00110000) /lib/ld-linux.so.2 (0x003ea000) On 14/05/2012, at 6:44 PM, Giovanni Maruzzelli wrote: > yes, it requires libspandsp, maybe the Makefile it's not yet tweaked > to build the library automatically. > > So, please first build mod_spandsp, then mod_gsmopen. > > We'll fixx the Makefile soon, sorry for the inconvenience. > > -giovanni > > On Mon, May 14, 2012 at 9:30 AM, Chris Mylonas wrote: >> Hi FS List, >> >> FYI - as a shortcut to building my freeswitch, I skip spandsp - but it looks >> like this mod_gsmopen wants it in there. >> >> >> [root at space build]# cd /usr/src/freeswitch/src/mod/endpoints/mod_gsmopen/ >> [root at space mod_gsmopen]# make clean >> [root at space mod_gsmopen]# make install >> Compiling gsmopen_protocol.cpp... >> Compiling >> /usr/src/freeswitch/src/mod/endpoints/mod_gsmopen/mod_gsmopen.cpp... >> mkdir .libs >> Compiling /usr/src/freeswitch/src/mod/endpoints/mod_gsmopen/mod_gsmopen.cpp >> ... >> Creating mod_gsmopen.so... >> /usr/bin/ld: cannot find -lspandsp >> collect2: ld returned 1 exit status >> g++ -I../../../../libs/spandsp/src -I../../../..//libs/tiff-3.8.2/libtiff >> -DGSMOPEN_C_VER=\"44fd0de\" -DMODGSMOPEN_C_VER=\"44fd0de\" >> -I/usr/src/freeswitch/libs/curl/include -I/usr/src/freeswitch/src/include >> -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/libs/libteletone/src >> -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g >> -O2 -D_GNU_SOURCE -shared -o .libs/mod_gsmopen.so -shared -Wl,-x >> .libs/mod_gsmopen.o gsmopen_protocol.o -lm >> /usr/src/freeswitch/.libs/libfreeswitch.so >> -L/usr/src/freeswitch/libs/apr-util/xml/expat/lib >> /usr/src/freeswitch/libs/apr-util/xml/expat/lib/.libs/libexpat.a >> /usr/src/freeswitch/libs/apr/.libs/libapr-1.a -luuid -lpthread >> -L/usr/src/freeswitch/libs/srtp -L/usr/kerberos/lib -lcrypt -lrt -lssl >> -lcrypto -ldl -lz -lncurses -ljpeg -L/usr/src/freeswitch/libs/spandsp/src >> -lspandsp -lctb-0.16 -lgsmme -Wl,--rpath -Wl,/usr/local/freeswitch/lib >> -Wl,--rpath -Wl,/usr/local/freeswitch/mod >> make[1]: *** [mod_gsmopen.so] Error 1 >> make: *** [install] Error 1 >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Sincerely, > > Giovanni Maruzzelli > Cell : +39-347-2665618 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120514/dd818881/attachment.html From chris at opencsta.org Mon May 14 14:15:39 2012 From: chris at opencsta.org (Chris Mylonas) Date: Mon, 14 May 2012 20:15:39 +1000 Subject: [Freeswitch-users] load average is hight (Very urgent please) In-Reply-To: <4FB0E4F1.2040500@gmail.com> References: <4FB0D599.4000200@gmail.com> <4FB0E4F1.2040500@gmail.com> Message-ID: <64E76B48-9CE9-4A6E-AD37-EC7FCD98983C@opencsta.org> If you have ngrep installed on CentOS 5 (you will need the epel repo) yum install ngrep You can view port 5060 traffic quite easily with ngrep -W byline port 5060 Your freeswitch logs probably have hundreds/thousands of registration attempts per minute - have a look at those These rego attempts need to be processed, so each one takes a small amount of resources. If you have one hundred or more rego attempts per second, then you'll certainly have a problem. Your CPU also has 512KB of cache (very fast memory) - modern Xeon's have 24x this amount, so are a little bit more resilient to a strain on resources** HTH Chris **I would imagine On 14/05/2012, at 8:56 PM, ghallab wrote: > Thank you for your quick replay. > Could you tell me please how do you come to this conclusion? > and if this is a brute-force, it can make my system slow all the time? > On 05/14/2012 09:21 AM, Muhammad Shahzad wrote: >> >> Seems like you system is under SIP brute-force attack. Somebody is trying to break into your freeswitch box. Just find that user's ip and block it in IPTables. >> >> If problem persists and hacker comes from another IP then you probably need to configure Fail2Ban service. >> >> Thank you. >> >> >> On Mon, May 14, 2012 at 11:51 AM, ghallab wrote: >> Hi all, >> This days am experiencing a problem with Freeswitch: it seems that it needs more hardware resources and voicemail don't work any more and it make a long delay to process call transfer. should I compile the last version to revolve the problem or try to increase hardware resources? I notice that I have anther instance of FS running fine in the same conditions. >> I tried to collect the maximum information that I can: >> OS version: CentOS release 5.4 >> the cpu info is: >> http://pastebin.freeswitch.org/19039 >> the mem info is: >> http://pastebin.freeswitch.org/19040 >> >> the out put of the command top is >> http://pastebin.freeswitch.org/19038 >> >> the number of calls when I made the test was: >> 5 total. >> >> the out put of gcore and gdb: >> http://pastebin.freeswitch.org/19041 >> >> number of sofia's profiles: >> 5 sofia profiles and 1 alias >> number of calls that I want: >> 30 simultaneous calls >> >> I want also to notice that I get all the time some warning messages like >> >> 2012-05-11 14:36:10.716722 [WARNING] sofia_reg.c:1442 SIP auth challenge (REGISTER) on sofia profile 'internal' for [7351 at domain] from ip @ip >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> -- >> Muhammad Shahzad >> ----------------------------------- >> CISCO Rich Media Communication Specialist (CRMCS) >> CISCO Certified Network Associate (CCNA) >> Cell: +92 334 422 40 88 >> MSN: shari_786pk at hotmail.com >> Email: shaheryarkh at googlemail.com >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120514/8cc7880a/attachment-0001.html From saami_mh at ymail.com Mon May 14 14:26:21 2012 From: saami_mh at ymail.com (Samira Mh) Date: Mon, 14 May 2012 03:26:21 -0700 (PDT) Subject: [Freeswitch-users] configuring gateway for making calls In-Reply-To: <005c01cd31b4$5bae8250$130b86f0$@speedingtrade.com> References: <1336392911.49162.YahooMailNeo@web120104.mail.ne1.yahoo.com> <1336986654.82652.YahooMailNeo@web120106.mail.ne1.yahoo.com> <005c01cd31b4$5bae8250$130b86f0$@speedingtrade.com> Message-ID: <1336991181.54394.YahooMailNeo@web120104.mail.ne1.yahoo.com> hi erkan; thanks so much for your reply; i think your mean is: ?i must configure file??01_custom.xml like?: ? and because i registered via extension:1000 on freeswitch so : ?must configure file 1000.xml ?like ?: ? ??? ????? ????? ??? ??? ????? ????? ?????? ????? ????? ????? ????? ????? ??? ? but i on't know what to be set as?? ? in the 1000.xml? please help, i am new in freeswitch? sorry for my english ;; ________________________________ From: Erkan ?nl? (simalube) To: 'FreeSWITCH Users Help' Sent: Monday, May 14, 2012 2:01 PM Subject: Re: [Freeswitch-users] configuring gateway for making calls Hi Samira, ? please build first of all them in the directory/default folder a XML file like this. ? ? ??? ????? ????? ??? ??? ????? ????? ????? ????? ????? ????? ????? ????? ??? ? ? Now you can register with your softphone to FS. See the user_context area in this section you must link it to your dialing plan The dialing plan is located in dialing plan folder. ? My simple dialing plan looks like this. ? ? ?? ???? ??????? ??????? ???? ?? ? ? The context name of the dialing plan must be the same in the directory XML user_context I color it in red. ? I hope that this info are usefully for you. ? Kind regards. ? Von:freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Samira Mh Gesendet: Montag, 14. Mai 2012 11:11 An: Free SWITCH Users Help Betreff: [Freeswitch-users] configuring gateway for making calls ? ? hi guys; i have configured gateway for making calls but it dosn't work; my config files are as follow: ? first create file that name " iptel.org.xml?" in the below path: ?/usr/local/freeswitch/conf/sip_profiles/external/ i have an account on?iptel.org like :(username:aimas;password:mypass) ?? --> ? ? ? ? ? ? ? the run : sofia profile external restart reloadxml reloxm and sofia status: ?Name ? ? ? ? ?Type ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? Data ? ? ?State ================================================================================================= ? ? ? ? ? ? ? ? ?external ? ? ? profile ? ? ? ? ? ? ? sip:mod_sofia at 10.0.3.15:5080 ? ? ?RUNNING (0) ? ? external::example.com ? ? ? gateway ? ? ? ? ? ? ? ? ? ?sip:joeuser at example.com ? ? ?NOREG ? ? ? ? ?external::custom ? ? ? gateway ? ? ? ? ? ? ? ? ? sip:arimas at sip.iptel.org ? ? ?REGED ? ? ? ? ? ? ? ? ?internal ? ? ? profile ? ? ? ? ? ? ? sip:mod_sofia at 10.0.3.15:5060 ? ? ?RUNNING (0) ? ? ? ? ? ? ? ? 10.0.3.15 ? ? ? ? alias ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? internal ? ? ?ALIASED ================================================================================================= then create file name "01_custom.xml " in the below path : /usr/local/freeswitch/conf/dialplan/default the content of the files ?01_custom.xml ? are: . now when i dial?9, 1-800-555-1212 on eyebeam that register on freeswitch server the call is failed? ? please help; thanks? _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120514/d9d095bf/attachment-0001.html From anton.jugatsu at gmail.com Mon May 14 14:43:15 2012 From: anton.jugatsu at gmail.com (Anton Kvashenkin) Date: Mon, 14 May 2012 14:43:15 +0400 Subject: [Freeswitch-users] unallocated number error after integrating FS with Kamailio In-Reply-To: <1336984433.38446.YahooMailClassic@web110806.mail.gq1.yahoo.com> References: <1336984433.38446.YahooMailClassic@web110806.mail.gq1.yahoo.com> Message-ID: About the second question: you should enable mod_voicemail, or compile it make mod_voicemail-install 2012/5/14 Sherif Omran > Hello Guys, > > I have Kamailio + FS running on the same server and integrated together > according to this tutorial as SBC + Media Services > > http://kb.asipto.com/freeswitch:kamailio-3.1.x-freeswitch-1.0.6d-sbc > > Issues: > 1- 2 Sip softphones are registered. I can see them online in kamailio. > When I call FS Echo service, it works. However, when I call the other > softphone number, it gives unallocated number. > 2- If user is unreachable, I should get to his voice mail. But I get also > unknown Invalid application voice mail. > > Any body figures this error? > > Thanks > Sherif Omran > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120514/1c164cf0/attachment.html From shaheryarkh at googlemail.com Mon May 14 15:16:53 2012 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Mon, 14 May 2012 13:16:53 +0200 Subject: [Freeswitch-users] load average is hight (Very urgent please) In-Reply-To: <4FB0E4F1.2040500@gmail.com> References: <4FB0D599.4000200@gmail.com> <4FB0E4F1.2040500@gmail.com> Message-ID: >From the warning message you mentioned you are getting on FS CLI, you need to block the IP address mentioned in the message. Thank you. On Mon, May 14, 2012 at 12:56 PM, ghallab wrote: > Thank you for your quick replay. > Could you tell me please how do you come to this conclusion? > and if this is a brute-force, it can make my system slow all the time? > > On 05/14/2012 09:21 AM, Muhammad Shahzad wrote: > > Seems like you system is under SIP brute-force attack. Somebody is trying > to break into your freeswitch box. Just find that user's ip and block it in > IPTables. > > If problem persists and hacker comes from another IP then you probably > need to configure Fail2Ban service. > > Thank you. > > > On Mon, May 14, 2012 at 11:51 AM, ghallab wrote: > >> Hi all, >> >> This days am experiencing a problem with Freeswitch: it seems that it >> needs more hardware resources and voicemail don't work any more and it make >> a long delay to process call transfer. should I compile the last version to >> revolve the problem or try to increase hardware resources? I notice that I >> have anther instance of FS running fine in the same conditions. >> >> I tried to collect the maximum information that I can: >> OS version: CentOS release 5.4 >> the cpu info is: >> http://pastebin.freeswitch.org/19039 >> the mem info is: >> http://pastebin.freeswitch.org/19040 >> >> the out put of the command top is >> http://pastebin.freeswitch.org/19038 >> >> the number of calls when I made the test was: >> 5 total. >> >> the out put of gcore and gdb: >> http://pastebin.freeswitch.org/19041 >> >> number of sofia's profiles: >> 5 sofia profiles and 1 alias >> number of calls that I want: >> 30 simultaneous calls >> >> I want also to notice that I get all the time some warning messages like >> >> 2012-05-11 14:36:10.716722 [WARNING] sofia_reg.c:1442 SIP auth challenge >> (REGISTER) on sofia profile 'internal' for [7351 at domain] from ip @ip >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +92 334 422 40 88 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.com > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120514/0b7effd3/attachment.html From gmaruzz at gmail.com Mon May 14 15:38:28 2012 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Mon, 14 May 2012 13:38:28 +0200 Subject: [Freeswitch-users] mod_gsmopen requires spandsp In-Reply-To: <3484676B-F4CA-47E3-A8B0-9743291C1056@opencsta.org> References: <1DE50890-EE01-4EEE-83A8-5B5F71BB3959@opencsta.org> <3484676B-F4CA-47E3-A8B0-9743291C1056@opencsta.org> Message-ID: you must first compile and install libctb, as per the wiki page ( http://wiki.freeswitch.org/wiki/gsmopen ) then, after installation of libctb and gsmlib (as per wiki), be sure to update your dinamic link cache, or compiler will not find then. Eg: ldconfig On 5/14/12, Chris Mylonas wrote: > All good GM, no inconvenience, just a minor gotcha ;) > I am unable to compile mod_gsmopen though. > > It complains about not being able to find ctb-0.16 > The actual filename is libctbd-0.16.so in /usr/local/lib as you can see from > the 2nd lot of stuff. > > How do I fix this? > > > [root at space mod_gsmopen]# make install > Creating mod_gsmopen.so... > /usr/bin/ld: cannot find -lctb-0.16 > collect2: ld returned 1 exit status > g++ -I../../../../libs/spandsp/src -I../../../..//libs/tiff-3.8.2/libtiff > -DGSMOPEN_C_VER=\"44fd0de\" -DMODGSMOPEN_C_VER=\"44fd0de\" > -I/usr/src/freeswitch/libs/curl/include -I/usr/src/freeswitch/src/include > -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/libs/libteletone/src > -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g > -O2 -D_GNU_SOURCE -shared -o .libs/mod_gsmopen.so -shared -Wl,-x > .libs/mod_gsmopen.o gsmopen_protocol.o > /usr/src/freeswitch/.libs/libfreeswitch.so > -L/usr/src/freeswitch/libs/apr-util/xml/expat/lib > /usr/src/freeswitch/libs/apr-util/xml/expat/lib/.libs/libexpat.a > /usr/src/freeswitch/libs/apr/.libs/libapr-1.a -luuid -lpthread > -L/usr/src/freeswitch/libs/srtp -L/usr/kerberos/lib -lcrypt -lrt -lssl > -lcrypto -ldl -lncurses -L/usr/src/freeswitch/libs/spandsp/src > /usr/src/freeswitch/libs/spandsp/src/.libs/libspandsp.a > -L/usr/src/freeswitch/libs/tiff-3.8.2/libtiff > /usr/src/freeswitch/libs/tiff-3.8.2/libtiff/.libs/libtiff.a -ljpeg -lz -lm > -lc -lctb-0.16 -lgsmme -Wl,--rpath -Wl,/usr/local/freeswitch/lib > -Wl,--rpath -Wl,/usr/local/freeswitch/mod > make[1]: *** [mod_gsmopen.so] Error 1 > make: *** [install] Error 1 > > > > [root at space mod_gsmopen]# ldd /usr/local/lib/libctbd-0.16.so > linux-gate.so.1 => (0x00754000) > libpthread.so.0 => /lib/libpthread.so.0 (0x00e83000) > libstdc++.so.6 => /usr/lib/libstdc++.so.6 (0x00d28000) > libm.so.6 => /lib/libm.so.6 (0x00964000) > libgcc_s.so.1 => /lib/libgcc_s.so.1 (0x00e1f000) > libc.so.6 => /lib/libc.so.6 (0x00110000) > /lib/ld-linux.so.2 (0x003ea000) > > > > On 14/05/2012, at 6:44 PM, Giovanni Maruzzelli wrote: > >> yes, it requires libspandsp, maybe the Makefile it's not yet tweaked >> to build the library automatically. >> >> So, please first build mod_spandsp, then mod_gsmopen. >> >> We'll fixx the Makefile soon, sorry for the inconvenience. >> >> -giovanni >> >> On Mon, May 14, 2012 at 9:30 AM, Chris Mylonas >> wrote: >>> Hi FS List, >>> >>> FYI - as a shortcut to building my freeswitch, I skip spandsp - but it >>> looks >>> like this mod_gsmopen wants it in there. >>> >>> >>> [root at space build]# cd >>> /usr/src/freeswitch/src/mod/endpoints/mod_gsmopen/ >>> [root at space mod_gsmopen]# make clean >>> [root at space mod_gsmopen]# make install >>> Compiling gsmopen_protocol.cpp... >>> Compiling >>> /usr/src/freeswitch/src/mod/endpoints/mod_gsmopen/mod_gsmopen.cpp... >>> mkdir .libs >>> Compiling >>> /usr/src/freeswitch/src/mod/endpoints/mod_gsmopen/mod_gsmopen.cpp >>> ... >>> Creating mod_gsmopen.so... >>> /usr/bin/ld: cannot find -lspandsp >>> collect2: ld returned 1 exit status >>> g++ -I../../../../libs/spandsp/src >>> -I../../../..//libs/tiff-3.8.2/libtiff >>> -DGSMOPEN_C_VER=\"44fd0de\" -DMODGSMOPEN_C_VER=\"44fd0de\" >>> -I/usr/src/freeswitch/libs/curl/include >>> -I/usr/src/freeswitch/src/include >>> -I/usr/src/freeswitch/src/include >>> -I/usr/src/freeswitch/libs/libteletone/src >>> -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 >>> -g >>> -O2 -D_GNU_SOURCE -shared -o .libs/mod_gsmopen.so -shared -Wl,-x >>> .libs/mod_gsmopen.o gsmopen_protocol.o -lm >>> /usr/src/freeswitch/.libs/libfreeswitch.so >>> -L/usr/src/freeswitch/libs/apr-util/xml/expat/lib >>> /usr/src/freeswitch/libs/apr-util/xml/expat/lib/.libs/libexpat.a >>> /usr/src/freeswitch/libs/apr/.libs/libapr-1.a -luuid -lpthread >>> -L/usr/src/freeswitch/libs/srtp -L/usr/kerberos/lib -lcrypt -lrt -lssl >>> -lcrypto -ldl -lz -lncurses -ljpeg >>> -L/usr/src/freeswitch/libs/spandsp/src >>> -lspandsp -lctb-0.16 -lgsmme -Wl,--rpath -Wl,/usr/local/freeswitch/lib >>> -Wl,--rpath -Wl,/usr/local/freeswitch/mod >>> make[1]: *** [mod_gsmopen.so] Error 1 >>> make: *** [install] Error 1 >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> Sincerely, >> >> Giovanni Maruzzelli >> Cell : +39-347-2665618 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From miha at softnet.si Mon May 14 15:38:40 2012 From: miha at softnet.si (Miha) Date: Mon, 14 May 2012 13:38:40 +0200 Subject: [Freeswitch-users] Phone not registering In-Reply-To: References: <4FABB4DF.5090200@softnet.si> <4FACB0EF.7070801@softnet.si> <4FACBD4C.4000307@softnet.si> Message-ID: <4FB0EEC0.1080801@softnet.si> @Michael, thank you for your explanation. Regards, Miha On 5/11/2012 5:24 PM, Michael Collins wrote: > You just have to be careful because the force rport can mess up phones > that properly handle NAT. If you plan to use only this model of phone > - or only phones that require force rport - then you'll be fine. If > you find that you have a mix of phone models then it's best to create > a separate profile to handle phones that are NAT friendly and then > keep the current profile for phones that need the force rport. > > -MC > > On Fri, May 11, 2012 at 12:18 AM, Miha > wrote: > > Hi @Michael, > > after I put > > > > in my internal sip profile FS send back to right dst_port. > > If I use this on live fs server, could this be causing any problems on registered phones or any easier abuse? > > Thanks! > > Miha > > > On 5/11/2012 8:25 AM, Miha wrote: >> Hi @Michael, >> >> thanks for your reply. >> >> I also noticed after I send you an email that this is NAT issue. >> Phone send registration packet on port 5060 (src port is 50006), >> but FS do not reply back to port 50006 but instead reply on 5060 >> due to this phone does not receive 401 and and send another >> REGISTER packet. >> >> How can I deal whit this issue? >> >> Thank you very much for all your help! >> >> p.s.: I also send you separately wireshark trace that you can see >> this issue. >> >> Regards, >> Miha >> >> On 5/10/2012 8:30 PM, Michael Collins wrote: >>> It's definitely a NAT issue. The phone is not responding to your >>> 401 and is instead just sending another REGISTER packet. Notice >>> that FS is responding on port 5060. Is that the port your phone >>> is expecting to receive on? >>> >>> -MC >>> >>> On Thu, May 10, 2012 at 5:30 AM, Miha >> > wrote: >>> >>> Hi, >>> >>> here is pastebin of siptrace >>> (http://pastebin.freeswitch.org/19029). >>> >>> Phone on local network are registered on FS. After I put >>> between local >>> network and Phone router, phones are unable to registered on FS. >>> >>> On other softswitch which is not FS phones are registering >>> (same port, >>> same scenario, etc.). >>> >>> Phones are SPA922. >>> >>> What could be causing the problem? >>> >>> Thanks! >>> >>> Miha >>> >>> >>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120514/1f637324/attachment-0001.html From chris at opencsta.org Mon May 14 15:53:53 2012 From: chris at opencsta.org (Chris Mylonas) Date: Mon, 14 May 2012 21:53:53 +1000 Subject: [Freeswitch-users] mod_gsmopen requires spandsp In-Reply-To: References: <1DE50890-EE01-4EEE-83A8-5B5F71BB3959@opencsta.org> <3484676B-F4CA-47E3-A8B0-9743291C1056@opencsta.org> Message-ID: <62D399FB-DFA0-4DD0-B86F-45E95026EF82@opencsta.org> Thanks for the reply Giovanni. I have done the steps on the wiki. tl;dr; should i put a prefix when i'm making the dependent mods so they don't go into /usr/local/lib, or this is why ldconfig is run - to tell the system where the libs are. All the relevant stuff is below Hope you can see something wrong, Cheers Chris e.g. here is my bash history 1050 cd freeswitch/ 1051 ls 1052 find . -name gsmlib 1053 cd src/mod/endpoints/mod_gsmopen/ 1054 ls 1055 cd gsmlib/ 1056 ls 1057 cd gsmlib-1.10-patched-13ubuntu/ 1058 ls 1059 ./configure 1060 make 1061 make install 1062 ldconfig 1063 cd /usr/src/freeswitch/src/mod/endpoints/mod_gsmopen/libctb-0.16/build 1064 make DEBUG=1 GPIB=0 1065 make DEBUG=1 GPIB=0 install 1066 ldconfig 1067 cd /usr/src/freeswitch/src/mod/endpoints/mod_gsmopen/ 1068 make clean 1069 make install Just to repeat the compilation error Creating mod_gsmopen.so... /usr/bin/ld: cannot find -lctb-0.16 collect2: ld returned 1 exit status g++ -I../../../../libs/spandsp/src -I../../../..//libs/tiff-3.8.2/libtiff -DGSMOPEN_C_VER=\"44fd0de\" -DMODGSMOPEN_C_VER=\"44fd0de\" -I/usr/src/freeswitch/libs/curl/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -shared -o .libs/mod_gsmopen.so -shared -Wl,-x .libs/mod_gsmopen.o gsmopen_protocol.o /usr/src/freeswitch/.libs/libfreeswitch.so -L/usr/src/freeswitch/libs/apr-util/xml/expat/lib /usr/src/freeswitch/libs/apr-util/xml/expat/lib/.libs/libexpat.a /usr/src/freeswitch/libs/apr/.libs/libapr-1.a -luuid -lpthread -L/usr/src/freeswitch/libs/srtp -L/usr/kerberos/lib -lcrypt -lrt -lssl -lcrypto -ldl -lncurses -L/usr/src/freeswitch/libs/spandsp/src /usr/src/freeswitch/libs/spandsp/src/.libs/libspandsp.a -L/usr/src/freeswitch/libs/tiff-3.8.2/libtiff /usr/src/freeswitch/libs/tiff-3.8.2/libtiff/.libs/libtiff.a -ljpeg -lz -lm -lc -lctb-0.16 -lgsmme -Wl,--rpath -Wl,/usr/local/freeswitch/lib -Wl,--rpath -Wl,/usr/local/freeswitch/mod make[1]: *** [mod_gsmopen.so] Error 1 make: *** [install] Error 1 All the gsmlib stuff that is NOT in the freeswitch src dir is here [root at space mod_gsmopen]# locate gsmlib | grep -v src /usr/local/include/gsmlib /usr/local/include/gsmlib/gsm_at.h /usr/local/include/gsmlib/gsm_cb.h /usr/local/include/gsmlib/gsm_error.h /usr/local/include/gsmlib/gsm_event.h /usr/local/include/gsmlib/gsm_map_key.h /usr/local/include/gsmlib/gsm_me_ta.h /usr/local/include/gsmlib/gsm_parser.h /usr/local/include/gsmlib/gsm_phonebook.h /usr/local/include/gsmlib/gsm_port.h /usr/local/include/gsmlib/gsm_sie_me.h /usr/local/include/gsmlib/gsm_sms.h /usr/local/include/gsmlib/gsm_sms_codec.h /usr/local/include/gsmlib/gsm_sms_store.h /usr/local/include/gsmlib/gsm_sorted_phonebook.h /usr/local/include/gsmlib/gsm_sorted_phonebook_base.h /usr/local/include/gsmlib/gsm_sorted_sms_store.h /usr/local/include/gsmlib/gsm_unix_serial.h /usr/local/include/gsmlib/gsm_util.h /usr/local/share/locale/de/LC_MESSAGES/gsmlib.mo And the ctb stuff is in /usr/local/lib /usr/local/include/ctb-0.16 /usr/local/include/ctb-0.16/ctb.h /usr/local/include/ctb-0.16/fifo.h /usr/local/include/ctb-0.16/getopt.h /usr/local/include/ctb-0.16/iobase.h /usr/local/include/ctb-0.16/linux /usr/local/include/ctb-0.16/portscan.h /usr/local/include/ctb-0.16/serport.h /usr/local/include/ctb-0.16/serportx.h /usr/local/include/ctb-0.16/timer.h /usr/local/include/ctb-0.16/linux/serport.h /usr/local/include/ctb-0.16/linux/timer.h /usr/local/lib/libctbd-0.16.a /usr/local/lib/libctbd-0.16.so gcc version 4.1.2 20080704 (Red Hat 4.1.2-52) GNU Make 3.81 CentOS release 5.8 (Final) On 14/05/2012, at 9:38 PM, Giovanni Maruzzelli wrote: > you must first compile and install libctb, as per the wiki page ( > http://wiki.freeswitch.org/wiki/gsmopen ) > then, after installation of libctb and gsmlib (as per wiki), be sure > to update your dinamic link cache, or compiler will not find then. > > Eg: ldconfig > > > On 5/14/12, Chris Mylonas wrote: >> All good GM, no inconvenience, just a minor gotcha ;) >> I am unable to compile mod_gsmopen though. >> >> It complains about not being able to find ctb-0.16 >> The actual filename is libctbd-0.16.so in /usr/local/lib as you can see from >> the 2nd lot of stuff. >> >> How do I fix this? >> >> >> [root at space mod_gsmopen]# make install >> Creating mod_gsmopen.so... >> /usr/bin/ld: cannot find -lctb-0.16 >> collect2: ld returned 1 exit status >> g++ -I../../../../libs/spandsp/src -I../../../..//libs/tiff-3.8.2/libtiff >> -DGSMOPEN_C_VER=\"44fd0de\" -DMODGSMOPEN_C_VER=\"44fd0de\" >> -I/usr/src/freeswitch/libs/curl/include -I/usr/src/freeswitch/src/include >> -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/libs/libteletone/src >> -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g >> -O2 -D_GNU_SOURCE -shared -o .libs/mod_gsmopen.so -shared -Wl,-x >> .libs/mod_gsmopen.o gsmopen_protocol.o >> /usr/src/freeswitch/.libs/libfreeswitch.so >> -L/usr/src/freeswitch/libs/apr-util/xml/expat/lib >> /usr/src/freeswitch/libs/apr-util/xml/expat/lib/.libs/libexpat.a >> /usr/src/freeswitch/libs/apr/.libs/libapr-1.a -luuid -lpthread >> -L/usr/src/freeswitch/libs/srtp -L/usr/kerberos/lib -lcrypt -lrt -lssl >> -lcrypto -ldl -lncurses -L/usr/src/freeswitch/libs/spandsp/src >> /usr/src/freeswitch/libs/spandsp/src/.libs/libspandsp.a >> -L/usr/src/freeswitch/libs/tiff-3.8.2/libtiff >> /usr/src/freeswitch/libs/tiff-3.8.2/libtiff/.libs/libtiff.a -ljpeg -lz -lm >> -lc -lctb-0.16 -lgsmme -Wl,--rpath -Wl,/usr/local/freeswitch/lib >> -Wl,--rpath -Wl,/usr/local/freeswitch/mod >> make[1]: *** [mod_gsmopen.so] Error 1 >> make: *** [install] Error 1 >> >> >> >> [root at space mod_gsmopen]# ldd /usr/local/lib/libctbd-0.16.so >> linux-gate.so.1 => (0x00754000) >> libpthread.so.0 => /lib/libpthread.so.0 (0x00e83000) >> libstdc++.so.6 => /usr/lib/libstdc++.so.6 (0x00d28000) >> libm.so.6 => /lib/libm.so.6 (0x00964000) >> libgcc_s.so.1 => /lib/libgcc_s.so.1 (0x00e1f000) >> libc.so.6 => /lib/libc.so.6 (0x00110000) >> /lib/ld-linux.so.2 (0x003ea000) >> >> >> >> On 14/05/2012, at 6:44 PM, Giovanni Maruzzelli wrote: >> >>> yes, it requires libspandsp, maybe the Makefile it's not yet tweaked >>> to build the library automatically. >>> >>> So, please first build mod_spandsp, then mod_gsmopen. >>> >>> We'll fixx the Makefile soon, sorry for the inconvenience. >>> >>> -giovanni >>> >>> On Mon, May 14, 2012 at 9:30 AM, Chris Mylonas >>> wrote: >>>> Hi FS List, >>>> >>>> FYI - as a shortcut to building my freeswitch, I skip spandsp - but it >>>> looks >>>> like this mod_gsmopen wants it in there. >>>> >>>> >>>> [root at space build]# cd >>>> /usr/src/freeswitch/src/mod/endpoints/mod_gsmopen/ >>>> [root at space mod_gsmopen]# make clean >>>> [root at space mod_gsmopen]# make install >>>> Compiling gsmopen_protocol.cpp... >>>> Compiling >>>> /usr/src/freeswitch/src/mod/endpoints/mod_gsmopen/mod_gsmopen.cpp... >>>> mkdir .libs >>>> Compiling >>>> /usr/src/freeswitch/src/mod/endpoints/mod_gsmopen/mod_gsmopen.cpp >>>> ... >>>> Creating mod_gsmopen.so... >>>> /usr/bin/ld: cannot find -lspandsp >>>> collect2: ld returned 1 exit status >>>> g++ -I../../../../libs/spandsp/src >>>> -I../../../..//libs/tiff-3.8.2/libtiff >>>> -DGSMOPEN_C_VER=\"44fd0de\" -DMODGSMOPEN_C_VER=\"44fd0de\" >>>> -I/usr/src/freeswitch/libs/curl/include >>>> -I/usr/src/freeswitch/src/include >>>> -I/usr/src/freeswitch/src/include >>>> -I/usr/src/freeswitch/libs/libteletone/src >>>> -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 >>>> -g >>>> -O2 -D_GNU_SOURCE -shared -o .libs/mod_gsmopen.so -shared -Wl,-x >>>> .libs/mod_gsmopen.o gsmopen_protocol.o -lm >>>> /usr/src/freeswitch/.libs/libfreeswitch.so >>>> -L/usr/src/freeswitch/libs/apr-util/xml/expat/lib >>>> /usr/src/freeswitch/libs/apr-util/xml/expat/lib/.libs/libexpat.a >>>> /usr/src/freeswitch/libs/apr/.libs/libapr-1.a -luuid -lpthread >>>> -L/usr/src/freeswitch/libs/srtp -L/usr/kerberos/lib -lcrypt -lrt -lssl >>>> -lcrypto -ldl -lz -lncurses -ljpeg >>>> -L/usr/src/freeswitch/libs/spandsp/src >>>> -lspandsp -lctb-0.16 -lgsmme -Wl,--rpath -Wl,/usr/local/freeswitch/lib >>>> -Wl,--rpath -Wl,/usr/local/freeswitch/mod >>>> make[1]: *** [mod_gsmopen.so] Error 1 >>>> make: *** [install] Error 1 >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> >>> -- >>> Sincerely, >>> >>> Giovanni Maruzzelli >>> Cell : +39-347-2665618 >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> > > > -- > Sincerely, > > Giovanni Maruzzelli > Cell : +39-347-2665618 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120514/e59ace8b/attachment-0001.html From michal.zubac at comgate.cz Mon May 14 16:57:09 2012 From: michal.zubac at comgate.cz (=?UTF-8?B?TWljaGFsIFp1YsOhxI0=?=) Date: Mon, 14 May 2012 14:57:09 +0200 Subject: [Freeswitch-users] disable SIP privacy In-Reply-To: <20120511150718.GA10459@eagle.cupis.co.uk> References: <4FAD22D0.5020009@comgate.cz> <20120511150718.GA10459@eagle.cupis.co.uk> Message-ID: <4FB10125.2050200@comgate.cz> thanks for answer, but app privacy=none didn't help. I figured out, that setting variable "origination_privacy=screen" will do the trick. :-) M. On 11.5.2012 17:07, Paul Cupis wrote: > On Fri, May 11, 2012 at 04:31:44PM +0200, Michal Zub???? wrote: >> I want to disable Privacy on outgoing SIP leg but I can't do it because >> incoming SIP leg has Privacy enabled. I want to set outgoing caller_id >> to some constant number for these cases. >> Can this even be done? I tried sip_Privacy variables, but with no effect. > Try something like this: > > > > > > Regards, > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From saami_mh at ymail.com Mon May 14 18:01:01 2012 From: saami_mh at ymail.com (Samira Mh) Date: Mon, 14 May 2012 07:01:01 -0700 (PDT) Subject: [Freeswitch-users] configuring gateway for making calls In-Reply-To: <009101cd31bd$dcc9e010$965da030$@speedingtrade.com> References: <1336392911.49162.YahooMailNeo@web120104.mail.ne1.yahoo.com> <1336986654.82652.YahooMailNeo@web120106.mail.ne1.yahoo.com> <005c01cd31b4$5bae8250$130b86f0$@speedingtrade.com> <1336991181.54394.YahooMailNeo@web120104.mail.ne1.yahoo.com> <009101cd31bd$dcc9e010$965da030$@speedingtrade.com> Message-ID: <1337004061.27952.YahooMailNeo@web120105.mail.ne1.yahoo.com> hi, thanks so much for great help but while dialing my umber using extensino:1000 that register on myfreeswitch server the error is as follow: 2012-05-13 20:23:55.098050 [NOTICE] switch_channel.c:669 New Channel sofia/internal/1000 at 192.168.1.100 [5c057400-2be8-4080-a401-25b5dc402dd0] 2012-05-13 20:23:55.106289 [INFO] mod_dialplan_xml.c:418 Processing 1000->00989191949637 in context default 2012-05-13 20:23:55.111170 [NOTICE] switch_ivr.c:1447 Transfer sofia/internal/1000 at 192.168.1.100 to enum[00989191949637 at default] 2012-05-13 20:23:55.248259 [INFO] switch_core_state_machine.c:142 No Route, Aborting 2012-05-13 20:23:55.248259 [NOTICE] switch_core_state_machine.c:143 Hangup sofia/internal/1000 at 192.168.1.100 [CS_ROUTING] [NO_ROUTE_DESTINATION] 2012-05-13 20:23:55.257505 [NOTICE] switch_core_session.c:1182 Session 59 (sofia/internal/1000 at 192.168.1.100) Ended 2012-05-13 20:23:55.257505 [NOTICE] switch_core_session.c:1184 Close Channel sofia/internal/1000 at 192.168.1.100 [CS_DESTROY] 1: --------------------------------------------------------- cat ? /usr/local/freeswitch/conf/dialplan/default/01_shatel.xml? 2:cat ? /usr/local/freeswitch/conf/directory/default/1000.xml -------------------------------------------------- ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ________________________________ From: Erkan ?nl? (simalube) To: 'Samira Mh' Sent: Monday, May 14, 2012 3:09 PM Subject: AW: [Freeswitch-users] configuring gateway for making calls Hi Samira, ? you need 3 xml files. ? 1.?????? Xml file is located at the directory/default ? Here put this xml sample and try after reloadxml to register your softphone ? ? ??? ????? ????? ??? ??? ????? ????? ?????? ????? ????? ????? ????? ????? ??? ? ? ? 2.?????? Build your dialingplan The dialingplan is located in the dialingplan folder. Create here a new xml? and put this sample inside the file. In the future you can study the default.xml in dialingplan folder to learn more about the dialingplan. In this section you see that the user_context in the directory folder XML file is the same with context_name So the user account is in association with the dialingplan. If you need you can build more different dialingplans for other requirements. ? ? ?? ???? ??????? ??????? ???? ?? ? ? 3.?????? The last step is the gateway Your gateway looks good and is reged correctly ? You see the gateway name you written ?custom? You can change it to gateway1 or myfavoriteGW or what you want. ? But the name of the gatewayname must be the same like this line in section 2 of my description ? ? Change the red gateway? to the right gatewayname in your sip profiles external folder located XML ? You see in my sample ? "sofia/gateway/gateway/00$1|sofia/gateway/gateway2/00$1? ? I have 2 gateways for failover if one gateway is down or gives errors. |? ?these character is an OR? and mean if the first gateway gives an error so try the second gateway ? You can use also only 1 gateway so delete the second gateway until to ?? character ? Hope that is now clear for you. ? Kind regards Erkan ? ? Von:Samira Mh [mailto:saami_mh at ymail.com] Gesendet: Montag, 14. Mai 2012 12:26 An: FreeSWITCH Users Help Cc: erkan at speedingtrade.com Betreff: Re: [Freeswitch-users] configuring gateway for making calls ? hi erkan; thanks so much for your reply; ? i think your mean is: ?i must configure file??01_custom.xml like?: ? ? ? and because i registered via extension:1000 on freeswitch so : ?must configure file 1000.xml ?like ?: ? ? ? ??? ????? ????? ??? ??? ????? ????? ?????? ????? ????? ????? ????? ????? ??? ? ? but i on't know what to be set as?? ? in the 1000.xml? please help, i am new in freeswitch? sorry for my english ;; ? ? ________________________________ From:Erkan ?nl? (simalube) To: 'FreeSWITCH Users Help' Sent: Monday, May 14, 2012 2:01 PM Subject: Re: [Freeswitch-users] configuring gateway for making calls ? Hi Samira, ? please build first of all them in the directory/default folder a XML file like this. ? ? ??? ????? ????? ??? ??? ????? ????? ????? ????? ????? ????? ????? ????? ??? ? ? Now you can register with your softphone to FS. See the user_context area in this section you must link it to your dialing plan The dialing plan is located in dialing plan folder. ? My simple dialing plan looks like this. ? ? ?? ???? ??????? ??????? ???? ?? ? ? The context name of the dialing plan must be the same in the directory XML user_context I color it in red. ? I hope that this info are usefully for you. ? Kind regards. ? Von:freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Samira Mh Gesendet: Montag, 14. Mai 2012 11:11 An: Free SWITCH Users Help Betreff: [Freeswitch-users] configuring gateway for making calls ? ? hi guys; i have configured gateway for making calls but it dosn't work; my config files are as follow: ? first create file that name " iptel.org.xml?" in the below path: ?/usr/local/freeswitch/conf/sip_profiles/external/ i have an account on?iptel.org like :(username:aimas;password:mypass) ?? --> ? ? ? ? ? ? ? the run : sofia profile external restart reloadxml reloxm and sofia status: ?Name ? ? ? ? ?Type ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? Data ? ? ?State ================================================================================================= ? ? ? ? ? ? ? ? ?external ? ? ? profile ? ? ? ? ? ? ? sip:mod_sofia at 10.0.3.15:5080 ? ? ?RUNNING (0) ? ? external::example.com ? ? ? gateway ? ? ? ? ? ? ? ? ? ?sip:joeuser at example.com ? ? ?NOREG ? ? ? ? ?external::custom ? ? ? gateway ? ? ? ? ? ? ? ? ? sip:arimas at sip.iptel.org ? ? ?REGED ? ? ? ? ? ? ? ? ?internal ? ? ? profile ? ? ? ? ? ? ? sip:mod_sofia at 10.0.3.15:5060 ? ? ?RUNNING (0) ? ? ? ? ? ? ? ? 10.0.3.15 ? ? ? ? alias ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? internal ? ? ?ALIASED ================================================================================================= then create file name "01_custom.xml " in the below path : /usr/local/freeswitch/conf/dialplan/default the content of the files ?01_custom.xml ? are: . now when i dial?9, 1-800-555-1212 on eyebeam that register on freeswitch server the call is failed? ? please help; thanks? _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120514/657930a0/attachment-0001.html From tculjaga at gmail.com Mon May 14 18:17:39 2012 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Mon, 14 May 2012 16:17:39 +0200 Subject: [Freeswitch-users] joining a bridged call to a conference In-Reply-To: References: Message-ID: yup, i was missing -both :=) next question if i may... say i do it all via event socket... originate user/1002 &conference(myConfRoom) how can i transfer an established call ( A-Leg + B-Leg) to that conference ? uuid_transfer uuid -both WHERE?? On Mon, May 14, 2012 at 7:40 AM, Peter Olsson < peter.olsson at visionutveckling.se> wrote: > I think "uuid_transfer -both" should work. If it's not working, > there might be some kind of media negotiation issue (for instance if bypass > media is used). > > /Peter > > 14 maj 2012 kl. 01:01 skrev "Tihomir Culjaga" : > > > hello, > > > > i got an established call A calling B. A is a number outside FS (any > number e.g. 16084191500), B is an extension registered to FS (e.g. 1002). > > > > after some time, i would like to join the call A => B to a conference > roon e.g. 3001 (of course using ESL) > > > > any idea how to do it smooth ? > > whats the best practice for this ? > > > > i tried uuid_transfer but whatever i do it keeps killing one leg... > > > > > > thanks for help, > > T. > > > > > > > > !DSPAM:4fb03a3f32761890320954! > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > !DSPAM:4fb03a3f32761890320954! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120514/b3a5a3c1/attachment.html From saami_mh at ymail.com Mon May 14 18:36:21 2012 From: saami_mh at ymail.com (Samira Mh) Date: Mon, 14 May 2012 07:36:21 -0700 (PDT) Subject: [Freeswitch-users] Fw: AW: configuring gateway for making calls In-Reply-To: <000301cd31dc$f938d570$ebaa8050$@speedingtrade.com> References: <1336392911.49162.YahooMailNeo@web120104.mail.ne1.yahoo.com> <1336986654.82652.YahooMailNeo@web120106.mail.ne1.yahoo.com> <005c01cd31b4$5bae8250$130b86f0$@speedingtrade.com> <1336991181.54394.YahooMailNeo@web120104.mail.ne1.yahoo.com> <009101cd31bd$dcc9e010$965da030$@speedingtrade.com> <1337004061.27952.YahooMailNeo@web120105.mail.ne1.yahoo.com> <000301cd31dc$f938d570$ebaa8050$@speedingtrade.com> Message-ID: <1337006181.98550.YahooMailNeo@web120102.mail.ne1.yahoo.com> dear erkan, i want to dial my mobile number (9191949637) via freeswitch myfreeswitch server is installed on vm ware with private ip-address(192.168.1.100) and the extension:1000 is register to freeswitch server now i want to call my phone number in order to ?my mobile phone will ringing; .such example in padf book :page 122 of freeswtch-1.0.6 ebook do you thins that needs worldwide call or internal call? for interal call need to set gateway ? ----- Forwarded Message ----- From: Erkan ?nl? (simalube) To: 'Samira Mh' Sent: Monday, May 14, 2012 6:52 PM Subject: RE: AW: [Freeswitch-users] configuring gateway for making calls Hi again, ? yes of course that is right. The dialingplan that I give it to you is only for worldwide calls. ? You need a second dialing plan section in the XML ? Look here all sections can be put in the dialing plan xml. ? Internalcall means (see destination number is 3) that means all 3 long number can be dialed. You can change it to 4? or 5 ????? ??????? ????? ??? ?????????????? ? Here a small sample for echo. Use GSM codec or G711a law codec to call. So you can hear you self. ??? ????? ??????? ??????? ????? ??? ? These sample gives you a beeeep tone. ??? ????? ??????? ??????? ????? ??? ? From:Samira Mh [mailto:saami_mh at ymail.com] Sent: Monday, May 14, 2012 4:01 PM To: Erkan ?nl? (simalube) Cc: Free SWITCH Users Help Subject: Re: AW: [Freeswitch-users] configuring gateway for making calls ? hi, thanks so much for great help but while dialing my umber using extensino:1000 that register on myfreeswitch server the error is as follow: ? 2012-05-13 20:23:55.098050 [NOTICE] switch_channel.c:669 New Channel sofia/internal/1000 at 192.168.1.100[5c057400-2be8-4080-a401-25b5dc402dd0] 2012-05-13 20:23:55.106289 [INFO] mod_dialplan_xml.c:418 Processing 1000->00989191949637 in context default 2012-05-13 20:23:55.111170 [NOTICE] switch_ivr.c:1447 Transfer sofia/internal/1000 at 192.168.1.100to enum[00989191949637 at default] 2012-05-13 20:23:55.248259 [INFO] switch_core_state_machine.c:142 No Route, Aborting 2012-05-13 20:23:55.248259 [NOTICE] switch_core_state_machine.c:143 Hangup sofia/internal/1000 at 192.168.1.100[CS_ROUTING] [NO_ROUTE_DESTINATION] 2012-05-13 20:23:55.257505 [NOTICE] switch_core_session.c:1182 Session 59 (sofia/internal/1000 at 192.168.1.100) Ended 2012-05-13 20:23:55.257505 [NOTICE] switch_core_session.c:1184 Close Channel sofia/internal/1000 at 192.168.1.100[CS_DESTROY] ? ? 1: --------------------------------------------------------- ? cat ? /usr/local/freeswitch/conf/dialplan/default/01_shatel.xml? ? ? 2:cat ? /usr/local/freeswitch/conf/directory/default/1000.xml -------------------------------------------------- ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ________________________________ From:Erkan ?nl? (simalube) To: 'Samira Mh' Sent: Monday, May 14, 2012 3:09 PM Subject: AW: [Freeswitch-users] configuring gateway for making calls ? Hi Samira, ? you need 3 xml files. ? 1.?????? Xml file is located at the directory/default ? Here put this xml sample and try after reloadxml to register your softphone ? ? ??? ????? ????? ??? ??? ????? ????? ?????? ????? ????? ????? ????? ????? ??? ? ? ? 2.?????? Build your dialingplan The dialingplan is located in the dialingplan folder. Create here a new xml? and put this sample inside the file. In the future you can study the default.xml in dialingplan folder to learn more about the dialingplan. In this section you see that the user_context in the directory folder XML file is the same with context_name So the user account is in association with the dialingplan. If you need you can build more different dialingplans for other requirements. ? ? ?? ???? ??????? ??????? ???? ?? ? ? 3.?????? The last step is the gateway Your gateway looks good and is reged correctly ? You see the gateway name you written ?custom? You can change it to gateway1 or myfavoriteGW or what you want. ? But the name of the gatewayname must be the same like this line in section 2 of my description ? ? Change the red gateway? to the right gatewayname in your sip profiles external folder located XML ? You see in my sample ? "sofia/gateway/gateway/00$1|sofia/gateway/gateway2/00$1? ? I have 2 gateways for failover if one gateway is down or gives errors. |? ?these character is an OR? and mean if the first gateway gives an error so try the second gateway ? You can use also only 1 gateway so delete the second gateway until to ?? character ? Hope that is now clear for you. ? Kind regards Erkan ? ? Von:Samira Mh [mailto:saami_mh at ymail.com] Gesendet: Montag, 14. Mai 2012 12:26 An: FreeSWITCH Users Help Cc: erkan at speedingtrade.com Betreff: Re: [Freeswitch-users] configuring gateway for making calls ? hi erkan; thanks so much for your reply; ? i think your mean is: ?i must configure file??01_custom.xml like?: ? ? ? and because i registered via extension:1000 on freeswitch so : ?must configure file 1000.xml ?like ?: ? ? ? ??? ????? ????? ??? ??? ????? ????? ?????? ????? ????? ????? ????? ????? ??? ? ? but i on't know what to be set as?? ? in the 1000.xml? please help, i am new in freeswitch? sorry for my english ;; ? ? ________________________________ From:Erkan ?nl? (simalube) To: 'FreeSWITCH Users Help' Sent: Monday, May 14, 2012 2:01 PM Subject: Re: [Freeswitch-users] configuring gateway for making calls ? Hi Samira, ? please build first of all them in the directory/default folder a XML file like this. ? ? ??? ????? ????? ??? ??? ????? ????? ????? ????? ????? ????? ????? ????? ??? ? ? Now you can register with your softphone to FS. See the user_context area in this section you must link it to your dialing plan The dialing plan is located in dialing plan folder. ? My simple dialing plan looks like this. ? ? ?? ???? ??????? ??????? ???? ?? ? ? The context name of the dialing plan must be the same in the directory XML user_context I color it in red. ? I hope that this info are usefully for you. ? Kind regards. ? Von:freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Samira Mh Gesendet: Montag, 14. Mai 2012 11:11 An: Free SWITCH Users Help Betreff: [Freeswitch-users] configuring gateway for making calls ? ? hi guys; i have configured gateway for making calls but it dosn't work; my config files are as follow: ? first create file that name " iptel.org.xml?" in the below path: ?/usr/local/freeswitch/conf/sip_profiles/external/ i have an account on?iptel.org like :(username:aimas;password:mypass) ?? --> ? ? ? ? ? ? ? the run : sofia profile external restart reloadxml reloxm and sofia status: ?Name ? ? ? ? ?Type ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? Data ? ? ?State ================================================================================================= ? ? ? ? ? ? ? ? ?external ? ? ? profile ? ? ? ? ? ? ? sip:mod_sofia at 10.0.3.15:5080? ? ?RUNNING (0) ? ? external::example.com ? ? ? gateway ? ? ? ? ? ? ? ? ? ?sip:joeuser at example.com? ? ?NOREG ? ? ? ? ?external::custom ? ? ? gateway ? ? ? ? ? ? ? ? ? sip:arimas at sip.iptel.org? ? ?REGED ? ? ? ? ? ? ? ? ?internal ? ? ? profile ? ? ? ? ? ? ? sip:mod_sofia at 10.0.3.15:5060? ? ?RUNNING (0) ? ? ? ? ? ? ? ? 10.0.3.15 ? ? ? ? alias ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? internal ? ? ?ALIASED ================================================================================================= then create file name "01_custom.xml " in the below path : /usr/local/freeswitch/conf/dialplan/default the content of the files ?01_custom.xml ? are: . now when i dial?9, 1-800-555-1212 on eyebeam that register on freeswitch server the call is failed? ? please help; thanks? _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120514/1b2aeb9d/attachment-0001.html From bdfoster at endigotech.com Mon May 14 18:51:58 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Mon, 14 May 2012 10:51:58 -0400 Subject: [Freeswitch-users] mod_gsmopen requires spandsp In-Reply-To: <62D399FB-DFA0-4DD0-B86F-45E95026EF82@opencsta.org> References: <1DE50890-EE01-4EEE-83A8-5B5F71BB3959@opencsta.org> <3484676B-F4CA-47E3-A8B0-9743291C1056@opencsta.org> <62D399FB-DFA0-4DD0-B86F-45E95026EF82@opencsta.org> Message-ID: if you compile libctb with debug it's actually libctbd, not libctb. On Mon, May 14, 2012 at 7:53 AM, Chris Mylonas wrote: > Thanks for the reply Giovanni. I have done the steps on the wiki. > > tl;dr; should i put a prefix when i'm making the dependent mods so they > don't go into /usr/local/lib, or this is why ldconfig is run - to tell the > system where the libs are. > > All the relevant stuff is below > > Hope you can see something wrong, > Cheers > Chris > > e.g. > here is my bash history > > 1050 cd freeswitch/ > 1051 ls > 1052 find . -name gsmlib > 1053 cd src/mod/endpoints/mod_gsmopen/ > 1054 ls > 1055 cd gsmlib/ > 1056 ls > 1057 cd gsmlib-1.10-patched-13ubuntu/ > 1058 ls > * 1059 ./configure* > * 1060 make* > * 1061 make install* > * 1062 ldconfig * > * 1063 cd > /usr/src/freeswitch/src/mod/endpoints/mod_gsmopen/libctb-0.16/build* > * 1064 make DEBUG=1 GPIB=0* > * 1065 make DEBUG=1 GPIB=0 install* > * 1066 ldconfig* > 1067 cd /usr/src/freeswitch/src/mod/endpoints/mod_gsmopen/ > 1068 make clean > 1069 make install > > Just to repeat the compilation error > Creating mod_gsmopen.so... > */usr/bin/ld: cannot find -lctb-0.16* > collect2: ld returned 1 exit status > g++ -I../../../../libs/spandsp/src -I../../../..//libs/tiff-3.8.2/libtiff > -DGSMOPEN_C_VER=\"44fd0de\" -DMODGSMOPEN_C_VER=\"44fd0de\" > -I/usr/src/freeswitch/libs/curl/include -I/usr/src/freeswitch/src/include > -I/usr/src/freeswitch/src/include > -I/usr/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden > -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -shared > -o .libs/mod_gsmopen.so -shared -Wl,-x .libs/mod_gsmopen.o > gsmopen_protocol.o /usr/src/freeswitch/.libs/libfreeswitch.so > -L/usr/src/freeswitch/libs/apr-util/xml/expat/lib > /usr/src/freeswitch/libs/apr-util/xml/expat/lib/.libs/libexpat.a > /usr/src/freeswitch/libs/apr/.libs/libapr-1.a -luuid -lpthread > -L/usr/src/freeswitch/libs/srtp -L/usr/kerberos/lib -lcrypt -lrt -lssl > -lcrypto -ldl -lncurses -L/usr/src/freeswitch/libs/spandsp/src > /usr/src/freeswitch/libs/spandsp/src/.libs/libspandsp.a > -L/usr/src/freeswitch/libs/tiff-3.8.2/libtiff > /usr/src/freeswitch/libs/tiff-3.8.2/libtiff/.libs/libtiff.a -ljpeg -lz -lm > -lc -lctb-0.16 -lgsmme -Wl,--rpath -Wl,/usr/local/freeswitch/lib > -Wl,--rpath -Wl,/usr/local/freeswitch/mod > make[1]: *** [mod_gsmopen.so] Error 1 > make: *** [install] Error 1 > > > > > All the gsmlib stuff that is NOT in the freeswitch src dir is here > > [root at space mod_gsmopen]# locate gsmlib | grep -v src > /usr/local/include/gsmlib > /usr/local/include/gsmlib/gsm_at.h > /usr/local/include/gsmlib/gsm_cb.h > /usr/local/include/gsmlib/gsm_error.h > /usr/local/include/gsmlib/gsm_event.h > /usr/local/include/gsmlib/gsm_map_key.h > /usr/local/include/gsmlib/gsm_me_ta.h > /usr/local/include/gsmlib/gsm_parser.h > /usr/local/include/gsmlib/gsm_phonebook.h > /usr/local/include/gsmlib/gsm_port.h > /usr/local/include/gsmlib/gsm_sie_me.h > /usr/local/include/gsmlib/gsm_sms.h > /usr/local/include/gsmlib/gsm_sms_codec.h > /usr/local/include/gsmlib/gsm_sms_store.h > /usr/local/include/gsmlib/gsm_sorted_phonebook.h > /usr/local/include/gsmlib/gsm_sorted_phonebook_base.h > /usr/local/include/gsmlib/gsm_sorted_sms_store.h > /usr/local/include/gsmlib/gsm_unix_serial.h > /usr/local/include/gsmlib/gsm_util.h > /usr/local/share/locale/de/LC_MESSAGES/gsmlib.mo > > > And the ctb stuff is in /usr/local/lib > > /usr/local/include/ctb-0.16 > /usr/local/include/ctb-0.16/ctb.h > /usr/local/include/ctb-0.16/fifo.h > /usr/local/include/ctb-0.16/getopt.h > /usr/local/include/ctb-0.16/iobase.h > /usr/local/include/ctb-0.16/linux > /usr/local/include/ctb-0.16/portscan.h > /usr/local/include/ctb-0.16/serport.h > /usr/local/include/ctb-0.16/serportx.h > /usr/local/include/ctb-0.16/timer.h > /usr/local/include/ctb-0.16/linux/serport.h > /usr/local/include/ctb-0.16/linux/timer.h > /usr/local/lib/libctbd-0.16.a > /usr/local/lib/libctbd-0.16.so > > > gcc version 4.1.2 20080704 (Red Hat 4.1.2-52) > GNU Make 3.81 > CentOS release 5.8 (Final) > > > > > On 14/05/2012, at 9:38 PM, Giovanni Maruzzelli wrote: > > you must first compile and install libctb, as per the wiki page ( > http://wiki.freeswitch.org/wiki/gsmopen ) > then, after installation of libctb and gsmlib (as per wiki), be sure > to update your dinamic link cache, or compiler will not find then. > > Eg: ldconfig > > > On 5/14/12, Chris Mylonas wrote: > > All good GM, no inconvenience, just a minor gotcha ;) > > I am unable to compile mod_gsmopen though. > > > It complains about not being able to find ctb-0.16 > > The actual filename is libctbd-0.16.so in /usr/local/lib as you can see > from > > the 2nd lot of stuff. > > > How do I fix this? > > > > [root at space mod_gsmopen]# make install > > Creating mod_gsmopen.so... > > /usr/bin/ld: cannot find -lctb-0.16 > > collect2: ld returned 1 exit status > > g++ -I../../../../libs/spandsp/src -I../../../..//libs/tiff-3.8.2/libtiff > > -DGSMOPEN_C_VER=\"44fd0de\" -DMODGSMOPEN_C_VER=\"44fd0de\" > > -I/usr/src/freeswitch/libs/curl/include -I/usr/src/freeswitch/src/include > > -I/usr/src/freeswitch/src/include > -I/usr/src/freeswitch/libs/libteletone/src > > -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g > > -O2 -D_GNU_SOURCE -shared -o .libs/mod_gsmopen.so -shared -Wl,-x > > .libs/mod_gsmopen.o gsmopen_protocol.o > > /usr/src/freeswitch/.libs/libfreeswitch.so > > -L/usr/src/freeswitch/libs/apr-util/xml/expat/lib > > /usr/src/freeswitch/libs/apr-util/xml/expat/lib/.libs/libexpat.a > > /usr/src/freeswitch/libs/apr/.libs/libapr-1.a -luuid -lpthread > > -L/usr/src/freeswitch/libs/srtp -L/usr/kerberos/lib -lcrypt -lrt -lssl > > -lcrypto -ldl -lncurses -L/usr/src/freeswitch/libs/spandsp/src > > /usr/src/freeswitch/libs/spandsp/src/.libs/libspandsp.a > > -L/usr/src/freeswitch/libs/tiff-3.8.2/libtiff > > /usr/src/freeswitch/libs/tiff-3.8.2/libtiff/.libs/libtiff.a -ljpeg -lz -lm > > -lc -lctb-0.16 -lgsmme -Wl,--rpath -Wl,/usr/local/freeswitch/lib > > -Wl,--rpath -Wl,/usr/local/freeswitch/mod > > make[1]: *** [mod_gsmopen.so] Error 1 > > make: *** [install] Error 1 > > > > > [root at space mod_gsmopen]# ldd /usr/local/lib/libctbd-0.16.so > > linux-gate.so.1 => (0x00754000) > > libpthread.so.0 => /lib/libpthread.so.0 (0x00e83000) > > libstdc++.so.6 => /usr/lib/libstdc++.so.6 (0x00d28000) > > libm.so.6 => /lib/libm.so.6 (0x00964000) > > libgcc_s.so.1 => /lib/libgcc_s.so.1 (0x00e1f000) > > libc.so.6 => /lib/libc.so.6 (0x00110000) > > /lib/ld-linux.so.2 (0x003ea000) > > > > > On 14/05/2012, at 6:44 PM, Giovanni Maruzzelli wrote: > > > yes, it requires libspandsp, maybe the Makefile it's not yet tweaked > > to build the library automatically. > > > So, please first build mod_spandsp, then mod_gsmopen. > > > We'll fixx the Makefile soon, sorry for the inconvenience. > > > -giovanni > > > On Mon, May 14, 2012 at 9:30 AM, Chris Mylonas > > wrote: > > Hi FS List, > > > FYI - as a shortcut to building my freeswitch, I skip spandsp - but it > > looks > > like this mod_gsmopen wants it in there. > > > > [root at space build]# cd > > /usr/src/freeswitch/src/mod/endpoints/mod_gsmopen/ > > [root at space mod_gsmopen]# make clean > > [root at space mod_gsmopen]# make install > > Compiling gsmopen_protocol.cpp... > > Compiling > > /usr/src/freeswitch/src/mod/endpoints/mod_gsmopen/mod_gsmopen.cpp... > > mkdir .libs > > Compiling > > /usr/src/freeswitch/src/mod/endpoints/mod_gsmopen/mod_gsmopen.cpp > > ... > > Creating mod_gsmopen.so... > > /usr/bin/ld: cannot find -lspandsp > > collect2: ld returned 1 exit status > > g++ -I../../../../libs/spandsp/src > > -I../../../..//libs/tiff-3.8.2/libtiff > > -DGSMOPEN_C_VER=\"44fd0de\" -DMODGSMOPEN_C_VER=\"44fd0de\" > > -I/usr/src/freeswitch/libs/curl/include > > -I/usr/src/freeswitch/src/include > > -I/usr/src/freeswitch/src/include > > -I/usr/src/freeswitch/libs/libteletone/src > > -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 > > -g > > -O2 -D_GNU_SOURCE -shared -o .libs/mod_gsmopen.so -shared -Wl,-x > > .libs/mod_gsmopen.o gsmopen_protocol.o -lm > > /usr/src/freeswitch/.libs/libfreeswitch.so > > -L/usr/src/freeswitch/libs/apr-util/xml/expat/lib > > /usr/src/freeswitch/libs/apr-util/xml/expat/lib/.libs/libexpat.a > > /usr/src/freeswitch/libs/apr/.libs/libapr-1.a -luuid -lpthread > > -L/usr/src/freeswitch/libs/srtp -L/usr/kerberos/lib -lcrypt -lrt -lssl > > -lcrypto -ldl -lz -lncurses -ljpeg > > -L/usr/src/freeswitch/libs/spandsp/src > > -lspandsp -lctb-0.16 -lgsmme -Wl,--rpath -Wl,/usr/local/freeswitch/lib > > -Wl,--rpath -Wl,/usr/local/freeswitch/mod > > make[1]: *** [mod_gsmopen.so] Error 1 > > make: *** [install] Error 1 > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > > Sincerely, > > > Giovanni Maruzzelli > > Cell : +39-347-2665618 > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > Sincerely, > > Giovanni Maruzzelli > Cell : +39-347-2665618 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-800-7876 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120514/385933c6/attachment-0001.html From krice at freeswitch.org Mon May 14 18:54:24 2012 From: krice at freeswitch.org (Ken Rice) Date: Mon, 14 May 2012 09:54:24 -0500 Subject: [Freeswitch-users] mod_gsmopen requires spandsp In-Reply-To: <1DE50890-EE01-4EEE-83A8-5B5F71BB3959@opencsta.org> Message-ID: You don?t want to skip mod_spandsp ... You really need it... Its as much of a core module as anything... Without it you are missing many many codecs you want... On 5/14/12 2:30 AM, "Chris Mylonas" wrote: > Hi FS List, > > FYI - as a shortcut to building my freeswitch, I skip spandsp - but it looks > like this mod_gsmopen wants it in there. > > > [root at space build]# cd /usr/src/freeswitch/src/mod/endpoints/mod_gsmopen/ > [root at space mod_gsmopen]# make clean > [root at space mod_gsmopen]# make install > Compiling gsmopen_protocol.cpp... > Compiling /usr/src/freeswitch/src/mod/endpoints/mod_gsmopen/mod_gsmopen.cpp... > mkdir .libs > Compiling /usr/src/freeswitch/src/mod/endpoints/mod_gsmopen/mod_gsmopen.cpp > ... > Creating mod_gsmopen.so... > /usr/bin/ld: cannot find -lspandsp > collect2: ld returned 1 exit status > g++ -I../../../../libs/spandsp/src -I../../../..//libs/tiff-3.8.2/libtiff > -DGSMOPEN_C_VER=\"44fd0de\" -DMODGSMOPEN_C_VER=\"44fd0de\" > -I/usr/src/freeswitch/libs/curl/include -I/usr/src/freeswitch/src/include > -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/libs/libteletone/src > -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 > -D_GNU_SOURCE -shared -o .libs/mod_gsmopen.so -shared -Wl,-x > .libs/mod_gsmopen.o gsmopen_protocol.o -lm > /usr/src/freeswitch/.libs/libfreeswitch.so > -L/usr/src/freeswitch/libs/apr-util/xml/expat/lib > /usr/src/freeswitch/libs/apr-util/xml/expat/lib/.libs/libexpat.a > /usr/src/freeswitch/libs/apr/.libs/libapr-1.a -luuid -lpthread > -L/usr/src/freeswitch/libs/srtp -L/usr/kerberos/lib -lcrypt -lrt -lssl > -lcrypto -ldl -lz -lncurses -ljpeg -L/usr/src/freeswitch/libs/spandsp/src > -lspandsp -lctb-0.16 -lgsmme -Wl,--rpath -Wl,/usr/local/freeswitch/lib > -Wl,--rpath -Wl,/usr/local/freeswitch/mod > make[1]: *** [mod_gsmopen.so] Error 1 > make: *** [install] Error 1 > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120514/19e7c14a/attachment.html From bdfoster at endigotech.com Mon May 14 19:03:21 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Mon, 14 May 2012 11:03:21 -0400 Subject: [Freeswitch-users] mod_xml_scgi compilation error In-Reply-To: <9B6666F4-AAA4-444B-8862-084C9EE32379@opencsta.org> References: <36BA7D4D-12BC-4676-B60F-F1A61B9003D8@opencsta.org> <9B6666F4-AAA4-444B-8862-084C9EE32379@opencsta.org> Message-ID: more than likely a bug, looks like something that would show up on a 32 bit machine. On Mon, May 14, 2012 at 3:32 AM, Chris Mylonas wrote: > I removed -Werror from the Makefile options line "SWITCH_AM_CFLAGS" and it > build successfully. > > > On 14/05/2012, at 3:03 PM, Chris Mylonas wrote: > > Hi FS Users, > > I was building FS on a box to test out the mod_gsmopen stuff that has been > mentioned on the list recently. > Updated CentOS 5 this morning, checked out a new git head. > > Getting this message that says "warnings being treated as errors" > Not sure if that is what is killing the build or if there genuinely is an > error. > > Any help would be great, > Cheers > Chris > > > making all mod_xml_scgi > Compiling ../../../../libs/libscgi/src/scgi.c... > cc1: warnings being treated as errors > ../../../../libs/libscgi/src/scgi.c: In function ?scgi_build_message?: > ../../../../libs/libscgi/src/scgi.c:94: warning: format ?%ld? expects type > ?long int?, but argument 4 has type ?size_t? > ../../../../libs/libscgi/src/scgi.c:94: warning: format ?%ld? expects type > ?long int?, but argument 4 has type ?size_t? > ../../../../libs/libscgi/src/scgi.c:103: warning: format ?%ld? expects > type ?long int?, but argument 4 has type ?size_t? > ../../../../libs/libscgi/src/scgi.c:103: warning: format ?%ld? expects > type ?long int?, but argument 4 has type ?size_t? > ../../../../libs/libscgi/src/scgi.c:110: warning: format ?%ld? expects > type ?long int?, but argument 4 has type ?size_t? > ../../../../libs/libscgi/src/scgi.c:110: warning: format ?%ld? expects > type ?long int?, but argument 4 has type ?size_t? > make[5]: *** [../../../../libs/libscgi/src/scgi.o] Error 1 > make[4]: *** [all] Error 1 > make[3]: *** [mod_xml_scgi-all] Error 1 > make[2]: *** [all-recursive] Error 1 > make[1]: *** [all-recursive] Error 1 > make: *** [all] Error 2 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-800-7876 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120514/d79ce651/attachment.html From chris at opencsta.org Mon May 14 19:16:38 2012 From: chris at opencsta.org (Chris Mylonas) Date: Tue, 15 May 2012 01:16:38 +1000 Subject: [Freeswitch-users] mod_gsmopen requires spandsp In-Reply-To: References: <1DE50890-EE01-4EEE-83A8-5B5F71BB3959@opencsta.org> <3484676B-F4CA-47E3-A8B0-9743291C1056@opencsta.org> <62D399FB-DFA0-4DD0-B86F-45E95026EF82@opencsta.org> Message-ID: <06870B57-67BE-4A1E-BE45-C56A8BC43CF5@opencsta.org> Excellent! That worked - thanks heaps! Some note in the wiki could be attached to that section re: DEBUG=1 creates ctbd. Is it editable by anyone registered or is there JIRA for documentation stuff? [root at space build]# make DEBUG=0 GPIB=0 [root at space build]# make DEBUG=0 GPIB=0 install [root at space build]# cd .. [root at space libctb-0.16]# cd .. [root at space mod_gsmopen]# make clean [root at space mod_gsmopen]# make install Compiling gsmopen_protocol.cpp... Compiling /usr/src/freeswitch/src/mod/endpoints/mod_gsmopen/mod_gsmopen.cpp... mkdir .libs Compiling /usr/src/freeswitch/src/mod/endpoints/mod_gsmopen/mod_gsmopen.cpp ... Creating mod_gsmopen.so... installing mod_gsmopen.so On 15/05/2012, at 12:51 AM, Brian Foster wrote: > if you compile libctb with debug it's actually libctbd, not libctb. > > On Mon, May 14, 2012 at 7:53 AM, Chris Mylonas wrote: > Thanks for the reply Giovanni. I have done the steps on the wiki. > > tl;dr; should i put a prefix when i'm making the dependent mods so they don't go into /usr/local/lib, or this is why ldconfig is run - to tell the system where the libs are. > > All the relevant stuff is below > > Hope you can see something wrong, > Cheers > Chris > > e.g. > here is my bash history > > 1050 cd freeswitch/ > 1051 ls > 1052 find . -name gsmlib > 1053 cd src/mod/endpoints/mod_gsmopen/ > 1054 ls > 1055 cd gsmlib/ > 1056 ls > 1057 cd gsmlib-1.10-patched-13ubuntu/ > 1058 ls > 1059 ./configure > 1060 make > 1061 make install > 1062 ldconfig > 1063 cd /usr/src/freeswitch/src/mod/endpoints/mod_gsmopen/libctb-0.16/build > 1064 make DEBUG=1 GPIB=0 > 1065 make DEBUG=1 GPIB=0 install > 1066 ldconfig > 1067 cd /usr/src/freeswitch/src/mod/endpoints/mod_gsmopen/ > 1068 make clean > 1069 make install > > Just to repeat the compilation error > Creating mod_gsmopen.so... > /usr/bin/ld: cannot find -lctb-0.16 > collect2: ld returned 1 exit status > g++ -I../../../../libs/spandsp/src -I../../../..//libs/tiff-3.8.2/libtiff -DGSMOPEN_C_VER=\"44fd0de\" -DMODGSMOPEN_C_VER=\"44fd0de\" -I/usr/src/freeswitch/libs/curl/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -shared -o .libs/mod_gsmopen.so -shared -Wl,-x .libs/mod_gsmopen.o gsmopen_protocol.o /usr/src/freeswitch/.libs/libfreeswitch.so -L/usr/src/freeswitch/libs/apr-util/xml/expat/lib /usr/src/freeswitch/libs/apr-util/xml/expat/lib/.libs/libexpat.a /usr/src/freeswitch/libs/apr/.libs/libapr-1.a -luuid -lpthread -L/usr/src/freeswitch/libs/srtp -L/usr/kerberos/lib -lcrypt -lrt -lssl -lcrypto -ldl -lncurses -L/usr/src/freeswitch/libs/spandsp/src /usr/src/freeswitch/libs/spandsp/src/.libs/libspandsp.a -L/usr/src/freeswitch/libs/tiff-3.8.2/libtiff /usr/src/freeswitch/libs/tiff-3.8.2/libtiff/.libs/libtiff.a -ljpeg -lz -lm -lc -lctb-0.16 -lgsmme -Wl,--rpath -Wl,/usr/local/freeswitch/lib -Wl,--rpath -Wl,/usr/local/freeswitch/mod > make[1]: *** [mod_gsmopen.so] Error 1 > make: *** [install] Error 1 > > > > > All the gsmlib stuff that is NOT in the freeswitch src dir is here > > [root at space mod_gsmopen]# locate gsmlib | grep -v src > /usr/local/include/gsmlib > /usr/local/include/gsmlib/gsm_at.h > /usr/local/include/gsmlib/gsm_cb.h > /usr/local/include/gsmlib/gsm_error.h > /usr/local/include/gsmlib/gsm_event.h > /usr/local/include/gsmlib/gsm_map_key.h > /usr/local/include/gsmlib/gsm_me_ta.h > /usr/local/include/gsmlib/gsm_parser.h > /usr/local/include/gsmlib/gsm_phonebook.h > /usr/local/include/gsmlib/gsm_port.h > /usr/local/include/gsmlib/gsm_sie_me.h > /usr/local/include/gsmlib/gsm_sms.h > /usr/local/include/gsmlib/gsm_sms_codec.h > /usr/local/include/gsmlib/gsm_sms_store.h > /usr/local/include/gsmlib/gsm_sorted_phonebook.h > /usr/local/include/gsmlib/gsm_sorted_phonebook_base.h > /usr/local/include/gsmlib/gsm_sorted_sms_store.h > /usr/local/include/gsmlib/gsm_unix_serial.h > /usr/local/include/gsmlib/gsm_util.h > /usr/local/share/locale/de/LC_MESSAGES/gsmlib.mo > > > And the ctb stuff is in /usr/local/lib > > /usr/local/include/ctb-0.16 > /usr/local/include/ctb-0.16/ctb.h > /usr/local/include/ctb-0.16/fifo.h > /usr/local/include/ctb-0.16/getopt.h > /usr/local/include/ctb-0.16/iobase.h > /usr/local/include/ctb-0.16/linux > /usr/local/include/ctb-0.16/portscan.h > /usr/local/include/ctb-0.16/serport.h > /usr/local/include/ctb-0.16/serportx.h > /usr/local/include/ctb-0.16/timer.h > /usr/local/include/ctb-0.16/linux/serport.h > /usr/local/include/ctb-0.16/linux/timer.h > /usr/local/lib/libctbd-0.16.a > /usr/local/lib/libctbd-0.16.so > > > gcc version 4.1.2 20080704 (Red Hat 4.1.2-52) > GNU Make 3.81 > CentOS release 5.8 (Final) > > > > > On 14/05/2012, at 9:38 PM, Giovanni Maruzzelli wrote: > >> you must first compile and install libctb, as per the wiki page ( >> http://wiki.freeswitch.org/wiki/gsmopen ) >> then, after installation of libctb and gsmlib (as per wiki), be sure >> to update your dinamic link cache, or compiler will not find then. >> >> Eg: ldconfig >> >> >> On 5/14/12, Chris Mylonas wrote: >>> All good GM, no inconvenience, just a minor gotcha ;) >>> I am unable to compile mod_gsmopen though. >>> >>> It complains about not being able to find ctb-0.16 >>> The actual filename is libctbd-0.16.so in /usr/local/lib as you can see from >>> the 2nd lot of stuff. >>> >>> How do I fix this? >>> >>> >>> [root at space mod_gsmopen]# make install >>> Creating mod_gsmopen.so... >>> /usr/bin/ld: cannot find -lctb-0.16 >>> collect2: ld returned 1 exit status >>> g++ -I../../../../libs/spandsp/src -I../../../..//libs/tiff-3.8.2/libtiff >>> -DGSMOPEN_C_VER=\"44fd0de\" -DMODGSMOPEN_C_VER=\"44fd0de\" >>> -I/usr/src/freeswitch/libs/curl/include -I/usr/src/freeswitch/src/include >>> -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/libs/libteletone/src >>> -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g >>> -O2 -D_GNU_SOURCE -shared -o .libs/mod_gsmopen.so -shared -Wl,-x >>> .libs/mod_gsmopen.o gsmopen_protocol.o >>> /usr/src/freeswitch/.libs/libfreeswitch.so >>> -L/usr/src/freeswitch/libs/apr-util/xml/expat/lib >>> /usr/src/freeswitch/libs/apr-util/xml/expat/lib/.libs/libexpat.a >>> /usr/src/freeswitch/libs/apr/.libs/libapr-1.a -luuid -lpthread >>> -L/usr/src/freeswitch/libs/srtp -L/usr/kerberos/lib -lcrypt -lrt -lssl >>> -lcrypto -ldl -lncurses -L/usr/src/freeswitch/libs/spandsp/src >>> /usr/src/freeswitch/libs/spandsp/src/.libs/libspandsp.a >>> -L/usr/src/freeswitch/libs/tiff-3.8.2/libtiff >>> /usr/src/freeswitch/libs/tiff-3.8.2/libtiff/.libs/libtiff.a -ljpeg -lz -lm >>> -lc -lctb-0.16 -lgsmme -Wl,--rpath -Wl,/usr/local/freeswitch/lib >>> -Wl,--rpath -Wl,/usr/local/freeswitch/mod >>> make[1]: *** [mod_gsmopen.so] Error 1 >>> make: *** [install] Error 1 >>> >>> >>> >>> [root at space mod_gsmopen]# ldd /usr/local/lib/libctbd-0.16.so >>> linux-gate.so.1 => (0x00754000) >>> libpthread.so.0 => /lib/libpthread.so.0 (0x00e83000) >>> libstdc++.so.6 => /usr/lib/libstdc++.so.6 (0x00d28000) >>> libm.so.6 => /lib/libm.so.6 (0x00964000) >>> libgcc_s.so.1 => /lib/libgcc_s.so.1 (0x00e1f000) >>> libc.so.6 => /lib/libc.so.6 (0x00110000) >>> /lib/ld-linux.so.2 (0x003ea000) >>> >>> >>> >>> On 14/05/2012, at 6:44 PM, Giovanni Maruzzelli wrote: >>> >>>> yes, it requires libspandsp, maybe the Makefile it's not yet tweaked >>>> to build the library automatically. >>>> >>>> So, please first build mod_spandsp, then mod_gsmopen. >>>> >>>> We'll fixx the Makefile soon, sorry for the inconvenience. >>>> >>>> -giovanni >>>> >>>> On Mon, May 14, 2012 at 9:30 AM, Chris Mylonas >>>> wrote: >>>>> Hi FS List, >>>>> >>>>> FYI - as a shortcut to building my freeswitch, I skip spandsp - but it >>>>> looks >>>>> like this mod_gsmopen wants it in there. >>>>> >>>>> >>>>> [root at space build]# cd >>>>> /usr/src/freeswitch/src/mod/endpoints/mod_gsmopen/ >>>>> [root at space mod_gsmopen]# make clean >>>>> [root at space mod_gsmopen]# make install >>>>> Compiling gsmopen_protocol.cpp... >>>>> Compiling >>>>> /usr/src/freeswitch/src/mod/endpoints/mod_gsmopen/mod_gsmopen.cpp... >>>>> mkdir .libs >>>>> Compiling >>>>> /usr/src/freeswitch/src/mod/endpoints/mod_gsmopen/mod_gsmopen.cpp >>>>> ... >>>>> Creating mod_gsmopen.so... >>>>> /usr/bin/ld: cannot find -lspandsp >>>>> collect2: ld returned 1 exit status >>>>> g++ -I../../../../libs/spandsp/src >>>>> -I../../../..//libs/tiff-3.8.2/libtiff >>>>> -DGSMOPEN_C_VER=\"44fd0de\" -DMODGSMOPEN_C_VER=\"44fd0de\" >>>>> -I/usr/src/freeswitch/libs/curl/include >>>>> -I/usr/src/freeswitch/src/include >>>>> -I/usr/src/freeswitch/src/include >>>>> -I/usr/src/freeswitch/libs/libteletone/src >>>>> -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 >>>>> -g >>>>> -O2 -D_GNU_SOURCE -shared -o .libs/mod_gsmopen.so -shared -Wl,-x >>>>> .libs/mod_gsmopen.o gsmopen_protocol.o -lm >>>>> /usr/src/freeswitch/.libs/libfreeswitch.so >>>>> -L/usr/src/freeswitch/libs/apr-util/xml/expat/lib >>>>> /usr/src/freeswitch/libs/apr-util/xml/expat/lib/.libs/libexpat.a >>>>> /usr/src/freeswitch/libs/apr/.libs/libapr-1.a -luuid -lpthread >>>>> -L/usr/src/freeswitch/libs/srtp -L/usr/kerberos/lib -lcrypt -lrt -lssl >>>>> -lcrypto -ldl -lz -lncurses -ljpeg >>>>> -L/usr/src/freeswitch/libs/spandsp/src >>>>> -lspandsp -lctb-0.16 -lgsmme -Wl,--rpath -Wl,/usr/local/freeswitch/lib >>>>> -Wl,--rpath -Wl,/usr/local/freeswitch/mod >>>>> make[1]: *** [mod_gsmopen.so] Error 1 >>>>> make: *** [install] Error 1 >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> >>>> -- >>>> Sincerely, >>>> >>>> Giovanni Maruzzelli >>>> Cell : +39-347-2665618 >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Sincerely, >> >> Giovanni Maruzzelli >> Cell : +39-347-2665618 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Brian D. Foster > Endigo Computer LLC > Email: bdfoster at endigotech.com > Phone: 317-800-7876 > Indianapolis, Indiana, USA > > This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120515/d5269e90/attachment-0001.html From chris at opencsta.org Mon May 14 19:17:44 2012 From: chris at opencsta.org (Chris Mylonas) Date: Tue, 15 May 2012 01:17:44 +1000 Subject: [Freeswitch-users] mod_xml_scgi compilation error In-Reply-To: References: <36BA7D4D-12BC-4676-B60F-F1A61B9003D8@opencsta.org> <9B6666F4-AAA4-444B-8862-084C9EE32379@opencsta.org> Message-ID: <76C3F38E-3D24-49B4-9006-433266F5E59C@opencsta.org> Are 32 bit machines JIRA-worthy for something like this? On 15/05/2012, at 1:03 AM, Brian Foster wrote: > more than likely a bug, looks like something that would show up on a 32 bit machine. > > On Mon, May 14, 2012 at 3:32 AM, Chris Mylonas wrote: > I removed -Werror from the Makefile options line "SWITCH_AM_CFLAGS" and it build successfully. > > > On 14/05/2012, at 3:03 PM, Chris Mylonas wrote: > >> Hi FS Users, >> >> I was building FS on a box to test out the mod_gsmopen stuff that has been mentioned on the list recently. >> Updated CentOS 5 this morning, checked out a new git head. >> >> Getting this message that says "warnings being treated as errors" >> Not sure if that is what is killing the build or if there genuinely is an error. >> >> Any help would be great, >> Cheers >> Chris >> >> >> making all mod_xml_scgi >> Compiling ../../../../libs/libscgi/src/scgi.c... >> cc1: warnings being treated as errors >> ../../../../libs/libscgi/src/scgi.c: In function ?scgi_build_message?: >> ../../../../libs/libscgi/src/scgi.c:94: warning: format ?%ld? expects type ?long int?, but argument 4 has type ?size_t? >> ../../../../libs/libscgi/src/scgi.c:94: warning: format ?%ld? expects type ?long int?, but argument 4 has type ?size_t? >> ../../../../libs/libscgi/src/scgi.c:103: warning: format ?%ld? expects type ?long int?, but argument 4 has type ?size_t? >> ../../../../libs/libscgi/src/scgi.c:103: warning: format ?%ld? expects type ?long int?, but argument 4 has type ?size_t? >> ../../../../libs/libscgi/src/scgi.c:110: warning: format ?%ld? expects type ?long int?, but argument 4 has type ?size_t? >> ../../../../libs/libscgi/src/scgi.c:110: warning: format ?%ld? expects type ?long int?, but argument 4 has type ?size_t? >> make[5]: *** [../../../../libs/libscgi/src/scgi.o] Error 1 >> make[4]: *** [all] Error 1 >> make[3]: *** [mod_xml_scgi-all] Error 1 >> make[2]: *** [all-recursive] Error 1 >> make[1]: *** [all-recursive] Error 1 >> make: *** [all] Error 2 >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Brian D. Foster > Endigo Computer LLC > Email: bdfoster at endigotech.com > Phone: 317-800-7876 > Indianapolis, Indiana, USA > > This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120515/017f1b5d/attachment.html From krice at freeswitch.org Mon May 14 19:19:24 2012 From: krice at freeswitch.org (Ken Rice) Date: Mon, 14 May 2012 10:19:24 -0500 Subject: [Freeswitch-users] mod_xml_scgi compilation error In-Reply-To: <76C3F38E-3D24-49B4-9006-433266F5E59C@opencsta.org> Message-ID: Yes we should hear about 32bit issues on Jira also... On 5/14/12 10:17 AM, "Chris Mylonas" wrote: > Are 32 bit machines JIRA-worthy for something like this? > > > On 15/05/2012, at 1:03 AM, Brian Foster wrote: > >> more than likely a bug, looks like something that would show up on a 32 bit >> machine. >> >> On Mon, May 14, 2012 at 3:32 AM, Chris Mylonas wrote: >>> I removed -Werror from the Makefile options line "SWITCH_AM_CFLAGS" and it >>> build successfully. >>> >>> >>> On 14/05/2012, at 3:03 PM, Chris Mylonas wrote: >>> >>>> Hi FS Users, >>>> >>>> I was building FS on a box to test out the mod_gsmopen stuff that has been >>>> mentioned on the list recently. >>>> Updated CentOS 5 this morning, checked out a new git head. >>>> >>>> Getting this message that says "warnings being treated as errors" >>>> Not sure if that is what is killing the build or if there genuinely is an >>>> error. >>>> >>>> Any help would be great, >>>> Cheers >>>> Chris >>>> >>>> >>>> making all mod_xml_scgi >>>> Compiling ../../../../libs/libscgi/src/scgi.c... >>>> cc1: warnings being treated as errors >>>> ../../../../libs/libscgi/src/scgi.c: In function ?scgi_build_message?: >>>> ../../../../libs/libscgi/src/scgi.c:94: warning: format ?%ld? expects type >>>> ?long int?, but argument 4 has type ?size_t? >>>> ../../../../libs/libscgi/src/scgi.c:94: warning: format ?%ld? expects type >>>> ?long int?, but argument 4 has type ?size_t? >>>> ../../../../libs/libscgi/src/scgi.c:103: warning: format ?%ld? expects type >>>> ?long int?, but argument 4 has type ?size_t? >>>> ../../../../libs/libscgi/src/scgi.c:103: warning: format ?%ld? expects type >>>> ?long int?, but argument 4 has type ?size_t? >>>> ../../../../libs/libscgi/src/scgi.c:110: warning: format ?%ld? expects type >>>> ?long int?, but argument 4 has type ?size_t? >>>> ../../../../libs/libscgi/src/scgi.c:110: warning: format ?%ld? expects type >>>> ?long int?, but argument 4 has type ?size_t? >>>> make[5]: *** [../../../../libs/libscgi/src/scgi.o] Error 1 >>>> make[4]: *** [all] Error 1 >>>> make[3]: *** [mod_xml_scgi-all] Error 1 >>>> make[2]: *** [all-recursive] Error 1 >>>> make[1]: *** [all-recursive] Error 1 >>>> make: *** [all] Error 2 >>>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120514/f84ae063/attachment.html From bote_radio at botecomm.com Mon May 14 19:20:13 2012 From: bote_radio at botecomm.com (Bote Man) Date: Mon, 14 May 2012 11:20:13 -0400 Subject: [Freeswitch-users] Huawei E169 Group Buy In-Reply-To: References: <4FAF49A5.5000203@coppice.org> Message-ID: <023801cd31e5$10cf3820$326da860$@com> An authoritative source of mine at AT&T Mobility said: HSDPA UMTS (850, 1900MHz) . GSM GPRS EDGE (850 , 1900MHz) are supported, which looks like his edited version of the E169 data sheet that I e-mailed to him. Bote From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian Foster Sent: Sunday, 13 May, 2012 02:39 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Huawei E169 Group Buy The Huawei E169 is confirmed to work with mod_gsmopen. Special thanks to Marcus Brown (mzb- on #freedoh) for helping me figure all of this stuff out and for donating a E169, a SIM card, and a virtual machine to run tests. Steve, understood. As far as I can tell the bands that the E169 supports SHOULD cover most of the US carriers that support GSM. Like I said in my last email, it is up to you guys to make sure that it will work in your situation. If it doesn't I'd sell it on ebay, I don't think you'll have a problem getting rid of it. From: http://3g-modem.wetpaint.com/page/Huawei+E169+(E169G,+E169V,+K3520) specifications Huawei E169 manufacturer: Huawei model: E169 interface: USB 2.0 (A plug) GSM frequency bands: 850, 900, 1800, 1900 UMTS frequency bands: 900, 2100 HSDPA: 7,2 MBit/s HSUPA: - EDGE: 236,8 KBit/s GPRS: 57,6 KBit//s CSD: ? connector for ext. antenna: + (on some versions covered by housing) connector type: SMK CRS5001 internal antenna diversity: + voice telephony: + NAND-flashmemory: + microSD-drive: + (up to 8GB) rebranded versions: Vodafone K3520 Vodafone E169V Also: http://www.alibaba.com/showroom/huawei-e169-modem.html -BDF On Sun, May 13, 2012 at 1:41 AM, Steve Underwood wrote: Hi, When looking at models of 3G dongles look carefully at whether they support the features you need. This basically comes down to supporting the voice features, and supporting the bands you need in your area. Most recent dongles with a Qualcomm chip set (which is most of them) support the voice features needed to work with FS. However, many SIM locked ones have the feature blocked. Unlocked ones should be OK. Most dongles now support quad band 2G operation, which covers most of the frequencies ever used for 2G GSM. However, most dongles only support one 3G band. Some support 2 bands. There are, however, several bands used for 3G around the planet. If the dongle you choose doesn't support your local 3G bands, you will only be able to use it for 2G communication. That might be OK. It might not. It depends on your local service offerings. Just beware. Steve On 05/13/2012 09:49 AM, Brian Foster wrote: > > Fellow freeswitchers, > > I am sending this email out to let you all know that I am setting up a > group buy for the Huawei E169. It should be compatable with > mod_gsmopen, and that will be tested before we go and do a buy. The > reason why there is interest in this particular model is the fact that > it does have an external antenna jack. Otherwise they are fairly close > to the E1550. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120514/bc84003c/attachment-0001.html From steveu at coppice.org Mon May 14 19:20:54 2012 From: steveu at coppice.org (Steve Underwood) Date: Mon, 14 May 2012 23:20:54 +0800 Subject: [Freeswitch-users] mod_gsmopen requires spandsp In-Reply-To: References: Message-ID: <4FB122D6.2090803@coppice.org> On 05/14/2012 10:54 PM, Ken Rice wrote: > You don?t want to skip mod_spandsp ... You really need it... Its as > much of a core module as anything... Without it you are missing many > many codecs you want... I wonder what inspires this "Oh, that looks important. Let's leave it out of the build" way of looking at thing? :-\ Steve From chris at opencsta.org Mon May 14 19:51:15 2012 From: chris at opencsta.org (Chris Mylonas) Date: Tue, 15 May 2012 01:51:15 +1000 Subject: [Freeswitch-users] mod_gsmopen requires spandsp In-Reply-To: <4FB122D6.2090803@coppice.org> References: <4FB122D6.2090803@coppice.org> Message-ID: <2F682649-B1FC-4CBE-A51E-8634D697E8BE@opencsta.org> i've never heard of sp & sp - what is this sp & sp for anyway :P there's a moderately good reason BTW. the inspiration came from installing on osx last week and missing libjpeg because "spandsp needed it" - so to shortcut my way through, i didn't install libjpeg (nor spandsp). it didn't build, and i installed libjpeg. it built. still minus spandsp. didn't initiate a call, so didn't come across the codec problem. and now that i'm building on linux, i've brought my bad osx habits with me... i'm just used to having a standalone fax machine don't shoot the t.38/FoIP luddite :) Cheers Chris On 15/05/2012, at 1:20 AM, Steve Underwood wrote: > On 05/14/2012 10:54 PM, Ken Rice wrote: >> You don?t want to skip mod_spandsp ... You really need it... Its as >> much of a core module as anything... Without it you are missing many >> many codecs you want... > I wonder what inspires this "Oh, that looks important. Let's leave it > out of the build" way of looking at thing? :-\ > > Steve > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From krice at freeswitch.org Mon May 14 19:57:12 2012 From: krice at freeswitch.org (Ken Rice) Date: Mon, 14 May 2012 10:57:12 -0500 Subject: [Freeswitch-users] mod_gsmopen requires spandsp In-Reply-To: <2F682649-B1FC-4CBE-A51E-8634D697E8BE@opencsta.org> Message-ID: Spandsp isnt just about Faxing and T38... Spandsp does that and much more... Its where we get codecs like G711, g722, g726, GSM, LPC-10, and ADPCM... K On 5/14/12 10:51 AM, "Chris Mylonas" wrote: > i've never heard of sp & sp - what is this sp & sp for anyway :P > there's a moderately good reason BTW. > > > the inspiration came from installing on osx last week and missing libjpeg > because "spandsp needed it" - so to shortcut my way through, i didn't install > libjpeg (nor spandsp). > it didn't build, and i installed libjpeg. > it built. still minus spandsp. didn't initiate a call, so didn't come across > the codec problem. > and now that i'm building on linux, i've brought my bad osx habits with me... > > > i'm just used to having a standalone fax machine > don't shoot the t.38/FoIP luddite :) > > Cheers > Chris > > On 15/05/2012, at 1:20 AM, Steve Underwood wrote: > >> On 05/14/2012 10:54 PM, Ken Rice wrote: >>> You don?t want to skip mod_spandsp ... You really need it... Its as >>> much of a core module as anything... Without it you are missing many >>> many codecs you want... >> I wonder what inspires this "Oh, that looks important. Let's leave it >> out of the build" way of looking at thing? :-\ >> >> Steve >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From chris at opencsta.org Mon May 14 20:07:20 2012 From: chris at opencsta.org (Chris Mylonas) Date: Tue, 15 May 2012 02:07:20 +1000 Subject: [Freeswitch-users] mod_xml_scgi compilation error In-Reply-To: References: Message-ID: <74FF7A89-8FC5-4D2D-9BA1-4D02196F7CDC@opencsta.org> Already reported earlier today, which I had read... - http://jira.freeswitch.org/browse/FS-4218 I just switched the compile flag for -Werror off by removing it /hacky On 15/05/2012, at 1:19 AM, Ken Rice wrote: > Yes we should hear about 32bit issues on Jira also... > > > On 5/14/12 10:17 AM, "Chris Mylonas" wrote: > >> Are 32 bit machines JIRA-worthy for something like this? >> >> >> On 15/05/2012, at 1:03 AM, Brian Foster wrote: >> >>> more than likely a bug, looks like something that would show up on a 32 bit machine. >>> >>> On Mon, May 14, 2012 at 3:32 AM, Chris Mylonas wrote: >>>> I removed -Werror from the Makefile options line "SWITCH_AM_CFLAGS" and it build successfully. >>>> >>>> >>>> On 14/05/2012, at 3:03 PM, Chris Mylonas wrote: >>>> >>>>> Hi FS Users, >>>>> >>>>> I was building FS on a box to test out the mod_gsmopen stuff that has been mentioned on the list recently. >>>>> Updated CentOS 5 this morning, checked out a new git head. >>>>> >>>>> Getting this message that says "warnings being treated as errors" >>>>> Not sure if that is what is killing the build or if there genuinely is an error. >>>>> >>>>> Any help would be great, >>>>> Cheers >>>>> Chris >>>>> >>>>> >>>>> making all mod_xml_scgi >>>>> Compiling ../../../../libs/libscgi/src/scgi.c... >>>>> cc1: warnings being treated as errors >>>>> ../../../../libs/libscgi/src/scgi.c: In function ?scgi_build_message?: >>>>> ../../../../libs/libscgi/src/scgi.c:94: warning: format ?%ld? expects type ?long int?, but argument 4 has type ?size_t? >>>>> ../../../../libs/libscgi/src/scgi.c:94: warning: format ?%ld? expects type ?long int?, but argument 4 has type ?size_t? >>>>> ../../../../libs/libscgi/src/scgi.c:103: warning: format ?%ld? expects type ?long int?, but argument 4 has type ?size_t? >>>>> ../../../../libs/libscgi/src/scgi.c:103: warning: format ?%ld? expects type ?long int?, but argument 4 has type ?size_t? >>>>> ../../../../libs/libscgi/src/scgi.c:110: warning: format ?%ld? expects type ?long int?, but argument 4 has type ?size_t? >>>>> ../../../../libs/libscgi/src/scgi.c:110: warning: format ?%ld? expects type ?long int?, but argument 4 has type ?size_t? >>>>> make[5]: *** [../../../../libs/libscgi/src/scgi.o] Error 1 >>>>> make[4]: *** [all] Error 1 >>>>> make[3]: *** [mod_xml_scgi-all] Error 1 >>>>> make[2]: *** [all-recursive] Error 1 >>>>> make[1]: *** [all-recursive] Error 1 >>>>> make: *** [all] Error 2 >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120515/a3426407/attachment.html From chris at opencsta.org Mon May 14 20:10:35 2012 From: chris at opencsta.org (Chris Mylonas) Date: Tue, 15 May 2012 02:10:35 +1000 Subject: [Freeswitch-users] mod_gsmopen requires spandsp In-Reply-To: References: Message-ID: Noted. I had no idea. Now it makes Steve's initial question on why anyone would want to leave it out make a whole lot more sense. Apologies & Respect! Cheers Chris On 15/05/2012, at 1:57 AM, Ken Rice wrote: > Spandsp isnt just about Faxing and T38... Spandsp does that and much > more... Its where we get codecs like G711, g722, g726, GSM, LPC-10, and > ADPCM... > > K > > > On 5/14/12 10:51 AM, "Chris Mylonas" wrote: > >> i've never heard of sp & sp - what is this sp & sp for anyway :P >> there's a moderately good reason BTW. >> >> >> the inspiration came from installing on osx last week and missing libjpeg >> because "spandsp needed it" - so to shortcut my way through, i didn't install >> libjpeg (nor spandsp). >> it didn't build, and i installed libjpeg. >> it built. still minus spandsp. didn't initiate a call, so didn't come across >> the codec problem. >> and now that i'm building on linux, i've brought my bad osx habits with me... >> >> >> i'm just used to having a standalone fax machine >> don't shoot the t.38/FoIP luddite :) >> >> Cheers >> Chris >> >> On 15/05/2012, at 1:20 AM, Steve Underwood wrote: >> >>> On 05/14/2012 10:54 PM, Ken Rice wrote: >>>> You don?t want to skip mod_spandsp ... You really need it... Its as >>>> much of a core module as anything... Without it you are missing many >>>> many codecs you want... >>> I wonder what inspires this "Oh, that looks important. Let's leave it >>> out of the build" way of looking at thing? :-\ >>> >>> Steve >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From bdfoster at endigotech.com Mon May 14 20:15:54 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Mon, 14 May 2012 12:15:54 -0400 Subject: [Freeswitch-users] Huawei E169 Group Buy In-Reply-To: <023801cd31e5$10cf3820$326da860$@com> References: <4FAF49A5.5000203@coppice.org> <023801cd31e5$10cf3820$326da860$@com> Message-ID: ...which renders the E169 useless for data but good for GSM so voice/SMS will work. I did look for one that was compatible on the data and GSM side of things but I'm turning up with nothing. Maybe there is a Huawei modem out there that either A) is E1550 compatible (but isn't on Giovanni's list) or B) support can be added fairly easily for it. 4G (LTS) is pretty much out of the question from what I can see as well, 3G (HSPA) would be a good starting point. So, recap: - Huawei Modem that is E1550 compatible (or support can be added) - Voice Enabled - UTMS on 850/1900 - GSM on 850/1900 - Unlocked (or unlockable) *Bonus Points: external antenna jack I'll keep searching around. I've got a friend who's also trying to help me out looking through this stuff (a lot more experienced than I am) but if anyone's got something better please feel free to share. -BDF On Mon, May 14, 2012 at 11:20 AM, Bote Man wrote: > An authoritative source of mine at AT&T Mobility said:**** > > ** ** > > HSDPA UMTS (850, 1900MHz) . GSM GPRS EDGE (850 , 1900MHz)**** > > ** ** > > are supported, which looks like his edited version of the E169 data sheet > that I e-mailed to him.**** > > ** ** > > Bote**** > > ** ** > > ** ** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Brian Foster > *Sent:* Sunday, 13 May, 2012 02:39 > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Huawei E169 Group Buy**** > > ** ** > > The Huawei E169 is confirmed to work with mod_gsmopen. Special thanks to > Marcus Brown (mzb- on #freedoh) for helping me figure all of this stuff out > and for donating a E169, a SIM card, and a virtual machine to run tests. > **** > > ** ** > > Steve, understood. As far as I can tell the bands that the E169 supports > SHOULD cover most of the US carriers that support GSM. Like I said in my > last email, it is up to you guys to make sure that it will work in your > situation. If it doesn't I'd sell it on ebay, I don't think you'll have a > problem getting rid of it.**** > > From: http://3g-modem.wetpaint.com/page/Huawei+E169+(E169G,+E169V,+K3520)* > *** > > ** ** > > specifications > > Huawei E169**** > > manufacturer:**** > > Huawei**** > > model:**** > > E169**** > > interface:**** > > USB 2.0 (A plug)**** > > GSM frequency bands:**** > > 850, 900, 1800, 1900**** > > UMTS frequency bands:**** > > 900, 2100**** > > HSDPA:**** > > 7,2 MBit/s**** > > HSUPA:**** > > -**** > > EDGE:**** > > 236,8 KBit/s**** > > GPRS:**** > > 57,6 KBit//s**** > > CSD:**** > > ?**** > > connector for ext. antenna:**** > > + (on some versions covered by housing)**** > > connector type:**** > > SMK CRS5001**** > > internal antenna diversity:**** > > +**** > > voice telephony:**** > > +**** > > NAND-flashmemory:**** > > +**** > > microSD-drive:**** > > + (up to 8GB)**** > > rebranded versions:**** > > Vodafone K3520 > Vodafone E169V**** > > Also: http://www.alibaba.com/showroom/huawei-e169-modem.html**** > > ** ** > > -BDF**** > > ** ** > > On Sun, May 13, 2012 at 1:41 AM, Steve Underwood > wrote:**** > > Hi, > > When looking at models of 3G dongles look carefully at whether they > support the features you need. This basically comes down to supporting > the voice features, and supporting the bands you need in your area. > > Most recent dongles with a Qualcomm chip set (which is most of them) > support the voice features needed to work with FS. However, many SIM > locked ones have the feature blocked. Unlocked ones should be OK. > > Most dongles now support quad band 2G operation, which covers most of > the frequencies ever used for 2G GSM. However, most dongles only support > one 3G band. Some support 2 bands. There are, however, several bands > used for 3G around the planet. If the dongle you choose doesn't support > your local 3G bands, you will only be able to use it for 2G > communication. That might be OK. It might not. It depends on your local > service offerings. Just beware. > > Steve**** > > > On 05/13/2012 09:49 AM, Brian Foster wrote: > > > > Fellow freeswitchers, > > > > I am sending this email out to let you all know that I am setting up a > > group buy for the Huawei E169. It should be compatable with > > mod_gsmopen, and that will be tested before we go and do a buy. The > > reason why there is interest in this particular model is the fact that > > it does have an external antenna jack. Otherwise they are fairly close > > to the E1550. > > > > **** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-800-7876 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120514/ffce459d/attachment-0001.html From mario_fs at mgtech.com Mon May 14 20:16:44 2012 From: mario_fs at mgtech.com (Mario G) Date: Mon, 14 May 2012 09:16:44 -0700 Subject: [Freeswitch-users] mod_gsmopen requires spandsp In-Reply-To: References: Message-ID: <802F385B-C86D-499C-ACFD-294A94964EA4@mgtech.com> Although the osX FS install wiki already says you need SPANDSP I will make it clearer when I update it (the wiki). Mario G On May 14, 2012, at 9:10 AM, Chris Mylonas wrote: > Noted. I had no idea. Now it makes Steve's initial question on why anyone would want to leave it out make a whole lot more sense. > Apologies & Respect! > > Cheers > Chris > > On 15/05/2012, at 1:57 AM, Ken Rice wrote: > >> Spandsp isnt just about Faxing and T38... Spandsp does that and much >> more... Its where we get codecs like G711, g722, g726, GSM, LPC-10, and >> ADPCM... >> >> K >> >> >> On 5/14/12 10:51 AM, "Chris Mylonas" wrote: >> >>> i've never heard of sp & sp - what is this sp & sp for anyway :P >>> there's a moderately good reason BTW. >>> >>> >>> the inspiration came from installing on osx last week and missing libjpeg >>> because "spandsp needed it" - so to shortcut my way through, i didn't install >>> libjpeg (nor spandsp). >>> it didn't build, and i installed libjpeg. >>> it built. still minus spandsp. didn't initiate a call, so didn't come across >>> the codec problem. >>> and now that i'm building on linux, i've brought my bad osx habits with me... >>> >>> >>> i'm just used to having a standalone fax machine >>> don't shoot the t.38/FoIP luddite :) >>> >>> Cheers >>> Chris >>> >>> On 15/05/2012, at 1:20 AM, Steve Underwood wrote: >>> >>>> On 05/14/2012 10:54 PM, Ken Rice wrote: >>>>> You don?t want to skip mod_spandsp ... You really need it... Its as >>>>> much of a core module as anything... Without it you are missing many >>>>> many codecs you want... >>>> I wonder what inspires this "Oh, that looks important. Let's leave it >>>> out of the build" way of looking at thing? :-\ >>>> >>>> Steve >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at 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 at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From bdfoster at endigotech.com Mon May 14 20:17:30 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Mon, 14 May 2012 12:17:30 -0400 Subject: [Freeswitch-users] mod_xml_scgi compilation error In-Reply-To: <74FF7A89-8FC5-4D2D-9BA1-4D02196F7CDC@opencsta.org> References: <74FF7A89-8FC5-4D2D-9BA1-4D02196F7CDC@opencsta.org> Message-ID: Regardless, it's still a bug and things might not work as intended even though all is well on the surface. Best to report (as you did) and get a fix, then remove the hack and update. -BDF On Mon, May 14, 2012 at 12:07 PM, Chris Mylonas wrote: > Already reported earlier today, which I had read... - > http://jira.freeswitch.org/browse/FS-4218 > I just switched the compile flag for -Werror off by removing it > > /hacky > > On 15/05/2012, at 1:19 AM, Ken Rice wrote: > > Yes we should hear about 32bit issues on Jira also... > > > On 5/14/12 10:17 AM, "Chris Mylonas" wrote: > > Are 32 bit machines JIRA-worthy for something like this? > > > On 15/05/2012, at 1:03 AM, Brian Foster wrote: > > more than likely a bug, looks like something that would show up on a 32 > bit machine. > > On Mon, May 14, 2012 at 3:32 AM, Chris Mylonas wrote: > > I removed -Werror from the Makefile options line "SWITCH_AM_CFLAGS" and it > build successfully. > > > On 14/05/2012, at 3:03 PM, Chris Mylonas wrote: > > Hi FS Users, > > I was building FS on a box to test out the mod_gsmopen stuff that has been > mentioned on the list recently. > Updated CentOS 5 this morning, checked out a new git head. > > Getting this message that says "warnings being treated as errors" > Not sure if that is what is killing the build or if there genuinely is an > error. > > Any help would be great, > Cheers > Chris > > > making all mod_xml_scgi > Compiling ../../../../libs/libscgi/src/scgi.c... > cc1: warnings being treated as errors > ../../../../libs/libscgi/src/scgi.c: In function ?scgi_build_message?: > ../../../../libs/libscgi/src/scgi.c:94: warning: format ?%ld? expects type > ?long int?, but argument 4 has type ?size_t? > ../../../../libs/libscgi/src/scgi.c:94: warning: format ?%ld? expects type > ?long int?, but argument 4 has type ?size_t? > ../../../../libs/libscgi/src/scgi.c:103: warning: format ?%ld? expects > type ?long int?, but argument 4 has type ?size_t? > ../../../../libs/libscgi/src/scgi.c:103: warning: format ?%ld? expects > type ?long int?, but argument 4 has type ?size_t? > ../../../../libs/libscgi/src/scgi.c:110: warning: format ?%ld? expects > type ?long int?, but argument 4 has type ?size_t? > ../../../../libs/libscgi/src/scgi.c:110: warning: format ?%ld? expects > type ?long int?, but argument 4 has type ?size_t? > make[5]: *** [../../../../libs/libscgi/src/scgi.o] Error 1 > make[4]: *** [all] Error 1 > make[3]: *** [mod_xml_scgi-all] Error 1 > make[2]: *** [all-recursive] Error 1 > make[1]: *** [all-recursive] Error 1 > make: *** [all] Error 2 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-800-7876 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120514/7a47b741/attachment.html From peter.olsson at visionutveckling.se Mon May 14 20:20:19 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Mon, 14 May 2012 16:20:19 +0000 Subject: [Freeswitch-users] joining a bridged call to a conference In-Reply-To: References: , Message-ID: Something like uuid_transfer uuid -both conference MyConfRoom inline. /Peter 14 maj 2012 kl. 16:23 skrev "Tihomir Culjaga" >: yup, i was missing -both :=) next question if i may... say i do it all via event socket... originate user/1002 &conference(myConfRoom) how can i transfer an established call ( A-Leg + B-Leg) to that conference ? uuid_transfer uuid -both WHERE?? On Mon, May 14, 2012 at 7:40 AM, Peter Olsson > wrote: I think "uuid_transfer -both" should work. If it's not working, there might be some kind of media negotiation issue (for instance if bypass media is used). /Peter 14 maj 2012 kl. 01:01 skrev "Tihomir Culjaga" >: > hello, > > i got an established call A calling B. A is a number outside FS (any number e.g. 16084191500), B is an extension registered to FS (e.g. 1002). > > after some time, i would like to join the call A => B to a conference roon e.g. 3001 (of course using ESL) > > any idea how to do it smooth ? > whats the best practice for this ? > > i tried uuid_transfer but whatever i do it keeps killing one leg... > > > thanks for help, > T. > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > !DSPAM:4fb03a3f32761890320954! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4fb1124c32761694115387! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4fb1124c32761694115387! From chris at opencsta.org Mon May 14 20:31:47 2012 From: chris at opencsta.org (Chris Mylonas) Date: Tue, 15 May 2012 02:31:47 +1000 Subject: [Freeswitch-users] mod_gsmopen requires spandsp In-Reply-To: References: <1DE50890-EE01-4EEE-83A8-5B5F71BB3959@opencsta.org> <3484676B-F4CA-47E3-A8B0-9743291C1056@opencsta.org> <62D399FB-DFA0-4DD0-B86F-45E95026EF82@opencsta.org> Message-ID: Hi FS List, Sorry to badger you with this again. I'm going to sleep, it's 2:30am - but I will leave you with more mod_gsmopen testing stuff. Getting this in my logs 2012-05-15 01:46:11.670033 [CRIT] switch_loadable_module.c:1300 Error Loading module /usr/local/freeswitch/mod/mod_gsmopen.so **libctb-0.16.so: cannot open shared object file: No such file or directory** It did compile and install and I am running as root. [root at space log]# ls -l /usr/local/freeswitch/mod/mod_gsmopen.so -rwxr-xr-x 1 root root 452284 May 15 01:38 /usr/local/freeswitch/mod/mod_gsmopen.so [root at space log]# ls -l /usr/local/lib/libctb-0.16.so -rwxr-xr-x 1 root root 47525 May 15 01:38 /usr/local/lib/libctb-0.16.so Just for completeness, Configuration file is from src example, modified only with the USB device And the /dev/usb stuff look like this (usbdev1.4_ep__ are the Huawei E173 dongle) - the config file is just my random stab in the dark at getting something going. /dev/usbdev1.1_ep00 /dev/usbdev1.1_ep81 /dev/usbdev1.4_ep00 /dev/usbdev1.4_ep01 /dev/usbdev1.4_ep02 /dev/usbdev1.4_ep81 /dev/usbdev1.4_ep82 /dev/usbdev2.1_ep00 /dev/usbdev2.1_ep81 Thanks Chris On 15/05/2012, at 12:51 AM, Brian Foster wrote: > if you compile libctb with debug it's actually libctbd, not libctb. > > On Mon, May 14, 2012 at 7:53 AM, Chris Mylonas wrote: > Thanks for the reply Giovanni. I have done the steps on the wiki. > > tl;dr; should i put a prefix when i'm making the dependent mods so they don't go into /usr/local/lib, or this is why ldconfig is run - to tell the system where the libs are. > > All the relevant stuff is below > > Hope you can see something wrong, > Cheers > Chris > > e.g. > here is my bash history > > 1050 cd freeswitch/ > 1051 ls > 1052 find . -name gsmlib > 1053 cd src/mod/endpoints/mod_gsmopen/ > 1054 ls > 1055 cd gsmlib/ > 1056 ls > 1057 cd gsmlib-1.10-patched-13ubuntu/ > 1058 ls > 1059 ./configure > 1060 make > 1061 make install > 1062 ldconfig > 1063 cd /usr/src/freeswitch/src/mod/endpoints/mod_gsmopen/libctb-0.16/build > 1064 make DEBUG=1 GPIB=0 > 1065 make DEBUG=1 GPIB=0 install > 1066 ldconfig > 1067 cd /usr/src/freeswitch/src/mod/endpoints/mod_gsmopen/ > 1068 make clean > 1069 make install > > Just to repeat the compilation error > Creating mod_gsmopen.so... > /usr/bin/ld: cannot find -lctb-0.16 > collect2: ld returned 1 exit status > g++ -I../../../../libs/spandsp/src -I../../../..//libs/tiff-3.8.2/libtiff -DGSMOPEN_C_VER=\"44fd0de\" -DMODGSMOPEN_C_VER=\"44fd0de\" -I/usr/src/freeswitch/libs/curl/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -shared -o .libs/mod_gsmopen.so -shared -Wl,-x .libs/mod_gsmopen.o gsmopen_protocol.o /usr/src/freeswitch/.libs/libfreeswitch.so -L/usr/src/freeswitch/libs/apr-util/xml/expat/lib /usr/src/freeswitch/libs/apr-util/xml/expat/lib/.libs/libexpat.a /usr/src/freeswitch/libs/apr/.libs/libapr-1.a -luuid -lpthread -L/usr/src/freeswitch/libs/srtp -L/usr/kerberos/lib -lcrypt -lrt -lssl -lcrypto -ldl -lncurses -L/usr/src/freeswitch/libs/spandsp/src /usr/src/freeswitch/libs/spandsp/src/.libs/libspandsp.a -L/usr/src/freeswitch/libs/tiff-3.8.2/libtiff /usr/src/freeswitch/libs/tiff-3.8.2/libtiff/.libs/libtiff.a -ljpeg -lz -lm -lc -lctb-0.16 -lgsmme -Wl,--rpath -Wl,/usr/local/freeswitch/lib -Wl,--rpath -Wl,/usr/local/freeswitch/mod > make[1]: *** [mod_gsmopen.so] Error 1 > make: *** [install] Error 1 > > > > > All the gsmlib stuff that is NOT in the freeswitch src dir is here > > [root at space mod_gsmopen]# locate gsmlib | grep -v src > /usr/local/include/gsmlib > /usr/local/include/gsmlib/gsm_at.h > /usr/local/include/gsmlib/gsm_cb.h > /usr/local/include/gsmlib/gsm_error.h > /usr/local/include/gsmlib/gsm_event.h > /usr/local/include/gsmlib/gsm_map_key.h > /usr/local/include/gsmlib/gsm_me_ta.h > /usr/local/include/gsmlib/gsm_parser.h > /usr/local/include/gsmlib/gsm_phonebook.h > /usr/local/include/gsmlib/gsm_port.h > /usr/local/include/gsmlib/gsm_sie_me.h > /usr/local/include/gsmlib/gsm_sms.h > /usr/local/include/gsmlib/gsm_sms_codec.h > /usr/local/include/gsmlib/gsm_sms_store.h > /usr/local/include/gsmlib/gsm_sorted_phonebook.h > /usr/local/include/gsmlib/gsm_sorted_phonebook_base.h > /usr/local/include/gsmlib/gsm_sorted_sms_store.h > /usr/local/include/gsmlib/gsm_unix_serial.h > /usr/local/include/gsmlib/gsm_util.h > /usr/local/share/locale/de/LC_MESSAGES/gsmlib.mo > > > And the ctb stuff is in /usr/local/lib > > /usr/local/include/ctb-0.16 > /usr/local/include/ctb-0.16/ctb.h > /usr/local/include/ctb-0.16/fifo.h > /usr/local/include/ctb-0.16/getopt.h > /usr/local/include/ctb-0.16/iobase.h > /usr/local/include/ctb-0.16/linux > /usr/local/include/ctb-0.16/portscan.h > /usr/local/include/ctb-0.16/serport.h > /usr/local/include/ctb-0.16/serportx.h > /usr/local/include/ctb-0.16/timer.h > /usr/local/include/ctb-0.16/linux/serport.h > /usr/local/include/ctb-0.16/linux/timer.h > /usr/local/lib/libctbd-0.16.a > /usr/local/lib/libctbd-0.16.so > > > gcc version 4.1.2 20080704 (Red Hat 4.1.2-52) > GNU Make 3.81 > CentOS release 5.8 (Final) > > > > > On 14/05/2012, at 9:38 PM, Giovanni Maruzzelli wrote: > >> you must first compile and install libctb, as per the wiki page ( >> http://wiki.freeswitch.org/wiki/gsmopen ) >> then, after installation of libctb and gsmlib (as per wiki), be sure >> to update your dinamic link cache, or compiler will not find then. >> >> Eg: ldconfig >> >> >> On 5/14/12, Chris Mylonas wrote: >>> All good GM, no inconvenience, just a minor gotcha ;) >>> I am unable to compile mod_gsmopen though. >>> >>> It complains about not being able to find ctb-0.16 >>> The actual filename is libctbd-0.16.so in /usr/local/lib as you can see from >>> the 2nd lot of stuff. >>> >>> How do I fix this? >>> >>> >>> [root at space mod_gsmopen]# make install >>> Creating mod_gsmopen.so... >>> /usr/bin/ld: cannot find -lctb-0.16 >>> collect2: ld returned 1 exit status >>> g++ -I../../../../libs/spandsp/src -I../../../..//libs/tiff-3.8.2/libtiff >>> -DGSMOPEN_C_VER=\"44fd0de\" -DMODGSMOPEN_C_VER=\"44fd0de\" >>> -I/usr/src/freeswitch/libs/curl/include -I/usr/src/freeswitch/src/include >>> -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/libs/libteletone/src >>> -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g >>> -O2 -D_GNU_SOURCE -shared -o .libs/mod_gsmopen.so -shared -Wl,-x >>> .libs/mod_gsmopen.o gsmopen_protocol.o >>> /usr/src/freeswitch/.libs/libfreeswitch.so >>> -L/usr/src/freeswitch/libs/apr-util/xml/expat/lib >>> /usr/src/freeswitch/libs/apr-util/xml/expat/lib/.libs/libexpat.a >>> /usr/src/freeswitch/libs/apr/.libs/libapr-1.a -luuid -lpthread >>> -L/usr/src/freeswitch/libs/srtp -L/usr/kerberos/lib -lcrypt -lrt -lssl >>> -lcrypto -ldl -lncurses -L/usr/src/freeswitch/libs/spandsp/src >>> /usr/src/freeswitch/libs/spandsp/src/.libs/libspandsp.a >>> -L/usr/src/freeswitch/libs/tiff-3.8.2/libtiff >>> /usr/src/freeswitch/libs/tiff-3.8.2/libtiff/.libs/libtiff.a -ljpeg -lz -lm >>> -lc -lctb-0.16 -lgsmme -Wl,--rpath -Wl,/usr/local/freeswitch/lib >>> -Wl,--rpath -Wl,/usr/local/freeswitch/mod >>> make[1]: *** [mod_gsmopen.so] Error 1 >>> make: *** [install] Error 1 >>> >>> >>> >>> [root at space mod_gsmopen]# ldd /usr/local/lib/libctbd-0.16.so >>> linux-gate.so.1 => (0x00754000) >>> libpthread.so.0 => /lib/libpthread.so.0 (0x00e83000) >>> libstdc++.so.6 => /usr/lib/libstdc++.so.6 (0x00d28000) >>> libm.so.6 => /lib/libm.so.6 (0x00964000) >>> libgcc_s.so.1 => /lib/libgcc_s.so.1 (0x00e1f000) >>> libc.so.6 => /lib/libc.so.6 (0x00110000) >>> /lib/ld-linux.so.2 (0x003ea000) >>> >>> >>> >>> On 14/05/2012, at 6:44 PM, Giovanni Maruzzelli wrote: >>> >>>> yes, it requires libspandsp, maybe the Makefile it's not yet tweaked >>>> to build the library automatically. >>>> >>>> So, please first build mod_spandsp, then mod_gsmopen. >>>> >>>> We'll fixx the Makefile soon, sorry for the inconvenience. >>>> >>>> -giovanni >>>> >>>> On Mon, May 14, 2012 at 9:30 AM, Chris Mylonas >>>> wrote: >>>>> Hi FS List, >>>>> >>>>> FYI - as a shortcut to building my freeswitch, I skip spandsp - but it >>>>> looks >>>>> like this mod_gsmopen wants it in there. >>>>> >>>>> >>>>> [root at space build]# cd >>>>> /usr/src/freeswitch/src/mod/endpoints/mod_gsmopen/ >>>>> [root at space mod_gsmopen]# make clean >>>>> [root at space mod_gsmopen]# make install >>>>> Compiling gsmopen_protocol.cpp... >>>>> Compiling >>>>> /usr/src/freeswitch/src/mod/endpoints/mod_gsmopen/mod_gsmopen.cpp... >>>>> mkdir .libs >>>>> Compiling >>>>> /usr/src/freeswitch/src/mod/endpoints/mod_gsmopen/mod_gsmopen.cpp >>>>> ... >>>>> Creating mod_gsmopen.so... >>>>> /usr/bin/ld: cannot find -lspandsp >>>>> collect2: ld returned 1 exit status >>>>> g++ -I../../../../libs/spandsp/src >>>>> -I../../../..//libs/tiff-3.8.2/libtiff >>>>> -DGSMOPEN_C_VER=\"44fd0de\" -DMODGSMOPEN_C_VER=\"44fd0de\" >>>>> -I/usr/src/freeswitch/libs/curl/include >>>>> -I/usr/src/freeswitch/src/include >>>>> -I/usr/src/freeswitch/src/include >>>>> -I/usr/src/freeswitch/libs/libteletone/src >>>>> -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 >>>>> -g >>>>> -O2 -D_GNU_SOURCE -shared -o .libs/mod_gsmopen.so -shared -Wl,-x >>>>> .libs/mod_gsmopen.o gsmopen_protocol.o -lm >>>>> /usr/src/freeswitch/.libs/libfreeswitch.so >>>>> -L/usr/src/freeswitch/libs/apr-util/xml/expat/lib >>>>> /usr/src/freeswitch/libs/apr-util/xml/expat/lib/.libs/libexpat.a >>>>> /usr/src/freeswitch/libs/apr/.libs/libapr-1.a -luuid -lpthread >>>>> -L/usr/src/freeswitch/libs/srtp -L/usr/kerberos/lib -lcrypt -lrt -lssl >>>>> -lcrypto -ldl -lz -lncurses -ljpeg >>>>> -L/usr/src/freeswitch/libs/spandsp/src >>>>> -lspandsp -lctb-0.16 -lgsmme -Wl,--rpath -Wl,/usr/local/freeswitch/lib >>>>> -Wl,--rpath -Wl,/usr/local/freeswitch/mod >>>>> make[1]: *** [mod_gsmopen.so] Error 1 >>>>> make: *** [install] Error 1 >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> >>>> -- >>>> Sincerely, >>>> >>>> Giovanni Maruzzelli >>>> Cell : +39-347-2665618 >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Sincerely, >> >> Giovanni Maruzzelli >> Cell : +39-347-2665618 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Brian D. Foster > Endigo Computer LLC > Email: bdfoster at endigotech.com > Phone: 317-800-7876 > Indianapolis, Indiana, USA > > This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120515/f9237d4e/attachment-0001.html From chris at opencsta.org Mon May 14 20:33:22 2012 From: chris at opencsta.org (Chris Mylonas) Date: Tue, 15 May 2012 02:33:22 +1000 Subject: [Freeswitch-users] Huawei E169 Group Buy In-Reply-To: References: <4FAF49A5.5000203@coppice.org> <023801cd31e5$10cf3820$326da860$@com> Message-ID: <90D7665C-274B-4DD1-BC8D-A9B97278A005@opencsta.org> E173 has an external antenna jack On 15/05/2012, at 2:15 AM, Brian Foster wrote: > ...which renders the E169 useless for data but good for GSM so voice/SMS will work. I did look for one that was compatible on the data and GSM side of things but I'm turning up with nothing. Maybe there is a Huawei modem out there that either A) is E1550 compatible (but isn't on Giovanni's list) or B) support can be added fairly easily for it. 4G (LTS) is pretty much out of the question from what I can see as well, 3G (HSPA) would be a good starting point. > > So, recap: > > - Huawei Modem that is E1550 compatible (or support can be added) > - Voice Enabled > - UTMS on 850/1900 > - GSM on 850/1900 > - Unlocked (or unlockable) > *Bonus Points: external antenna jack > > I'll keep searching around. I've got a friend who's also trying to help me out looking through this stuff (a lot more experienced than I am) but if anyone's got something better please feel free to share. > > -BDF > > > On Mon, May 14, 2012 at 11:20 AM, Bote Man wrote: > An authoritative source of mine at AT&T Mobility said: > > > > HSDPA UMTS (850, 1900MHz) . GSM GPRS EDGE (850 , 1900MHz) > > > > are supported, which looks like his edited version of the E169 data sheet that I e-mailed to him. > > > > Bote > > > > > > > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian Foster > Sent: Sunday, 13 May, 2012 02:39 > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Huawei E169 Group Buy > > > > The Huawei E169 is confirmed to work with mod_gsmopen. Special thanks to Marcus Brown (mzb- on #freedoh) for helping me figure all of this stuff out and for donating a E169, a SIM card, and a virtual machine to run tests. > > > > Steve, understood. As far as I can tell the bands that the E169 supports SHOULD cover most of the US carriers that support GSM. Like I said in my last email, it is up to you guys to make sure that it will work in your situation. If it doesn't I'd sell it on ebay, I don't think you'll have a problem getting rid of it. > > From: http://3g-modem.wetpaint.com/page/Huawei+E169+(E169G,+E169V,+K3520) > > > > specifications > > Huawei E169 > > manufacturer: > > Huawei > > model: > > E169 > > interface: > > USB 2.0 (A plug) > > GSM frequency bands: > > 850, 900, 1800, 1900 > > UMTS frequency bands: > > 900, 2100 > > HSDPA: > > 7,2 MBit/s > > HSUPA: > > - > > EDGE: > > 236,8 KBit/s > > GPRS: > > 57,6 KBit//s > > CSD: > > ? > > connector for ext. antenna: > > + (on some versions covered by housing) > > connector type: > > SMK CRS5001 > > internal antenna diversity: > > + > > voice telephony: > > + > > NAND-flashmemory: > > + > > microSD-drive: > > + (up to 8GB) > > rebranded versions: > > Vodafone K3520 > Vodafone E169V > > Also: http://www.alibaba.com/showroom/huawei-e169-modem.html > > > > -BDF > > > > On Sun, May 13, 2012 at 1:41 AM, Steve Underwood wrote: > > Hi, > > When looking at models of 3G dongles look carefully at whether they > support the features you need. This basically comes down to supporting > the voice features, and supporting the bands you need in your area. > > Most recent dongles with a Qualcomm chip set (which is most of them) > support the voice features needed to work with FS. However, many SIM > locked ones have the feature blocked. Unlocked ones should be OK. > > Most dongles now support quad band 2G operation, which covers most of > the frequencies ever used for 2G GSM. However, most dongles only support > one 3G band. Some support 2 bands. There are, however, several bands > used for 3G around the planet. If the dongle you choose doesn't support > your local 3G bands, you will only be able to use it for 2G > communication. That might be OK. It might not. It depends on your local > service offerings. Just beware. > > Steve > > > On 05/13/2012 09:49 AM, Brian Foster wrote: > > > > Fellow freeswitchers, > > > > I am sending this email out to let you all know that I am setting up a > > group buy for the Huawei E169. It should be compatable with > > mod_gsmopen, and that will be tested before we go and do a buy. The > > reason why there is interest in this particular model is the fact that > > it does have an external antenna jack. Otherwise they are fairly close > > to the E1550. > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Brian D. Foster > Endigo Computer LLC > Email: bdfoster at endigotech.com > Phone: 317-800-7876 > Indianapolis, Indiana, USA > > This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120515/f8f1f444/attachment-0001.html From chris at opencsta.org Mon May 14 20:41:17 2012 From: chris at opencsta.org (Chris Mylonas) Date: Tue, 15 May 2012 02:41:17 +1000 Subject: [Freeswitch-users] mod_gsmopen requires spandsp In-Reply-To: <802F385B-C86D-499C-ACFD-294A94964EA4@mgtech.com> References: <802F385B-C86D-499C-ACFD-294A94964EA4@mgtech.com> Message-ID: <926060AE-E7D1-4D31-B39D-FA5ACC4920E0@opencsta.org> I reckon a "SpanDSP provides lots of stuff" sentence would be good somewhere on the wiki, not specifically on the OSX install page, just saying. I've got 10.6 installed with Xcode Git only will run with sudo using the git d/l from the google code site. xcode didn't install any, but again - i went with the defaults off the dvd. I'm doing the "minimum" to get a base install going so I know what relies on what. I'll send you what gets me through tomorrow (..if I get through) Thanks for the replies - 2:40am - might call it a day CM On 15/05/2012, at 2:16 AM, Mario G wrote: > Although the osX FS install wiki already says you need SPANDSP I will make it clearer when I update it (the wiki). > Mario G > > On May 14, 2012, at 9:10 AM, Chris Mylonas wrote: > >> Noted. I had no idea. Now it makes Steve's initial question on why anyone would want to leave it out make a whole lot more sense. >> Apologies & Respect! >> >> Cheers >> Chris >> >> On 15/05/2012, at 1:57 AM, Ken Rice wrote: >> >>> Spandsp isnt just about Faxing and T38... Spandsp does that and much >>> more... Its where we get codecs like G711, g722, g726, GSM, LPC-10, and >>> ADPCM... >>> >>> K >>> >>> >>> On 5/14/12 10:51 AM, "Chris Mylonas" wrote: >>> >>>> i've never heard of sp & sp - what is this sp & sp for anyway :P >>>> there's a moderately good reason BTW. >>>> >>>> >>>> the inspiration came from installing on osx last week and missing libjpeg >>>> because "spandsp needed it" - so to shortcut my way through, i didn't install >>>> libjpeg (nor spandsp). >>>> it didn't build, and i installed libjpeg. >>>> it built. still minus spandsp. didn't initiate a call, so didn't come across >>>> the codec problem. >>>> and now that i'm building on linux, i've brought my bad osx habits with me... >>>> >>>> >>>> i'm just used to having a standalone fax machine >>>> don't shoot the t.38/FoIP luddite :) >>>> >>>> Cheers >>>> Chris >>>> >>>> On 15/05/2012, at 1:20 AM, Steve Underwood wrote: >>>> >>>>> On 05/14/2012 10:54 PM, Ken Rice wrote: >>>>>> You don?t want to skip mod_spandsp ... You really need it... Its as >>>>>> much of a core module as anything... Without it you are missing many >>>>>> many codecs you want... >>>>> I wonder what inspires this "Oh, that looks important. Let's leave it >>>>> out of the build" way of looking at thing? :-\ >>>>> >>>>> Steve >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at 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 at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at 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 at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From bdfoster at endigotech.com Mon May 14 20:42:00 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Mon, 14 May 2012 12:42:00 -0400 Subject: [Freeswitch-users] Huawei E169 Group Buy In-Reply-To: <90D7665C-274B-4DD1-BC8D-A9B97278A005@opencsta.org> References: <4FAF49A5.5000203@coppice.org> <023801cd31e5$10cf3820$326da860$@com> <90D7665C-274B-4DD1-BC8D-A9B97278A005@opencsta.org> Message-ID: So does the E169. On Mon, May 14, 2012 at 12:33 PM, Chris Mylonas wrote: > E173 has an external antenna jack > > > On 15/05/2012, at 2:15 AM, Brian Foster wrote: > > ...which renders the E169 useless for data but good for GSM so voice/SMS > will work. I did look for one that was compatible on the data and GSM side > of things but I'm turning up with nothing. Maybe there is a Huawei modem > out there that either A) is E1550 compatible (but isn't on Giovanni's list) > or B) support can be added fairly easily for it. 4G (LTS) is pretty much > out of the question from what I can see as well, 3G (HSPA) would be a good > starting point. > > So, recap: > > - Huawei Modem that is E1550 compatible (or support can be added) > - Voice Enabled > - UTMS on 850/1900 > - GSM on 850/1900 > - Unlocked (or unlockable) > *Bonus Points: external antenna jack > > I'll keep searching around. I've got a friend who's also trying to help me > out looking through this stuff (a lot more experienced than I am) but if > anyone's got something better please feel free to share. > > -BDF > > > On Mon, May 14, 2012 at 11:20 AM, Bote Man wrote: > >> An authoritative source of mine at AT&T Mobility said:**** >> >> ** ** >> >> HSDPA UMTS (850, 1900MHz) . GSM GPRS EDGE (850 , 1900MHz)**** >> >> ** ** >> >> are supported, which looks like his edited version of the E169 data sheet >> that I e-mailed to him.**** >> >> ** ** >> >> Bote**** >> >> ** ** >> >> ** ** >> >> ** ** >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Brian >> Foster >> *Sent:* Sunday, 13 May, 2012 02:39 >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] Huawei E169 Group Buy**** >> >> ** ** >> >> The Huawei E169 is confirmed to work with mod_gsmopen. Special thanks to >> Marcus Brown (mzb- on #freedoh) for helping me figure all of this stuff out >> and for donating a E169, a SIM card, and a virtual machine to run tests. >> **** >> >> ** ** >> >> Steve, understood. As far as I can tell the bands that the E169 supports >> SHOULD cover most of the US carriers that support GSM. Like I said in my >> last email, it is up to you guys to make sure that it will work in your >> situation. If it doesn't I'd sell it on ebay, I don't think you'll have a >> problem getting rid of it.**** >> >> From: http://3g-modem.wetpaint.com/page/Huawei+E169+(E169G,+E169V,+K3520) >> **** >> >> ** ** >> >> specifications >> >> Huawei E169**** >> >> manufacturer:**** >> >> Huawei**** >> >> model:**** >> >> E169**** >> >> interface:**** >> >> USB 2.0 (A plug)**** >> >> GSM frequency bands:**** >> >> 850, 900, 1800, 1900**** >> >> UMTS frequency bands:**** >> >> 900, 2100**** >> >> HSDPA:**** >> >> 7,2 MBit/s**** >> >> HSUPA:**** >> >> -**** >> >> EDGE:**** >> >> 236,8 KBit/s**** >> >> GPRS:**** >> >> 57,6 KBit//s**** >> >> CSD:**** >> >> ?**** >> >> connector for ext. antenna:**** >> >> + (on some versions covered by housing)**** >> >> connector type:**** >> >> SMK CRS5001**** >> >> internal antenna diversity:**** >> >> +**** >> >> voice telephony:**** >> >> +**** >> >> NAND-flashmemory:**** >> >> +**** >> >> microSD-drive:**** >> >> + (up to 8GB)**** >> >> rebranded versions:**** >> >> Vodafone K3520 >> Vodafone E169V**** >> >> Also: http://www.alibaba.com/showroom/huawei-e169-modem.html**** >> >> ** ** >> >> -BDF**** >> >> ** ** >> >> On Sun, May 13, 2012 at 1:41 AM, Steve Underwood >> wrote:**** >> >> Hi, >> >> When looking at models of 3G dongles look carefully at whether they >> support the features you need. This basically comes down to supporting >> the voice features, and supporting the bands you need in your area. >> >> Most recent dongles with a Qualcomm chip set (which is most of them) >> support the voice features needed to work with FS. However, many SIM >> locked ones have the feature blocked. Unlocked ones should be OK. >> >> Most dongles now support quad band 2G operation, which covers most of >> the frequencies ever used for 2G GSM. However, most dongles only support >> one 3G band. Some support 2 bands. There are, however, several bands >> used for 3G around the planet. If the dongle you choose doesn't support >> your local 3G bands, you will only be able to use it for 2G >> communication. That might be OK. It might not. It depends on your local >> service offerings. Just beware. >> >> Steve**** >> >> >> On 05/13/2012 09:49 AM, Brian Foster wrote: >> > >> > Fellow freeswitchers, >> > >> > I am sending this email out to let you all know that I am setting up a >> > group buy for the Huawei E169. It should be compatable with >> > mod_gsmopen, and that will be tested before we go and do a buy. The >> > reason why there is interest in this particular model is the fact that >> > it does have an external antenna jack. Otherwise they are fairly close >> > to the E1550. >> > >> >> **** >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Brian D. Foster > Endigo Computer LLC > Email: bdfoster at endigotech.com > Phone: 317-800-7876 > Indianapolis, Indiana, USA > > This message contains confidential information and is intended for those > listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If > you are not the intended recipient you are notified that disclosing, > copying, distributing or taking any action in reliance on the contents of > this information is strictly prohibited. E-mail transmission cannot be > guaranteed to be secure or error-free as information could be intercepted, > corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. > The sender therefore does not accept liability for any errors or omissions > in the contents of this message, which arise as a result of e-mail > transmission. If verification is required please request a hard-copy > version. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-800-7876 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120514/71d8d111/attachment-0001.html From sarahig1985 at gmail.com Mon May 14 20:47:36 2012 From: sarahig1985 at gmail.com (Sara Higfler) Date: Mon, 14 May 2012 17:47:36 +0100 Subject: [Freeswitch-users] Javascript Outbound Event Socket - Event Locking Message-ID: Hi, I'm continuing to make progress with the development of some dialplan applications using the Outbound Javascript Event Socket Library. The new challenge I've encountered is (I believe) related to event locking. I'm using the Outbound event socket in async mode. A simplified example of what I want to do is to play an announcement, wait for it to finish, then hangup the channel. Announcements are working fine in other parts of the application, but I'm having trouble forcing the announcement to play before the channel is disconnected. I believe this can be solved by activating event locking per: http://wiki.freeswitch.org/wiki/Event_Socket_Library#setEventLock However, I cannot find an example of this being implemented within javascript... I thought there would be a means of appending it to the execute command, but have been unable to determine how. req.execute('playback', 'conference/conf-enter_conf_number.wav'); Any help would be sincerely appreciated. I'm in the process of writing a guide describing basic examples of using Javascript with outbound event sockets. I'll add this to the list of hints for the newbies like myself once I've got to the bottom of it. Kind regards, Sara. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120514/8d02fe50/attachment.html From steveu at coppice.org Mon May 14 20:53:09 2012 From: steveu at coppice.org (Steve Underwood) Date: Tue, 15 May 2012 00:53:09 +0800 Subject: [Freeswitch-users] Huawei E169 Group Buy In-Reply-To: References: <4FAF49A5.5000203@coppice.org> <023801cd31e5$10cf3820$326da860$@com> <90D7665C-274B-4DD1-BC8D-A9B97278A005@opencsta.org> Message-ID: <4FB13875.4040907@coppice.org> But the E173 supports the US bands. Its looks like it just the US version of the E169. Steve On 05/15/2012 12:42 AM, Brian Foster wrote: > So does the E169. > > On Mon, May 14, 2012 at 12:33 PM, Chris Mylonas > wrote: > > E173 has an external antenna jack > > > On 15/05/2012, at 2:15 AM, Brian Foster wrote: > >> ...which renders the E169 useless for data but good for GSM so >> voice/SMS will work. I did look for one that was compatible on >> the data and GSM side of things but I'm turning up with nothing. >> Maybe there is a Huawei modem out there that either A) is E1550 >> compatible (but isn't on Giovanni's list) or B) support can be >> added fairly easily for it. 4G (LTS) is pretty much out of the >> question from what I can see as well, 3G (HSPA) would be a good >> starting point. >> >> So, recap: >> >> - Huawei Modem that is E1550 compatible (or support can be added) >> - Voice Enabled >> - UTMS on 850/1900 >> - GSM on 850/1900 >> - Unlocked (or unlockable) >> *Bonus Points: external antenna jack >> >> I'll keep searching around. I've got a friend who's also trying >> to help me out looking through this stuff (a lot more experienced >> than I am) but if anyone's got something better please feel free >> to share. >> >> -BDF >> >> >> On Mon, May 14, 2012 at 11:20 AM, Bote Man >> > wrote: >> >> An authoritative source of mine at AT&T Mobility said: >> >> HSDPA UMTS (850, 1900MHz) . GSM GPRS EDGE (850 , 1900MHz) >> >> are supported, which looks like his edited version of the >> E169 data sheet that I e-mailed to him. >> >> Bote >> >> *From:*freeswitch-users-bounces at lists.freeswitch.org >> >> [mailto:freeswitch-users-bounces at lists.freeswitch.org >> ] *On >> Behalf Of *Brian Foster >> *Sent:* Sunday, 13 May, 2012 02:39 >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] Huawei E169 Group Buy >> >> The Huawei E169 is confirmed to work with mod_gsmopen. >> Special thanks to Marcus Brown (mzb- on #freedoh) for helping >> me figure all of this stuff out and for donating a E169, a >> SIM card, and a virtual machine to run tests. >> >> Steve, understood. As far as I can tell the bands that the >> E169 supports SHOULD cover most of the US carriers that >> support GSM. Like I said in my last email, it is up to you >> guys to make sure that it will work in your situation. If it >> doesn't I'd sell it on ebay, I don't think you'll have a >> problem getting rid of it. >> >> From: >> http://3g-modem.wetpaint.com/page/Huawei+E169+(E169G,+E169V,+K3520) >> >> >> specifications >> >> Huawei E169 >> >> manufacturer: >> >> >> >> Huawei >> >> model: >> >> >> >> E169 >> >> interface: >> >> >> >> USB 2.0 (A plug) >> >> GSM frequency bands: >> >> >> >> 850, 900, 1800, 1900 >> >> UMTS frequency bands: >> >> >> >> 900, 2100 >> >> HSDPA: >> >> >> >> 7,2 MBit/s >> >> HSUPA: >> >> >> >> - >> >> EDGE: >> >> >> >> 236,8 KBit/s >> >> GPRS: >> >> >> >> 57,6 KBit//s >> >> CSD: >> >> >> >> ? >> >> connector for ext. antenna: >> >> >> >> + (on some versions covered by housing) >> >> connector type: >> >> >> >> SMK CRS5001 >> >> internal antenna diversity: >> >> >> >> + >> >> voice telephony: >> >> >> >> + >> >> NAND-flashmemory: >> >> >> >> + >> >> microSD-drive: >> >> >> >> + (up to 8GB) >> >> rebranded versions: >> >> >> >> Vodafone K3520 >> Vodafone E169V >> >> Also: http://www.alibaba.com/showroom/huawei-e169-modem.html >> >> -BDF >> >> On Sun, May 13, 2012 at 1:41 AM, Steve Underwood >> > wrote: >> >> Hi, >> >> When looking at models of 3G dongles look carefully at >> whether they >> support the features you need. This basically comes down to >> supporting >> the voice features, and supporting the bands you need in your >> area. >> >> Most recent dongles with a Qualcomm chip set (which is most >> of them) >> support the voice features needed to work with FS. However, >> many SIM >> locked ones have the feature blocked. Unlocked ones should be OK. >> >> Most dongles now support quad band 2G operation, which covers >> most of >> the frequencies ever used for 2G GSM. However, most dongles >> only support >> one 3G band. Some support 2 bands. There are, however, >> several bands >> used for 3G around the planet. If the dongle you choose >> doesn't support >> your local 3G bands, you will only be able to use it for 2G >> communication. That might be OK. It might not. It depends on >> your local >> service offerings. Just beware. >> >> Steve >> >> >> On 05/13/2012 09:49 AM, Brian Foster wrote: >> > >> > Fellow freeswitchers, >> > >> > I am sending this email out to let you all know that I am >> setting up a >> > group buy for the Huawei E169. It should be compatable with >> > mod_gsmopen, and that will be tested before we go and do a >> buy. The >> > reason why there is interest in this particular model is >> the fact that >> > it does have an external antenna jack. Otherwise they are >> fairly close >> > to the E1550. >> > >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> -- >> Brian D. Foster >> Endigo Computer LLC >> Email: bdfoster at endigotech.com >> Phone: 317-800-7876 >> Indianapolis, Indiana, USA >> >> This message contains confidential information and is intended >> for those listed in the "To:", "CC:", and/or "BCC:" fields of the >> message header. If you are not the intended recipient you are >> notified that disclosing, copying, distributing or taking any >> action in reliance on the contents of this information is >> strictly prohibited. E-mail transmission cannot be guaranteed to >> be secure or error-free as information could be intercepted, >> corrupted, lost, destroyed, arrive late or incomplete, or contain >> viruses. The sender therefore does not accept liability for any >> errors or omissions in the contents of this message, which arise >> as a result of e-mail transmission. If verification is required >> please request a hard-copy version. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Brian D. Foster > Endigo Computer LLC > Email: bdfoster at endigotech.com > Phone: 317-800-7876 > Indianapolis, Indiana, USA > > This message contains confidential information and is intended for > those listed in the "To:", "CC:", and/or "BCC:" fields of the message > header. If you are not the intended recipient you are notified that > disclosing, copying, distributing or taking any action in reliance on > the contents of this information is strictly prohibited. E-mail > transmission cannot be guaranteed to be secure or error-free as > information could be intercepted, corrupted, lost, destroyed, arrive > late or incomplete, or contain viruses. The sender therefore does not > accept liability for any errors or omissions in the contents of this > message, which arise as a result of e-mail transmission. If > verification is required please request a hard-copy version. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From bdfoster at endigotech.com Mon May 14 20:59:33 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Mon, 14 May 2012 12:59:33 -0400 Subject: [Freeswitch-users] Huawei E169 Group Buy In-Reply-To: <4FB13875.4040907@coppice.org> References: <4FAF49A5.5000203@coppice.org> <023801cd31e5$10cf3820$326da860$@com> <90D7665C-274B-4DD1-BC8D-A9B97278A005@opencsta.org> <4FB13875.4040907@coppice.org> Message-ID: Steve, *manufacturer:**Huawei**Huawei**Huawei**Huawei**model:**E173u-1**E173u-2** E173s-1**E173s-2*interface:USB 2.0 (A plug)USB 2.0 (A plug)USB 2.0 (A plug)USB 2.0 (A plug)GSM frequency bands:850, 900, 1800, 1900850, 900, 1800, 1900850, 900, 1800, 1900850, 900, 1800, 1900UMTS frequency bands:2100900, 21002100900, 2100HSDPA:7.2 Mbps7.2 Mbps7.2 Mb/s7.2 Mb/sHSUPA:5,76 Mbit/s5,76 Mbit/s1.92 Mb/s2 Mb/s EDGE:236,8 KBit/s236,8 KBit/s236,8 KBit/s236,8 KBit/sGPRS:57,6 KBit//s57,6 KBit//s57,6 KBit//s57,6 KBit//sCSD:????connector for ext. antenna:noyesno* / yes**yesconnector type:-CRC9-* / CRC9**CRC9internal antenna diversity:????voice telephony:?yes??NAND-flashmemory:++ (96 MB)+ (32 MB)+microSD-drive:++++rebranded versions: Play Online (PL)*) T-Mobile.hr, **) BlueconnectOrange.pl T-Mobile.pl They look fairly similar. UTMS wouldn't be covered in either case. -BDF On Mon, May 14, 2012 at 12:53 PM, Steve Underwood wrote: > But the E173 supports the US bands. Its looks like it just the US > version of the E169. > > Steve > > > On 05/15/2012 12:42 AM, Brian Foster wrote: > > So does the E169. > > > > On Mon, May 14, 2012 at 12:33 PM, Chris Mylonas > > wrote: > > > > E173 has an external antenna jack > > > > > > On 15/05/2012, at 2:15 AM, Brian Foster wrote: > > > >> ...which renders the E169 useless for data but good for GSM so > >> voice/SMS will work. I did look for one that was compatible on > >> the data and GSM side of things but I'm turning up with nothing. > >> Maybe there is a Huawei modem out there that either A) is E1550 > >> compatible (but isn't on Giovanni's list) or B) support can be > >> added fairly easily for it. 4G (LTS) is pretty much out of the > >> question from what I can see as well, 3G (HSPA) would be a good > >> starting point. > >> > >> So, recap: > >> > >> - Huawei Modem that is E1550 compatible (or support can be added) > >> - Voice Enabled > >> - UTMS on 850/1900 > >> - GSM on 850/1900 > >> - Unlocked (or unlockable) > >> *Bonus Points: external antenna jack > >> > >> I'll keep searching around. I've got a friend who's also trying > >> to help me out looking through this stuff (a lot more experienced > >> than I am) but if anyone's got something better please feel free > >> to share. > >> > >> -BDF > >> > >> > >> On Mon, May 14, 2012 at 11:20 AM, Bote Man > >> > wrote: > >> > >> An authoritative source of mine at AT&T Mobility said: > >> > >> HSDPA UMTS (850, 1900MHz) . GSM GPRS EDGE (850 , 1900MHz) > >> > >> are supported, which looks like his edited version of the > >> E169 data sheet that I e-mailed to him. > >> > >> Bote > >> > >> *From:*freeswitch-users-bounces at lists.freeswitch.org > >> > >> [mailto:freeswitch-users-bounces at lists.freeswitch.org > >> ] *On > >> Behalf Of *Brian Foster > >> *Sent:* Sunday, 13 May, 2012 02:39 > >> *To:* FreeSWITCH Users Help > >> *Subject:* Re: [Freeswitch-users] Huawei E169 Group Buy > >> > >> The Huawei E169 is confirmed to work with mod_gsmopen. > >> Special thanks to Marcus Brown (mzb- on #freedoh) for helping > >> me figure all of this stuff out and for donating a E169, a > >> SIM card, and a virtual machine to run tests. > >> > >> Steve, understood. As far as I can tell the bands that the > >> E169 supports SHOULD cover most of the US carriers that > >> support GSM. Like I said in my last email, it is up to you > >> guys to make sure that it will work in your situation. If it > >> doesn't I'd sell it on ebay, I don't think you'll have a > >> problem getting rid of it. > >> > >> From: > >> > http://3g-modem.wetpaint.com/page/Huawei+E169+(E169G,+E169V,+K3520) > >> < > http://3g-modem.wetpaint.com/page/Huawei+E169+%28E169G,+E169V,+K3520%29> > >> > >> specifications > >> > >> Huawei E169 > >> > >> manufacturer: > >> > >> > >> > >> Huawei > >> > >> model: > >> > >> > >> > >> E169 > >> > >> interface: > >> > >> > >> > >> USB 2.0 (A plug) > >> > >> GSM frequency bands: > >> > >> > >> > >> 850, 900, 1800, 1900 > >> > >> UMTS frequency bands: > >> > >> > >> > >> 900, 2100 > >> > >> HSDPA: > >> > >> > >> > >> 7,2 MBit/s > >> > >> HSUPA: > >> > >> > >> > >> - > >> > >> EDGE: > >> > >> > >> > >> 236,8 KBit/s > >> > >> GPRS: > >> > >> > >> > >> 57,6 KBit//s > >> > >> CSD: > >> > >> > >> > >> ? > >> > >> connector for ext. antenna: > >> > >> > >> > >> + (on some versions covered by housing) > >> > >> connector type: > >> > >> > >> > >> SMK CRS5001 > >> > >> internal antenna diversity: > >> > >> > >> > >> + > >> > >> voice telephony: > >> > >> > >> > >> + > >> > >> NAND-flashmemory: > >> > >> > >> > >> + > >> > >> microSD-drive: > >> > >> > >> > >> + (up to 8GB) > >> > >> rebranded versions: > >> > >> > >> > >> Vodafone K3520 > >> Vodafone E169V > >> > >> Also: http://www.alibaba.com/showroom/huawei-e169-modem.html > >> > >> -BDF > >> > >> On Sun, May 13, 2012 at 1:41 AM, Steve Underwood > >> > wrote: > >> > >> Hi, > >> > >> When looking at models of 3G dongles look carefully at > >> whether they > >> support the features you need. This basically comes down to > >> supporting > >> the voice features, and supporting the bands you need in your > >> area. > >> > >> Most recent dongles with a Qualcomm chip set (which is most > >> of them) > >> support the voice features needed to work with FS. However, > >> many SIM > >> locked ones have the feature blocked. Unlocked ones should be > OK. > >> > >> Most dongles now support quad band 2G operation, which covers > >> most of > >> the frequencies ever used for 2G GSM. However, most dongles > >> only support > >> one 3G band. Some support 2 bands. There are, however, > >> several bands > >> used for 3G around the planet. If the dongle you choose > >> doesn't support > >> your local 3G bands, you will only be able to use it for 2G > >> communication. That might be OK. It might not. It depends on > >> your local > >> service offerings. Just beware. > >> > >> Steve > >> > >> > >> On 05/13/2012 09:49 AM, Brian Foster wrote: > >> > > >> > Fellow freeswitchers, > >> > > >> > I am sending this email out to let you all know that I am > >> setting up a > >> > group buy for the Huawei E169. It should be compatable with > >> > mod_gsmopen, and that will be tested before we go and do a > >> buy. The > >> > reason why there is interest in this particular model is > >> the fact that > >> > it does have an external antenna jack. Otherwise they are > >> fairly close > >> > to the E1550. > >> > > >> > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > >> > >> > >> > >> -- > >> Brian D. Foster > >> Endigo Computer LLC > >> Email: bdfoster at endigotech.com > >> Phone: 317-800-7876 > >> Indianapolis, Indiana, USA > >> > >> This message contains confidential information and is intended > >> for those listed in the "To:", "CC:", and/or "BCC:" fields of the > >> message header. If you are not the intended recipient you are > >> notified that disclosing, copying, distributing or taking any > >> action in reliance on the contents of this information is > >> strictly prohibited. E-mail transmission cannot be guaranteed to > >> be secure or error-free as information could be intercepted, > >> corrupted, lost, destroyed, arrive late or incomplete, or contain > >> viruses. The sender therefore does not accept liability for any > >> errors or omissions in the contents of this message, which arise > >> as a result of e-mail transmission. If verification is required > >> please request a hard-copy version. > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at 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 at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > -- > > Brian D. Foster > > Endigo Computer LLC > > Email: bdfoster at endigotech.com > > Phone: 317-800-7876 > > Indianapolis, Indiana, USA > > > > This message contains confidential information and is intended for > > those listed in the "To:", "CC:", and/or "BCC:" fields of the message > > header. If you are not the intended recipient you are notified that > > disclosing, copying, distributing or taking any action in reliance on > > the contents of this information is strictly prohibited. E-mail > > transmission cannot be guaranteed to be secure or error-free as > > information could be intercepted, corrupted, lost, destroyed, arrive > > late or incomplete, or contain viruses. The sender therefore does not > > accept liability for any errors or omissions in the contents of this > > message, which arise as a result of e-mail transmission. If > > verification is required please request a hard-copy version. > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-800-7876 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120514/94f57152/attachment-0001.html From marketing at cluecon.com Mon May 14 21:11:48 2012 From: marketing at cluecon.com (Michael Collins) Date: Mon, 14 May 2012 10:11:48 -0700 Subject: [Freeswitch-users] FreeSWITCH Weekly News and Notes Message-ID: Greetings from the FreeSWITCH team. We hope your Monday is going well! This week we have an interesting topic in our community conference call. We will be joined by Fred Dixon, lead developer of the Big Blue Buttonproject. Big Blue Button is open software that enables distance learning and remote education. Fred will do a live demonstration of how it works as well as discussing how FreeSWITCH fits into the picture. We invite everyone to join us Wednesday to see this cool project in action. In ClueCon news, we are pleased to announce that we have two more sponsors: NACT and HarQen . NACT uses FreeSWITCH in carrier class solutions, such as pre-paid and their commercial Vinci Class 4/5 softswitch. HarQen provides innovative voice-based solutions in their line of Intelligent Voice Services products.Their flagship offering is Voice Advantage, a tool that allows creation of custom recorded interviews. Managers and recruiters can quickly review these interviews, greatly speeding up the recruiting process. We look forward to seeing representatives from NACT and HarQen at ClueCon this August. As a reminder, our ClueCon PowerPlay is still in effect. Register one attendee by the end of May and you'll receive three entries in our great ClueCon prize giveaway. Register two attendees from the same company and they *each *will receive six entries. The more people from your company who register, the more entries each person receives. Visit the ClueCon siteor call us at 877-742-2583 if you have any questions about attending, sponsoring, or speaking at this year's event. See you this August! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE cc12-0514 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120514/01923940/attachment.html From anthony.minessale at gmail.com Mon May 14 21:18:24 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 14 May 2012 12:18:24 -0500 Subject: [Freeswitch-users] Javascript Outbound Event Socket - Event Locking In-Reply-To: References: Message-ID: you just need the header Event-Lock: true present in the message you send out On Mon, May 14, 2012 at 11:47 AM, Sara Higfler wrote: > Hi, > > I'm continuing to make progress with the development of some dialplan > applications using the Outbound Javascript Event Socket?Library.? The new > challenge I've encountered is (I believe) related to event locking.??I'm > using the Outbound event socket in async mode.? A simplified example of what > I want to do is to play an announcement, wait for it to finish, then hangup > the channel.?? Announcements are working fine in other parts of the > application, but I'm having trouble forcing the announcement to play before > the channel is disconnected.? I believe this can be solved by activating > event locking per: > > http://wiki.freeswitch.org/wiki/Event_Socket_Library#setEventLock > > However, I cannot find an example of this being implemented within > javascript... I thought there would be a means of appending it to the > execute command, but have been unable to determine how. > > ????????????????????? req.execute('playback', > 'conference/conf-enter_conf_number.wav'); > Any help would be sincerely appreciated.? I'm in the process of writing a > guide describing basic examples of using Javascript with outbound event > sockets.? I'll add this to the list of hints for the newbies like myself > once I've got to the bottom of it. > > Kind regards, > Sara. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From wesleyakio at tuntscorp.com Mon May 14 22:04:38 2012 From: wesleyakio at tuntscorp.com (Wesley Akio) Date: Mon, 14 May 2012 15:04:38 -0300 Subject: [Freeswitch-users] mod_gsmopen requires spandsp In-Reply-To: References: <1DE50890-EE01-4EEE-83A8-5B5F71BB3959@opencsta.org> <3484676B-F4CA-47E3-A8B0-9743291C1056@opencsta.org> <62D399FB-DFA0-4DD0-B86F-45E95026EF82@opencsta.org> Message-ID: I got to that at some point(E173s-6)... Wich distro are you in? Anyway, you're probably missing the device drivers. Mine (CentOS 5.8) looked just like that... with the drivers it gives me: crw------- 1 root root 188, 2 Mai 14 15:01 ttyUSB_utps_diag crw------- 1 root root 188, 1 Mai 14 15:01 ttyUSB_utps_modem crw------- 1 root root 188, 3 Mai 14 15:01 ttyUSB_utps_pcui I got SMS working after that. Modem is, of course, the modem... I could'n get audio working though so I have no clue on each of the other two is the RAW device. If you get it to work please let me know! Best, Wesley Akio TuntsCorp.com On Mon, May 14, 2012 at 1:31 PM, Chris Mylonas wrote: > Hi FS List, > > Sorry to badger you with this again. I'm going to sleep, it's 2:30am - > but I will leave you with more mod_gsmopen testing stuff. > > > Getting this in my logs > > 2012-05-15 01:46:11.670033 [CRIT] switch_loadable_module.c:1300 Error > Loading module /usr/local/freeswitch/mod/mod_gsmopen.so > **libctb-0.16.so: cannot open shared object file: No such file or > directory** > > It did compile and install and I am running as root. > > [root at space log]# ls -l /usr/local/freeswitch/mod/mod_gsmopen.so > -rwxr-xr-x 1 root root 452284 May 15 01:38 > /usr/local/freeswitch/mod/mod_gsmopen.so > > [root at space log]# ls -l /usr/local/lib/libctb-0.16.so > -rwxr-xr-x 1 root root 47525 May 15 01:38 /usr/local/lib/libctb-0.16.so > > Just for completeness, > > Configuration file is from src example, modified only with the USB device > > > > > > > > > > > > > > > > > > > > > > > > > And the /dev/usb stuff look like this (usbdev1.4_ep__ are the Huawei E173 > dongle) - the config file is just my random stab in the dark at getting > something going. > > /dev/usbdev1.1_ep00 > /dev/usbdev1.1_ep81 > */dev/usbdev1.4_ep00* > */dev/usbdev1.4_ep01* > */dev/usbdev1.4_ep02* > */dev/usbdev1.4_ep81* > */dev/usbdev1.4_ep82* > /dev/usbdev2.1_ep00 > /dev/usbdev2.1_ep81 > > Thanks > Chris > > > On 15/05/2012, at 12:51 AM, Brian Foster wrote: > > if you compile libctb with debug it's actually libctbd, not libctb. > > On Mon, May 14, 2012 at 7:53 AM, Chris Mylonas wrote: > >> Thanks for the reply Giovanni. I have done the steps on the wiki. >> >> tl;dr; should i put a prefix when i'm making the dependent mods so they >> don't go into /usr/local/lib, or this is why ldconfig is run - to tell the >> system where the libs are. >> >> All the relevant stuff is below >> >> Hope you can see something wrong, >> Cheers >> Chris >> >> e.g. >> here is my bash history >> >> 1050 cd freeswitch/ >> 1051 ls >> 1052 find . -name gsmlib >> 1053 cd src/mod/endpoints/mod_gsmopen/ >> 1054 ls >> 1055 cd gsmlib/ >> 1056 ls >> 1057 cd gsmlib-1.10-patched-13ubuntu/ >> 1058 ls >> * 1059 ./configure* >> * 1060 make* >> * 1061 make install* >> * 1062 ldconfig * >> * 1063 cd >> /usr/src/freeswitch/src/mod/endpoints/mod_gsmopen/libctb-0.16/build* >> * 1064 make DEBUG=1 GPIB=0* >> * 1065 make DEBUG=1 GPIB=0 install* >> * 1066 ldconfig* >> 1067 cd /usr/src/freeswitch/src/mod/endpoints/mod_gsmopen/ >> 1068 make clean >> 1069 make install >> >> Just to repeat the compilation error >> Creating mod_gsmopen.so... >> */usr/bin/ld: cannot find -lctb-0.16* >> collect2: ld returned 1 exit status >> g++ -I../../../../libs/spandsp/src -I../../../..//libs/tiff-3.8.2/libtiff >> -DGSMOPEN_C_VER=\"44fd0de\" -DMODGSMOPEN_C_VER=\"44fd0de\" >> -I/usr/src/freeswitch/libs/curl/include -I/usr/src/freeswitch/src/include >> -I/usr/src/freeswitch/src/include >> -I/usr/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden >> -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -shared >> -o .libs/mod_gsmopen.so -shared -Wl,-x .libs/mod_gsmopen.o >> gsmopen_protocol.o /usr/src/freeswitch/.libs/libfreeswitch.so >> -L/usr/src/freeswitch/libs/apr-util/xml/expat/lib >> /usr/src/freeswitch/libs/apr-util/xml/expat/lib/.libs/libexpat.a >> /usr/src/freeswitch/libs/apr/.libs/libapr-1.a -luuid -lpthread >> -L/usr/src/freeswitch/libs/srtp -L/usr/kerberos/lib -lcrypt -lrt -lssl >> -lcrypto -ldl -lncurses -L/usr/src/freeswitch/libs/spandsp/src >> /usr/src/freeswitch/libs/spandsp/src/.libs/libspandsp.a >> -L/usr/src/freeswitch/libs/tiff-3.8.2/libtiff >> /usr/src/freeswitch/libs/tiff-3.8.2/libtiff/.libs/libtiff.a -ljpeg -lz -lm >> -lc -lctb-0.16 -lgsmme -Wl,--rpath -Wl,/usr/local/freeswitch/lib >> -Wl,--rpath -Wl,/usr/local/freeswitch/mod >> make[1]: *** [mod_gsmopen.so] Error 1 >> make: *** [install] Error 1 >> >> >> >> >> All the gsmlib stuff that is NOT in the freeswitch src dir is here >> >> [root at space mod_gsmopen]# locate gsmlib | grep -v src >> /usr/local/include/gsmlib >> /usr/local/include/gsmlib/gsm_at.h >> /usr/local/include/gsmlib/gsm_cb.h >> /usr/local/include/gsmlib/gsm_error.h >> /usr/local/include/gsmlib/gsm_event.h >> /usr/local/include/gsmlib/gsm_map_key.h >> /usr/local/include/gsmlib/gsm_me_ta.h >> /usr/local/include/gsmlib/gsm_parser.h >> /usr/local/include/gsmlib/gsm_phonebook.h >> /usr/local/include/gsmlib/gsm_port.h >> /usr/local/include/gsmlib/gsm_sie_me.h >> /usr/local/include/gsmlib/gsm_sms.h >> /usr/local/include/gsmlib/gsm_sms_codec.h >> /usr/local/include/gsmlib/gsm_sms_store.h >> /usr/local/include/gsmlib/gsm_sorted_phonebook.h >> /usr/local/include/gsmlib/gsm_sorted_phonebook_base.h >> /usr/local/include/gsmlib/gsm_sorted_sms_store.h >> /usr/local/include/gsmlib/gsm_unix_serial.h >> /usr/local/include/gsmlib/gsm_util.h >> /usr/local/share/locale/de/LC_MESSAGES/gsmlib.mo >> >> >> And the ctb stuff is in /usr/local/lib >> >> /usr/local/include/ctb-0.16 >> /usr/local/include/ctb-0.16/ctb.h >> /usr/local/include/ctb-0.16/fifo.h >> /usr/local/include/ctb-0.16/getopt.h >> /usr/local/include/ctb-0.16/iobase.h >> /usr/local/include/ctb-0.16/linux >> /usr/local/include/ctb-0.16/portscan.h >> /usr/local/include/ctb-0.16/serport.h >> /usr/local/include/ctb-0.16/serportx.h >> /usr/local/include/ctb-0.16/timer.h >> /usr/local/include/ctb-0.16/linux/serport.h >> /usr/local/include/ctb-0.16/linux/timer.h >> /usr/local/lib/libctbd-0.16.a >> /usr/local/lib/libctbd-0.16.so >> >> >> gcc version 4.1.2 20080704 (Red Hat 4.1.2-52) >> GNU Make 3.81 >> CentOS release 5.8 (Final) >> >> >> >> >> On 14/05/2012, at 9:38 PM, Giovanni Maruzzelli wrote: >> >> you must first compile and install libctb, as per the wiki page ( >> http://wiki.freeswitch.org/wiki/gsmopen ) >> then, after installation of libctb and gsmlib (as per wiki), be sure >> to update your dinamic link cache, or compiler will not find then. >> >> Eg: ldconfig >> >> >> On 5/14/12, Chris Mylonas wrote: >> >> All good GM, no inconvenience, just a minor gotcha ;) >> >> I am unable to compile mod_gsmopen though. >> >> >> It complains about not being able to find ctb-0.16 >> >> The actual filename is libctbd-0.16.so in /usr/local/lib as you can see >> from >> >> the 2nd lot of stuff. >> >> >> How do I fix this? >> >> >> >> [root at space mod_gsmopen]# make install >> >> Creating mod_gsmopen.so... >> >> /usr/bin/ld: cannot find -lctb-0.16 >> >> collect2: ld returned 1 exit status >> >> g++ -I../../../../libs/spandsp/src -I../../../..//libs/tiff-3.8.2/libtiff >> >> -DGSMOPEN_C_VER=\"44fd0de\" -DMODGSMOPEN_C_VER=\"44fd0de\" >> >> -I/usr/src/freeswitch/libs/curl/include -I/usr/src/freeswitch/src/include >> >> -I/usr/src/freeswitch/src/include >> -I/usr/src/freeswitch/libs/libteletone/src >> >> -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g >> >> -O2 -D_GNU_SOURCE -shared -o .libs/mod_gsmopen.so -shared -Wl,-x >> >> .libs/mod_gsmopen.o gsmopen_protocol.o >> >> /usr/src/freeswitch/.libs/libfreeswitch.so >> >> -L/usr/src/freeswitch/libs/apr-util/xml/expat/lib >> >> /usr/src/freeswitch/libs/apr-util/xml/expat/lib/.libs/libexpat.a >> >> /usr/src/freeswitch/libs/apr/.libs/libapr-1.a -luuid -lpthread >> >> -L/usr/src/freeswitch/libs/srtp -L/usr/kerberos/lib -lcrypt -lrt -lssl >> >> -lcrypto -ldl -lncurses -L/usr/src/freeswitch/libs/spandsp/src >> >> /usr/src/freeswitch/libs/spandsp/src/.libs/libspandsp.a >> >> -L/usr/src/freeswitch/libs/tiff-3.8.2/libtiff >> >> /usr/src/freeswitch/libs/tiff-3.8.2/libtiff/.libs/libtiff.a -ljpeg -lz -lm >> >> -lc -lctb-0.16 -lgsmme -Wl,--rpath -Wl,/usr/local/freeswitch/lib >> >> -Wl,--rpath -Wl,/usr/local/freeswitch/mod >> >> make[1]: *** [mod_gsmopen.so] Error 1 >> >> make: *** [install] Error 1 >> >> >> >> >> [root at space mod_gsmopen]# ldd /usr/local/lib/libctbd-0.16.so >> >> linux-gate.so.1 => (0x00754000) >> >> libpthread.so.0 => /lib/libpthread.so.0 (0x00e83000) >> >> libstdc++.so.6 => /usr/lib/libstdc++.so.6 (0x00d28000) >> >> libm.so.6 => /lib/libm.so.6 (0x00964000) >> >> libgcc_s.so.1 => /lib/libgcc_s.so.1 (0x00e1f000) >> >> libc.so.6 => /lib/libc.so.6 (0x00110000) >> >> /lib/ld-linux.so.2 (0x003ea000) >> >> >> >> >> On 14/05/2012, at 6:44 PM, Giovanni Maruzzelli wrote: >> >> >> yes, it requires libspandsp, maybe the Makefile it's not yet tweaked >> >> to build the library automatically. >> >> >> So, please first build mod_spandsp, then mod_gsmopen. >> >> >> We'll fixx the Makefile soon, sorry for the inconvenience. >> >> >> -giovanni >> >> >> On Mon, May 14, 2012 at 9:30 AM, Chris Mylonas >> >> wrote: >> >> Hi FS List, >> >> >> FYI - as a shortcut to building my freeswitch, I skip spandsp - but it >> >> looks >> >> like this mod_gsmopen wants it in there. >> >> >> >> [root at space build]# cd >> >> /usr/src/freeswitch/src/mod/endpoints/mod_gsmopen/ >> >> [root at space mod_gsmopen]# make clean >> >> [root at space mod_gsmopen]# make install >> >> Compiling gsmopen_protocol.cpp... >> >> Compiling >> >> /usr/src/freeswitch/src/mod/endpoints/mod_gsmopen/mod_gsmopen.cpp... >> >> mkdir .libs >> >> Compiling >> >> /usr/src/freeswitch/src/mod/endpoints/mod_gsmopen/mod_gsmopen.cpp >> >> ... >> >> Creating mod_gsmopen.so... >> >> /usr/bin/ld: cannot find -lspandsp >> >> collect2: ld returned 1 exit status >> >> g++ -I../../../../libs/spandsp/src >> >> -I../../../..//libs/tiff-3.8.2/libtiff >> >> -DGSMOPEN_C_VER=\"44fd0de\" -DMODGSMOPEN_C_VER=\"44fd0de\" >> >> -I/usr/src/freeswitch/libs/curl/include >> >> -I/usr/src/freeswitch/src/include >> >> -I/usr/src/freeswitch/src/include >> >> -I/usr/src/freeswitch/libs/libteletone/src >> >> -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 >> >> -g >> >> -O2 -D_GNU_SOURCE -shared -o .libs/mod_gsmopen.so -shared -Wl,-x >> >> .libs/mod_gsmopen.o gsmopen_protocol.o -lm >> >> /usr/src/freeswitch/.libs/libfreeswitch.so >> >> -L/usr/src/freeswitch/libs/apr-util/xml/expat/lib >> >> /usr/src/freeswitch/libs/apr-util/xml/expat/lib/.libs/libexpat.a >> >> /usr/src/freeswitch/libs/apr/.libs/libapr-1.a -luuid -lpthread >> >> -L/usr/src/freeswitch/libs/srtp -L/usr/kerberos/lib -lcrypt -lrt -lssl >> >> -lcrypto -ldl -lz -lncurses -ljpeg >> >> -L/usr/src/freeswitch/libs/spandsp/src >> >> -lspandsp -lctb-0.16 -lgsmme -Wl,--rpath -Wl,/usr/local/freeswitch/lib >> >> -Wl,--rpath -Wl,/usr/local/freeswitch/mod >> >> make[1]: *** [mod_gsmopen.so] Error 1 >> >> make: *** [install] Error 1 >> >> >> >> >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> >> >> >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> http://wiki.freeswitch.org >> >> http://www.cluecon.com >> >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> >> >> >> >> >> -- >> >> Sincerely, >> >> >> Giovanni Maruzzelli >> >> Cell : +39-347-2665618 >> >> >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> >> >> >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> http://wiki.freeswitch.org >> >> http://www.cluecon.com >> >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> >> >> >> >> >> -- >> Sincerely, >> >> Giovanni Maruzzelli >> Cell : +39-347-2665618 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Brian D. Foster > Endigo Computer LLC > Email: bdfoster at endigotech.com > Phone: 317-800-7876 > Indianapolis, Indiana, USA > > This message contains confidential information and is intended for those > listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If > you are not the intended recipient you are notified that disclosing, > copying, distributing or taking any action in reliance on the contents of > this information is strictly prohibited. E-mail transmission cannot be > guaranteed to be secure or error-free as information could be intercepted, > corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. > The sender therefore does not accept liability for any errors or omissions > in the contents of this message, which arise as a result of e-mail > transmission. If verification is required please request a hard-copy > version. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120514/859c41f6/attachment-0001.html From andrew at cassidywebservices.co.uk Mon May 14 23:25:00 2012 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Mon, 14 May 2012 20:25:00 +0100 Subject: [Freeswitch-users] Voice Activity Detection Message-ID: Hi Guys, What's the default for VAD in the sip profiles if none is specified? I get choppy audio without explicity setting it to none, is there anything I can check/try to improve the situation? Hardware wise it's a rackspace cloud server (but it does have a 1000Hz preemptible kernel, and otherwise seems fine) but I'll be testing again on a proper machine in the coming days and network wise it's over the internet (just in case latency causes issues with VAD). It's not a massive issue in reality, just that rackspace charge for traffic, so thought it might be worth trying to cut it down. -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T *03300 100 960 *F *03300 100 961 *E *andrew at cassidywebservices.co.uk *W *www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120514/332e9822/attachment.html From tculjaga at gmail.com Tue May 15 00:12:04 2012 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Mon, 14 May 2012 22:12:04 +0200 Subject: [Freeswitch-users] joining a bridged call to a conference In-Reply-To: References: Message-ID: cool, so i don't need to transfer it to an extension ... i can transfer it to a conf room do i understand it well ? On Mon, May 14, 2012 at 6:20 PM, Peter Olsson < peter.olsson at visionutveckling.se> wrote: > Something like uuid_transfer uuid -both conference MyConfRoom inline. > > > /Peter > > 14 maj 2012 kl. 16:23 skrev "Tihomir Culjaga" tculjaga at gmail.com>>: > > yup, i was missing -both :=) > > next question if i may... > > say i do it all via event socket... > > > > originate user/1002 &conference(myConfRoom) > > > how can i transfer an established call ( A-Leg + B-Leg) to that conference > ? > > uuid_transfer uuid -both WHERE?? > > > > > > On Mon, May 14, 2012 at 7:40 AM, Peter Olsson < > peter.olsson at visionutveckling.se> > wrote: > I think "uuid_transfer -both" should work. If it's not working, > there might be some kind of media negotiation issue (for instance if bypass > media is used). > > /Peter > > 14 maj 2012 kl. 01:01 skrev "Tihomir Culjaga" tculjaga at gmail.com>>: > > > hello, > > > > i got an established call A calling B. A is a number outside FS (any > number e.g. 16084191500), B is an extension registered to FS (e.g. 1002). > > > > after some time, i would like to join the call A => B to a conference > roon e.g. 3001 (of course using ESL) > > > > any idea how to do it smooth ? > > whats the best practice for this ? > > > > i tried uuid_transfer but whatever i do it keeps killing one leg... > > > > > > thanks for help, > > T. > > > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org FreeSWITCH-users at lists.freeswitch.org> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > !DSPAM:4fb03a3f32761890320954! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org FreeSWITCH-users at lists.freeswitch.org> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > !DSPAM:4fb1124c32761694115387! > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org FreeSWITCH-users at lists.freeswitch.org> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > !DSPAM:4fb1124c32761694115387! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120514/3573a6e3/attachment.html From cmason at frontiernetworks.ca Tue May 15 00:12:24 2012 From: cmason at frontiernetworks.ca (Colin Mason) Date: Mon, 14 May 2012 16:12:24 -0400 Subject: [Freeswitch-users] Thanks FreeSWITCH Message-ID: <0D1C698866F66045A6201FD0F59CAC900146149B65@EX.frontier.local> UP 0 years, 7 days, 10 hours, 7 minutes, 58 seconds, 481 milliseconds, 883 microseconds FreeSWITCH is ready 1009971 session(s) since startup 376 session(s) 4/60 1000 session(s) max min idle cpu 0.00/69.00 This is running on a XenServer 6 Virtual Machine using CentOS 6.2 It uses 2 cores: Intel(R) Xeon(R) CPU E5430 @ 2.66GHz and 4GB of RAM. Using: FreeSWITCH Version 1.0.head (git-d827cfe 2012-03-04 17-48-30 -0600) I run freeswitch in -hp mode. I use track-calls in the SIP profile to store calls for HA support and mysql master-slave replication to transfer the database to the standby freeswitch virtual machine (different physical hardware). I use heartbeat and mon for network and service monitoring. If the network or a service drops it fails over to the standby freeswitch virtual machine without dropping the calls. Great product. Quality and Stability is rock solid. Thanks Anthony, Brian and everyone I forgot. This userlist is great and assisted me in troubleshooting many issues and for getting this beast up and running. Colin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120514/913af3fd/attachment.html From krice at freeswitch.org Tue May 15 00:32:40 2012 From: krice at freeswitch.org (Ken Rice) Date: Mon, 14 May 2012 15:32:40 -0500 Subject: [Freeswitch-users] Thanks FreeSWITCH In-Reply-To: <0D1C698866F66045A6201FD0F59CAC900146149B65@EX.frontier.local> Message-ID: Hey Colin, Could you document your setup on the wiki maybe? K On 5/14/12 3:12 PM, "Colin Mason" wrote: > UP 0 years, 7 days, 10 hours, 7 minutes, 58 seconds, 481 milliseconds, 883 > microseconds > FreeSWITCH is ready > 1009971 session(s) since startup > 376 session(s) 4/60 > 1000 session(s) max > min idle cpu 0.00/69.00 > > This is running on a XenServer 6 Virtual Machine using CentOS 6.2 > It uses 2 cores: Intel(R) Xeon(R) CPU E5430 @ 2.66GHz and 4GB of RAM. > > Using: FreeSWITCH Version 1.0.head (git-d827cfe 2012-03-04 17-48-30 -0600) > > I run freeswitch in ?hp mode. I use track-calls in the SIP profile to store > calls for HA support and mysql master-slave replication to transfer the > database to the standby freeswitch virtual machine (different physical > hardware). I use heartbeat and mon for network and service monitoring. If the > network or a service drops it fails over to the standby freeswitch virtual > machine without dropping the calls. > > Great product. Quality and Stability is rock solid. Thanks Anthony, Brian and > everyone I forgot. > > This userlist is great and assisted me in troubleshooting many issues and for > getting this beast up and running. > > Colin > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120514/6828be3a/attachment-0001.html From anthony.minessale at gmail.com Tue May 15 00:36:57 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 14 May 2012 15:36:57 -0500 Subject: [Freeswitch-users] Thanks FreeSWITCH In-Reply-To: References: <0D1C698866F66045A6201FD0F59CAC900146149B65@EX.frontier.local> Message-ID: One .... Meeelyun .... phone calls. On Mon, May 14, 2012 at 3:32 PM, Ken Rice wrote: > Hey Colin, > > Could you document your setup on the wiki maybe? > > K > > > > On 5/14/12 3:12 PM, "Colin Mason" wrote: > > UP 0 years, 7 days, 10 hours, 7 minutes, 58 seconds, 481 milliseconds, 883 > microseconds > FreeSWITCH is ready > 1009971 session(s) since startup > 376 session(s) 4/60 > 1000 session(s) max > min idle cpu 0.00/69.00 > > This is running on a XenServer 6 Virtual Machine using CentOS 6.2 > It uses 2 cores: Intel(R) Xeon(R) CPU E5430 @ 2.66GHz and 4GB of RAM. > > Using: FreeSWITCH Version 1.0.head (git-d827cfe 2012-03-04 17-48-30 -0600) > > I run freeswitch in ?hp mode. I use track-calls in the SIP profile to store > calls for HA support and mysql master-slave replication to transfer the > database to the standby freeswitch virtual machine (different physical > hardware). I use heartbeat and mon for network and service monitoring. If > the network or a service drops it fails over to the standby freeswitch > virtual machine without dropping the calls. > > Great product. Quality and Stability is rock solid. Thanks Anthony, Brian > and everyone I forgot. > > This userlist is great and assisted me in troubleshooting many issues and > for getting this beast up and running. > > Colin > > ________________________________ > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- A non-text attachment was scrubbed... Name: dr_evil.png Type: image/png Size: 195158 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120514/a49a7e7b/attachment-0001.png From acrow at integrafin.co.uk Tue May 15 00:38:44 2012 From: acrow at integrafin.co.uk (Alex Crow) Date: Mon, 14 May 2012 21:38:44 +0100 Subject: [Freeswitch-users] Thanks FreeSWITCH In-Reply-To: References: Message-ID: <4FB16D54.1000903@integrafin.co.uk> Seconded. The more documented ways of doing HA the better. Cheers Alex On 14/05/12 21:32, Ken Rice wrote: > Hey Colin, > > Could you document your setup on the wiki maybe? > > K > > > On 5/14/12 3:12 PM, "Colin Mason" wrote: > > UP 0 years, 7 days, 10 hours, 7 minutes, 58 seconds, 481 > milliseconds, 883 microseconds > FreeSWITCH is ready > 1009971 session(s) since startup > 376 session(s) 4/60 > 1000 session(s) max > min idle cpu 0.00/69.00 > > This is running on a XenServer 6 Virtual Machine using CentOS 6.2 > It uses 2 cores: Intel(R) Xeon(R) CPU E5430 @ 2.66GHz and 4GB of RAM. > > Using: FreeSWITCH Version 1.0.head (git-d827cfe 2012-03-04 > 17-48-30 -0600) > > I run freeswitch in --hp mode. I use track-calls in the SIP > profile to store calls for HA support and mysql master-slave > replication to transfer the database to the standby freeswitch > virtual machine (different physical hardware). I use heartbeat and > mon for network and service monitoring. If the network or a > service drops it fails over to the standby freeswitch virtual > machine without dropping the calls. > > Great product. Quality and Stability is rock solid. Thanks > Anthony, Brian and everyone I forgot. > > This userlist is great and assisted me in troubleshooting many > issues and for getting this beast up and running. > > Colin > > ------------------------------------------------------------------------ > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > -- > This message has been scanned for viruses and > dangerous content by *MailScanner* , and is > believed to be clean. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- This message is intended only for the addressee and may contain confidential information. Unless you are that person, you may not disclose its contents or use it in any way and are requested to delete the message along with any attachments and notify us immediately. "Transact" is operated by Integrated Financial Arrangements plc Domain House, 5-7 Singer Street, London EC2A 4BQ Tel: (020) 7608 4900 Fax: (020) 7608 5300 (Registered office: as above; Registered in England and Wales under number: 3727592) Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120514/3e0bdc10/attachment.html From msc at freeswitch.org Tue May 15 00:42:33 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 14 May 2012 13:42:33 -0700 Subject: [Freeswitch-users] Thanks FreeSWITCH In-Reply-To: References: <0D1C698866F66045A6201FD0F59CAC900146149B65@EX.frontier.local> Message-ID: On Mon, May 14, 2012 at 1:32 PM, Ken Rice wrote: > Hey Colin, > > Could you document your setup on the wiki maybe? > > K > > Also, if you want to come join the Wednesday conference call and give us the lowdown on how you pulled this off we'd love to hear about it. You've got much more than just FreeSWITCH in your environment. Putting all those pieces together was probably a lot of "fun". :) -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120514/4d4664ad/attachment.html From peter.olsson at visionutveckling.se Tue May 15 00:44:05 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Mon, 14 May 2012 20:44:05 +0000 Subject: [Freeswitch-users] joining a bridged call to a conference In-Reply-To: References: , Message-ID: <1FFF97C269757C458224B7C895F35F150CA6B5@cantor.std.visionutv.se> Yes, correct. When you use "inline", it should work by executing the application directly. /Peter ________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f?r Tihomir Culjaga [tculjaga at gmail.com] Skickat: den 14 maj 2012 22:12 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] joining a bridged call to a conference cool, so i don't need to transfer it to an extension ... i can transfer it to a conf room do i understand it well ? On Mon, May 14, 2012 at 6:20 PM, Peter Olsson > wrote: Something like uuid_transfer uuid -both conference MyConfRoom inline. /Peter 14 maj 2012 kl. 16:23 skrev "Tihomir Culjaga" >>: yup, i was missing -both :=) next question if i may... say i do it all via event socket... originate user/1002 &conference(myConfRoom) how can i transfer an established call ( A-Leg + B-Leg) to that conference ? uuid_transfer uuid -both WHERE?? On Mon, May 14, 2012 at 7:40 AM, Peter Olsson >> wrote: I think "uuid_transfer -both" should work. If it's not working, there might be some kind of media negotiation issue (for instance if bypass media is used). /Peter 14 maj 2012 kl. 01:01 skrev "Tihomir Culjaga" >>: > hello, > > i got an established call A calling B. A is a number outside FS (any number e.g. 16084191500), B is an extension registered to FS (e.g. 1002). > > after some time, i would like to join the call A => B to a conference roon e.g. 3001 (of course using ESL) > > any idea how to do it smooth ? > whats the best practice for this ? > > i tried uuid_transfer but whatever i do it keeps killing one leg... > > > thanks for help, > T. > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org> > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org> http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4fb1124c32761694115387! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org> http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4fb1124c32761694115387! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4fb165b532761043710233! From mytemike72 at gmail.com Tue May 15 01:31:24 2012 From: mytemike72 at gmail.com (Michael Lutz) Date: Mon, 14 May 2012 23:31:24 +0200 Subject: [Freeswitch-users] Using start_dtmf on an originated call via ESL. Message-ID: Hi All, I am having some trouble trying to figure out how to enable the start_dtmf (or equivalant) for DTMF detection on generated calls using the API (ESL) with originate. My inbound leg have set this on in the dialplan based on the provider of the incomming call (1 have one provider which does not support RFC2833) My problem is that my outbound calls use that specific provider. The thing is that I can only detect DTMF from my original inbound leg (these are bridged) but not from the outbound leg. I assume I have to enable the start_dtmf on the outbound leg after the originate. But have no clue how to do that becuase I park the call after originate so my ESL server can pick it up and handle the call. I can not find an API function or equivalant for this, and I have, of course, no session object a I originate the call from ESL. The code I am using is: string cDialString = "{origination_uuid=" + thisDial.CallId_LegB + ",origination_caller_id_number=" + thisDial.CLIP + ",origination_caller_id_name=" + thisDial.CLIP + "}sofia/external/" + thisExtension.Destination + " &park()"; eslEvent = thisFSDial.Connection.Api("originate", cDialString); After the call is succesfully answered I (depending on the script might play some audio to the dialed leg, and) after that I bridge using a uuid_bridge on an inbound uuid and this origination_uuid. (all this works perfectly, it's just the detection of the dtmf's on the b-leg!) I need to fix this by tommorow as I have a live customer who urgently requires this to work, so any help appreciated! Best regards, Mike. From bclark at grasshopper.com Tue May 15 01:07:02 2012 From: bclark at grasshopper.com (Brett Clark - Grasshopper) Date: Mon, 14 May 2012 16:07:02 -0500 Subject: [Freeswitch-users] Conferences without mod_conference? Message-ID: Hello, Is it possible to conference together multiple calls into a conference call without using the mod_conference module? Or am I restricted to bridged calls only when I disable mod_conference? I was just wondering what options I had if I was using an event socket, or the like. Thanks! Brett -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120514/03140671/attachment-0001.html From spencer at 5ninesolutions.com Tue May 15 01:43:49 2012 From: spencer at 5ninesolutions.com (Spencer Thomason) Date: Mon, 14 May 2012 14:43:49 -0700 Subject: [Freeswitch-users] Subscribe for MWI Message-ID: <9D1F6AAA-D64F-45BC-A70D-C6E469D38C30@5ninesolutions.com> Hello, I'm working on setting up FreeSWITCH as a media server behind Kamailio. All endpoints register to Kamailio and SUBSCRIBEs are forwarded to FreeSWITCH for MWI. FreeSWITCH is unaware of any registrations. When an endpoint has an active subscription and a new message is left, the NOTIFY is sent and everything works correctly. The problem arises when there is no subscription and a message is left. In this case when the endpoint creates an active subscription after the message, there is no NOTIFY sent. The there a way I can send NOTIFYs upon creation of a subscription if a message exists? Is there a better way to accomplish this? I'd like to avoid forwarding REGISTERs if possible. Thanks, Spencer From msc at freeswitch.org Tue May 15 01:53:43 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 14 May 2012 14:53:43 -0700 Subject: [Freeswitch-users] Using start_dtmf on an originated call via ESL. In-Reply-To: References: Message-ID: How about setting execute_on_answer=start_dtmf in your dialstring? string cDialString = "{execute_on_answer=start_dtmf,origination_uuid=" + thisDial.CallId_LegB ... Try it and let us know. -MC On Mon, May 14, 2012 at 2:31 PM, Michael Lutz wrote: > Hi All, > > I am having some trouble trying to figure out how to enable the > start_dtmf (or equivalant) for DTMF detection on generated calls using > the API (ESL) with originate. > My inbound leg have set this on in the dialplan based on the provider > of the incomming call (1 have one provider which does not support > RFC2833) > > My problem is that my outbound calls use that specific provider. > > The thing is that I can only detect DTMF from my original inbound leg > (these are bridged) but not from the outbound leg. > > I assume I have to enable the start_dtmf on the outbound leg after the > originate. But have no clue how to do that becuase I park the call > after originate so my ESL server can pick it up and handle the call. > I can not find an API function or equivalant for this, and I have, of > course, no session object a I originate the call from ESL. > > The code I am using is: > > string cDialString = "{origination_uuid=" + > thisDial.CallId_LegB > + > ",origination_caller_id_number=" + thisDial.CLIP > + > ",origination_caller_id_name=" + thisDial.CLIP > + "}sofia/external/" + > thisExtension.Destination > + " &park()"; > > eslEvent = thisFSDial.Connection.Api("originate", > cDialString); > > After the call is succesfully answered I (depending on the script > might play some audio to the dialed leg, and) after that I bridge > using a uuid_bridge on an inbound uuid and this origination_uuid. (all > this works perfectly, it's just the detection of the dtmf's on the > b-leg!) > > I need to fix this by tommorow as I have a live customer who urgently > requires this to work, so any help appreciated! > > > Best regards, > Mike. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120514/fcc8a026/attachment.html From msc at freeswitch.org Tue May 15 01:55:37 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 14 May 2012 14:55:37 -0700 Subject: [Freeswitch-users] Conferences without mod_conference? In-Reply-To: References: Message-ID: You'll need something to mix the audio of all these channels. If it's not mod_conference then it would need to be on one of the devices you are using. Question: what value is there in disabling mod_conference? It's about as robust an audio conferencing module as you'll find on the planet... -MC On Mon, May 14, 2012 at 2:07 PM, Brett Clark - Grasshopper < bclark at grasshopper.com> wrote: > Hello,**** > > ** ** > > Is it possible to conference together multiple calls into a conference > call without using the mod_conference module? Or am I restricted to > bridged calls only when I disable mod_conference? I was just wondering > what options I had if I was using an event socket, or the like.**** > > ** ** > > Thanks! > Brett**** > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120514/e1e61471/attachment.html From mytemike72 at gmail.com Tue May 15 02:53:27 2012 From: mytemike72 at gmail.com (Michael Lutz) Date: Tue, 15 May 2012 00:53:27 +0200 Subject: [Freeswitch-users] Using start_dtmf on an originated call via ESL. In-Reply-To: References: Message-ID: Hi Michael, Thanks for your quick response, that seems to have done the trick indead! Again thanks for your help! Regards, Mike. 2012/5/14 Michael Collins : > How about setting execute_on_answer=start_dtmf in your dialstring? > > string cDialString = "{execute_on_answer=start_dtmf,origination_uuid=" + > thisDial.CallId_LegB > ... > > Try it and let us know. > -MC > > > On Mon, May 14, 2012 at 2:31 PM, Michael Lutz wrote: >> >> Hi All, >> >> I am having some trouble trying to figure out how to enable the >> start_dtmf (or equivalant) for DTMF detection on generated calls using >> the API (ESL) with originate. >> My inbound leg have set this on in the dialplan based on the provider >> of the incomming call (1 have one provider which does not support >> RFC2833) >> >> My problem is that my outbound calls use that specific provider. >> >> The thing is that I can only detect DTMF from my original inbound leg >> (these are bridged) but not from the outbound leg. >> >> I assume I have to enable the start_dtmf on the outbound leg after the >> originate. But have no clue how to do that becuase I park the call >> after originate so my ESL server can pick it up and handle the call. >> I can not find an API function or equivalant for this, and I have, of >> course, no session object a I originate the call from ESL. >> >> The code I am using is: >> >> ? ? ? ? ? ? ? ? ? ?string cDialString = "{origination_uuid=" + >> thisDial.CallId_LegB >> ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?+ >> ",origination_caller_id_number=" + thisDial.CLIP >> ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?+ >> ",origination_caller_id_name=" + thisDial.CLIP >> ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?+ "}sofia/external/" + >> thisExtension.Destination >> ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?+ " &park()"; >> >> ? ? ? ? ? ? ? ? ? ?eslEvent = thisFSDial.Connection.Api("originate", >> cDialString); >> >> After the call is succesfully answered I (depending on the script >> might play some audio to the dialed leg, and) after that I bridge >> using a uuid_bridge on an inbound uuid and this origination_uuid. (all >> this works perfectly, it's just the detection of the dtmf's on the >> b-leg!) >> >> I need to fix this by tommorow as I have a live customer who urgently >> requires this to work, so any help appreciated! >> >> >> Best regards, >> Mike. > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From chris at opencsta.org Tue May 15 03:33:07 2012 From: chris at opencsta.org (Chris Mylonas) Date: Tue, 15 May 2012 09:33:07 +1000 Subject: [Freeswitch-users] mod_gsmopen requires spandsp In-Reply-To: References: <1DE50890-EE01-4EEE-83A8-5B5F71BB3959@opencsta.org> <3484676B-F4CA-47E3-A8B0-9743291C1056@opencsta.org> <62D399FB-DFA0-4DD0-B86F-45E95026EF82@opencsta.org> Message-ID: <90F18D04-F0A8-41DE-8785-CA4C56C42A10@opencsta.org> Thanks for the encouraging news/info Wesley! It's reminding me of the USB/Serial dongle days when serial ports first started disappearing off motherboards and Siemens resellers would buy the cheapest PC they could get their hands on to run CSTA software :) Mine is an E173u-2 I'll give it another crack today - I go away for a few weeks and will be taking the modem with me for on-the-road Internet - so I'd like to be able o relay success back to the list prior to the group buy. Cheers Chris On 15/05/2012, at 4:04 AM, Wesley Akio wrote: > I got to that at some point(E173s-6)... > > Wich distro are you in? > > Anyway, you're probably missing the device drivers. Mine (CentOS 5.8) looked just like that... with the drivers it gives me: > > crw------- 1 root root 188, 2 Mai 14 15:01 ttyUSB_utps_diag > crw------- 1 root root 188, 1 Mai 14 15:01 ttyUSB_utps_modem > crw------- 1 root root 188, 3 Mai 14 15:01 ttyUSB_utps_pcui > > I got SMS working after that. Modem is, of course, the modem... > > I could'n get audio working though so I have no clue on each of the other two is the RAW device. > > If you get it to work please let me know! > > Best, > > Wesley Akio > TuntsCorp.com > > > On Mon, May 14, 2012 at 1:31 PM, Chris Mylonas wrote: > Hi FS List, > > Sorry to badger you with this again. I'm going to sleep, it's 2:30am - but I will leave you with more mod_gsmopen testing stuff. > > > Getting this in my logs > > 2012-05-15 01:46:11.670033 [CRIT] switch_loadable_module.c:1300 Error Loading module /usr/local/freeswitch/mod/mod_gsmopen.so > **libctb-0.16.so: cannot open shared object file: No such file or directory** > > It did compile and install and I am running as root. > > [root at space log]# ls -l /usr/local/freeswitch/mod/mod_gsmopen.so > -rwxr-xr-x 1 root root 452284 May 15 01:38 /usr/local/freeswitch/mod/mod_gsmopen.so > > [root at space log]# ls -l /usr/local/lib/libctb-0.16.so > -rwxr-xr-x 1 root root 47525 May 15 01:38 /usr/local/lib/libctb-0.16.so > > Just for completeness, > > Configuration file is from src example, modified only with the USB device > > > > > > > > > > > > > > > > > > > > > > > > > And the /dev/usb stuff look like this (usbdev1.4_ep__ are the Huawei E173 dongle) - the config file is just my random stab in the dark at getting something going. > > /dev/usbdev1.1_ep00 > /dev/usbdev1.1_ep81 > /dev/usbdev1.4_ep00 > /dev/usbdev1.4_ep01 > /dev/usbdev1.4_ep02 > /dev/usbdev1.4_ep81 > /dev/usbdev1.4_ep82 > /dev/usbdev2.1_ep00 > /dev/usbdev2.1_ep81 > > Thanks > Chris > > > On 15/05/2012, at 12:51 AM, Brian Foster wrote: > >> if you compile libctb with debug it's actually libctbd, not libctb. >> >> On Mon, May 14, 2012 at 7:53 AM, Chris Mylonas wrote: >> Thanks for the reply Giovanni. I have done the steps on the wiki. >> >> tl;dr; should i put a prefix when i'm making the dependent mods so they don't go into /usr/local/lib, or this is why ldconfig is run - to tell the system where the libs are. >> >> All the relevant stuff is below >> >> Hope you can see something wrong, >> Cheers >> Chris >> >> e.g. >> here is my bash history >> >> 1050 cd freeswitch/ >> 1051 ls >> 1052 find . -name gsmlib >> 1053 cd src/mod/endpoints/mod_gsmopen/ >> 1054 ls >> 1055 cd gsmlib/ >> 1056 ls >> 1057 cd gsmlib-1.10-patched-13ubuntu/ >> 1058 ls >> 1059 ./configure >> 1060 make >> 1061 make install >> 1062 ldconfig >> 1063 cd /usr/src/freeswitch/src/mod/endpoints/mod_gsmopen/libctb-0.16/build >> 1064 make DEBUG=1 GPIB=0 >> 1065 make DEBUG=1 GPIB=0 install >> 1066 ldconfig >> 1067 cd /usr/src/freeswitch/src/mod/endpoints/mod_gsmopen/ >> 1068 make clean >> 1069 make install >> >> Just to repeat the compilation error >> Creating mod_gsmopen.so... >> /usr/bin/ld: cannot find -lctb-0.16 >> collect2: ld returned 1 exit status >> g++ -I../../../../libs/spandsp/src -I../../../..//libs/tiff-3.8.2/libtiff -DGSMOPEN_C_VER=\"44fd0de\" -DMODGSMOPEN_C_VER=\"44fd0de\" -I/usr/src/freeswitch/libs/curl/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -shared -o .libs/mod_gsmopen.so -shared -Wl,-x .libs/mod_gsmopen.o gsmopen_protocol.o /usr/src/freeswitch/.libs/libfreeswitch.so -L/usr/src/freeswitch/libs/apr-util/xml/expat/lib /usr/src/freeswitch/libs/apr-util/xml/expat/lib/.libs/libexpat.a /usr/src/freeswitch/libs/apr/.libs/libapr-1.a -luuid -lpthread -L/usr/src/freeswitch/libs/srtp -L/usr/kerberos/lib -lcrypt -lrt -lssl -lcrypto -ldl -lncurses -L/usr/src/freeswitch/libs/spandsp/src /usr/src/freeswitch/libs/spandsp/src/.libs/libspandsp.a -L/usr/src/freeswitch/libs/tiff-3.8.2/libtiff /usr/src/freeswitch/libs/tiff-3.8.2/libtiff/.libs/libtiff.a -ljpeg -lz -lm -lc -lctb-0.16 -lgsmme -Wl,--rpath -Wl,/usr/local/freeswitch/lib -Wl,--rpath -Wl,/usr/local/freeswitch/mod >> make[1]: *** [mod_gsmopen.so] Error 1 >> make: *** [install] Error 1 >> >> >> >> >> All the gsmlib stuff that is NOT in the freeswitch src dir is here >> >> [root at space mod_gsmopen]# locate gsmlib | grep -v src >> /usr/local/include/gsmlib >> /usr/local/include/gsmlib/gsm_at.h >> /usr/local/include/gsmlib/gsm_cb.h >> /usr/local/include/gsmlib/gsm_error.h >> /usr/local/include/gsmlib/gsm_event.h >> /usr/local/include/gsmlib/gsm_map_key.h >> /usr/local/include/gsmlib/gsm_me_ta.h >> /usr/local/include/gsmlib/gsm_parser.h >> /usr/local/include/gsmlib/gsm_phonebook.h >> /usr/local/include/gsmlib/gsm_port.h >> /usr/local/include/gsmlib/gsm_sie_me.h >> /usr/local/include/gsmlib/gsm_sms.h >> /usr/local/include/gsmlib/gsm_sms_codec.h >> /usr/local/include/gsmlib/gsm_sms_store.h >> /usr/local/include/gsmlib/gsm_sorted_phonebook.h >> /usr/local/include/gsmlib/gsm_sorted_phonebook_base.h >> /usr/local/include/gsmlib/gsm_sorted_sms_store.h >> /usr/local/include/gsmlib/gsm_unix_serial.h >> /usr/local/include/gsmlib/gsm_util.h >> /usr/local/share/locale/de/LC_MESSAGES/gsmlib.mo >> >> >> And the ctb stuff is in /usr/local/lib >> >> /usr/local/include/ctb-0.16 >> /usr/local/include/ctb-0.16/ctb.h >> /usr/local/include/ctb-0.16/fifo.h >> /usr/local/include/ctb-0.16/getopt.h >> /usr/local/include/ctb-0.16/iobase.h >> /usr/local/include/ctb-0.16/linux >> /usr/local/include/ctb-0.16/portscan.h >> /usr/local/include/ctb-0.16/serport.h >> /usr/local/include/ctb-0.16/serportx.h >> /usr/local/include/ctb-0.16/timer.h >> /usr/local/include/ctb-0.16/linux/serport.h >> /usr/local/include/ctb-0.16/linux/timer.h >> /usr/local/lib/libctbd-0.16.a >> /usr/local/lib/libctbd-0.16.so >> >> >> gcc version 4.1.2 20080704 (Red Hat 4.1.2-52) >> GNU Make 3.81 >> CentOS release 5.8 (Final) >> >> >> >> >> On 14/05/2012, at 9:38 PM, Giovanni Maruzzelli wrote: >> >>> you must first compile and install libctb, as per the wiki page ( >>> http://wiki.freeswitch.org/wiki/gsmopen ) >>> then, after installation of libctb and gsmlib (as per wiki), be sure >>> to update your dinamic link cache, or compiler will not find then. >>> >>> Eg: ldconfig >>> >>> >>> On 5/14/12, Chris Mylonas wrote: >>>> All good GM, no inconvenience, just a minor gotcha ;) >>>> I am unable to compile mod_gsmopen though. >>>> >>>> It complains about not being able to find ctb-0.16 >>>> The actual filename is libctbd-0.16.so in /usr/local/lib as you can see from >>>> the 2nd lot of stuff. >>>> >>>> How do I fix this? >>>> >>>> >>>> [root at space mod_gsmopen]# make install >>>> Creating mod_gsmopen.so... >>>> /usr/bin/ld: cannot find -lctb-0.16 >>>> collect2: ld returned 1 exit status >>>> g++ -I../../../../libs/spandsp/src -I../../../..//libs/tiff-3.8.2/libtiff >>>> -DGSMOPEN_C_VER=\"44fd0de\" -DMODGSMOPEN_C_VER=\"44fd0de\" >>>> -I/usr/src/freeswitch/libs/curl/include -I/usr/src/freeswitch/src/include >>>> -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/libs/libteletone/src >>>> -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g >>>> -O2 -D_GNU_SOURCE -shared -o .libs/mod_gsmopen.so -shared -Wl,-x >>>> .libs/mod_gsmopen.o gsmopen_protocol.o >>>> /usr/src/freeswitch/.libs/libfreeswitch.so >>>> -L/usr/src/freeswitch/libs/apr-util/xml/expat/lib >>>> /usr/src/freeswitch/libs/apr-util/xml/expat/lib/.libs/libexpat.a >>>> /usr/src/freeswitch/libs/apr/.libs/libapr-1.a -luuid -lpthread >>>> -L/usr/src/freeswitch/libs/srtp -L/usr/kerberos/lib -lcrypt -lrt -lssl >>>> -lcrypto -ldl -lncurses -L/usr/src/freeswitch/libs/spandsp/src >>>> /usr/src/freeswitch/libs/spandsp/src/.libs/libspandsp.a >>>> -L/usr/src/freeswitch/libs/tiff-3.8.2/libtiff >>>> /usr/src/freeswitch/libs/tiff-3.8.2/libtiff/.libs/libtiff.a -ljpeg -lz -lm >>>> -lc -lctb-0.16 -lgsmme -Wl,--rpath -Wl,/usr/local/freeswitch/lib >>>> -Wl,--rpath -Wl,/usr/local/freeswitch/mod >>>> make[1]: *** [mod_gsmopen.so] Error 1 >>>> make: *** [install] Error 1 >>>> >>>> >>>> >>>> [root at space mod_gsmopen]# ldd /usr/local/lib/libctbd-0.16.so >>>> linux-gate.so.1 => (0x00754000) >>>> libpthread.so.0 => /lib/libpthread.so.0 (0x00e83000) >>>> libstdc++.so.6 => /usr/lib/libstdc++.so.6 (0x00d28000) >>>> libm.so.6 => /lib/libm.so.6 (0x00964000) >>>> libgcc_s.so.1 => /lib/libgcc_s.so.1 (0x00e1f000) >>>> libc.so.6 => /lib/libc.so.6 (0x00110000) >>>> /lib/ld-linux.so.2 (0x003ea000) >>>> >>>> >>>> >>>> On 14/05/2012, at 6:44 PM, Giovanni Maruzzelli wrote: >>>> >>>>> yes, it requires libspandsp, maybe the Makefile it's not yet tweaked >>>>> to build the library automatically. >>>>> >>>>> So, please first build mod_spandsp, then mod_gsmopen. >>>>> >>>>> We'll fixx the Makefile soon, sorry for the inconvenience. >>>>> >>>>> -giovanni >>>>> >>>>> On Mon, May 14, 2012 at 9:30 AM, Chris Mylonas >>>>> wrote: >>>>>> Hi FS List, >>>>>> >>>>>> FYI - as a shortcut to building my freeswitch, I skip spandsp - but it >>>>>> looks >>>>>> like this mod_gsmopen wants it in there. >>>>>> >>>>>> >>>>>> [root at space build]# cd >>>>>> /usr/src/freeswitch/src/mod/endpoints/mod_gsmopen/ >>>>>> [root at space mod_gsmopen]# make clean >>>>>> [root at space mod_gsmopen]# make install >>>>>> Compiling gsmopen_protocol.cpp... >>>>>> Compiling >>>>>> /usr/src/freeswitch/src/mod/endpoints/mod_gsmopen/mod_gsmopen.cpp... >>>>>> mkdir .libs >>>>>> Compiling >>>>>> /usr/src/freeswitch/src/mod/endpoints/mod_gsmopen/mod_gsmopen.cpp >>>>>> ... >>>>>> Creating mod_gsmopen.so... >>>>>> /usr/bin/ld: cannot find -lspandsp >>>>>> collect2: ld returned 1 exit status >>>>>> g++ -I../../../../libs/spandsp/src >>>>>> -I../../../..//libs/tiff-3.8.2/libtiff >>>>>> -DGSMOPEN_C_VER=\"44fd0de\" -DMODGSMOPEN_C_VER=\"44fd0de\" >>>>>> -I/usr/src/freeswitch/libs/curl/include >>>>>> -I/usr/src/freeswitch/src/include >>>>>> -I/usr/src/freeswitch/src/include >>>>>> -I/usr/src/freeswitch/libs/libteletone/src >>>>>> -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 >>>>>> -g >>>>>> -O2 -D_GNU_SOURCE -shared -o .libs/mod_gsmopen.so -shared -Wl,-x >>>>>> .libs/mod_gsmopen.o gsmopen_protocol.o -lm >>>>>> /usr/src/freeswitch/.libs/libfreeswitch.so >>>>>> -L/usr/src/freeswitch/libs/apr-util/xml/expat/lib >>>>>> /usr/src/freeswitch/libs/apr-util/xml/expat/lib/.libs/libexpat.a >>>>>> /usr/src/freeswitch/libs/apr/.libs/libapr-1.a -luuid -lpthread >>>>>> -L/usr/src/freeswitch/libs/srtp -L/usr/kerberos/lib -lcrypt -lrt -lssl >>>>>> -lcrypto -ldl -lz -lncurses -ljpeg >>>>>> -L/usr/src/freeswitch/libs/spandsp/src >>>>>> -lspandsp -lctb-0.16 -lgsmme -Wl,--rpath -Wl,/usr/local/freeswitch/lib >>>>>> -Wl,--rpath -Wl,/usr/local/freeswitch/mod >>>>>> make[1]: *** [mod_gsmopen.so] Error 1 >>>>>> make: *** [install] Error 1 >>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>> >>>>> >>>>> >>>>> -- >>>>> Sincerely, >>>>> >>>>> Giovanni Maruzzelli >>>>> Cell : +39-347-2665618 >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Sincerely, >>> >>> Giovanni Maruzzelli >>> Cell : +39-347-2665618 >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> -- >> Brian D. Foster >> Endigo Computer LLC >> Email: bdfoster at endigotech.com >> Phone: 317-800-7876 >> Indianapolis, Indiana, USA >> >> This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120515/97ef53ea/attachment-0001.html From anthony.minessale at gmail.com Tue May 15 05:32:02 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 14 May 2012 20:32:02 -0500 Subject: [Freeswitch-users] Subscribe for MWI In-Reply-To: <9D1F6AAA-D64F-45BC-A70D-C6E469D38C30@5ninesolutions.com> References: <9D1F6AAA-D64F-45BC-A70D-C6E469D38C30@5ninesolutions.com> Message-ID: are you on the latest GIT HEAD? Looking at the code suggests that should work. Its subbing to message-summary events? On Mon, May 14, 2012 at 4:43 PM, Spencer Thomason wrote: > Hello, > I'm working on setting up FreeSWITCH as a media server behind Kamailio. ?All endpoints register to Kamailio and SUBSCRIBEs are forwarded to FreeSWITCH for MWI. ?FreeSWITCH is unaware of any registrations. ?When an endpoint has an active subscription and a new message is left, the NOTIFY is sent and everything works correctly. ?The problem arises when there is no subscription and a message is left. ?In this case when the endpoint creates an active subscription after the message, there is no NOTIFY sent. ?The there a way I can send NOTIFYs upon creation of a subscription if a message exists? ?Is there a better way to accomplish this? ?I'd like to avoid forwarding REGISTERs if possible. > > Thanks, > Spencer > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From chris at opencsta.org Tue May 15 06:35:51 2012 From: chris at opencsta.org (Chris Mylonas) Date: Tue, 15 May 2012 12:35:51 +1000 Subject: [Freeswitch-users] mod_gsmopen requires spandsp In-Reply-To: References: <1DE50890-EE01-4EEE-83A8-5B5F71BB3959@opencsta.org> <3484676B-F4CA-47E3-A8B0-9743291C1056@opencsta.org> <62D399FB-DFA0-4DD0-B86F-45E95026EF82@opencsta.org> Message-ID: <074EF57F-FF04-4BAE-BFD8-A9729C4EBB8B@opencsta.org> Device drivers installed and I get the ttyUSB devices Wesley mentioned in the previous email. I've just recompiled FS (and trying again currently with a ./configure --with-libctb=/usr/local/lib/libctb-0.16.so in the hope that will inform FS where to look) and get the following error - which is the same as yesterday when loading mod_gsmopen The dependent libs are in place as per the wiki page and run ldconfig after install. 2012-05-15 12:52:44.119783 [CRIT] switch_loadable_module.c:1300 Error Loading module /usr/local/freeswitch/mod/mod_gsmopen.so **libctb-0.16.so: cannot open shared object file: No such file or directory** I'm using FreeSWITCH Version 1.2.0 (git-44fd0de 2012-05-14 02-04-36 +0200) I have a CentOS 6 64 bit virtualised system running in USA I will test on that to see if the module loads. Thanks Chris On 15/05/2012, at 4:04 AM, Wesley Akio wrote: > I got to that at some point(E173s-6)... > > Wich distro are you in? > > Anyway, you're probably missing the device drivers. Mine (CentOS 5.8) looked just like that... with the drivers it gives me: > > crw------- 1 root root 188, 2 Mai 14 15:01 ttyUSB_utps_diag > crw------- 1 root root 188, 1 Mai 14 15:01 ttyUSB_utps_modem > crw------- 1 root root 188, 3 Mai 14 15:01 ttyUSB_utps_pcui > > I got SMS working after that. Modem is, of course, the modem... > > I could'n get audio working though so I have no clue on each of the other two is the RAW device. > > If you get it to work please let me know! > > Best, > > Wesley Akio > TuntsCorp.com > > > On Mon, May 14, 2012 at 1:31 PM, Chris Mylonas wrote: > Hi FS List, > > Sorry to badger you with this again. I'm going to sleep, it's 2:30am - but I will leave you with more mod_gsmopen testing stuff. > > > Getting this in my logs > > 2012-05-15 01:46:11.670033 [CRIT] switch_loadable_module.c:1300 Error Loading module /usr/local/freeswitch/mod/mod_gsmopen.so > **libctb-0.16.so: cannot open shared object file: No such file or directory** > > It did compile and install and I am running as root. > > [root at space log]# ls -l /usr/local/freeswitch/mod/mod_gsmopen.so > -rwxr-xr-x 1 root root 452284 May 15 01:38 /usr/local/freeswitch/mod/mod_gsmopen.so > > [root at space log]# ls -l /usr/local/lib/libctb-0.16.so > -rwxr-xr-x 1 root root 47525 May 15 01:38 /usr/local/lib/libctb-0.16.so > > Just for completeness, > > Configuration file is from src example, modified only with the USB device > > > > > > > > > > > > > > > > > > > > > > > > > And the /dev/usb stuff look like this (usbdev1.4_ep__ are the Huawei E173 dongle) - the config file is just my random stab in the dark at getting something going. > > /dev/usbdev1.1_ep00 > /dev/usbdev1.1_ep81 > /dev/usbdev1.4_ep00 > /dev/usbdev1.4_ep01 > /dev/usbdev1.4_ep02 > /dev/usbdev1.4_ep81 > /dev/usbdev1.4_ep82 > /dev/usbdev2.1_ep00 > /dev/usbdev2.1_ep81 > > Thanks > Chris > > > On 15/05/2012, at 12:51 AM, Brian Foster wrote: > >> if you compile libctb with debug it's actually libctbd, not libctb. >> >> On Mon, May 14, 2012 at 7:53 AM, Chris Mylonas wrote: >> Thanks for the reply Giovanni. I have done the steps on the wiki. >> >> tl;dr; should i put a prefix when i'm making the dependent mods so they don't go into /usr/local/lib, or this is why ldconfig is run - to tell the system where the libs are. >> >> All the relevant stuff is below >> >> Hope you can see something wrong, >> Cheers >> Chris >> >> e.g. >> here is my bash history >> >> 1050 cd freeswitch/ >> 1051 ls >> 1052 find . -name gsmlib >> 1053 cd src/mod/endpoints/mod_gsmopen/ >> 1054 ls >> 1055 cd gsmlib/ >> 1056 ls >> 1057 cd gsmlib-1.10-patched-13ubuntu/ >> 1058 ls >> 1059 ./configure >> 1060 make >> 1061 make install >> 1062 ldconfig >> 1063 cd /usr/src/freeswitch/src/mod/endpoints/mod_gsmopen/libctb-0.16/build >> 1064 make DEBUG=1 GPIB=0 >> 1065 make DEBUG=1 GPIB=0 install >> 1066 ldconfig >> 1067 cd /usr/src/freeswitch/src/mod/endpoints/mod_gsmopen/ >> 1068 make clean >> 1069 make install >> >> Just to repeat the compilation error >> Creating mod_gsmopen.so... >> /usr/bin/ld: cannot find -lctb-0.16 >> collect2: ld returned 1 exit status >> g++ -I../../../../libs/spandsp/src -I../../../..//libs/tiff-3.8.2/libtiff -DGSMOPEN_C_VER=\"44fd0de\" -DMODGSMOPEN_C_VER=\"44fd0de\" -I/usr/src/freeswitch/libs/curl/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -shared -o .libs/mod_gsmopen.so -shared -Wl,-x .libs/mod_gsmopen.o gsmopen_protocol.o /usr/src/freeswitch/.libs/libfreeswitch.so -L/usr/src/freeswitch/libs/apr-util/xml/expat/lib /usr/src/freeswitch/libs/apr-util/xml/expat/lib/.libs/libexpat.a /usr/src/freeswitch/libs/apr/.libs/libapr-1.a -luuid -lpthread -L/usr/src/freeswitch/libs/srtp -L/usr/kerberos/lib -lcrypt -lrt -lssl -lcrypto -ldl -lncurses -L/usr/src/freeswitch/libs/spandsp/src /usr/src/freeswitch/libs/spandsp/src/.libs/libspandsp.a -L/usr/src/freeswitch/libs/tiff-3.8.2/libtiff /usr/src/freeswitch/libs/tiff-3.8.2/libtiff/.libs/libtiff.a -ljpeg -lz -lm -lc -lctb-0.16 -lgsmme -Wl,--rpath -Wl,/usr/local/freeswitch/lib -Wl,--rpath -Wl,/usr/local/freeswitch/mod >> make[1]: *** [mod_gsmopen.so] Error 1 >> make: *** [install] Error 1 >> >> >> >> >> All the gsmlib stuff that is NOT in the freeswitch src dir is here >> >> [root at space mod_gsmopen]# locate gsmlib | grep -v src >> /usr/local/include/gsmlib >> /usr/local/include/gsmlib/gsm_at.h >> /usr/local/include/gsmlib/gsm_cb.h >> /usr/local/include/gsmlib/gsm_error.h >> /usr/local/include/gsmlib/gsm_event.h >> /usr/local/include/gsmlib/gsm_map_key.h >> /usr/local/include/gsmlib/gsm_me_ta.h >> /usr/local/include/gsmlib/gsm_parser.h >> /usr/local/include/gsmlib/gsm_phonebook.h >> /usr/local/include/gsmlib/gsm_port.h >> /usr/local/include/gsmlib/gsm_sie_me.h >> /usr/local/include/gsmlib/gsm_sms.h >> /usr/local/include/gsmlib/gsm_sms_codec.h >> /usr/local/include/gsmlib/gsm_sms_store.h >> /usr/local/include/gsmlib/gsm_sorted_phonebook.h >> /usr/local/include/gsmlib/gsm_sorted_phonebook_base.h >> /usr/local/include/gsmlib/gsm_sorted_sms_store.h >> /usr/local/include/gsmlib/gsm_unix_serial.h >> /usr/local/include/gsmlib/gsm_util.h >> /usr/local/share/locale/de/LC_MESSAGES/gsmlib.mo >> >> >> And the ctb stuff is in /usr/local/lib >> >> /usr/local/include/ctb-0.16 >> /usr/local/include/ctb-0.16/ctb.h >> /usr/local/include/ctb-0.16/fifo.h >> /usr/local/include/ctb-0.16/getopt.h >> /usr/local/include/ctb-0.16/iobase.h >> /usr/local/include/ctb-0.16/linux >> /usr/local/include/ctb-0.16/portscan.h >> /usr/local/include/ctb-0.16/serport.h >> /usr/local/include/ctb-0.16/serportx.h >> /usr/local/include/ctb-0.16/timer.h >> /usr/local/include/ctb-0.16/linux/serport.h >> /usr/local/include/ctb-0.16/linux/timer.h >> /usr/local/lib/libctbd-0.16.a >> /usr/local/lib/libctbd-0.16.so >> >> >> gcc version 4.1.2 20080704 (Red Hat 4.1.2-52) >> GNU Make 3.81 >> CentOS release 5.8 (Final) >> >> >> >> >> On 14/05/2012, at 9:38 PM, Giovanni Maruzzelli wrote: >> >>> you must first compile and install libctb, as per the wiki page ( >>> http://wiki.freeswitch.org/wiki/gsmopen ) >>> then, after installation of libctb and gsmlib (as per wiki), be sure >>> to update your dinamic link cache, or compiler will not find then. >>> >>> Eg: ldconfig >>> >>> >>> On 5/14/12, Chris Mylonas wrote: >>>> All good GM, no inconvenience, just a minor gotcha ;) >>>> I am unable to compile mod_gsmopen though. >>>> >>>> It complains about not being able to find ctb-0.16 >>>> The actual filename is libctbd-0.16.so in /usr/local/lib as you can see from >>>> the 2nd lot of stuff. >>>> >>>> How do I fix this? >>>> >>>> >>>> [root at space mod_gsmopen]# make install >>>> Creating mod_gsmopen.so... >>>> /usr/bin/ld: cannot find -lctb-0.16 >>>> collect2: ld returned 1 exit status >>>> g++ -I../../../../libs/spandsp/src -I../../../..//libs/tiff-3.8.2/libtiff >>>> -DGSMOPEN_C_VER=\"44fd0de\" -DMODGSMOPEN_C_VER=\"44fd0de\" >>>> -I/usr/src/freeswitch/libs/curl/include -I/usr/src/freeswitch/src/include >>>> -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/libs/libteletone/src >>>> -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g >>>> -O2 -D_GNU_SOURCE -shared -o .libs/mod_gsmopen.so -shared -Wl,-x >>>> .libs/mod_gsmopen.o gsmopen_protocol.o >>>> /usr/src/freeswitch/.libs/libfreeswitch.so >>>> -L/usr/src/freeswitch/libs/apr-util/xml/expat/lib >>>> /usr/src/freeswitch/libs/apr-util/xml/expat/lib/.libs/libexpat.a >>>> /usr/src/freeswitch/libs/apr/.libs/libapr-1.a -luuid -lpthread >>>> -L/usr/src/freeswitch/libs/srtp -L/usr/kerberos/lib -lcrypt -lrt -lssl >>>> -lcrypto -ldl -lncurses -L/usr/src/freeswitch/libs/spandsp/src >>>> /usr/src/freeswitch/libs/spandsp/src/.libs/libspandsp.a >>>> -L/usr/src/freeswitch/libs/tiff-3.8.2/libtiff >>>> /usr/src/freeswitch/libs/tiff-3.8.2/libtiff/.libs/libtiff.a -ljpeg -lz -lm >>>> -lc -lctb-0.16 -lgsmme -Wl,--rpath -Wl,/usr/local/freeswitch/lib >>>> -Wl,--rpath -Wl,/usr/local/freeswitch/mod >>>> make[1]: *** [mod_gsmopen.so] Error 1 >>>> make: *** [install] Error 1 >>>> >>>> >>>> >>>> [root at space mod_gsmopen]# ldd /usr/local/lib/libctbd-0.16.so >>>> linux-gate.so.1 => (0x00754000) >>>> libpthread.so.0 => /lib/libpthread.so.0 (0x00e83000) >>>> libstdc++.so.6 => /usr/lib/libstdc++.so.6 (0x00d28000) >>>> libm.so.6 => /lib/libm.so.6 (0x00964000) >>>> libgcc_s.so.1 => /lib/libgcc_s.so.1 (0x00e1f000) >>>> libc.so.6 => /lib/libc.so.6 (0x00110000) >>>> /lib/ld-linux.so.2 (0x003ea000) >>>> >>>> >>>> >>>> On 14/05/2012, at 6:44 PM, Giovanni Maruzzelli wrote: >>>> >>>>> yes, it requires libspandsp, maybe the Makefile it's not yet tweaked >>>>> to build the library automatically. >>>>> >>>>> So, please first build mod_spandsp, then mod_gsmopen. >>>>> >>>>> We'll fixx the Makefile soon, sorry for the inconvenience. >>>>> >>>>> -giovanni >>>>> >>>>> On Mon, May 14, 2012 at 9:30 AM, Chris Mylonas >>>>> wrote: >>>>>> Hi FS List, >>>>>> >>>>>> FYI - as a shortcut to building my freeswitch, I skip spandsp - but it >>>>>> looks >>>>>> like this mod_gsmopen wants it in there. >>>>>> >>>>>> >>>>>> [root at space build]# cd >>>>>> /usr/src/freeswitch/src/mod/endpoints/mod_gsmopen/ >>>>>> [root at space mod_gsmopen]# make clean >>>>>> [root at space mod_gsmopen]# make install >>>>>> Compiling gsmopen_protocol.cpp... >>>>>> Compiling >>>>>> /usr/src/freeswitch/src/mod/endpoints/mod_gsmopen/mod_gsmopen.cpp... >>>>>> mkdir .libs >>>>>> Compiling >>>>>> /usr/src/freeswitch/src/mod/endpoints/mod_gsmopen/mod_gsmopen.cpp >>>>>> ... >>>>>> Creating mod_gsmopen.so... >>>>>> /usr/bin/ld: cannot find -lspandsp >>>>>> collect2: ld returned 1 exit status >>>>>> g++ -I../../../../libs/spandsp/src >>>>>> -I../../../..//libs/tiff-3.8.2/libtiff >>>>>> -DGSMOPEN_C_VER=\"44fd0de\" -DMODGSMOPEN_C_VER=\"44fd0de\" >>>>>> -I/usr/src/freeswitch/libs/curl/include >>>>>> -I/usr/src/freeswitch/src/include >>>>>> -I/usr/src/freeswitch/src/include >>>>>> -I/usr/src/freeswitch/libs/libteletone/src >>>>>> -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 >>>>>> -g >>>>>> -O2 -D_GNU_SOURCE -shared -o .libs/mod_gsmopen.so -shared -Wl,-x >>>>>> .libs/mod_gsmopen.o gsmopen_protocol.o -lm >>>>>> /usr/src/freeswitch/.libs/libfreeswitch.so >>>>>> -L/usr/src/freeswitch/libs/apr-util/xml/expat/lib >>>>>> /usr/src/freeswitch/libs/apr-util/xml/expat/lib/.libs/libexpat.a >>>>>> /usr/src/freeswitch/libs/apr/.libs/libapr-1.a -luuid -lpthread >>>>>> -L/usr/src/freeswitch/libs/srtp -L/usr/kerberos/lib -lcrypt -lrt -lssl >>>>>> -lcrypto -ldl -lz -lncurses -ljpeg >>>>>> -L/usr/src/freeswitch/libs/spandsp/src >>>>>> -lspandsp -lctb-0.16 -lgsmme -Wl,--rpath -Wl,/usr/local/freeswitch/lib >>>>>> -Wl,--rpath -Wl,/usr/local/freeswitch/mod >>>>>> make[1]: *** [mod_gsmopen.so] Error 1 >>>>>> make: *** [install] Error 1 >>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>> >>>>> >>>>> >>>>> -- >>>>> Sincerely, >>>>> >>>>> Giovanni Maruzzelli >>>>> Cell : +39-347-2665618 >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Sincerely, >>> >>> Giovanni Maruzzelli >>> Cell : +39-347-2665618 >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> -- >> Brian D. Foster >> Endigo Computer LLC >> Email: bdfoster at endigotech.com >> Phone: 317-800-7876 >> Indianapolis, Indiana, USA >> >> This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120515/cc2a6eb5/attachment-0001.html From spencer at 5ninesolutions.com Tue May 15 06:47:38 2012 From: spencer at 5ninesolutions.com (Spencer Thomason) Date: Mon, 14 May 2012 19:47:38 -0700 Subject: [Freeswitch-users] Subscribe for MWI In-Reply-To: References: <9D1F6AAA-D64F-45BC-A70D-C6E469D38C30@5ninesolutions.com> Message-ID: <8b33f052-3895-4252-8f47-6ce672e810c6@blur> I am not. I am using a build from February. I will update in the morning and retest. Thanks, Spencer Connected by DROID on Verizon Wireless -----Original message----- From: Anthony Minessale To: FreeSWITCH Users Help Sent: Tue, May 15, 2012 01:33:43 GMT+00:00 Subject: Re: [Freeswitch-users] Subscribe for MWI are you on the latest GIT HEAD? Looking at the code suggests that should work. Its subbing to message-summary events? On Mon, May 14, 2012 at 4:43 PM, Spencer Thomason wrote: > Hello, > I'm working on setting up FreeSWITCH as a media server behind Kamailio. ?All endpoints register to Kamailio and SUBSCRIBEs are forwarded to FreeSWITCH for MWI. ?FreeSWITCH is unaware of any registrations. ?When an endpoint has an active subscription and a new message is left, the NOTIFY is sent and everything works correctly. ?The problem arises when there is no subscription and a message is left. ?In this case when the endpoint creates an active subscription after the message, there is no NOTIFY sent. ?The there a way I can send NOTIFYs upon creation of a subscription if a message exists? ?Is there a better way to accomplish this? ?I'd like to avoid forwarding REGISTERs if possible. > > Thanks, > Spencer > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120514/c90c477e/attachment.html From msc at freeswitch.org Tue May 15 07:01:26 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 14 May 2012 20:01:26 -0700 Subject: [Freeswitch-users] Using start_dtmf on an originated call via ESL. In-Reply-To: References: Message-ID: :) On Mon, May 14, 2012 at 3:53 PM, Michael Lutz wrote: > Hi Michael, > > Thanks for your quick response, that seems to have done the trick indead! > Again thanks for your help! > > Regards, > Mike. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120514/9c0d0cbd/attachment.html From govoiper at gmail.com Tue May 15 09:32:51 2012 From: govoiper at gmail.com (SamyGo) Date: Tue, 15 May 2012 10:32:51 +0500 Subject: [Freeswitch-users] Thanks FreeSWITCH In-Reply-To: References: <0D1C698866F66045A6201FD0F59CAC900146149B65@EX.frontier.local> Message-ID: WOW, I'm pretty sure there are other FS deployments giving quality performances, but just 7 days... !! It will be great if this kind of setup-guide is documented for the benefit for all of us here. Regards, Sammy On Tue, May 15, 2012 at 1:42 AM, Michael Collins wrote: > > > On Mon, May 14, 2012 at 1:32 PM, Ken Rice wrote: > >> Hey Colin, >> >> Could you document your setup on the wiki maybe? >> >> K >> >> > Also, if you want to come join the Wednesday conference call and give us > the lowdown on how you pulled this off we'd love to hear about it. You've > got much more than just FreeSWITCH in your environment. Putting all those > pieces together was probably a lot of "fun". :) > > -MC > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120515/426612ae/attachment.html From bote_radio at botecomm.com Tue May 15 10:13:46 2012 From: bote_radio at botecomm.com (Bote Man) Date: Tue, 15 May 2012 02:13:46 -0400 Subject: [Freeswitch-users] detect if conference is active Message-ID: <007301cd3261$e3b85f60$ab291e20$@com> I have a situation where I am dialing a number of outbound calls using the conference module. This works for me. BUT! If someone misses the call or just wants to join the existing outbound conference they can't just dial the extension that I have set up for it since I have coded logic in the dialplan to exclude the conference from dialing the originator--there's no sense in dialing the person who is initiating the conference after all. This is a small list of fixed, known extensions in a closed collection, nothing random. So how could I detect if the conference room is already active so that I can just dump the late entry right into the room instead of falling through my initiator logic? I'm hoping it can all be done in the dialplan because I'm not proficient in the scripting and event socket approaches. As a hacky workaround I could have the users dial a different extension to make it a standard inbound conference, but I want the switch to be smart enough to do the Right Thing when one extension number is dialed, either initiate the outbound group call or join the same existing conference room late. Thanks! Bote http://www.botecomm.com/bote/radio ? my hobby radio pages http://www.trackstreamer.com ? my streaming scanner feeds -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120515/54f80c9a/attachment.html From shaheryarkh at googlemail.com Tue May 15 11:19:52 2012 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Tue, 15 May 2012 09:19:52 +0200 Subject: [Freeswitch-users] Thanks FreeSWITCH In-Reply-To: References: <0D1C698866F66045A6201FD0F59CAC900146149B65@EX.frontier.local> Message-ID: Here are stats from my production box. It been runing over a year now, without any maintenance. :-) UP 1 year, 68 days, 11 hours, 58 minutes, 31 seconds, 333 milliseconds, 996 microseconds 11131103 session(s) since startup 496 session(s) 0/30 1000 session(s) max min idle cpu 0.00/100.00 Yes! FreeSWITCH is damn stable. Thank you. On Tue, May 15, 2012 at 7:32 AM, SamyGo wrote: > WOW, > I'm pretty sure there are other FS deployments giving quality > performances, but just 7 days... !! > It will be great if this kind of setup-guide is documented for the > benefit for all of us here. > Regards, > Sammy > > On Tue, May 15, 2012 at 1:42 AM, Michael Collins wrote: > >> >> >> On Mon, May 14, 2012 at 1:32 PM, Ken Rice wrote: >> >>> Hey Colin, >>> >>> Could you document your setup on the wiki maybe? >>> >>> K >>> >>> >> Also, if you want to come join the Wednesday conference call and give us >> the lowdown on how you pulled this off we'd love to hear about it. You've >> got much more than just FreeSWITCH in your environment. Putting all those >> pieces together was probably a lot of "fun". :) >> >> -MC >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120515/965159c5/attachment-0001.html From bobc at devassert.com Tue May 15 11:22:56 2012 From: bobc at devassert.com (Bob Coleman) Date: Tue, 15 May 2012 07:22:56 +0000 Subject: [Freeswitch-users] Thanks FreeSWITCH In-Reply-To: <0D1C698866F66045A6201FD0F59CAC900146149B65@EX.frontier.local> References: <0D1C698866F66045A6201FD0F59CAC900146149B65@EX.frontier.local> Message-ID: Nice, agree with you there, great product across many different platforms. We run FreeSWITCH on Windows 2008 Standard with SQL Server 2008, in a load balanced fashion across two physical machines, using C# ESL based ivr applications. Although our stats arent based on one machine, they arent too bad either, the limiting factor for us is not enough customers :-) Node One:UP 0 years, 35 days, 5 hours, 2 minutes, 29 seconds, 798 milliseconds, 961 microseconds3036283 session(s) since startupmin idle cpu 0.00/86.72 Node Two:UP 0 years, 35 days, 5 hours, 7 minutes, 20 seconds, 676 milliseconds, 891 microseconds3025205 session(s) since startupmin idle cpu 0.00/97.66 Total of 6,061,488 calls in 35 days. We have more problems with our SIP trunk suppliers than we do with FreeSWITCH :-) BobFrom: cmason at frontiernetworks.ca To: freeswitch-users at lists.freeswitch.org Date: Mon, 14 May 2012 16:12:24 -0400 Subject: [Freeswitch-users] Thanks FreeSWITCH UP 0 years, 7 days, 10 hours, 7 minutes, 58 seconds, 481 milliseconds, 883 microsecondsFreeSWITCH is ready1009971 session(s) since startup376 session(s) 4/601000 session(s) maxmin idle cpu 0.00/69.00 This is running on a XenServer 6 Virtual Machine using CentOS 6.2It uses 2 cores: Intel(R) Xeon(R) CPU E5430 @ 2.66GHz and 4GB of RAM. Using: FreeSWITCH Version 1.0.head (git-d827cfe 2012-03-04 17-48-30 -0600) I run freeswitch in ?hp mode. I use track-calls in the SIP profile to store calls for HA support and mysql master-slave replication to transfer the database to the standby freeswitch virtual machine (different physical hardware). I use heartbeat and mon for network and service monitoring. If the network or a service drops it fails over to the standby freeswitch virtual machine without dropping the calls. Great product. Quality and Stability is rock solid. Thanks Anthony, Brian and everyone I forgot. This userlist is great and assisted me in troubleshooting many issues and for getting this beast up and running. Colin _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120515/8909a82e/attachment.html From garbytrash at gmail.com Tue May 15 11:30:28 2012 From: garbytrash at gmail.com (Zenny) Date: Tue, 15 May 2012 09:30:28 +0200 Subject: [Freeswitch-users] Thanks FreeSWITCH In-Reply-To: References: <0D1C698866F66045A6201FD0F59CAC900146149B65@EX.frontier.local> Message-ID: @Colin Mason: Very impressive setup and in seven days more than a million calls. Mind blowing! Hats off to FS developers and your achievement. I also second others in the list that you need to document it for the benefits of the community because it is more than FS! /z --- Support http://thehumanape.com On 5/15/12, SamyGo wrote: > WOW, > I'm pretty sure there are other FS deployments giving quality performances, > but just 7 days... !! > It will be great if this kind of setup-guide is documented for the benefit > for all of us here. > Regards, > Sammy > > On Tue, May 15, 2012 at 1:42 AM, Michael Collins > wrote: > >> >> >> On Mon, May 14, 2012 at 1:32 PM, Ken Rice wrote: >> >>> Hey Colin, >>> >>> Could you document your setup on the wiki maybe? >>> >>> K >>> >>> >> Also, if you want to come join the Wednesday conference call and give us >> the lowdown on how you pulled this off we'd love to hear about it. You've >> got much more than just FreeSWITCH in your environment. Putting all those >> pieces together was probably a lot of "fun". :) >> >> -MC >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > From piyush.sharma at coraltele.com Tue May 15 11:27:52 2012 From: piyush.sharma at coraltele.com (Piyush Sharma) Date: Tue, 15 May 2012 12:57:52 +0530 (IST) Subject: [Freeswitch-users] Thanks FreeSWITCH In-Reply-To: <0D1C698866F66045A6201FD0F59CAC900146149B65@EX.frontier.local> Message-ID: <2c1b6681-f326-49cb-8333-38c33fbf8db6@mail.coraltele.com> It amazing, Such a great performance of FreeSWITCH. It is really awesome. If it is documented, we all can achieve the same. Thanks ----- Original Message ----- From: "Colin Mason" To: freeswitch-users at lists.freeswitch.org Sent: Tuesday, May 15, 2012 1:42:24 AM Subject: [Freeswitch-users] Thanks FreeSWITCH UP 0 years, 7 days, 10 hours, 7 minutes, 58 seconds, 481 milliseconds, 883 microseconds FreeSWITCH is ready 1009971 session(s) since startup 376 session(s) 4/60 1000 session(s) max min idle cpu 0.00/69.00 This is running on a XenServer 6 Virtual Machine using CentOS 6.2 It uses 2 cores: Intel(R) Xeon(R) CPU E5430 @ 2.66GHz and 4GB of RAM. Using: FreeSWITCH Version 1.0.head (git-d827cfe 2012-03-04 17-48-30 -0600) I run freeswitch in ?hp mode. I use track-calls in the SIP profile to store calls for HA support and mysql master-slave replication to transfer the database to the standby freeswitch virtual machine (different physical hardware). I use heartbeat and mon for network and service monitoring. If the network or a service drops it fails over to the standby freeswitch virtual machine without dropping the calls. Great product. Quality and Stability is rock solid. Thanks Anthony, Brian and everyone I forgot. This userlist is great and assisted me in troubleshooting many issues and for getting this beast up and running. Colin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120515/d5b9b940/attachment.html From nbhatti at gmail.com Tue May 15 11:43:21 2012 From: nbhatti at gmail.com (Muhammad Naseer Bhatti) Date: Tue, 15 May 2012 10:43:21 +0300 Subject: [Freeswitch-users] Thanks FreeSWITCH In-Reply-To: <2c1b6681-f326-49cb-8333-38c33fbf8db6@mail.coraltele.com> References: <0D1C698866F66045A6201FD0F59CAC900146149B65@EX.frontier.local> <2c1b6681-f326-49cb-8333-38c33fbf8db6@mail.coraltele.com> Message-ID: I think for the sake of the project, and make it evolve even more, such configurations should be documented and possibly should be made public on wiki. This will help other users compare and possibly one can also enhance their existing setup too. -B On Tue, May 15, 2012 at 10:27 AM, Piyush Sharma wrote: > It amazing, Such a great performance of FreeSWITCH. It is really awesome. If > it is documented, we all can achieve the same. > Thanks > > ________________________________ > From: "Colin Mason" > To: freeswitch-users at lists.freeswitch.org > Sent: Tuesday, May 15, 2012 1:42:24 AM > Subject: [Freeswitch-users] Thanks FreeSWITCH > > > UP 0 years, 7 days, 10 hours, 7 minutes, 58 seconds, 481 milliseconds, 883 > microseconds > > FreeSWITCH is ready > > 1009971 session(s) since startup > > 376 session(s) 4/60 > > 1000 session(s) max > > min idle cpu 0.00/69.00 > > > > This is running on a XenServer 6 Virtual Machine using CentOS 6.2 > > It uses 2 cores: Intel(R) Xeon(R) CPU E5430 @ 2.66GHz and 4GB of RAM. > > > > Using: FreeSWITCH Version 1.0.head (git-d827cfe 2012-03-04 17-48-30 -0600) > > > > I run freeswitch in ?hp mode. I use track-calls in the SIP profile to store > calls for HA support and mysql master-slave replication to transfer the > database to the standby freeswitch virtual machine (different physical > hardware). I use heartbeat and mon for network and service monitoring. If > the network or a service drops it fails over to the standby freeswitch > virtual machine without dropping the calls. > > > > Great product. Quality and Stability is rock solid. Thanks Anthony, Brian > and everyone I forgot. > > > > This userlist is great and assisted me in troubleshooting many issues and > for getting this beast up and running. > > > > Colin > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From sharad at coraltele.com Tue May 15 11:56:02 2012 From: sharad at coraltele.com (Sharad Garg) Date: Tue, 15 May 2012 13:26:02 +0530 Subject: [Freeswitch-users] Thanks FreeSWITCH References: <0D1C698866F66045A6201FD0F59CAC900146149B65@EX.frontier.local> Message-ID: <7012D3EB08DC49D98FAD465E59D570E3@sharad> Hi all We are using Freeswitch as voicemail with a IP based call server & I do not remember a single instance when Freeswitch was crashed. Yes, 1-2 times, server hardware was down but that was due to some power or some other hardware issue. But as far as freeswitch is concerned, not even a single failure... ....rock stable. regards sharad ----- Original Message ----- From: Muhammad Shahzad To: FreeSWITCH Users Help Sent: Tuesday, May 15, 2012 12:49 PM Subject: Re: [Freeswitch-users] Thanks FreeSWITCH Here are stats from my production box. It been runing over a year now, without any maintenance. :-) UP 1 year, 68 days, 11 hours, 58 minutes, 31 seconds, 333 milliseconds, 996 microseconds 11131103 session(s) since startup 496 session(s) 0/30 1000 session(s) max min idle cpu 0.00/100.00 Yes! FreeSWITCH is damn stable. Thank you. On Tue, May 15, 2012 at 7:32 AM, SamyGo wrote: WOW, I'm pretty sure there are other FS deployments giving quality performances, but just 7 days... !! It will be great if this kind of setup-guide is documented for the benefit for all of us here. Regards, Sammy On Tue, May 15, 2012 at 1:42 AM, Michael Collins wrote: On Mon, May 14, 2012 at 1:32 PM, Ken Rice wrote: Hey Colin, Could you document your setup on the wiki maybe? K Also, if you want to come join the Wednesday conference call and give us the lowdown on how you pulled this off we'd love to hear about it. You've got much more than just FreeSWITCH in your environment. Putting all those pieces together was probably a lot of "fun". :) -MC _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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 at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120515/1f8e9521/attachment-0001.html From bote_radio at botecomm.com Tue May 15 12:09:33 2012 From: bote_radio at botecomm.com (Bote Man) Date: Tue, 15 May 2012 04:09:33 -0400 Subject: [Freeswitch-users] Thanks FreeSWITCH In-Reply-To: References: <0D1C698866F66045A6201FD0F59CAC900146149B65@EX.frontier.local> Message-ID: <009601cd3272$10a40e60$31ec2b20$@com> That's central office-grade. Bote From: Muhammad Shahzad Sent: Tuesday, 15 May, 2012 03:20 Here are stats from my production box. It been runing over a year now, without any maintenance. :-) UP 1 year, 68 days, 11 hours, 58 minutes, 31 seconds, 333 milliseconds, 996 microseconds 11131103 session(s) since startup 496 session(s) 0/30 1000 session(s) max min idle cpu 0.00/100.00 Yes! FreeSWITCH is damn stable. Thank you. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120515/1e6e847d/attachment.html From hynek.cihlar at gmail.com Tue May 15 12:37:09 2012 From: hynek.cihlar at gmail.com (Hynek Cihlar) Date: Tue, 15 May 2012 10:37:09 +0200 Subject: [Freeswitch-users] Thanks FreeSWITCH In-Reply-To: <0D1C698866F66045A6201FD0F59CAC900146149B65@EX.frontier.local> References: <0D1C698866F66045A6201FD0F59CAC900146149B65@EX.frontier.local> Message-ID: Impressive! I would be interested how did you setup the connection from FreeSWITCH to MySQL. We had to give up HA for now, because of the MySQL ODBC driver stability issues. Hynek On Mon, May 14, 2012 at 10:12 PM, Colin Mason wrote: > UP 0 years, 7 days, 10 hours, 7 minutes, 58 seconds, 481 milliseconds, 883 > microseconds**** > > FreeSWITCH is ready**** > > 1009971 session(s) since startup**** > > 376 session(s) 4/60**** > > 1000 session(s) max**** > > min idle cpu 0.00/69.00**** > > ** ** > > This is running on a XenServer 6 Virtual Machine using CentOS 6.2**** > > It uses 2 cores: Intel(R) Xeon(R) CPU E5430 @ 2.66GHz and 4GB of RAM.**** > > ** ** > > Using: FreeSWITCH Version 1.0.head (git-d827cfe 2012-03-04 17-48-30 -0600) > **** > > ** ** > > I run freeswitch in ?hp mode. I use track-calls in the SIP profile to > store calls for HA support and mysql master-slave replication to transfer > the database to the standby freeswitch virtual machine (different physical > hardware). I use heartbeat and mon for network and service monitoring. If > the network or a service drops it fails over to the standby freeswitch > virtual machine without dropping the calls.**** > > ** ** > > Great product. Quality and Stability is rock solid. Thanks Anthony, Brian > and everyone I forgot.**** > > ** ** > > This userlist is great and assisted me in troubleshooting many issues and > for getting this beast up and running.**** > > ** ** > > Colin**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120515/71695902/attachment.html From B.Tietz at pinguin.ag Tue May 15 12:56:18 2012 From: B.Tietz at pinguin.ag (B.Tietz at pinguin.ag) Date: Tue, 15 May 2012 10:56:18 +0200 Subject: [Freeswitch-users] Thanks FreeSWITCH In-Reply-To: References: <0D1C698866F66045A6201FD0F59CAC900146149B65@EX.frontier.local> Message-ID: <07BF4904977CC645B485E970424193AD10EDC37E63@localhost> Hi, in my setup the unixODBC 2.3.1 and mysql odbc connector 5.1.9 is stable Benjamin Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Hynek Cihlar Gesendet: Dienstag, 15. Mai 2012 10:37 An: FreeSWITCH Users Help Betreff: Re: [Freeswitch-users] Thanks FreeSWITCH Impressive! I would be interested how did you setup the connection from FreeSWITCH to MySQL. We had to give up HA for now, because of the MySQL ODBC driver stability issues. Hynek On Mon, May 14, 2012 at 10:12 PM, Colin Mason > wrote: UP 0 years, 7 days, 10 hours, 7 minutes, 58 seconds, 481 milliseconds, 883 microseconds FreeSWITCH is ready 1009971 session(s) since startup 376 session(s) 4/60 1000 session(s) max min idle cpu 0.00/69.00 This is running on a XenServer 6 Virtual Machine using CentOS 6.2 It uses 2 cores: Intel(R) Xeon(R) CPU E5430 @ 2.66GHz and 4GB of RAM. Using: FreeSWITCH Version 1.0.head (git-d827cfe 2012-03-04 17-48-30 -0600) I run freeswitch in -hp mode. I use track-calls in the SIP profile to store calls for HA support and mysql master-slave replication to transfer the database to the standby freeswitch virtual machine (different physical hardware). I use heartbeat and mon for network and service monitoring. If the network or a service drops it fails over to the standby freeswitch virtual machine without dropping the calls. Great product. Quality and Stability is rock solid. Thanks Anthony, Brian and everyone I forgot. This userlist is great and assisted me in troubleshooting many issues and for getting this beast up and running. Colin _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120515/255fb441/attachment.html From chris at opencsta.org Tue May 15 13:12:25 2012 From: chris at opencsta.org (Chris Mylonas) Date: Tue, 15 May 2012 19:12:25 +1000 Subject: [Freeswitch-users] 2012 VoIP With Virtualization Message-ID: <550DADC3-9636-42FF-B391-BB00E84AC403@opencsta.org> Hi FS Users, What is the consensus about using virtualized servers for real-time voice (RTP)? Even up until a few years ago it was hard to guarantee the CPU cycles to the voice nodes. Virtualizing the signalling (SIP) has always been favourable in terms of HA. If the entire host is under control, is it safe to say - we can virtualise voice? Does it depend on what else is shared in the host - say it would be silly to put a high load database server on the same hardware node... Thanks for your inputs Chris From joohny at mail.ru Tue May 15 13:15:13 2012 From: joohny at mail.ru (=?UTF-8?B?0JXQstCz0LXQvdC40Lk=?=) Date: Tue, 15 May 2012 13:15:13 +0400 Subject: [Freeswitch-users] =?utf-8?q?mod=5Fcallcenter_real_answer?= Message-ID: Hi, colleagues. Could I have real answer for client on mod_callcenter, when agent answers? Or there is another way to queue call between 2 FreeSwitch. For example, client's call has to be routed to two FS servers, which has different agents in queues. But answer comes from first server when application CALLCENTER is called. How can I do this? Best Regards, Evginey. http://blog.buchnev.ru -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120515/efb4cc6a/attachment.html From haloha201 at gmail.com Tue May 15 13:55:19 2012 From: haloha201 at gmail.com (haloha) Date: Tue, 15 May 2012 16:55:19 +0700 Subject: [Freeswitch-users] create rpm fail on freeswitch-1.2.rc2 Message-ID: hi list i am try to build rpm from source and i get error look like this: /usr/bin/install -c 'scripts/fsxs' '/var/tmp/freeswitch-1.2.rc2-%{BUILD_NUMBER}-root-ha/usr/bin/fsxs' mkdir: cannot create directory `/var/tmp/freeswitch-1.2.rc2-%{BUILD_NUMBER}-root-ha/usr/lib/freeswitch/mod': No such file or directory make[3]: *** [install-exec-local] Error 1 make[2]: *** [install-am] Error 2 Making install in src Making install in mod making install mod_abstraction installing mod_abstraction.so /usr/bin/install: cannot create regular file `/var/tmp/freeswitch-1.2.rc2-%{BUILD_NUMBER}-root-ha/usr/lib/freeswitch/mod': No such file or directory make[5]: *** [/var/tmp/freeswitch-1.2.rc2-%{BUILD_NUMBER}-root-ha/usr/lib/freeswitch/mod/mod_abstraction.so] Error 1 make[4]: *** [install] Error 1 make[3]: *** [mod_abstraction-install] Error 1 make[2]: *** [install-recursive] Error 1 make[1]: *** [install-recursive] Error 1 make: *** [install] Error 2 error: Bad exit status from /var/tmp/rpm-tmp.46706 (%install) RPM build errors: Bad exit status from /var/tmp/rpm-tmp.46706 (%install) please help thank you From koralu at gmail.com Tue May 15 14:08:54 2012 From: koralu at gmail.com (Adrian Andrei) Date: Tue, 15 May 2012 13:08:54 +0300 Subject: [Freeswitch-users] origination_caller_id_number - internal user bridging Message-ID: Hello, I have one xlite(1000), a FS and Cisco box that is registred as a user(1001) to FS. When I dial ^5555$ from user/1000 I want to bridge the call to user/1001 and change the number to 7777. My dial plan looks: After the call is made, the cisco sees that the origination number = 1000 instead of 7777 and I think is fair because the sip call is 1000 at ip_cisco:5060. But I want the cisco box to see 7777. Is any solution to make this setup? Ty From avi at avimarcus.net Tue May 15 14:28:55 2012 From: avi at avimarcus.net (Avi Marcus) Date: Tue, 15 May 2012 13:28:55 +0300 Subject: [Freeswitch-users] origination_caller_id_number - internal user bridging In-Reply-To: References: Message-ID: I always use effective_caller_id_number -Avi (This message was painstakingly thumbed out on my mobile, so apologies for brevity and errors.) On May 15, 2012 1:10 PM, "Adrian Andrei" wrote: Hello, I have one xlite(1000), a FS and Cisco box that is registred as a user(1001) to FS. When I dial ^5555$ from user/1000 I want to bridge the call to user/1001 and change the number to 7777. My dial plan looks: After the call is made, the cisco sees that the origination number = 1000 instead of 7777 and I think is fair because the sip call is 1000 at ip_cisco:5060. But I want the cisco box to see 7777. Is any solution to make this setup? Ty _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120515/4b4fe6f2/attachment.html From jaybinks at gmail.com Tue May 15 14:32:00 2012 From: jaybinks at gmail.com (jay binks) Date: Tue, 15 May 2012 20:32:00 +1000 Subject: [Freeswitch-users] 2012 VoIP With Virtualization In-Reply-To: <550DADC3-9636-42FF-B391-BB00E84AC403@opencsta.org> References: <550DADC3-9636-42FF-B391-BB00E84AC403@opencsta.org> Message-ID: basically it depends ... on MANY things.. a) how much you care about your voice quality b) how many concurrent calls c) do you modify the RTP stream or just pass it through. basically if your real picky or have more than a handful of concurrent calls.. then dont do it. if your using conferences etc... dont do it. that being said, you can get away with a lot if you dont mind the occasional blip and chirp in your media.. Thing to not though... above is for virtualization ... ( ESX, Xen, etc ) Jails like OpenVZ, BSD Jails, etc work well and provide many of the benefits found in virtualization. Jay On 15 May 2012 19:12, Chris Mylonas wrote: > Hi FS Users, > > What is the consensus about using virtualized servers for real-time voice > (RTP)? Even up until a few years ago it was hard to guarantee the CPU > cycles to the voice nodes. > Virtualizing the signalling (SIP) has always been favourable in terms of > HA. > > If the entire host is under control, is it safe to say - we can virtualise > voice? > Does it depend on what else is shared in the host - say it would be silly > to put a high load database server on the same hardware node... > > Thanks for your inputs > Chris > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sincerely Jay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120515/76a4f7e7/attachment.html From chris at opencsta.org Tue May 15 15:14:21 2012 From: chris at opencsta.org (Chris Mylonas) Date: Tue, 15 May 2012 21:14:21 +1000 Subject: [Freeswitch-users] 2012 VoIP With Virtualization In-Reply-To: References: <550DADC3-9636-42FF-B391-BB00E84AC403@opencsta.org> Message-ID: Beauty - thanks Jay - I'll check out what jails can do for me. I've just compiled FS on 2 openvz servers (Denver/LA, I'm in Sydney) for testing a bunch of stuff, firstly loading mod_gsmopen without it complaining about ctb not being available. In my world, voice is precious ;) No virtualisation continues to be my modus operandi for now then :) Cheers Chris On 15/05/2012, at 8:32 PM, jay binks wrote: > basically it depends ... on MANY things.. > > a) how much you care about your voice quality > b) how many concurrent calls > c) do you modify the RTP stream or just pass it through. > > basically if your real picky or have more than a handful of concurrent calls.. then dont do it. > if your using conferences etc... dont do it. > > that being said, you can get away with a lot if you dont mind the occasional blip and chirp in your media.. > > Thing to not though... above is for virtualization ... ( ESX, Xen, etc ) > > Jails like OpenVZ, BSD Jails, etc work well and provide many of the benefits found in virtualization. > > Jay > > > > On 15 May 2012 19:12, Chris Mylonas wrote: > Hi FS Users, > > What is the consensus about using virtualized servers for real-time voice (RTP)? Even up until a few years ago it was hard to guarantee the CPU cycles to the voice nodes. > Virtualizing the signalling (SIP) has always been favourable in terms of HA. > > If the entire host is under control, is it safe to say - we can virtualise voice? > Does it depend on what else is shared in the host - say it would be silly to put a high load database server on the same hardware node... > > Thanks for your inputs > Chris > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -- > Sincerely > > Jay > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120515/17627afb/attachment-0001.html From koralu at gmail.com Tue May 15 15:17:37 2012 From: koralu at gmail.com (Adrian Andrei) Date: Tue, 15 May 2012 14:17:37 +0300 Subject: [Freeswitch-users] origination_caller_id_number - internal user bridging In-Reply-To: References: Message-ID: Doesn't work. On 5/15/12, Avi Marcus wrote: > I always use effective_caller_id_number > > -Avi > (This message was painstakingly thumbed out on my mobile, so apologies for > brevity and errors.) > > On May 15, 2012 1:10 PM, "Adrian Andrei" wrote: > > Hello, > > I have one xlite(1000), a FS and Cisco box that is registred as a > user(1001) to FS. > > When I dial ^5555$ from user/1000 I want to bridge the call to > user/1001 and change the number to 7777. My dial plan looks: > > > > > > > > > After the call is made, the cisco sees that the origination number = > 1000 instead of 7777 and I think is fair because the sip call is > 1000 at ip_cisco:5060. > > But I want the cisco box to see 7777. Is any solution to make this setup? > > Ty > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From andrew at cassidywebservices.co.uk Tue May 15 15:40:01 2012 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Tue, 15 May 2012 12:40:01 +0100 Subject: [Freeswitch-users] origination_caller_id_number - internal user bridging In-Reply-To: References: Message-ID: try application="export" rather than application="set" On 15 May 2012 12:17, Adrian Andrei wrote: > Doesn't work. > > On 5/15/12, Avi Marcus wrote: > > I always use effective_caller_id_number > > > > -Avi > > (This message was painstakingly thumbed out on my mobile, so apologies > for > > brevity and errors.) > > > > On May 15, 2012 1:10 PM, "Adrian Andrei" wrote: > > > > Hello, > > > > I have one xlite(1000), a FS and Cisco box that is registred as a > > user(1001) to FS. > > > > When I dial ^5555$ from user/1000 I want to bridge the call to > > user/1001 and change the number to 7777. My dial plan looks: > > > > > > > > > > > > > > > > > > After the call is made, the cisco sees that the origination number = > > 1000 instead of 7777 and I think is fair because the sip call is > > 1000 at ip_cisco:5060. > > > > But I want the cisco box to see 7777. Is any solution to make this setup? > > > > Ty > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T *03300 100 960 *F *03300 100 961 *E *andrew at cassidywebservices.co.uk *W *www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120515/1badd1bf/attachment.html From chris at opencsta.org Tue May 15 15:51:52 2012 From: chris at opencsta.org (Chris Mylonas) Date: Tue, 15 May 2012 21:51:52 +1000 Subject: [Freeswitch-users] mod_gsmopen requires spandsp In-Reply-To: References: <1DE50890-EE01-4EEE-83A8-5B5F71BB3959@opencsta.org> <3484676B-F4CA-47E3-A8B0-9743291C1056@opencsta.org> <62D399FB-DFA0-4DD0-B86F-45E95026EF82@opencsta.org> Message-ID: <98FF7FEB-32D4-414B-914A-9E819EB97801@opencsta.org> I had no luck today at home with CentOS 5 (32bit) and loading mod_gsmopen I tried it with CentOS 6 both 32/64 bit on a remote machine but the same error message comes up After compiling mod_gsmopen, this message came: quiet_libtool: install: warning: relinking `mod_gsmopen.la' Does this mean anything significant? Kind Regards, Chris On 15/05/2012, at 4:04 AM, Wesley Akio wrote: > I got to that at some point(E173s-6)... > > Wich distro are you in? > > Anyway, you're probably missing the device drivers. Mine (CentOS 5.8) looked just like that... with the drivers it gives me: > > crw------- 1 root root 188, 2 Mai 14 15:01 ttyUSB_utps_diag > crw------- 1 root root 188, 1 Mai 14 15:01 ttyUSB_utps_modem > crw------- 1 root root 188, 3 Mai 14 15:01 ttyUSB_utps_pcui > > I got SMS working after that. Modem is, of course, the modem... > > I could'n get audio working though so I have no clue on each of the other two is the RAW device. > > If you get it to work please let me know! > > Best, > > Wesley Akio > TuntsCorp.com > > > On Mon, May 14, 2012 at 1:31 PM, Chris Mylonas wrote: > Hi FS List, > > Sorry to badger you with this again. I'm going to sleep, it's 2:30am - but I will leave you with more mod_gsmopen testing stuff. > > > Getting this in my logs > > 2012-05-15 01:46:11.670033 [CRIT] switch_loadable_module.c:1300 Error Loading module /usr/local/freeswitch/mod/mod_gsmopen.so > **libctb-0.16.so: cannot open shared object file: No such file or directory** > > It did compile and install and I am running as root. > > [root at space log]# ls -l /usr/local/freeswitch/mod/mod_gsmopen.so > -rwxr-xr-x 1 root root 452284 May 15 01:38 /usr/local/freeswitch/mod/mod_gsmopen.so > > [root at space log]# ls -l /usr/local/lib/libctb-0.16.so > -rwxr-xr-x 1 root root 47525 May 15 01:38 /usr/local/lib/libctb-0.16.so > > Just for completeness, > > Configuration file is from src example, modified only with the USB device > > > > > > > > > > > > > > > > > > > > > > > > > And the /dev/usb stuff look like this (usbdev1.4_ep__ are the Huawei E173 dongle) - the config file is just my random stab in the dark at getting something going. > > /dev/usbdev1.1_ep00 > /dev/usbdev1.1_ep81 > /dev/usbdev1.4_ep00 > /dev/usbdev1.4_ep01 > /dev/usbdev1.4_ep02 > /dev/usbdev1.4_ep81 > /dev/usbdev1.4_ep82 > /dev/usbdev2.1_ep00 > /dev/usbdev2.1_ep81 > > Thanks > Chris > > > On 15/05/2012, at 12:51 AM, Brian Foster wrote: > >> if you compile libctb with debug it's actually libctbd, not libctb. >> >> On Mon, May 14, 2012 at 7:53 AM, Chris Mylonas wrote: >> Thanks for the reply Giovanni. I have done the steps on the wiki. >> >> tl;dr; should i put a prefix when i'm making the dependent mods so they don't go into /usr/local/lib, or this is why ldconfig is run - to tell the system where the libs are. >> >> All the relevant stuff is below >> >> Hope you can see something wrong, >> Cheers >> Chris >> >> e.g. >> here is my bash history >> >> 1050 cd freeswitch/ >> 1051 ls >> 1052 find . -name gsmlib >> 1053 cd src/mod/endpoints/mod_gsmopen/ >> 1054 ls >> 1055 cd gsmlib/ >> 1056 ls >> 1057 cd gsmlib-1.10-patched-13ubuntu/ >> 1058 ls >> 1059 ./configure >> 1060 make >> 1061 make install >> 1062 ldconfig >> 1063 cd /usr/src/freeswitch/src/mod/endpoints/mod_gsmopen/libctb-0.16/build >> 1064 make DEBUG=1 GPIB=0 >> 1065 make DEBUG=1 GPIB=0 install >> 1066 ldconfig >> 1067 cd /usr/src/freeswitch/src/mod/endpoints/mod_gsmopen/ >> 1068 make clean >> 1069 make install >> >> Just to repeat the compilation error >> Creating mod_gsmopen.so... >> /usr/bin/ld: cannot find -lctb-0.16 >> collect2: ld returned 1 exit status >> g++ -I../../../../libs/spandsp/src -I../../../..//libs/tiff-3.8.2/libtiff -DGSMOPEN_C_VER=\"44fd0de\" -DMODGSMOPEN_C_VER=\"44fd0de\" -I/usr/src/freeswitch/libs/curl/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -shared -o .libs/mod_gsmopen.so -shared -Wl,-x .libs/mod_gsmopen.o gsmopen_protocol.o /usr/src/freeswitch/.libs/libfreeswitch.so -L/usr/src/freeswitch/libs/apr-util/xml/expat/lib /usr/src/freeswitch/libs/apr-util/xml/expat/lib/.libs/libexpat.a /usr/src/freeswitch/libs/apr/.libs/libapr-1.a -luuid -lpthread -L/usr/src/freeswitch/libs/srtp -L/usr/kerberos/lib -lcrypt -lrt -lssl -lcrypto -ldl -lncurses -L/usr/src/freeswitch/libs/spandsp/src /usr/src/freeswitch/libs/spandsp/src/.libs/libspandsp.a -L/usr/src/freeswitch/libs/tiff-3.8.2/libtiff /usr/src/freeswitch/libs/tiff-3.8.2/libtiff/.libs/libtiff.a -ljpeg -lz -lm -lc -lctb-0.16 -lgsmme -Wl,--rpath -Wl,/usr/local/freeswitch/lib -Wl,--rpath -Wl,/usr/local/freeswitch/mod >> make[1]: *** [mod_gsmopen.so] Error 1 >> make: *** [install] Error 1 >> >> >> >> >> All the gsmlib stuff that is NOT in the freeswitch src dir is here >> >> [root at space mod_gsmopen]# locate gsmlib | grep -v src >> /usr/local/include/gsmlib >> /usr/local/include/gsmlib/gsm_at.h >> /usr/local/include/gsmlib/gsm_cb.h >> /usr/local/include/gsmlib/gsm_error.h >> /usr/local/include/gsmlib/gsm_event.h >> /usr/local/include/gsmlib/gsm_map_key.h >> /usr/local/include/gsmlib/gsm_me_ta.h >> /usr/local/include/gsmlib/gsm_parser.h >> /usr/local/include/gsmlib/gsm_phonebook.h >> /usr/local/include/gsmlib/gsm_port.h >> /usr/local/include/gsmlib/gsm_sie_me.h >> /usr/local/include/gsmlib/gsm_sms.h >> /usr/local/include/gsmlib/gsm_sms_codec.h >> /usr/local/include/gsmlib/gsm_sms_store.h >> /usr/local/include/gsmlib/gsm_sorted_phonebook.h >> /usr/local/include/gsmlib/gsm_sorted_phonebook_base.h >> /usr/local/include/gsmlib/gsm_sorted_sms_store.h >> /usr/local/include/gsmlib/gsm_unix_serial.h >> /usr/local/include/gsmlib/gsm_util.h >> /usr/local/share/locale/de/LC_MESSAGES/gsmlib.mo >> >> >> And the ctb stuff is in /usr/local/lib >> >> /usr/local/include/ctb-0.16 >> /usr/local/include/ctb-0.16/ctb.h >> /usr/local/include/ctb-0.16/fifo.h >> /usr/local/include/ctb-0.16/getopt.h >> /usr/local/include/ctb-0.16/iobase.h >> /usr/local/include/ctb-0.16/linux >> /usr/local/include/ctb-0.16/portscan.h >> /usr/local/include/ctb-0.16/serport.h >> /usr/local/include/ctb-0.16/serportx.h >> /usr/local/include/ctb-0.16/timer.h >> /usr/local/include/ctb-0.16/linux/serport.h >> /usr/local/include/ctb-0.16/linux/timer.h >> /usr/local/lib/libctbd-0.16.a >> /usr/local/lib/libctbd-0.16.so >> >> >> gcc version 4.1.2 20080704 (Red Hat 4.1.2-52) >> GNU Make 3.81 >> CentOS release 5.8 (Final) >> >> >> >> >> On 14/05/2012, at 9:38 PM, Giovanni Maruzzelli wrote: >> >>> you must first compile and install libctb, as per the wiki page ( >>> http://wiki.freeswitch.org/wiki/gsmopen ) >>> then, after installation of libctb and gsmlib (as per wiki), be sure >>> to update your dinamic link cache, or compiler will not find then. >>> >>> Eg: ldconfig >>> >>> >>> On 5/14/12, Chris Mylonas wrote: >>>> All good GM, no inconvenience, just a minor gotcha ;) >>>> I am unable to compile mod_gsmopen though. >>>> >>>> It complains about not being able to find ctb-0.16 >>>> The actual filename is libctbd-0.16.so in /usr/local/lib as you can see from >>>> the 2nd lot of stuff. >>>> >>>> How do I fix this? >>>> >>>> >>>> [root at space mod_gsmopen]# make install >>>> Creating mod_gsmopen.so... >>>> /usr/bin/ld: cannot find -lctb-0.16 >>>> collect2: ld returned 1 exit status >>>> g++ -I../../../../libs/spandsp/src -I../../../..//libs/tiff-3.8.2/libtiff >>>> -DGSMOPEN_C_VER=\"44fd0de\" -DMODGSMOPEN_C_VER=\"44fd0de\" >>>> -I/usr/src/freeswitch/libs/curl/include -I/usr/src/freeswitch/src/include >>>> -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/libs/libteletone/src >>>> -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g >>>> -O2 -D_GNU_SOURCE -shared -o .libs/mod_gsmopen.so -shared -Wl,-x >>>> .libs/mod_gsmopen.o gsmopen_protocol.o >>>> /usr/src/freeswitch/.libs/libfreeswitch.so >>>> -L/usr/src/freeswitch/libs/apr-util/xml/expat/lib >>>> /usr/src/freeswitch/libs/apr-util/xml/expat/lib/.libs/libexpat.a >>>> /usr/src/freeswitch/libs/apr/.libs/libapr-1.a -luuid -lpthread >>>> -L/usr/src/freeswitch/libs/srtp -L/usr/kerberos/lib -lcrypt -lrt -lssl >>>> -lcrypto -ldl -lncurses -L/usr/src/freeswitch/libs/spandsp/src >>>> /usr/src/freeswitch/libs/spandsp/src/.libs/libspandsp.a >>>> -L/usr/src/freeswitch/libs/tiff-3.8.2/libtiff >>>> /usr/src/freeswitch/libs/tiff-3.8.2/libtiff/.libs/libtiff.a -ljpeg -lz -lm >>>> -lc -lctb-0.16 -lgsmme -Wl,--rpath -Wl,/usr/local/freeswitch/lib >>>> -Wl,--rpath -Wl,/usr/local/freeswitch/mod >>>> make[1]: *** [mod_gsmopen.so] Error 1 >>>> make: *** [install] Error 1 >>>> >>>> >>>> >>>> [root at space mod_gsmopen]# ldd /usr/local/lib/libctbd-0.16.so >>>> linux-gate.so.1 => (0x00754000) >>>> libpthread.so.0 => /lib/libpthread.so.0 (0x00e83000) >>>> libstdc++.so.6 => /usr/lib/libstdc++.so.6 (0x00d28000) >>>> libm.so.6 => /lib/libm.so.6 (0x00964000) >>>> libgcc_s.so.1 => /lib/libgcc_s.so.1 (0x00e1f000) >>>> libc.so.6 => /lib/libc.so.6 (0x00110000) >>>> /lib/ld-linux.so.2 (0x003ea000) >>>> >>>> >>>> >>>> On 14/05/2012, at 6:44 PM, Giovanni Maruzzelli wrote: >>>> >>>>> yes, it requires libspandsp, maybe the Makefile it's not yet tweaked >>>>> to build the library automatically. >>>>> >>>>> So, please first build mod_spandsp, then mod_gsmopen. >>>>> >>>>> We'll fixx the Makefile soon, sorry for the inconvenience. >>>>> >>>>> -giovanni >>>>> >>>>> On Mon, May 14, 2012 at 9:30 AM, Chris Mylonas >>>>> wrote: >>>>>> Hi FS List, >>>>>> >>>>>> FYI - as a shortcut to building my freeswitch, I skip spandsp - but it >>>>>> looks >>>>>> like this mod_gsmopen wants it in there. >>>>>> >>>>>> >>>>>> [root at space build]# cd >>>>>> /usr/src/freeswitch/src/mod/endpoints/mod_gsmopen/ >>>>>> [root at space mod_gsmopen]# make clean >>>>>> [root at space mod_gsmopen]# make install >>>>>> Compiling gsmopen_protocol.cpp... >>>>>> Compiling >>>>>> /usr/src/freeswitch/src/mod/endpoints/mod_gsmopen/mod_gsmopen.cpp... >>>>>> mkdir .libs >>>>>> Compiling >>>>>> /usr/src/freeswitch/src/mod/endpoints/mod_gsmopen/mod_gsmopen.cpp >>>>>> ... >>>>>> Creating mod_gsmopen.so... >>>>>> /usr/bin/ld: cannot find -lspandsp >>>>>> collect2: ld returned 1 exit status >>>>>> g++ -I../../../../libs/spandsp/src >>>>>> -I../../../..//libs/tiff-3.8.2/libtiff >>>>>> -DGSMOPEN_C_VER=\"44fd0de\" -DMODGSMOPEN_C_VER=\"44fd0de\" >>>>>> -I/usr/src/freeswitch/libs/curl/include >>>>>> -I/usr/src/freeswitch/src/include >>>>>> -I/usr/src/freeswitch/src/include >>>>>> -I/usr/src/freeswitch/libs/libteletone/src >>>>>> -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 >>>>>> -g >>>>>> -O2 -D_GNU_SOURCE -shared -o .libs/mod_gsmopen.so -shared -Wl,-x >>>>>> .libs/mod_gsmopen.o gsmopen_protocol.o -lm >>>>>> /usr/src/freeswitch/.libs/libfreeswitch.so >>>>>> -L/usr/src/freeswitch/libs/apr-util/xml/expat/lib >>>>>> /usr/src/freeswitch/libs/apr-util/xml/expat/lib/.libs/libexpat.a >>>>>> /usr/src/freeswitch/libs/apr/.libs/libapr-1.a -luuid -lpthread >>>>>> -L/usr/src/freeswitch/libs/srtp -L/usr/kerberos/lib -lcrypt -lrt -lssl >>>>>> -lcrypto -ldl -lz -lncurses -ljpeg >>>>>> -L/usr/src/freeswitch/libs/spandsp/src >>>>>> -lspandsp -lctb-0.16 -lgsmme -Wl,--rpath -Wl,/usr/local/freeswitch/lib >>>>>> -Wl,--rpath -Wl,/usr/local/freeswitch/mod >>>>>> make[1]: *** [mod_gsmopen.so] Error 1 >>>>>> make: *** [install] Error 1 >>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>> >>>>> >>>>> >>>>> -- >>>>> Sincerely, >>>>> >>>>> Giovanni Maruzzelli >>>>> Cell : +39-347-2665618 >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Sincerely, >>> >>> Giovanni Maruzzelli >>> Cell : +39-347-2665618 >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> -- >> Brian D. Foster >> Endigo Computer LLC >> Email: bdfoster at endigotech.com >> Phone: 317-800-7876 >> Indianapolis, Indiana, USA >> >> This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120515/c81e75e8/attachment-0001.html From wesleyakio at tuntscorp.com Tue May 15 15:57:49 2012 From: wesleyakio at tuntscorp.com (Wesley Akio) Date: Tue, 15 May 2012 08:57:49 -0300 Subject: [Freeswitch-users] mod_gsmopen requires spandsp In-Reply-To: <074EF57F-FF04-4BAE-BFD8-A9729C4EBB8B@opencsta.org> References: <1DE50890-EE01-4EEE-83A8-5B5F71BB3959@opencsta.org> <3484676B-F4CA-47E3-A8B0-9743291C1056@opencsta.org> <62D399FB-DFA0-4DD0-B86F-45E95026EF82@opencsta.org> <074EF57F-FF04-4BAE-BFD8-A9729C4EBB8B@opencsta.org> Message-ID: Sent from mobile, sorry for the typos.... Em 14/05/2012 23:37, "Chris Mylonas" escreveu: > Device drivers installed and I get the ttyUSB devices Wesley mentioned in > the previous email. > > I've just recompiled FS (and trying again currently with a ./configure > --with-libctb=/usr/local/lib/libctb-0.16.so in the hope that will inform > FS where to look) and get the following error - which is the same as > yesterday when loading mod_gsmopen > The dependent libs are in place as per the wiki page and run ldconfig > after install. > > 2012-05-15 12:52:44.119783 [CRIT] switch_loadable_module.c:1300 Error > Loading module /usr/local/freeswitch/mod/mod_gsmopen.so > **libctb-0.16.so: cannot open shared object file: No such file or > directory** > > I'm using FreeSWITCH Version 1.2.0 (git-44fd0de 2012-05-14 02-04-36 +0200) > > > I have a CentOS 6 64 bit virtualised system running in USA I will test on > that to see if the module loads. > > > Thanks > Chris > > > On 15/05/2012, at 4:04 AM, Wesley Akio wrote: > > I got to that at some point(E173s-6)... > > Wich distro are you in? > > Anyway, you're probably missing the device drivers. Mine (CentOS 5.8) > looked just like that... with the drivers it gives me: > > crw------- 1 root root 188, 2 Mai 14 15:01 ttyUSB_utps_diag > crw------- 1 root root 188, 1 Mai 14 15:01 ttyUSB_utps_modem > crw------- 1 root root 188, 3 Mai 14 15:01 ttyUSB_utps_pcui > > I got SMS working after that. Modem is, of course, the modem... > > I could'n get audio working though so I have no clue on each of the other > two is the RAW device. > > If you get it to work please let me know! > > Best, > > Wesley Akio > TuntsCorp.com > > > On Mon, May 14, 2012 at 1:31 PM, Chris Mylonas wrote: > >> Hi FS List, >> >> Sorry to badger you with this again. I'm going to sleep, it's 2:30am - >> but I will leave you with more mod_gsmopen testing stuff. >> >> >> Getting this in my logs >> >> 2012-05-15 01:46:11.670033 [CRIT] switch_loadable_module.c:1300 Error >> Loading module /usr/local/freeswitch/mod/mod_gsmopen.so >> **libctb-0.16.so: cannot open shared object file: No such file or >> directory** >> >> It did compile and install and I am running as root. >> >> [root at space log]# ls -l /usr/local/freeswitch/mod/mod_gsmopen.so >> -rwxr-xr-x 1 root root 452284 May 15 01:38 >> /usr/local/freeswitch/mod/mod_gsmopen.so >> >> [root at space log]# ls -l /usr/local/lib/libctb-0.16.so >> -rwxr-xr-x 1 root root 47525 May 15 01:38 /usr/local/lib/libctb-0.16.so >> >> Just for completeness, >> >> Configuration file is from src example, modified only with the USB device >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> And the /dev/usb stuff look like this (usbdev1.4_ep__ are the Huawei E173 >> dongle) - the config file is just my random stab in the dark at getting >> something going. >> >> /dev/usbdev1.1_ep00 >> /dev/usbdev1.1_ep81 >> */dev/usbdev1.4_ep00* >> */dev/usbdev1.4_ep01* >> */dev/usbdev1.4_ep02* >> */dev/usbdev1.4_ep81* >> */dev/usbdev1.4_ep82* >> /dev/usbdev2.1_ep00 >> /dev/usbdev2.1_ep81 >> >> Thanks >> Chris >> >> >> On 15/05/2012, at 12:51 AM, Brian Foster wrote: >> >> if you compile libctb with debug it's actually libctbd, not libctb. >> >> On Mon, May 14, 2012 at 7:53 AM, Chris Mylonas wrote: >> >>> Thanks for the reply Giovanni. I have done the steps on the wiki. >>> >>> tl;dr; should i put a prefix when i'm making the dependent mods so they >>> don't go into /usr/local/lib, or this is why ldconfig is run - to tell the >>> system where the libs are. >>> >>> All the relevant stuff is below >>> >>> Hope you can see something wrong, >>> Cheers >>> Chris >>> >>> e.g. >>> here is my bash history >>> >>> 1050 cd freeswitch/ >>> 1051 ls >>> 1052 find . -name gsmlib >>> 1053 cd src/mod/endpoints/mod_gsmopen/ >>> 1054 ls >>> 1055 cd gsmlib/ >>> 1056 ls >>> 1057 cd gsmlib-1.10-patched-13ubuntu/ >>> 1058 ls >>> * 1059 ./configure* >>> * 1060 make* >>> * 1061 make install* >>> * 1062 ldconfig * >>> * 1063 cd >>> /usr/src/freeswitch/src/mod/endpoints/mod_gsmopen/libctb-0.16/build* >>> * 1064 make DEBUG=1 GPIB=0* >>> * 1065 make DEBUG=1 GPIB=0 install* >>> * 1066 ldconfig* >>> 1067 cd /usr/src/freeswitch/src/mod/endpoints/mod_gsmopen/ >>> 1068 make clean >>> 1069 make install >>> >>> Just to repeat the compilation error >>> Creating mod_gsmopen.so... >>> */usr/bin/ld: cannot find -lctb-0.16* >>> collect2: ld returned 1 exit status >>> g++ -I../../../../libs/spandsp/src >>> -I../../../..//libs/tiff-3.8.2/libtiff -DGSMOPEN_C_VER=\"44fd0de\" >>> -DMODGSMOPEN_C_VER=\"44fd0de\" -I/usr/src/freeswitch/libs/curl/include >>> -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/src/include >>> -I/usr/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden >>> -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -shared >>> -o .libs/mod_gsmopen.so -shared -Wl,-x .libs/mod_gsmopen.o >>> gsmopen_protocol.o /usr/src/freeswitch/.libs/libfreeswitch.so >>> -L/usr/src/freeswitch/libs/apr-util/xml/expat/lib >>> /usr/src/freeswitch/libs/apr-util/xml/expat/lib/.libs/libexpat.a >>> /usr/src/freeswitch/libs/apr/.libs/libapr-1.a -luuid -lpthread >>> -L/usr/src/freeswitch/libs/srtp -L/usr/kerberos/lib -lcrypt -lrt -lssl >>> -lcrypto -ldl -lncurses -L/usr/src/freeswitch/libs/spandsp/src >>> /usr/src/freeswitch/libs/spandsp/src/.libs/libspandsp.a >>> -L/usr/src/freeswitch/libs/tiff-3.8.2/libtiff >>> /usr/src/freeswitch/libs/tiff-3.8.2/libtiff/.libs/libtiff.a -ljpeg -lz -lm >>> -lc -lctb-0.16 -lgsmme -Wl,--rpath -Wl,/usr/local/freeswitch/lib >>> -Wl,--rpath -Wl,/usr/local/freeswitch/mod >>> make[1]: *** [mod_gsmopen.so] Error 1 >>> make: *** [install] Error 1 >>> >>> >>> >>> >>> All the gsmlib stuff that is NOT in the freeswitch src dir is here >>> >>> [root at space mod_gsmopen]# locate gsmlib | grep -v src >>> /usr/local/include/gsmlib >>> /usr/local/include/gsmlib/gsm_at.h >>> /usr/local/include/gsmlib/gsm_cb.h >>> /usr/local/include/gsmlib/gsm_error.h >>> /usr/local/include/gsmlib/gsm_event.h >>> /usr/local/include/gsmlib/gsm_map_key.h >>> /usr/local/include/gsmlib/gsm_me_ta.h >>> /usr/local/include/gsmlib/gsm_parser.h >>> /usr/local/include/gsmlib/gsm_phonebook.h >>> /usr/local/include/gsmlib/gsm_port.h >>> /usr/local/include/gsmlib/gsm_sie_me.h >>> /usr/local/include/gsmlib/gsm_sms.h >>> /usr/local/include/gsmlib/gsm_sms_codec.h >>> /usr/local/include/gsmlib/gsm_sms_store.h >>> /usr/local/include/gsmlib/gsm_sorted_phonebook.h >>> /usr/local/include/gsmlib/gsm_sorted_phonebook_base.h >>> /usr/local/include/gsmlib/gsm_sorted_sms_store.h >>> /usr/local/include/gsmlib/gsm_unix_serial.h >>> /usr/local/include/gsmlib/gsm_util.h >>> /usr/local/share/locale/de/LC_MESSAGES/gsmlib.mo >>> >>> >>> And the ctb stuff is in /usr/local/lib >>> >>> /usr/local/include/ctb-0.16 >>> /usr/local/include/ctb-0.16/ctb.h >>> /usr/local/include/ctb-0.16/fifo.h >>> /usr/local/include/ctb-0.16/getopt.h >>> /usr/local/include/ctb-0.16/iobase.h >>> /usr/local/include/ctb-0.16/linux >>> /usr/local/include/ctb-0.16/portscan.h >>> /usr/local/include/ctb-0.16/serport.h >>> /usr/local/include/ctb-0.16/serportx.h >>> /usr/local/include/ctb-0.16/timer.h >>> /usr/local/include/ctb-0.16/linux/serport.h >>> /usr/local/include/ctb-0.16/linux/timer.h >>> /usr/local/lib/libctbd-0.16.a >>> /usr/local/lib/libctbd-0.16.so >>> >>> >>> gcc version 4.1.2 20080704 (Red Hat 4.1.2-52) >>> GNU Make 3.81 >>> CentOS release 5.8 (Final) >>> >>> >>> >>> >>> On 14/05/2012, at 9:38 PM, Giovanni Maruzzelli wrote: >>> >>> you must first compile and install libctb, as per the wiki page ( >>> http://wiki.freeswitch.org/wiki/gsmopen ) >>> then, after installation of libctb and gsmlib (as per wiki), be sure >>> to update your dinamic link cache, or compiler will not find then. >>> >>> Eg: ldconfig >>> >>> >>> On 5/14/12, Chris Mylonas wrote: >>> >>> All good GM, no inconvenience, just a minor gotcha ;) >>> >>> I am unable to compile mod_gsmopen though. >>> >>> >>> It complains about not being able to find ctb-0.16 >>> >>> The actual filename is libctbd-0.16.so in /usr/local/lib as you can see >>> from >>> >>> the 2nd lot of stuff. >>> >>> >>> How do I fix this? >>> >>> >>> >>> [root at space mod_gsmopen]# make install >>> >>> Creating mod_gsmopen.so... >>> >>> /usr/bin/ld: cannot find -lctb-0.16 >>> >>> collect2: ld returned 1 exit status >>> >>> g++ -I../../../../libs/spandsp/src -I../../../..//libs/tiff-3.8.2/libtiff >>> >>> -DGSMOPEN_C_VER=\"44fd0de\" -DMODGSMOPEN_C_VER=\"44fd0de\" >>> >>> -I/usr/src/freeswitch/libs/curl/include -I/usr/src/freeswitch/src/include >>> >>> -I/usr/src/freeswitch/src/include >>> -I/usr/src/freeswitch/libs/libteletone/src >>> >>> -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 >>> -g >>> >>> -O2 -D_GNU_SOURCE -shared -o .libs/mod_gsmopen.so -shared -Wl,-x >>> >>> .libs/mod_gsmopen.o gsmopen_protocol.o >>> >>> /usr/src/freeswitch/.libs/libfreeswitch.so >>> >>> -L/usr/src/freeswitch/libs/apr-util/xml/expat/lib >>> >>> /usr/src/freeswitch/libs/apr-util/xml/expat/lib/.libs/libexpat.a >>> >>> /usr/src/freeswitch/libs/apr/.libs/libapr-1.a -luuid -lpthread >>> >>> -L/usr/src/freeswitch/libs/srtp -L/usr/kerberos/lib -lcrypt -lrt -lssl >>> >>> -lcrypto -ldl -lncurses -L/usr/src/freeswitch/libs/spandsp/src >>> >>> /usr/src/freeswitch/libs/spandsp/src/.libs/libspandsp.a >>> >>> -L/usr/src/freeswitch/libs/tiff-3.8.2/libtiff >>> >>> /usr/src/freeswitch/libs/tiff-3.8.2/libtiff/.libs/libtiff.a -ljpeg -lz >>> -lm >>> >>> -lc -lctb-0.16 -lgsmme -Wl,--rpath -Wl,/usr/local/freeswitch/lib >>> >>> -Wl,--rpath -Wl,/usr/local/freeswitch/mod >>> >>> make[1]: *** [mod_gsmopen.so] Error 1 >>> >>> make: *** [install] Error 1 >>> >>> >>> >>> >>> [root at space mod_gsmopen]# ldd /usr/local/lib/libctbd-0.16.so >>> >>> linux-gate.so.1 => (0x00754000) >>> >>> libpthread.so.0 => /lib/libpthread.so.0 (0x00e83000) >>> >>> libstdc++.so.6 => /usr/lib/libstdc++.so.6 (0x00d28000) >>> >>> libm.so.6 => /lib/libm.so.6 (0x00964000) >>> >>> libgcc_s.so.1 => /lib/libgcc_s.so.1 (0x00e1f000) >>> >>> libc.so.6 => /lib/libc.so.6 (0x00110000) >>> >>> /lib/ld-linux.so.2 (0x003ea000) >>> >>> >>> >>> >>> On 14/05/2012, at 6:44 PM, Giovanni Maruzzelli wrote: >>> >>> >>> yes, it requires libspandsp, maybe the Makefile it's not yet tweaked >>> >>> to build the library automatically. >>> >>> >>> So, please first build mod_spandsp, then mod_gsmopen. >>> >>> >>> We'll fixx the Makefile soon, sorry for the inconvenience. >>> >>> >>> -giovanni >>> >>> >>> On Mon, May 14, 2012 at 9:30 AM, Chris Mylonas >>> >>> wrote: >>> >>> Hi FS List, >>> >>> >>> FYI - as a shortcut to building my freeswitch, I skip spandsp - but it >>> >>> looks >>> >>> like this mod_gsmopen wants it in there. >>> >>> >>> >>> [root at space build]# cd >>> >>> /usr/src/freeswitch/src/mod/endpoints/mod_gsmopen/ >>> >>> [root at space mod_gsmopen]# make clean >>> >>> [root at space mod_gsmopen]# make install >>> >>> Compiling gsmopen_protocol.cpp... >>> >>> Compiling >>> >>> /usr/src/freeswitch/src/mod/endpoints/mod_gsmopen/mod_gsmopen.cpp... >>> >>> mkdir .libs >>> >>> Compiling >>> >>> /usr/src/freeswitch/src/mod/endpoints/mod_gsmopen/mod_gsmopen.cpp >>> >>> ... >>> >>> Creating mod_gsmopen.so... >>> >>> /usr/bin/ld: cannot find -lspandsp >>> >>> collect2: ld returned 1 exit status >>> >>> g++ -I../../../../libs/spandsp/src >>> >>> -I../../../..//libs/tiff-3.8.2/libtiff >>> >>> -DGSMOPEN_C_VER=\"44fd0de\" -DMODGSMOPEN_C_VER=\"44fd0de\" >>> >>> -I/usr/src/freeswitch/libs/curl/include >>> >>> -I/usr/src/freeswitch/src/include >>> >>> -I/usr/src/freeswitch/src/include >>> >>> -I/usr/src/freeswitch/libs/libteletone/src >>> >>> -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 >>> >>> -g >>> >>> -O2 -D_GNU_SOURCE -shared -o .libs/mod_gsmopen.so -shared -Wl,-x >>> >>> .libs/mod_gsmopen.o gsmopen_protocol.o -lm >>> >>> /usr/src/freeswitch/.libs/libfreeswitch.so >>> >>> -L/usr/src/freeswitch/libs/apr-util/xml/expat/lib >>> >>> /usr/src/freeswitch/libs/apr-util/xml/expat/lib/.libs/libexpat.a >>> >>> /usr/src/freeswitch/libs/apr/.libs/libapr-1.a -luuid -lpthread >>> >>> -L/usr/src/freeswitch/libs/srtp -L/usr/kerberos/lib -lcrypt -lrt -lssl >>> >>> -lcrypto -ldl -lz -lncurses -ljpeg >>> >>> -L/usr/src/freeswitch/libs/spandsp/src >>> >>> -lspandsp -lctb-0.16 -lgsmme -Wl,--rpath -Wl,/usr/local/freeswitch/lib >>> >>> -Wl,--rpath -Wl,/usr/local/freeswitch/mod >>> >>> make[1]: *** [mod_gsmopen.so] Error 1 >>> >>> make: *** [install] Error 1 >>> >>> >>> >>> >>> _________________________________________________________________________ >>> >>> Professional FreeSWITCH Consulting Services: >>> >>> consulting at freeswitch.org >>> >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> >>> http://www.freeswitch.org >>> >>> http://wiki.freeswitch.org >>> >>> http://www.cluecon.com >>> >>> >>> FreeSWITCH-users mailing list >>> >>> FreeSWITCH-users at lists.freeswitch.org >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >>> UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >>> http://www.freeswitch.org >>> >>> >>> >>> >>> >>> -- >>> >>> Sincerely, >>> >>> >>> Giovanni Maruzzelli >>> >>> Cell : +39-347-2665618 >>> >>> >>> _________________________________________________________________________ >>> >>> Professional FreeSWITCH Consulting Services: >>> >>> consulting at freeswitch.org >>> >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> >>> http://www.freeswitch.org >>> >>> http://wiki.freeswitch.org >>> >>> http://www.cluecon.com >>> >>> >>> FreeSWITCH-users mailing list >>> >>> FreeSWITCH-users at lists.freeswitch.org >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >>> http://www.freeswitch.org >>> >>> >>> >>> >>> >>> -- >>> Sincerely, >>> >>> Giovanni Maruzzelli >>> Cell : +39-347-2665618 >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Brian D. Foster >> Endigo Computer LLC >> Email: bdfoster at endigotech.com >> Phone: 317-800-7876 >> Indianapolis, Indiana, USA >> >> This message contains confidential information and is intended for those >> listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If >> you are not the intended recipient you are notified that disclosing, >> copying, distributing or taking any action in reliance on the contents of >> this information is strictly prohibited. E-mail transmission cannot be >> guaranteed to be secure or error-free as information could be intercepted, >> corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. >> The sender therefore does not accept liability for any errors or omissions >> in the contents of this message, which arise as a result of e-mail >> transmission. If verification is required please request a hard-copy >> version. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120515/c29fec1a/attachment-0001.html From wesleyakio at tuntscorp.com Tue May 15 16:01:39 2012 From: wesleyakio at tuntscorp.com (Wesley Akio) Date: Tue, 15 May 2012 09:01:39 -0300 Subject: [Freeswitch-users] mod_gsmopen requires spandsp In-Reply-To: References: <1DE50890-EE01-4EEE-83A8-5B5F71BB3959@opencsta.org> <3484676B-F4CA-47E3-A8B0-9743291C1056@opencsta.org> <62D399FB-DFA0-4DD0-B86F-45E95026EF82@opencsta.org> <074EF57F-FF04-4BAE-BFD8-A9729C4EBB8B@opencsta.org> Message-ID: On CentOS 5 I just edited the Makefile to use ctbd and all went smooth... Kind of... Sent from mobile, sorry for the typos.... Em 14/05/2012 23:37, "Chris Mylonas" escreveu: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120515/d0c63ef7/attachment.html From sdame at 207me.com Tue May 15 16:31:17 2012 From: sdame at 207me.com (Stephen Dame) Date: Tue, 15 May 2012 08:31:17 -0400 Subject: [Freeswitch-users] detect if conference is active In-Reply-To: <007301cd3261$e3b85f60$ab291e20$@com> References: <007301cd3261$e3b85f60$ab291e20$@com> Message-ID: <001f01cd3296$a0bbae30$e2330a90$@com> Bote, I had asked the same question a few weeks ago, If you have the latest freeswitch Will only allow some to join if conference exists. It supports the conf xxxx list count in new versions of freeswitch. I?m using an older version of freeswitch that doesn?t have that list count command yet so I use this, which also allows user to only join an existing conf if they know the number. Regards, Stephen From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Bote Man Sent: Tuesday, May 15, 2012 2:14 AM To: 'FreeSWITCH Users Help' Subject: [Freeswitch-users] detect if conference is active I have a situation where I am dialing a number of outbound calls using the conference module. This works for me. BUT! If someone misses the call or just wants to join the existing outbound conference they can't just dial the extension that I have set up for it since I have coded logic in the dialplan to exclude the conference from dialing the originator--there's no sense in dialing the person who is initiating the conference after all. This is a small list of fixed, known extensions in a closed collection, nothing random. So how could I detect if the conference room is already active so that I can just dump the late entry right into the room instead of falling through my initiator logic? I'm hoping it can all be done in the dialplan because I'm not proficient in the scripting and event socket approaches. As a hacky workaround I could have the users dial a different extension to make it a standard inbound conference, but I want the switch to be smart enough to do the Right Thing when one extension number is dialed, either initiate the outbound group call or join the same existing conference room late. Thanks! Bote http://www.botecomm.com/bote/radio ? my hobby radio pages http://www.trackstreamer.com ? my streaming scanner feeds -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120515/ddc00ea3/attachment.html From koralu at gmail.com Tue May 15 16:33:45 2012 From: koralu at gmail.com (Adrian Andrei) Date: Tue, 15 May 2012 15:33:45 +0300 Subject: [Freeswitch-users] origination_caller_id_number - internal user bridging In-Reply-To: References: Message-ID: I think my approch is wrong. Every modification to effective_caller_id_number or origination_caller_id_number makes the same result to cisco box. It thinks that the caller number is 1001 not 7777. I'm run out of ideas. On 5/15/12, Andrew Cassidy wrote: > try application="export" rather than application="set" > > On 15 May 2012 12:17, Adrian Andrei wrote: > >> Doesn't work. >> >> On 5/15/12, Avi Marcus wrote: >> > I always use effective_caller_id_number >> > >> > -Avi >> > (This message was painstakingly thumbed out on my mobile, so apologies >> for >> > brevity and errors.) >> > >> > On May 15, 2012 1:10 PM, "Adrian Andrei" wrote: >> > >> > Hello, >> > >> > I have one xlite(1000), a FS and Cisco box that is registred as a >> > user(1001) to FS. >> > >> > When I dial ^5555$ from user/1000 I want to bridge the call to >> > user/1001 and change the number to 7777. My dial plan looks: >> > >> > >> > >> > > > data="origination_caller_id_number=7777"/> >> > >> > >> > >> > >> > After the call is made, the cisco sees that the origination number = >> > 1000 instead of 7777 and I think is fair because the sip call is >> > 1000 at ip_cisco:5060. >> > >> > But I want the cisco box to see 7777. Is any solution to make this >> > setup? >> > >> > Ty >> > >> > _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at 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 at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > *Andrew Cassidy BSc (Hons) MBCS SSCA* > Managing Director > > > *T *03300 100 960 > *F > *03300 100 961 > *E *andrew at cassidywebservices.co.uk > *W *www.cassidywebservices.co.uk > From anton.jugatsu at gmail.com Tue May 15 16:38:07 2012 From: anton.jugatsu at gmail.com (Anton Kvashenkin) Date: Tue, 15 May 2012 16:38:07 +0400 Subject: [Freeswitch-users] origination_caller_id_number - internal user bridging In-Reply-To: References: Message-ID: Show your directory config for user 1000. 2012/5/15 Adrian Andrei > I think my approch is wrong. Every modification to > effective_caller_id_number or origination_caller_id_number makes the > same result to cisco box. It thinks that the caller number is 1001 > not 7777. > > I'm run out of ideas. > > On 5/15/12, Andrew Cassidy wrote: > > try application="export" rather than application="set" > > > > On 15 May 2012 12:17, Adrian Andrei wrote: > > > >> Doesn't work. > >> > >> On 5/15/12, Avi Marcus wrote: > >> > I always use effective_caller_id_number > >> > > >> > -Avi > >> > (This message was painstakingly thumbed out on my mobile, so apologies > >> for > >> > brevity and errors.) > >> > > >> > On May 15, 2012 1:10 PM, "Adrian Andrei" wrote: > >> > > >> > Hello, > >> > > >> > I have one xlite(1000), a FS and Cisco box that is registred as a > >> > user(1001) to FS. > >> > > >> > When I dial ^5555$ from user/1000 I want to bridge the call to > >> > user/1001 and change the number to 7777. My dial plan looks: > >> > > >> > > >> > > >> > >> > data="origination_caller_id_number=7777"/> > >> > > >> > > >> > > >> > > >> > After the call is made, the cisco sees that the origination number = > >> > 1000 instead of 7777 and I think is fair because the sip call is > >> > 1000 at ip_cisco:5060. > >> > > >> > But I want the cisco box to see 7777. Is any solution to make this > >> > setup? > >> > > >> > Ty > >> > > >> > > _________________________________________________________________________ > >> > Professional FreeSWITCH Consulting Services: > >> > consulting at freeswitch.org > >> > http://www.freeswitchsolutions.com > >> > > >> > > >> > > >> > > >> > Official FreeSWITCH Sites > >> > http://www.freeswitch.org > >> > http://wiki.freeswitch.org > >> > http://www.cluecon.com > >> > > >> > FreeSWITCH-users mailing list > >> > FreeSWITCH-users at 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 at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > > > > > -- > > *Andrew Cassidy BSc (Hons) MBCS SSCA* > > Managing Director > > > > > > *T *03300 100 960 > > *F > > *03300 100 961 > > *E *andrew at cassidywebservices.co.uk > > *W *www.cassidywebservices.co.uk > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120515/d9052912/attachment-0001.html From avi at avimarcus.net Tue May 15 16:40:00 2012 From: avi at avimarcus.net (Avi Marcus) Date: Tue, 15 May 2012 15:40:00 +0300 Subject: [Freeswitch-users] origination_caller_id_number - internal user bridging In-Reply-To: References: Message-ID: Did you try changing the caller ID type? http://wiki.freeswitch.org/wiki/Cid -- see setting method. -Avi On Tue, May 15, 2012 at 3:33 PM, Adrian Andrei wrote: > I think my approch is wrong. Every modification to > effective_caller_id_number or origination_caller_id_number makes the > same result to cisco box. It thinks that the caller number is 1001 > not 7777. > > I'm run out of ideas. > > On 5/15/12, Andrew Cassidy wrote: > > try application="export" rather than application="set" > > > > On 15 May 2012 12:17, Adrian Andrei wrote: > > > >> Doesn't work. > >> > >> On 5/15/12, Avi Marcus wrote: > >> > I always use effective_caller_id_number > >> > > >> > -Avi > >> > (This message was painstakingly thumbed out on my mobile, so apologies > >> for > >> > brevity and errors.) > >> > > >> > On May 15, 2012 1:10 PM, "Adrian Andrei" wrote: > >> > > >> > Hello, > >> > > >> > I have one xlite(1000), a FS and Cisco box that is registred as a > >> > user(1001) to FS. > >> > > >> > When I dial ^5555$ from user/1000 I want to bridge the call to > >> > user/1001 and change the number to 7777. My dial plan looks: > >> > > >> > > >> > > >> > >> > data="origination_caller_id_number=7777"/> > >> > > >> > > >> > > >> > > >> > After the call is made, the cisco sees that the origination number = > >> > 1000 instead of 7777 and I think is fair because the sip call is > >> > 1000 at ip_cisco:5060. > >> > > >> > But I want the cisco box to see 7777. Is any solution to make this > >> > setup? > >> > > >> > Ty > >> > > >> > > _________________________________________________________________________ > >> > Professional FreeSWITCH Consulting Services: > >> > consulting at freeswitch.org > >> > http://www.freeswitchsolutions.com > >> > > >> > > >> > > >> > > >> > Official FreeSWITCH Sites > >> > http://www.freeswitch.org > >> > http://wiki.freeswitch.org > >> > http://www.cluecon.com > >> > > >> > FreeSWITCH-users mailing list > >> > FreeSWITCH-users at 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 at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > > > > > -- > > *Andrew Cassidy BSc (Hons) MBCS SSCA* > > Managing Director > > > > > > *T *03300 100 960 > > *F > > *03300 100 961 > > *E *andrew at cassidywebservices.co.uk > > *W *www.cassidywebservices.co.uk > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120515/b4b2cb95/attachment.html From dgarcia at anew.com.ve Tue May 15 16:44:39 2012 From: dgarcia at anew.com.ve (Saugort Dario Garcia Tovar) Date: Tue, 15 May 2012 08:14:39 -0430 Subject: [Freeswitch-users] 2012 VoIP With Virtualization In-Reply-To: <550DADC3-9636-42FF-B391-BB00E84AC403@opencsta.org> References: <550DADC3-9636-42FF-B391-BB00E84AC403@opencsta.org> Message-ID: <4FB24FB7.10204@anew.com.ve> Hi, Chris That depend of your enviroment. First you have to take a sort of scientific approach: define, trial and test again and again until the result be consistent. Second tell us about your enviroment: - Real hardware: the server that will run your vm. ?Are blades, small server, etc? VM software used (VMware, OpenVZ, XEN, Jails...)? - ?How you set your VM? cpu, cpu cores, memory, disk, #nic? - ?How many calls have you handle in the real world? - ?Transcoding? - ?Have you tested your VM? Virtualized enviroment for VoIP are growing. I am consultant for contact center services, and I have receive a lot of questions about virtualization. A solution provider can give some guidence about sizing and technical considerations but at the end the final word is: YOU HAVE MAKE A LOAD TEST TO TEST YOUR ENVIROMENT AND IDENTIFY POSSIBLE ISSUES BEFORE GO TO PRODUCTION!!!! Virtualization solutions will do their homework to make VoIP a secure option. Look this paper for wmware: http://blogs.vmware.com/performance/2012/01/voip-performance-on-vsphere-5.html http://www.vmware.com/files/pdf/techpaper/voip-perf-vsphere5.pdf On 5/15/2012 4:42 AM, Chris Mylonas wrote: > Hi FS Users, > > What is the consensus about using virtualized servers for real-time voice (RTP)? Even up until a few years ago it was hard to guarantee the CPU cycles to the voice nodes. > Virtualizing the signalling (SIP) has always been favourable in terms of HA. > > If the entire host is under control, is it safe to say - we can virtualise voice? > Does it depend on what else is shared in the host - say it would be silly to put a high load database server on the same hardware node... > > Thanks for your inputs > Chris > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ----- > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.2171 / Virus Database: 2425/4999 - Release Date: 05/14/12 > > -- Atentamente, *Dario Garc?a* Consultor. CCCT, Nivel C2, Sector Yarey, Mz, Ofc. MZ03a. Caracas-Venezuela. Tel?fono: +58 212 9081842 Cel: +58 412 2221515 dgarcia at anew.com.ve http://www.anew.com.ve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120515/3575716c/attachment.html From wesleyakio at tuntscorp.com Tue May 15 16:49:31 2012 From: wesleyakio at tuntscorp.com (Wesley Akio) Date: Tue, 15 May 2012 09:49:31 -0300 Subject: [Freeswitch-users] 2012 VoIP With Virtualization In-Reply-To: <4FB24FB7.10204@anew.com.ve> References: <550DADC3-9636-42FF-B391-BB00E84AC403@opencsta.org> <4FB24FB7.10204@anew.com.ve> Message-ID: Sent from mobile, sorry for the typos.... Em 15/05/2012 09:42, "Saugort Dario Garcia Tovar" escreveu: > Hi, Chris > > That depend of your enviroment. First you have to take a sort of > scientific approach: define, trial and test again and again until the > result be consistent. > Second tell us about your enviroment: > - Real hardware: the server that will run your vm. ?Are blades, small > server, etc? VM software used (VMware, OpenVZ, XEN, Jails...)? > - ?How you set your VM? cpu, cpu cores, memory, disk, #nic? > - ?How many calls have you handle in the real world? > - ?Transcoding? > - ?Have you tested your VM? > > Virtualized enviroment for VoIP are growing. I am consultant for contact > center services, and I have receive a lot of questions about > virtualization. A solution provider can give some guidence about sizing and > technical considerations but at the end the final word is: YOU HAVE MAKE > A LOAD TEST TO TEST YOUR ENVIROMENT AND IDENTIFY POSSIBLE ISSUES BEFORE GO > TO PRODUCTION!!!! > > Virtualization solutions will do their homework to make VoIP a secure > option. Look this paper for wmware: > > http://blogs.vmware.com/performance/2012/01/voip-performance-on-vsphere-5.html > http://www.vmware.com/files/pdf/techpaper/voip-perf-vsphere5.pdf > > On 5/15/2012 4:42 AM, Chris Mylonas wrote: > > Hi FS Users, > > What is the consensus about using virtualized servers for real-time voice (RTP)? Even up until a few years ago it was hard to guarantee the CPU cycles to the voice nodes. > Virtualizing the signalling (SIP) has always been favourable in terms of HA. > > If the entire host is under control, is it safe to say - we can virtualise voice? > Does it depend on what else is shared in the host - say it would be silly to put a high load database server on the same hardware node... > > Thanks for your inputs > Chris > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > ----- > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.2171 / Virus Database: 2425/4999 - Release Date: 05/14/12 > > > > > > -- > Atentamente, > *Dario Garc?a* > Consultor. > > CCCT, Nivel C2, Sector Yarey, Mz, > Ofc. MZ03a. > Caracas-Venezuela. > Tel?fono: +58 212 9081842 > Cel: +58 412 2221515 > dgarcia at anew.com.ve > http://www.anew.com.ve > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120515/17403631/attachment-0001.html From andrew.paul85 at gmail.com Tue May 15 16:50:13 2012 From: andrew.paul85 at gmail.com (Andrew Paul) Date: Tue, 15 May 2012 18:20:13 +0530 Subject: [Freeswitch-users] Transfer Failure in Proxy Mode Message-ID: Hai , I am using freeswitch in proxy mode . I configured my freeswitch dialplan like this ( http://wiki.freeswitch.org/wiki/Proxy_Media ) In my sip profile i made inbound-proxy-media equals true. When i am calling B from A and i am putting A side hold ,B side am able to get Music on hold ,then from A side am making one Blind transfer to C . This time freeswitch processing the refer and it is sending BYE to B and it is not furthur porcessing INVITE to C . Is there anything required to add more in configuration . -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120515/14bdb69b/attachment.html From wesleyakio at tuntscorp.com Tue May 15 16:54:14 2012 From: wesleyakio at tuntscorp.com (Wesley Akio) Date: Tue, 15 May 2012 09:54:14 -0300 Subject: [Freeswitch-users] 2012 VoIP With Virtualization In-Reply-To: <4FB24FB7.10204@anew.com.ve> References: <550DADC3-9636-42FF-B391-BB00E84AC403@opencsta.org> <4FB24FB7.10204@anew.com.ve> Message-ID: As said before, it depends heavily in your setup... We do virtualize FreeSWITCH and aside from a few glitches it has been good... We have approx 400 calls per VM. Later when I hit the office I can describe my setup in detail. Sent from mobile, sorry for the typos.... Em 15/05/2012 09:42, "Saugort Dario Garcia Tovar" escreveu: > Hi, Chris > > That depend of your enviroment. First you have to take a sort of > scientific approach: define, trial and test again and again until the > result be consistent. > Second tell us about your enviroment: > - Real hardware: the server that will run your vm. ?Are blades, small > server, etc? VM software used (VMware, OpenVZ, XEN, Jails...)? > - ?How you set your VM? cpu, cpu cores, memory, disk, #nic? > - ?How many calls have you handle in the real world? > - ?Transcoding? > - ?Have you tested your VM? > > Virtualized enviroment for VoIP are growing. I am consultant for contact > center services, and I have receive a lot of questions about > virtualization. A solution provider can give some guidence about sizing and > technical considerations but at the end the final word is: YOU HAVE MAKE > A LOAD TEST TO TEST YOUR ENVIROMENT AND IDENTIFY POSSIBLE ISSUES BEFORE GO > TO PRODUCTION!!!! > > Virtualization solutions will do their homework to make VoIP a secure > option. Look this paper for wmware: > > http://blogs.vmware.com/performance/2012/01/voip-performance-on-vsphere-5.html > http://www.vmware.com/files/pdf/techpaper/voip-perf-vsphere5.pdf > > On 5/15/2012 4:42 AM, Chris Mylonas wrote: > > Hi FS Users, > > What is the consensus about using virtualized servers for real-time voice (RTP)? Even up until a few years ago it was hard to guarantee the CPU cycles to the voice nodes. > Virtualizing the signalling (SIP) has always been favourable in terms of HA. > > If the entire host is under control, is it safe to say - we can virtualise voice? > Does it depend on what else is shared in the host - say it would be silly to put a high load database server on the same hardware node... > > Thanks for your inputs > Chris > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > ----- > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.2171 / Virus Database: 2425/4999 - Release Date: 05/14/12 > > > > > > -- > Atentamente, > *Dario Garc?a* > Consultor. > > CCCT, Nivel C2, Sector Yarey, Mz, > Ofc. MZ03a. > Caracas-Venezuela. > Tel?fono: +58 212 9081842 > Cel: +58 412 2221515 > dgarcia at anew.com.ve > http://www.anew.com.ve > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120515/62244d1e/attachment.html From krice at freeswitch.org Tue May 15 17:04:45 2012 From: krice at freeswitch.org (Ken Rice) Date: Tue, 15 May 2012 08:04:45 -0500 Subject: [Freeswitch-users] create rpm fail on freeswitch-1.2.rc2 In-Reply-To: Message-ID: That's because you are not passing the required variables to the spec file... You should look at it a little closer... This whole process is currently not document On 5/15/12 4:55 AM, "haloha" wrote: > hi list > > i am try to build rpm from source and i get error look like this: > > /usr/bin/install -c 'scripts/fsxs' > '/var/tmp/freeswitch-1.2.rc2-%{BUILD_NUMBER}-root-ha/usr/bin/fsxs' > mkdir: cannot create directory > `/var/tmp/freeswitch-1.2.rc2-%{BUILD_NUMBER}-root-ha/usr/lib/freeswitch/mod': > No such file or directory > make[3]: *** [install-exec-local] Error 1 > make[2]: *** [install-am] Error 2 > Making install in src > Making install in mod > > making install mod_abstraction > installing mod_abstraction.so > /usr/bin/install: cannot create regular file > `/var/tmp/freeswitch-1.2.rc2-%{BUILD_NUMBER}-root-ha/usr/lib/freeswitch/mod': > No such file or directory > make[5]: *** > [/var/tmp/freeswitch-1.2.rc2-%{BUILD_NUMBER}-root-ha/usr/lib/freeswitch/mod/mo > d_abstraction.so] > Error 1 > make[4]: *** [install] Error 1 > make[3]: *** [mod_abstraction-install] Error 1 > make[2]: *** [install-recursive] Error 1 > make[1]: *** [install-recursive] Error 1 > make: *** [install] Error 2 > error: Bad exit status from /var/tmp/rpm-tmp.46706 (%install) > > > RPM build errors: > Bad exit status from /var/tmp/rpm-tmp.46706 (%install) > > > please help > thank you > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From chris at opencsta.org Tue May 15 17:08:10 2012 From: chris at opencsta.org (Chris Mylonas) Date: Tue, 15 May 2012 23:08:10 +1000 Subject: [Freeswitch-users] 2012 VoIP With Virtualization In-Reply-To: References: <550DADC3-9636-42FF-B391-BB00E84AC403@opencsta.org> <4FB24FB7.10204@anew.com.ve> Message-ID: OK 400 calls is on the very high end of what I would have put on a VM. If you wouldn't mind explaining your VM set up that would be great. On 15/05/2012, at 10:54 PM, Wesley Akio wrote: > As said before, it depends heavily in your setup... We do virtualize FreeSWITCH and aside from a few glitches it has been good... We have approx 400 calls per VM. Later when I hit the office I can describe my setup in detail. > > > Sent from mobile, sorry for the typos.... > > Em 15/05/2012 09:42, "Saugort Dario Garcia Tovar" escreveu: > Hi, Chris > > That depend of your enviroment. First you have to take a sort of scientific approach: define, trial and test again and again until the result be consistent. > Second tell us about your enviroment: > - Real hardware: the server that will run your vm. ?Are blades, small server, etc? VM software used (VMware, OpenVZ, XEN, Jails...)? > - ?How you set your VM? cpu, cpu cores, memory, disk, #nic? > - ?How many calls have you handle in the real world? > - ?Transcoding? > - ?Have you tested your VM? > > Virtualized enviroment for VoIP are growing. I am consultant for contact center services, and I have receive a lot of questions about virtualization. A solution provider can give some guidence about sizing and technical considerations but at the end the final word is: YOU HAVE MAKE A LOAD TEST TO TEST YOUR ENVIROMENT AND IDENTIFY POSSIBLE ISSUES BEFORE GO TO PRODUCTION!!!! > > Virtualization solutions will do their homework to make VoIP a secure option. Look this paper for wmware: > http://blogs.vmware.com/performance/2012/01/voip-performance-on-vsphere-5.html > http://www.vmware.com/files/pdf/techpaper/voip-perf-vsphere5.pdf > > On 5/15/2012 4:42 AM, Chris Mylonas wrote: >> >> Hi FS Users, >> >> What is the consensus about using virtualized servers for real-time voice (RTP)? Even up until a few years ago it was hard to guarantee the CPU cycles to the voice nodes. >> Virtualizing the signalling (SIP) has always been favourable in terms of HA. >> >> If the entire host is under control, is it safe to say - we can virtualise voice? >> Does it depend on what else is shared in the host - say it would be silly to put a high load database server on the same hardware node... >> >> Thanks for your inputs >> Chris >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> ----- >> No virus found in this message. >> Checked by AVG - www.avg.com >> Version: 2012.0.2171 / Virus Database: 2425/4999 - Release Date: 05/14/12 >> >> > > > -- > Atentamente, > Dario Garc?a > Consultor. > > CCCT, Nivel C2, Sector Yarey, Mz, > Ofc. MZ03a. > Caracas-Venezuela. > Tel?fono: +58 212 9081842 > Cel: +58 412 2221515 > dgarcia at anew.com.ve > http://www.anew.com.ve > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120515/b6c32ed5/attachment-0001.html From bote_radio at botecomm.com Tue May 15 17:16:03 2012 From: bote_radio at botecomm.com (Bote Man) Date: Tue, 15 May 2012 09:16:03 -0400 Subject: [Freeswitch-users] detect if conference is active In-Reply-To: <001f01cd3296$a0bbae30$e2330a90$@com> References: <007301cd3261$e3b85f60$ab291e20$@com> <001f01cd3296$a0bbae30$e2330a90$@com> Message-ID: <011b01cd329c$e2db6b10$a8924130$@com> Excellent! I will try that today. Thanks! Bote From: Stephen Dame Sent: Tuesday, 15 May, 2012 08:31 Bote, I had asked the same question a few weeks ago, If you have the latest freeswitch Will only allow some to join if conference exists. It supports the conf xxxx list count in new versions of freeswitch. I?m using an older version of freeswitch that doesn?t have that list count command yet so I use this, which also allows user to only join an existing conf if they know the number. Regards, Stephen From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Bote Man Sent: Tuesday, May 15, 2012 2:14 AM To: 'FreeSWITCH Users Help' Subject: [Freeswitch-users] detect if conference is active I have a situation where I am dialing a number of outbound calls using the conference module. This works for me. BUT! If someone misses the call or just wants to join the existing outbound conference they can't just dial the extension that I have set up for it since I have coded logic in the dialplan to exclude the conference from dialing the originator--there's no sense in dialing the person who is initiating the conference after all. This is a small list of fixed, known extensions in a closed collection, nothing random. So how could I detect if the conference room is already active so that I can just dump the late entry right into the room instead of falling through my initiator logic? I'm hoping it can all be done in the dialplan because I'm not proficient in the scripting and event socket approaches. As a hacky workaround I could have the users dial a different extension to make it a standard inbound conference, but I want the switch to be smart enough to do the Right Thing when one extension number is dialed, either initiate the outbound group call or join the same existing conference room late. Thanks! Bote http://www.botecomm.com/bote/radio ? my hobby radio pages http://www.trackstreamer.com ? my streaming scanner feeds -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120515/5cc6b158/attachment.html From marcdecorny at gmail.com Tue May 15 17:34:54 2012 From: marcdecorny at gmail.com (Marc de Corny) Date: Tue, 15 May 2012 14:34:54 +0100 Subject: [Freeswitch-users] Querying presence from third party SIP platform Message-ID: Hi All we are using Freeswitch typically to add feature to our core platform in particular around queuing with mod_fifo. we have come up against an issue whereby the FS does not have visibility of the state of the phones as they are registered on another sip platform. I was wondering if there was a way for FS to send SUBSCRIBE messages out through a SIP trunk to the other platform to query the state of the lines? any help is very welcome thanks marc -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120515/b4e3ffbf/attachment.html From chris at opencsta.org Tue May 15 17:43:27 2012 From: chris at opencsta.org (Chris Mylonas) Date: Tue, 15 May 2012 23:43:27 +1000 Subject: [Freeswitch-users] 2012 VoIP With Virtualization In-Reply-To: <4FB24FB7.10204@anew.com.ve> References: <550DADC3-9636-42FF-B391-BB00E84AC403@opencsta.org> <4FB24FB7.10204@anew.com.ve> Message-ID: <9F1CE2B4-DFD1-4733-8DE2-99095869A6FF@opencsta.org> tl;dr; 3 opportunities, none are urgent. virtualization isn't a need-to-have but it would be nice to know what is do-able these days... i've had the best part of 4 years away from 100% voip, and spent 2 years in academia on gentoo plus haven't worked for nearly 12 months just faffing about with J2EE stuff and a bit of recent asterisk/kamailio and now freeswitch. Hi Dario - I am creating a load test at the moment for a queue of 90 calls and comparing FS with asterisk for queuing - just reading over my old notes for SIPp. I am very familiar with asterisk through to 1.4 but have been doing plain old web sys admin with gentoo in academia for 2 years and stayed away from voip and the industry - but times/opportunities are changing up to get back into it. I have lurked in the FS irc channel for about 6 months whilst doing this other stuff and now is the time to pull the finger out so to speak. Current opportunities are 2 potential network providers and a start up i'm the tech guy for: A) This 90 call queue scenario will not be virtualised as it probably would not suit queuing which I understand to be a bit hungry for resources - it will be a failover set up for an Avaya where the avaya is on-site and this hosted-queue will be at the DC. B) I am working on another startup that does purely conferencing and sms (hence my persistence with mod_gsmopen over the last couple of days) C) Am getting some information on a hosted pabx environment where the current set up is 2 asterisk boxes with 700 registrations and about 60 concurrent calls and where to go with it. the people that run that show haven't told me too much - i.e. we want to give access to "IT Guys" to administer their own customers, we just want to bill the minutes VS they have their own hosted apps and want to keep everything 100% theirs. The great thing with (C) is that I will be doing system engineering and no support. I will read through the links you sent. I am unfamiliar with virtualisation in terms of real world usage / performance etc. but have dabbled in a convenience-for-development xen and virtualbox. Using openvz in a "you can manage your own customer handsets" is appealing - i just had a quick read of it all. But yes -- it's a load testing thing at the end of the day, and there can be no blips, or other voiping artifacts (i.e. digitized voice and the like) In general: 1. No transcoding (ever) - it will all be alaw all the way. 2. All servers have Xeon CPUs. Whether they are recent or from the last 5 years is another story depending on the availability of servers and/or cash. I think there are more servers than cash to spend - rackspace is not a problem. That being said, I just priced up some dells and supermicros to have some kind of figures ready. 3. Currently, there are 60 concurrent calls across about 700 registrations. Switches and routers are none of my concern, but they use a range of some cisco switches, a redback router, extremenetworks switch and some juniper firewalls will be in before end of year. There is no rush on any of the above, it's all early days, plus I still have some of this non-computer project work I've been doing - manual labour upgrades...fun fun fun!!! Thanks for your links and comments - I look forward to hearing more. On 15/05/2012, at 10:44 PM, Saugort Dario Garcia Tovar wrote: > Hi, Chris > > That depend of your enviroment. First you have to take a sort of scientific approach: define, trial and test again and again until the result be consistent. > Second tell us about your enviroment: > - Real hardware: the server that will run your vm. ?Are blades, small server, etc? VM software used (VMware, OpenVZ, XEN, Jails...)? > - ?How you set your VM? cpu, cpu cores, memory, disk, #nic? > - ?How many calls have you handle in the real world? > - ?Transcoding? > - ?Have you tested your VM? > > Virtualized enviroment for VoIP are growing. I am consultant for contact center services, and I have receive a lot of questions about virtualization. A solution provider can give some guidence about sizing and technical considerations but at the end the final word is: YOU HAVE MAKE A LOAD TEST TO TEST YOUR ENVIROMENT AND IDENTIFY POSSIBLE ISSUES BEFORE GO TO PRODUCTION!!!! > > Virtualization solutions will do their homework to make VoIP a secure option. Look this paper for wmware: > http://blogs.vmware.com/performance/2012/01/voip-performance-on-vsphere-5.html > http://www.vmware.com/files/pdf/techpaper/voip-perf-vsphere5.pdf > > On 5/15/2012 4:42 AM, Chris Mylonas wrote: >> >> Hi FS Users, >> >> What is the consensus about using virtualized servers for real-time voice (RTP)? Even up until a few years ago it was hard to guarantee the CPU cycles to the voice nodes. >> Virtualizing the signalling (SIP) has always been favourable in terms of HA. >> >> If the entire host is under control, is it safe to say - we can virtualise voice? >> Does it depend on what else is shared in the host - say it would be silly to put a high load database server on the same hardware node... >> >> Thanks for your inputs >> Chris >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> ----- >> No virus found in this message. >> Checked by AVG - www.avg.com >> Version: 2012.0.2171 / Virus Database: 2425/4999 - Release Date: 05/14/12 >> >> > > > -- > Atentamente, > Dario Garc?a > Consultor. > > CCCT, Nivel C2, Sector Yarey, Mz, > Ofc. MZ03a. > Caracas-Venezuela. > Tel?fono: +58 212 9081842 > Cel: +58 412 2221515 > dgarcia at anew.com.ve > http://www.anew.com.ve > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120515/5540bcd7/attachment-0001.html From koralu at gmail.com Tue May 15 17:56:11 2012 From: koralu at gmail.com (Adrian Andrei) Date: Tue, 15 May 2012 16:56:11 +0300 Subject: [Freeswitch-users] origination_caller_id_number - internal user bridging In-Reply-To: References: Message-ID: On 5/15/12, Anton Kvashenkin wrote: > Show your directory config for user 1000. > > 2012/5/15 Adrian Andrei > >> I think my approch is wrong. Every modification to >> effective_caller_id_number or origination_caller_id_number makes the >> same result to cisco box. It thinks that the caller number is 1001 >> not 7777. >> >> I'm run out of ideas. >> >> On 5/15/12, Andrew Cassidy wrote: >> > try application="export" rather than application="set" >> > >> > On 15 May 2012 12:17, Adrian Andrei wrote: >> > >> >> Doesn't work. >> >> >> >> On 5/15/12, Avi Marcus wrote: >> >> > I always use effective_caller_id_number >> >> > >> >> > -Avi >> >> > (This message was painstakingly thumbed out on my mobile, so >> >> > apologies >> >> for >> >> > brevity and errors.) >> >> > >> >> > On May 15, 2012 1:10 PM, "Adrian Andrei" wrote: >> >> > >> >> > Hello, >> >> > >> >> > I have one xlite(1000), a FS and Cisco box that is registred as a >> >> > user(1001) to FS. >> >> > >> >> > When I dial ^5555$ from user/1000 I want to bridge the call to >> >> > user/1001 and change the number to 7777. My dial plan looks: >> >> > >> >> > >> >> > >> >> > > >> > data="origination_caller_id_number=7777"/> >> >> > >> >> > >> >> > >> >> > >> >> > After the call is made, the cisco sees that the origination number = >> >> > 1000 instead of 7777 and I think is fair because the sip call is >> >> > 1000 at ip_cisco:5060. >> >> > >> >> > But I want the cisco box to see 7777. Is any solution to make this >> >> > setup? >> >> > >> >> > Ty >> >> > >> >> > >> _________________________________________________________________________ >> >> > Professional FreeSWITCH Consulting Services: >> >> > consulting at freeswitch.org >> >> > http://www.freeswitchsolutions.com >> >> > >> >> > >> >> > >> >> > >> >> > Official FreeSWITCH Sites >> >> > http://www.freeswitch.org >> >> > http://wiki.freeswitch.org >> >> > http://www.cluecon.com >> >> > >> >> > FreeSWITCH-users mailing list >> >> > FreeSWITCH-users at 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 at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> >> >> >> >> >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> http://wiki.freeswitch.org >> >> http://www.cluecon.com >> >> >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> >> >> > >> > >> > >> > -- >> > *Andrew Cassidy BSc (Hons) MBCS SSCA* >> > Managing Director >> > >> > >> > *T *03300 100 960 >> > *F >> > *03300 100 961 >> > *E *andrew at cassidywebservices.co.uk >> > *W *www.cassidywebservices.co.uk >> > >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > From bclark at grasshopper.com Tue May 15 18:29:05 2012 From: bclark at grasshopper.com (Brett Clark - Grasshopper) Date: Tue, 15 May 2012 09:29:05 -0500 Subject: [Freeswitch-users] Conferences without mod_conference? In-Reply-To: References: Message-ID: Hey, Thanks for the reply. Apparently there is not value in disabling mod_conference. The question arose from ignorance of the system's architecture. :) Thanks for the help! Brett From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Monday, May 14, 2012 5:56 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Conferences without mod_conference? You'll need something to mix the audio of all these channels. If it's not mod_conference then it would need to be on one of the devices you are using. Question: what value is there in disabling mod_conference? It's about as robust an audio conferencing module as you'll find on the planet... -MC On Mon, May 14, 2012 at 2:07 PM, Brett Clark - Grasshopper > wrote: Hello, Is it possible to conference together multiple calls into a conference call without using the mod_conference module? Or am I restricted to bridged calls only when I disable mod_conference? I was just wondering what options I had if I was using an event socket, or the like. Thanks! Brett -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120515/f6083822/attachment.html From cmason at frontiernetworks.ca Tue May 15 18:30:47 2012 From: cmason at frontiernetworks.ca (Colin Mason) Date: Tue, 15 May 2012 10:30:47 -0400 Subject: [Freeswitch-users] Thanks FreeSWITCH In-Reply-To: References: <0D1C698866F66045A6201FD0F59CAC900146149B65@EX.frontier.local> Message-ID: <0D1C698866F66045A6201FD0F59CAC900146149B96@EX.frontier.local> Sure! Give me a week or so and I will document my setup. My next test will be a baremetal freeswitch setup. I will compare the performance between baremetal and virtual machine. Colin From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of SamyGo Sent: Tuesday, May 15, 2012 1:33 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Thanks FreeSWITCH WOW, I'm pretty sure there are other FS deployments giving quality performances, but just 7 days... !! It will be great if this kind of setup-guide is documented for the benefit for all of us here. Regards, Sammy On Tue, May 15, 2012 at 1:42 AM, Michael Collins > wrote: On Mon, May 14, 2012 at 1:32 PM, Ken Rice > wrote: Hey Colin, Could you document your setup on the wiki maybe? K Also, if you want to come join the Wednesday conference call and give us the lowdown on how you pulled this off we'd love to hear about it. You've got much more than just FreeSWITCH in your environment. Putting all those pieces together was probably a lot of "fun". :) -MC _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120515/cf3b2c36/attachment.html From andrew at cassidywebservices.co.uk Tue May 15 18:48:08 2012 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Tue, 15 May 2012 15:48:08 +0100 Subject: [Freeswitch-users] 2012 VoIP With Virtualization In-Reply-To: <9F1CE2B4-DFD1-4733-8DE2-99095869A6FF@opencsta.org> References: <550DADC3-9636-42FF-B391-BB00E84AC403@opencsta.org> <4FB24FB7.10204@anew.com.ve> <9F1CE2B4-DFD1-4733-8DE2-99095869A6FF@opencsta.org> Message-ID: I currently run a testing setup on rackspace cloud, doesn't seem to be too bad, but then again the call volume is low. On 15 May 2012 14:43, Chris Mylonas wrote: > tl;dr; 3 opportunities, none are urgent. virtualization isn't a > need-to-have but it would be nice to know what is do-able these days... > i've had the best part of 4 years away from 100% voip, and spent 2 years in > academia on gentoo plus haven't worked for nearly 12 months just faffing > about with J2EE stuff and a bit of recent asterisk/kamailio and now > freeswitch. > > > > Hi Dario - I am creating a load test at the moment for a queue of 90 calls > and comparing FS with asterisk for queuing - just reading over my old notes > for SIPp. I am very familiar with asterisk through to 1.4 but have been > doing plain old web sys admin with gentoo in academia for 2 years and > stayed away from voip and the industry - but times/opportunities are > changing up to get back into it. I have lurked in the FS irc channel for > about 6 months whilst doing this other stuff and now is the time to pull > the finger out so to speak. > > Current opportunities are 2 potential network providers and a start up i'm > the tech guy for: > A) This 90 call queue scenario will not be virtualised as it probably > would not suit queuing which I understand to be a bit hungry for resources > - it will be a failover set up for an Avaya where the avaya is on-site and > this hosted-queue will be at the DC. > > B) I am working on another startup that does purely conferencing and sms > (hence my persistence with mod_gsmopen over the last couple of days) > > C) Am getting some information on a hosted pabx environment where the > current set up is 2 asterisk boxes with 700 registrations and about 60 > concurrent calls and where to go with it. the people that run that show > haven't told me too much - i.e. we want to give access to "IT Guys" to > administer their own customers, we just want to bill the minutes VS they > have their own hosted apps and want to keep everything 100% theirs. > > The great thing with (C) is that I will be doing system engineering and no > support. > > I will read through the links you sent. I am unfamiliar with > virtualisation in terms of real world usage / performance etc. but have > dabbled in a convenience-for-development xen and virtualbox. > Using openvz in a "you can manage your own customer handsets" is appealing > - i just had a quick read of it all. > > But yes -- it's a load testing thing at the end of the day, and there can > be no blips, or other voiping artifacts (i.e. digitized voice and the like) > > In general: > > 1. No transcoding (ever) - it will all be alaw all the way. > 2. All servers have Xeon CPUs. Whether they are recent or from the last > 5 years is another story depending on the availability of servers and/or > cash. I think there are more servers than cash to spend - rackspace is not > a problem. That being said, I just priced up some dells and supermicros to > have some kind of figures ready. > 3. Currently, there are 60 concurrent calls across about 700 > registrations. > > Switches and routers are none of my concern, but they use a range of some > cisco switches, a redback router, extremenetworks switch and some juniper > firewalls will be in before end of year. > > > There is no rush on any of the above, it's all early days, plus I still > have some of this non-computer project work I've been doing - manual labour > upgrades...fun fun fun!!! > > Thanks for your links and comments - I look forward to hearing more. > > > > On 15/05/2012, at 10:44 PM, Saugort Dario Garcia Tovar wrote: > > Hi, Chris > > That depend of your enviroment. First you have to take a sort of > scientific approach: define, trial and test again and again until the > result be consistent. > Second tell us about your enviroment: > - Real hardware: the server that will run your vm. ?Are blades, small > server, etc? VM software used (VMware, OpenVZ, XEN, Jails...)? > - ?How you set your VM? cpu, cpu cores, memory, disk, #nic? > - ?How many calls have you handle in the real world? > - ?Transcoding? > - ?Have you tested your VM? > > Virtualized enviroment for VoIP are growing. I am consultant for contact > center services, and I have receive a lot of questions about > virtualization. A solution provider can give some guidence about sizing and > technical considerations but at the end the final word is: YOU HAVE MAKE > A LOAD TEST TO TEST YOUR ENVIROMENT AND IDENTIFY POSSIBLE ISSUES BEFORE GO > TO PRODUCTION!!!! > > Virtualization solutions will do their homework to make VoIP a secure > option. Look this paper for wmware: > > http://blogs.vmware.com/performance/2012/01/voip-performance-on-vsphere-5.html > http://www.vmware.com/files/pdf/techpaper/voip-perf-vsphere5.pdf > > On 5/15/2012 4:42 AM, Chris Mylonas wrote: > > Hi FS Users, > > What is the consensus about using virtualized servers for real-time voice (RTP)? Even up until a few years ago it was hard to guarantee the CPU cycles to the voice nodes. > Virtualizing the signalling (SIP) has always been favourable in terms of HA. > > If the entire host is under control, is it safe to say - we can virtualise voice? > Does it depend on what else is shared in the host - say it would be silly to put a high load database server on the same hardware node... > > Thanks for your inputs > Chris > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > ----- > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.2171 / Virus Database: 2425/4999 - Release Date: 05/14/12 > > > > > > -- > Atentamente, > *Dario Garc?a* > Consultor. > > CCCT, Nivel C2, Sector Yarey, Mz, > Ofc. MZ03a. > Caracas-Venezuela. > Tel?fono: +58 212 9081842 > Cel: +58 412 2221515 > dgarcia at anew.com.ve > http://www.anew.com.ve > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T *03300 100 960 *F *03300 100 961 *E *andrew at cassidywebservices.co.uk *W *www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120515/e93202d8/attachment-0001.html From brett at launch3.net Tue May 15 19:03:04 2012 From: brett at launch3.net (Brett Wilson) Date: Tue, 15 May 2012 11:03:04 -0400 Subject: [Freeswitch-users] Adding portaudio Message-ID: <012701cd32ab$d47de7f0$7d79b7d0$@launch3.net> Portaudio seems to be missing from my fusionpbx Ubuntu install. Is there an easy way to get the module without recompiling everything? I don't want to mess up my fs install because I have not had good luck with the build scripts in the past. I'd rather just somehow compile the portaudio module. Is there any way someone can give me a prebuilt module? The pc is i386, 32 bit Pentium 4. ******************************************* Brett Wilson IT Department Launch 3 Ventures, LLC 134 Myer Street Hackensack, NJ 07601 Phone: 877.878.9134 Fax: 646.536.3866 Email: Brett.Wilson at launch3.net AOL IM: Brett.Wilson at launch3.net www.Launch3.net www.Launch3telecom.com ******************************************* Description: Description: Description: Blogger-logo Description: Description: Description: FaceBook-Logo Description: Description: Description: Twitter-Logo Description: Description: Description: GPlus-Logo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120515/0f5e63f4/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... 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Name: not available Type: image/png Size: 3063 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120515/0f5e63f4/attachment-0003.png From anthony.minessale at gmail.com Tue May 15 19:04:09 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 15 May 2012 10:04:09 -0500 Subject: [Freeswitch-users] Thanks FreeSWITCH In-Reply-To: <0D1C698866F66045A6201FD0F59CAC900146149B96@EX.frontier.local> References: <0D1C698866F66045A6201FD0F59CAC900146149B65@EX.frontier.local> <0D1C698866F66045A6201FD0F59CAC900146149B96@EX.frontier.local> Message-ID: With all the attention this thread is getting, I can't resist the chance to remind you about ClueCon in August! My last 2 talks were about tuning FS for performance, I might switch it up this year. On Tue, May 15, 2012 at 9:30 AM, Colin Mason wrote: > Sure! Give me a week or so and I will document my setup. My next test will > be a baremetal freeswitch setup. I will compare the performance between > baremetal and virtual machine. > > > > Colin > > > > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of SamyGo > Sent: Tuesday, May 15, 2012 1:33 AM > To: FreeSWITCH Users Help > > > Subject: Re: [Freeswitch-users] Thanks FreeSWITCH > > > > WOW, > > I'm pretty sure there are other FS deployments giving quality performances, > but just 7 days... !! > > ?It will be great if this kind of setup-guide is documented for the benefit > for all of us here. > Regards, > > Sammy > > > > On Tue, May 15, 2012 at 1:42 AM, Michael Collins wrote: > > > > On Mon, May 14, 2012 at 1:32 PM, Ken Rice wrote: > > Hey Colin, > > Could you document your setup on the wiki maybe? > > K > > > > > > Also, if you want to come join the Wednesday conference call and give us the > lowdown on how you pulled this off we'd love to hear about it. You've got > much more than just FreeSWITCH in your environment. Putting all those pieces > together was probably a lot of "fun". :) > > -MC > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From vipkilla at gmail.com Tue May 15 19:05:54 2012 From: vipkilla at gmail.com (Vik Killa) Date: Tue, 15 May 2012 11:05:54 -0400 Subject: [Freeswitch-users] Adding portaudio In-Reply-To: <012701cd32ab$d47de7f0$7d79b7d0$@launch3.net> References: <012701cd32ab$d47de7f0$7d79b7d0$@launch3.net> Message-ID: make mod_portaudio-install fs_cli>load mod_portaudio -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120515/230d95ab/attachment.html From bdfoster at endigotech.com Tue May 15 19:34:33 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Tue, 15 May 2012 11:34:33 -0400 Subject: [Freeswitch-users] origination_caller_id_number - internal user bridging In-Reply-To: References: Message-ID: couple of things, 1. effective_caller_id_* is already automatically exported to the next leg. 2. you need to be executing the change to the cid inline: 3. if that doesnt work please submit a debug level log and a siptrace http://pastebin.freeswitch.org -BDF On Tue, May 15, 2012 at 9:56 AM, Adrian Andrei wrote: > > > > > > > > > > > > > value="$${outbound_caller_name}"/> > value="$${outbound_caller_id}"/> > > > > > > > On 5/15/12, Anton Kvashenkin wrote: > > Show your directory config for user 1000. > > > > 2012/5/15 Adrian Andrei > > > >> I think my approch is wrong. Every modification to > >> effective_caller_id_number or origination_caller_id_number makes the > >> same result to cisco box. It thinks that the caller number is 1001 > >> not 7777. > >> > >> I'm run out of ideas. > >> > >> On 5/15/12, Andrew Cassidy wrote: > >> > try application="export" rather than application="set" > >> > > >> > On 15 May 2012 12:17, Adrian Andrei wrote: > >> > > >> >> Doesn't work. > >> >> > >> >> On 5/15/12, Avi Marcus wrote: > >> >> > I always use effective_caller_id_number > >> >> > > >> >> > -Avi > >> >> > (This message was painstakingly thumbed out on my mobile, so > >> >> > apologies > >> >> for > >> >> > brevity and errors.) > >> >> > > >> >> > On May 15, 2012 1:10 PM, "Adrian Andrei" wrote: > >> >> > > >> >> > Hello, > >> >> > > >> >> > I have one xlite(1000), a FS and Cisco box that is registred as a > >> >> > user(1001) to FS. > >> >> > > >> >> > When I dial ^5555$ from user/1000 I want to bridge the call to > >> >> > user/1001 and change the number to 7777. My dial plan looks: > >> >> > > >> >> > > >> >> > > >> >> > >> >> > data="origination_caller_id_number=7777"/> > >> >> > > >> >> > > >> >> > > >> >> > > >> >> > After the call is made, the cisco sees that the origination number > = > >> >> > 1000 instead of 7777 and I think is fair because the sip call is > >> >> > 1000 at ip_cisco:5060. > >> >> > > >> >> > But I want the cisco box to see 7777. Is any solution to make this > >> >> > setup? > >> >> > > >> >> > Ty > >> >> > > >> >> > > >> > _________________________________________________________________________ > >> >> > Professional FreeSWITCH Consulting Services: > >> >> > consulting at freeswitch.org > >> >> > http://www.freeswitchsolutions.com > >> >> > > >> >> > > >> >> > > >> >> > > >> >> > Official FreeSWITCH Sites > >> >> > http://www.freeswitch.org > >> >> > http://wiki.freeswitch.org > >> >> > http://www.cluecon.com > >> >> > > >> >> > FreeSWITCH-users mailing list > >> >> > FreeSWITCH-users at 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 at freeswitch.org > >> >> http://www.freeswitchsolutions.com > >> >> > >> >> > >> >> > >> >> > >> >> Official FreeSWITCH Sites > >> >> http://www.freeswitch.org > >> >> http://wiki.freeswitch.org > >> >> http://www.cluecon.com > >> >> > >> >> FreeSWITCH-users mailing list > >> >> FreeSWITCH-users at lists.freeswitch.org > >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >> UNSUBSCRIBE: > >> http://lists.freeswitch.org/mailman/options/freeswitch-users > >> >> http://www.freeswitch.org > >> >> > >> > > >> > > >> > > >> > -- > >> > *Andrew Cassidy BSc (Hons) MBCS SSCA* > >> > Managing Director > >> > > >> > > >> > *T *03300 100 960 > >> > *F > >> > *03300 100 961 > >> > *E *andrew at cassidywebservices.co.uk > >> > *W *www.cassidywebservices.co.uk > >> > > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-800-7876 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120515/a91416d2/attachment.html From bdfoster at endigotech.com Tue May 15 19:35:22 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Tue, 15 May 2012 11:35:22 -0400 Subject: [Freeswitch-users] Conferences without mod_conference? In-Reply-To: References: Message-ID: You have much to learn, grasshopper. On Tue, May 15, 2012 at 10:29 AM, Brett Clark - Grasshopper < bclark at grasshopper.com> wrote: > Hey,**** > > ** ** > > Thanks for the reply. Apparently there is not value in disabling > mod_conference. The question arose from ignorance of the system?s > architecture. :)**** > > ** ** > > Thanks for the help!**** > > Brett**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael > Collins > *Sent:* Monday, May 14, 2012 5:56 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Conferences without mod_conference?**** > > ** ** > > You'll need something to mix the audio of all these channels. If it's not > mod_conference then it would need to be on one of the devices you are using. > > Question: what value is there in disabling mod_conference? It's about as > robust an audio conferencing module as you'll find on the planet... > > -MC**** > > On Mon, May 14, 2012 at 2:07 PM, Brett Clark - Grasshopper < > bclark at grasshopper.com> wrote:**** > > Hello,**** > > **** > > Is it possible to conference together multiple calls into a conference > call without using the mod_conference module? Or am I restricted to > bridged calls only when I disable mod_conference? I was just wondering > what options I had if I was using an event socket, or the like.**** > > **** > > Thanks! > Brett**** > > ** ** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-800-7876 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120515/71cbac27/attachment-0001.html From avi at avimarcus.net Tue May 15 19:39:07 2012 From: avi at avimarcus.net (Avi Marcus) Date: Tue, 15 May 2012 18:39:07 +0300 Subject: [Freeswitch-users] origination_caller_id_number - internal user bridging In-Reply-To: References: Message-ID: A set that is purely used as a bridge variable never needs to be inline, because bridges are never inline. -Avi On Tue, May 15, 2012 at 6:34 PM, Brian Foster wrote: > couple of things, > > 1. effective_caller_id_* is already automatically exported to the next leg. > 2. you need to be executing the change to the cid inline: > inline="true"/> > 3. if that doesnt work please submit a debug level log and a siptrace > http://pastebin.freeswitch.org > > -BDF > > > On Tue, May 15, 2012 at 9:56 AM, Adrian Andrei wrote: > >> >> >> >> >> >> >> >> >> >> >> >> >> > value="$${outbound_caller_name}"/> >> > value="$${outbound_caller_id}"/> >> >> >> >> >> >> >> On 5/15/12, Anton Kvashenkin wrote: >> > Show your directory config for user 1000. >> > >> > 2012/5/15 Adrian Andrei >> > >> >> I think my approch is wrong. Every modification to >> >> effective_caller_id_number or origination_caller_id_number makes the >> >> same result to cisco box. It thinks that the caller number is 1001 >> >> not 7777. >> >> >> >> I'm run out of ideas. >> >> >> >> On 5/15/12, Andrew Cassidy wrote: >> >> > try application="export" rather than application="set" >> >> > >> >> > On 15 May 2012 12:17, Adrian Andrei wrote: >> >> > >> >> >> Doesn't work. >> >> >> >> >> >> On 5/15/12, Avi Marcus wrote: >> >> >> > I always use effective_caller_id_number >> >> >> > >> >> >> > -Avi >> >> >> > (This message was painstakingly thumbed out on my mobile, so >> >> >> > apologies >> >> >> for >> >> >> > brevity and errors.) >> >> >> > >> >> >> > On May 15, 2012 1:10 PM, "Adrian Andrei" >> wrote: >> >> >> > >> >> >> > Hello, >> >> >> > >> >> >> > I have one xlite(1000), a FS and Cisco box that is registred as a >> >> >> > user(1001) to FS. >> >> >> > >> >> >> > When I dial ^5555$ from user/1000 I want to bridge the call to >> >> >> > user/1001 and change the number to 7777. My dial plan looks: >> >> >> > >> >> >> > >> >> >> > >> >> >> > > >> >> > data="origination_caller_id_number=7777"/> >> >> >> > >> >> >> > >> >> >> > >> >> >> > >> >> >> > After the call is made, the cisco sees that the origination >> number = >> >> >> > 1000 instead of 7777 and I think is fair because the sip call is >> >> >> > 1000 at ip_cisco:5060. >> >> >> > >> >> >> > But I want the cisco box to see 7777. Is any solution to make this >> >> >> > setup? >> >> >> > >> >> >> > Ty >> >> >> > >> >> >> > >> >> >> _________________________________________________________________________ >> >> >> > Professional FreeSWITCH Consulting Services: >> >> >> > consulting at freeswitch.org >> >> >> > http://www.freeswitchsolutions.com >> >> >> > >> >> >> > >> >> >> > >> >> >> > >> >> >> > Official FreeSWITCH Sites >> >> >> > http://www.freeswitch.org >> >> >> > http://wiki.freeswitch.org >> >> >> > http://www.cluecon.com >> >> >> > >> >> >> > FreeSWITCH-users mailing list >> >> >> > FreeSWITCH-users at 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 at freeswitch.org >> >> >> http://www.freeswitchsolutions.com >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> Official FreeSWITCH Sites >> >> >> http://www.freeswitch.org >> >> >> http://wiki.freeswitch.org >> >> >> http://www.cluecon.com >> >> >> >> >> >> FreeSWITCH-users mailing list >> >> >> FreeSWITCH-users at lists.freeswitch.org >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> UNSUBSCRIBE: >> >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> >> http://www.freeswitch.org >> >> >> >> >> > >> >> > >> >> > >> >> > -- >> >> > *Andrew Cassidy BSc (Hons) MBCS SSCA* >> >> > Managing Director >> >> > >> >> > >> >> > *T *03300 100 960 >> >> > *F >> >> > *03300 100 961 >> >> > *E *andrew at cassidywebservices.co.uk >> >> > *W *www.cassidywebservices.co.uk >> >> > >> >> >> >> >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> >> >> >> >> >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> http://wiki.freeswitch.org >> >> http://www.cluecon.com >> >> >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at 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 at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Brian D. Foster > Endigo Computer LLC > Email: bdfoster at endigotech.com > Phone: 317-800-7876 > Indianapolis, Indiana, USA > > This message contains confidential information and is intended for those > listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If > you are not the intended recipient you are notified that disclosing, > copying, distributing or taking any action in reliance on the contents of > this information is strictly prohibited. E-mail transmission cannot be > guaranteed to be secure or error-free as information could be intercepted, > corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. > The sender therefore does not accept liability for any errors or omissions > in the contents of this message, which arise as a result of e-mail > transmission. If verification is required please request a hard-copy > version. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120515/778ec9ac/attachment.html From bdfoster at endigotech.com Tue May 15 19:40:47 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Tue, 15 May 2012 11:40:47 -0400 Subject: [Freeswitch-users] Adding portaudio In-Reply-To: References: <012701cd32ab$d47de7f0$7d79b7d0$@launch3.net> Message-ID: also, make sure you are editing modules.conf and modules.conf.xml to keep the changes between restarts and upgrades. -BDF On Tue, May 15, 2012 at 11:05 AM, Vik Killa wrote: > make mod_portaudio-install > > fs_cli>load mod_portaudio > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-800-7876 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120515/76459fff/attachment-0001.html From cristian.re at adctelecom.it Tue May 15 18:23:30 2012 From: cristian.re at adctelecom.it (Cristian Re) Date: Tue, 15 May 2012 16:23:30 +0200 Subject: [Freeswitch-users] Detect answer on FXO device Message-ID: <4FB266E2.4060408@adctelecom.it> I have a Grandstream GXW4108 device, connected to a PSTN line and I use it to make outbound calls via freeswitch. The problem I have is that this device sends SIP 200/OK message immediatly after freeswitch INVITE. There is a way to configure freeswitch to detect the answer (with voice detetection?) and propagate the 200/OK message only on answered calls? Thanks Cristian From gabe at gundy.org Tue May 15 21:19:28 2012 From: gabe at gundy.org (Gabriel Gunderson) Date: Tue, 15 May 2012 11:19:28 -0600 Subject: [Freeswitch-users] Thanks FreeSWITCH In-Reply-To: References: <0D1C698866F66045A6201FD0F59CAC900146149B65@EX.frontier.local> <0D1C698866F66045A6201FD0F59CAC900146149B96@EX.frontier.local> Message-ID: On Tue, May 15, 2012 at 9:04 AM, Anthony Minessale wrote: > With all the attention this thread is getting, I can't resist the > chance to remind you about ClueCon in August! > My last 2 talks were about tuning FS for performance, I might switch > it up this year. Link for the lazy :) http://www.cluecon.com/ I hope to see you all there! Gabe From bclark at grasshopper.com Tue May 15 21:35:49 2012 From: bclark at grasshopper.com (Brett Clark - Grasshopper) Date: Tue, 15 May 2012 12:35:49 -0500 Subject: [Freeswitch-users] 2012 VoIP With Virtualization In-Reply-To: References: <550DADC3-9636-42FF-B391-BB00E84AC403@opencsta.org> <4FB24FB7.10204@anew.com.ve> <9F1CE2B4-DFD1-4733-8DE2-99095869A6FF@opencsta.org> Message-ID: What volume are you able to do on Rackspace? And, if I could ask, what service tier/# of cores/OS are you using there? Thanks! From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Andrew Cassidy Sent: Tuesday, May 15, 2012 10:48 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] 2012 VoIP With Virtualization I currently run a testing setup on rackspace cloud, doesn't seem to be too bad, but then again the call volume is low. On 15 May 2012 14:43, Chris Mylonas > wrote: tl;dr; 3 opportunities, none are urgent. virtualization isn't a need-to-have but it would be nice to know what is do-able these days... i've had the best part of 4 years away from 100% voip, and spent 2 years in academia on gentoo plus haven't worked for nearly 12 months just faffing about with J2EE stuff and a bit of recent asterisk/kamailio and now freeswitch. Hi Dario - I am creating a load test at the moment for a queue of 90 calls and comparing FS with asterisk for queuing - just reading over my old notes for SIPp. I am very familiar with asterisk through to 1.4 but have been doing plain old web sys admin with gentoo in academia for 2 years and stayed away from voip and the industry - but times/opportunities are changing up to get back into it. I have lurked in the FS irc channel for about 6 months whilst doing this other stuff and now is the time to pull the finger out so to speak. Current opportunities are 2 potential network providers and a start up i'm the tech guy for: A) This 90 call queue scenario will not be virtualised as it probably would not suit queuing which I understand to be a bit hungry for resources - it will be a failover set up for an Avaya where the avaya is on-site and this hosted-queue will be at the DC. B) I am working on another startup that does purely conferencing and sms (hence my persistence with mod_gsmopen over the last couple of days) C) Am getting some information on a hosted pabx environment where the current set up is 2 asterisk boxes with 700 registrations and about 60 concurrent calls and where to go with it. the people that run that show haven't told me too much - i.e. we want to give access to "IT Guys" to administer their own customers, we just want to bill the minutes VS they have their own hosted apps and want to keep everything 100% theirs. The great thing with (C) is that I will be doing system engineering and no support. I will read through the links you sent. I am unfamiliar with virtualisation in terms of real world usage / performance etc. but have dabbled in a convenience-for-development xen and virtualbox. Using openvz in a "you can manage your own customer handsets" is appealing - i just had a quick read of it all. But yes -- it's a load testing thing at the end of the day, and there can be no blips, or other voiping artifacts (i.e. digitized voice and the like) In general: 1. No transcoding (ever) - it will all be alaw all the way. 2. All servers have Xeon CPUs. Whether they are recent or from the last 5 years is another story depending on the availability of servers and/or cash. I think there are more servers than cash to spend - rackspace is not a problem. That being said, I just priced up some dells and supermicros to have some kind of figures ready. 3. Currently, there are 60 concurrent calls across about 700 registrations. Switches and routers are none of my concern, but they use a range of some cisco switches, a redback router, extremenetworks switch and some juniper firewalls will be in before end of year. There is no rush on any of the above, it's all early days, plus I still have some of this non-computer project work I've been doing - manual labour upgrades...fun fun fun!!! Thanks for your links and comments - I look forward to hearing more. On 15/05/2012, at 10:44 PM, Saugort Dario Garcia Tovar wrote: Hi, Chris That depend of your enviroment. First you have to take a sort of scientific approach: define, trial and test again and again until the result be consistent. Second tell us about your enviroment: - Real hardware: the server that will run your vm. ?Are blades, small server, etc? VM software used (VMware, OpenVZ, XEN, Jails...)? - ?How you set your VM? cpu, cpu cores, memory, disk, #nic? - ?How many calls have you handle in the real world? - ?Transcoding? - ?Have you tested your VM? Virtualized enviroment for VoIP are growing. I am consultant for contact center services, and I have receive a lot of questions about virtualization. A solution provider can give some guidence about sizing and technical considerations but at the end the final word is: YOU HAVE MAKE A LOAD TEST TO TEST YOUR ENVIROMENT AND IDENTIFY POSSIBLE ISSUES BEFORE GO TO PRODUCTION!!!! Virtualization solutions will do their homework to make VoIP a secure option. Look this paper for wmware: http://blogs.vmware.com/performance/2012/01/voip-performance-on-vsphere-5.html http://www.vmware.com/files/pdf/techpaper/voip-perf-vsphere5.pdf On 5/15/2012 4:42 AM, Chris Mylonas wrote: Hi FS Users, What is the consensus about using virtualized servers for real-time voice (RTP)? Even up until a few years ago it was hard to guarantee the CPU cycles to the voice nodes. Virtualizing the signalling (SIP) has always been favourable in terms of HA. If the entire host is under control, is it safe to say - we can virtualise voice? Does it depend on what else is shared in the host - say it would be silly to put a high load database server on the same hardware node... Thanks for your inputs Chris _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ----- No virus found in this message. Checked by AVG - www.avg.com Version: 2012.0.2171 / Virus Database: 2425/4999 - Release Date: 05/14/12 -- Atentamente, Dario Garc?a Consultor. CCCT, Nivel C2, Sector Yarey, Mz, Ofc. MZ03a. Caracas-Venezuela. Tel?fono: +58 212 9081842 Cel: +58 412 2221515 dgarcia at anew.com.ve http://www.anew.com.ve _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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 at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Andrew Cassidy BSc (Hons) MBCS SSCA Managing Director [http://www.cassidywebservices.co.uk/media/emailsig.png] T 03300 100 960 F 03300 100 961 E andrew at cassidywebservices.co.uk W www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120515/352d6d4d/attachment-0001.html From msc at freeswitch.org Tue May 15 22:23:02 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 15 May 2012 11:23:02 -0700 Subject: [Freeswitch-users] Detect answer on FXO device In-Reply-To: <4FB266E2.4060408@adctelecom.it> References: <4FB266E2.4060408@adctelecom.it> Message-ID: Technically, answer detection is the job of the ATA. The problem is that it is an inexact science. I would look in the GXW docs to see if they have a section on 'answer supervision'. It might also be in a section along with 'disconnect supervision.' -MC On Tue, May 15, 2012 at 7:23 AM, Cristian Re wrote: > I have a Grandstream GXW4108 device, connected to a PSTN line and I use > it to make outbound calls via freeswitch. > The problem I have is that this device sends SIP 200/OK message > immediatly after freeswitch INVITE. > There is a way to configure freeswitch to detect the answer (with voice > detetection?) and propagate the 200/OK message only on answered calls? > > Thanks > Cristian > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120515/738e6f63/attachment.html From brett at launch3.net Tue May 15 22:55:22 2012 From: brett at launch3.net (Brett Wilson) Date: Tue, 15 May 2012 14:55:22 -0400 Subject: [Freeswitch-users] Adding portaudio In-Reply-To: References: <012701cd32ab$d47de7f0$7d79b7d0$@launch3.net> Message-ID: <0da701cd32cc$47df0dd0$d79d2970$@launch3.net> I need the source for this correct? I do not think this fusionpbx distro includes source. Where should I get it from? ******************************************* Brett Wilson IT Department Launch 3 Ventures, LLC 134 Myer Street Hackensack, NJ 07601 Phone: 877.878.9134 Fax: 646.536.3866 Email: Brett.Wilson at launch3.net AOL IM: Brett.Wilson at launch3.net www.Launch3.net www.Launch3telecom.com ******************************************* Description: Description: Description: Blogger-logo Description: Description: Description: FaceBook-Logo Description: Description: Description: Twitter-Logo Description: Description: Description: GPlus-Logo From: Vik Killa [mailto:vipkilla at gmail.com] Sent: Tuesday, May 15, 2012 11:06 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Adding portaudio make mod_portaudio-install fs_cli>load mod_portaudio -------------- next part -------------- An HTML attachment was scrubbed... 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Name: not available Type: image/png Size: 3063 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120515/40a24e01/attachment-0007.png From msc at freeswitch.org Tue May 15 23:00:21 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 15 May 2012 12:00:21 -0700 Subject: [Freeswitch-users] Adding portaudio In-Reply-To: <0da701cd32cc$47df0dd0$d79d2970$@launch3.net> References: <012701cd32ab$d47de7f0$7d79b7d0$@launch3.net> <0da701cd32cc$47df0dd0$d79d2970$@launch3.net> Message-ID: I'm almost positive that FusionPBX does a git clone. Look in the modules section. (I forget the actual menu - it's been a while.) Also, check out #FusionPBX on irc.freenode.net -MC On Tue, May 15, 2012 at 11:55 AM, Brett Wilson wrote: > I need the source for this correct? I do not think this fusionpbx distro > includes source. Where should I get it from?**** > > ** ** > > ******************************************* > *Brett Wilson* > *IT Department* > *Launch 3 Ventures, LLC* > 134 Myer Street > Hackensack, NJ 07601 > *Phone:* 877.878.9134 > *Fax:* 646.536.3866 > *Email:* Brett.Wilson at launch3.net > *AOL IM:* Brett.Wilson at launch3.net > www.Launch3.net > *www.Launch3telecom.com * > ******************************************* > [image: Description: Description: Description: Blogger-logo][image: > Description: Description: Description: FaceBook-Logo][image: > Description: Description: Description: Twitter-Logo][image: > Description: Description: Description: GPlus-Logo] > **** > > ** ** > > *From:* Vik Killa [mailto:vipkilla at gmail.com] > *Sent:* Tuesday, May 15, 2012 11:06 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Adding portaudio**** > > ** ** > > make mod_portaudio-install**** > > ** ** > > fs_cli>load mod_portaudio**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... 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Name: not available Type: image/png Size: 3063 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120515/ee82b621/attachment-0003.png From chris at opencsta.org Tue May 15 23:05:10 2012 From: chris at opencsta.org (Chris Mylonas) Date: Wed, 16 May 2012 05:05:10 +1000 Subject: [Freeswitch-users] FreeSWITCH Queue Scenario - Caller Puts MoH on hold Message-ID: <2AA25D7C-E49F-4597-8C3F-1EAE48646F75@opencsta.org> Hi FS Users, What happens when the caller that is waiting in the queue puts the MoH on hold.....and then the Internet drops out. Does FS detect that the caller is gone? The reason I ask is I was doing some testing with __asterisk__ and a couple of soft phones, a snom, and had SIPp scenarios for agents on a remote machine. My snom had 6 calls in the queue (but they were on hold) and my router gave up. On reconnecting, my phone had also rebooted. I dialled into the queue to my surprise I was the 10th caller. I put another half a dozen calls in the queue and then rebooted it to see if the queue would recognise the drop out. It didn't. Is it known that FreeSWITCH works better than this behaviour? I will test it myself - I'm just still in early cookbook pages with it though.... Thanks for the interest, Chris From bdfoster at endigotech.com Wed May 16 00:24:56 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Tue, 15 May 2012 16:24:56 -0400 Subject: [Freeswitch-users] Adding portaudio In-Reply-To: References: <012701cd32ab$d47de7f0$7d79b7d0$@launch3.net> <0da701cd32cc$47df0dd0$d79d2970$@launch3.net> Message-ID: Yes it is downloaded via git if you are using soapee's script (and the CD as it is based on soapee's script) Also the instructions on their site have you download the master branch. -BDF On May 15, 2012 3:01 PM, "Michael Collins" wrote: > I'm almost positive that FusionPBX does a git clone. Look in the modules > section. (I forget the actual menu - it's been a while.) Also, check out > #FusionPBX on irc.freenode.net > > -MC > > On Tue, May 15, 2012 at 11:55 AM, Brett Wilson wrote: > >> I need the source for this correct? I do not think this fusionpbx distro >> includes source. Where should I get it from?**** >> >> ** ** >> >> ******************************************* >> *Brett Wilson* >> *IT Department* >> *Launch 3 Ventures, LLC* >> 134 Myer Street >> Hackensack, NJ 07601 >> *Phone:* 877.878.9134 >> *Fax:* 646.536.3866 >> *Email:* Brett.Wilson at launch3.net >> *AOL IM:* Brett.Wilson at launch3.net >> www.Launch3.net >> *www.Launch3telecom.com * >> ******************************************* >> [image: Description: Description: Description: Blogger-logo][image: >> Description: Description: Description: FaceBook-Logo][image: >> Description: Description: Description: Twitter-Logo][image: >> Description: Description: Description: GPlus-Logo] >> **** >> >> ** ** >> >> *From:* Vik Killa [mailto:vipkilla at gmail.com] >> *Sent:* Tuesday, May 15, 2012 11:06 AM >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] Adding portaudio**** >> >> ** ** >> >> make mod_portaudio-install**** >> >> ** ** >> >> fs_cli>load mod_portaudio**** >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... 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Name: not available Type: image/png Size: 2952 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120515/737ed087/attachment-0007.png From spencer at 5ninesolutions.com Wed May 16 01:10:03 2012 From: spencer at 5ninesolutions.com (Spencer Thomason) Date: Tue, 15 May 2012 14:10:03 -0700 Subject: [Freeswitch-users] Subscribe for MWI In-Reply-To: <8b33f052-3895-4252-8f47-6ce672e810c6@blur> References: <9D1F6AAA-D64F-45BC-A70D-C6E469D38C30@5ninesolutions.com> <8b33f052-3895-4252-8f47-6ce672e810c6@blur> Message-ID: <850DE19F-0F8B-4E4F-9B22-534311287478@5ninesolutions.com> I updated to latest git and everything is working correctly. Thanks, Spencer On May 14, 2012, at 7:47 PM, Spencer Thomason wrote: > I am not. I am using a build from February. I will update in the morning and retest. > > Thanks, > Spencer > > Connected by DROID on Verizon Wireless > > > -----Original message----- > From: Anthony Minessale > To: FreeSWITCH Users Help > Sent: Tue, May 15, 2012 01:33:43 GMT+00:00 > Subject: Re: [Freeswitch-users] Subscribe for MWI > > are you on the latest GIT HEAD? Looking at the code suggests that > should work. Its subbing to message-summary events? > > > On Mon, May 14, 2012 at 4:43 PM, Spencer Thomason > wrote: > > Hello, > > I'm working on setting up FreeSWITCH as a media server behind Kamailio. All endpoints register to Kamailio and SUBSCRIBEs are forwarded to FreeSWITCH for MWI. FreeSWITCH is unaware of any registrations. When an endpoint has an active subscription and a new message is left, the NOTIFY is sent and everything works correctly. The problem arises when there is no subscription and a message is left. In this case when the endpoint creates an active subscription after the message, there is no NOTIFY sent. The there a way I can send NOTIFYs upon creation of a subscription if a message exists? Is there a better way to accomplish this? I'd like to avoid forwarding REGISTERs if possible. > > > > Thanks, > > Spencer > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120515/3dbef7f3/attachment.html From jmesquita at freeswitch.org Wed May 16 01:15:47 2012 From: jmesquita at freeswitch.org (=?utf-8?Q?Jo=C3=A3o_Mesquita?=) Date: Tue, 15 May 2012 18:15:47 -0300 Subject: [Freeswitch-users] Thanks FreeSWITCH In-Reply-To: References: <0D1C698866F66045A6201FD0F59CAC900146149B65@EX.frontier.local> <0D1C698866F66045A6201FD0F59CAC900146149B96@EX.frontier.local> Message-ID: Why shouldn't you? You are one of the creators of this little monster we all love anyway? -- Jo?o Mesquita Sent with Sparrow (http://www.sparrowmailapp.com/?sig) On Tuesday, May 15, 2012 at 12:04 PM, Anthony Minessale wrote: > With all the attention this thread is getting, I can't resist the > chance to remind you about ClueCon in August! > My last 2 talks were about tuning FS for performance, I might switch > it up this year. > > > On Tue, May 15, 2012 at 9:30 AM, Colin Mason wrote: > > Sure! Give me a week or so and I will document my setup. My next test will > > be a baremetal freeswitch setup. I will compare the performance between > > baremetal and virtual machine. > > > > > > > > Colin > > > > > > > > From: freeswitch-users-bounces at lists.freeswitch.org (mailto:freeswitch-users-bounces at lists.freeswitch.org) > > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of SamyGo > > Sent: Tuesday, May 15, 2012 1:33 AM > > To: FreeSWITCH Users Help > > > > > > Subject: Re: [Freeswitch-users] Thanks FreeSWITCH > > > > > > > > WOW, > > > > I'm pretty sure there are other FS deployments giving quality performances, > > but just 7 days... !! > > > > It will be great if this kind of setup-guide is documented for the benefit > > for all of us here. > > Regards, > > > > Sammy > > > > > > > > On Tue, May 15, 2012 at 1:42 AM, Michael Collins wrote: > > > > > > > > On Mon, May 14, 2012 at 1:32 PM, Ken Rice wrote: > > > > Hey Colin, > > > > Could you document your setup on the wiki maybe? > > > > K > > > > > > > > > > > > Also, if you want to come join the Wednesday conference call and give us the > > lowdown on how you pulled this off we'd love to hear about it. You've got > > much more than just FreeSWITCH in your environment. Putting all those pieces > > together was probably a lot of "fun". :) > > > > -MC > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at 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 at freeswitch.org (mailto:consulting at freeswitch.org) > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com (mailto:anthony_minessale at hotmail.com) > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com (mailto:anthony.minessale at gmail.com) > IRC: irc.freenode.net (http://irc.freenode.net) #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org (mailto:888 at conference.freeswitch.org) > googletalk:conf+888 at conference.freeswitch.org (mailto:conf+888 at conference.freeswitch.org) > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120515/73f4f03a/attachment-0001.html From gustavomarsico at gmail.com Wed May 16 02:19:30 2012 From: gustavomarsico at gmail.com (=?iso-8859-1?Q?Gustavo_M=E1rsico?=) Date: Tue, 15 May 2012 19:19:30 -0300 Subject: [Freeswitch-users] mod_callcenter real answer In-Reply-To: References: Message-ID: <38F79ADF-3F23-4E66-B41A-310CE4094218@gmail.com> Hi Evginey I made that change a few days ago, due a scenario where dialer is used. In fact, dialer dials to a queue first instead of customer. All I did was replace in the code "answer" by "pre_answer" in some place. I've no the code right now, but I can send it to you later. Regards Gustavo On May 15, 2012, at 6:15 AM, ??????? wrote: > Hi, colleagues. > > Could I have real answer for client on mod_callcenter, when agent answers? > Or there is another way to queue call between 2 FreeSwitch. > For example, client's call has to be routed to two FS servers, which has different agents in queues. But answer comes from first server when application CALLCENTER is called. How can I do this? > > Best Regards, Evginey. > http://blog.buchnev.ru > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120515/68072867/attachment.html From chris at gonumina.com Wed May 16 02:51:03 2012 From: chris at gonumina.com (Chris Ferreira) Date: Tue, 15 May 2012 18:51:03 -0400 Subject: [Freeswitch-users] Fax Issues with Cisco SPA112 and T.38 In-Reply-To: References: Message-ID: Hello Brian and Everyone, Here is a link to the siptrace output. Sorry for the delay, it was my birthday and I was away for the weekend. http://pastebin.freeswitch.org/19057 Thanks, -Chris On Thu, May 10, 2012 at 10:16 PM, Brian Foster wrote: > 1. Run fs_cli and do 'sofia global siptrace on' without quotes. Stay in > the cli and run a call in an instance where it would fail. Make sure your > scroll buffer on your SSH terminal (PuTTY or what ever you have) to > unlimited or something bog like 20K lines. Copy the logs out and put them > on http://pastebin.freeswitch.org. If prompted for a username/password, > please read the dialog box. SpanDSP doesn't have a great ability to be > degugged on the click but it will give is something to go on. > > 2. There usually isn't much you can do as far as ensuring the VPS is > capable of keeping the timing right. VoIP is reliant heavily on a good > timing source. Faxing is even more dependent. > > I'm not really sure what the MOS score is useful for, maybe someone else > can chime in on that. > > Something you need to make sure you are doing (specifically with > flowroute) is making sure you are doing a t.38 reinvite. Flowroute does not > auto detect t.38. Also freeswitch might have to transcode from audio > ulaw/alaw to t.38 if your FAX machine doesn't actually support t.38. > > I don't really know much about the new SPA 112/122's but usually the ATA > has a t.38 passthru mode that is auto detected. That might be what you are > referring to. > > -BDF > On May 10, 2012 2:35 PM, "Chris Ferreira" wrote: > >> Hi Bryan, >> >> >> Thanks for your response, much appreciated. >> >> >> 1. I have never had to do a debug other than analyze fs_cli output and >> iv'e never done a sip trace. I have started reading up on how to do these >> on the Wiki and will generate some files to take a look at. >> >> >> 2. I have heard this before about VPS's and even VMWare servers running >> on local physical hardware. I hope to be able to overcome these issues. >> >> >> 3. Other calls work fine throughout the switch and between carriers. >> >> >> >> I don't know if this is a good actual measure of anything, but >> VoIPSpear.com says that the servers MOS Score is a 4.0 . >> >> >> Other than the server running on a VPS, is my setup with the ATA >> registering to FreeSWITCH as an intermediary between Flowroute.com a >> correct practice? Or is there something else I could be doing? >> >> >> I know that if I was not needing to use an Analog Fax machine for >> sending/receiving that I would have other options. But unfortunately the >> Dinosaur must have it's Dial Tone. >> >> >> >> Thanks, >> >> -Chris >> >> >> >> >> On Thu, May 10, 2012 at 2:03 PM, Brian Foster wrote: >> >>> 1. Debug tells all. We need the siptrace as well. >>> >>> 2. VPS's are usually bad at sending/receiving faxes even if the media is >>> just being routed through it. >>> >>> 3. You could also look at session timers. Are other calls working? >>> >>> The only real way to know is if we have number 1 in our possession, >>> otherwise we just start guessing. >>> >>> -BDF >>> >>> On Thu, May 10, 2012 at 1:30 PM, Chris Ferreira wrote: >>> >>>> Hello All, >>>> >>>> >>>> >>>> I have a FreeSWITCH install running on CentOS 5.6 on a Linode VPS. I am >>>> trying to get my SPA112 (Version 1.1 Firmware) to send faxes (and >>>> eventually receive) successfully. I have the ATA registered as an extension >>>> and it's primary outgoing route is through Flowroute. I have T.38 enabled >>>> on the ATA and I have diabled ECM on the analog fax machine. All of the >>>> CDR's in Flowroute show that the test faxes are all ending their calls at >>>> 39 seconds. >>>> >>>> >>>> I can post screen shots of the ATA config or provide any other config >>>> info. >>>> >>>> >>>> Is there something I am missing, or should this setup work? >>>> >>>> I have poked around for a while for info, but unlike the PAP2T there is >>>> little info on these SPA112's. This is my first post asking for help as I >>>> usually try to resolve things on my own. But for this, I will defer >>>> to everyone's experience. >>>> >>>> >>>> >>>> Thanks, >>>> >>>> -Chris >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Brian D. Foster >>> Endigo Computer LLC >>> Email: bdfoster at endigotech.com >>> Phone: 317-800-7876 >>> Indianapolis, Indiana, USA >>> >>> This message contains confidential information and is intended for those >>> listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If >>> you are not the intended recipient you are notified that disclosing, >>> copying, distributing or taking any action in reliance on the contents of >>> this information is strictly prohibited. E-mail transmission cannot be >>> guaranteed to be secure or error-free as information could be intercepted, >>> corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. >>> The sender therefore does not accept liability for any errors or omissions >>> in the contents of this message, which arise as a result of e-mail >>> transmission. If verification is required please request a hard-copy >>> version. >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120515/c3abaf97/attachment.html From wesleyakio at tuntscorp.com Wed May 16 03:02:14 2012 From: wesleyakio at tuntscorp.com (Wesley Akio) Date: Tue, 15 May 2012 20:02:14 -0300 Subject: [Freeswitch-users] 2012 VoIP With Virtualization In-Reply-To: References: <550DADC3-9636-42FF-B391-BB00E84AC403@opencsta.org> <4FB24FB7.10204@anew.com.ve> Message-ID: Sorry for the delay... For the setup in question: The hardware: IBM SystemX 3650. 8 core Xeons, E5*. 48GB RAM Virtualization is done by VMWare ESXi, mostly 4 or 5. Network is all Cisco. 3 VM Boxes that reach an average of 400 calls during working hours. FreesSWITCH there is very old(1.0.6). 4 CPU Cores 2GB 64 bit Centos 5.2 local dialplan XML no transcoding xml_cdr very stable. Servers have several virtual machines with a lot of different things running, FSs have priority on hardware resources... We have some other VM boxes(FS Sep 2011) with less load(270 calls avg) that do trancoding(25 calls avg g729<->g711), no quality issues but the FS clock drifts a lot(up to 10 minutes a day), VMware's fault -> http://www.vmware.com/files/pdf/Timekeeping-In-VirtualMachines.pdf fsctl sync_clock solves the problem but might ruin CDRs so BEWARE! There are several other VMs in quite a few hardware setups, but none of them has impressive call volumes... P.S.: Please don't shout at the age of the installs... they have served us well! :D. [ ]s, Wesley Akio TuntsCorp.com On Tue, May 15, 2012 at 10:08 AM, Chris Mylonas wrote: > OK 400 calls is on the very high end of what I would have put on a VM. If > you wouldn't mind explaining your VM set up that would be great. > > > On 15/05/2012, at 10:54 PM, Wesley Akio wrote: > > As said before, it depends heavily in your setup... We do virtualize > FreeSWITCH and aside from a few glitches it has been good... We have approx > 400 calls per VM. Later when I hit the office I can describe my setup in > detail. > > Sent from mobile, sorry for the typos.... > Em 15/05/2012 09:42, "Saugort Dario Garcia Tovar" > escreveu: > >> Hi, Chris >> >> That depend of your enviroment. First you have to take a sort of >> scientific approach: define, trial and test again and again until the >> result be consistent. >> Second tell us about your enviroment: >> - Real hardware: the server that will run your vm. ?Are blades, small >> server, etc? VM software used (VMware, OpenVZ, XEN, Jails...)? >> - ?How you set your VM? cpu, cpu cores, memory, disk, #nic? >> - ?How many calls have you handle in the real world? >> - ?Transcoding? >> - ?Have you tested your VM? >> >> Virtualized enviroment for VoIP are growing. I am consultant for contact >> center services, and I have receive a lot of questions about >> virtualization. A solution provider can give some guidence about sizing and >> technical considerations but at the end the final word is: YOU HAVE >> MAKE A LOAD TEST TO TEST YOUR ENVIROMENT AND IDENTIFY POSSIBLE ISSUES >> BEFORE GO TO PRODUCTION!!!! >> >> Virtualization solutions will do their homework to make VoIP a secure >> option. Look this paper for wmware: >> >> http://blogs.vmware.com/performance/2012/01/voip-performance-on-vsphere-5.html >> http://www.vmware.com/files/pdf/techpaper/voip-perf-vsphere5.pdf >> >> On 5/15/2012 4:42 AM, Chris Mylonas wrote: >> >> Hi FS Users, >> >> What is the consensus about using virtualized servers for real-time voice (RTP)? Even up until a few years ago it was hard to guarantee the CPU cycles to the voice nodes. >> Virtualizing the signalling (SIP) has always been favourable in terms of HA. >> >> If the entire host is under control, is it safe to say - we can virtualise voice? >> Does it depend on what else is shared in the host - say it would be silly to put a high load database server on the same hardware node... >> >> Thanks for your inputs >> Chris >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com >> >> >> >> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com >> >> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >> >> >> ----- >> No virus found in this message. >> Checked by AVG - www.avg.com >> Version: 2012.0.2171 / Virus Database: 2425/4999 - Release Date: 05/14/12 >> >> >> >> >> >> -- >> Atentamente, >> *Dario Garc?a* >> Consultor. >> >> CCCT, Nivel C2, Sector Yarey, Mz, >> Ofc. MZ03a. >> Caracas-Venezuela. >> Tel?fono: +58 212 9081842 >> Cel: +58 412 2221515 >> dgarcia at anew.com.ve >> http://www.anew.com.ve >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120515/a8058377/attachment-0001.html From fvillarroel at yahoo.com Wed May 16 04:31:24 2012 From: fvillarroel at yahoo.com (FERNANDO VILLARROEL) Date: Tue, 15 May 2012 17:31:24 -0700 (PDT) Subject: [Freeswitch-users] Nibblebill users prepaid or postpaid Message-ID: <1337128284.54664.YahooMailClassic@web160305.mail.bf1.yahoo.com> Dear All. I am trying to use mod_nibblebill http://wiki.freeswitch.org/wiki/Mod_nibblebill I have installed it succesfully, but i do not understand how i config my users like prepaid or postpaid. I appreciate some tips. Regadrs From jmesquita at freeswitch.org Wed May 16 04:35:25 2012 From: jmesquita at freeswitch.org (=?utf-8?Q?Jo=C3=A3o_Mesquita?=) Date: Tue, 15 May 2012 21:35:25 -0300 Subject: [Freeswitch-users] Nibblebill users prepaid or postpaid In-Reply-To: <1337128284.54664.YahooMailClassic@web160305.mail.bf1.yahoo.com> References: <1337128284.54664.YahooMailClassic@web160305.mail.bf1.yahoo.com> Message-ID: Fernando, Argentinian greetings from old timers! :-) Have you started with the wiki page? If you know a bit of FreeSWITCH, it should be pretty straightforward. Regards, -- Jo?o Mesquita Sent with Sparrow (http://www.sparrowmailapp.com/?sig) On Tuesday, May 15, 2012 at 9:31 PM, FERNANDO VILLARROEL wrote: > Dear All. > > I am trying to use mod_nibblebill > > http://wiki.freeswitch.org/wiki/Mod_nibblebill > > I have installed it succesfully, but i do not understand how i config my users like prepaid or postpaid. > > I appreciate some tips. > > Regadrs > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120515/65a7ad97/attachment.html From mario_fs at mgtech.com Wed May 16 04:36:58 2012 From: mario_fs at mgtech.com (Mario G) Date: Tue, 15 May 2012 17:36:58 -0700 Subject: [Freeswitch-users] How to set "keep-alives re-use the TCP connection" ? Message-ID: Using Bria on iPad with a TCP connection, all works except after several hours it no longer registers to FreeSwitch. The Bria support says if this happens set the server PBX to use "keep-alives re-use the TCP connection". I could not find this options in the wiki, only thing close dealt with NAT but FS and the phones are all on local lan. Anyone know how to set this in the user definition? Thanks, Mario G From fvillarroel at yahoo.com Wed May 16 06:05:15 2012 From: fvillarroel at yahoo.com (FERNANDO VILLARROEL) Date: Tue, 15 May 2012 19:05:15 -0700 (PDT) Subject: [Freeswitch-users] Nibblebill users prepaid or postpaid In-Reply-To: Message-ID: <1337133915.9240.YahooMailClassic@web160302.mail.bf1.yahoo.com> Dear?Jo?o. Yes i see the wiki page, but i do not understand.? For example how i tell to FS that some customer is postpaid so FS authorize the call and if some customer is prepaid and balance cash is 0 so the calls should be declined. How i can do ? Fernando --- On Tue, 5/15/12, Jo?o Mesquita wrote: From: Jo?o Mesquita Subject: Re: [Freeswitch-users] Nibblebill users prepaid or postpaid To: "FreeSWITCH Users Help" Date: Tuesday, May 15, 2012, 9:35 PM Fernando, Argentinian greetings from old timers! :-) Have you started with the wiki page? If you know a bit of FreeSWITCH, it should be pretty straightforward. Regards, --?Jo?o MesquitaSent with Sparrow On Tuesday, May 15, 2012 at 9:31 PM, FERNANDO VILLARROEL wrote: Dear All. I am trying to use mod_nibblebill http://wiki.freeswitch.org/wiki/Mod_nibblebill I have installed it succesfully, but i do not understand how i config my users like prepaid or postpaid. I appreciate some tips. Regadrs _________________________________________________________________________Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org -----Inline Attachment Follows----- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120515/7bcacfbb/attachment.html From bdfoster at endigotech.com Wed May 16 06:10:22 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Tue, 15 May 2012 22:10:22 -0400 Subject: [Freeswitch-users] Fax Issues with Cisco SPA112 and T.38 In-Reply-To: References: Message-ID: I think your problem is probably the fact that your FAX machine doesn't support t.38. In that case you'd have to have freeswitch transcode from pcmu to t.38. It doesn't give good results when your server isn't on the LAN that your ATA is on. The point really of transmitting to flowroute as t.38 is to get better results when transmitting a FAX via VoIP. You would essentially be taking a detour to get a transcode and it really doesn't help because like I said your server is somewhere out in cyberspace. Also mix in that it is indeed a VPS, and that's pro ably what you are seeing the result of. Maybe the Atari actually transcodes to t.38 locally, I'm not sure and I don't have enough experience with that device to be able to tell you. If the ATA is indeed locally trandcoding to t.38 then my suggestion would be to bypass media on the server and sent it straight to flowroute. I hope this helps. If you have any questions let us know! -BDF On May 15, 2012 6:52 PM, "Chris Ferreira" wrote: > Hello Brian and Everyone, > > > Here is a link to the siptrace output. Sorry for the delay, it was my > birthday and I was away for the weekend. > > > http://pastebin.freeswitch.org/19057 > > > Thanks, > > -Chris > > > > > On Thu, May 10, 2012 at 10:16 PM, Brian Foster wrote: > >> 1. Run fs_cli and do 'sofia global siptrace on' without quotes. Stay in >> the cli and run a call in an instance where it would fail. Make sure your >> scroll buffer on your SSH terminal (PuTTY or what ever you have) to >> unlimited or something bog like 20K lines. Copy the logs out and put them >> on http://pastebin.freeswitch.org. If prompted for a username/password, >> please read the dialog box. SpanDSP doesn't have a great ability to be >> degugged on the click but it will give is something to go on. >> >> 2. There usually isn't much you can do as far as ensuring the VPS is >> capable of keeping the timing right. VoIP is reliant heavily on a good >> timing source. Faxing is even more dependent. >> >> I'm not really sure what the MOS score is useful for, maybe someone else >> can chime in on that. >> >> Something you need to make sure you are doing (specifically with >> flowroute) is making sure you are doing a t.38 reinvite. Flowroute does not >> auto detect t.38. Also freeswitch might have to transcode from audio >> ulaw/alaw to t.38 if your FAX machine doesn't actually support t.38. >> >> I don't really know much about the new SPA 112/122's but usually the ATA >> has a t.38 passthru mode that is auto detected. That might be what you are >> referring to. >> >> -BDF >> On May 10, 2012 2:35 PM, "Chris Ferreira" wrote: >> >>> Hi Bryan, >>> >>> >>> Thanks for your response, much appreciated. >>> >>> >>> 1. I have never had to do a debug other than analyze fs_cli output and >>> iv'e never done a sip trace. I have started reading up on how to do these >>> on the Wiki and will generate some files to take a look at. >>> >>> >>> 2. I have heard this before about VPS's and even VMWare servers running >>> on local physical hardware. I hope to be able to overcome these issues. >>> >>> >>> 3. Other calls work fine throughout the switch and between carriers. >>> >>> >>> >>> I don't know if this is a good actual measure of anything, but >>> VoIPSpear.com says that the servers MOS Score is a 4.0 . >>> >>> >>> Other than the server running on a VPS, is my setup with the ATA >>> registering to FreeSWITCH as an intermediary between Flowroute.com a >>> correct practice? Or is there something else I could be doing? >>> >>> >>> I know that if I was not needing to use an Analog Fax machine for >>> sending/receiving that I would have other options. But unfortunately the >>> Dinosaur must have it's Dial Tone. >>> >>> >>> >>> Thanks, >>> >>> -Chris >>> >>> >>> >>> >>> On Thu, May 10, 2012 at 2:03 PM, Brian Foster wrote: >>> >>>> 1. Debug tells all. We need the siptrace as well. >>>> >>>> 2. VPS's are usually bad at sending/receiving faxes even if the media >>>> is just being routed through it. >>>> >>>> 3. You could also look at session timers. Are other calls working? >>>> >>>> The only real way to know is if we have number 1 in our possession, >>>> otherwise we just start guessing. >>>> >>>> -BDF >>>> >>>> On Thu, May 10, 2012 at 1:30 PM, Chris Ferreira wrote: >>>> >>>>> Hello All, >>>>> >>>>> >>>>> >>>>> I have a FreeSWITCH install running on CentOS 5.6 on a Linode VPS. I >>>>> am trying to get my SPA112 (Version 1.1 Firmware) to send faxes (and >>>>> eventually receive) successfully. I have the ATA registered as an extension >>>>> and it's primary outgoing route is through Flowroute. I have T.38 enabled >>>>> on the ATA and I have diabled ECM on the analog fax machine. All of the >>>>> CDR's in Flowroute show that the test faxes are all ending their calls at >>>>> 39 seconds. >>>>> >>>>> >>>>> I can post screen shots of the ATA config or provide any other config >>>>> info. >>>>> >>>>> >>>>> Is there something I am missing, or should this setup work? >>>>> >>>>> I have poked around for a while for info, but unlike the PAP2T there >>>>> is little info on these SPA112's. This is my first post asking for help as >>>>> I usually try to resolve things on my own. But for this, I will defer >>>>> to everyone's experience. >>>>> >>>>> >>>>> >>>>> Thanks, >>>>> >>>>> -Chris >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> -- >>>> Brian D. Foster >>>> Endigo Computer LLC >>>> Email: bdfoster at endigotech.com >>>> Phone: 317-800-7876 >>>> Indianapolis, Indiana, USA >>>> >>>> This message contains confidential information and is intended for >>>> those listed in the "To:", "CC:", and/or "BCC:" fields of the message >>>> header. If you are not the intended recipient you are notified that >>>> disclosing, copying, distributing or taking any action in reliance on the >>>> contents of this information is strictly prohibited. E-mail transmission >>>> cannot be guaranteed to be secure or error-free as information could be >>>> intercepted, corrupted, lost, destroyed, arrive late or incomplete, or >>>> contain viruses. The sender therefore does not accept liability for any >>>> errors or omissions in the contents of this message, which arise as a >>>> result of e-mail transmission. If verification is required please request a >>>> hard-copy version. >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at 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 at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120515/349d0ce7/attachment-0001.html From bdfoster at endigotech.com Wed May 16 06:12:21 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Tue, 15 May 2012 22:12:21 -0400 Subject: [Freeswitch-users] Fax Issues with Cisco SPA112 and T.38 In-Reply-To: References: Message-ID: s/Atari/ATA/g s/pro ably/probably/ Probably other mistakes in there. Gotta love android spell check. -BDF On May 15, 2012 10:10 PM, "Brian Foster" wrote: > I think your problem is probably the fact that your FAX machine doesn't > support t.38. In that case you'd have to have freeswitch transcode from > pcmu to t.38. It doesn't give good results when your server isn't on the > LAN that your ATA is on. The point really of transmitting to flowroute as > t.38 is to get better results when transmitting a FAX via VoIP. You would > essentially be taking a detour to get a transcode and it really doesn't > help because like I said your server is somewhere out in cyberspace. Also > mix in that it is indeed a VPS, and that's pro ably what you are seeing the > result of. Maybe the Atari actually transcodes to t.38 locally, I'm not > sure and I don't have enough experience with that device to be able to tell > you. If the ATA is indeed locally trandcoding to t.38 then my suggestion > would be to bypass media on the server and sent it straight to flowroute. > > I hope this helps. If you have any questions let us know! > > -BDF > On May 15, 2012 6:52 PM, "Chris Ferreira" wrote: > >> Hello Brian and Everyone, >> >> >> Here is a link to the siptrace output. Sorry for the delay, it was my >> birthday and I was away for the weekend. >> >> >> http://pastebin.freeswitch.org/19057 >> >> >> Thanks, >> >> -Chris >> >> >> >> >> On Thu, May 10, 2012 at 10:16 PM, Brian Foster wrote: >> >>> 1. Run fs_cli and do 'sofia global siptrace on' without quotes. Stay in >>> the cli and run a call in an instance where it would fail. Make sure your >>> scroll buffer on your SSH terminal (PuTTY or what ever you have) to >>> unlimited or something bog like 20K lines. Copy the logs out and put them >>> on http://pastebin.freeswitch.org. If prompted for a username/password, >>> please read the dialog box. SpanDSP doesn't have a great ability to be >>> degugged on the click but it will give is something to go on. >>> >>> 2. There usually isn't much you can do as far as ensuring the VPS is >>> capable of keeping the timing right. VoIP is reliant heavily on a good >>> timing source. Faxing is even more dependent. >>> >>> I'm not really sure what the MOS score is useful for, maybe someone else >>> can chime in on that. >>> >>> Something you need to make sure you are doing (specifically with >>> flowroute) is making sure you are doing a t.38 reinvite. Flowroute does not >>> auto detect t.38. Also freeswitch might have to transcode from audio >>> ulaw/alaw to t.38 if your FAX machine doesn't actually support t.38. >>> >>> I don't really know much about the new SPA 112/122's but usually the ATA >>> has a t.38 passthru mode that is auto detected. That might be what you are >>> referring to. >>> >>> -BDF >>> On May 10, 2012 2:35 PM, "Chris Ferreira" wrote: >>> >>>> Hi Bryan, >>>> >>>> >>>> Thanks for your response, much appreciated. >>>> >>>> >>>> 1. I have never had to do a debug other than analyze fs_cli output and >>>> iv'e never done a sip trace. I have started reading up on how to do these >>>> on the Wiki and will generate some files to take a look at. >>>> >>>> >>>> 2. I have heard this before about VPS's and even VMWare servers running >>>> on local physical hardware. I hope to be able to overcome these issues. >>>> >>>> >>>> 3. Other calls work fine throughout the switch and between carriers. >>>> >>>> >>>> >>>> I don't know if this is a good actual measure of anything, but >>>> VoIPSpear.com says that the servers MOS Score is a 4.0 . >>>> >>>> >>>> Other than the server running on a VPS, is my setup with the ATA >>>> registering to FreeSWITCH as an intermediary between Flowroute.com a >>>> correct practice? Or is there something else I could be doing? >>>> >>>> >>>> I know that if I was not needing to use an Analog Fax machine for >>>> sending/receiving that I would have other options. But unfortunately the >>>> Dinosaur must have it's Dial Tone. >>>> >>>> >>>> >>>> Thanks, >>>> >>>> -Chris >>>> >>>> >>>> >>>> >>>> On Thu, May 10, 2012 at 2:03 PM, Brian Foster wrote: >>>> >>>>> 1. Debug tells all. We need the siptrace as well. >>>>> >>>>> 2. VPS's are usually bad at sending/receiving faxes even if the media >>>>> is just being routed through it. >>>>> >>>>> 3. You could also look at session timers. Are other calls working? >>>>> >>>>> The only real way to know is if we have number 1 in our possession, >>>>> otherwise we just start guessing. >>>>> >>>>> -BDF >>>>> >>>>> On Thu, May 10, 2012 at 1:30 PM, Chris Ferreira wrote: >>>>> >>>>>> Hello All, >>>>>> >>>>>> >>>>>> >>>>>> I have a FreeSWITCH install running on CentOS 5.6 on a Linode VPS. I >>>>>> am trying to get my SPA112 (Version 1.1 Firmware) to send faxes (and >>>>>> eventually receive) successfully. I have the ATA registered as an extension >>>>>> and it's primary outgoing route is through Flowroute. I have T.38 enabled >>>>>> on the ATA and I have diabled ECM on the analog fax machine. All of the >>>>>> CDR's in Flowroute show that the test faxes are all ending their calls at >>>>>> 39 seconds. >>>>>> >>>>>> >>>>>> I can post screen shots of the ATA config or provide any other config >>>>>> info. >>>>>> >>>>>> >>>>>> Is there something I am missing, or should this setup work? >>>>>> >>>>>> I have poked around for a while for info, but unlike the PAP2T there >>>>>> is little info on these SPA112's. This is my first post asking for help as >>>>>> I usually try to resolve things on my own. But for this, I will defer >>>>>> to everyone's experience. >>>>>> >>>>>> >>>>>> >>>>>> Thanks, >>>>>> >>>>>> -Chris >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>> >>>>> >>>>> -- >>>>> Brian D. Foster >>>>> Endigo Computer LLC >>>>> Email: bdfoster at endigotech.com >>>>> Phone: 317-800-7876 >>>>> Indianapolis, Indiana, USA >>>>> >>>>> This message contains confidential information and is intended for >>>>> those listed in the "To:", "CC:", and/or "BCC:" fields of the message >>>>> header. If you are not the intended recipient you are notified that >>>>> disclosing, copying, distributing or taking any action in reliance on the >>>>> contents of this information is strictly prohibited. E-mail transmission >>>>> cannot be guaranteed to be secure or error-free as information could be >>>>> intercepted, corrupted, lost, destroyed, arrive late or incomplete, or >>>>> contain viruses. The sender therefore does not accept liability for any >>>>> errors or omissions in the contents of this message, which arise as a >>>>> result of e-mail transmission. If verification is required please request a >>>>> hard-copy version. >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at 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 at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at 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 at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120515/b14fe9fa/attachment.html From bdfoster at endigotech.com Wed May 16 06:14:05 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Tue, 15 May 2012 22:14:05 -0400 Subject: [Freeswitch-users] FreeSWITCH Queue Scenario - Caller Puts MoH on hold In-Reply-To: <2AA25D7C-E49F-4597-8C3F-1EAE48646F75@opencsta.org> References: <2AA25D7C-E49F-4597-8C3F-1EAE48646F75@opencsta.org> Message-ID: Session timers would be where I would start. -BDF On May 15, 2012 3:06 PM, "Chris Mylonas" wrote: > Hi FS Users, > > What happens when the caller that is waiting in the queue puts the MoH on > hold.....and then the Internet drops out. Does FS detect that the caller > is gone? > > The reason I ask is I was doing some testing with __asterisk__ and a > couple of soft phones, a snom, and had SIPp scenarios for agents on a > remote machine. My snom had 6 calls in the queue (but they were on hold) > and my router gave up. On reconnecting, my phone had also rebooted. I > dialled into the queue to my surprise I was the 10th caller. I put another > half a dozen calls in the queue and then rebooted it to see if the queue > would recognise the drop out. It didn't. > > Is it known that FreeSWITCH works better than this behaviour? > > I will test it myself - I'm just still in early cookbook pages with it > though.... > > Thanks for the interest, > Chris > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120515/3216c5c4/attachment-0001.html From bdfoster at endigotech.com Wed May 16 06:22:41 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Tue, 15 May 2012 22:22:41 -0400 Subject: [Freeswitch-users] Nibblebill users prepaid or postpaid In-Reply-To: <1337133915.9240.YahooMailClassic@web160302.mail.bf1.yahoo.com> References: <1337133915.9240.YahooMailClassic@web160302.mail.bf1.yahoo.com> Message-ID: As it says on the wiki you can set it up to run negative balance. -BDF On May 15, 2012 10:06 PM, "FERNANDO VILLARROEL" wrote: > Dear Jo?o. > > Yes i see the wiki page, but i do not understand. > > For example how i tell to FS that some customer is postpaid so FS > authorize the call and if some customer is prepaid and balance cash is 0 so > the calls should be declined. How i can do ? > > Fernando > > --- On *Tue, 5/15/12, Jo?o Mesquita * wrote: > > > From: Jo?o Mesquita > Subject: Re: [Freeswitch-users] Nibblebill users prepaid or postpaid > To: "FreeSWITCH Users Help" > Date: Tuesday, May 15, 2012, 9:35 PM > > Fernando, > > Argentinian greetings from old timers! :-) > > Have you started with the wiki page? If you know a bit of FreeSWITCH, it > should be pretty straightforward. > > Regards, > > -- > Jo?o Mesquita > Sent with Sparrow > > On Tuesday, May 15, 2012 at 9:31 PM, FERNANDO VILLARROEL wrote: > > Dear All. > > I am trying to use mod_nibblebill > > http://wiki.freeswitch.org/wiki/Mod_nibblebill > > I have installed it succesfully, but i do not understand how i config my > users like prepaid or postpaid. > > I appreciate some tips. > > Regadrs > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -----Inline Attachment Follows----- > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120515/85dc5c00/attachment.html From mgg at giagnocavo.net Wed May 16 06:31:45 2012 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Tue, 15 May 2012 22:31:45 -0400 Subject: [Freeswitch-users] Freeswitch + Lync In-Reply-To: References: Message-ID: <83FF8D7C9F526E44B77C97DD2891652A53D9795A@mse17be1.mse17.exchange.ms> When using Lync to terminate outbound to FreeSWITCH, remember by default Lync will CANCEL the call if it does not get 183 progress within 10 seconds. Reason is some stuff about ICE/NAT traversal and missing the first second of audio. (Also realise Lync will try to use ICE, so if you're on a VPN with a similar subnet, it'll pick that path over others. Only way to fix is to firewall off RTP from the Lync server, so that ICE fails. This is the only time I've seen ICE work, and then it resulted in horrible audio quality (RTP over SSL VPN)). -Michael From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Shawn Yu Sent: Tuesday, May 08, 2012 11:34 AM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Freeswitch + Lync We have Lync in our environment, and have successfully configured the following to work with Freeswitch: - Outbound call from Lync via Freeswitch - Inbound call rings both Lync and Freeswitch endpoint (Polycom phone) - Calls between Lync and Polycom phone We would also like to have Lync to remote call control the Polycom phone. Is this possible? Shawn -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120515/bfc078a6/attachment.html From chris at opencsta.org Wed May 16 06:46:59 2012 From: chris at opencsta.org (Chris Mylonas) Date: Wed, 16 May 2012 12:46:59 +1000 Subject: [Freeswitch-users] FreeSWITCH Queue Scenario - Caller Puts MoH on hold In-Reply-To: References: <2AA25D7C-E49F-4597-8C3F-1EAE48646F75@opencsta.org> Message-ID: <76C79753-4704-4846-99EE-041F3E3CC761@opencsta.org> Cool thanks Brian. It's been a couple of years since I was last seen digging into protocol stuff :) On 16/05/2012, at 12:14 PM, Brian Foster wrote: > Session timers would be where I would start. > > -BDF > > On May 15, 2012 3:06 PM, "Chris Mylonas" wrote: > Hi FS Users, > > What happens when the caller that is waiting in the queue puts the MoH on hold.....and then the Internet drops out. Does FS detect that the caller is gone? > > The reason I ask is I was doing some testing with __asterisk__ and a couple of soft phones, a snom, and had SIPp scenarios for agents on a remote machine. My snom had 6 calls in the queue (but they were on hold) and my router gave up. On reconnecting, my phone had also rebooted. I dialled into the queue to my surprise I was the 10th caller. I put another half a dozen calls in the queue and then rebooted it to see if the queue would recognise the drop out. It didn't. > > Is it known that FreeSWITCH works better than this behaviour? > > I will test it myself - I'm just still in early cookbook pages with it though.... > > Thanks for the interest, > Chris > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120516/9adb366d/attachment-0001.html From chris at gonumina.com Wed May 16 07:51:52 2012 From: chris at gonumina.com (Chris Ferreira) Date: Tue, 15 May 2012 23:51:52 -0400 Subject: [Freeswitch-users] Fax Issues with Cisco SPA112 and T.38 In-Reply-To: References: Message-ID: <-7345408630261820439@unknownmsgid> I will look into this, and I am testing it with an older fax right now because I brought the ATA home from the office to work on. I think the test fax at the office supports newer standards. Thanks for correcting the spelling errors, I had started digging through some boxes for my Atari 2600 to fit into the equation. Lol. Much appreciated Brian. Thanks, -Chris ___________________ Mobile Reply On May 15, 2012, at 10:14 PM, Brian Foster wrote: s/Atari/ATA/g s/pro ably/probably/ Probably other mistakes in there. Gotta love android spell check. -BDF On May 15, 2012 10:10 PM, "Brian Foster" wrote: > I think your problem is probably the fact that your FAX machine doesn't > support t.38. In that case you'd have to have freeswitch transcode from > pcmu to t.38. It doesn't give good results when your server isn't on the > LAN that your ATA is on. The point really of transmitting to flowroute as > t.38 is to get better results when transmitting a FAX via VoIP. You would > essentially be taking a detour to get a transcode and it really doesn't > help because like I said your server is somewhere out in cyberspace. Also > mix in that it is indeed a VPS, and that's pro ably what you are seeing the > result of. Maybe the Atari actually transcodes to t.38 locally, I'm not > sure and I don't have enough experience with that device to be able to tell > you. If the ATA is indeed locally trandcoding to t.38 then my suggestion > would be to bypass media on the server and sent it straight to flowroute. > > I hope this helps. If you have any questions let us know! > > -BDF > On May 15, 2012 6:52 PM, "Chris Ferreira" wrote: > >> Hello Brian and Everyone, >> >> >> Here is a link to the siptrace output. Sorry for the delay, it was my >> birthday and I was away for the weekend. >> >> >> http://pastebin.freeswitch.org/19057 >> >> >> Thanks, >> >> -Chris >> >> >> >> >> On Thu, May 10, 2012 at 10:16 PM, Brian Foster wrote: >> >>> 1. Run fs_cli and do 'sofia global siptrace on' without quotes. Stay in >>> the cli and run a call in an instance where it would fail. Make sure your >>> scroll buffer on your SSH terminal (PuTTY or what ever you have) to >>> unlimited or something bog like 20K lines. Copy the logs out and put them >>> on http://pastebin.freeswitch.org. If prompted for a username/password, >>> please read the dialog box. SpanDSP doesn't have a great ability to be >>> degugged on the click but it will give is something to go on. >>> >>> 2. There usually isn't much you can do as far as ensuring the VPS is >>> capable of keeping the timing right. VoIP is reliant heavily on a good >>> timing source. Faxing is even more dependent. >>> >>> I'm not really sure what the MOS score is useful for, maybe someone else >>> can chime in on that. >>> >>> Something you need to make sure you are doing (specifically with >>> flowroute) is making sure you are doing a t.38 reinvite. Flowroute does not >>> auto detect t.38. Also freeswitch might have to transcode from audio >>> ulaw/alaw to t.38 if your FAX machine doesn't actually support t.38. >>> >>> I don't really know much about the new SPA 112/122's but usually the ATA >>> has a t.38 passthru mode that is auto detected. That might be what you are >>> referring to. >>> >>> -BDF >>> On May 10, 2012 2:35 PM, "Chris Ferreira" wrote: >>> >>>> Hi Bryan, >>>> >>>> >>>> Thanks for your response, much appreciated. >>>> >>>> >>>> 1. I have never had to do a debug other than analyze fs_cli output and >>>> iv'e never done a sip trace. I have started reading up on how to do these >>>> on the Wiki and will generate some files to take a look at. >>>> >>>> >>>> 2. I have heard this before about VPS's and even VMWare servers running >>>> on local physical hardware. I hope to be able to overcome these issues. >>>> >>>> >>>> 3. Other calls work fine throughout the switch and between carriers. >>>> >>>> >>>> >>>> I don't know if this is a good actual measure of anything, but >>>> VoIPSpear.com says that the servers MOS Score is a 4.0 . >>>> >>>> >>>> Other than the server running on a VPS, is my setup with the ATA >>>> registering to FreeSWITCH as an intermediary between Flowroute.com a >>>> correct practice? Or is there something else I could be doing? >>>> >>>> >>>> I know that if I was not needing to use an Analog Fax machine for >>>> sending/receiving that I would have other options. But unfortunately the >>>> Dinosaur must have it's Dial Tone. >>>> >>>> >>>> >>>> Thanks, >>>> >>>> -Chris >>>> >>>> >>>> >>>> >>>> On Thu, May 10, 2012 at 2:03 PM, Brian Foster wrote: >>>> >>>>> 1. Debug tells all. We need the siptrace as well. >>>>> >>>>> 2. VPS's are usually bad at sending/receiving faxes even if the media >>>>> is just being routed through it. >>>>> >>>>> 3. You could also look at session timers. Are other calls working? >>>>> >>>>> The only real way to know is if we have number 1 in our possession, >>>>> otherwise we just start guessing. >>>>> >>>>> -BDF >>>>> >>>>> On Thu, May 10, 2012 at 1:30 PM, Chris Ferreira wrote: >>>>> >>>>>> Hello All, >>>>>> >>>>>> >>>>>> >>>>>> I have a FreeSWITCH install running on CentOS 5.6 on a Linode VPS. I >>>>>> am trying to get my SPA112 (Version 1.1 Firmware) to send faxes (and >>>>>> eventually receive) successfully. I have the ATA registered as an extension >>>>>> and it's primary outgoing route is through Flowroute. I have T.38 enabled >>>>>> on the ATA and I have diabled ECM on the analog fax machine. All of the >>>>>> CDR's in Flowroute show that the test faxes are all ending their calls at >>>>>> 39 seconds. >>>>>> >>>>>> >>>>>> I can post screen shots of the ATA config or provide any other config >>>>>> info. >>>>>> >>>>>> >>>>>> Is there something I am missing, or should this setup work? >>>>>> >>>>>> I have poked around for a while for info, but unlike the PAP2T there >>>>>> is little info on these SPA112's. This is my first post asking for help as >>>>>> I usually try to resolve things on my own. But for this, I will defer >>>>>> to everyone's experience. >>>>>> >>>>>> >>>>>> >>>>>> Thanks, >>>>>> >>>>>> -Chris >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>> >>>>> >>>>> -- >>>>> Brian D. Foster >>>>> Endigo Computer LLC >>>>> Email: bdfoster at endigotech.com >>>>> Phone: 317-800-7876 >>>>> Indianapolis, Indiana, USA >>>>> >>>>> This message contains confidential information and is intended for >>>>> those listed in the "To:", "CC:", and/or "BCC:" fields of the message >>>>> header. If you are not the intended recipient you are notified that >>>>> disclosing, copying, distributing or taking any action in reliance on the >>>>> contents of this information is strictly prohibited. E-mail transmission >>>>> cannot be guaranteed to be secure or error-free as information could be >>>>> intercepted, corrupted, lost, destroyed, arrive late or incomplete, or >>>>> contain viruses. The sender therefore does not accept liability for any >>>>> errors or omissions in the contents of this message, which arise as a >>>>> result of e-mail transmission. If verification is required please request a >>>>> hard-copy version. >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at 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 at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at 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 at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120515/3e37cdb2/attachment-0001.html From thaddeus at thogan.com Wed May 16 08:33:38 2012 From: thaddeus at thogan.com (Thaddeus Hogan) Date: Tue, 15 May 2012 23:33:38 -0500 Subject: [Freeswitch-users] 2012 VoIP With Virtualization In-Reply-To: <550DADC3-9636-42FF-B391-BB00E84AC403@opencsta.org> References: <550DADC3-9636-42FF-B391-BB00E84AC403@opencsta.org> Message-ID: I have Freeswitch running on a KVM based VM right now, and it doesn't do too hot. Voicemail, MoH, and IVR prompts are choppy often but call media proxy (no transcode) works okay, though long calls start to deteriorate after about 30 minutes. I haven't tried on Xen since I bailed to KVM to get to a later kernel in 2010 when blktap2 progress was unapparent. However Xen has always been much better at I/O in my opinion. Rackspace runs Xen for their cloud servers service, and people on this list seem to report good results with Freeswitch on Rackspace cloud. Most recently I have been testing on LXC guests on Ubuntu 12.04. This has worked out as well as LXC doesn't do any hardware virtualization. Instead it is a method of isolating execution of processes in different namespaces and all guests share the host kernel. I have been migrating all of my KVM guests to LXC, and my Freeswitch install is on the list. With LXC you can set memory and CPU usage restrictions, so in my experience provides all the same benefits of full virt minus the ability to run Windows. OpenVZ is also based on process isolation rather than hardware virtualization, and would probably have the same benefits over full virt as LXC with Freeswitch. -- Thaddeus On 5/15/2012 4:12 AM, Chris Mylonas wrote: > Hi FS Users, > > What is the consensus about using virtualized servers for real-time voice (RTP)? Even up until a few years ago it was hard to guarantee the CPU cycles to the voice nodes. > Virtualizing the signalling (SIP) has always been favourable in terms of HA. > > If the entire host is under control, is it safe to say - we can virtualise voice? > Does it depend on what else is shared in the host - say it would be silly to put a high load database server on the same hardware node... > > Thanks for your inputs > Chris > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From gabe at gundy.org Wed May 16 09:53:30 2012 From: gabe at gundy.org (Gabriel Gunderson) Date: Tue, 15 May 2012 23:53:30 -0600 Subject: [Freeswitch-users] 2012 VoIP With Virtualization In-Reply-To: References: <550DADC3-9636-42FF-B391-BB00E84AC403@opencsta.org> Message-ID: On Tue, May 15, 2012 at 10:33 PM, Thaddeus Hogan wrote: > I have Freeswitch running on a KVM based VM right now, and it doesn't do > too hot. Voicemail, MoH, and IVR prompts are choppy often but call media > proxy (no transcode) works okay, though long calls start to deteriorate > after about 30 minutes. I've used KVM with good results... what kind of load are we talking about? Gabe From noel at vwci.com Wed May 16 04:46:23 2012 From: noel at vwci.com (Noel Morgan) Date: Tue, 15 May 2012 19:46:23 -0500 Subject: [Freeswitch-users] ASR weirdness In-Reply-To: Message-ID: FreeSWITCH Version 1.2.0 (git-44fd0de 2012-05-14 02-04-36 +0200) on CentOS 6.2 Here is the scenario: Caller calls into dialplan, lua script gets called, their number gets parsed, does a db look up, then the caller is transferred to a recording then eventually hold music in dialplan. During the original callers session, script 2 gets called (see script 1). Script 2 executes lua script 3 on the origination. Speech is always detected on the session:answer(). I have even tried separate/new/independent sessions of the original session? No go. The idea is to bridge the calls in a uuid bridge eventually and it works fine when it gets there, but I cannot trap when the call is answered under any scenario (no early media signaling and disposition is always 'ANSWER'), hence the ASR (tried all the execute_on_answer variants plus nolocal: yada yada, even tried it with python and javascript as well). I am cross-eyed by now so it is probably missing something stupid. I need the async behavior to keep the user on hold for as little time as possible. Any ideas are appreciated. TIA, Noel 2012-05-15 17:01:09.748559 [DEBUG] switch_rtp.c:3253 Correct ip/port confirmed. INFO: cmd_ln.c(691): Parsing command line: \ -samprate 8000 \ -hmm /usr/local/freeswitch/grammar/model/communicator \ -jsgf /usr/local/freeswitch/grammar/xxxxxx.gram \ -lw 6.5 \ -dict /usr/local/freeswitch/grammar/default.dic \ -frate 50 \ -silprob 0.005 Current configuration: [NAME] [DEFLT] [VALUE] -agc none none -agcthresh 2.0 2.000000e+00 -alpha 0.97 9.700000e-01 -ascale 20.0 2.000000e+01 -aw 1 1 -backtrace no no -beam 1e-48 1.000000e-48 -bestpath yes yes -bestpathlw 9.5 9.500000e+00 -bghist no no -ceplen 13 13 -cmn current current -cmninit 8.0 8.0 -compallsen no no -debug 0 -dict /usr/local/freeswitch/grammar/default.dic -dictcase no no -dither no no -doublebw no no -ds 1 1 -fdict -feat 1s_c_d_dd 1s_c_d_dd -featparams -fillprob 1e-8 1.000000e-08 -frate 100 50 -fsg -fsgusealtpron yes yes -fsgusefiller yes yes -fwdflat yes yes -fwdflatbeam 1e-64 1.000000e-64 -fwdflatefwid 4 4 -fwdflatlw 8.5 8.500000e+00 -fwdflatsfwin 25 25 -fwdflatwbeam 7e-29 7.000000e-29 -fwdtree yes yes -hmm /usr/local/freeswitch/grammar/model/communicator -input_endian little little -jsgf /usr/local/freeswitch/grammar/xxxxxx.gram -kdmaxbbi -1 -1 -kdmaxdepth 0 0 -kdtree -latsize 5000 5000 -lda -ldadim 0 0 -lextreedump 0 0 -lifter 0 0 -lm -lmctl -lmname default default -logbase 1.0001 1.000100e+00 -logfn -logspec no no -lowerf 133.33334 1.333333e+02 -lpbeam 1e-40 1.000000e-40 -lponlybeam 7e-29 7.000000e-29 -lw 6.5 6.500000e+00 -maxhmmpf -1 -1 -maxnewoov 20 20 -maxwpf -1 -1 -mdef -mean -mfclogdir -min_endfr 0 0 -mixw -mixwfloor 0.0000001 1.000000e-07 -mllr -mmap yes yes -ncep 13 13 -nfft 512 512 -nfilt 40 40 -nwpen 1.0 1.000000e+00 -pbeam 1e-48 1.000000e-48 -pip 1.0 1.000000e+00 -pl_beam 1e-10 1.000000e-10 -pl_pbeam 1e-5 1.000000e-05 -pl_window 0 0 -rawlogdir -remove_dc no no -round_filters yes yes -samprate 16000 8.000000e+03 -seed -1 -1 -sendump -senlogdir -senmgau -silprob 0.005 5.000000e-03 -smoothspec no no -svspec -tmat -tmatfloor 0.0001 1.000000e-04 -topn 4 4 -topn_beam 0 0 -toprule -transform legacy legacy -unit_area yes yes -upperf 6855.4976 6.855498e+03 -usewdphones no no -uw 1.0 1.000000e+00 -var -varfloor 0.0001 1.000000e-04 -varnorm no no -verbose no no -warp_params -warp_type inverse_linear inverse_linear -wbeam 7e-29 7.000000e-29 -wip 0.65 6.500000e-01 -wlen 0.025625 2.562500e-02 EXECUTE sofia/xxxxxx/2143909113 detect_speech(pocketsphinx xxxxxx default) INFO: cmd_ln.c(691): Parsing command line: \ -alpha 0.97 \ -dither yes \ -doublebw no \ -nfilt 31 \ -ncep 13 \ -lowerf 200 \ -upperf 3500 \ -nfft 256 \ -wlen 0.0256 \ -transform legacy \ -feat s2_4x \ -agc none \ -cmn current \ -varnorm no Current configuration: [NAME] [DEFLT] [VALUE] -agc none none -agcthresh 2.0 2.000000e+00 -alpha 0.97 9.700000e-01 -ceplen 13 13 -cmn current current -cmninit 8.0 8.0 -dither no yes -doublebw no no -feat 1s_c_d_dd s2_4x -frate 100 50 -input_endian little little -lda -ldadim 0 0 -lifter 0 0 -logspec no no -lowerf 133.33334 2.000000e+02 -ncep 13 13 -nfft 512 256 -nfilt 40 31 -remove_dc no no -round_filters yes yes -samprate 16000 8.000000e+03 -seed -1 -1 -smoothspec no no -svspec -transform legacy legacy -unit_area yes yes -upperf 6855.4976 3.500000e+03 -varnorm no no -verbose no no -warp_params -warp_type inverse_linear inverse_linear -wlen 0.025625 2.560000e-02 INFO: acmod.c(242): Parsed model-specific feature parameters from /usr/local/freeswitch/grammar/model/communicator/feat.params INFO: fe_interface.c(289): You are using the internal mechanism to generate the seed. INFO: feat.c(684): Initializing feature stream to type: 's2_4x', ceplen=13, CMN='current', VARNORM='no', AGC='none' INFO: cmn.c(142): mean[0]= 12.00, mean[1..12]= 0.0 INFO: mdef.c(520): Reading model definition: /usr/local/freeswitch/grammar/model/communicator/mdef INFO: bin_mdef.c(173): Allocating 104160 * 8 bytes (813 KiB) for CD tree INFO: tmat.c(205): Reading HMM transition probability matrices: /usr/local/freeswitch/grammar/model/communicator/transition_matrices INFO: acmod.c(117): Attempting to use SCHMM computation module INFO: ms_gauden.c(198): Reading mixture gaussian parameter: /usr/local/freeswitch/grammar/model/communicator/means INFO: ms_gauden.c(292): 1 codebook, 4 feature, size: INFO: ms_gauden.c(294): 256x12 INFO: ms_gauden.c(294): 256x24 INFO: ms_gauden.c(294): 256x3 INFO: ms_gauden.c(294): 256x12 INFO: ms_gauden.c(198): Reading mixture gaussian parameter: /usr/local/freeswitch/grammar/model/communicator/variances INFO: ms_gauden.c(292): 1 codebook, 4 feature, size: INFO: ms_gauden.c(294): 256x12 INFO: ms_gauden.c(294): 256x24 INFO: ms_gauden.c(294): 256x3 INFO: ms_gauden.c(294): 256x12 INFO: ms_gauden.c(354): 59 variance values floored INFO: s2_semi_mgau.c(908): Loading senones from dump file /usr/local/freeswitch/grammar/model/communicator/sendump INFO: s2_semi_mgau.c(932): BEGIN FILE FORMAT DESCRIPTION INFO: s2_semi_mgau.c(995): Rows: 256, Columns: 6256 INFO: s2_semi_mgau.c(1027): Using memory-mapped I/O for senones INFO: s2_semi_mgau.c(1304): Maximum top-N: 4 Top-N beams: 0 0 0 0 INFO: dict.c(306): Allocating 137548 * 32 bytes (4298 KiB) for word entries INFO: dict.c(321): Reading main dictionary: /usr/local/freeswitch/grammar/default.dic INFO: dict.c(212): Allocated 1010 KiB for strings, 1664 KiB for phones INFO: dict.c(324): 133436 words read INFO: dict.c(330): Reading filler dictionary: /usr/local/freeswitch/grammar/model/communicator/noisedict INFO: dict.c(212): Allocated 0 KiB for strings, 0 KiB for phones INFO: dict.c(333): 17 words read INFO: dict2pid.c(396): Building PID tables for dictionary INFO: dict2pid.c(404): Allocating 51^3 * 2 bytes (259 KiB) for word-initial triphones INFO: dict2pid.c(131): Allocated 62832 bytes (61 KiB) for word-final triphones INFO: dict2pid.c(195): Allocated 62832 bytes (61 KiB) for single-phone word triphones INFO: fsg_search.c(145): FSG(beam: -1080, pbeam: -1080, wbeam: -634; wip: -26, pip: 0) INFO: jsgf.c(546): Defined rule: INFO: jsgf.c(546): Defined rule: INFO: jsgf.c(546): Defined rule: INFO: jsgf.c(546): Defined rule: INFO: jsgf.c(546): Defined rule: INFO: jsgf.c(546): Defined rule: INFO: jsgf.c(546): Defined rule: INFO: jsgf.c(546): Defined rule: PUBLIC INFO: fsg_model.c(213): Computing transitive closure for null transitions INFO: fsg_model.c(264): 87 null transitions added INFO: fsg_model.c(411): Adding silence transitions for to FSG INFO: fsg_model.c(431): Added 23 silence word transitions INFO: fsg_model.c(411): Adding silence transitions for ++AE++ to FSG INFO: fsg_model.c(431): Added 23 silence word transitions INFO: fsg_model.c(411): Adding silence transitions for ++AH++ to FSG INFO: fsg_model.c(431): Added 23 silence word transitions INFO: fsg_model.c(411): Adding silence transitions for ++BACKGROUND++ to FSG INFO: fsg_model.c(431): Added 23 silence word transitions INFO: fsg_model.c(411): Adding silence transitions for ++BREATH++ to FSG INFO: fsg_model.c(431): Added 23 silence word transitions INFO: fsg_model.c(411): Adding silence transitions for ++COUGH++ to FSG INFO: fsg_model.c(431): Added 23 silence word transitions INFO: fsg_model.c(411): Adding silence transitions for ++EH++ to FSG INFO: fsg_model.c(431): Added 23 silence word transitions INFO: fsg_model.c(411): Adding silence transitions for ++ER++ to FSG INFO: fsg_model.c(431): Added 23 silence word transitions INFO: fsg_model.c(411): Adding silence transitions for ++LAUGH++ to FSG INFO: fsg_model.c(431): Added 23 silence word transitions INFO: fsg_model.c(411): Adding silence transitions for ++MM++ to FSG INFO: fsg_model.c(431): Added 23 silence word transitions INFO: fsg_model.c(411): Adding silence transitions for ++MUMBLE++ to FSG INFO: fsg_model.c(431): Added 23 silence word transitions INFO: fsg_model.c(411): Adding silence transitions for ++NOISE++ to FSG INFO: fsg_model.c(431): Added 23 silence word transitions INFO: fsg_model.c(411): Adding silence transitions for ++OH++ to FSG INFO: fsg_model.c(431): Added 23 silence word transitions INFO: fsg_model.c(411): Adding silence transitions for ++SMACK++ to FSG INFO: fsg_model.c(431): Added 23 silence word transitions INFO: fsg_model.c(411): Adding silence transitions for ++UH++ to FSG INFO: fsg_model.c(431): Added 23 silence word transitions INFO: fsg_model.c(411): Adding silence transitions for ++UH_NOISE++ to FSG INFO: fsg_model.c(431): Added 23 silence word transitions INFO: fsg_model.c(411): Adding silence transitions for ++UM++ to FSG INFO: fsg_model.c(431): Added 23 silence word transitions INFO: fsg_model.c(411): Adding silence transitions for ++UM_NOISE++ to FSG INFO: fsg_model.c(431): Added 23 silence word transitions INFO: fsg_search.c(364): Added 1 alternate word transitions INFO: fsg_lextree.c(108): Allocated 2392 bytes (2 KiB) for left and right context phones INFO: fsg_lextree.c(251): 505 HMM nodes in lextree (496 leaves) INFO: fsg_lextree.c(253): Allocated 64640 bytes (63 KiB) for all lextree nodes INFO: fsg_lextree.c(256): Allocated 63488 bytes (62 KiB) for lextree leafnodes 2012-05-15 17:01:11.428551 [DEBUG] switch_core_media_bug.c:502 Attaching BUG to sofia/xxxxxx/2143909113 2012-05-15 17:01:11.448552 [DEBUG] switch_core_io.c:353 Setting BUG Codec PCMU:0 2012-05-15 17:01:13.448551 [NOTICE] switch_cpp.cpp:1227 ----------------->call answered is false...<------------- EXECUTE sofia/xxxxxx/2143909113 detect_speech(resume) 2012-05-15 17:01:15.008548 [INFO] switch_cpp.cpp:1227 Callback with type event 2012-05-15 17:01:15.008548 [INFO] switch_cpp.cpp:1227 'Event-Name: DETECTED_SPEECH Core-UUID: 0a387574-9ee8-11e1-8fa1-43045b098f7a FreeSWITCH-Hostname: xxxxxx FreeSWITCH-Switchname: xxxxxx FreeSWITCH-IPv4: 199.15.96.29 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2012-05-15%2017%3A01%3A14 Event-Date-GMT: Wed,%2016%20May%202012%2000%3A01%3A14%20GMT Event-Date-Timestamp: 1337126474988540 Event-Calling-File: switch_ivr_async.c Event-Calling-Function: speech_thread Event-Calling-Line-Number: 3549 Event-Sequence: 1204 Speech-Type: begin-speaking ' 2012-05-15 17:01:15.008548 [NOTICE] switch_cpp.cpp:1227 ----------------->call answered is false...<------------- EXECUTE sofia/xxxxxx/2143909113 detect_speech(resume) 2012-05-15 17:01:17.008548 [NOTICE] switch_cpp.cpp:1227 ----------------->call answered is false...<------------- EXECUTE sofia/xxxxxx/2143909113 detect_speech(resume) INFO: cmn_prior.c(121): cmn_prior_update: from < 8.00 0.00 0.00 0.00 0.00 0.00 0.00 0.00 0.00 0.00 0.00 0.00 0.00 > INFO: cmn_prior.c(139): cmn_prior_update: to < 8.11 -0.00 0.40 -0.00 -0.39 -0.64 -0.79 -0.65 -0.51 -0.32 -0.08 0.12 0.11 > INFO: fsg_search.c(1030): 133 frames, 1846 HMMs (13/fr), 5417 senones (40/fr), 1099 history entries (8/fr) INFO: fsg_search.c(1407): Start node hello.0:14:96 INFO: fsg_search.c(1446): End node .95:105:132 (-2846) INFO: fsg_search.c(1662): lattice start node hello.0 end node .95 INFO: ps_lattice.c(1352): Normalizer P(O) = alpha(:95:132) = -886169 INFO: ps_lattice.c(1390): Joint P(O,S) = -886169 P(S|O) = 0 2012-05-15 17:01:17.648543 [DEBUG] mod_pocketsphinx.c:383 Recognized: hello, Confidence: 100 2012-05-15 17:01:17.648543 [INFO] switch_cpp.cpp:1227 Callback with type event 2012-05-15 17:01:17.648543 [INFO] switch_cpp.cpp:1227 'Event-Name: DETECTED_SPEECH Core-UUID: 0a387574-9ee8-11e1-8fa1-43045b098f7a FreeSWITCH-Hostname: xxxxxx FreeSWITCH-Switchname: xxxxxx FreeSWITCH-IPv4: 199.15.96.29 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2012-05-15%2017%3A01%3A17 Event-Date-GMT: Wed,%2016%20May%202012%2000%3A01%3A17%20GMT Event-Date-Timestamp: 1337126477648543 Event-Calling-File: switch_ivr_async.c Event-Calling-Function: speech_thread Event-Calling-Line-Number: 3549 Event-Sequence: 1209 Speech-Type: detected-speech Content-Length: 172 hello ' EXECUTE sofia/xxxxxx/2143909113 detect_speech(pause) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120515/f8afe2cf/attachment-0001.html From azza.miled at gmail.com Wed May 16 12:41:52 2012 From: azza.miled at gmail.com (azza miled) Date: Wed, 16 May 2012 09:41:52 +0100 Subject: [Freeswitch-users] music on hold for opensips users Message-ID: I am using FS as a media server for my OpenSIPS server. I want to know if I can make opensips' users listen to music via FreeSWITCH, until the opensips callee picks up. Yhe point that I don't get it, is how FS knows when the opensips' callee change from busy state to available. Thanks a lot -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120516/fe1012d9/attachment.html From joohny at mail.ru Wed May 16 13:10:59 2012 From: joohny at mail.ru (=?UTF-8?B?0JXQstCz0LXQvdC40Lk=?=) Date: Wed, 16 May 2012 13:10:59 +0400 Subject: [Freeswitch-users] =?utf-8?q?mod=5Fcallcenter_real_answer?= Message-ID: Hello, Gustavo! Thank you very much. I thought about fixing src, but was little afraid, your message made me to do it :) I Think this place(2446 line): /* Make sure we answer the channel before getting the switch_channel_time_table_t answer time */ ? ? ?//switch_channel_answer(member_channel); switch_channel_pre_answer(member_channel); ? ? ?// switch_channel_mark_ring_ready(member_channel); ? ? ?//switch_channel_ring_ready(member_channel); Now it is good with answer(in console logging?at least). I need also ring_ready signals. I tried to switch_channel_mark_ring_ready(member_channel); switch_channel_ring_ready(member_channel); but NO luck. Silence on members side. Best Regards, Evginey. http://blog.buchnev.ru --------------------------------------- Hi Evginey I made that change a few days ago, due a scenario where dialer is used. In fact, dialer dials to a queue first instead of customer. All I did was replace in the code "answer" by "pre_answer" in some place. I've no the code right now, but I can send it to you later. Regards Gustavo Hi, colleagues. Could I have real answer for client on mod_callcenter, when agent answers? Or there is another way to queue call between 2 FreeSwitch.? For example, client's call has to be routed to two FS servers, which has different agents in queues. But answer comes from first server when application CALLCENTER is called. How can I do this? Best Regards, Evginey. http://blog.buchnev.ru -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120516/80708dfe/attachment.html From modesto at isimples.com.br Wed May 16 16:08:53 2012 From: modesto at isimples.com.br (Antonio Modesto) Date: Wed, 16 May 2012 09:08:53 -0300 Subject: [Freeswitch-users] Portuguese Sounds Message-ID: <1337170133.2975.12.camel@modesto.localdomain.net> Hi, Does anybody knows if there is a portuguese version of the default freeswitch sounds? I only have found it in English and Russian. From miha at softnet.si Wed May 16 16:15:34 2012 From: miha at softnet.si (Miha) Date: Wed, 16 May 2012 14:15:34 +0200 Subject: [Freeswitch-users] bridge a call to registered user Message-ID: <4FB39A66.5070809@softnet.si> Hi, just one question. I have created two profiles (suggested by @Michael) as I was having same NAT issue. Profiles are basically the same only registration port is different (5060, 5070) and NAT profile have set: . In my dialplan I have made bridge like this: freeswitch at default> show re [ registrations] freeswitch at default> show registrations reg_user,realm,token,url,expires,network_ip,network_port,network_proto,hostname 018108753.fs_test,fs_test.fs2.blabla.com,40804176-fb1f176 at 172.31.1.190,sofia/internal/sip:018108753.fs_test at xxx.xxx.xxx.xxx:1265,1337169317,xxx.xxx.xxx.xxx,1265,udp,localhost.localdomain 018108752.fs_test,fs_test.fs2.blabla.comi,873cfe40-1224f25 at 172.31.1.171,sofia/nat/sip:018108752.fs_test at xxx.xxx.xxx.xxx:2097,1337170569,xxx.xxx.xxx.xxx,1920,udp,localhost.localdomain 2 total. Why only phone work which is registered on NAT profile. IF I call users which is registered on internal profile I get user not registered? Thanks! Miha From ccesario at tecnomega.com.br Wed May 16 16:37:54 2012 From: ccesario at tecnomega.com.br (Carlos Cesario) Date: Wed, 16 May 2012 09:37:54 -0300 Subject: [Freeswitch-users] Portuguese Sounds In-Reply-To: <1337170133.2975.12.camel@modesto.localdomain.net> References: <1337170133.2975.12.camel@modesto.localdomain.net> Message-ID: <4FB39FA2.7040801@tecnomega.com.br> Hi Antonio, Yes, you can download it in http://wirelessmundi.com/freeswitch-sounds-pt-BR-karina-48000-1.0.15.tar.gz att, Em 16-05-2012 09:08, Antonio Modesto escreveu: > Hi, > > Does anybody knows if there is a portuguese version of the default > freeswitch sounds? I only have found it in English and Russian. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120516/afe7cf5e/attachment.html From ahe.sanath at gmail.com Wed May 16 16:59:18 2012 From: ahe.sanath at gmail.com (Sanath Prasanna) Date: Wed, 16 May 2012 18:29:18 +0530 Subject: [Freeswitch-users] Run LUA script in different server Message-ID: Hi all, I have 2 servers. One server has SIP GW connection From Operator & IVR applications need to build in other server. How to call distributed LUA applications with Mysql Databases from the SIP GW server ? Pls advice. Main idea is, maintaining SIP connection in one server & all the IVR applications in other server. Br, Sanath -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120516/b1310c90/attachment.html From wesleyakio at tuntscorp.com Wed May 16 18:00:24 2012 From: wesleyakio at tuntscorp.com (Wesley Akio) Date: Wed, 16 May 2012 11:00:24 -0300 Subject: [Freeswitch-users] FS Core Dump using GSM Hardware Message-ID: Hi all, I'm experiencing several core dumps a day in one of our boxes. The problem was happening with FS trunk from Apr 30. Updated yesterday and the problem continues. I suspect it has something to do with the GSM hardware(Khomp) in use but I'm not sure... If someone could take a look at the backtraces I would be very thankfull. http://pastebin.freeswitch.org/19059 FreeSWITCH Version 1.2.0 (git-d015395 2012-05-15 11-26-16 -0500) Best, Wesley Akio TuntsCorp.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120516/3812be2d/attachment-0001.html From jmesquita at freeswitch.org Wed May 16 18:05:01 2012 From: jmesquita at freeswitch.org (=?utf-8?Q?Jo=C3=A3o_Mesquita?=) Date: Wed, 16 May 2012 11:05:01 -0300 Subject: [Freeswitch-users] FS Core Dump using GSM Hardware In-Reply-To: References: Message-ID: <8BA004618E4E4FE0ACE37755AC1C9610@freeswitch.org> Wesley, Why do you believe it to be related to the GSM hardware? What version of the mod_khomp are you using? Regards, -- Jo?o Mesquita Sent with Sparrow (http://www.sparrowmailapp.com/?sig) On Wednesday, May 16, 2012 at 11:00 AM, Wesley Akio wrote: > Hi all, > > I'm experiencing several core dumps a day in one of our boxes. > > The problem was happening with FS trunk from Apr 30. Updated yesterday and the problem continues. > > I suspect it has something to do with the GSM hardware(Khomp) in use but I'm not sure... > > If someone could take a look at the backtraces I would be very thankfull. > > http://pastebin.freeswitch.org/19059 > FreeSWITCH Version 1.2.0 (git-d015395 2012-05-15 11-26-16 -0500) > > Best, > > Wesley Akio > TuntsCorp.com (http://TuntsCorp.com) > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120516/19709c6c/attachment.html From wesleyakio at tuntscorp.com Wed May 16 18:43:01 2012 From: wesleyakio at tuntscorp.com (Wesley Akio) Date: Wed, 16 May 2012 11:43:01 -0300 Subject: [Freeswitch-users] FS Core Dump using GSM Hardware In-Reply-To: <8BA004618E4E4FE0ACE37755AC1C9610@freeswitch.org> References: <8BA004618E4E4FE0ACE37755AC1C9610@freeswitch.org> Message-ID: Hi Jo?o, Latest from Khomp(mod_khomp_2.0_004_x86-64). Guessing wildly here... For two reasons only: -I dont get more than a few SMSs/hour and there are several smsThreads. -Some cores show mod_khomp in the backtrace, like the BT below. #0 0x0000003820430285 in raise () from /lib64/libc.so.6 #1 0x0000003820431d30 in abort () from /lib64/libc.so.6 #2 0x0000003820429706 in __assert_fail () from /lib64/libc.so.6 #3 0x00002aafdde16122 in switch_channel_perform_set_state (channel=0xd20f4c0, file=0x2aaaac55ea7e "src/khomp_pvt.cpp", func=0x2aaaac561096 "justStart", line=854, state=CS_INIT) at src/switch_channel.c:2145 #4 0x00002aaaac509989 in Board::KhompPvt::justStart (this=0xd0c5810, profile=0x0) at src/khomp_pvt.cpp:854 #5 0x00002aaaac50b2ea in Board::KhompPvt::onNewCall (this=0xd0c5810, e=) at src/khomp_pvt.cpp:1663 #6 0x00002aaaac4fb22e in Board::KhompPvt::eventHandler (this=0xd0c5810, e=0xd0bae50) at src/khomp_pvt.cpp:2038 #7 0x00002aaaac4c9dfc in BoardGSM::KhompPvtGSM::eventHandler (this=0xd0c5810, e=0xd0bae50) at ./include/khomp_pvt_gsm.h:191 #8 0x00002aaaac5148c6 in Board::eventHandler (this=0x2aaab0218c40, obj=0, e=0xd0bae50) at ./include/khomp_pvt.h:1264 #9 0x00002aaaac4f9047 in Board::eventThread (void_evt=) at src/khomp_pvt.cpp:2403 #10 0x00002aaaac46575d in Thread::run (thread=, obj=0x2072) at ./commons/base/system/freeswitch/thread.hpp:314 #11 0x0000003820c0677d in start_thread () from /lib64/libpthread.so.0 #12 0x00000038204d325d in clone () from /lib64/libc.so.6 Sorry but the core dump from where I got this BT is no loger among us... [ ]s, Wesley Akio TuntsCorp.com On Wed, May 16, 2012 at 11:05 AM, Jo?o Mesquita wrote: > Wesley, > > Why do you believe it to be related to the GSM hardware? What version of > the mod_khomp are you using? > > Regards, > > -- > Jo?o Mesquita > Sent with Sparrow > > On Wednesday, May 16, 2012 at 11:00 AM, Wesley Akio wrote: > > Hi all, > > I'm experiencing several core dumps a day in one of our boxes. > > The problem was happening with FS trunk from Apr 30. Updated yesterday and > the problem continues. > > I suspect it has something to do with the GSM hardware(Khomp) in use but > I'm not sure... > > If someone could take a look at the backtraces I would be very thankfull. > > http://pastebin.freeswitch.org/19059 > FreeSWITCH Version 1.2.0 (git-d015395 2012-05-15 11-26-16 -0500) > > Best, > > Wesley Akio > TuntsCorp.com > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120516/36052c19/attachment.html From lewisjoshua2 at gmail.com Wed May 16 17:49:04 2012 From: lewisjoshua2 at gmail.com (lewisjoshua2) Date: Wed, 16 May 2012 06:49:04 -0700 (PDT) Subject: [Freeswitch-users] Polycom SCA/SLA Trouble Message-ID: <33858252.post@talk.nabble.com> Hello all, I have two Polycom Soundpoint IP 501 phones running firmware 3.1.x and registered to FreeSwitch. I have the phones set up with two lines each, one private and one shared. For instance: Phone1: ext 1006 (private) and ext 1009 (shared) Phone2: ext 1007 (private) and ext 1009 (shared) When I dial extension 1009, both phones ring as expected. When I pick up the call on one phone, the other phone shows the busy status of the line as expected. However, when I place the call on hold from either phone and attempt to "unhold" the call from the other phone, it attempts to dial the shared extension (1009) instead of bridging the call from the original calling phone (say ext 1005). This is the case when there are two callsPerLine setup in the -.cfg file for each phone. When there is only 1 callsPerLine, attemtping to "unhold" the call will simply drop the call on that particular phone. I have the following configuration in conf/sip_profiles/internal.xml ... "/> ... I have the following configuration in conf/dialplan/default.xml for ext 1009 The following is my FreeSwitch version: FreeSWITCH Version 1.1.beta1 (git-4283408 2012-04-29 11-33-24 -0400) Any help is greatly appreciated! -- View this message in context: http://old.nabble.com/Polycom-SCA-SLA-Trouble-tp33858252p33858252.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From lewisjoshua2 at gmail.com Wed May 16 18:28:40 2012 From: lewisjoshua2 at gmail.com (lewisjoshua2) Date: Wed, 16 May 2012 07:28:40 -0700 (PDT) Subject: [Freeswitch-users] Polycom SCA/SLA Trouble In-Reply-To: <33858252.post@talk.nabble.com> References: <33858252.post@talk.nabble.com> Message-ID: <33858493.post@talk.nabble.com> I fixed this problem by changing the domain variable in conf/vars.xml from $${local_ip_v4} to the static IP of my PBX. I am glad this fixed the problem; however, I would still like some insight on why FreeSwitch behaved in the manner I described. Thanks. lewisjoshua2 wrote: > > Hello all, > > I have two Polycom Soundpoint IP 501 phones running firmware 3.1.x and > registered to FreeSwitch. I have the phones set up with two lines each, > one private and one shared. For instance: > Phone1: ext 1006 (private) and ext 1009 (shared) > Phone2: ext 1007 (private) and ext 1009 (shared) > > When I dial extension 1009, both phones ring as expected. When I pick up > the call on one phone, the other phone shows the busy status of the line > as expected. However, when I place the call on hold from either phone and > attempt to "unhold" the call from the other phone, it attempts to dial the > shared extension (1009) instead of bridging the call from the original > calling phone (say ext 1005). This is the case when there are two > callsPerLine setup in the -.cfg file for each phone. When there > is only 1 callsPerLine, attemtping to "unhold" the call will simply drop > the call on that particular phone. > > I have the following configuration in conf/sip_profiles/internal.xml > ... > > > "/> > > ... > > I have the following configuration in conf/dialplan/default.xml for ext > 1009 > > > > > > > > The following is my FreeSwitch version: > FreeSWITCH Version 1.1.beta1 (git-4283408 2012-04-29 11-33-24 -0400) > > Any help is greatly appreciated! > > -- View this message in context: http://old.nabble.com/Polycom-SCA-SLA-Trouble-tp33858252p33858493.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From vivekm at packtpub.com Wed May 16 11:51:01 2012 From: vivekm at packtpub.com (Vivek Menon) Date: Wed, 16 May 2012 13:21:01 +0530 Subject: [Freeswitch-users] Review Request for new book on FreeSWITCH Message-ID: <4FB35C65.1030504@packtpub.com> Hi Everyone, My name is Vivek and I work for Packt, a UK based publishing company specializing in publishing books based on Information Technology (I.T) with a heavy emphasis on Open Source. We have recently published a new book on FreeSWITCH titled, " *FreeSWITCH Cookbook* *"*. Written by members of the FreeSWITCH development team, this book is for readers who want to Control FreeSWITCH remotely with the powerful event socket interface, route inbound and outbound calls, monitoring calls via the FreeSWITCH Web interface, configuring users and phones as well as connections to VoIP providers and even Google Voice and many more. You can read more about this book here: http://www.packtpub.com/freeswitch-telephony-advanced-cookbook/book Considering your expertise in the subject matter, your thoughts about this book would prove extremely valuable to us and will be very much appreciated. In case this subject interests you and you'd be interested in writing a review on your website or Amazon, do let me know. I'd be delighted to send you an e-copy of the book on your confirmation. In case you have any queries, do let me know and I'd be happy to assist. I look forward to hearing from you. Regards, Vivek -- *Vivek Menon* *Marketing Research Executive* | Packt Publishing | www.PacktPub.com *MSN:* vivekm at packtpub.com | Interested in becoming an author? Visit Packt's Author Website for all the information you need about writing for Packt. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120516/9ec9a6f8/attachment-0001.html From noel at vwci.com Wed May 16 17:16:50 2012 From: noel at vwci.com (Noel Morgan) Date: Wed, 16 May 2012 08:16:50 -0500 Subject: [Freeswitch-users] ASR Weirdness Message-ID: Sorry for the cross-thread; it was an accident. FreeSWITCH Version 1.2.0 (git-44fd0de 2012-05-14 02-04-36 +0200) on CentOS 6.2 Here is the scenario: Caller calls into dialplan, lua script gets called, their number gets parsed, does a db look up, then the caller is transferred to a recording then eventually hold music in dialplan. During the original callers session, script 2 gets called (see script 1). Script 2 executes lua script 3 on the origination. Speech is always detected on the session:answer(). I have even tried separate/new/independent sessions of the original session? No go. The idea is to bridge the calls in a uuid bridge eventually and it works fine when it gets there, but I cannot trap when the call is answered under any scenario (no early media signaling and disposition is always 'ANSWER'), hence the ASR (tried all the execute_on_answer variants plus nolocal: yada yada, even tried it with python and javascript as well). I am cross-eyed by now so it is probably missing something stupid. I need the async behavior to keep the user on hold for as little time as possible. Any ideas are appreciated. TIA, Noel 2012-05-15 17:01:09.748559 [DEBUG] switch_rtp.c:3253 Correct ip/port confirmed. INFO: cmd_ln.c(691): Parsing command line: \ -samprate 8000 \ -hmm /usr/local/freeswitch/grammar/model/communicator \ -jsgf /usr/local/freeswitch/grammar/xxxxxx.gram \ -lw 6.5 \ -dict /usr/local/freeswitch/grammar/default.dic \ -frate 50 \ -silprob 0.005 Current configuration: [NAME] [DEFLT] [VALUE] -agc none none -agcthresh 2.0 2.000000e+00 -alpha 0.97 9.700000e-01 -ascale 20.0 2.000000e+01 -aw 1 1 -backtrace no no -beam 1e-48 1.000000e-48 -bestpath yes yes -bestpathlw 9.5 9.500000e+00 -bghist no no -ceplen 13 13 -cmn current current -cmninit 8.0 8.0 -compallsen no no -debug 0 -dict /usr/local/freeswitch/grammar/default.dic -dictcase no no -dither no no -doublebw no no -ds 1 1 -fdict -feat 1s_c_d_dd 1s_c_d_dd -featparams -fillprob 1e-8 1.000000e-08 -frate 100 50 -fsg -fsgusealtpron yes yes -fsgusefiller yes yes -fwdflat yes yes -fwdflatbeam 1e-64 1.000000e-64 -fwdflatefwid 4 4 -fwdflatlw 8.5 8.500000e+00 -fwdflatsfwin 25 25 -fwdflatwbeam 7e-29 7.000000e-29 -fwdtree yes yes -hmm /usr/local/freeswitch/grammar/model/communicator -input_endian little little -jsgf /usr/local/freeswitch/grammar/xxxxxx.gram -kdmaxbbi -1 -1 -kdmaxdepth 0 0 -kdtree -latsize 5000 5000 -lda -ldadim 0 0 -lextreedump 0 0 -lifter 0 0 -lm -lmctl -lmname default default -logbase 1.0001 1.000100e+00 -logfn -logspec no no -lowerf 133.33334 1.333333e+02 -lpbeam 1e-40 1.000000e-40 -lponlybeam 7e-29 7.000000e-29 -lw 6.5 6.500000e+00 -maxhmmpf -1 -1 -maxnewoov 20 20 -maxwpf -1 -1 -mdef -mean -mfclogdir -min_endfr 0 0 -mixw -mixwfloor 0.0000001 1.000000e-07 -mllr -mmap yes yes -ncep 13 13 -nfft 512 512 -nfilt 40 40 -nwpen 1.0 1.000000e+00 -pbeam 1e-48 1.000000e-48 -pip 1.0 1.000000e+00 -pl_beam 1e-10 1.000000e-10 -pl_pbeam 1e-5 1.000000e-05 -pl_window 0 0 -rawlogdir -remove_dc no no -round_filters yes yes -samprate 16000 8.000000e+03 -seed -1 -1 -sendump -senlogdir -senmgau -silprob 0.005 5.000000e-03 -smoothspec no no -svspec -tmat -tmatfloor 0.0001 1.000000e-04 -topn 4 4 -topn_beam 0 0 -toprule -transform legacy legacy -unit_area yes yes -upperf 6855.4976 6.855498e+03 -usewdphones no no -uw 1.0 1.000000e+00 -var -varfloor 0.0001 1.000000e-04 -varnorm no no -verbose no no -warp_params -warp_type inverse_linear inverse_linear -wbeam 7e-29 7.000000e-29 -wip 0.65 6.500000e-01 -wlen 0.025625 2.562500e-02 EXECUTE sofia/xxxxxx/2143909113 detect_speech(pocketsphinx xxxxxx default) INFO: cmd_ln.c(691): Parsing command line: \ -alpha 0.97 \ -dither yes \ -doublebw no \ -nfilt 31 \ -ncep 13 \ -lowerf 200 \ -upperf 3500 \ -nfft 256 \ -wlen 0.0256 \ -transform legacy \ -feat s2_4x \ -agc none \ -cmn current \ -varnorm no Current configuration: [NAME] [DEFLT] [VALUE] -agc none none -agcthresh 2.0 2.000000e+00 -alpha 0.97 9.700000e-01 -ceplen 13 13 -cmn current current -cmninit 8.0 8.0 -dither no yes -doublebw no no -feat 1s_c_d_dd s2_4x -frate 100 50 -input_endian little little -lda -ldadim 0 0 -lifter 0 0 -logspec no no -lowerf 133.33334 2.000000e+02 -ncep 13 13 -nfft 512 256 -nfilt 40 31 -remove_dc no no -round_filters yes yes -samprate 16000 8.000000e+03 -seed -1 -1 -smoothspec no no -svspec -transform legacy legacy -unit_area yes yes -upperf 6855.4976 3.500000e+03 -varnorm no no -verbose no no -warp_params -warp_type inverse_linear inverse_linear -wlen 0.025625 2.560000e-02 INFO: acmod.c(242): Parsed model-specific feature parameters from /usr/local/freeswitch/grammar/model/communicator/feat.params INFO: fe_interface.c(289): You are using the internal mechanism to generate the seed. INFO: feat.c(684): Initializing feature stream to type: 's2_4x', ceplen=13, CMN='current', VARNORM='no', AGC='none' INFO: cmn.c(142): mean[0]= 12.00, mean[1..12]= 0.0 INFO: mdef.c(520): Reading model definition: /usr/local/freeswitch/grammar/model/communicator/mdef INFO: bin_mdef.c(173): Allocating 104160 * 8 bytes (813 KiB) for CD tree INFO: tmat.c(205): Reading HMM transition probability matrices: /usr/local/freeswitch/grammar/model/communicator/transition_matrices INFO: acmod.c(117): Attempting to use SCHMM computation module INFO: ms_gauden.c(198): Reading mixture gaussian parameter: /usr/local/freeswitch/grammar/model/communicator/means INFO: ms_gauden.c(292): 1 codebook, 4 feature, size: INFO: ms_gauden.c(294): 256x12 INFO: ms_gauden.c(294): 256x24 INFO: ms_gauden.c(294): 256x3 INFO: ms_gauden.c(294): 256x12 INFO: ms_gauden.c(198): Reading mixture gaussian parameter: /usr/local/freeswitch/grammar/model/communicator/variances INFO: ms_gauden.c(292): 1 codebook, 4 feature, size: INFO: ms_gauden.c(294): 256x12 INFO: ms_gauden.c(294): 256x24 INFO: ms_gauden.c(294): 256x3 INFO: ms_gauden.c(294): 256x12 INFO: ms_gauden.c(354): 59 variance values floored INFO: s2_semi_mgau.c(908): Loading senones from dump file /usr/local/freeswitch/grammar/model/communicator/sendump INFO: s2_semi_mgau.c(932): BEGIN FILE FORMAT DESCRIPTION INFO: s2_semi_mgau.c(995): Rows: 256, Columns: 6256 INFO: s2_semi_mgau.c(1027): Using memory-mapped I/O for senones INFO: s2_semi_mgau.c(1304): Maximum top-N: 4 Top-N beams: 0 0 0 0 INFO: dict.c(306): Allocating 137548 * 32 bytes (4298 KiB) for word entries INFO: dict.c(321): Reading main dictionary: /usr/local/freeswitch/grammar/default.dic INFO: dict.c(212): Allocated 1010 KiB for strings, 1664 KiB for phones INFO: dict.c(324): 133436 words read INFO: dict.c(330): Reading filler dictionary: /usr/local/freeswitch/grammar/model/communicator/noisedict INFO: dict.c(212): Allocated 0 KiB for strings, 0 KiB for phones INFO: dict.c(333): 17 words read INFO: dict2pid.c(396): Building PID tables for dictionary INFO: dict2pid.c(404): Allocating 51^3 * 2 bytes (259 KiB) for word-initial triphones INFO: dict2pid.c(131): Allocated 62832 bytes (61 KiB) for word-final triphones INFO: dict2pid.c(195): Allocated 62832 bytes (61 KiB) for single-phone word triphones INFO: fsg_search.c(145): FSG(beam: -1080, pbeam: -1080, wbeam: -634; wip: -26, pip: 0) INFO: jsgf.c(546): Defined rule: INFO: jsgf.c(546): Defined rule: INFO: jsgf.c(546): Defined rule: INFO: jsgf.c(546): Defined rule: INFO: jsgf.c(546): Defined rule: INFO: jsgf.c(546): Defined rule: INFO: jsgf.c(546): Defined rule: INFO: jsgf.c(546): Defined rule: PUBLIC INFO: fsg_model.c(213): Computing transitive closure for null transitions INFO: fsg_model.c(264): 87 null transitions added INFO: fsg_model.c(411): Adding silence transitions for to FSG INFO: fsg_model.c(431): Added 23 silence word transitions INFO: fsg_model.c(411): Adding silence transitions for ++AE++ to FSG INFO: fsg_model.c(431): Added 23 silence word transitions INFO: fsg_model.c(411): Adding silence transitions for ++AH++ to FSG INFO: fsg_model.c(431): Added 23 silence word transitions INFO: fsg_model.c(411): Adding silence transitions for ++BACKGROUND++ to FSG INFO: fsg_model.c(431): Added 23 silence word transitions INFO: fsg_model.c(411): Adding silence transitions for ++BREATH++ to FSG INFO: fsg_model.c(431): Added 23 silence word transitions INFO: fsg_model.c(411): Adding silence transitions for ++COUGH++ to FSG INFO: fsg_model.c(431): Added 23 silence word transitions INFO: fsg_model.c(411): Adding silence transitions for ++EH++ to FSG INFO: fsg_model.c(431): Added 23 silence word transitions INFO: fsg_model.c(411): Adding silence transitions for ++ER++ to FSG INFO: fsg_model.c(431): Added 23 silence word transitions INFO: fsg_model.c(411): Adding silence transitions for ++LAUGH++ to FSG INFO: fsg_model.c(431): Added 23 silence word transitions INFO: fsg_model.c(411): Adding silence transitions for ++MM++ to FSG INFO: fsg_model.c(431): Added 23 silence word transitions INFO: fsg_model.c(411): Adding silence transitions for ++MUMBLE++ to FSG INFO: fsg_model.c(431): Added 23 silence word transitions INFO: fsg_model.c(411): Adding silence transitions for ++NOISE++ to FSG INFO: fsg_model.c(431): Added 23 silence word transitions INFO: fsg_model.c(411): Adding silence transitions for ++OH++ to FSG INFO: fsg_model.c(431): Added 23 silence word transitions INFO: fsg_model.c(411): Adding silence transitions for ++SMACK++ to FSG INFO: fsg_model.c(431): Added 23 silence word transitions INFO: fsg_model.c(411): Adding silence transitions for ++UH++ to FSG INFO: fsg_model.c(431): Added 23 silence word transitions INFO: fsg_model.c(411): Adding silence transitions for ++UH_NOISE++ to FSG INFO: fsg_model.c(431): Added 23 silence word transitions INFO: fsg_model.c(411): Adding silence transitions for ++UM++ to FSG INFO: fsg_model.c(431): Added 23 silence word transitions INFO: fsg_model.c(411): Adding silence transitions for ++UM_NOISE++ to FSG INFO: fsg_model.c(431): Added 23 silence word transitions INFO: fsg_search.c(364): Added 1 alternate word transitions INFO: fsg_lextree.c(108): Allocated 2392 bytes (2 KiB) for left and right context phones INFO: fsg_lextree.c(251): 505 HMM nodes in lextree (496 leaves) INFO: fsg_lextree.c(253): Allocated 64640 bytes (63 KiB) for all lextree nodes INFO: fsg_lextree.c(256): Allocated 63488 bytes (62 KiB) for lextree leafnodes 2012-05-15 17:01:11.428551 [DEBUG] switch_core_media_bug.c:502 Attaching BUG to sofia/xxxxxx/2143909113 2012-05-15 17:01:11.448552 [DEBUG] switch_core_io.c:353 Setting BUG Codec PCMU:0 2012-05-15 17:01:13.448551 [NOTICE] switch_cpp.cpp:1227 ----------------->call answered is false...<------------- EXECUTE sofia/xxxxxx/2143909113 detect_speech(resume) 2012-05-15 17:01:15.008548 [INFO] switch_cpp.cpp:1227 Callback with type event 2012-05-15 17:01:15.008548 [INFO] switch_cpp.cpp:1227 'Event-Name: DETECTED_SPEECH Core-UUID: 0a387574-9ee8-11e1-8fa1-43045b098f7a FreeSWITCH-Hostname: xxxxxx FreeSWITCH-Switchname: xxxxxx FreeSWITCH-IPv4: 199.15.96.29 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2012-05-15%2017%3A01%3A14 Event-Date-GMT: Wed,%2016%20May%202012%2000%3A01%3A14%20GMT Event-Date-Timestamp: 1337126474988540 Event-Calling-File: switch_ivr_async.c Event-Calling-Function: speech_thread Event-Calling-Line-Number: 3549 Event-Sequence: 1204 Speech-Type: begin-speaking ' 2012-05-15 17:01:15.008548 [NOTICE] switch_cpp.cpp:1227 ----------------->call answered is false...<------------- EXECUTE sofia/xxxxxx/2143909113 detect_speech(resume) 2012-05-15 17:01:17.008548 [NOTICE] switch_cpp.cpp:1227 ----------------->call answered is false...<------------- EXECUTE sofia/xxxxxx/2143909113 detect_speech(resume) INFO: cmn_prior.c(121): cmn_prior_update: from < 8.00 0.00 0.00 0.00 0.00 0.00 0.00 0.00 0.00 0.00 0.00 0.00 0.00 > INFO: cmn_prior.c(139): cmn_prior_update: to < 8.11 -0.00 0.40 -0.00 -0.39 -0.64 -0.79 -0.65 -0.51 -0.32 -0.08 0.12 0.11 > INFO: fsg_search.c(1030): 133 frames, 1846 HMMs (13/fr), 5417 senones (40/fr), 1099 history entries (8/fr) INFO: fsg_search.c(1407): Start node hello.0:14:96 INFO: fsg_search.c(1446): End node .95:105:132 (-2846) INFO: fsg_search.c(1662): lattice start node hello.0 end node .95 INFO: ps_lattice.c(1352): Normalizer P(O) = alpha(:95:132) = -886169 INFO: ps_lattice.c(1390): Joint P(O,S) = -886169 P(S|O) = 0 2012-05-15 17:01:17.648543 [DEBUG] mod_pocketsphinx.c:383 Recognized: hello, Confidence: 100 2012-05-15 17:01:17.648543 [INFO] switch_cpp.cpp:1227 Callback with type event 2012-05-15 17:01:17.648543 [INFO] switch_cpp.cpp:1227 'Event-Name: DETECTED_SPEECH Core-UUID: 0a387574-9ee8-11e1-8fa1-43045b098f7a FreeSWITCH-Hostname: xxxxxx FreeSWITCH-Switchname: xxxxxx FreeSWITCH-IPv4: 199.15.96.29 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2012-05-15%2017%3A01%3A17 Event-Date-GMT: Wed,%2016%20May%202012%2000%3A01%3A17%20GMT Event-Date-Timestamp: 1337126477648543 Event-Calling-File: switch_ivr_async.c Event-Calling-Function: speech_thread Event-Calling-Line-Number: 3549 Event-Sequence: 1209 Speech-Type: detected-speech Content-Length: 172 hello ' EXECUTE sofia/xxxxxx/2143909113 detect_speech(pause) From gmaruzz at gmail.com Wed May 16 19:22:19 2012 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Wed, 16 May 2012 17:22:19 +0200 Subject: [Freeswitch-users] mod_gsmopen requires spandsp In-Reply-To: References: <1DE50890-EE01-4EEE-83A8-5B5F71BB3959@opencsta.org> <3484676B-F4CA-47E3-A8B0-9743291C1056@opencsta.org> <62D399FB-DFA0-4DD0-B86F-45E95026EF82@opencsta.org> <074EF57F-FF04-4BAE-BFD8-A9729C4EBB8B@opencsta.org> Message-ID: sorry people, now I fixed the wiki page. to compile libctb you: cd /usr/src/freeswitch/src/mod/endpoints/mod_gsmopen/libctb-0.16/build make DEBUG=0 GPIB=0 make DEBUG=0 GPIB=0 install ldconfig this will work :) -giovanni On Tue, May 15, 2012 at 2:01 PM, Wesley Akio wrote: > On CentOS 5 I just edited the Makefile to use ctbd and all went smooth... > Kind of... > > Sent from mobile, sorry for the typos.... > > Em 14/05/2012 23:37, "Chris Mylonas" escreveu: > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From gmaruzz at gmail.com Wed May 16 19:55:53 2012 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Wed, 16 May 2012 17:55:53 +0200 Subject: [Freeswitch-users] 2012 VoIP With Virtualization In-Reply-To: References: <550DADC3-9636-42FF-B391-BB00E84AC403@opencsta.org> Message-ID: On Tue, May 15, 2012 at 1:14 PM, Chris Mylonas wrote: > Beauty - thanks Jay - I'll check out what jails can do for me. ?I've just > compiled FS on 2 openvz servers (Denver/LA, I'm in Sydney) for testing a > bunch of stuff, firstly loading mod_gsmopen without it complaining about ctb > not being available. Chris, obviously do not forget for mod_gsmopen to "export" the serial devices from the "host" to the "guest", eg: from the bare metal to the containers. I use that daily, works perfect. -giovanni -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From andrew at cassidywebservices.co.uk Wed May 16 19:58:27 2012 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Wed, 16 May 2012 16:58:27 +0100 Subject: [Freeswitch-users] 2012 VoIP With Virtualization In-Reply-To: References: <550DADC3-9636-42FF-B391-BB00E84AC403@opencsta.org> Message-ID: I'm on the 512MB RAM tier. Occasionally FreeSWITCH craps itself and crashes leaving no logs but that's like every few weeks. I currently have my work calls routed through it, and take a fair few of them (one at a time mind, but I have call waiting set up on my phone so can have 2 sip sessions at once. Voicemail seems fine, and I have a conference call set up but it needs further testing. tel:+443300100002, sip:03300100002 at sip.v-pbx.net:5080 or ISN 1234*1432 will get you into the conference which is probably the best test of the media capabilities. I've not done a proper load test on it either, I've mainly been using it to build a front end. I've been told by customers that my voice breaks up a little, but since setting vad=none I've seen better results. Incoming audio seems to have no issues, mind so it could just be my home internet connection which today was upgraded to 30Mbps down 2Mbps up so I may see an improvement yet. I also installed voipmonitor yesterday, so will take a look at the logs later. On 16 May 2012 06:53, Gabriel Gunderson wrote: > On Tue, May 15, 2012 at 10:33 PM, Thaddeus Hogan > wrote: > > I have Freeswitch running on a KVM based VM right now, and it doesn't do > > too hot. Voicemail, MoH, and IVR prompts are choppy often but call media > > proxy (no transcode) works okay, though long calls start to deteriorate > > after about 30 minutes. > > I've used KVM with good results... what kind of load are we talking about? > > Gabe > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T *03300 100 960 *F *03300 100 961 *E *andrew at cassidywebservices.co.uk *W *www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120516/a4a82133/attachment-0001.html From gmaruzz at gmail.com Wed May 16 20:01:07 2012 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Wed, 16 May 2012 18:01:07 +0200 Subject: [Freeswitch-users] 2012 VoIP With Virtualization In-Reply-To: References: <550DADC3-9636-42FF-B391-BB00E84AC403@opencsta.org> Message-ID: I mean, OpenVZ works perfect for mod_gsmopen, that I was meaning. Also, there is consensus from the core developers that OpenVZ is the virtualization platform works with FreeSWITCH (mod_gsmopen and mod_skypopen included) without particular tweak. That's because OpenVZ is a kind of jail, so applications in guests have direct access to the IRQs of the one only kernel running on the bare metal (as opposed to Xen, VMWare, KVM, etc that runs many concurrent kernels on software simulated hardware machines). -giovanni On Wed, May 16, 2012 at 5:55 PM, Giovanni Maruzzelli wrote: > On Tue, May 15, 2012 at 1:14 PM, Chris Mylonas wrote: >> Beauty - thanks Jay - I'll check out what jails can do for me. ?I've just >> compiled FS on 2 openvz servers (Denver/LA, I'm in Sydney) for testing a >> bunch of stuff, firstly loading mod_gsmopen without it complaining about ctb >> not being available. > > Chris, obviously do not forget for mod_gsmopen to "export" the serial > devices from the "host" to the "guest", eg: from the bare metal to the > containers. I use that daily, works perfect. > > -giovanni > > > -- > Sincerely, > > Giovanni Maruzzelli > Cell : +39-347-2665618 -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From b2m at a-cti.com Wed May 16 20:01:52 2012 From: b2m at a-cti.com (Balamurugan Mahendran) Date: Wed, 16 May 2012 21:31:52 +0530 Subject: [Freeswitch-users] GSM - module Message-ID: Hi, Gsm got worked, two way signaling work. But now answer after answering the call. 2012-05-16 21:28:26.009824 [ERR] gsmopen_protocol.cpp:1853 rev 44fd0de|44fd0de[(nil)|37 ][ERRORA 1853 ][ ][-1,24,28] Error RE-sending (A): 10 -1 (Bad file descriptor) 2012-05-16 21:28:26.009824 [ERR] gsmopen_protocol.cpp:1859 rev 44fd0de|44fd0de[(nil)|37 ][ERRORA 1859 ][ ][-1,24,28] wrote -1 bytes!!! Nenormalno! Marking this gsmopen_serial_device /dev/ttyUSB3 as dead, andif it is owned by a channel, hanging up. Maybe the phone is stuck, switched off, power down or battery exhausted 2012-05-16 21:28:26.009824 [ERR] gsmopen_protocol.cpp:1862 rev 44fd0de|44fd0de[(nil)|37 ][ERRORA 1862 ][ ][-1,24,28] gsmopen_serial_monitor failed, declaring /dev/ttyUSB3 dead 2012-05-16 21:28:26.009824 [ERR] mod_gsmopen.cpp:2883 rev 44fd0de|44fd0de[(nil)|37 ][ERRORA 2883 ][ ][-1,24,28] ALARM on interface : 2012-05-16 21:28:27.009857 [ERR] gsmopen_protocol.cpp:2131 rev 44fd0de|44fd0de[(nil)|37 ][ERRORA 2131 ][ ][-1,24,28] Error sending data... (Bad file descriptor) 2012-05-16 21:28:27.009857 [ERR] gsmopen_protocol.cpp:2136 rev 44fd0de|44fd0de[(nil)|37 ][ERRORA 2136 ][ ][-1,24,28] wrote -1 bytes!!! Nenormalno! Marking this gsmopen_serial_device /dev/ttyUSB3 as dead, andif it is owned by a channel, hanging up. Maybe the phone is stuck, switched off, power down or battery exhausted 2012-05-16 21:28:27.009857 [ERR] gsmopen_protocol.cpp:2139 rev 44fd0de|44fd0de[(nil)|37 ][ERRORA 2139 ][ ][-1,24,28] gsmopen_serial_monitor failed, declaring /dev/ttyUSB3 dead 2012-05-16 21:28:27.009857 [ERR] mod_gsmopen.cpp:2883 rev 44fd0de|44fd0de[(nil)|37 ][ERRORA 2883 ][ ][-1,24,28] ALARM on interface : Thanks, Bala -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120516/defd645d/attachment.html From chris at opencsta.org Wed May 16 20:13:22 2012 From: chris at opencsta.org (Chris Mylonas) Date: Thu, 17 May 2012 02:13:22 +1000 Subject: [Freeswitch-users] 2012 VoIP With Virtualization In-Reply-To: References: <550DADC3-9636-42FF-B391-BB00E84AC403@opencsta.org> Message-ID: <07CE0BC3-5E2E-4557-A8BA-22C2E9955E5F@opencsta.org> Thanks G - the virtualization is a different set of testing. My iSCSI target (file server) has FS on it as well with the USB dongle (btw, i already had DEBUG=0, so will review). The virtualization is testing for queuing - have SIPp agents answer calls and echo the media. I only have a 100Mbit switch though. Thanks for the thought though :) On 17/05/2012, at 1:55 AM, Giovanni Maruzzelli wrote: > On Tue, May 15, 2012 at 1:14 PM, Chris Mylonas wrote: >> Beauty - thanks Jay - I'll check out what jails can do for me. I've just >> compiled FS on 2 openvz servers (Denver/LA, I'm in Sydney) for testing a >> bunch of stuff, firstly loading mod_gsmopen without it complaining about ctb >> not being available. > > Chris, obviously do not forget for mod_gsmopen to "export" the serial > devices from the "host" to the "guest", eg: from the bare metal to the > containers. I use that daily, works perfect. > > -giovanni > > > -- > Sincerely, > > Giovanni Maruzzelli > Cell : +39-347-2665618 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From gmaruzz at gmail.com Wed May 16 20:13:32 2012 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Wed, 16 May 2012 18:13:32 +0200 Subject: [Freeswitch-users] GSM - module In-Reply-To: References: Message-ID: It's evident it's not working, serial port is gone. Please follow exactly the steps in the wiki page for building and installing, and if your problem persists, follow this instructions: http://wiki.freeswitch.org/wiki/GSMopen#How_to_Report_Bugs_and_Feature_Requests -giovanni On Wed, May 16, 2012 at 6:01 PM, Balamurugan Mahendran wrote: > Hi, > > Gsm got worked, two way?signaling?work. But now answer after?answering?the > call. > > 2012-05-16 21:28:26.009824 [ERR] gsmopen_protocol.cpp:1853 rev > 44fd0de|44fd0de[(nil)|37 ? ? ][ERRORA ?1853 ][ ? ? ? ? ?][-1,24,28] Error > RE-sending (A): 10 -1 (Bad file descriptor) > 2012-05-16 21:28:26.009824 [ERR] gsmopen_protocol.cpp:1859 rev > 44fd0de|44fd0de[(nil)|37 ? ? ][ERRORA ?1859 ][ ? ? ? ? ?][-1,24,28] wrote -1 > bytes!!! Nenormalno! Marking this gsmopen_serial_device /dev/ttyUSB3 as > dead, andif it is owned by a channel, hanging up. Maybe the phone is stuck, > switched off, power down or battery exhausted > 2012-05-16 21:28:26.009824 [ERR] gsmopen_protocol.cpp:1862 rev > 44fd0de|44fd0de[(nil)|37 ? ? ][ERRORA ?1862 ][ ? ? ? ? ?][-1,24,28] > gsmopen_serial_monitor failed, declaring /dev/ttyUSB3 dead > 2012-05-16 21:28:26.009824 [ERR] mod_gsmopen.cpp:2883 rev > 44fd0de|44fd0de[(nil)|37 ? ? ][ERRORA ?2883 ][ ? ? ? ? ?][-1,24,28] ALARM on > interface : > 2012-05-16 21:28:27.009857 [ERR] gsmopen_protocol.cpp:2131 rev > 44fd0de|44fd0de[(nil)|37 ? ? ][ERRORA ?2131 ][ ? ? ? ? ?][-1,24,28] Error > sending data... (Bad file descriptor) > 2012-05-16 21:28:27.009857 [ERR] gsmopen_protocol.cpp:2136 rev > 44fd0de|44fd0de[(nil)|37 ? ? ][ERRORA ?2136 ][ ? ? ? ? ?][-1,24,28] wrote -1 > bytes!!! Nenormalno! Marking this gsmopen_serial_device /dev/ttyUSB3 as > dead, andif it is owned by a channel, hanging up. Maybe the phone is stuck, > switched off, power down or battery exhausted > 2012-05-16 21:28:27.009857 [ERR] gsmopen_protocol.cpp:2139 rev > 44fd0de|44fd0de[(nil)|37 ? ? ][ERRORA ?2139 ][ ? ? ? ? ?][-1,24,28] > gsmopen_serial_monitor failed, declaring /dev/ttyUSB3 dead > 2012-05-16 21:28:27.009857 [ERR] mod_gsmopen.cpp:2883 rev > 44fd0de|44fd0de[(nil)|37 ? ? ][ERRORA ?2883 ][ ? ? ? ? ?][-1,24,28] ALARM on > interface : > > > Thanks, > Bala > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From chris at opencsta.org Wed May 16 20:14:53 2012 From: chris at opencsta.org (Chris Mylonas) Date: Thu, 17 May 2012 02:14:53 +1000 Subject: [Freeswitch-users] 2012 VoIP With Virtualization In-Reply-To: References: <550DADC3-9636-42FF-B391-BB00E84AC403@opencsta.org> Message-ID: <7C76E769-E1F3-44CE-81D3-4C527CF2D2CF@opencsta.org> I'm in the process of setting up proxmox.com's KVM & OpenVZ on an old core 2 duo - so that will test the pants off the resources :) Thanks for the heads up. On 17/05/2012, at 2:01 AM, Giovanni Maruzzelli wrote: > I mean, OpenVZ works perfect for mod_gsmopen, that I was meaning. > > Also, there is consensus from the core developers that OpenVZ is the > virtualization platform works with FreeSWITCH (mod_gsmopen and > mod_skypopen included) without particular tweak. > > That's because OpenVZ is a kind of jail, so applications in guests > have direct access to the IRQs of the one only kernel running on the > bare metal (as opposed to Xen, VMWare, KVM, etc that runs many > concurrent kernels on software simulated hardware machines). > > -giovanni > > On Wed, May 16, 2012 at 5:55 PM, Giovanni Maruzzelli wrote: >> On Tue, May 15, 2012 at 1:14 PM, Chris Mylonas wrote: >>> Beauty - thanks Jay - I'll check out what jails can do for me. I've just >>> compiled FS on 2 openvz servers (Denver/LA, I'm in Sydney) for testing a >>> bunch of stuff, firstly loading mod_gsmopen without it complaining about ctb >>> not being available. >> >> Chris, obviously do not forget for mod_gsmopen to "export" the serial >> devices from the "host" to the "guest", eg: from the bare metal to the >> containers. I use that daily, works perfect. >> >> -giovanni >> >> >> -- >> Sincerely, >> >> Giovanni Maruzzelli >> Cell : +39-347-2665618 > > > > -- > Sincerely, > > Giovanni Maruzzelli > Cell : +39-347-2665618 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From chris at opencsta.org Wed May 16 20:16:20 2012 From: chris at opencsta.org (Chris Mylonas) Date: Thu, 17 May 2012 02:16:20 +1000 Subject: [Freeswitch-users] 2012 VoIP With Virtualization In-Reply-To: References: <550DADC3-9636-42FF-B391-BB00E84AC403@opencsta.org> Message-ID: <030952E7-AEE1-4DE1-BA1A-C9294E412813@opencsta.org> When it's not office hours, why don't you sent a trunk in from another box and pound the sh.. out of it to see where it breaks? On 17/05/2012, at 1:58 AM, Andrew Cassidy wrote: > I'm on the 512MB RAM tier. Occasionally FreeSWITCH craps itself and crashes leaving no logs but that's like every few weeks. I currently have my work calls routed through it, and take a fair few of them (one at a time mind, but I have call waiting set up on my phone so can have 2 sip sessions at once. Voicemail seems fine, and I have a conference call set up but it needs further testing. > > tel:+443300100002, sip:03300100002 at sip.v-pbx.net:5080 or ISN 1234*1432 will get you into the conference which is probably the best test of the media capabilities. > > I've not done a proper load test on it either, I've mainly been using it to build a front end. > > I've been told by customers that my voice breaks up a little, but since setting vad=none I've seen better results. Incoming audio seems to have no issues, mind so it could just be my home internet connection which today was upgraded to 30Mbps down 2Mbps up so I may see an improvement yet. > > I also installed voipmonitor yesterday, so will take a look at the logs later. > > On 16 May 2012 06:53, Gabriel Gunderson wrote: > On Tue, May 15, 2012 at 10:33 PM, Thaddeus Hogan wrote: > > I have Freeswitch running on a KVM based VM right now, and it doesn't do > > too hot. Voicemail, MoH, and IVR prompts are choppy often but call media > > proxy (no transcode) works okay, though long calls start to deteriorate > > after about 30 minutes. > > I've used KVM with good results... what kind of load are we talking about? > > Gabe > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -- > Andrew Cassidy BSc (Hons) MBCS SSCA > Managing Director > > > T 03300 100 960 F 03300 100 961 > E andrew at cassidywebservices.co.uk > W www.cassidywebservices.co.uk > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120517/c0abecbf/attachment-0001.html From gmaruzz at gmail.com Wed May 16 20:17:46 2012 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Wed, 16 May 2012 18:17:46 +0200 Subject: [Freeswitch-users] 2012 VoIP With Virtualization In-Reply-To: <7C76E769-E1F3-44CE-81D3-4C527CF2D2CF@opencsta.org> References: <550DADC3-9636-42FF-B391-BB00E84AC403@opencsta.org> <7C76E769-E1F3-44CE-81D3-4C527CF2D2CF@opencsta.org> Message-ID: please let us know all you will know :) On Wed, May 16, 2012 at 6:14 PM, Chris Mylonas wrote: > I'm in the process of setting up proxmox.com's KVM & OpenVZ on an old core 2 duo - so that will test the pants off the resources :) > Thanks for the heads up. > > On 17/05/2012, at 2:01 AM, Giovanni Maruzzelli wrote: > >> I mean, OpenVZ works perfect for mod_gsmopen, that I was meaning. >> >> Also, there is consensus from the core developers that OpenVZ is the >> virtualization platform works with FreeSWITCH (mod_gsmopen and >> mod_skypopen included) without particular tweak. >> >> That's because OpenVZ is a kind of jail, so applications in guests >> have direct access to the IRQs of the one only kernel running on the >> bare metal (as opposed to Xen, VMWare, KVM, etc that runs many >> concurrent kernels on software simulated hardware machines). >> >> -giovanni >> >> On Wed, May 16, 2012 at 5:55 PM, Giovanni Maruzzelli wrote: >>> On Tue, May 15, 2012 at 1:14 PM, Chris Mylonas wrote: >>>> Beauty - thanks Jay - I'll check out what jails can do for me. ?I've just >>>> compiled FS on 2 openvz servers (Denver/LA, I'm in Sydney) for testing a >>>> bunch of stuff, firstly loading mod_gsmopen without it complaining about ctb >>>> not being available. >>> >>> Chris, obviously do not forget for mod_gsmopen to "export" the serial >>> devices from the "host" to the "guest", eg: from the bare metal to the >>> containers. I use that daily, works perfect. >>> >>> -giovanni >>> >>> >>> -- >>> Sincerely, >>> >>> Giovanni Maruzzelli >>> Cell : +39-347-2665618 >> >> >> >> -- >> Sincerely, >> >> Giovanni Maruzzelli >> Cell : +39-347-2665618 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From dgarcia at anew.com.ve Wed May 16 20:24:00 2012 From: dgarcia at anew.com.ve (Saugort Dario Garcia Tovar) Date: Wed, 16 May 2012 11:54:00 -0430 Subject: [Freeswitch-users] About execute_on_tone_detect Message-ID: <4FB3D4A0.6090806@anew.com.ve> Hi, Could anyone give an example of "execute_on_tone_detect" use? for example to detect busy tone 425Hz or SIT I check the the wiki but I not sure about it. There is a lot of examples using the old format Thanks in advance guys From chris at opencsta.org Wed May 16 20:25:12 2012 From: chris at opencsta.org (Chris Mylonas) Date: Thu, 17 May 2012 02:25:12 +1000 Subject: [Freeswitch-users] 2012 VoIP With Virtualization In-Reply-To: References: <550DADC3-9636-42FF-B391-BB00E84AC403@opencsta.org> Message-ID: Check out http://wiki.opencsta.org/index.php/SIPp the registration scenario, or just set up IP based ACL without auth to send you a call every 10 seconds or so in a queue. (or set up another box and some sort of call generation mechanism) - I'm currently using SIPp to register and bailing out of executing after the first registration, then listening for any media and echoing it back for queues. They are very obedient agents :) I'm sure something similar could be done for the conference, just bridge one conference channel to a file - the SIPp buggers will echo it back! You'd be best putting some monitoring software on it too (zabbix.com is pretty good - might be a bit of a learning curve, but if you wanted to put the agent/proxy on there and route it back to my install i can give you a login - it's phoning home back to the zabbix server is over ssl tunnels) Maybe set debug on as well, hopefully that tells you a bit more. On 17/05/2012, at 1:58 AM, Andrew Cassidy wrote: > I'm on the 512MB RAM tier. Occasionally FreeSWITCH craps itself and crashes leaving no logs but that's like every few weeks. I currently have my work calls routed through it, and take a fair few of them (one at a time mind, but I have call waiting set up on my phone so can have 2 sip sessions at once. Voicemail seems fine, and I have a conference call set up but it needs further testing. > > tel:+443300100002, sip:03300100002 at sip.v-pbx.net:5080 or ISN 1234*1432 will get you into the conference which is probably the best test of the media capabilities. > > I've not done a proper load test on it either, I've mainly been using it to build a front end. > > I've been told by customers that my voice breaks up a little, but since setting vad=none I've seen better results. Incoming audio seems to have no issues, mind so it could just be my home internet connection which today was upgraded to 30Mbps down 2Mbps up so I may see an improvement yet. > > I also installed voipmonitor yesterday, so will take a look at the logs later. > > On 16 May 2012 06:53, Gabriel Gunderson wrote: > On Tue, May 15, 2012 at 10:33 PM, Thaddeus Hogan wrote: > > I have Freeswitch running on a KVM based VM right now, and it doesn't do > > too hot. Voicemail, MoH, and IVR prompts are choppy often but call media > > proxy (no transcode) works okay, though long calls start to deteriorate > > after about 30 minutes. > > I've used KVM with good results... what kind of load are we talking about? > > Gabe > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -- > Andrew Cassidy BSc (Hons) MBCS SSCA > Managing Director > > > T 03300 100 960 F 03300 100 961 > E andrew at cassidywebservices.co.uk > W www.cassidywebservices.co.uk > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120517/3598a7d1/attachment.html From gozdal at gmail.com Wed May 16 20:45:44 2012 From: gozdal at gmail.com (Marcin Gozdalik) Date: Wed, 16 May 2012 18:45:44 +0200 Subject: [Freeswitch-users] T.38/spandsp_fax interop problem Message-ID: Hi I'm trying to send fax from FreeSWITCH to PSTN using T.38. FS is connected to Broadsoft softswitch. The softswitch is connected to PSTN using SIP-trunk to external provider. Apparently there is no T.38 on Broadsoft and it is provided by the external provider (i.e. Broadsoft provides only T.38 pass-through). I can receive faxes destined at FS without a problem. I call rxfax and FS sends re-INVITE to Broadsoft and T.38 transmission begins. However, when I want to send fax, I can't get re-INVITE from Broadsoft with T.38. From the company that operates the Broadsoft softswitch I received a PCAP of working transmission with Linksys/SPA2102-5.2.10: SPA2102 issues re-INVITE itself when sending fax and T.38 is negotiated correctly. Unfortunately I can't get FS to issue the re-INVITE. I thought that a part of the problem is that FS receives from Broadsoft RTP-event Fax ANS (32) messages and it somehow interferes with fax-detecting routing in spandsp. Unfortunately, disabling telephone-event altogether didn't help - Broadsoft does not sent RTP-event packets but still FS does not re-INVITE. My sending script is as follows: var variables = "{"; variables += "loopback_bowout=false,"; variables += "ignore_early_media=true,"; variables += "origination_caller_id_number='" + callerid + "',"; variables += "origination_caller_id_name='" + callername + "',"; variables += "fax_ident='" + callerid + "',"; variables += "fax_header='" + callername + "',"; variables += "fax_email='" + email + "',"; variables += "fax_dest='" + dest + "',"; variables += "fax_tiff='" + tiff + "',"; variables += "fax_attempt='" + attempt + "',"; variables += "fax_max_attempts='" + max_attempts + "',"; variables += "fax_delay='" + delay + "',"; variables += "fax_enable_t38=true,"; variables += "fax_enable_t38_insist=true,"; variables += "fax_enable_t38_request=true,"; variables += "fax_disable_v17=false,"; variables += "fax_use_ecm=true,"; //variables += "fax_force_caller=true,"; variables += "fax_verbose=true"; variables += "}"; var s = new Session(variables + "loopback/"+dest+"/pbx/XML"); hook_run = 0; s.setHangupHook(hangup_hook); if (s.ready()) { console_log("DEBUG", "calling txfax\n"); s.execute("txfax", tiff); console_log("DEBUG", "returning from txfax" + "\n"); } else { console_log("INFO", "sending fax from " + callerid + " to " + dest + ": could not connect: " + s.cause + "(" + s.causecode + ")" + "\n"); return; } I do send the fax through loopback as I need it to establish rating session. I have experimented with various settings in fax_* variables. The only combination where anything happens (there is some T.30 negotiation in the logs) is when I set fax_enable_t38_request to false - then FS tries to negotiate T.30 transmission over G.711. I've never observed T.38 connection when sending fax, but it always ends with (20) Received no response to DCS or TCF. Usually the only thing there is in the logs is: 2012-05-16 17:27:19.824842 [DEBUG] mod_spandsp_fax.c:1357 Raw read codec activation Success L16 20000 2012-05-16 17:27:19.824842 [DEBUG] mod_spandsp_fax.c:1373 Raw write codec activation Success L16 2012-05-16 17:27:22.124834 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC signal status is Carrier up (-2) in state 18 2012-05-16 17:27:22.884832 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC signal status is Carrier down (-1) in state 18 2012-05-16 17:27:23.444834 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC signal status is Carrier up (-2) in state 18 2012-05-16 17:27:23.784833 [DEBUG] switch_rtp.c:3452 RTP RECV DTMF 2012-05-16 17:27:23.804833 [DEBUG] switch_rtp.c:3452 RTP RECV DTMF 2012-05-16 17:27:26.544834 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC sig nal status is Carrier down (-1) in state 18 2012-05-16 17:27:26.544834 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC signal status is Carrier up (-2) in state 18 2012-05-16 17:27:27.264833 [DEBUG] switch_rtp.c:3452 RTP RECV DTMF 2012-05-16 17:27:27.284873 [DEBUG] switch_rtp.c:3452 RTP RECV DTMF 2012-05-16 17:27:27.604841 [DEBUG] switch_rtp.c:3452 RTP RECV DTMF 2012-05- 16 17:27:28.884836 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC signal status is Abort (-8) in state 18 2012-05-16 17:27:28.884836 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC signal status is Carrier down (-1) in state 18 2012-05-16 17:27:30.704933 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC signal status is Carrier up (-2) in state 18 2012-05-16 17:27:30.764833 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC signal status is Carrier down (-1) in state 18 2012-05-16 17:27:31.904833 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC signal status is Carrier up (-2) in state 18 2012-05-16 17:27:34.544835 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC signal status is Carrier down (-1) in state 18 2012-05-16 17:27:37.544837 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC signal status is Carrier up (-2) in state 18 2012-05-16 17:27:40.164835 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC signal status is Carrier down (-1) in state 18 2012-05-16 17:27:41.184835 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC signal status is Carrier up (-2) in state 18 2012-05-16 17:27:41.224837 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC signal status is Abort (-8) in state 18 2012-05-16 17:27:41.224837 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC signal status is Carrier down (-1) in state 18 2012-05-16 17:27:43.184834 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC signal status is Carrier up (-2) in state 18 2012-05-16 17:27:44.824838 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC signal status is Carrier down (-1) in state 18 2012-05-16 17:27:45.664833 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC signal status is Carrier up (-2) in state 18 2012-05-16 17:27:45.664833 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC signal status is Carrier down (-1) in state 18 2012-05-16 17:27:45.724837 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC signal status is Carrier up (-2) in state 18 2012-05-16 17:27:45.724837 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC signal status is Carrier down (-1) in state 18 2012-05-16 17:27:45.784837 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC signal status is Carrier up (-2) in state 18 2012-05-16 17:27:45.804837 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 HDLC signal status is Carrier down (-1) in state 18 2012-05-16 17:27:46.304847 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 Status changing to 'The call dropped prematurely' 2012-05-16 17:27:46.304847 [DEBUG] mod_spandsp_fax.c:491 ============================================================================== 2012-05-16 17:27:46.304847 [DEBUG] mod_spandsp_fax.c:504 Fax processing not successful - result (49) The call dropped prematurely. 2012-05-16 17:27:46.304847 [DEBUG] mod_spandsp_fax.c:509 Remote station id: 2012-05-16 17:27:46.304847 [DEBUG] mod_spandsp_fax.c:510 Local station id: 2012-05-16 17:27:46.304847 [DEBUG] mod_spandsp_fax.c:511 Pages transferred: 0 2012-05-16 17:27:46.304847 [DEBUG] mod_spandsp_fax.c:513 Total fax pages: 0 2012-05-16 17:27:46.304847 [DEBUG] mod_spandsp_fax.c:514 Image resolution: 0x0 2012-05-16 17:27:46.304847 [DEBUG] mod_spandsp_fax.c:515 Transfer Rate: 14400 2012-05-16 17:27:46.304847 [DEBUG] mod_spandsp_fax.c:517 ECM status off 2012-05-16 17:27:46.304847 [DEBUG] mod_spandsp_fax.c:518 remote country: 2012-05-16 17:27:46.304847 [DEBUG] mod_spandsp_fax.c:519 remote vendor: 2012-05-16 17:27:46.304847 [DEBUG] mod_spandsp_fax.c:520 remote model: 2012-05-16 17:27:46.304847 [DEBUG] mod_spandsp_fax.c:522 ============================================================================== 2012-05-16 17:27:46.304847 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 Changing from state 18 to 32 2012-05-16 17:27:46.304847 [DEBUG] mod_spandsp_fax.c:286 FLOW T.30 Changing from phase T30_PHASE_A_CNG to T30_PHASE_CALL_FINISHED 2012-05-16 17:27:46.304847 [DEBUG] mod_spandsp_fax.c:286 FLOW FAX Set rx type 9 2012-05-16 17:27:46.304847 [DEBUG] mod_spandsp_fax.c:286 FLOW FAX FAX exchange complete 2012-05-16 17:27:46.304847 [DEBUG] mod_spandsp_fax.c:286 FLOW FAX Set tx type 9 2012-05-16 17:27:46.304847 [DEBUG] mod_spandsp_fax.c:286 FLOW FAX FAX exchange complete The same script transmits faxes correctly with Cisco gateway, but Cisco correctly re-INVITEs FS when FS is transmitting fax. PCAP dumps available, but I'd like to keep them private and not post on public pastebin. I'd appreciate any help on this issue. Best regards -- Marcin Gozdalik -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120516/41893014/attachment-0001.html From chris at opencsta.org Wed May 16 20:49:59 2012 From: chris at opencsta.org (Chris Mylonas) Date: Thu, 17 May 2012 02:49:59 +1000 Subject: [Freeswitch-users] 2012 VoIP With Virtualization In-Reply-To: References: <550DADC3-9636-42FF-B391-BB00E84AC403@opencsta.org> <7C76E769-E1F3-44CE-81D3-4C527CF2D2CF@opencsta.org> Message-ID: <7E2CB89D-7C21-4982-8C3B-9A43D1905118@opencsta.org> Sure... when I get stuff done. In my best efforts not to stuff up what I have, I resized my RAID from 2.8TB to 790GB accidentally. I wanted it set to 2000GB, not reduced by 2000GB!!! No damage luckily :) Here are proxmox's features: http://proxmox.com/products/proxmox-ve/features and they are impressive. The same host can do both KVM and OpenVZ - great for testing I guess. Parts are AGPL licensed, so if you were to offer any network services to clients (i.e. the REST API) - you know what that means... Nothing stopping you from making a front end for it that interacts with that web service though. The GUI is very polished from my couple of hours of waiting for my poor-man's-SAN to resize :) I won't have solid info on it for a while (i'm going up north for a few weeks so my output is greatly reduced), I'll blog about most of my stuff, including the telephony load testing. Cheers Chris On 17/05/2012, at 2:17 AM, Giovanni Maruzzelli wrote: > please let us know all you will know :) > > > > On Wed, May 16, 2012 at 6:14 PM, Chris Mylonas wrote: >> I'm in the process of setting up proxmox.com's KVM & OpenVZ on an old core 2 duo - so that will test the pants off the resources :) >> Thanks for the heads up. >> >> On 17/05/2012, at 2:01 AM, Giovanni Maruzzelli wrote: >> >>> I mean, OpenVZ works perfect for mod_gsmopen, that I was meaning. >>> >>> Also, there is consensus from the core developers that OpenVZ is the >>> virtualization platform works with FreeSWITCH (mod_gsmopen and >>> mod_skypopen included) without particular tweak. >>> >>> That's because OpenVZ is a kind of jail, so applications in guests >>> have direct access to the IRQs of the one only kernel running on the >>> bare metal (as opposed to Xen, VMWare, KVM, etc that runs many >>> concurrent kernels on software simulated hardware machines). >>> >>> -giovanni >>> >>> On Wed, May 16, 2012 at 5:55 PM, Giovanni Maruzzelli wrote: >>>> On Tue, May 15, 2012 at 1:14 PM, Chris Mylonas wrote: >>>>> Beauty - thanks Jay - I'll check out what jails can do for me. I've just >>>>> compiled FS on 2 openvz servers (Denver/LA, I'm in Sydney) for testing a >>>>> bunch of stuff, firstly loading mod_gsmopen without it complaining about ctb >>>>> not being available. >>>> >>>> Chris, obviously do not forget for mod_gsmopen to "export" the serial >>>> devices from the "host" to the "guest", eg: from the bare metal to the >>>> containers. I use that daily, works perfect. >>>> >>>> -giovanni >>>> >>>> >>>> -- >>>> Sincerely, >>>> >>>> Giovanni Maruzzelli >>>> Cell : +39-347-2665618 >>> >>> >>> >>> -- >>> Sincerely, >>> >>> Giovanni Maruzzelli >>> Cell : +39-347-2665618 >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> Join Us At ClueCon - Aug 7-9, 2012 >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > -- > Sincerely, > > Giovanni Maruzzelli > Cell : +39-347-2665618 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120517/94492cc0/attachment.html From philq at qsystemsengineering.com Wed May 16 20:57:48 2012 From: philq at qsystemsengineering.com (Phil Quesinberry) Date: Wed, 16 May 2012 12:57:48 -0400 Subject: [Freeswitch-users] Loud white noise during sleep Message-ID: <01a801cd3385$06487050$12d950f0$@com> Is there a setting to reduce the white noise generated during sleep? Setting suppress-cng to true has no effect on that noise. Setting send_silence_when_idle to a non-zero value appears to disable ringback in early media, so that's not a good solution. Is there another variable that can be set to an appropriate value, i.e. "silence"? Thanks, Phil Quesinberry Q Systems Engineering, Inc. Electronic Controls and Embedded Systems Development (410) 969-8002 http://www.qsystemsengineering.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120516/6437f1b6/attachment.html From gustavomarsico at gmail.com Wed May 16 23:46:34 2012 From: gustavomarsico at gmail.com (=?iso-8859-1?Q?Gustavo_M=E1rsico?=) Date: Wed, 16 May 2012 16:46:34 -0300 Subject: [Freeswitch-users] mod_callcenter real answer In-Reply-To: <38F79ADF-3F23-4E66-B41A-310CE4094218@gmail.com> References: <38F79ADF-3F23-4E66-B41A-310CE4094218@gmail.com> Message-ID: <8436E1D0-4017-4405-AE28-C3E11EDE1376@gmail.com> Evginey: These are the changes we made: --- mod_callcenter.c.orig 2012-01-11 20:23:34.000000000 -0300 +++ mod_callcenter.c 2012-01-26 15:31:15.000000000 -0300 @@ -1420,6 +1420,7 @@ switch_event_add_header_string(event, SWITCH_STACK_BOTTOM, "CC-Member-CID-Name", h->member_cid_name); switch_event_add_header_string(event, SWITCH_STACK_BOTTOM, "CC-Member-CID-Number", h->member_cid_number); switch_event_add_header_string(event, SWITCH_STACK_BOTTOM, "CC-Member-DNIS", member_dnis); + switch_event_add_header(event, SWITCH_STACK_BOTTOM, "CC-Member-Joined-Time", "%" SWITCH_TIME_T_FMT, t_member_called); switch_event_fire(&event); } @@ -2408,7 +2409,7 @@ } /* Make sure we answer the channel before getting the switch_channel_time_table_t answer time */ - switch_channel_answer(member_channel); + switch_channel_pre_answer(member_channel); /* Grab the start epoch of a channel */ times = switch_channel_get_timetable(member_channel); On May 15, 2012, at 7:19 PM, Gustavo M?rsico wrote: > Hi Evginey > > I made that change a few days ago, due a scenario where dialer is used. In fact, dialer dials to a queue first instead of customer. > All I did was replace in the code "answer" by "pre_answer" in some place. I've no the code right now, but I can send it to you later. > > Regards > > Gustavo > > > On May 15, 2012, at 6:15 AM, ??????? wrote: > >> Hi, colleagues. >> >> Could I have real answer for client on mod_callcenter, when agent answers? >> Or there is another way to queue call between 2 FreeSwitch. >> For example, client's call has to be routed to two FS servers, which has different agents in queues. But answer comes from first server when application CALLCENTER is called. How can I do this? >> >> Best Regards, Evginey. >> http://blog.buchnev.ru >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120516/16569c76/attachment-0001.html From member at linkedin.com Thu May 17 00:55:06 2012 From: member at linkedin.com (michael mendel via LinkedIn) Date: Wed, 16 May 2012 20:55:06 +0000 (UTC) Subject: [Freeswitch-users] Invitation to connect on LinkedIn Message-ID: <24038215.22333521.1337201706884.JavaMail.app@ela4-bed80.prod> LinkedIn ------------ michael mendel requested to add you as a connection on LinkedIn: ------------------------------------------ Zohair, I'd like to add you to my professional network on LinkedIn. - michael Accept invitation from michael mendel http://www.linkedin.com/e/xbphn8-h2avdchs-46/vPtmrrfmcGvVxWv84eLqdl0FlSkzdWO84S6qfAKHOSUhfgvFvWjMOLQ/blk/I101429536_45/1BpC5vrmRLoRZcjkkZt5YCpnlOt3RApnhMpmdzgmhxrSNBszYRd5YScPkVczgNc359bSR2h5xGdn1EbPwScz4OczoOcjsLrCBxbOYWrSlI/EML_comm_afe/?hs=false&tok=3CIA22Fr9D4lg1 View invitation from michael mendel http://www.linkedin.com/e/xbphn8-h2avdchs-46/vPtmrrfmcGvVxWv84eLqdl0FlSkzdWO84S6qfAKHOSUhfgvFvWjMOLQ/blk/I101429536_45/3kQnPoPdjAOd34MckALqnpPbOYWrSlI/svi/?hs=false&tok=2IQD2Fxu5D4lg1 ------------------------------------------ Why might connecting with michael mendel be a good idea? michael mendel's connections could be useful to you: After accepting michael mendel's invitation, check michael mendel's connections to see who else you may know and who you might want an introduction to. Building these connections can create opportunities in the future. -- (c) 2012, LinkedIn Corporation -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120516/d25cd3b6/attachment.html From thaddeus at thogan.com Thu May 17 02:07:47 2012 From: thaddeus at thogan.com (Thaddeus Hogan) Date: Wed, 16 May 2012 17:07:47 -0500 Subject: [Freeswitch-users] 2012 VoIP With Virtualization In-Reply-To: References: <550DADC3-9636-42FF-B391-BB00E84AC403@opencsta.org> Message-ID: Not much on this particular box. 20 registered clients, all behind NAT and from all over the US. Average of about 5 concurrent calls. Host is an Opteron 6100 series 8 core w/16 GB RAM running Ubuntu 10.04, guest has 4 GB RAM and exclusive access to 2 cores on the host. I figured the issue was KVM because the same setup on the same hardware configuration with an LXC guest is handles a testing load of similar proportions without issue. Everything here is moving to LXC over the next three months anyway. My observations with KVM must be considered under the premise that I have not identified any specific cause for the issues related to KVM. Also I have not run Freeswitch on Ubuntu 10.04 on bare metal. -- Thaddeus On 5/16/2012 12:53 AM, Gabriel Gunderson wrote: > On Tue, May 15, 2012 at 10:33 PM, Thaddeus Hogan wrote: >> I have Freeswitch running on a KVM based VM right now, and it doesn't do >> too hot. Voicemail, MoH, and IVR prompts are choppy often but call media >> proxy (no transcode) works okay, though long calls start to deteriorate >> after about 30 minutes. > > I've used KVM with good results... what kind of load are we talking about? > > Gabe > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From msc at freeswitch.org Thu May 17 02:17:08 2012 From: msc at freeswitch.org (Michael Collins) Date: Wed, 16 May 2012 15:17:08 -0700 Subject: [Freeswitch-users] bridge a call to registered user In-Reply-To: <4FB39A66.5070809@softnet.si> References: <4FB39A66.5070809@softnet.si> Message-ID: This is probably because the two different profiles are not set to use the same register domain. Check out this wiki page and try setting the value to be the same for all your profiles: http://wiki.freeswitch.org/wiki/Sofia.conf.xml#force-register-domain -MC On Wed, May 16, 2012 at 5:15 AM, Miha wrote: > Hi, > > just one question. I have created two profiles (suggested by @Michael) > as I was having same NAT issue. Profiles are basically the same only > registration port is different (5060, 5070) and NAT profile have set: > . > > In my dialplan I have made bridge like this: > > > > data="{origination_callee_id_name='${effective_caller_id_name}',origination_calee_id_number='${effective_caller_id_name}'}user/${ > destination_number}.fs_test at fs_test.fs2.blabla.com" > /> > > freeswitch at default> show re > > [ registrations] > > > freeswitch at default> show registrations > > reg_user,realm,token,url,expires,network_ip,network_port,network_proto,hostname > 018108753.fs_test,fs_test.fs2.blabla.com,40804176-fb1f176 at 172.31.1.190 > ,sofia/internal/sip:018108753.fs_test at xxx.xxx.xxx.xxx > :1265,1337169317,xxx.xxx.xxx.xxx,1265,udp,localhost.localdomain > 018108752.fs_test,fs_test.fs2.blabla.comi,873cfe40-1224f25 at 172.31.1.171 > ,sofia/nat/sip:018108752.fs_test at xxx.xxx.xxx.xxx > :2097,1337170569,xxx.xxx.xxx.xxx,1920,udp,localhost.localdomain > > 2 total. > > Why only phone work which is registered on NAT profile. IF I call users > which is registered on internal profile I get user not registered? > > Thanks! > > Miha > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120516/d473be43/attachment.html From jerry.richards at teotech.com Thu May 17 02:50:29 2012 From: jerry.richards at teotech.com (Jerry Richards) Date: Wed, 16 May 2012 22:50:29 +0000 Subject: [Freeswitch-users] SIP MESSAGE Source IP For Internal/External Config Message-ID: <1545146083A72C4DB7B66584B7E5D98402BD4B5B@BY2PRD0410MB377.namprd04.prod.outlook.com> Hello, I'm using CentOS 5.7 and the internal sip_profile for eth0 and the external sip_profile for eth1. I can register phones on either network, make calls between either network, observe presence updates on either network. Everything is working well, except I have one-way Instant-Messaging (IM) between a softphone on the eth0 network and a softphone on the eth1 network. An IM from internal-to-external works fine. An IM from external-to-internal is not displayed at the softphone. In the latter case, I see Freeswitch sending the SIP MESSAGE with correct content, but it's source IP address is the eth1/external IP, instead of the eth0/internal IP. I'm not sure if this is a bug or a configuration issue? Do you know where Freeswitch pulls the source IP in this case? Thanks, Jerry -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120516/61b98f13/attachment.html From msc at freeswitch.org Thu May 17 03:07:12 2012 From: msc at freeswitch.org (Michael Collins) Date: Wed, 16 May 2012 16:07:12 -0700 Subject: [Freeswitch-users] Run LUA script in different server In-Reply-To: References: Message-ID: I don't think you can directly do what you are describing. However, you might be able to use mod_httapi for this. There's some documentation on the wiki and in the module. Keep in mind that this is a relatively new module so we don't have lots of examples yet, so you'll probably be doing a fair amount of research and testing. -MC On Wed, May 16, 2012 at 5:59 AM, Sanath Prasanna wrote: > Hi all, > I have 2 servers. One server has SIP GW connection From Operator & IVR > applications need to build in other server. How to call distributed LUA > applications with Mysql Databases from the SIP GW server ? Pls advice. > Main idea is, maintaining SIP connection in one server & all the IVR > applications in other server. > Br, > Sanath > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120516/e500e29e/attachment-0001.html From mario_fs at mgtech.com Thu May 17 03:53:51 2012 From: mario_fs at mgtech.com (Mario G) Date: Wed, 16 May 2012 16:53:51 -0700 Subject: [Freeswitch-users] How to set "keep-alives re-use the TCP connection" ? In-Reply-To: References: Message-ID: <5B3DD049-C7AA-4084-9082-DABF85BA9720@mgtech.com> OK, does anyone know if this is even possible in FS? On May 15, 2012, at 5:36 PM, Mario G wrote: > Using Bria on iPad with a TCP connection, all works except after several hours it no longer registers to FreeSwitch. The Bria support says if this happens set the server PBX to use "keep-alives re-use the TCP connection". I could not find this options in the wiki, only thing close dealt with NAT but FS and the phones are all on local lan. Anyone know how to set this in the user definition? Thanks, > Mario G > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From bdfoster at endigotech.com Thu May 17 03:59:43 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Wed, 16 May 2012 19:59:43 -0400 Subject: [Freeswitch-users] How to set "keep-alives re-use the TCP connection" ? In-Reply-To: <5B3DD049-C7AA-4084-9082-DABF85BA9720@mgtech.com> References: <5B3DD049-C7AA-4084-9082-DABF85BA9720@mgtech.com> Message-ID: http://wiki.freeswitch.org/wiki/Sofia.conf.xml#nat-options-ping On May 16, 2012 7:54 PM, "Mario G" wrote: > OK, does anyone know if this is even possible in FS? > > On May 15, 2012, at 5:36 PM, Mario G wrote: > > > Using Bria on iPad with a TCP connection, all works except after several > hours it no longer registers to FreeSwitch. The Bria support says if this > happens set the server PBX to use "keep-alives re-use the TCP connection". > I could not find this options in the wiki, only thing close dealt with NAT > but FS and the phones are all on local lan. Anyone know how to set this in > the user definition? Thanks, > > Mario G > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120516/501b25b2/attachment.html From albert_nguyen16 at hotmail.com Thu May 17 05:50:32 2012 From: albert_nguyen16 at hotmail.com (Albert Nguyen) Date: Thu, 17 May 2012 01:50:32 +0000 Subject: [Freeswitch-users] Help on Explaination of the configurations parrametters Message-ID: Hi, I am new to FS and trying to setup a SBC using the example 2 in the FS wiki website. The link is http://wiki.freeswitch.org/wiki/SBC_FreeSWITCH_Configuration_Example_2. There are parameters in the example that I have to replaces for it to work with my own scenarios. However I have problem understanding the following lines. Is there anyone able to explain what does this mean The bits I am not sure is )\d+\.\d+(:\d+)(;dtg=\w+)?). What does this do? Thanks in advance. Al -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120517/ce8b967b/attachment.html From joohny at mail.ru Thu May 17 09:47:12 2012 From: joohny at mail.ru (=?UTF-8?B?0JXQstCz0LXQvdC40Lk=?=) Date: Thu, 17 May 2012 09:47:12 +0400 Subject: [Freeswitch-users] =?utf-8?q?mod=5Fcallcenter_real_answer?= Message-ID: Hi, Gustavo. Thank you for the patch. For information - I think there is new version of mod_callcenter for you in GIT, because lines in your patch is on the new places in new mod_callcenter.c. Do you have issue with no ring_ready sound on client side?Best Regards, Evginey. http://blog.buchnev.ru ------------------------------------------------------------- Evginey: These are the changes we made: --- mod_callcenter.c.orig2012-01-11 20:23:34.000000000 -0300 +++ mod_callcenter.c2012-01-26 15:31:15.000000000 -0300 @@ -1420,6 +1420,7 @@ ?switch_event_add_header_string(event, SWITCH_STACK_BOTTOM, "CC-Member-CID-Name", h->member_cid_name); ?switch_event_add_header_string(event, SWITCH_STACK_BOTTOM, "CC-Member-CID-Number", h->member_cid_number); ?switch_event_add_header_string(event, SWITCH_STACK_BOTTOM, "CC-Member-DNIS", member_dnis); +switch_event_add_header(event, SWITCH_STACK_BOTTOM, "CC-Member-Joined-Time", "%" SWITCH_TIME_T_FMT, t_member_called); ?switch_event_fire(&event); ?} ? @@ -2408,7 +2409,7 @@ ?} ? ?? /* Make sure we answer the channel before getting the switch_channel_time_table_t answer time */ -switch_channel_answer(member_channel); +switch_channel_pre_answer(member_channel); ? ?? /* Grab the start epoch of a channel */ ?times = switch_channel_get_timetable(member_channel); On May 15, 2012, at 7:19 PM, Gustavo M?rsico wrote: Hi Evginey I made that change a few days ago, due a scenario where dialer is used. In fact, dialer dials to a queue first instead of customer. All I did was replace in the code "answer" by "pre_answer" in some place. I've no the code right now, but I can send it to you later. Regards Gustavo On May 15, 2012, at 6:15 AM, ??????? wrote: Hi, colleagues. Could I have real answer for client on mod_callcenter, when agent answers? Or there is another way to queue call between 2 FreeSwitch.? For example, client's call has to be routed to two FS servers, which has different agents in queues. But answer comes from first server when application CALLCENTER is called. How can I do this? Best Regards, Evginey. http://blog.buchnev.ru -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120517/8356f790/attachment.html From miha at softnet.si Thu May 17 10:38:15 2012 From: miha at softnet.si (Miha) Date: Thu, 17 May 2012 08:38:15 +0200 Subject: [Freeswitch-users] bridge a call to registered user In-Reply-To: References: <4FB39A66.5070809@softnet.si> Message-ID: <4FB49CD7.3040509@softnet.si> HI @Michael, I tried to uncomment this due to Multi-tenant configuration. As this did not helped me I tried to use Presence variable (http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#Presence). After this FS does not log any more as user not registered. Could this function be causing any other problems? THanks! Miha On 5/17/2012 12:17 AM, Michael Collins wrote: > This is probably because the two different profiles are not set to use > the same register domain. Check out this wiki page and try setting the > value to be the same for all your profiles: > http://wiki.freeswitch.org/wiki/Sofia.conf.xml#force-register-domain > > -MC > > On Wed, May 16, 2012 at 5:15 AM, Miha > wrote: > > Hi, > > just one question. I have created two profiles (suggested by @Michael) > as I was having same NAT issue. Profiles are basically the same only > registration port is different (5060, 5070) and NAT profile have set: > . > > In my dialplan I have made bridge like this: > > > data="{origination_callee_id_name='${effective_caller_id_name}',origination_calee_id_number='${effective_caller_id_name}'}user/${destination_number}.fs_test at fs_test.fs2.blabla.com > " > /> > > freeswitch at default> show re > > [ registrations] > > > freeswitch at default> show registrations > reg_user,realm,token,url,expires,network_ip,network_port,network_proto,hostname > 018108753.fs_test,fs_test.fs2.blabla.com > ,40804176-fb1f176 at 172.31.1.190 > ,sofia/internal/sip:018108753.fs_test at xxx.xxx.xxx.xxx:1265,1337169317,xxx.xxx.xxx.xxx,1265,udp,localhost.localdomain > 018108752.fs_test,fs_test.fs2.blabla.comi,873cfe40-1224f25 at 172.31.1.171 > ,sofia/nat/sip:018108752.fs_test at xxx.xxx.xxx.xxx:2097,1337170569,xxx.xxx.xxx.xxx,1920,udp,localhost.localdomain > > 2 total. > > Why only phone work which is registered on NAT profile. IF I call > users > which is registered on internal profile I get user not registered? > > Thanks! > > Miha > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120517/b1d5bbb7/attachment-0001.html From vitaliy.davudov at vts24.ru Thu May 17 13:36:26 2012 From: vitaliy.davudov at vts24.ru (=?UTF-8?B?0JLQuNGC0LDQu9C40Lkg0JTQsNCy0YPQtNC+0LI=?=) Date: Thu, 17 May 2012 13:36:26 +0400 Subject: [Freeswitch-users] Variable continue on fail In-Reply-To: <4FAB719E.9010800@vts24.ru> References: <4F87EF60.3070105@vts24.ru> <4F8822A5.9090402@vts24.ru> <4FA4D2D6.1010908@vts24.ru> <4FAB719E.9010800@vts24.ru> Message-ID: <4FB4C69A.7070108@vts24.ru> Hi! Is there any hope that this problem will be solved? 10.05.2012 11:43, ??????? ??????? ???????: > Hi, Michael! > > I added a new post on pastebin: > > http://pastebin.freeswitch.org/19026 > > > 07.05.2012 5:14, Michael Collins ???????: >> I don't recall you ever posting a console debug log to >> pastebin.freeswitch.org . Could you >> please do that? It will help us to know exactly what is happening. >> -MC >> >> On Sat, May 5, 2012 at 12:12 AM, ??????? ??????? >> > wrote: >> >> Hi! >> Is there any news on this issue? >> >> 13.04.2012 16:57, ??????? ??????? ???????: >>> Yes, you are right! >>> >>> I did it: http://pastebin.freeswitch.org/18863 >>> >>> Additionally: >>> I've included in this extension new line: >>> >>> >>> >>> >>> ** >>> >> data="continue_on_fail=1,2,3,6,25,34,38,41,42,44,47,63,66,500,501"/> >>> >>> >>> >>> >>> >>> >>> Without that line a similar situation occurs if FS recieve >>> /NORMAL_CLEARING./ >>> >>> 13.04.2012 13:44, Anton Kvashenkin ???????: >>>> Ok, i got it. Even that there is no USER_BUSY at >>>> continue_on_fail variable, FS still tries to reach the second >>>> action, am i right? >>>> >>>> So, for better debugging, i suggest to paste full call log with >>>> enabled siptrace and /log 7 to pastebin.freeswitch.org >>>> . >>>> >>>> 13 ?????? 2012 ?. 13:18 ???????????? ??????? ??????? >>>> > >>>> ???????: >>>> >>>> Hi all! >>>> >>>> In my dialplan I've included variable continue on fail: >>>> >>>> >>>> >>>> >>>> >>> data="continue_on_fail=1,2,3,6,25,34,38,41,42,44,47,63,66,500,501"/> >>>> >>>> >>>> >>>> >>>> >>>> >>>> And if FS recieve from first gateway USER_BUSY, then FS >>>> try to bridge >>>> this call to another gateway. Although in line >>> application="set" >>>> data="continue_on_fail=1,2,3,6,25,34,38,41,42,44,47,63,66,500,501"/> >>>> there is no code Q.850 = 17. >>>> How resolve this issue? >>>> >>>> -- >>>> Best regards, >>>> Vitaly Davudov >>>> "VIP-TELECOM-SERVICE" Ltd. >>>> ("ETERIA" Group of companies) >>>> http://www.vts24.ru >>>> >>>> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > -- > ? ?????????? ???????????, > ??????? ??????? ????????? > ??? "???-???????-??????" > (?????? ???????? "ETERIA") > http://www.vts24.ru > ???: (495) 989-47-00 -- ? ?????????? ???????????, ??????? ??????? ????????? ??? "???-???????-??????" (?????? ???????? "ETERIA") http://www.vts24.ru ???: (495) 989-47-00 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120517/0bb76c61/attachment.html From anton.jugatsu at gmail.com Thu May 17 13:42:05 2012 From: anton.jugatsu at gmail.com (Anton Kvashenkin) Date: Thu, 17 May 2012 13:42:05 +0400 Subject: [Freeswitch-users] Variable continue on fail In-Reply-To: <4FB4C69A.7070108@vts24.ru> References: <4F87EF60.3070105@vts24.ru> <4F8822A5.9090402@vts24.ru> <4FA4D2D6.1010908@vts24.ru> <4FAB719E.9010800@vts24.ru> <4FB4C69A.7070108@vts24.ru> Message-ID: To solve this issue, I think it would be better if you will open the ticket at bugtracker http://jira.freeswitch.org/secure/Dashboard.jspa . 2012/5/17 ??????? ??????? > Hi! > Is there any hope that this problem will be solved? > > 10.05.2012 11:43, ??????? ??????? ???????: > > Hi, Michael! > > I added a new post on pastebin: > > http://pastebin.freeswitch.org/19026 > > > 07.05.2012 5:14, Michael Collins ???????: > > I don't recall you ever posting a console debug log to > pastebin.freeswitch.org. Could you please do that? It will help us to > know exactly what is happening. > -MC > > On Sat, May 5, 2012 at 12:12 AM, ??????? ??????? < > vitaliy.davudov at vts24.ru> wrote: > >> Hi! >> Is there any news on this issue? >> >> 13.04.2012 16:57, ??????? ??????? ???????: >> >> Yes, you are right! >> >> I did it: http://pastebin.freeswitch.org/18863 >> >> Additionally: >> I've included in this extension new line: >> >> >> >> >> ** >> > data="continue_on_fail=1,2,3,6,25,34,38,41,42,44,47,63,66,500,501"/> >> >> >> >> >> >> >> Without that line a similar situation occurs if FS recieve * >> NORMAL_CLEARING.* >> >> 13.04.2012 13:44, Anton Kvashenkin ???????: >> >> Ok, i got it. Even that there is no USER_BUSY at continue_on_fail >> variable, FS still tries to reach the second action, am i right? >> >> So, for better debugging, i suggest to paste full call log with enabled >> siptrace and /log 7 to pastebin.freeswitch.org. >> >> 13 ?????? 2012 ?. 13:18 ???????????? ??????? ??????? < >> vitaliy.davudov at vts24.ru> ???????: >> >>> Hi all! >>> >>> In my dialplan I've included variable continue on fail: >>> >>> >>> >>> >>> >> data="continue_on_fail=1,2,3,6,25,34,38,41,42,44,47,63,66,500,501"/> >>> >>> >>> >>> >>> >>> >>> And if FS recieve from first gateway USER_BUSY, then FS try to bridge >>> this call to another gateway. Although in line >> data="continue_on_fail=1,2,3,6,25,34,38,41,42,44,47,63,66,500,501"/> >>> there is no code Q.850 = 17. >>> How resolve this issue? >>> >>> -- >>> Best regards, >>> Vitaly Davudov >>> "VIP-TELECOM-SERVICE" Ltd. >>> ("ETERIA" Group of companies) >>> http://www.vts24.ru >>> >>> >>> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > -- > ? ?????????? ???????????, > ??????? ??????? ????????? > ??? "???-???????-??????" > (?????? ???????? "ETERIA")http://www.vts24.ru > ???: (495) 989-47-00 > > > -- > ? ?????????? ???????????, > ??????? ??????? ????????? > ??? "???-???????-??????" > (?????? ???????? "ETERIA")http://www.vts24.ru > ???: (495) 989-47-00 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120517/b391bc30/attachment-0001.html From anita.hall at simmortel.com Thu May 17 15:33:58 2012 From: anita.hall at simmortel.com (Anita Hall) Date: Thu, 17 May 2012 17:03:58 +0530 Subject: [Freeswitch-users] Fax tones for India ? Message-ID: Hi In spandsp.conf.xml, the following tones are given for North America. And similarly, others are given for UK and Germany. What will be the Fax tones for India ? 20 21 22 23 24 25 26 27 28 29 30 31 32 33 34 35 36 37 regards, Anita -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120517/0c3b0d9b/attachment.html From anita.hall at simmortel.com Thu May 17 15:35:39 2012 From: anita.hall at simmortel.com (Anita Hall) Date: Thu, 17 May 2012 17:05:39 +0530 Subject: [Freeswitch-users] Run LUA script in different server In-Reply-To: References: Message-ID: You could run a Lua ESL server on a different machine but this will not be the same as running a Lua script. http://wiki.freeswitch.org/wiki/Event_Socket_Library regards, Anita On Thu, May 17, 2012 at 4:37 AM, Michael Collins wrote: > I don't think you can directly do what you are describing. However, you > might be able to use mod_httapi for this. There's some documentation on the > wiki and in the module. Keep in mind that this is a relatively new module > so we don't have lots of examples yet, so you'll probably be doing a fair > amount of research and testing. > > -MC > > > On Wed, May 16, 2012 at 5:59 AM, Sanath Prasanna wrote: > >> Hi all, >> I have 2 servers. One server has SIP GW connection From Operator & IVR >> applications need to build in other server. How to call distributed LUA >> applications with Mysql Databases from the SIP GW server ? Pls advice. >> Main idea is, maintaining SIP connection in one server & all the IVR >> applications in other server. >> Br, >> Sanath >> >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120517/e9869e60/attachment.html From dgarcia at anew.com.ve Thu May 17 16:51:24 2012 From: dgarcia at anew.com.ve (Saugort Dario Garcia Tovar) Date: Thu, 17 May 2012 08:21:24 -0430 Subject: [Freeswitch-users] Fax tones for India ? In-Reply-To: References: Message-ID: <4FB4F44C.3060601@anew.com.ve> mmm, check i this links helps you http://www.3amsystems.com/wireline/tone-search.htm On 5/17/2012 7:03 AM, Anita Hall wrote: > Hi > > In spandsp.conf.xml, the following tones are given for North America. > And similarly, others are given for UK and Germany. What will be the > Fax tones for India ? > > > 20 > 21 > 22 > 23 > 24 > 25 > 26 > 27 > 28 > 29 > 30 > 31 > 32 > 33 > 34 > 35 > 36 > 37 > > > regards, > Anita > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.2176 / Virus Database: 2425/5004 - Release Date: 05/16/12 > -- Atentamente, *Dario Garc?a* Consultor. CCCT, Nivel C2, Sector Yarey, Mz, Ofc. MZ03a. Caracas-Venezuela. Tel?fono: +58 212 9081842 Cel: +58 412 2221515 dgarcia at anew.com.ve http://www.anew.com.ve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120517/f4584d28/attachment.html From freeswitch-list at puzzled.xs4all.nl Thu May 17 17:02:06 2012 From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists) Date: Thu, 17 May 2012 15:02:06 +0200 Subject: [Freeswitch-users] Fax tones for India ? In-Reply-To: References: Message-ID: <4FB4F6CE.2000507@puzzled.xs4all.nl> On 17-05-12 13:33, Anita Hall wrote: > Hi > > In spandsp.conf.xml, the following tones are given for North America. > And similarly, others are given for UK and Germany. What will be the Fax > tones for India ? http://www.itu.int/dms_pub/itu-t/opb/sp/T-SP-E.180-2010-PDF-E.pdf Regards, Patrick From cristian.re.work at gmail.com Thu May 17 17:23:43 2012 From: cristian.re.work at gmail.com (cristian re) Date: Thu, 17 May 2012 15:23:43 +0200 Subject: [Freeswitch-users] Detect answer on FXO device Message-ID: The ATA has both "call answer supervision" and "polarity reversal" but they doesn't works worldwide but only with some carriers. Is there a way to delegate this job to Freeswitch? using voice activity detection? Cristian >Technically, answer detection is the job of the ATA. The problem is that it >is an inexact science. I would look in the GXW docs to see if they have a >section on 'answer supervision'. It might also be in a section along with >'disconnect supervision.' > >-MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120517/bd702c7b/attachment.html From anita.hall at simmortel.com Thu May 17 17:51:32 2012 From: anita.hall at simmortel.com (Anita Hall) Date: Thu, 17 May 2012 19:21:32 +0530 Subject: [Freeswitch-users] Spandsp mulated modems inbound/no /dev/FS* device? In-Reply-To: <4FA696B1.8050706@coppice.org> References: <4FA24100.7040908@integrafin.co.uk> <20120503151946.GA824@eagle.cupis.co.uk> <4FA3EE97.1090908@integrafin.co.uk> <4FA596BF.4090703@integrafin.co.uk> <4FA635AE.5050006@integrafin.co.uk> <4FA66862.1090300@coppice.org> <4FA6871F.5050909@puzzled.xs4all.nl> <4FA696B1.8050706@coppice.org> Message-ID: Hi Alex Did you get the answer to your first question ? Did you succeed in using the emulated modem option for taking fax calls to hylafax ? I put the following in spandsp.conf.xml and got /dev/FS[0-4] devices as soft links to /dev/pts/[4-8]. My freeswitch is running as root user so I did not face the issue you did. What next? Could you point me to some doc in hylafax? (And before Steve lashes out at me again, I must clarify, I do not want to just play around, but my boss is clueless and wants me to evaluate hylafax :() regards, Anita On Sun, May 6, 2012 at 8:50 PM, Steve Underwood wrote: > On 05/06/2012 10:13 PM, Patrick Lists wrote: > > On 06-05-12 14:02, Steve Underwood wrote: > >> Hi Alex, > >> > >> Its the user ID which is causing your problem. If you just add the > following > >> > >> > >> > >> to your configuration you will get five /dev/FS* devices if you run as > >> root, but none if you run as freeswitch. This ought to be fixed. I think > > Could a udev rule with user=... (and group=...) perhaps fix that? > > > >> the device names are a bit questionable, too. Calling the /dev/FS/modem* > >> or even /dev/FS/t31-* might increase flexibility for the future where > >> more device types might be needed. > > +1 > Its not the actual pts devices which cause problems. Those are happily > created with any user name that's tried. It is the /dev/FS* links which > won't create unless you are root. I'm not clear how udev can be used to > make those creatable by someone like the freeswitch user. If you create > a directory /dev/FS, make that writable by all, change FS to create the > link as /dev/FS/*, and run as user freeswitch things work OK. However, > /dev/FS is lost on reboot, because of the dynamic nature of /dev. > > Steve > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120517/a669ccdb/attachment-0001.html From steveu at coppice.org Thu May 17 17:55:27 2012 From: steveu at coppice.org (Steve Underwood) Date: Thu, 17 May 2012 21:55:27 +0800 Subject: [Freeswitch-users] Fax tones for India ? In-Reply-To: References: Message-ID: <4FB5034F.4010705@coppice.org> Hi, I don't know why that small file was created from the original in spandsp. If you look in spandsp/global-tones.xml you will find all the world's tones. Steve On 05/17/2012 07:33 PM, Anita Hall wrote: > Hi > > In spandsp.conf.xml, the following tones are given for North America. > And similarly, others are given for UK and Germany. What will be the > Fax tones for India ? > > > 20 > 21 > 22 > 23 > 24 > 25 > 26 > 27 > 28 > 29 > 30 > 31 > 32 > 33 > 34 > 35 > 36 > 37 From anita.hall at simmortel.com Thu May 17 18:23:31 2012 From: anita.hall at simmortel.com (Anita Hall) Date: Thu, 17 May 2012 19:53:31 +0530 Subject: [Freeswitch-users] Fax tones for India ? In-Reply-To: <4FB5034F.4010705@coppice.org> References: <4FB5034F.4010705@coppice.org> Message-ID: Thanks Steve! The xml tags for tone description in spandsp/global-tones.xml and spandsp.conf.xml are different. Will that make a difference ? regards, Anita On Thu, May 17, 2012 at 7:25 PM, Steve Underwood wrote: > Hi, > > I don't know why that small file was created from the original in > spandsp. If you look in spandsp/global-tones.xml you will find all the > world's tones. > > Steve > > > On 05/17/2012 07:33 PM, Anita Hall wrote: > > Hi > > > > In spandsp.conf.xml, the following tones are given for North America. > > And similarly, others are given for UK and Germany. What will be the > > Fax tones for India ? > > > > > > 20 > > 21 > > 22 > > 23 > > 24 > > 25 > > 26 > > 27 > > 28 > > 29 > > 30 > > 31 > > 32 > > 33 > > 34 > > 35 > > 36 > > 37 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120517/b0991bba/attachment.html From spencer at 5ninesolutions.com Thu May 17 19:58:16 2012 From: spencer at 5ninesolutions.com (Spencer Thomason) Date: Thu, 17 May 2012 08:58:16 -0700 Subject: [Freeswitch-users] Polycom auto answer on active call Message-ID: <6987271E-B52C-4D9C-8B2E-D37808D9B6F9@5ninesolutions.com> Hello, I'm having a problem with Polycom endpoints where the phone auto answers a page even if another call is in progress. My dialplan looks like this: Does anyone have any suggestions for an elegant way to solve this? Thanks, Spencer From yehavi.bourvine at gmail.com Thu May 17 20:09:36 2012 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Thu, 17 May 2012 19:09:36 +0300 Subject: [Freeswitch-users] Polycom auto answer on active call In-Reply-To: <6987271E-B52C-4D9C-8B2E-D37808D9B6F9@5ninesolutions.com> References: <6987271E-B52C-4D9C-8B2E-D37808D9B6F9@5ninesolutions.com> Message-ID: Use the mod_limit to check whether there is already a call on his extension, and if so - do not try to bridge with auto answer. __Yehavi: 2012/5/17 Spencer Thomason > Hello, I'm having a problem with Polycom endpoints where the phone auto > answers a page even if another call is in progress. My dialplan looks like > this: > > > > > > > > > > Does anyone have any suggestions for an elegant way to solve this? > > Thanks, > Spencer > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120517/b9690175/attachment.html From philq at qsystemsengineering.com Thu May 17 20:13:05 2012 From: philq at qsystemsengineering.com (Phil Quesinberry) Date: Thu, 17 May 2012 12:13:05 -0400 Subject: [Freeswitch-users] Loud whilte noise during sleep Message-ID: <00d201cd3447$f3ecb0b0$dbc61210$@com> ** Disclaimer: this is probably a really bad idea ** Just in case this is frustrating anyone else: Silence is generated by the switch_generate_sln_silence function(s) in switch_resample.c, 'divisor' is the variable which determines the noise volume - the higher the value, the lower the noise volume. So, a quick fix for this problem would be to hard-code a value like 1400 for 'divisor' within those functions, by adding divisor = 1400 before the for loop and recompiling. This will affect ALL silence generated by FreeSwitch so it would be better to find where this is set within the sleep function so it only affects noise generated there. Better yet, if there's a way to accomplish this without hacking the code, (i.e. a variable that can be set), then by all means do that instead (and let me know what it is). Setting send_silence_when_idle to the desired value curbs the noise during sleep but it sends silence during ringback too, at least in my configuration here. Regards, Phil Quesinberry Q Systems Engineering, Inc. Electronic Controls and Embedded Systems Development (410) 969-8002 http://www.qsystemsengineering.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120517/5f564068/attachment.html From anthony.minessale at gmail.com Thu May 17 20:21:35 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 17 May 2012 11:21:35 -0500 Subject: [Freeswitch-users] Loud whilte noise during sleep In-Reply-To: <00d201cd3447$f3ecb0b0$dbc61210$@com> References: <00d201cd3447$f3ecb0b0$dbc61210$@com> Message-ID: Which revision of the code are you looking at? The sleep sends dummy frames unless you set that send_silence_when_idle var, if its not set its probably not sending anything. You can see in the code that unless that var is set it wont be generating anything. On Thu, May 17, 2012 at 11:13 AM, Phil Quesinberry wrote: > ** Disclaimer: this is probably a really bad idea ** > > Just in case this is frustrating anyone else: > > Silence is generated by the switch_generate_sln_silence function(s) in > switch_resample.c, 'divisor' is the variable which determines the noise > volume - the higher the value, the lower the noise volume. > > So, a quick fix for this problem would be to hard-code a value like 1400 for > 'divisor' within those functions, by adding divisor = 1400 before the for > loop and recompiling.? This will affect ALL silence generated by FreeSwitch > so it would be better to find where this is set within the sleep function so > it only affects noise generated there. > > Better yet, if there's a way to accomplish this without hacking the code, > (i.e. a variable that can be set), then by all means do that instead (and > let me know what it is). > > Setting send_silence_when_idle to the desired value curbs the noise during > sleep but it sends silence during ringback too, at least in my configuration > here. > > Regards, > > Phil Quesinberry > > Q Systems Engineering, Inc. > > Electronic Controls and Embedded Systems Development > > (410) 969-8002 > > http://www.qsystemsengineering.com > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From rrodolfos at gmail.com Thu May 17 20:21:57 2012 From: rrodolfos at gmail.com (RrodolfoS .) Date: Thu, 17 May 2012 11:51:57 -0430 Subject: [Freeswitch-users] mod_say_es fail when tray pronounce voicemails messages dates Message-ID: mod_say_es fail when tray pronounce voicemails messages dates, in english work well but in espanish fail. In spanish the ivr and voicemail intro work fine, except the voicemails messages. The logs is the same of this http://www.freeswitch.es/node/14755 Configs details: Version FreeSWITCH Version 1.1.beta1 (git-4283408 2012-04-29 11-33-24 -0400) /usr/src/freeswitch/modules.conf say/mod_say_es # make mod_say_es-install /usr/local/freeswitch/conf/freeswitch.xml
/usr/local/freeswitch/conf/vars.xml /usr/local/freeswitch/conf/lang/es/es.xml /usr/local/freeswitch/conf/autoload_configs/modules.conf.xml The spanish sounds are in /usr/local/freeswitch/sounds/es/mx/maria/ with 8000, 16000, 32000, 44100 and 48000 sample rate mono. /usr/local/freeswitch/conf/lang/es are a modified copy of /usr/local/freeswitch/conf/lang/en From modesto at isimples.com.br Thu May 17 20:25:04 2012 From: modesto at isimples.com.br (Antonio Modesto) Date: Thu, 17 May 2012 13:25:04 -0300 Subject: [Freeswitch-users] Portuguese Sounds In-Reply-To: <4FB39FA2.7040801@tecnomega.com.br> References: <1337170133.2975.12.camel@modesto.localdomain.net> <4FB39FA2.7040801@tecnomega.com.br> Message-ID: <1337271904.2977.36.camel@modesto.localdomain.net> Thank you very much! Regards. On Wed, 2012-05-16 at 09:37 -0300, Carlos Cesario wrote: > Hi Antonio, > > Yes, you can download it in > http://wirelessmundi.com/freeswitch-sounds-pt-BR-karina-48000-1.0.15.tar.gz > > att, > > Em 16-05-2012 09:08, Antonio Modesto escreveu: > > > Hi, > > > > Does anybody knows if there is a portuguese version of the default > > freeswitch sounds? I only have found it in English and Russian. > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Atenciosamente, Ant?nio Modesto Gerente de TI Pra?a Get?lio Vargas, 77 ? Sala 308 ? Centro Santo Ant?nio do Monte ? MG ? CEP: 35560-000 Tel:(37) 3281-2800 Contato: isimples at isimples.com.br http://www.isimples.com.br Aviso:Esta mensagem e quaisquer arquivos em anexo podem conter informa??es confidenciais e/ou privilegiadas. Se voc? n?o for o destinat?rio ou a pessoa autorizada a receber esta mensagem, por favor, n?o leia, copie, repasse, imprima, guarde, nem tome qualquer a??o baseada nessas informa??es. Notifique o remetente imediatamente por e-mail e apague a mensagem permanentemente. Aten??o: embora a Isimples Telecom, tome seus cuidados para garantir a aus?ncia de v?rus neste e-mail, a empresa n?o se responsabiliza por quaisquer perdas ou danos decorrentes do uso da mensagem e seus anexos. A seguran?a e aus?ncia de erros na transmiss?o do e-mail n?o podem ser garantidas, j? que as informa??es podem ser interceptadas, corrompidas, perdidas, destru?das, atrasadas, chegarem incompletas, ou, ainda, conter v?rus. Recomendamos checar se o e-mail e seus anexos cont?m v?rus, uma vez que nem a Isimples Telecom ou o remetente se responsabilizam pela transmiss?o destes. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120517/a12931a0/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: logo_isimples.png Type: image/png Size: 18197 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120517/a12931a0/attachment-0001.png From mario_fs at mgtech.com Thu May 17 20:27:34 2012 From: mario_fs at mgtech.com (Mario G) Date: Thu, 17 May 2012 09:27:34 -0700 Subject: [Freeswitch-users] How to set "keep-alives re-use the TCP connection" ? In-Reply-To: References: <5B3DD049-C7AA-4084-9082-DABF85BA9720@mgtech.com> Message-ID: This is what is confusing, there is no NATing involved... So it still applies to non-natted as well? Mario G On May 16, 2012, at 4:59 PM, Brian Foster wrote: > http://wiki.freeswitch.org/wiki/Sofia.conf.xml#nat-options-ping > > On May 16, 2012 7:54 PM, "Mario G" wrote: > OK, does anyone know if this is even possible in FS? > > On May 15, 2012, at 5:36 PM, Mario G wrote: > > > Using Bria on iPad with a TCP connection, all works except after several hours it no longer registers to FreeSwitch. The Bria support says if this happens set the server PBX to use "keep-alives re-use the TCP connection". I could not find this options in the wiki, only thing close dealt with NAT but FS and the phones are all on local lan. Anyone know how to set this in the user definition? Thanks, > > Mario G > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120517/74b67615/attachment.html From spencer at 5ninesolutions.com Thu May 17 20:37:13 2012 From: spencer at 5ninesolutions.com (Spencer Thomason) Date: Thu, 17 May 2012 09:37:13 -0700 Subject: [Freeswitch-users] Polycom auto answer on active call In-Reply-To: References: <6987271E-B52C-4D9C-8B2E-D37808D9B6F9@5ninesolutions.com> Message-ID: Thanks that should work. How did you keep track of the calls? Calls from an extension are easy enough but an extension can be called many ways, i.e. hunts groups, transfers, etc that would need to be accounted using limit. Could you do this by modifying the dial-string? Spencer On May 17, 2012, at 9:09 AM, Yehavi Bourvine wrote: > Use the mod_limit to check whether there is already a call on his extension, and if so - do not try to bridge with auto answer. > > __Yehavi: > > 2012/5/17 Spencer Thomason > Hello, I'm having a problem with Polycom endpoints where the phone auto answers a page even if another call is in progress. My dialplan looks like this: > > > > > > > > > > Does anyone have any suggestions for an elegant way to solve this? > > Thanks, > Spencer > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120517/7f0cd66c/attachment.html From peter.olsson at visionutveckling.se Thu May 17 20:47:56 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Thu, 17 May 2012 16:47:56 +0000 Subject: [Freeswitch-users] Loud whilte noise during sleep In-Reply-To: <00d201cd3447$f3ecb0b0$dbc61210$@com> References: <00d201cd3447$f3ecb0b0$dbc61210$@com> Message-ID: <62E9B82F-6236-48F7-B83E-05069FC66BA8@visionutveckling.se> If i remember correctly, the default divisor used when setting send_silence_when_idle=true is 1400. It sounds to me that you are using a a very old version or something? I've never had any problems with loud white noice, and I use send_silence quite alot. /Peter 17 maj 2012 kl. 18:18 skrev "Phil Quesinberry" >: ** Disclaimer: this is probably a really bad idea ** Just in case this is frustrating anyone else: Silence is generated by the switch_generate_sln_silence function(s) in switch_resample.c, 'divisor' is the variable which determines the noise volume - the higher the value, the lower the noise volume. So, a quick fix for this problem would be to hard-code a value like 1400 for 'divisor' within those functions, by adding divisor = 1400 before the for loop and recompiling. This will affect ALL silence generated by FreeSwitch so it would be better to find where this is set within the sleep function so it only affects noise generated there. Better yet, if there's a way to accomplish this without hacking the code, (i.e. a variable that can be set), then by all means do that instead (and let me know what it is). Setting send_silence_when_idle to the desired value curbs the noise during sleep but it sends silence during ringback too, at least in my configuration here. Regards, Phil Quesinberry Q Systems Engineering, Inc. Electronic Controls and Embedded Systems Development (410) 969-8002 http://www.qsystemsengineering.com !DSPAM:4fb521ba32762069614621! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4fb521ba32762069614621! From yehavi.bourvine at gmail.com Thu May 17 20:52:43 2012 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Thu, 17 May 2012 19:52:43 +0300 Subject: [Freeswitch-users] Polycom auto answer on active call In-Reply-To: References: <6987271E-B52C-4D9C-8B2E-D37808D9B6F9@5ninesolutions.com> Message-ID: You can call mod_limit by yourself or at the dialstring. Mod_limit counts all calls to/from an extension, so it is quite accurate. For example, we use: Which either returns "user busy" or calls the extension. __Yehavi: 2012/5/17 Spencer Thomason > Thanks that should work. How did you keep track of the calls? Calls from > an extension are easy enough but an extension can be called many ways, i.e. > hunts groups, transfers, etc that would need to be accounted using limit. > Could you do this by modifying the dial-string? > > Spencer > > On May 17, 2012, at 9:09 AM, Yehavi Bourvine wrote: > > Use the mod_limit to check whether there is already a call on his > extension, and if so - do not try to bridge with auto answer. > > __Yehavi: > > 2012/5/17 Spencer Thomason > >> Hello, I'm having a problem with Polycom endpoints where the phone auto >> answers a page even if another call is in progress. My dialplan looks like >> this: >> >> >> >> >> >> >> >> >> >> Does anyone have any suggestions for an elegant way to solve this? >> >> Thanks, >> Spencer >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120517/a8f17dc5/attachment-0001.html From daniel at pocock.com.au Thu May 17 20:52:06 2012 From: daniel at pocock.com.au (Daniel Pocock) Date: Thu, 17 May 2012 16:52:06 +0000 Subject: [Freeswitch-users] dlz-ldap-enum - expose LDAP data to FreeSWITCH via ENUM Message-ID: <4FB52CB6.4050204@pocock.com.au> I've recently released a dlz ENUM module for the bind9 nameserver: http://www.opentelecoms.org/dlz-ldap-enum Basically, it handles ENUM queries from FreeSWITCH, Kamailio, Asterisk, repro, Lumicall, searches for the phone number in LDAP, and if found, returns the email address as both a SIP address and Jabber address This should make it even easier than ever before to get federated VoIP up and running using email addresses interchangeably with phone numbers. If the data already exists in LDAP as an address book, then just install bind9, install the module and you're up and running. Regards, Daniel From msc at freeswitch.org Thu May 17 21:40:59 2012 From: msc at freeswitch.org (Michael Collins) Date: Thu, 17 May 2012 10:40:59 -0700 Subject: [Freeswitch-users] mod_say_es fail when tray pronounce voicemails messages dates In-Reply-To: References: Message-ID: Go ahead and update to latest git and make sure the problem still exists. (It probably will. :) The go to jira.freeswitch.org and open a ticket. I know that English is not your first language but that's okay, just do your best. If you really need to express something and are not sure how to do it in English then do it in Spanish and we'll use Google translate to help out. Thanks, MC On Thu, May 17, 2012 at 9:21 AM, RrodolfoS . wrote: > mod_say_es fail when tray pronounce voicemails messages dates, in > english work well but in espanish fail. > > In spanish the ivr and voicemail intro work fine, except the > voicemails messages. > > The logs is the same of this http://www.freeswitch.es/node/14755 > > Configs details: > > Version > FreeSWITCH Version 1.1.beta1 (git-4283408 2012-04-29 11-33-24 -0400) > > /usr/src/freeswitch/modules.conf > say/mod_say_es > > # make mod_say_es-install > > /usr/local/freeswitch/conf/freeswitch.xml >
> > > > > > >
> > /usr/local/freeswitch/conf/vars.xml > > > /usr/local/freeswitch/conf/lang/es/es.xml > > sound-prefix="$${sounds_dir}/es/mx/maria" tts-engine="cepstral" > tts-voice="maria"> > > > > > > > > > > > > > > /usr/local/freeswitch/conf/autoload_configs/modules.conf.xml > > > > > > > The spanish sounds are in /usr/local/freeswitch/sounds/es/mx/maria/ > with 8000, 16000, 32000, 44100 and 48000 sample rate mono. > > /usr/local/freeswitch/conf/lang/es are a modified copy of > /usr/local/freeswitch/conf/lang/en > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120517/9944e8a2/attachment.html From philq at qsystemsengineering.com Thu May 17 21:45:47 2012 From: philq at qsystemsengineering.com (Phil Quesinberry) Date: Thu, 17 May 2012 13:45:47 -0400 Subject: [Freeswitch-users] Loud whilte noise during sleep Message-ID: <016e01cd3454$e73d64b0$b5b82e10$@com> Actually it's one of the latest 1.2.0 revisions - b653c21. I didn't realize that with the quick look I had at the code. Curious. that makes me wonder if the Aastra phones are generating the noise when they don't see any media. A user complained about hearing the noise when first logging into voicemail, before she is asked for her PIN. I hadn't noticed it before because I normally do that on speaker but on the handset, it is quite noticeable. Is there a way to use send_silence_when_idle without having it kill ringback when set to a non-zero value? ---------- Anthony Minessale Thu May 17 20:21:35 MSD 2012 Which revision of the code are you looking at? The sleep sends dummy frames unless you set that send_silence_when_idle var, if its not set its probably not sending anything. You can see in the code that unless that var is set it wont be generating anything. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120517/f276a8e4/attachment.html From spencer at 5ninesolutions.com Thu May 17 22:15:06 2012 From: spencer at 5ninesolutions.com (Spencer Thomason) Date: Thu, 17 May 2012 11:15:06 -0700 Subject: [Freeswitch-users] Polycom auto answer on active call In-Reply-To: References: <6987271E-B52C-4D9C-8B2E-D37808D9B6F9@5ninesolutions.com> Message-ID: <5F355E02-B837-4326-8675-E90929221653@5ninesolutions.com> The problem is that I have now way of incrementing the number of calls that limit is keeping track of from an API command. I believe that I need to increment the call count from the directory dial-string because there are several places an extension could be called with a bridge directly to the user (i.e. bridge/user@${domain_name}) as apposed to a transfer. If a call is transferred then I can call the dialplan application limit before the bridge to the user and everything works great. Is there a way I can call a dialplan application from the directory dial-string? or is there any other way to make sure that limit is called on every bridge to a registered user? Thanks again, Spencer On May 17, 2012, at 9:52 AM, Yehavi Bourvine wrote: > You can call mod_limit by yourself or at the dialstring. Mod_limit counts all calls to/from an extension, so it is quite accurate. > > For example, we use: > > > Which either returns "user busy" or calls the extension. > > __Yehavi: > > 2012/5/17 Spencer Thomason > Thanks that should work. How did you keep track of the calls? Calls from an extension are easy enough but an extension can be called many ways, i.e. hunts groups, transfers, etc that would need to be accounted using limit. Could you do this by modifying the dial-string? > > Spencer > > On May 17, 2012, at 9:09 AM, Yehavi Bourvine wrote: > >> Use the mod_limit to check whether there is already a call on his extension, and if so - do not try to bridge with auto answer. >> >> __Yehavi: >> >> 2012/5/17 Spencer Thomason >> Hello, I'm having a problem with Polycom endpoints where the phone auto answers a page even if another call is in progress. My dialplan looks like this: >> >> >> >> >> >> >> >> >> >> Does anyone have any suggestions for an elegant way to solve this? >> >> Thanks, >> Spencer >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120517/1312856b/attachment-0001.html From tarik.bts.gi at gmail.com Thu May 17 23:08:06 2012 From: tarik.bts.gi at gmail.com (ghallab) Date: Thu, 17 May 2012 19:08:06 +0000 Subject: [Freeswitch-users] transfer caller ID problem and plycom Message-ID: <4FB54C96.4010702@gmail.com> When their Polycom 550 rings the user see the caller ID of the person who transferred the call, once they answer the phone the caller ID of the transferred caller is displayed. I already tried the ignore display updates set to faulse and true in the vars.xml file. But don't have effect! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120517/a15ed61a/attachment.html From msc at freeswitch.org Thu May 17 22:39:20 2012 From: msc at freeswitch.org (Michael Collins) Date: Thu, 17 May 2012 11:39:20 -0700 Subject: [Freeswitch-users] bridge a call to registered user In-Reply-To: <4FB49CD7.3040509@softnet.si> References: <4FB39A66.5070809@softnet.si> <4FB49CD7.3040509@softnet.si> Message-ID: I'll have to defer to those with more experience than I. SIP presence, multitenancy, etc. is outside my area of expertise... -MC On Wed, May 16, 2012 at 11:38 PM, Miha wrote: > HI @Michael, > > I tried to uncomment this due to Multi-tenant configuration. As this did > not helped me I tried to use Presence variable ( > http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#Presence). > After this FS does not log any more as user not registered. > > Could this function be causing any other problems? > > THanks! > > Miha > > > > On 5/17/2012 12:17 AM, Michael Collins wrote: > > This is probably because the two different profiles are not set to use the > same register domain. Check out this wiki page and try setting the value to > be the same for all your profiles: > http://wiki.freeswitch.org/wiki/Sofia.conf.xml#force-register-domain > > -MC > > On Wed, May 16, 2012 at 5:15 AM, Miha wrote: > >> Hi, >> >> just one question. I have created two profiles (suggested by @Michael) >> as I was having same NAT issue. Profiles are basically the same only >> registration port is different (5060, 5070) and NAT profile have set: >> . >> >> In my dialplan I have made bridge like this: >> >> >> > >> data="{origination_callee_id_name='${effective_caller_id_name}',origination_calee_id_number='${effective_caller_id_name}'}user/${ >> destination_number}.fs_test at fs_test.fs2.blabla.com" >> /> >> >> freeswitch at default> show re >> >> [ registrations] >> >> >> freeswitch at default> show registrations >> >> reg_user,realm,token,url,expires,network_ip,network_port,network_proto,hostname >> 018108753.fs_test,fs_test.fs2.blabla.com,40804176-fb1f176 at 172.31.1.190, >> sofia/internal/sip:018108753.fs_test at xxx.xxx.xxx.xxx:1265,1337169317,xxx.xxx.xxx.xxx,1265,udp,localhost.localdomain >> 018108752.fs_test,fs_test.fs2.blabla.comi,873cfe40-1224f25 at 172.31.1.171, >> sofia/nat/sip:018108752.fs_test at xxx.xxx.xxx.xxx:2097,1337170569,xxx.xxx.xxx.xxx,1920,udp,localhost.localdomain >> >> 2 total. >> >> Why only phone work which is registered on NAT profile. IF I call users >> which is registered on internal profile I get user not registered? >> >> Thanks! >> >> Miha >> >> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120517/a8457e3a/attachment.html From msc at freeswitch.org Thu May 17 22:42:16 2012 From: msc at freeswitch.org (Michael Collins) Date: Thu, 17 May 2012 11:42:16 -0700 Subject: [Freeswitch-users] Detect answer on FXO device In-Reply-To: References: Message-ID: Not reliably. The VAD might detect "We're sorry, you have reached a number that is disconnected..." and consider that an "answer." You might possibly be able to use the spandsp tone detect stuff to listen for ringback and then act accordingly if you no longer hear ringback, but that's a scary scenario. You are better off figuring out why the ATA isn't working in the first place. -MC On Thu, May 17, 2012 at 6:23 AM, cristian re wrote: > The ATA has both "call answer supervision" and "polarity reversal" but > they doesn't works worldwide but only with some carriers. > Is there a way to delegate this job to Freeswitch? using voice activity > detection? > > Cristian > > > >Technically, answer detection is the job of the ATA. The problem is that > it > >is an inexact science. I would look in the GXW docs to see if they have a > >section on 'answer supervision'. It might also be in a section along with > >'disconnect supervision.' > > > >-MC > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120517/661f8050/attachment.html From spencer at 5ninesolutions.com Thu May 17 22:47:03 2012 From: spencer at 5ninesolutions.com (Spencer Thomason) Date: Thu, 17 May 2012 11:47:03 -0700 Subject: [Freeswitch-users] mod_voicemail vmain from dialplan Message-ID: <8C7BCC25-8321-4F72-B0A9-2D3E00DEE593@5ninesolutions.com> Hello all, I'm working on a multi tenant setup where on one of the domains I need to have a different action for the vmain key in mod_voicemail. I cannot modify the vmain extension in the default context as this would affect all domains. Is it possible to set the vmain extension from the dialplan or is there another way to handle this? Essentially I need to get the caller the option of pressing a key to be connected to after hours support when they get voicemail. Thanks, Spencer From yehavi.bourvine at gmail.com Thu May 17 22:49:47 2012 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Thu, 17 May 2012 21:49:47 +0300 Subject: [Freeswitch-users] Polycom auto answer on active call In-Reply-To: <5F355E02-B837-4326-8675-E90929221653@5ninesolutions.com> References: <6987271E-B52C-4D9C-8B2E-D37808D9B6F9@5ninesolutions.com> <5F355E02-B837-4326-8675-E90929221653@5ninesolutions.com> Message-ID: Sorry, I did not understand what you exactly mean. Can you give some example? About the bridge application - you have to check it in each bridge call, or do one centralized check before you enter the logic of the dialplan: If the user is above limit - make it busy; if not, continue with the dialplan as usuall. __Yehavi: 2012/5/17 Spencer Thomason > The problem is that I have now way of incrementing the number of calls > that limit is keeping track of from an API command. I believe that I need > to increment the call count from the directory dial-string because there > are several places an extension could be called with a bridge directly to > the user (i.e. bridge/user@${domain_name}) as apposed to a transfer. If > a call is transferred then I can call the dialplan application limit before > the bridge to the user and everything works great. Is there a way I can > call a dialplan application from the directory dial-string? or is there > any other way to make sure that limit is called on every bridge to a > registered user? > > Thanks again, > Spencer > > > On May 17, 2012, at 9:52 AM, Yehavi Bourvine wrote: > > You can call mod_limit by yourself or at the dialstring. Mod_limit counts > all calls to/from an extension, so it is quite accurate. > > For example, we use: > > > Which either returns "user busy" or calls the extension. > > __Yehavi: > > 2012/5/17 Spencer Thomason > >> Thanks that should work. How did you keep track of the calls? Calls >> from an extension are easy enough but an extension can be called many ways, >> i.e. hunts groups, transfers, etc that would need to be accounted using >> limit. Could you do this by modifying the dial-string? >> >> Spencer >> >> On May 17, 2012, at 9:09 AM, Yehavi Bourvine wrote: >> >> Use the mod_limit to check whether there is already a call on his >> extension, and if so - do not try to bridge with auto answer. >> >> __Yehavi: >> >> 2012/5/17 Spencer Thomason >> >>> Hello, I'm having a problem with Polycom endpoints where the phone auto >>> answers a page even if another call is in progress. My dialplan looks like >>> this: >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> Does anyone have any suggestions for an elegant way to solve this? >>> >>> Thanks, >>> Spencer >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> Join Us At ClueCon - Aug 7-9, 2012 >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120517/0f9c6fe2/attachment-0001.html From paul at cupis.co.uk Thu May 17 23:01:35 2012 From: paul at cupis.co.uk (Paul Cupis) Date: Thu, 17 May 2012 20:01:35 +0100 Subject: [Freeswitch-users] Help on Explaination of the configurations parrametters In-Reply-To: References: Message-ID: <4FB54B0F.3050000@cupis.co.uk> On 17/05/12 02:50, Albert Nguyen wrote: > I am new to FS and trying to setup a SBC using the example 2 in the FS > wiki website. The link is > http://wiki.freeswitch.org/wiki/SBC_FreeSWITCH_Configuration_Example_2. > > There are parameters in the example that I have to replaces for it to > work with my own scenarios. However I have problem understanding the > following lines. Is there anyone able to explain what does this mean > > expression="(((192.168.)|(172.24.)|(10.10.))\d+\.\d+(:\d+)(;dtg=\w+)?)"> > > The bits I am not sure is )\d+\.\d+(:\d+)(;dtg=\w+)?). What does this do? This will match certain (IPv4) IP addresses, with ports specified: 192.168.x.x:y 172.24.x.x:y 10.10.x.x:y optionally ending in: ;dtg=xxxx In the given example, the rest of the XML is executed conditionally on this match. If the sip_redirect_contact_0 matches the expression, then the are executed, otherwise the is executed. Regards, From spencer at 5ninesolutions.com Thu May 17 23:03:53 2012 From: spencer at 5ninesolutions.com (Spencer Thomason) Date: Thu, 17 May 2012 12:03:53 -0700 Subject: [Freeswitch-users] Polycom auto answer on active call In-Reply-To: References: <6987271E-B52C-4D9C-8B2E-D37808D9B6F9@5ninesolutions.com> <5F355E02-B837-4326-8675-E90929221653@5ninesolutions.com> Message-ID: Yes, The main problem I'm have trouble with is tracking calls from a hunt group. I use a lua script to call multiple users at once. For example: session:execute("bridge", "[leg_timeout=30]user/1001@"..domain_name..",".."[leg_timeout=30]user/1002@"..domain_name..",".."[leg_timeout=30]user/1003@"..domain_name); It is impossible to know which user will be on the phone before the lua script is called. I can execute an api application from the directory dial-string but I am unaware of a way to execute another dialplan application from there. Since there is no transfer, I cannot do something like: before I hope that helps explain my problem :-) Thanks, Spencer On May 17, 2012, at 11:49 AM, Yehavi Bourvine wrote: > Sorry, I did not understand what you exactly mean. Can you give some example? > > About the bridge application - you have to check it in each bridge call, or do one centralized check before you enter the logic of the dialplan: If the user is above limit - make it busy; if not, continue with the dialplan as usuall. > > __Yehavi: > > 2012/5/17 Spencer Thomason > The problem is that I have now way of incrementing the number of calls that limit is keeping track of from an API command. I believe that I need to increment the call count from the directory dial-string because there are several places an extension could be called with a bridge directly to the user (i.e. bridge/user@${domain_name}) as apposed to a transfer. If a call is transferred then I can call the dialplan application limit before the bridge to the user and everything works great. Is there a way I can call a dialplan application from the directory dial-string? or is there any other way to make sure that limit is called on every bridge to a registered user? > > Thanks again, > Spencer > > > On May 17, 2012, at 9:52 AM, Yehavi Bourvine wrote: > >> You can call mod_limit by yourself or at the dialstring. Mod_limit counts all calls to/from an extension, so it is quite accurate. >> >> For example, we use: >> >> >> Which either returns "user busy" or calls the extension. >> >> __Yehavi: >> >> 2012/5/17 Spencer Thomason >> Thanks that should work. How did you keep track of the calls? Calls from an extension are easy enough but an extension can be called many ways, i.e. hunts groups, transfers, etc that would need to be accounted using limit. Could you do this by modifying the dial-string? >> >> Spencer >> >> On May 17, 2012, at 9:09 AM, Yehavi Bourvine wrote: >> >>> Use the mod_limit to check whether there is already a call on his extension, and if so - do not try to bridge with auto answer. >>> >>> __Yehavi: >>> >>> 2012/5/17 Spencer Thomason >>> Hello, I'm having a problem with Polycom endpoints where the phone auto answers a page even if another call is in progress. My dialplan looks like this: >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> Does anyone have any suggestions for an elegant way to solve this? >>> >>> Thanks, >>> Spencer >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> Join Us At ClueCon - Aug 7-9, 2012 >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> Join Us At ClueCon - Aug 7-9, 2012 >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120517/220025ea/attachment.html From acrow at integrafin.co.uk Thu May 17 23:38:11 2012 From: acrow at integrafin.co.uk (Alex Crow) Date: Thu, 17 May 2012 20:38:11 +0100 Subject: [Freeswitch-users] Spandsp mulated modems inbound/no /dev/FS* device? In-Reply-To: References: <4FA24100.7040908@integrafin.co.uk> <20120503151946.GA824@eagle.cupis.co.uk> <4FA3EE97.1090908@integrafin.co.uk> <4FA596BF.4090703@integrafin.co.uk> <4FA635AE.5050006@integrafin.co.uk> <4FA66862.1090300@coppice.org> <4FA6871F.5050909@puzzled.xs4all.nl> <4FA696B1.8050706@coppice.org> Message-ID: <4FB553A3.5010807@integrafin.co.uk> On 17/05/12 14:51, Anita Hall wrote: > Hi Alex > > Did you get the answer to your first question ? Did you succeed in > using the emulated modem option for taking fax calls to hylafax ? > > I put the following in spandsp.conf.xml and got /dev/FS[0-4] devices > as soft links to /dev/pts/[4-8]. My freeswitch is running as root user > so I did not face the issue you did. > > > > > > What next? Could you point me to some doc in hylafax? > > (And before Steve lashes out at me again, I must clarify, I do not > want to just play around, but my boss is clueless and wants me to > evaluate hylafax :() > > regards, > Anita > > Anita, The answer I got was that inbound is not yet supported by that mechanism, so I'd advise trying the Hylafax/T38Modem option if you need to use Hylafax. The T38Modem site provides tarballs of the OPAL and required dependencies to build it. The one main advantage we see for it is that there are quite a few free and commercial clients for Hylafax that can behave as a printer in Windows, so the user selects to print say a .doc file, chooses the fax printer, enters a number and off it goes. We used WHFC with XP, works great. We even have chaps batch faxing 100+ faxes a day using it from Excel with a macro. The other route for inbound is well documented, it using the inbuilt FreeSWITCH faxing and sending to email. Outbound is equally simple but not IMHO as easy for our end users as Hylafax (email-to-fax is simple but printing to fax from Windows not so simple). But with some work on your side, the spandsp stuff could probably do anything HF can do. As I said, Hylafax is excellent software and has been deployed on major production sites for probably 15 years+ with billions of total faxes delivered. We've used it for more than ten years with about 400 faxes in/out per day (which is piddling compared with some other sites, 10,000 per day is not difficult). If you want to avoid using t38modem you'd probably have to set up a dedicated HF box with an ISDN/POTS multichannel card (and if you want to fax from/to FS use a BRI or T1/E1 card to send audio to and from it). There are other ways of integrating it with backoffice systems including FTP, email, etc. To try to deploy a business fax system without first evaluating HylaFAX is in my opinion insane, so maybe your boss has a point. If, however, your boss is the one driving you to ask all these very diverse questions on this list (and they are indeed frequent and often completely unrelated to each other), perhaps you should ask him/her to focus on those that are his top priorities, and pick the first 3 to 5 and focus on those first, or in fact just compose a list of required features, post them up in a single email rather than 50, and ask "is this possible?". Just being a bit more targeted with your questions and how they link together might elicit more responses. Try to pick one solution at a time, test it, ask questions on that and stick with it for a bit and I think you'll get more help. Cheers Alex -- This message is intended only for the addressee and may contain confidential information. Unless you are that person, you may not disclose its contents or use it in any way and are requested to delete the message along with any attachments and notify us immediately. "Transact" is operated by Integrated Financial Arrangements plc Domain House, 5-7 Singer Street, London EC2A 4BQ Tel: (020) 7608 4900 Fax: (020) 7608 5300 (Registered office: as above; Registered in England and Wales under number: 3727592) Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856) From philq at qsystemsengineering.com Thu May 17 23:42:55 2012 From: philq at qsystemsengineering.com (Phil Quesinberry) Date: Thu, 17 May 2012 15:42:55 -0400 Subject: [Freeswitch-users] Bypass media succeeds from extension to gateway but fails from extension to extension Message-ID: <020801cd3465$43e08020$cba18060$@com> Nevermind! As it turns out, FS does send the correct external/NAT IP addresses as long as BOTH extensions are set for bypass media. Forgive me for ever doubting... Of course it still doesn't work for reasons discussed earlier, although I think that it should work as long as there's no symmetric NAT involved. I'm guessing session border controller(s) will be the next step if we want to remove the endpoint to endpoint traffic from our network. Any suggestions? OpenSIPS certainly looks promising. Thanks, - Phil -----Original Message----- From: Phil Quesinberry Sent: Friday, May 11, 2012 2:41 PM To: 'freeswitch-users at lists.freeswitch.org' Subject: Bypass media succeeds from extension to gateway but fails from extension to extension Ken, Thanks for taking the time to write that out. I understand that and that makes perfect sense, although the endpoints in this case are configured to report their external IP addresses, either through STUN or a static NAT IP entry. What still is unclear to me is why this failed when FS was able to successfully bridge the same extension directly to the PSTN gateway. ... >From the traffic I pastebinned, we can see that Endpoint A is sending its external WAN address info to FS, right before FS sends Endpoint A?s internal LAN address to Endpoint B, so I would think that FS should be passing Endpoint A?s WAN address along instead of its LAN address for media: ... ---------- Ken Rice Thu May 10 19:08:53 MSD 2012 The contact IP has nothing to do with where the media goes... That?s entirely defined in the SDP... Consider this Endpoint A (192.168.100.100) -> NAT A -> FreeSWITCH (4.2.2.2) -> NAT B -> Endpoint B (192.168.100.200) Now lets assume that NAT A and NAT B are 2 separate nat gateways and that Endpoint A and Endpoint B are on 2 different physical LANs... Telling Endpoint A to talk directly to Endpoint B without proxying media will never work since the endpoints think they are on the same LAN. There is no mechanism there to allow for the redirection and automagic adjustments of ports etc so that they can talk directly to each other... Now lets change this slightly so that endpoint B is 192.168.200.200. Unless NAT A knows how to get to 192.168.200.0/24 (assuming class C sized block) and NAT B knows how to get to 192.168.100.0/24 they are both going to use their default routing which is to NAT the outgoing RTP, and forward it to the next hop... Again, RTP will not make it to other side in either direction... FreeSWITCH cant compensate due to a number of factors... Your Endpoints have to be smart enough to actually compensate for the NAT in this situation OR your NAT boxes have to compensate for it... The simple answer, don?t use bypass media in this situation, the complex answer I wont get into here... Stop by IRC and ask around... There is a 3rd option here you might want to consider, contact consulting at freeswitch.org for some professional help... This may not be specifially what you need to get going as I have no clue what your skill level happens to be, and you did say you are still learning. Good Luck! From spencer at 5ninesolutions.com Fri May 18 00:04:02 2012 From: spencer at 5ninesolutions.com (Spencer Thomason) Date: Thu, 17 May 2012 13:04:02 -0700 Subject: [Freeswitch-users] Polycom auto answer on active call In-Reply-To: References: <6987271E-B52C-4D9C-8B2E-D37808D9B6F9@5ninesolutions.com> <5F355E02-B837-4326-8675-E90929221653@5ninesolutions.com> Message-ID: <6641633A-039B-45C7-B571-5668CD7611BE@5ninesolutions.com> Hello, You can use the following to track all calls placed to a registered user: I use the following dial-string: I will update the wiki. Thanks, Spencer On May 17, 2012, at 12:03 PM, Spencer Thomason wrote: > Yes, > The main problem I'm have trouble with is tracking calls from a hunt group. I use a lua script to call multiple users at once. For example: > session:execute("bridge", "[leg_timeout=30]user/1001@"..domain_name..",".."[leg_timeout=30]user/1002@"..domain_name..",".."[leg_timeout=30]user/1003@"..domain_name); > > It is impossible to know which user will be on the phone before the lua script is called. I can execute an api application from the directory dial-string but I am unaware of a way to execute another dialplan application from there. Since there is no transfer, I cannot do something like: > > before > > > I hope that helps explain my problem :-) > > Thanks, > Spencer > > > On May 17, 2012, at 11:49 AM, Yehavi Bourvine wrote: > >> Sorry, I did not understand what you exactly mean. Can you give some example? >> >> About the bridge application - you have to check it in each bridge call, or do one centralized check before you enter the logic of the dialplan: If the user is above limit - make it busy; if not, continue with the dialplan as usuall. >> >> __Yehavi: >> >> 2012/5/17 Spencer Thomason >> The problem is that I have now way of incrementing the number of calls that limit is keeping track of from an API command. I believe that I need to increment the call count from the directory dial-string because there are several places an extension could be called with a bridge directly to the user (i.e. bridge/user@${domain_name}) as apposed to a transfer. If a call is transferred then I can call the dialplan application limit before the bridge to the user and everything works great. Is there a way I can call a dialplan application from the directory dial-string? or is there any other way to make sure that limit is called on every bridge to a registered user? >> >> Thanks again, >> Spencer >> >> >> On May 17, 2012, at 9:52 AM, Yehavi Bourvine wrote: >> >>> You can call mod_limit by yourself or at the dialstring. Mod_limit counts all calls to/from an extension, so it is quite accurate. >>> >>> For example, we use: >>> >>> >>> Which either returns "user busy" or calls the extension. >>> >>> __Yehavi: >>> >>> 2012/5/17 Spencer Thomason >>> Thanks that should work. How did you keep track of the calls? Calls from an extension are easy enough but an extension can be called many ways, i.e. hunts groups, transfers, etc that would need to be accounted using limit. Could you do this by modifying the dial-string? >>> >>> Spencer >>> >>> On May 17, 2012, at 9:09 AM, Yehavi Bourvine wrote: >>> >>>> Use the mod_limit to check whether there is already a call on his extension, and if so - do not try to bridge with auto answer. >>>> >>>> __Yehavi: >>>> >>>> 2012/5/17 Spencer Thomason >>>> Hello, I'm having a problem with Polycom endpoints where the phone auto answers a page even if another call is in progress. My dialplan looks like this: >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> Does anyone have any suggestions for an elegant way to solve this? >>>> >>>> Thanks, >>>> Spencer >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> Join Us At ClueCon - Aug 7-9, 2012 >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at 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 at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> Join Us At ClueCon - Aug 7-9, 2012 >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at 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 at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> Join Us At ClueCon - Aug 7-9, 2012 >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> Join Us At ClueCon - Aug 7-9, 2012 >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120517/4e931798/attachment-0001.html From nickolayr at gmail.com Fri May 18 00:08:27 2012 From: nickolayr at gmail.com (Nikolay Rogoshchenkov) Date: Thu, 17 May 2012 16:08:27 -0400 Subject: [Freeswitch-users] Play wav before bridge in IVR, possible? Message-ID: Hello, This is part of IVR config, and I would like to play announcement before bridge to 123 at 12.3.4.56, is it possible to do? =============================================== [...] . =============================================== Thank you. -- Nikolay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120517/a36f4eb1/attachment.html From wstephen80 at gmail.com Fri May 18 00:34:57 2012 From: wstephen80 at gmail.com (Stephen Wilde) Date: Thu, 17 May 2012 22:34:57 +0200 Subject: [Freeswitch-users] One core at 100% usage Message-ID: I have done a git pull (now git-83e090c 2012-05-17 16-40-30 +0000) and I have a strange load distribution over cores: one core at 100%. Here the htop: 1 [|||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||100.0%] 2 [||||||||||||||||| 19.1%] 3 [|| 1.3%] 4 [|| 1.3%] 5 [ 0.0%] 6 [ 0.0%] 7 [|||| 3.9%] 8 [||||||||||||| 13.2%] 9 [ 0.0%] 10 [||||||||||||||||| 18.3%] 11 [||||| 4.6%] 12 [|||||||||||||||| 17.0%] 13 [|||||||| 8.6%] 14 [|||||||||||| 12.5%] 15 [|||||||| 8.6%] 16 [||||||||||||||||||| 21.1%] After revert to an old git I have this htop: 1 [||||||| 7.2%] 2 [|||||||||||| 11.8%] 3 [||||| 5.3%] 4 [||||||| 6.6%] 5 [|||| 3.3%] 6 [||||||||| 9.0%] 7 [||||| 4.6%] 8 [|||||||||||||| 15.1%] 9 [||||||| 6.5%] 10 [||||||||| 9.3%] 11 [|||||| 6.5%] 12 [|||||||||||||| 14.9%] 13 [|||||||| 7.8%] 14 [|||||||||||||||| 17.0%] 15 [|||||||| 7.3%] 16 [||||||||||| 11.8%] anyone with same issue? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120517/b9d2745d/attachment.html From msc at freeswitch.org Fri May 18 00:44:33 2012 From: msc at freeswitch.org (Michael Collins) Date: Thu, 17 May 2012 13:44:33 -0700 Subject: [Freeswitch-users] Play wav before bridge in IVR, possible? In-Reply-To: References: Message-ID: You could transfer to a new extension and have that extension execute the playback and then do the bridge. That is simple, clean, and extensible. -MC On Thu, May 17, 2012 at 1:08 PM, Nikolay Rogoshchenkov wrote: > Hello, > > This is part of IVR config, and I would like to play announcement before > bridge to 123 at 12.3.4.56, is it possible to do? > > =============================================== > [...] > . > > > > > > > =============================================== > > Thank you. > -- > Nikolay > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120517/0ff48a8a/attachment.html From nickolayr at gmail.com Fri May 18 00:48:29 2012 From: nickolayr at gmail.com (Nikolay Rogoshchenkov) Date: Thu, 17 May 2012 16:48:29 -0400 Subject: [Freeswitch-users] Play wav before bridge in IVR, possible? In-Reply-To: References: Message-ID: But if I would like to bridge/transfer to external number via gateway? may be I can you something like this? [...] [...] -- Nikolay On Thu, May 17, 2012 at 4:44 PM, Michael Collins wrote: > You could transfer to a new extension and have that extension execute the > playback and then do the bridge. That is simple, clean, and extensible. > > -MC > > > On Thu, May 17, 2012 at 1:08 PM, Nikolay Rogoshchenkov < > nickolayr at gmail.com> wrote: > >> Hello, >> >> This is part of IVR config, and I would like to play announcement before >> bridge to 123 at 12.3.4.56, is it possible to do? >> >> =============================================== >> [...] >> . >> >> >> >> >> >> >> =============================================== >> >> Thank you. >> -- >> Nikolay >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120517/6f2cc9c0/attachment-0001.html From nickolayr at gmail.com Fri May 18 00:57:12 2012 From: nickolayr at gmail.com (Nikolay Rogoshchenkov) Date: Thu, 17 May 2012 16:57:12 -0400 Subject: [Freeswitch-users] Play wav before bridge in IVR, possible? In-Reply-To: References: Message-ID: Looks like it works with that config: [...] -- Nikolay On Thu, May 17, 2012 at 4:48 PM, Nikolay Rogoshchenkov wrote: > But if I would like to bridge/transfer to external number via gateway? > > may be I can you something like this? > [...] > > > [...] > -- > Nikolay > > > > On Thu, May 17, 2012 at 4:44 PM, Michael Collins wrote: > >> You could transfer to a new extension and have that extension execute the >> playback and then do the bridge. That is simple, clean, and extensible. >> >> -MC >> >> >> On Thu, May 17, 2012 at 1:08 PM, Nikolay Rogoshchenkov < >> nickolayr at gmail.com> wrote: >> >>> Hello, >>> >>> This is part of IVR config, and I would like to play announcement before >>> bridge to 123 at 12.3.4.56, is it possible to do? >>> >>> =============================================== >>> [...] >>> . >>> >>> >>> >>> >>> >>> >>> =============================================== >>> >>> Thank you. >>> -- >>> Nikolay >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> Join Us At ClueCon - Aug 7-9, 2012 >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120517/16afcbb1/attachment.html From msc at freeswitch.org Fri May 18 01:03:59 2012 From: msc at freeswitch.org (Michael Collins) Date: Thu, 17 May 2012 14:03:59 -0700 Subject: [Freeswitch-users] Play wav before bridge in IVR, possible? In-Reply-To: References: Message-ID: Ah very interesting. I have never tried that. I will lab it up and confirm, and if it really does work like this then I will update the wiki and make a note in chapter 6 of my copy of the FS book. (If we ever do a 2e of the bridge book I'lll have a stack of new items to put into it. My copy his marked up all over the place. :) Thanks Nikolay, MC On Thu, May 17, 2012 at 1:57 PM, Nikolay Rogoshchenkov wrote: > Looks like it works with that config: > > [...] > > > > param="transfer $1 XML someserver"/> > > > > > -- > Nikolay > > > > On Thu, May 17, 2012 at 4:48 PM, Nikolay Rogoshchenkov < > nickolayr at gmail.com> wrote: > >> But if I would like to bridge/transfer to external number via gateway? >> >> may be I can you something like this? >> [...] >> >> >> [...] >> -- >> Nikolay >> >> >> >> On Thu, May 17, 2012 at 4:44 PM, Michael Collins wrote: >> >>> You could transfer to a new extension and have that extension execute >>> the playback and then do the bridge. That is simple, clean, and extensible. >>> >>> -MC >>> >>> >>> On Thu, May 17, 2012 at 1:08 PM, Nikolay Rogoshchenkov < >>> nickolayr at gmail.com> wrote: >>> >>>> Hello, >>>> >>>> This is part of IVR config, and I would like to play announcement >>>> before bridge to 123 at 12.3.4.56, is it possible to do? >>>> >>>> =============================================== >>>> [...] >>>> . >>>> >>>> >>>> >>>> >>>> >>>> >>>> =============================================== >>>> >>>> Thank you. >>>> -- >>>> Nikolay >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> Join Us At ClueCon - Aug 7-9, 2012 >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at 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 at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> Join Us At ClueCon - Aug 7-9, 2012 >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120517/73fa365a/attachment.html From msc at freeswitch.org Fri May 18 01:29:03 2012 From: msc at freeswitch.org (Michael Collins) Date: Thu, 17 May 2012 14:29:03 -0700 Subject: [Freeswitch-users] mod_voicemail vmain from dialplan In-Reply-To: <8C7BCC25-8321-4F72-B0A9-2D3E00DEE593@5ninesolutions.com> References: <8C7BCC25-8321-4F72-B0A9-2D3E00DEE593@5ninesolutions.com> Message-ID: You could create a separate voicemail profile. Look in autoload_configs/voicemail.conf, specifically at the section. There is a "default" profile already that you are using. You can copy & paste it into a custom profile, edit the "vmain-extension" parameter, and then just use that profile when you call the voicemail app from the dialplan for the tenant in question -MC On Thu, May 17, 2012 at 11:47 AM, Spencer Thomason < spencer at 5ninesolutions.com> wrote: > Hello all, > I'm working on a multi tenant setup where on one of the domains I need to > have a different action for the vmain key in mod_voicemail. I cannot > modify the vmain extension in the default context as this would affect all > domains. Is it possible to set the vmain extension from the dialplan or is > there another way to handle this? Essentially I need to get the caller the > option of pressing a key to be connected to after hours support when they > get voicemail. > > Thanks, > Spencer > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120517/ca6af302/attachment-0001.html From albert_nguyen16 at hotmail.com Fri May 18 05:32:57 2012 From: albert_nguyen16 at hotmail.com (Albert Nguyen) Date: Fri, 18 May 2012 01:32:57 +0000 Subject: [Freeswitch-users] Help on Explaination of the configurations parrametters In-Reply-To: <4FB54B0F.3050000@cupis.co.uk> References: , <4FB54B0F.3050000@cupis.co.uk> Message-ID: Hi Paul, Thanks very much for your help on this. Now I know what to replace with this parrametter. Kind regards, Al > Date: Thu, 17 May 2012 20:01:35 +0100 > From: paul at cupis.co.uk > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Help on Explaination of the configurations parrametters > > On 17/05/12 02:50, Albert Nguyen wrote: > > I am new to FS and trying to setup a SBC using the example 2 in the FS > > wiki website. The link is > > http://wiki.freeswitch.org/wiki/SBC_FreeSWITCH_Configuration_Example_2. > > > > There are parameters in the example that I have to replaces for it to > > work with my own scenarios. However I have problem understanding the > > following lines. Is there anyone able to explain what does this mean > > > > > expression="(((192.168.)|(172.24.)|(10.10.))\d+\.\d+(:\d+)(;dtg=\w+)?)"> > > > > The bits I am not sure is )\d+\.\d+(:\d+)(;dtg=\w+)?). What does this do? > > This will match certain (IPv4) IP addresses, with ports specified: > > 192.168.x.x:y > 172.24.x.x:y > 10.10.x.x:y > > optionally ending in: > > ;dtg=xxxx > > In the given example, the rest of the XML is executed conditionally on > this match. If the sip_redirect_contact_0 matches the expression, then > the are executed, otherwise the is executed. > > Regards, > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120518/af339939/attachment.html From ahe.sanath at gmail.com Fri May 18 08:07:11 2012 From: ahe.sanath at gmail.com (Sanath Prasanna) Date: Fri, 18 May 2012 09:37:11 +0530 Subject: [Freeswitch-users] Run LUA script in different server In-Reply-To: References: Message-ID: Tx for advice MC & Anita. Can I do work around like this . Another freeswitch instant will be start in other server & calls will be transfer from operator connected freeswitch instance to this new freeswitch instance & vise versa. Pls advice. On Thu, May 17, 2012 at 5:05 PM, Anita Hall wrote: > You could run a Lua ESL server on a different machine but this will not be > the same as running a Lua script. > http://wiki.freeswitch.org/wiki/Event_Socket_Library > > regards, > Anita > > > > On Thu, May 17, 2012 at 4:37 AM, Michael Collins wrote: > >> I don't think you can directly do what you are describing. However, you >> might be able to use mod_httapi for this. There's some documentation on the >> wiki and in the module. Keep in mind that this is a relatively new module >> so we don't have lots of examples yet, so you'll probably be doing a fair >> amount of research and testing. >> >> -MC >> >> >> On Wed, May 16, 2012 at 5:59 AM, Sanath Prasanna wrote: >> >>> Hi all, >>> I have 2 servers. One server has SIP GW connection From Operator & IVR >>> applications need to build in other server. How to call distributed LUA >>> applications with Mysql Databases from the SIP GW server ? Pls advice. >>> Main idea is, maintaining SIP connection in one server & all the IVR >>> applications in other server. >>> Br, >>> Sanath >>> >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120518/0d9d1f91/attachment.html From msc at freeswitch.org Fri May 18 08:27:24 2012 From: msc at freeswitch.org (Michael Collins) Date: Thu, 17 May 2012 21:27:24 -0700 Subject: [Freeswitch-users] Run LUA script in different server In-Reply-To: References: Message-ID: If I understand your question correctly, yes you can do this. You can send calls from one FreeSWITCH server to another. Start here: http://wiki.freeswitch.org/wiki/Connect_Two_FreeSWITCH_Boxes Best way to learn is to get the FreeSWITCH books from Packt Publishing and just start hacking code. -MC On Thu, May 17, 2012 at 9:07 PM, Sanath Prasanna wrote: > Tx for advice MC & Anita. Can I do work around like this . > Another freeswitch instant will be start in other server & calls will be > transfer from operator connected freeswitch instance to this new freeswitch > instance & vise versa. Pls advice. > > > On Thu, May 17, 2012 at 5:05 PM, Anita Hall wrote: > >> You could run a Lua ESL server on a different machine but this will not >> be the same as running a Lua script. >> http://wiki.freeswitch.org/wiki/Event_Socket_Library >> >> regards, >> Anita >> >> >> >> On Thu, May 17, 2012 at 4:37 AM, Michael Collins wrote: >> >>> I don't think you can directly do what you are describing. However, you >>> might be able to use mod_httapi for this. There's some documentation on the >>> wiki and in the module. Keep in mind that this is a relatively new module >>> so we don't have lots of examples yet, so you'll probably be doing a fair >>> amount of research and testing. >>> >>> -MC >>> >>> >>> On Wed, May 16, 2012 at 5:59 AM, Sanath Prasanna wrote: >>> >>>> Hi all, >>>> I have 2 servers. One server has SIP GW connection From Operator & IVR >>>> applications need to build in other server. How to call distributed LUA >>>> applications with Mysql Databases from the SIP GW server ? Pls advice. >>>> Main idea is, maintaining SIP connection in one server & all the IVR >>>> applications in other server. >>>> Br, >>>> Sanath >>>> >>>> >>>> >>> >>> >>> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120517/dbfa2f2b/attachment.html From rrodolfos at gmail.com Fri May 18 08:32:54 2012 From: rrodolfos at gmail.com (RrodolfoS .) Date: Fri, 18 May 2012 00:02:54 -0430 Subject: [Freeswitch-users] mod_say_es fail when tray pronounce voicemails messages dates In-Reply-To: References: Message-ID: Michael, I update Freeswitch today last git version: FreeSWITCH Version 1.2.0 (git-bbdcd33 2012-05-17 20-10-53 -0500) but fail persist. I did the bug report in jira, will wait for a response on failure. Thanks, RrodolfoS PD.: Actualice a la ultima versi?n de Freswitch desde Git, la falla persiste, tambi?n lo reporte en jira, esperare una respuesta sobre la falla, Muchas gracias por el inter?s! Saludos. On Thu, May 17, 2012 at 1:10 PM, Michael Collins wrote: > Go ahead and update to latest git and make sure the problem still exists. > (It probably will. :) The go to jira.freeswitch.org and open a ticket. I > know that English is not your first language but that's okay, just do your > best. If you really need to express something and are not sure how to do it > in English then do it in Spanish and we'll use Google translate to help out. > > Thanks, > MC > > On Thu, May 17, 2012 at 9:21 AM, RrodolfoS . wrote: >> >> mod_say_es fail when tray pronounce voicemails messages dates, in >> english work well but in espanish fail. >> >> In spanish the ivr and voicemail intro work fine, except the >> voicemails messages. >> >> The logs is the same of this http://www.freeswitch.es/node/14755 >> >> Configs details: >> >> Version >> FreeSWITCH Version 1.1.beta1 (git-4283408 2012-04-29 11-33-24 -0400) >> >> /usr/src/freeswitch/modules.conf >> ?say/mod_say_es >> >> # make mod_say_es-install >> >> /usr/local/freeswitch/conf/freeswitch.xml >> ?
>> ? ? >> ? ? >> ? ? >> ? ? >> ? ? >> ? ? >> ?
>> >> /usr/local/freeswitch/conf/vars.xml >> ? >> >> /usr/local/freeswitch/conf/lang/es/es.xml >> >> ?> sound-prefix="$${sounds_dir}/es/mx/maria" tts-engine="cepstral" >> tts-voice="maria"> >> ? ? >> ? ? ? >> ? ? ? ? >> ? ? ? ? >> ? ? ? ? >> ? ? ? ? >> ? ? ? ? ? >> ? ? ? >> ? ? ? >> ? ? >> ? >> >> >> /usr/local/freeswitch/conf/autoload_configs/modules.conf.xml >> ? ? >> ? ? >> ? ? >> ? ? >> ? ? >> >> The spanish sounds are in /usr/local/freeswitch/sounds/es/mx/maria/ >> with 8000, 16000, 32000, 44100 and 48000 sample rate mono. >> >> /usr/local/freeswitch/conf/lang/es are a modified copy of >> /usr/local/freeswitch/conf/lang/en >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From philq at qsystemsengineering.com Fri May 18 08:37:06 2012 From: philq at qsystemsengineering.com (Phil Quesinberry) Date: Fri, 18 May 2012 00:37:06 -0400 Subject: [Freeswitch-users] 183 and no ringback when send_silence_when_idle is non-zero Message-ID: <002a01cd34af$e15f1b00$a41d5100$@com> Ok. more info on this. In default Proxy-Media mode: When making a call through the gateway for PSTN termination, FS sends 180 - Ringing to the calling endpoint when 'send_silence_when_idle' is set to 0 and ringback tone is heard. FS sends 183 and early media to the calling endpoint when 'send_silence_when_idle' is set to a non-zero value and no ringback tone is heard. In Bypass-Media mode: Making that same call through the gateway, FS sends 180 - Ringing to the calling endpoint regardless of the 'send_silence_when_idle' value and ringback tone is heard. Is this the intended behavior? - Phil _____________________________________________ From: Phil Quesinberry Sent: Thursday, May 17, 2012 1:46 PM To: 'freeswitch-users at lists.freeswitch.org' Subject: Re: Loud whilte noise during sleep Actually it's one of the latest 1.2.0 revisions - b653c21. I didn't realize that with the quick look I had at the code. Curious. that makes me wonder if the Aastra phones are generating the noise when they don't see any media. A user complained about hearing the noise when first logging into voicemail, before she is asked for her PIN. I hadn't noticed it before because I normally do that on speaker but on the handset, it is quite noticeable. Is there a way to use send_silence_when_idle without having it kill ringback when set to a non-zero value? ---------- Anthony Minessale Thu May 17 20:21:35 MSD 2012 Which revision of the code are you looking at? The sleep sends dummy frames unless you set that send_silence_when_idle var, if its not set its probably not sending anything. You can see in the code that unless that var is set it wont be generating anything. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120518/92f71a6f/attachment.html From miha at softnet.si Fri May 18 10:05:35 2012 From: miha at softnet.si (Miha) Date: Fri, 18 May 2012 08:05:35 +0200 Subject: [Freeswitch-users] bridge a call to registered user In-Reply-To: References: <4FB39A66.5070809@softnet.si> <4FB49CD7.3040509@softnet.si> Message-ID: <4FB5E6AF.7030907@softnet.si> On 5/17/2012 8:39 PM, Michael Collins wrote: > I'll have to defer to those with more experience than I. SIP presence, > multitenancy, etc. is outside my area of expertise... > > -MC > > On Wed, May 16, 2012 at 11:38 PM, Miha > wrote: > > HI @Michael, > > I tried to uncomment this due to Multi-tenant configuration. As > this did not helped me I tried to use Presence variable > (http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#Presence). > After this FS does not log any more as user not registered. > > Could this function be causing any other problems? > > THanks! > > Miha > > > > On 5/17/2012 12:17 AM, Michael Collins wrote: >> This is probably because the two different profiles are not set >> to use the same register domain. Check out this wiki page and try >> setting the value to be the same for all your profiles: >> http://wiki.freeswitch.org/wiki/Sofia.conf.xml#force-register-domain >> >> -MC >> >> On Wed, May 16, 2012 at 5:15 AM, Miha > > wrote: >> >> Hi, >> >> just one question. I have created two profiles (suggested by >> @Michael) >> as I was having same NAT issue. Profiles are basically the >> same only >> registration port is different (5060, 5070) and NAT profile >> have set: >> . >> >> In my dialplan I have made bridge like this: >> >> >> > data="{origination_callee_id_name='${effective_caller_id_name}',origination_calee_id_number='${effective_caller_id_name}'}user/${destination_number}.fs_test at fs_test.fs2.blabla.com >> " >> /> >> >> freeswitch at default> show re >> >> [ registrations] >> >> >> freeswitch at default> show registrations >> reg_user,realm,token,url,expires,network_ip,network_port,network_proto,hostname >> 018108753.fs_test,fs_test.fs2.blabla.com >> ,40804176-fb1f176 at 172.31.1.190 >> ,sofia/internal/sip:018108753.fs_test at xxx.xxx.xxx.xxx:1265,1337169317,xxx.xxx.xxx.xxx,1265,udp,localhost.localdomain >> >> 018108752.fs_test,fs_test.fs2.blabla.comi,873cfe40-1224f25 at 172.31.1.171 >> ,sofia/nat/sip:018108752.fs_test at xxx.xxx.xxx.xxx:2097,1337170569,xxx.xxx.xxx.xxx,1920,udp,localhost.localdomain >> >> >> 2 total. >> >> Why only phone work which is registered on NAT profile. IF I >> call users >> which is registered on internal profile I get user not >> registered? >> >> Thanks! >> >> Miha >> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org @Michael, thanks anyway:) Reagrds, Miha -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120518/7c626bf8/attachment.html From anita.hall at simmortel.com Fri May 18 12:34:51 2012 From: anita.hall at simmortel.com (Anita Hall) Date: Fri, 18 May 2012 14:04:51 +0530 Subject: [Freeswitch-users] Spandsp mulated modems inbound/no /dev/FS* device? In-Reply-To: <4FB553A3.5010807@integrafin.co.uk> References: <4FA24100.7040908@integrafin.co.uk> <20120503151946.GA824@eagle.cupis.co.uk> <4FA3EE97.1090908@integrafin.co.uk> <4FA596BF.4090703@integrafin.co.uk> <4FA635AE.5050006@integrafin.co.uk> <4FA66862.1090300@coppice.org> <4FA6871F.5050909@puzzled.xs4all.nl> <4FA696B1.8050706@coppice.org> <4FB553A3.5010807@integrafin.co.uk> Message-ID: Hi Alex Many thanks for the reply. Do I need t38modem if I am already receiving the Fax calls over E1 via a Sangoma Card in my FS machine ? I read on the list that Hylafax takes audio and the only reason you needed t38modem was because your FS was getting calls over SIP. This is my topology ~~~~~~~~~ Hylafax <-------->|______| <-------> FreeSWITCH <---------> E1 ~~~~~~~~~ Since Hylafax is already taking audio calls, may be I do not need the t38modem ? If not, can Hylafax take audio calls directly from FreeSWITCH ? This is your topology (taken from your mails in the list) ~~~~~~~~~ Hylafax<---Audio---->t38modem<---T38--->Freeswitch/t38gateway<----G.711/SIP--->Mitel 3300(T38 unsupported, grr)<---ISDN30---->PSTN ~~~~~~~~~ Many thanks for the discussion. regards, Anita On Fri, May 18, 2012 at 1:08 AM, Alex Crow wrote: > On 17/05/12 14:51, Anita Hall wrote: > > Hi Alex > > > > Did you get the answer to your first question ? Did you succeed in > > using the emulated modem option for taking fax calls to hylafax ? > > > > I put the following in spandsp.conf.xml and got /dev/FS[0-4] devices > > as soft links to /dev/pts/[4-8]. My freeswitch is running as root user > > so I did not face the issue you did. > > > > > > > > > > > > What next? Could you point me to some doc in hylafax? > > > > (And before Steve lashes out at me again, I must clarify, I do not > > want to just play around, but my boss is clueless and wants me to > > evaluate hylafax :() > > > > regards, > > Anita > > > > > > Anita, > > The answer I got was that inbound is not yet supported by that > mechanism, so I'd advise trying the Hylafax/T38Modem option if you need > to use Hylafax. The T38Modem site provides tarballs of the OPAL and > required dependencies to build it. The one main advantage we see for it > is that there are quite a few free and commercial clients for Hylafax > that can behave as a printer in Windows, so the user selects to print > say a .doc file, chooses the fax printer, enters a number and off it > goes. We used WHFC with XP, works great. We even have chaps batch faxing > 100+ faxes a day using it from Excel with a macro. > > The other route for inbound is well documented, it using the inbuilt > FreeSWITCH faxing and sending to email. Outbound is equally simple but > not IMHO as easy for our end users as Hylafax (email-to-fax is simple > but printing to fax from Windows not so simple). But with some work on > your side, the spandsp stuff could probably do anything HF can do. > > As I said, Hylafax is excellent software and has been deployed on major > production sites for probably 15 years+ with billions of total faxes > delivered. We've used it for more than ten years with about 400 faxes > in/out per day (which is piddling compared with some other sites, 10,000 > per day is not difficult). If you want to avoid using t38modem you'd > probably have to set up a dedicated HF box with an ISDN/POTS > multichannel card (and if you want to fax from/to FS use a BRI or T1/E1 > card to send audio to and from it). There are other ways of integrating > it with backoffice systems including FTP, email, etc. > > To try to deploy a business fax system without first evaluating HylaFAX > is in my opinion insane, so maybe your boss has a point. If, however, > your boss is the one driving you to ask all these very diverse questions > on this list (and they are indeed frequent and often completely > unrelated to each other), perhaps you should ask him/her to focus on > those that are his top priorities, and pick the first 3 to 5 and focus > on those first, or in fact just compose a list of required features, > post them up in a single email rather than 50, and ask "is this possible?". > > Just being a bit more targeted with your questions and how they link > together might elicit more responses. Try to pick one solution at a > time, test it, ask questions on that and stick with it for a bit and I > think you'll get more help. > > Cheers > > Alex > > > -- > This message is intended only for the addressee and may contain > confidential information. Unless you are that person, you may not > disclose its contents or use it in any way and are requested to delete > the message along with any attachments and notify us immediately. > > "Transact" is operated by Integrated Financial Arrangements plc > Domain House, 5-7 Singer Street, London EC2A 4BQ > Tel: (020) 7608 4900 Fax: (020) 7608 5300 > (Registered office: as above; Registered in England and Wales under > number: 3727592) > Authorised and regulated by the Financial Services Authority (entered on > the FSA Register; number: 190856) > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120518/7ed0de34/attachment-0001.html From tarik.bts.gi at gmail.com Fri May 18 13:37:09 2012 From: tarik.bts.gi at gmail.com (ghallab) Date: Fri, 18 May 2012 09:37:09 +0000 Subject: [Freeswitch-users] transfer caller ID problem and plycom In-Reply-To: <4FB54C96.4010702@gmail.com> References: <4FB54C96.4010702@gmail.com> Message-ID: <4FB61845.9010600@gmail.com> You don't have any idea? what should I do to figure out this problem? On 05/17/2012 07:08 PM, ghallab wrote: > When their Polycom 550 rings the user see the caller ID of the person > who transferred the call, once they answer the phone the caller ID of > the transferred caller is displayed. > I already tried the ignore display updates set to faulse and true in > the vars.xml file. But don't have effect! > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120518/38766d86/attachment.html From tarik.bts.gi at gmail.com Fri May 18 13:49:41 2012 From: tarik.bts.gi at gmail.com (ghallab) Date: Fri, 18 May 2012 09:49:41 +0000 Subject: [Freeswitch-users] transfer caller ID problem and plycom In-Reply-To: <4FB61845.9010600@gmail.com> References: <4FB54C96.4010702@gmail.com> <4FB61845.9010600@gmail.com> Message-ID: <4FB61B35.5040208@gmail.com> Just to explain more I will give a scenario: A is an outside user that call an internal user B then B answer and transfer the call to anther internal user C. When C phone is ringing he see the caller ID of B and when he answer he see the caller ID of A. I tried to set ignore_display_updates in vars.xml, But don't have effect. Could help please? On 05/18/2012 09:37 AM, ghallab wrote: > You don't have any idea? what should I do to figure out this problem? > > On 05/17/2012 07:08 PM, ghallab wrote: >> When their Polycom 550 rings the user see the caller ID of the person >> who transferred the call, once they answer the phone the caller ID of >> the transferred caller is displayed. >> I already tried the ignore display updates set to faulse and true in >> the vars.xml file. But don't have effect! >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120518/066dfdad/attachment.html From peter.olsson at visionutveckling.se Fri May 18 13:03:41 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Fri, 18 May 2012 09:03:41 +0000 Subject: [Freeswitch-users] transfer caller ID problem and plycom In-Reply-To: <4FB61B35.5040208@gmail.com> References: <4FB54C96.4010702@gmail.com> <4FB61845.9010600@gmail.com>,<4FB61B35.5040208@gmail.com> Message-ID: I guess you should be setting the variable in the dialplan, before you execute the bridge app. /Peter 18 maj 2012 kl. 10:56 skrev "ghallab" >: Just to explain more I will give a scenario: A is an outside user that call an internal user B then B answer and transfer the call to anther internal user C. When C phone is ringing he see the caller ID of B and when he answer he see the caller ID of A. I tried to set ignore_display_updates in vars.xml, But don't have effect. Could help please? On 05/18/2012 09:37 AM, ghallab wrote: You don't have any idea? what should I do to figure out this problem? On 05/17/2012 07:08 PM, ghallab wrote: When their Polycom 550 rings the user see the caller ID of the person who transferred the call, once they answer the phone the caller ID of the transferred caller is displayed. I already tried the ignore display updates set to faulse and true in the vars.xml file. But don't have effect! !DSPAM:4fb60b9532761134834124! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4fb60b9532761134834124! From acrow at integrafin.co.uk Fri May 18 13:15:05 2012 From: acrow at integrafin.co.uk (Alex Crow) Date: Fri, 18 May 2012 10:15:05 +0100 Subject: [Freeswitch-users] Spandsp mulated modems inbound/no /dev/FS* device? In-Reply-To: References: <4FA24100.7040908@integrafin.co.uk> <20120503151946.GA824@eagle.cupis.co.uk> <4FA3EE97.1090908@integrafin.co.uk> <4FA596BF.4090703@integrafin.co.uk> <4FA635AE.5050006@integrafin.co.uk> <4FA66862.1090300@coppice.org> <4FA6871F.5050909@puzzled.xs4all.nl> <4FA696B1.8050706@coppice.org> <4FB553A3.5010807@integrafin.co.uk> Message-ID: <4FB61319.6060205@integrafin.co.uk> On 18/05/12 09:34, Anita Hall wrote: > Hi Alex > > Many thanks for the reply. > > Do I need t38modem if I am already receiving the Fax calls over E1 via > a Sangoma Card in my FS machine ? I read on the list that Hylafax > takes audio and the only reason you needed t38modem was because your > FS was getting calls over SIP. > Anita, Yes, you do if you want to use Hylafax and need inbound faxing, because it needs a modem device. T38Modem provides that (eg /dev/ttyT38x). > This is my topology > ~~~~~~~~~ > Hylafax <-------->|______| <-------> FreeSWITCH <---------> E1 Pretty much the same as mine in effect. So you would need Hylafax <-------->T38Modem<-------> FreeSWITCH/t38gateway <---------> E1 So hylafax speaks "Fax data" to T38modem, which speaks T38 to FreeSwitch, which gateways to/from audio over the E1. You may need to change my configs a bit to get that last bit working, maybe not. > ~~~~~~~~~ > > Since Hylafax is already taking audio calls, may be I do not need the > t38modem ? If not, can Hylafax take audio calls directly from FreeSWITCH ? Not unless the inbuilt modem can be made to be an endpoint for calls from coming in from the E1, which I understand at present it cannot. This is because Hylafax does not send and receive audio itself, it talks fax control commands to modems and sends and receives data over a serial port. > > This is your topology (taken from your mails in the list) > ~~~~~~~~~ > Hylafax<---Audio---->t38modem<---T38--->Freeswitch/t38gateway<----G.711/SIP--->Mitel > 3300(T38 unsupported, grr)<---ISDN30---->PSTN > ~~~~~~~~~ Actually the Mitel does support T38 with a license. The thing I'm not sure about is how good its gatewaying to ISDN is. Cheers Alex -- This message is intended only for the addressee and may contain confidential information. Unless you are that person, you may not disclose its contents or use it in any way and are requested to delete the message along with any attachments and notify us immediately. "Transact" is operated by Integrated Financial Arrangements plc Domain House, 5-7 Singer Street, London EC2A 4BQ Tel: (020) 7608 4900 Fax: (020) 7608 5300 (Registered office: as above; Registered in England and Wales under number: 3727592) Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856) From albert_nguyen16 at hotmail.com Fri May 18 13:34:02 2012 From: albert_nguyen16 at hotmail.com (Albert Nguyen) Date: Fri, 18 May 2012 09:34:02 +0000 Subject: [Freeswitch-users] [WARNING] switch_core.c:1206 can not locate domain 10.239.236.1 In-Reply-To: References: Message-ID: Hi When I start FS, I get a warning message from FS like [WARNING] switch_core.c:1206 can not locate domain 10.239.236.1 but 10.239.236.1 is the local NIC ip address. How can I fix this problem? Thanks in advance. Al -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120518/6382de7f/attachment.html From tarik.bts.gi at gmail.com Fri May 18 14:36:07 2012 From: tarik.bts.gi at gmail.com (ghallab) Date: Fri, 18 May 2012 10:36:07 +0000 Subject: [Freeswitch-users] transfer caller ID problem and plycom In-Reply-To: References: <4FB54C96.4010702@gmail.com> <4FB61845.9010600@gmail.com>, <4FB61B35.5040208@gmail.com> Message-ID: <4FB62617.6040304@gmail.com> Thanks for tour replay. But normally when you configure some thing in vars.xml it apply to all calls. On 05/18/2012 09:03 AM, Peter Olsson wrote: > I guess you should be setting the variable in the dialplan, before you execute the bridge app. > > /Peter > > 18 maj 2012 kl. 10:56 skrev "ghallab">: > > Just to explain more I will give a scenario: > A is an outside user that call an internal user B then B answer and transfer the call to anther internal user C. When C phone is ringing he see the caller ID of B and when he answer he see the caller ID of A. I tried to set ignore_display_updates in vars.xml, But don't have effect. > Could help please? > > > On 05/18/2012 09:37 AM, ghallab wrote: > You don't have any idea? what should I do to figure out this problem? > > On 05/17/2012 07:08 PM, ghallab wrote: > When their Polycom 550 rings the user see the caller ID of the person who transferred the call, once they answer the phone the caller ID of the transferred caller is displayed. > I already tried the ignore display updates set to faulse and true in the vars.xml file. But don't have effect! > > > > > !DSPAM:4fb60b9532761134834124! > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > !DSPAM:4fb60b9532761134834124! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From peter.olsson at visionutveckling.se Fri May 18 13:50:06 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Fri, 18 May 2012 09:50:06 +0000 Subject: [Freeswitch-users] transfer caller ID problem and plycom In-Reply-To: <4FB62617.6040304@gmail.com> References: <4FB54C96.4010702@gmail.com> <4FB61845.9010600@gmail.com>, <4FB61B35.5040208@gmail.com> , <4FB62617.6040304@gmail.com> Message-ID: <13AFEEB9-AA80-4253-A9D0-136CC6CA82FE@visionutveckling.se> Yes, for some things this is true - but only if something later on handles that configured variable. Preparser variables can be used by using the syntax $${variable}, so in this case you will still ned to modify the actual dialplan to use that var. /Peter 18 maj 2012 kl. 11:41 skrev "ghallab" : > Thanks for tour replay. > But normally when you configure some thing in vars.xml it apply to all > calls. > > On 05/18/2012 09:03 AM, Peter Olsson wrote: >> I guess you should be setting the variable in the dialplan, before you execute the bridge app. >> >> /Peter >> >> 18 maj 2012 kl. 10:56 skrev "ghallab">: >> >> Just to explain more I will give a scenario: >> A is an outside user that call an internal user B then B answer and transfer the call to anther internal user C. When C phone is ringing he see the caller ID of B and when he answer he see the caller ID of A. I tried to set ignore_display_updates in vars.xml, But don't have effect. >> Could help please? >> >> >> On 05/18/2012 09:37 AM, ghallab wrote: >> You don't have any idea? what should I do to figure out this problem? >> >> On 05/17/2012 07:08 PM, ghallab wrote: >> When their Polycom 550 rings the user see the caller ID of the person who transferred the call, once they answer the phone the caller ID of the transferred caller is displayed. >> I already tried the ignore display updates set to faulse and true in the vars.xml file. But don't have effect! >> >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> !DSPAM:4fb60b9532761134834124! >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > !DSPAM:4fb6162632768978511028! > From peter.olsson at visionutveckling.se Fri May 18 14:10:07 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Fri, 18 May 2012 10:10:07 +0000 Subject: [Freeswitch-users] One core at 100% usage In-Reply-To: References: Message-ID: <4B4CF310-EC50-46A9-8406-F043E0BFA4B2@visionutveckling.se> If this is still reproducable on latest git, please file a Jira and supply as much details as possible. At least you will need to upload debug log, and a stack trace when this happens. /Peter 17 maj 2012 kl. 22:41 skrev "Stephen Wilde" : > I have done a git pull (now git-83e090c 2012-05-17 16-40-30 +0000) and I have a strange load distribution over cores: one core at 100%. > > Here the htop: > > > 1 [|||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||100.0%] > 2 [||||||||||||||||| 19.1%] > 3 [|| 1.3%] > 4 [|| 1.3%] > 5 [ 0.0%] > 6 [ 0.0%] > 7 [|||| 3.9%] > 8 [||||||||||||| 13.2%] > 9 [ 0.0%] > 10 [||||||||||||||||| 18.3%] > 11 [||||| 4.6%] > 12 [|||||||||||||||| 17.0%] > 13 [|||||||| 8.6%] > 14 [|||||||||||| 12.5%] > 15 [|||||||| 8.6%] > 16 [||||||||||||||||||| 21.1%] > > > After revert to an old git I have this htop: > > 1 [||||||| 7.2%] > 2 [|||||||||||| 11.8%] > 3 [||||| 5.3%] > 4 [||||||| 6.6%] > 5 [|||| 3.3%] > 6 [||||||||| 9.0%] > 7 [||||| 4.6%] > 8 [|||||||||||||| 15.1%] > 9 [||||||| 6.5%] > 10 [||||||||| 9.3%] > 11 [|||||| 6.5%] > 12 [|||||||||||||| 14.9%] > 13 [|||||||| 7.8%] > 14 [|||||||||||||||| 17.0%] > 15 [|||||||| 7.3%] > 16 [||||||||||| 11.8%] > > anyone with same issue? > !DSPAM:4fb55f8732762045212077! > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > !DSPAM:4fb55f8732762045212077! From gozdal at gmail.com Fri May 18 14:24:15 2012 From: gozdal at gmail.com (Marcin Gozdalik) Date: Fri, 18 May 2012 12:24:15 +0200 Subject: [Freeswitch-users] T.38/spandsp_fax interop problem In-Reply-To: References: Message-ID: 2012/5/16 Marcin Gozdalik > I'm trying to send fax from FreeSWITCH to PSTN using T.38. FS is connected > to Broadsoft softswitch. The softswitch is connected to PSTN using > SIP-trunk to external provider. Apparently there is no T.38 on Broadsoft > and it is provided by the external provider (i.e. Broadsoft provides only > T.38 pass-through). > > [snip] > For anybody struggling with the same problem: it turned out to be loopback problem. It is possible to transmit a fax through loopback using txfax when the other side is issuing reINVITE. It is however not possible to force FS to issue reINVITE in txfax over loopback (using enable_t38_request). Best regards -- Marcin Gozdalik -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120518/8f06104b/attachment.html From anita.hall at simmortel.com Fri May 18 16:01:32 2012 From: anita.hall at simmortel.com (Anita Hall) Date: Fri, 18 May 2012 17:31:32 +0530 Subject: [Freeswitch-users] set dtmf type without sip profile Message-ID: Hi I am taking DID from voxbone.com, they do not need authentication by login and password, so I have not created a SIP profile. By allowing their IP addresses in acl.conf.xml, I am able to receive incoming calls and media. However, dtmf is not coming. Doing tshark, no RTP event corresponding to dtmf is being received. Voxbone sends DTMF over rfc2833 I have set in conf/sip_profiles/internal.xml and done reload mod_sofia But I am not sure if this is the correct place. Is there any other pace where dtmf type can be specified in the absence of a sip profile ? regards, Anita -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120518/d545142a/attachment.html From saami_mh at ymail.com Fri May 18 16:15:53 2012 From: saami_mh at ymail.com (Samira Mh) Date: Fri, 18 May 2012 05:15:53 -0700 (PDT) Subject: [Freeswitch-users] how to fix : RECOVERY_ON_TIMER_EXPIRE?? Message-ID: <1337343353.86641.YahooMailNeo@web120106.mail.ne1.yahoo.com> hi guys, every time i want to make call via gateway on freeswitch the following error occured: 2012-05-18 16:17:16.296384 [INFO] mod_dptools.c:2355 Originate Failed. ?Cause: RECOVERY_ON_TIMER_EXPIRE ?plz help,thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120518/3de80789/attachment.html From tarik.bts.gi at gmail.com Fri May 18 17:19:33 2012 From: tarik.bts.gi at gmail.com (ghallab) Date: Fri, 18 May 2012 13:19:33 +0000 Subject: [Freeswitch-users] how to fix : RECOVERY_ON_TIMER_EXPIRE?? In-Reply-To: <1337343353.86641.YahooMailNeo@web120106.mail.ne1.yahoo.com> References: <1337343353.86641.YahooMailNeo@web120106.mail.ne1.yahoo.com> Message-ID: <4FB64C65.80900@gmail.com> At this link you will find the reason why this is happens: http://wiki.freeswitch.org/wiki/Hangup_causes On 05/18/2012 12:15 PM, Samira Mh wrote: > hi guys, > every time i want to make call via gateway on freeswitch the following > error occured: > > 2012-05-18 16:17:16.296384 [INFO] mod_dptools.c:2355 Originate Failed. > Cause: RECOVERY_ON_TIMER_EXPIRE > plz help,thanks > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120518/67c725e0/attachment.html From yehavi.bourvine at gmail.com Fri May 18 16:24:59 2012 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Fri, 18 May 2012 15:24:59 +0300 Subject: [Freeswitch-users] transfer caller ID problem and plycom In-Reply-To: <4FB61845.9010600@gmail.com> References: <4FB54C96.4010702@gmail.com> <4FB61845.9010600@gmail.com> Message-ID: To me it looks like the intended and correct behavior... _Yehavi: 2012/5/18 ghallab > You don't have any idea? what should I do to figure out this problem? > > > On 05/17/2012 07:08 PM, ghallab wrote: > > When their Polycom 550 rings the user see the caller ID of the person who > transferred the call, once they answer the phone the caller ID of the > transferred caller is displayed. > I already tried the ignore display updates set to faulse and true in the > vars.xml file. But don't have effect!**** > > ** ** > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120518/037c5c10/attachment-0001.html From saami_mh at ymail.com Fri May 18 16:34:52 2012 From: saami_mh at ymail.com (Samira Mh) Date: Fri, 18 May 2012 05:34:52 -0700 (PDT) Subject: [Freeswitch-users] how to fix : RECOVERY_ON_TIMER_EXPIRE?? In-Reply-To: <4FB64C65.80900@gmail.com> References: <1337343353.86641.YahooMailNeo@web120106.mail.ne1.yahoo.com> <4FB64C65.80900@gmail.com> Message-ID: <1337344492.47593.YahooMailNeo@web120101.mail.ne1.yahoo.com> i read the link but i couldn't fix it? ________________________________ From: ghallab To: FreeSWITCH Users Help Sent: Friday, May 18, 2012 5:49 PM Subject: Re: [Freeswitch-users] how to fix : RECOVERY_ON_TIMER_EXPIRE?? ?At this link you will find the reason why this is happens: http://wiki.freeswitch.org/wiki/Hangup_causes On 05/18/2012 12:15 PM, Samira Mh wrote: hi guys, >every time i want to make call via gateway on freeswitch the following error occured: > > >2012-05-18 16:17:16.296384 [INFO] mod_dptools.c:2355 Originate Failed. ?Cause: RECOVERY_ON_TIMER_EXPIRE >?plz help,thanks > > >_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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 at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120518/83f44584/attachment.html From saami_mh at ymail.com Fri May 18 16:35:43 2012 From: saami_mh at ymail.com (Samira Mh) Date: Fri, 18 May 2012 05:35:43 -0700 (PDT) Subject: [Freeswitch-users] how to fix : RECOVERY_ON_TIMER_EXPIRE?? In-Reply-To: <4FB64C65.80900@gmail.com> References: <1337343353.86641.YahooMailNeo@web120106.mail.ne1.yahoo.com> <4FB64C65.80900@gmail.com> Message-ID: <1337344543.52305.YahooMailNeo@web120104.mail.ne1.yahoo.com> i read the link but i could't solve the problem ________________________________ From: ghallab To: FreeSWITCH Users Help Sent: Friday, May 18, 2012 5:49 PM Subject: Re: [Freeswitch-users] how to fix : RECOVERY_ON_TIMER_EXPIRE?? ?At this link you will find the reason why this is happens: http://wiki.freeswitch.org/wiki/Hangup_causes On 05/18/2012 12:15 PM, Samira Mh wrote: hi guys, >every time i want to make call via gateway on freeswitch the following error occured: > > >2012-05-18 16:17:16.296384 [INFO] mod_dptools.c:2355 Originate Failed. ?Cause: RECOVERY_ON_TIMER_EXPIRE >?plz help,thanks > > >_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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 at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120518/2ce68637/attachment.html From wesleyakio at tuntscorp.com Fri May 18 17:10:05 2012 From: wesleyakio at tuntscorp.com (Wesley Akio) Date: Fri, 18 May 2012 10:10:05 -0300 Subject: [Freeswitch-users] FS Core Dump using GSM Hardware In-Reply-To: References: Message-ID: Hi all, The manufacturer is giving me support on this one but they are not sure there is any relation to the hardware... Did someone have the chance of a quick look and seen something odd at all in the backtraces? Anything that can point me in the right direction would appreciated... Best, Wesley Akio TuntsCorp.com On Wed, May 16, 2012 at 11:00 AM, Wesley Akio wrote: > Hi all, > > I'm experiencing several core dumps a day in one of our boxes. > > The problem was happening with FS trunk from Apr 30. Updated yesterday and > the problem continues. > > I suspect it has something to do with the GSM hardware(Khomp) in use but > I'm not sure... > > If someone could take a look at the backtraces I would be very thankfull. > > http://pastebin.freeswitch.org/19059 > FreeSWITCH Version 1.2.0 (git-d015395 2012-05-15 11-26-16 -0500) > > Best, > > Wesley Akio > TuntsCorp.com > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120518/b40daefd/attachment.html From B.Tietz at pinguin.ag Fri May 18 17:38:07 2012 From: B.Tietz at pinguin.ag (B.Tietz at pinguin.ag) Date: Fri, 18 May 2012 15:38:07 +0200 Subject: [Freeswitch-users] start perl-script after receiving fax Message-ID: <07BF4904977CC645B485E970424193AD10EE2A3B81@localhost> Hi, I would like to receive (T38/T30) faxes with rxfax. I think this won't be a problem to do so in dialplan. But, after (succesfull) receiving I would like to call a perl-script which needs some variables, at least the fax-file name, and some other infos. How can I call this script with variables afte receiving? regards, Benjamin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120518/d4a5ce03/attachment-0001.html From dnotivol at gmail.com Fri May 18 18:22:17 2012 From: dnotivol at gmail.com (David Notivol) Date: Fri, 18 May 2012 16:22:17 +0200 Subject: [Freeswitch-users] HA using redis for tracking calls status Message-ID: Hello, I'm thinking on going to a High-Availability scenario with FreeSwitch. Reading the wiki I see the solution for sharing the call status between the master/slave servers is an ODBC source, as Postgresql/MySQL. But I'm a bit concerned of how this could impact the performance of the machine. My idea was to have redis doing this function in this scenario. Do you think this is possible? Does anyone have any expierence with this? Any advice? Thanks a lot. -- Regards, David Notivol -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120518/845f3844/attachment.html From vipkilla at gmail.com Fri May 18 18:25:47 2012 From: vipkilla at gmail.com (Vik Killa) Date: Fri, 18 May 2012 10:25:47 -0400 Subject: [Freeswitch-users] cannot set variable on att_xfer bridge Message-ID: Hi, I can't seem to set variables on the att_xfer call leg. I've tried the following: or or None seem to work. Is it even possible to send variables to the att_xfer call leg? I don't see anything in the wiki about it. From mitch.capper at gmail.com Fri May 18 18:34:25 2012 From: mitch.capper at gmail.com (Mitch Capper) Date: Fri, 18 May 2012 07:34:25 -0700 Subject: [Freeswitch-users] how to fix : RECOVERY_ON_TIMER_EXPIRE?? In-Reply-To: <1337344543.52305.YahooMailNeo@web120104.mail.ne1.yahoo.com> References: <1337343353.86641.YahooMailNeo@web120106.mail.ne1.yahoo.com> <4FB64C65.80900@gmail.com> <1337344543.52305.YahooMailNeo@web120104.mail.ne1.yahoo.com> Message-ID: Did you try a no-nat situation? Did you try another carrier? Both of those are suggested in the wiki as the sources of the problem. ~Mitch On Fri, May 18, 2012 at 5:35 AM, Samira Mh wrote: > i read the link but i could't solve the problem > > ________________________________ > From: ghallab > To: FreeSWITCH Users Help > Sent: Friday, May 18, 2012 5:49 PM > Subject: Re: [Freeswitch-users] how to fix : RECOVERY_ON_TIMER_EXPIRE?? > > ?At this link you will find the reason why this is happens: > http://wiki.freeswitch.org/wiki/Hangup_causes > > On 05/18/2012 12:15 PM, Samira Mh wrote: > > hi guys, > every time i want to make call via gateway on freeswitch the following error > occured: > > 2012-05-18 16:17:16.296384 [INFO] mod_dptools.c:2355 Originate Failed. > ?Cause: RECOVERY_ON_TIMER_EXPIRE > ?plz help,thanks > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From avi at avimarcus.net Fri May 18 18:39:37 2012 From: avi at avimarcus.net (Avi Marcus) Date: Fri, 18 May 2012 17:39:37 +0300 Subject: [Freeswitch-users] HA using redis for tracking calls status In-Reply-To: References: Message-ID: Best would be something that uses the event system. I don't know if mod_ha_cluster is even in alpha yet, though... -Avi On Fri, May 18, 2012 at 5:22 PM, David Notivol wrote: > Hello, > > I'm thinking on going to a High-Availability scenario with FreeSwitch. > Reading the wiki I see the solution for sharing the call status between the > master/slave servers is an ODBC source, as Postgresql/MySQL. But I'm a bit > concerned of how this could impact the performance of the machine. > > My idea was to have redis doing this function in this scenario. Do you > think this is possible? Does anyone have any expierence with this? Any > advice? > > Thanks a lot. > > -- > Regards, > David Notivol > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120518/4866b6d6/attachment.html From fluixab at bellsouth.net Fri May 18 19:00:13 2012 From: fluixab at bellsouth.net (Bernard Fluixa) Date: Fri, 18 May 2012 11:00:13 -0400 Subject: [Freeswitch-users] LINGER Message-ID: <72CBCB40-6B65-4C4C-9F26-7409D6E3DA6B@bellsouth.net> Hello, I am working on an outbound ESL application in C on Linux. I can send commands and receive events normally. My problem is that the "linger" command returns the "-ERR not controlling a session" error message. The socket application is set to be in async full mode. (). Any idea? Thank you B From anthony.minessale at gmail.com Fri May 18 19:04:40 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 18 May 2012 10:04:40 -0500 Subject: [Freeswitch-users] One core at 100% usage In-Reply-To: <4B4CF310-EC50-46A9-8406-F043E0BFA4B2@visionutveckling.se> References: <4B4CF310-EC50-46A9-8406-F043E0BFA4B2@visionutveckling.se> Message-ID: update to latest reproduce cmd: top -H `cat /usr/local/freeswitch/run/freeswitch.pid` find the process that is 100% and note its pid cmd: gcore `cat /usr/local/freeswitch/run/freeswitch.pid` cmd: gdb `which freeswitch` enter: "info threads" look for the one with the same pid as the one you noted above enter: thread enter: bt but you should be using jira, I have to say this 200 times a week. On Fri, May 18, 2012 at 5:10 AM, Peter Olsson wrote: > If this is still reproducable on latest git, please file a Jira and supply as much details as possible. At least you will need to upload debug log, and a stack trace when this happens. > > /Peter > > 17 maj 2012 kl. 22:41 skrev "Stephen Wilde" : > >> I have done a git pull (now git-83e090c 2012-05-17 16-40-30 +0000) and I have a strange load distribution over cores: one core at 100%. >> >> Here the htop: >> >> >> ? 1 ?[|||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||100.0%] >> ? 2 ?[||||||||||||||||| ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? 19.1%] >> ? 3 ?[|| ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? 1.3%] >> ? 4 ?[|| ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? 1.3%] >> ? 5 ?[ ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? 0.0%] >> ? 6 ?[ ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? 0.0%] >> ? 7 ?[|||| ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? 3.9%] >> ? 8 ?[||||||||||||| ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? 13.2%] >> ? 9 ?[ ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? 0.0%] >> ? 10 [||||||||||||||||| ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? 18.3%] >> ? 11 [||||| ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?4.6%] >> ? 12 [|||||||||||||||| ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?17.0%] >> ? 13 [|||||||| ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? 8.6%] >> ? 14 [|||||||||||| ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?12.5%] >> ? 15 [|||||||| ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? 8.6%] >> ? 16 [||||||||||||||||||| ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? 21.1%] >> >> >> After revert to an old git I have this htop: >> >> ? 1 ?[||||||| ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?7.2%] >> ? 2 ?[|||||||||||| ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?11.8%] >> ? 3 ?[||||| ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?5.3%] >> ? 4 ?[||||||| ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?6.6%] >> ? 5 ?[|||| ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? 3.3%] >> ? 6 ?[||||||||| ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?9.0%] >> ? 7 ?[||||| ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?4.6%] >> ? 8 ?[|||||||||||||| ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?15.1%] >> ? 9 ?[||||||| ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?6.5%] >> ? 10 [||||||||| ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?9.3%] >> ? 11 [|||||| ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? 6.5%] >> ? 12 [|||||||||||||| ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?14.9%] >> ? 13 [|||||||| ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? 7.8%] >> ? 14 [|||||||||||||||| ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?17.0%] >> ? 15 [|||||||| ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? 7.3%] >> ? 16 [||||||||||| ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? 11.8%] >> >> anyone with same issue? >> !DSPAM:4fb55f8732762045212077! >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> !DSPAM:4fb55f8732762045212077! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From william.suffill at gmail.com Fri May 18 19:06:45 2012 From: william.suffill at gmail.com (William Suffill) Date: Fri, 18 May 2012 11:06:45 -0400 Subject: [Freeswitch-users] Huawei E169 Group Buy In-Reply-To: References: <4FAF49A5.5000203@coppice.org> <023801cd31e5$10cf3820$326da860$@com> <90D7665C-274B-4DD1-BC8D-A9B97278A005@opencsta.org> <4FB13875.4040907@coppice.org> Message-ID: Did we figure out what model is best for the US market? Looks quite interesting. -- W -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120518/85125358/attachment.html From msc at freeswitch.org Fri May 18 19:19:51 2012 From: msc at freeswitch.org (Michael Collins) Date: Fri, 18 May 2012 08:19:51 -0700 Subject: [Freeswitch-users] [WARNING] switch_core.c:1206 can not locate domain 10.239.236.1 In-Reply-To: References: Message-ID: What shows up when you type "sofia status" at the fs_cli? -MC On Fri, May 18, 2012 at 2:34 AM, Albert Nguyen wrote: > > > ------------------------------ > Hi > > When I start FS, I get a warning message from FS like > > [WARNING] switch_core.c:1206 can not locate domain 10.239.236.1 > > but 10.239.236.1 is the local NIC ip address. How can I fix this problem? > > Thanks in advance. > > Al > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120518/6e2213a7/attachment.html From msc at freeswitch.org Fri May 18 19:24:52 2012 From: msc at freeswitch.org (Michael Collins) Date: Fri, 18 May 2012 08:24:52 -0700 Subject: [Freeswitch-users] mod_say_es fail when tray pronounce voicemails messages dates In-Reply-To: References: Message-ID: Thanks for doing the report. It may take some time as we have a very small number of engineers who are capable of doing this kind of work and they all have jobs, lives, families, etc. :) -MC On Thu, May 17, 2012 at 9:32 PM, RrodolfoS . wrote: > Michael, > > I update Freeswitch today last git version: > > FreeSWITCH Version 1.2.0 (git-bbdcd33 2012-05-17 20-10-53 -0500) > > but fail persist. > > I did the bug report in jira, will wait for a response on failure. > > Thanks, > > RrodolfoS > > PD.: > Actualice a la ultima versi?n de Freswitch desde Git, la falla > persiste, tambi?n lo reporte en jira, esperare una respuesta sobre la > falla, > > Muchas gracias por el inter?s! > > Saludos. > > On Thu, May 17, 2012 at 1:10 PM, Michael Collins > wrote: > > Go ahead and update to latest git and make sure the problem still exists. > > (It probably will. :) The go to jira.freeswitch.org and open a ticket. I > > know that English is not your first language but that's okay, just do > your > > best. If you really need to express something and are not sure how to do > it > > in English then do it in Spanish and we'll use Google translate to help > out. > > > > Thanks, > > MC > > > > On Thu, May 17, 2012 at 9:21 AM, RrodolfoS . > wrote: > >> > >> mod_say_es fail when tray pronounce voicemails messages dates, in > >> english work well but in espanish fail. > >> > >> In spanish the ivr and voicemail intro work fine, except the > >> voicemails messages. > >> > >> The logs is the same of this http://www.freeswitch.es/node/14755 > >> > >> Configs details: > >> > >> Version > >> FreeSWITCH Version 1.1.beta1 (git-4283408 2012-04-29 11-33-24 -0400) > >> > >> /usr/src/freeswitch/modules.conf > >> say/mod_say_es > >> > >> # make mod_say_es-install > >> > >> /usr/local/freeswitch/conf/freeswitch.xml > >>
> >> > >> > >> > >> > >> > >> > >>
> >> > >> /usr/local/freeswitch/conf/vars.xml > >> > >> > >> /usr/local/freeswitch/conf/lang/es/es.xml > >> > >> >> sound-prefix="$${sounds_dir}/es/mx/maria" tts-engine="cepstral" > >> tts-voice="maria"> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> /usr/local/freeswitch/conf/autoload_configs/modules.conf.xml > >> > >> > >> > >> > >> > >> > >> The spanish sounds are in /usr/local/freeswitch/sounds/es/mx/maria/ > >> with 8000, 16000, 32000, 44100 and 48000 sample rate mono. > >> > >> /usr/local/freeswitch/conf/lang/es are a modified copy of > >> /usr/local/freeswitch/conf/lang/en > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> Join Us At ClueCon - Aug 7-9, 2012 > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at 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 at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120518/143f45ba/attachment.html From anthony.minessale at gmail.com Fri May 18 20:11:30 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 18 May 2012 11:11:30 -0500 Subject: [Freeswitch-users] FS Core Dump using GSM Hardware In-Reply-To: References: Message-ID: Do you have the logs on your box? look in freeswitch.log for Invalid State Change this log line should spell out exactly where the bug is in mod_khomp On Fri, May 18, 2012 at 8:10 AM, Wesley Akio wrote: > Hi all, > > The manufacturer is giving me support on this one but they are not sure > there is any relation to the hardware... > > Did someone have the chance of a quick look and seen something odd at all in > the backtraces? Anything that can point me in the right direction > would?appreciated... > > Best, > > Wesley Akio > TuntsCorp.com > > > > On Wed, May 16, 2012 at 11:00 AM, Wesley Akio > wrote: >> >> Hi all, >> >> I'm experiencing several core dumps a day in one of our boxes. >> >> The problem was happening with FS trunk from Apr 30. Updated yesterday and >> the problem continues. >> >> I suspect it has something to do with the GSM hardware(Khomp) in use but >> I'm not sure... >> >> If someone could take a look at the backtraces I would be very thankfull. >> >> http://pastebin.freeswitch.org/19059 >> FreeSWITCH Version 1.2.0 (git-d015395 2012-05-15 11-26-16 -0500) >> >> Best, >> >> Wesley Akio >> TuntsCorp.com > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From anthony.minessale at gmail.com Fri May 18 20:14:49 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 18 May 2012 11:14:49 -0500 Subject: [Freeswitch-users] Loud whilte noise during sleep In-Reply-To: <016e01cd3454$e73d64b0$b5b82e10$@com> References: <016e01cd3454$e73d64b0$b5b82e10$@com> Message-ID: you could try setting supress_cng on the leg facing the aastra, that will take the cng negotiation out of the sdp and possibly disable the locally generated silence. On Thu, May 17, 2012 at 12:45 PM, Phil Quesinberry wrote: > Actually it?s one of the latest 1.2.0 revisions ? b653c21. > > I didn?t realize that with the quick look I had at the code.? Curious? that > makes me wonder if the Aastra phones are generating the noise when they > don?t see any media.? A user complained about hearing the noise when first > logging into voicemail, before she is asked for her PIN.? I hadn?t noticed > it before because I normally do that on speaker but on the handset, it is > quite noticeable. > > Is there a way to use send_silence_when_idle without having it kill ringback > when set to a non-zero value? > > ---------- > > Anthony Minessale > Thu May 17 20:21:35 MSD 2012 > > Which revision of the code are you looking at? > > The sleep sends dummy frames unless you set that > > send_silence_when_idle var, if its not set its probably not sending > > anything. > > You can see in the code that unless that var is set it wont be > > generating anything. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From anthony.minessale at gmail.com Fri May 18 20:15:25 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 18 May 2012 11:15:25 -0500 Subject: [Freeswitch-users] Loud whilte noise during sleep In-Reply-To: References: <016e01cd3454$e73d64b0$b5b82e10$@com> Message-ID: err: suppress_cng On Fri, May 18, 2012 at 11:14 AM, Anthony Minessale wrote: > you could try setting supress_cng on the leg facing the aastra, that > will take the cng negotiation out of the sdp and possibly disable the > locally generated silence. > > > On Thu, May 17, 2012 at 12:45 PM, Phil Quesinberry > wrote: >> Actually it?s one of the latest 1.2.0 revisions ? b653c21. >> >> I didn?t realize that with the quick look I had at the code.? Curious? that >> makes me wonder if the Aastra phones are generating the noise when they >> don?t see any media.? A user complained about hearing the noise when first >> logging into voicemail, before she is asked for her PIN.? I hadn?t noticed >> it before because I normally do that on speaker but on the handset, it is >> quite noticeable. >> >> Is there a way to use send_silence_when_idle without having it kill ringback >> when set to a non-zero value? >> >> ---------- >> >> Anthony Minessale >> Thu May 17 20:21:35 MSD 2012 >> >> Which revision of the code are you looking at? >> >> The sleep sends dummy frames unless you set that >> >> send_silence_when_idle var, if its not set its probably not sending >> >> anything. >> >> You can see in the code that unless that var is set it wont be >> >> generating anything. >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From vipkilla at gmail.com Fri May 18 20:15:36 2012 From: vipkilla at gmail.com (Vik Killa) Date: Fri, 18 May 2012 12:15:36 -0400 Subject: [Freeswitch-users] cannot set variable on att_xfer bridge In-Reply-To: References: Message-ID: For anyone having the same issue, I was able to send variables in the SIP Header using: From msc at freeswitch.org Fri May 18 20:31:46 2012 From: msc at freeswitch.org (Michael Collins) Date: Fri, 18 May 2012 09:31:46 -0700 Subject: [Freeswitch-users] start perl-script after receiving fax In-Reply-To: <07BF4904977CC645B485E970424193AD10EE2A3B81@localhost> References: <07BF4904977CC645B485E970424193AD10EE2A3B81@localhost> Message-ID: Check out the stuff you can do with execute_on_fax_XXX chan vars: http://wiki.freeswitch.org/wiki/Mod_spandsp#Execute_based_on_fax_session_outcome -MC On Fri, May 18, 2012 at 6:38 AM, wrote: > Hi,**** > > ** ** > > I would like to receive (T38/T30) faxes with rxfax.**** > > I think this won?t be a problem to do so in dialplan. But, after > (succesfull) receiving I would like to call a perl-script which needs some > variables, at least the fax-file name, and some other infos.**** > > ** ** > > How can I call this script with variables afte receiving?**** > > ** ** > > regards,**** > > Benjamin**** > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120518/922d1bbe/attachment.html From wesleyakio at tuntscorp.com Fri May 18 20:32:22 2012 From: wesleyakio at tuntscorp.com (Wesley Akio) Date: Fri, 18 May 2012 13:32:22 -0300 Subject: [Freeswitch-users] FS Core Dump using GSM Hardware In-Reply-To: References: Message-ID: Thank you very much for the reply Anthony! Although I have almost 30 core dumps in my hands(from the last 3 days only) that message appears in the logs only twice(maybe it crashed before writing to log?) The times are consistent with crashes thought... freeswitch.log.2012-05-17-14-25-41.1-2012-05-17 13:54:17.996544 [DEBUG] switch_core_state_machine.c:685 (Khomp/0/14/s) State REPORTING going to sleep freeswitch.log.2012-05-17-14-25-41.1:2012-05-17 13:54:17.996544 [WARNING] switch_core_state_machine.c:410 (Khomp/0/14/s) Invalid State Change CS_EXECUTE -> CS_DESTROY -- freeswitch.log.2012-05-17-17-14-38.1-2012-05-17 16:38:28.697842 [DEBUG] switch_core_state_machine.c:685 (Khomp/0/10/s) State REPORTING going to sleep freeswitch.log.2012-05-17-17-14-38.1:2012-05-17 16:38:28.697842 [WARNING] switch_core_state_machine.c:410 (Khomp/0/10/s) Invalid State Change CS_EXECUTE -> CS_DESTROY I've changed the log configuration to add the uuid in order to sort the logs for the lifetime of the call. Best, Wesley Akio TuntsCorp.com On Fri, May 18, 2012 at 1:11 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > Do you have the logs on your box? > look in freeswitch.log for Invalid State Change this log line should > spell out exactly where the bug is in mod_khomp > > On Fri, May 18, 2012 at 8:10 AM, Wesley Akio > wrote: > > Hi all, > > > > The manufacturer is giving me support on this one but they are not sure > > there is any relation to the hardware... > > > > Did someone have the chance of a quick look and seen something odd at > all in > > the backtraces? Anything that can point me in the right direction > > would appreciated... > > > > Best, > > > > Wesley Akio > > TuntsCorp.com > > > > > > > > On Wed, May 16, 2012 at 11:00 AM, Wesley Akio > > wrote: > >> > >> Hi all, > >> > >> I'm experiencing several core dumps a day in one of our boxes. > >> > >> The problem was happening with FS trunk from Apr 30. Updated yesterday > and > >> the problem continues. > >> > >> I suspect it has something to do with the GSM hardware(Khomp) in use but > >> I'm not sure... > >> > >> If someone could take a look at the backtraces I would be very > thankfull. > >> > >> http://pastebin.freeswitch.org/19059 > >> FreeSWITCH Version 1.2.0 (git-d015395 2012-05-15 11-26-16 -0500) > >> > >> Best, > >> > >> Wesley Akio > >> TuntsCorp.com > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120518/7ea8135e/attachment-0001.html From msc at freeswitch.org Fri May 18 20:37:05 2012 From: msc at freeswitch.org (Michael Collins) Date: Fri, 18 May 2012 09:37:05 -0700 Subject: [Freeswitch-users] set dtmf type without sip profile In-Reply-To: References: Message-ID: If this is a vanilla install then the internal.xml profile will be listening on port 5060. Assuming that's where the calls come in from voxbone then what you've set is correct. I would verify your assertions: is voxbone really sending DTMFs via 2833? If so then they should appear in your tshark captures. I would double-check, and if indeed you are not seeing those 2833 packets from voxbone then I would call their support and ask them to verify that they really are set up for 2833. Also, just for confirmation, I would listen to see if the DTMFs come inband and also look in the SIP traffic to see if maybe they are doing something silly like using SIP INFO packets to send DTMF. -MC On Fri, May 18, 2012 at 5:01 AM, Anita Hall wrote: > Hi > > I am taking DID from voxbone.com, they do not need authentication by > login and password, so I have not created a SIP profile. By allowing their > IP addresses in acl.conf.xml, I am able to receive incoming calls and > media. However, dtmf is not coming. Doing tshark, no RTP event > corresponding to dtmf is being received. > > Voxbone sends DTMF over rfc2833 > > I have set in conf/sip_profiles/internal.xml and done reload mod_sofia > > > But I am not sure if this is the correct place. Is there any other pace > where dtmf type can be specified in the absence of a sip profile ? > > regards, > Anita > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120518/56d0ea4b/attachment.html From saami_mh at ymail.com Fri May 18 20:51:39 2012 From: saami_mh at ymail.com (Samira Mh) Date: Fri, 18 May 2012 09:51:39 -0700 (PDT) Subject: [Freeswitch-users] how to fix : RECOVERY_ON_TIMER_EXPIRE?? In-Reply-To: References: <1337343353.86641.YahooMailNeo@web120106.mail.ne1.yahoo.com> <4FB64C65.80900@gmail.com> <1337344543.52305.YahooMailNeo@web120104.mail.ne1.yahoo.com> Message-ID: <1337359899.50425.YahooMailNeo@web120104.mail.ne1.yahoo.com> hi guys, my problem was solved by enabling "sofia profile external siptrace on " and i found that the ?my phone number dialled matched ?in "01_exapmle.com"? in the path " /usr/local/freeswitch/conf/dialplan/default/? " instead of used "01_myprovider.com" on the same path; so called : 0912******@example.com insted of 0912******@myprovider.com ; and disabled ? in 01_example.com so my problem solved because myphone number matched the condition?^(\d{11})$"? in 01_example.com instead of ?01_myprovider.com; any way,thanks alot ? for your help and replies ; ________________________________ From: Mitch Capper To: FreeSWITCH Users Help Sent: Friday, May 18, 2012 7:04 PM Subject: Re: [Freeswitch-users] how to fix : RECOVERY_ON_TIMER_EXPIRE?? Did you try a no-nat situation?? Did you try another carrier? Both of those are suggested in the wiki as the sources of the problem. ~Mitch On Fri, May 18, 2012 at 5:35 AM, Samira Mh wrote: > i read the link but i could't solve the problem > > ________________________________ > From: ghallab > To: FreeSWITCH Users Help > Sent: Friday, May 18, 2012 5:49 PM > Subject: Re: [Freeswitch-users] how to fix : RECOVERY_ON_TIMER_EXPIRE?? > > ?At this link you will find the reason why this is happens: > http://wiki.freeswitch.org/wiki/Hangup_causes > > On 05/18/2012 12:15 PM, Samira Mh wrote: > > hi guys, > every time i want to make call via gateway on freeswitch the following error > occured: > > 2012-05-18 16:17:16.296384 [INFO] mod_dptools.c:2355 Originate Failed. > ?Cause: RECOVERY_ON_TIMER_EXPIRE > ?plz help,thanks > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120518/643454fe/attachment.html From msc at freeswitch.org Fri May 18 21:38:10 2012 From: msc at freeswitch.org (Michael Collins) Date: Fri, 18 May 2012 10:38:10 -0700 Subject: [Freeswitch-users] One core at 100% usage In-Reply-To: References: <4B4CF310-EC50-46A9-8406-F043E0BFA4B2@visionutveckling.se> Message-ID: FYI, I've added this information to the "Reporting Bugs" page. Everyone please be ready to point those with "my CPU is at 100%" questions to this section: http://wiki.freeswitch.org/wiki/Reporting_Bugs#CPU_Usage -MC On Fri, May 18, 2012 at 8:04 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > update to latest > reproduce > > cmd: top -H `cat /usr/local/freeswitch/run/freeswitch.pid` > find the process that is 100% and note its pid > cmd: gcore `cat /usr/local/freeswitch/run/freeswitch.pid` > cmd: gdb `which freeswitch` > enter: "info threads" > look for the one with the same pid as the one you noted above > enter: thread > enter: bt > > but you should be using jira, I have to say this 200 times a week. > > > On Fri, May 18, 2012 at 5:10 AM, Peter Olsson > wrote: > > If this is still reproducable on latest git, please file a Jira and > supply as much details as possible. At least you will need to upload debug > log, and a stack trace when this happens. > > > > /Peter > > > > 17 maj 2012 kl. 22:41 skrev "Stephen Wilde" : > > > >> I have done a git pull (now git-83e090c 2012-05-17 16-40-30 +0000) and > I have a strange load distribution over cores: one core at 100%. > >> > >> Here the htop: > >> > >> > >> 1 > [|||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||||100.0%] > >> 2 [||||||||||||||||| > 19.1%] > >> 3 [|| > 1.3%] > >> 4 [|| > 1.3%] > >> 5 [ > 0.0%] > >> 6 [ > 0.0%] > >> 7 [|||| > 3.9%] > >> 8 [||||||||||||| > 13.2%] > >> 9 [ > 0.0%] > >> 10 [||||||||||||||||| > 18.3%] > >> 11 [||||| > 4.6%] > >> 12 [|||||||||||||||| > 17.0%] > >> 13 [|||||||| > 8.6%] > >> 14 [|||||||||||| > 12.5%] > >> 15 [|||||||| > 8.6%] > >> 16 [||||||||||||||||||| > 21.1%] > >> > >> > >> After revert to an old git I have this htop: > >> > >> 1 [||||||| > 7.2%] > >> 2 [|||||||||||| > 11.8%] > >> 3 [||||| > 5.3%] > >> 4 [||||||| > 6.6%] > >> 5 [|||| > 3.3%] > >> 6 [||||||||| > 9.0%] > >> 7 [||||| > 4.6%] > >> 8 [|||||||||||||| > 15.1%] > >> 9 [||||||| > 6.5%] > >> 10 [||||||||| > 9.3%] > >> 11 [|||||| > 6.5%] > >> 12 [|||||||||||||| > 14.9%] > >> 13 [|||||||| > 7.8%] > >> 14 [|||||||||||||||| > 17.0%] > >> 15 [|||||||| > 7.3%] > >> 16 [||||||||||| > 11.8%] > >> > >> anyone with same issue? > >> !DSPAM:4fb55f8732762045212077! > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120518/0e19c2c8/attachment-0001.html From anthony.minessale at gmail.com Fri May 18 21:40:51 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 18 May 2012 12:40:51 -0500 Subject: [Freeswitch-users] FS Core Dump using GSM Hardware In-Reply-To: References: Message-ID: You should collect samples of distinct cores and post them. Also you should be doing this on jira not here. I have already told one person this today. Always report crashes and bugs to http://jira.freeswitch.org On Fri, May 18, 2012 at 11:32 AM, Wesley Akio wrote: > Thank you very much for the reply Anthony! > > Although I have almost 30 core dumps in my hands(from the last 3 days only) > that message appears in the logs only twice(maybe it crashed before writing > to log?) > > The times are consistent with crashes thought... > > freeswitch.log.2012-05-17-14-25-41.1-2012-05-17 13:54:17.996544 [DEBUG] > switch_core_state_machine.c:685 (Khomp/0/14/s) State REPORTING going to > sleep > freeswitch.log.2012-05-17-14-25-41.1:2012-05-17 13:54:17.996544 [WARNING] > switch_core_state_machine.c:410 (Khomp/0/14/s) Invalid State Change > CS_EXECUTE -> CS_DESTROY > -- > freeswitch.log.2012-05-17-17-14-38.1-2012-05-17 16:38:28.697842 [DEBUG] > switch_core_state_machine.c:685 (Khomp/0/10/s) State REPORTING going to > sleep > freeswitch.log.2012-05-17-17-14-38.1:2012-05-17 16:38:28.697842 [WARNING] > switch_core_state_machine.c:410 (Khomp/0/10/s) Invalid State Change > CS_EXECUTE -> CS_DESTROY > > I've changed the log configuration to add the uuid in order to sort the logs > for the lifetime of the call. > > Best, > > Wesley Akio > TuntsCorp.com > > > > On Fri, May 18, 2012 at 1:11 PM, Anthony Minessale > wrote: >> >> Do you have the logs on your box? >> look in freeswitch.log for Invalid State Change this log line should >> spell out exactly where the bug is in mod_khomp >> >> On Fri, May 18, 2012 at 8:10 AM, Wesley Akio >> wrote: >> > Hi all, >> > >> > The manufacturer is giving me support on this one but they are not sure >> > there is any relation to the hardware... >> > >> > Did someone have the chance of a quick look and seen something odd at >> > all in >> > the backtraces? Anything that can point me in the right direction >> > would?appreciated... >> > >> > Best, >> > >> > Wesley Akio >> > TuntsCorp.com >> > >> > >> > >> > On Wed, May 16, 2012 at 11:00 AM, Wesley Akio >> > wrote: >> >> >> >> Hi all, >> >> >> >> I'm experiencing several core dumps a day in one of our boxes. >> >> >> >> The problem was happening with FS trunk from Apr 30. Updated yesterday >> >> and >> >> the problem continues. >> >> >> >> I suspect it has something to do with the GSM hardware(Khomp) in use >> >> but >> >> I'm not sure... >> >> >> >> If someone could take a look at the backtraces I would be very >> >> thankfull. >> >> >> >> http://pastebin.freeswitch.org/19059 >> >> FreeSWITCH Version 1.2.0 (git-d015395 2012-05-15 11-26-16 -0500) >> >> >> >> Best, >> >> >> >> Wesley Akio >> >> TuntsCorp.com >> > >> > >> > >> > >> > _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > Join Us At ClueCon - Aug 7-9, 2012 >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From anthony.minessale at gmail.com Fri May 18 21:42:14 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 18 May 2012 12:42:14 -0500 Subject: [Freeswitch-users] FS Core Dump using GSM Hardware In-Reply-To: References: Message-ID: Also, I forgot. You should clean out your FS distro just to make sure you do not have build skew. rm /usr/local/freeswitch/lib/* rm /usr/local/freeswitch/bin/* rm /usr/local/freeswitch/mod/* make current On Fri, May 18, 2012 at 12:40 PM, Anthony Minessale wrote: > You should collect samples of distinct cores and post them. > Also you should be doing this on jira not here. ?I have already told > one person this today. > Always report crashes and bugs to http://jira.freeswitch.org > > On Fri, May 18, 2012 at 11:32 AM, Wesley Akio wrote: >> Thank you very much for the reply Anthony! >> >> Although I have almost 30 core dumps in my hands(from the last 3 days only) >> that message appears in the logs only twice(maybe it crashed before writing >> to log?) >> >> The times are consistent with crashes thought... >> >> freeswitch.log.2012-05-17-14-25-41.1-2012-05-17 13:54:17.996544 [DEBUG] >> switch_core_state_machine.c:685 (Khomp/0/14/s) State REPORTING going to >> sleep >> freeswitch.log.2012-05-17-14-25-41.1:2012-05-17 13:54:17.996544 [WARNING] >> switch_core_state_machine.c:410 (Khomp/0/14/s) Invalid State Change >> CS_EXECUTE -> CS_DESTROY >> -- >> freeswitch.log.2012-05-17-17-14-38.1-2012-05-17 16:38:28.697842 [DEBUG] >> switch_core_state_machine.c:685 (Khomp/0/10/s) State REPORTING going to >> sleep >> freeswitch.log.2012-05-17-17-14-38.1:2012-05-17 16:38:28.697842 [WARNING] >> switch_core_state_machine.c:410 (Khomp/0/10/s) Invalid State Change >> CS_EXECUTE -> CS_DESTROY >> >> I've changed the log configuration to add the uuid in order to sort the logs >> for the lifetime of the call. >> >> Best, >> >> Wesley Akio >> TuntsCorp.com >> >> >> >> On Fri, May 18, 2012 at 1:11 PM, Anthony Minessale >> wrote: >>> >>> Do you have the logs on your box? >>> look in freeswitch.log for Invalid State Change this log line should >>> spell out exactly where the bug is in mod_khomp >>> >>> On Fri, May 18, 2012 at 8:10 AM, Wesley Akio >>> wrote: >>> > Hi all, >>> > >>> > The manufacturer is giving me support on this one but they are not sure >>> > there is any relation to the hardware... >>> > >>> > Did someone have the chance of a quick look and seen something odd at >>> > all in >>> > the backtraces? Anything that can point me in the right direction >>> > would?appreciated... >>> > >>> > Best, >>> > >>> > Wesley Akio >>> > TuntsCorp.com >>> > >>> > >>> > >>> > On Wed, May 16, 2012 at 11:00 AM, Wesley Akio >>> > wrote: >>> >> >>> >> Hi all, >>> >> >>> >> I'm experiencing several core dumps a day in one of our boxes. >>> >> >>> >> The problem was happening with FS trunk from Apr 30. Updated yesterday >>> >> and >>> >> the problem continues. >>> >> >>> >> I suspect it has something to do with the GSM hardware(Khomp) in use >>> >> but >>> >> I'm not sure... >>> >> >>> >> If someone could take a look at the backtraces I would be very >>> >> thankfull. >>> >> >>> >> http://pastebin.freeswitch.org/19059 >>> >> FreeSWITCH Version 1.2.0 (git-d015395 2012-05-15 11-26-16 -0500) >>> >> >>> >> Best, >>> >> >>> >> Wesley Akio >>> >> TuntsCorp.com >>> > >>> > >>> > >>> > >>> > _________________________________________________________________________ >>> > Professional FreeSWITCH Consulting Services: >>> > consulting at freeswitch.org >>> > http://www.freeswitchsolutions.com >>> > >>> > >>> > >>> > >>> > Official FreeSWITCH Sites >>> > http://www.freeswitch.org >>> > http://wiki.freeswitch.org >>> > http://www.cluecon.com >>> > >>> > Join Us At ClueCon - Aug 7-9, 2012 >>> > >>> > FreeSWITCH-users mailing list >>> > FreeSWITCH-users at lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.org >>> > >>> >>> >>> >>> -- >>> Anthony Minessale II >>> >>> FreeSWITCH http://www.freeswitch.org/ >>> ClueCon http://www.cluecon.com/ >>> Twitter: http://twitter.com/FreeSWITCH_wire >>> >>> AIM: anthm >>> MSN:anthony_minessale at hotmail.com >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>> IRC: irc.freenode.net #freeswitch >>> >>> FreeSWITCH Developer Conference >>> sip:888 at conference.freeswitch.org >>> googletalk:conf+888 at conference.freeswitch.org >>> pstn:+19193869900 >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> Join Us At ClueCon - Aug 7-9, 2012 >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From nbhatti at gmail.com Fri May 18 21:45:44 2012 From: nbhatti at gmail.com (Muhammad Naseer Bhatti) Date: Fri, 18 May 2012 20:45:44 +0300 Subject: [Freeswitch-users] Stuck calls Message-ID: I got like 46 calls stuck in FreeSWITCH. The switch is not doing any traffic right now. It is running FreeSWITCH Version 1.2.0 (git-0709cc6 2012-05-16 02-50-13 +0000). Should I file jira or look for something else? They are mostly in CS_INIT,,,,,,,RINGING -- CS_ROUTING,,,,,,,DOWN and CS_CONSUME_MEDIA state. Thanks. From anthony.minessale at gmail.com Fri May 18 21:52:10 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 18 May 2012 12:52:10 -0500 Subject: [Freeswitch-users] Stuck calls In-Reply-To: References: Message-ID: compare "show channels" to "status" if status says there are 0 calls, look for a db error somewhere in fs log [or db specific log if you have odbc] cat freeswitch.log | grep CRIT if status sees channels too, see if you have a cdr module that is doing some post processing that could be stuck. On Fri, May 18, 2012 at 12:45 PM, Muhammad Naseer Bhatti wrote: > I got like 46 calls stuck in FreeSWITCH. The switch is not doing any > traffic right now. It is running FreeSWITCH Version 1.2.0 (git-0709cc6 > 2012-05-16 02-50-13 +0000). Should I file jira or look for something > else? They are mostly in CS_INIT,,,,,,,RINGING -- > CS_ROUTING,,,,,,,DOWN and CS_CONSUME_MEDIA state. > > Thanks. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From nbhatti at gmail.com Fri May 18 23:18:04 2012 From: nbhatti at gmail.com (Muhammad Naseer Bhatti) Date: Fri, 18 May 2012 22:18:04 +0300 Subject: [Freeswitch-users] Stuck calls In-Reply-To: References: Message-ID: Show channels shows 46 calls, status shows 0. pb @ http://pastebin.freeswitch.org/19093 Logging was turned of from this production server to gain some disk I/Os. Using mod_xml_cdr for cdr posting. Web server could possibly the reason and if so, why the calls are in RINGING state? CDR is only posted at the end of the call. Looks like we don't have much evidence. Goni On Fri, May 18, 2012 at 8:52 PM, Anthony Minessale wrote: > compare "show channels" to "status" if status says there are 0 calls, > look for a db error somewhere in fs log [or db specific log if you > have odbc] > cat freeswitch.log | grep CRIT > > if status sees channels too, see if you have a cdr module that is > doing some post processing that could be stuck. > > > > > On Fri, May 18, 2012 at 12:45 PM, Muhammad Naseer Bhatti > wrote: >> I got like 46 calls stuck in FreeSWITCH. The switch is not doing any >> traffic right now. It is running FreeSWITCH Version 1.2.0 (git-0709cc6 >> 2012-05-16 02-50-13 +0000). Should I file jira or look for something >> else? They are mostly in CS_INIT,,,,,,,RINGING -- >> CS_ROUTING,,,,,,,DOWN and CS_CONSUME_MEDIA state. >> >> Thanks. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From msc at freeswitch.org Fri May 18 23:50:15 2012 From: msc at freeswitch.org (Michael Collins) Date: Fri, 18 May 2012 12:50:15 -0700 Subject: [Freeswitch-users] Stuck calls In-Reply-To: References: Message-ID: Can you turn debugging back on for a period of time and re-test? Also, any chance you can get to latest git for testing? -MC On Fri, May 18, 2012 at 12:18 PM, Muhammad Naseer Bhatti wrote: > Show channels shows 46 calls, status shows 0. pb @ > http://pastebin.freeswitch.org/19093 > Logging was turned of from this production server to gain some disk > I/Os. Using mod_xml_cdr for cdr posting. Web server could possibly the > reason and if so, why the calls are in RINGING state? CDR is only > posted at the end of the call. Looks like we don't have much evidence. > > Goni > > On Fri, May 18, 2012 at 8:52 PM, Anthony Minessale > wrote: > > compare "show channels" to "status" if status says there are 0 calls, > > look for a db error somewhere in fs log [or db specific log if you > > have odbc] > > cat freeswitch.log | grep CRIT > > > > if status sees channels too, see if you have a cdr module that is > > doing some post processing that could be stuck. > > > > > > > > > > On Fri, May 18, 2012 at 12:45 PM, Muhammad Naseer Bhatti > > wrote: > >> I got like 46 calls stuck in FreeSWITCH. The switch is not doing any > >> traffic right now. It is running FreeSWITCH Version 1.2.0 (git-0709cc6 > >> 2012-05-16 02-50-13 +0000). Should I file jira or look for something > >> else? They are mostly in CS_INIT,,,,,,,RINGING -- > >> CS_ROUTING,,,,,,,DOWN and CS_CONSUME_MEDIA state. > >> > >> Thanks. > >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120518/abf94b1a/attachment-0001.html From nbhatti at gmail.com Sat May 19 00:01:38 2012 From: nbhatti at gmail.com (Muhammad Naseer Bhatti) Date: Fri, 18 May 2012 23:01:38 +0300 Subject: [Freeswitch-users] Stuck calls In-Reply-To: References: Message-ID: I was trying to dig some information out of it before a restart. There is no traffic on this box right now, so I can restart/retest anything. I'll update, grab some logs and will post back. Thanks. On Fri, May 18, 2012 at 10:50 PM, Michael Collins wrote: > Can you turn debugging back on for a period of time and re-test? Also, any > chance you can get to latest git for testing? > -MC > > > On Fri, May 18, 2012 at 12:18 PM, Muhammad Naseer Bhatti > wrote: >> >> Show channels shows 46 calls, status shows 0. pb @ >> http://pastebin.freeswitch.org/19093 >> Logging was turned of from this production server to gain some disk >> I/Os. Using mod_xml_cdr for cdr posting. Web server could possibly the >> reason and ?if so, why the calls are in RINGING state? CDR is only >> posted at the end of the call. Looks like we don't have much evidence. >> >> Goni >> >> On Fri, May 18, 2012 at 8:52 PM, Anthony Minessale >> wrote: >> > compare "show channels" to "status" if status says there are 0 calls, >> > look for a db error somewhere in fs log [or db specific log if you >> > have odbc] >> > cat freeswitch.log | grep CRIT >> > >> > if status sees channels too, see if you have a cdr module that is >> > doing some post processing that could be stuck. >> > >> > >> > >> > >> > On Fri, May 18, 2012 at 12:45 PM, Muhammad Naseer Bhatti >> > wrote: >> >> I got like 46 calls stuck in FreeSWITCH. The switch is not doing any >> >> traffic right now. It is running FreeSWITCH Version 1.2.0 (git-0709cc6 >> >> 2012-05-16 02-50-13 +0000). Should I file jira or look for something >> >> else? They are mostly in CS_INIT,,,,,,,RINGING -- >> >> CS_ROUTING,,,,,,,DOWN and CS_CONSUME_MEDIA state. >> >> >> >> Thanks. >> >> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From slava at tangramltd.com Sat May 19 00:15:57 2012 From: slava at tangramltd.com (Dubrovskiy Viacheslav) Date: Fri, 18 May 2012 16:15:57 -0400 Subject: [Freeswitch-users] Calls to the external phone Message-ID: <4FB6ADFD.5030801@tangramltd.com> Hi. Using the default configuration. There are internal and external real IP. I have several registered account from the internal network. I want to connect remote user from Ethernet. For this added external real IP to the external sip profile (param name="sip-ip" and param name="rtp-ip") and copied the directory/default.xml in directory/externa.xml which changed the Now remote users can register and make calls. But there's a problem: No one can call to this external user. Tell me, what am I doing wrong? Thank you -- WBR, Dubrovskiy Viacheslav From anthony.minessale at gmail.com Sat May 19 00:42:13 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 18 May 2012 15:42:13 -0500 Subject: [Freeswitch-users] Stuck calls In-Reply-To: References: Message-ID: That spells out SQL error. The show channels is just a db keeping track of the channel status based on events. If you can reproduce it on git HEAD open a jira On Fri, May 18, 2012 at 3:01 PM, Muhammad Naseer Bhatti wrote: > I was trying to dig some information out of it before a restart. There > is no traffic on this box right now, so I can restart/retest anything. > I'll update, grab some logs and will post back. > > Thanks. > > On Fri, May 18, 2012 at 10:50 PM, Michael Collins wrote: >> Can you turn debugging back on for a period of time and re-test? Also, any >> chance you can get to latest git for testing? >> -MC >> >> >> On Fri, May 18, 2012 at 12:18 PM, Muhammad Naseer Bhatti >> wrote: >>> >>> Show channels shows 46 calls, status shows 0. pb @ >>> http://pastebin.freeswitch.org/19093 >>> Logging was turned of from this production server to gain some disk >>> I/Os. Using mod_xml_cdr for cdr posting. Web server could possibly the >>> reason and ?if so, why the calls are in RINGING state? CDR is only >>> posted at the end of the call. Looks like we don't have much evidence. >>> >>> Goni >>> >>> On Fri, May 18, 2012 at 8:52 PM, Anthony Minessale >>> wrote: >>> > compare "show channels" to "status" if status says there are 0 calls, >>> > look for a db error somewhere in fs log [or db specific log if you >>> > have odbc] >>> > cat freeswitch.log | grep CRIT >>> > >>> > if status sees channels too, see if you have a cdr module that is >>> > doing some post processing that could be stuck. >>> > >>> > >>> > >>> > >>> > On Fri, May 18, 2012 at 12:45 PM, Muhammad Naseer Bhatti >>> > wrote: >>> >> I got like 46 calls stuck in FreeSWITCH. The switch is not doing any >>> >> traffic right now. It is running FreeSWITCH Version 1.2.0 (git-0709cc6 >>> >> 2012-05-16 02-50-13 +0000). Should I file jira or look for something >>> >> else? They are mostly in CS_INIT,,,,,,,RINGING -- >>> >> CS_ROUTING,,,,,,,DOWN and CS_CONSUME_MEDIA state. >>> >> >>> >> Thanks. >>> >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From nickolayr at gmail.com Sat May 19 00:51:18 2012 From: nickolayr at gmail.com (Nikolay Rogoshchenkov) Date: Fri, 18 May 2012 16:51:18 -0400 Subject: [Freeswitch-users] Calls to the external phone In-Reply-To: <4FB6ADFD.5030801@tangramltd.com> References: <4FB6ADFD.5030801@tangramltd.com> Message-ID: Could you show >sofia status and what did you see in logs? -- Nikolay On Fri, May 18, 2012 at 4:15 PM, Dubrovskiy Viacheslav wrote: > Hi. > > Using the default configuration. There are internal and external real IP. > I have several registered account from the internal network. I want to > connect remote user from Ethernet. > For this added external real IP to the external sip profile (param > name="sip-ip" and param name="rtp-ip") and copied the > directory/default.xml in directory/externa.xml which changed the name="external_IP"> > > Now remote users can register and make calls. But there's a problem: No > one can call to this external user. > > Tell me, what am I doing wrong? > Thank you > > -- > WBR, > Dubrovskiy Viacheslav > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120518/4fc12e04/attachment.html From freeswitch at zoho.com Sat May 19 01:32:29 2012 From: freeswitch at zoho.com (dingdong) Date: Fri, 18 May 2012 14:32:29 -0700 Subject: [Freeswitch-users] correct way to send out dtmf type in gateway Message-ID: <13761ddb6c4.3940845884435026170.7401235021465964981@zoho.com> Hello all, what is the correct way to send out a dtmf type other than rcf2833? via a gateway? i have this telepacific number that goes to IVR when called,it doesn't accept any options pressed when i call from a sip ipphone(flowroute) it will start working though when start_dtmf_generate is started. dp <extension name="telepacific" > <condition field="destination_number" expression="^(18774878349)$" > <action application="start_dtmf_generate" /> <action application="bridge" data="sofia/gateway/Flowroute/$1" /> </condition> </extension> how do i exactly set a dtmf type? ding -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120518/04391b80/attachment.html From mel0torme at gmail.com Sat May 19 03:36:04 2012 From: mel0torme at gmail.com (Tom C) Date: Fri, 18 May 2012 16:36:04 -0700 Subject: [Freeswitch-users] What is the digit for in ./bootstrap.sh -j4 ? Message-ID: On the Installation Guide page of the FreeSwitch wiki, it says the following: ------------------------- You can use multiple cores for your bootstrap/config/build, by specifying it at the start, e.g.: (note this may make build errors harder to spot) ./bootstrap.sh -j4 -------------------------- I looked through the bootstrap.sh code, and it appears that the trailing digit in the above example is ignored? Or am I missing something? I was thinking/hoping it was a way to specify the maximum # of background processes that get created. At this point I assume it is just an error in the wiki. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120518/da8ed65a/attachment.html From slava at tangramltd.com Sat May 19 03:39:33 2012 From: slava at tangramltd.com (Dubrovskiy Viacheslav) Date: Fri, 18 May 2012 19:39:33 -0400 Subject: [Freeswitch-users] Calls to the external phone In-Reply-To: References: <4FB6ADFD.5030801@tangramltd.com> Message-ID: <4FB6DDB5.4030104@tangramltd.com> 18.05.2012 16:51, Nikolay Rogoshchenkov ???????: > Could you show > >sofia status freeswitch at internal> sofia status Name Type Data State ================================================================================================= internal profile sip:mod_sofia at 192.168.1.120:5060 RUNNING (0) host alias internal ALIASED internal-ipv6 profile sip:mod_sofia@[::1]:5060 RUNNING (0) external profile sip:mod_sofia at 195.211.131.140:5080 RUNNING (0) external::gold gateway sip:joeuser at gold NOREG ================================================================================================= 3 profiles 1 alias > > and what did you see in logs? I think I understand. Do not need that external users connected through an external profile (port 5080). Is necessary to add another internal profile (to the external IP 195.211.131.140 port 5060) and that the two internal and internal profile 2 (port 5060) were in the same context and domain. Is it possible? > > -- > Nikolay > > > On Fri, May 18, 2012 at 4:15 PM, Dubrovskiy Viacheslav > > wrote: > > Hi. > > Using the default configuration. There are internal and external > real IP. > I have several registered account from the internal network. I want to > connect remote user from Ethernet. > For this added external real IP to the external sip profile (param > name="sip-ip" and param name="rtp-ip") and copied the > directory/default.xml in directory/externa.xml which changed the > name="external_IP"> > > Now remote users can register and make calls. But there's a > problem: No > one can call to this external user. > > Tell me, what am I doing wrong? > Thank you > > -- > WBR, > Dubrovskiy Viacheslav > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- WBR, Dubrovskiy Viacheslav -------------- next part -------------- An HTML attachment was scrubbed... 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S/MIME Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120518/d819e1ab/attachment-0001.bin From djbinter at gmail.com Sat May 19 03:40:07 2012 From: djbinter at gmail.com (DJB International) Date: Fri, 18 May 2012 16:40:07 -0700 Subject: [Freeswitch-users] correct way to send out dtmf type in gateway In-Reply-To: <13761ddb6c4.3940845884435026170.7401235021465964981@zoho.com> References: <13761ddb6c4.3940845884435026170.7401235021465964981@zoho.com> Message-ID: http://wiki.freeswitch.org/wiki/Sofia.conf.xml#DTMF -djbinter On Fri, May 18, 2012 at 2:32 PM, dingdong wrote: > ** > Hello all, > what is the correct way to send out a dtmf type other than rcf2833? via a > gateway? i have this telepacific number that goes to IVR when called,it > doesn't accept any options pressed when i call from a sip > ipphone(flowroute) it will start working though when start_dtmf_generate is > started. > > dp > > > > > > > > > > how do i exactly set a dtmf type? > > > ding > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120518/7e886815/attachment.html From slava at tangramltd.com Sat May 19 04:05:03 2012 From: slava at tangramltd.com (Dubrovskiy Viacheslav) Date: Fri, 18 May 2012 20:05:03 -0400 Subject: [Freeswitch-users] Calls to the external phone In-Reply-To: <4FB6DDB5.4030104@tangramltd.com> References: <4FB6ADFD.5030801@tangramltd.com> <4FB6DDB5.4030104@tangramltd.com> Message-ID: <4FB6E3AF.4020906@tangramltd.com> 18.05.2012 19:39, Dubrovskiy Viacheslav ???????: > I think I understand. Do not need that external users connected > through an external profile (port 5080). Is necessary to add another > internal profile (to the external IP 195.211.131.140 port 5060) and > that the two internal and internal profile 2 (port 5060) were in the > same context and domain. > > Is it possible? I found a solution. Just added to the unternal profile And internal profile begin listen external IP. Thank you. -- WBR, Dubrovskiy Viacheslav -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120518/caf563c3/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/pkcs7-signature Size: 4916 bytes Desc: ?????????????????????????????????? ?????????????? S/MIME Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120518/caf563c3/attachment.bin From jkomar at jbox.ca Sat May 19 04:27:10 2012 From: jkomar at jbox.ca (Komar, Jason) Date: Fri, 18 May 2012 18:27:10 -0600 Subject: [Freeswitch-users] What is the digit for in ./bootstrap.sh -j4 ? In-Reply-To: References: Message-ID: The -j4 is a compiler flag, so it probably just passes through to the compiler without being touched by the shell script. On May 18, 2012 5:36 PM, "Tom C" wrote: > On the Installation Guide page of the FreeSwitch wiki, it says the > following: > > ------------------------- > > You can use multiple cores for your bootstrap/config/build, by specifying > it at the start, e.g.: (note this may make build errors harder to spot) > > > ./bootstrap.sh -j4 > > -------------------------- > > > I looked through the bootstrap.sh code, and it appears that the trailing > digit in the above example is ignored? Or am I missing something? > > I was thinking/hoping it was a way to specify the maximum # of background > processes that get created. At this point I assume it is just an error in > the wiki. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120518/24b18910/attachment.html From chris at opencsta.org Sat May 19 04:53:16 2012 From: chris at opencsta.org (Chris Mylonas) Date: Sat, 19 May 2012 10:53:16 +1000 Subject: [Freeswitch-users] What is the digit for in ./bootstrap.sh -j4 ? In-Reply-To: References: Message-ID: if it's not used now, it may have been in the past - someone here that has been around for a few years may recall and probably why the wiki page says (note this may make build errors harder to spot) and it was removed because of that reason. either way, just to clarify, here's my 2cents use it with 'make' command e.g. make -j4 will use 4 cores for compiling if there is a make command in bootstrap.sh it may be passed there. configure and bootstrap are just scripts to organise things. I haven't looked inside them whether either calls 'make'. HTH Chris On 19/05/2012, at 10:27 AM, Komar, Jason wrote: > The -j4 is a compiler flag, so it probably just passes through to the compiler without being touched by the shell script. > > On May 18, 2012 5:36 PM, "Tom C" wrote: > On the Installation Guide page of the FreeSwitch wiki, it says the following: > > ------------------------- > You can use multiple cores for your bootstrap/config/build, by specifying it at the start, e.g.: (note this may make build errors harder to spot) > > ./bootstrap.sh -j4 > -------------------------- > > > I looked through the bootstrap.sh code, and it appears that the trailing digit in the above example is ignored? Or am I missing something? > > I was thinking/hoping it was a way to specify the maximum # of background processes that get created. At this point I assume it is just an error in the wiki. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120519/4738dea7/attachment-0001.html From krice at freeswitch.org Sat May 19 05:20:39 2012 From: krice at freeswitch.org (Ken Rice) Date: Fri, 18 May 2012 20:20:39 -0500 Subject: [Freeswitch-users] What is the digit for in ./bootstrap.sh -j4 ? In-Reply-To: Message-ID: The digit on the bootstrap doesn?t do anything, its ignored... The ?j says parallelize the bootstrap... This will spawn off a pile of background shell scripts to make the bootstrap go faster. K On 5/18/12 6:36 PM, "Tom C" wrote: > On the Installation Guide page of the FreeSwitch wiki, it says the following: > > ------------------------- > You can use multiple cores for your bootstrap/config/build, by specifying it > at the start, e.g.: (note this may make build errors harder to spot) > > ./bootstrap.sh -j4 > -------------------------- > > > I looked through the bootstrap.sh code, and it appears that the trailing digit > in the above example is ignored? ?Or am I missing something? > > I was thinking/hoping it was a way to specify the maximum # of background > processes that get created. ?At this point I assume it is just an error in the > wiki. > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120518/f944d788/attachment.html From krice at freeswitch.org Sat May 19 05:21:43 2012 From: krice at freeswitch.org (Ken Rice) Date: Fri, 18 May 2012 20:21:43 -0500 Subject: [Freeswitch-users] What is the digit for in ./bootstrap.sh -j4 ? In-Reply-To: Message-ID: Make ?j works in most cases... If there are errors we would like to know about them.... As I said in the earlier reply tho, ./bootstrap.sh ?j ignores any numbers after the ?j K On 5/18/12 7:53 PM, "Chris Mylonas" wrote: > if it's not used now, it may have been in the past - someone here that has > been around for a few years may recall and probably why the wiki page says > (note this may make build errors harder to spot) and it was removed because of > that reason. > > > > either way, just to clarify, here's my 2cents > > use it with 'make' command > e.g. > > make -j4 > > will use 4 cores for compiling > if there is a make command in bootstrap.sh it may be passed there. > configure and bootstrap are just scripts to organise things. I haven't looked > inside them whether either calls 'make'. > > HTH > Chris > > On 19/05/2012, at 10:27 AM, Komar, Jason wrote: > >> >> The -j4 is a compiler flag, so it probably just passes through to the >> compiler without being touched by the shell script. >> >> On May 18, 2012 5:36 PM, "Tom C" wrote: >>> On the Installation Guide page of the FreeSwitch wiki, it says the >>> following: >>> >>> ------------------------- >>> You can use multiple cores for your bootstrap/config/build, by specifying it >>> at the start, e.g.: (note this may make build errors harder to spot) >>> >>> ./bootstrap.sh -j4 >>> -------------------------- >>> >>> >>> I looked through the bootstrap.sh code, and it appears that the trailing >>> digit in the above example is ignored? Or am I missing something? >>> >>> I was thinking/hoping it was a way to specify the maximum # of background >>> processes that get created. At this point I assume it is just an error in >>> the wiki. >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> Join Us At ClueCon - Aug 7-9, 2012 >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120518/ff1a4479/attachment.html From nickolayr at gmail.com Sat May 19 05:54:46 2012 From: nickolayr at gmail.com (Nikolay Rogoshchenkov) Date: Fri, 18 May 2012 21:54:46 -0400 Subject: [Freeswitch-users] Calls to the external phone In-Reply-To: <4FB6DDB5.4030104@tangramltd.com> References: <4FB6ADFD.5030801@tangramltd.com> <4FB6DDB5.4030104@tangramltd.com> Message-ID: If needed, you can change the default external port in vars.xml (data="external_sip_port=5080") -- Nikolay On Fri, May 18, 2012 at 7:39 PM, Dubrovskiy Viacheslav wrote: > 18.05.2012 16:51, Nikolay Rogoshchenkov ???????: > > Could you show > >sofia status > > freeswitch at internal> sofia status > Name > Type Data State > > ================================================================================================= > internal profile > sip:mod_sofia at 192.168.1.120:5060 RUNNING (0) > host alias > internal ALIASED > internal-ipv6 profile sip:mod_sofia@[::1]:5060 > RUNNING (0) > external profile > sip:mod_sofia at 195.211.131.140:5080 RUNNING (0) > external::gold gateway > sip:joeuser at gold NOREG > > ================================================================================================= > 3 profiles 1 alias > > > > > and what did you see in logs? > > I think I understand. Do not need that external users connected through an > external profile (port 5080). Is necessary to add another internal profile > (to the external IP 195.211.131.140 port 5060) and that the two internal > and internal profile 2 (port 5060) were in the same context and domain. > > Is it possible? > > > > -- > Nikolay > > > On Fri, May 18, 2012 at 4:15 PM, Dubrovskiy Viacheslav < > slava at tangramltd.com> wrote: > >> Hi. >> >> Using the default configuration. There are internal and external real IP. >> I have several registered account from the internal network. I want to >> connect remote user from Ethernet. >> For this added external real IP to the external sip profile (param >> name="sip-ip" and param name="rtp-ip") and copied the >> directory/default.xml in directory/externa.xml which changed the > name="external_IP"> >> >> Now remote users can register and make calls. But there's a problem: No >> one can call to this external user. >> >> Tell me, what am I doing wrong? >> Thank you >> >> -- >> WBR, >> Dubrovskiy Viacheslav >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > -- > WBR, > Dubrovskiy Viacheslav > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120518/c7114e0e/attachment-0001.html From sharad at coraltele.com Sat May 19 14:36:02 2012 From: sharad at coraltele.com (Sharad Garg) Date: Sat, 19 May 2012 16:06:02 +0530 Subject: [Freeswitch-users] local_ip_v4 is resoved with 127.0.0.1 References: Message-ID: <76A356CCE67C44DCB143D6F3CB2831B4@sharad> Hi All Just installed the freeswitch with centos. I found that freeswitch is getting the local_ip_v4 = 127.0.0.1 & thats why the calls are not handled by freeswitch. I have set auto detection. With the same centos / freeswitch, I checked it up with another pc & found everything working. This time local_ip_v4 is getting the ip of eth0. In both scenerio, pc is having eth0 & eth1. >From this excercise, I concluded that there is something wrong in installation at OS side. But I do not know what is wrong. Actually I do not want to configure the eth0 IP manually in xml files. I want auto detection only. So could someone help me to diagnose this issue ? Thanks in advance. regards Sharad From koralu at gmail.com Sat May 19 16:15:24 2012 From: koralu at gmail.com (Adrian Andrei) Date: Sat, 19 May 2012 15:15:24 +0300 Subject: [Freeswitch-users] bridge_early_media and bypass_media Message-ID: Hello, In my dial plan I use bypass_media set to true because I want the codec negotiation be made by end points. Also I want the early_media to be bridge from Bleg to Aleg. For this setup I design this dialplan: Setting bypass_media and bridge_early_media to true cause no early media bridge. If I set only bridge_early_media=true and uncomment bypass_media line then early media is bridged but no codec negotiation between end points. How can I have also codec neg. between end points and early media bridged? Thank you -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120519/7d647d2f/attachment.html From anthony.minessale at gmail.com Sat May 19 18:16:50 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 19 May 2012 09:16:50 -0500 Subject: [Freeswitch-users] bridge_early_media and bypass_media In-Reply-To: References: Message-ID: The best you can do is set {inherit_codec=true,bypass_media_after_bridge=true,bridge_early_media=true} and use the inbound-late-negotiation sofia profile param. On Sat, May 19, 2012 at 7:15 AM, Adrian Andrei wrote: > Hello, > > In my dial plan I use bypass_media set to true because I want the codec > negotiation be made by end points. Also I want the early_media to be bridge > from Bleg to Aleg. For this setup I design this dialplan: > > > > > > > > Setting bypass_media and bridge_early_media to true cause no early media > bridge. > If I set only bridge_early_media=true and uncomment bypass_media line then > early media is bridged but no codec negotiation between end points. > > How can I have also codec neg. between end points and early media bridged? > > Thank you > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From aftnix at gmail.com Sat May 19 14:19:26 2012 From: aftnix at gmail.com (Arif Hossain) Date: Sat, 19 May 2012 16:19:26 +0600 Subject: [Freeswitch-users] Fwd: RTP stats explaination In-Reply-To: References: Message-ID: -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all, We are getting very poor quality of voice during testing of a new filtering application of us. The application receives packets from kernel using netfilter_queue library. Then insert the packets into a new user managed queue and does some transformations on it, like concatenation of udp payload. The network is healthy. Its inside our lab. And it does not drop packets or anything . In our app we do not forward packet immediately. After enough packet received to increase rtp packetization time (ptime) the we forward the message over raw socket and set dscp to be 10 so that this time packets can escape iptable rules. >From client side the RTP stream analysis shows nearly every stream as problematic. summery for some streams are given below : Stream 1: Max delta = 1758.72 ms at packet no. 40506 Max jitter = 231.07 ms. Mean jitter = 9.27 ms. Max skew = -2066.18 ms. Total RTP packets = 468 ? (expected 468) ? Lost RTP packets = 0 (0.00%) ? Sequence errors = 0 Duration 23.45 s (-22628 ms clock drift, corresponding to 281 Hz (-96.49%) Stream 2: Max delta = 1750.96 ms at packet no. 45453 Max jitter = 230.90 ms. Mean jitter = 7.50 ms. Max skew = -2076.96 ms. Total RTP packets = 468 ? (expected 468) ? Lost RTP packets = 0 (0.00%) ? Sequence errors = 0 Duration 23.46 s (-22715 ms clock drift, corresponding to 253 Hz (-96.84%) Stream 3: Max delta = 71.47 ms at packet no. 25009 Max jitter = 6.05 ms. Mean jitter = 2.33 ms. Max skew = -29.09 ms. Total RTP packets = 258 ? (expected 258) ? Lost RTP packets = 0 (0.00%) ? Sequence errors = 0 Duration 10.28 s (-10181 ms clock drift, corresponding to 76 Hz (-99.05%) Any idea where should we look for the problem? - -- - -aft -----BEGIN PGP SIGNATURE----- Version: OpenPGP.js v0.1 Comment: http://openpgpjs.org wsBcBAEBAgAQBQJPtiJ7CRCJVJ6A/SK8awAAHJgH/jRURuZWygqJTX7zafjK 807MvWxODhtw2p4ZLg+sCNJ0OpZlV5GftPKwH9XB4GfUKhmTPfCZ3n8vF6PU nWKMdBDM3D/K+DktpiGuAK90A5lwniFptN8DGt/ltqdAaSAlGbz1E6LGGdlw 1w/6lPeO3J9HKpkNQA2nfn3q/s5ZHQiYoGvgFofkm85LBSYgyn4b7YDY7qnp wtsGkSR1h66+q37Hb+1iZyWrHj26YJ+sNRBe8GzKdEccwrKMX1gJFNl5BBfl frGsZ3Y02cSYFg5L1vVstl8F/yTz8JXTGk6yKbtA0N+TRhI7gIJ/VItm1SJt JcF94ylPjIWiw9Iacv8ec88= =jiIm -----END PGP SIGNATURE----- -- -aft From anthony.minessale at gmail.com Sat May 19 18:37:45 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 19 May 2012 09:37:45 -0500 Subject: [Freeswitch-users] LINGER In-Reply-To: <72CBCB40-6B65-4C4C-9F26-7409D6E3DA6B@bellsouth.net> References: <72CBCB40-6B65-4C4C-9F26-7409D6E3DA6B@bellsouth.net> Message-ID: Are you using the C esl lib in tree? The testsetver.c in libs/esl listens on port 8084 and uses linger properly. The only way it would be saying that is if the connection was not On Fri, May 18, 2012 at 10:00 AM, Bernard Fluixa wrote: > Hello, > > I am working on an outbound ESL application in C on Linux. I can send commands and receive events normally. My problem is that the "linger" command returns > the "-ERR not controlling a session" error message. The socket application is set to be in async full mode. (). > > Any idea? > > Thank you > > B > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From koralu at gmail.com Sat May 19 20:01:17 2012 From: koralu at gmail.com (Adrian Andrei) Date: Sat, 19 May 2012 19:01:17 +0300 Subject: [Freeswitch-users] bridge_early_media and bypass_media In-Reply-To: References: Message-ID: Thank you for the fast answer but doesn't work. The dial plan looks: If I set the codec to G711ulaw for endpoints the call is ok. If I change the codec to something FS doesn't know I receive: 2012-05-19 19:50:52.130039 [INFO] mod_dptools.c:2954 Originate Failed. Cause: INCOMPATIBLE_DESTINATION Param inbound_late_negotiation is true in sofia internal profile. TY On Sat, May 19, 2012 at 5:16 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > The best you can do is set > {inherit_codec=true,bypass_media_after_bridge=true,bridge_early_media=true} > and use the inbound-late-negotiation sofia profile param. > > > On Sat, May 19, 2012 at 7:15 AM, Adrian Andrei wrote: > > Hello, > > > > In my dial plan I use bypass_media set to true because I want the codec > > negotiation be made by end points. Also I want the early_media to be > bridge > > from Bleg to Aleg. For this setup I design this dialplan: > > > > > > > > > > > > > > > > Setting bypass_media and bridge_early_media to true cause no early media > > bridge. > > If I set only bridge_early_media=true and uncomment bypass_media line > then > > early media is bridged but no codec negotiation between end points. > > > > How can I have also codec neg. between end points and early media > bridged? > > > > Thank you > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120519/51f2c140/attachment-0001.html From vetali100 at gmail.com Sat May 19 20:46:23 2012 From: vetali100 at gmail.com (Vitalie Colosov) Date: Sat, 19 May 2012 09:46:23 -0700 Subject: [Freeswitch-users] bridge_early_media and bypass_media In-Reply-To: References: Message-ID: Maybe this is the case when it is required to set the variables into the b-leg using {...} in the bridge command (or use "export" instead of "set"). Try this: (not sure it will work, but worth to test) 2012/5/19 Adrian Andrei > Thank you for the fast answer but doesn't work. The dial plan looks: > > > > > > > > > > > > If I set the codec to G711ulaw for endpoints the call is ok. If I change > the codec to something FS doesn't know I receive: > > 2012-05-19 19:50:52.130039 [INFO] mod_dptools.c:2954 Originate Failed. > Cause: INCOMPATIBLE_DESTINATION > > Param inbound_late_negotiation is true in sofia internal profile. > > TY > > > On Sat, May 19, 2012 at 5:16 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> The best you can do is set >> >> {inherit_codec=true,bypass_media_after_bridge=true,bridge_early_media=true} >> and use the inbound-late-negotiation sofia profile param. >> >> >> On Sat, May 19, 2012 at 7:15 AM, Adrian Andrei wrote: >> > Hello, >> > >> > In my dial plan I use bypass_media set to true because I want the codec >> > negotiation be made by end points. Also I want the early_media to be >> bridge >> > from Bleg to Aleg. For this setup I design this dialplan: >> > >> > >> > >> > >> > >> > >> > >> > Setting bypass_media and bridge_early_media to true cause no early media >> > bridge. >> > If I set only bridge_early_media=true and uncomment bypass_media line >> then >> > early media is bridged but no codec negotiation between end points. >> > >> > How can I have also codec neg. between end points and early media >> bridged? >> > >> > Thank you >> > >> > >> _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > Join Us At ClueCon - Aug 7-9, 2012 >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120519/4af0f85d/attachment.html From koralu at gmail.com Sat May 19 21:34:28 2012 From: koralu at gmail.com (Adrian Andrei) Date: Sat, 19 May 2012 20:34:28 +0300 Subject: [Freeswitch-users] bridge_early_media and bypass_media In-Reply-To: References: Message-ID: Thx but same result. No codec negotiation. On Sat, May 19, 2012 at 7:46 PM, Vitalie Colosov wrote: > Maybe this is the case when it is required to set the variables into the > b-leg using {...} in the bridge command (or use "export" instead of "set"). > Try this: (not sure it will work, but worth to test) > > > > > > > > > > > 2012/5/19 Adrian Andrei > >> Thank you for the fast answer but doesn't work. The dial plan looks: >> >> >> >> >> >> >> >> >> >> >> >> If I set the codec to G711ulaw for endpoints the call is ok. If I change >> the codec to something FS doesn't know I receive: >> >> 2012-05-19 19:50:52.130039 [INFO] mod_dptools.c:2954 Originate Failed. >> Cause: INCOMPATIBLE_DESTINATION >> >> Param inbound_late_negotiation is true in sofia internal profile. >> >> TY >> >> >> On Sat, May 19, 2012 at 5:16 PM, Anthony Minessale < >> anthony.minessale at gmail.com> wrote: >> >>> The best you can do is set >>> >>> {inherit_codec=true,bypass_media_after_bridge=true,bridge_early_media=true} >>> and use the inbound-late-negotiation sofia profile param. >>> >>> >>> On Sat, May 19, 2012 at 7:15 AM, Adrian Andrei wrote: >>> > Hello, >>> > >>> > In my dial plan I use bypass_media set to true because I want the codec >>> > negotiation be made by end points. Also I want the early_media to be >>> bridge >>> > from Bleg to Aleg. For this setup I design this dialplan: >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > Setting bypass_media and bridge_early_media to true cause no early >>> media >>> > bridge. >>> > If I set only bridge_early_media=true and uncomment bypass_media line >>> then >>> > early media is bridged but no codec negotiation between end points. >>> > >>> > How can I have also codec neg. between end points and early media >>> bridged? >>> > >>> > Thank you >>> > >>> > >>> _________________________________________________________________________ >>> > Professional FreeSWITCH Consulting Services: >>> > consulting at freeswitch.org >>> > http://www.freeswitchsolutions.com >>> > >>> > >>> > >>> > >>> > Official FreeSWITCH Sites >>> > http://www.freeswitch.org >>> > http://wiki.freeswitch.org >>> > http://www.cluecon.com >>> > >>> > Join Us At ClueCon - Aug 7-9, 2012 >>> > >>> > FreeSWITCH-users mailing list >>> > FreeSWITCH-users at lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.org >>> > >>> >>> >>> >>> -- >>> Anthony Minessale II >>> >>> FreeSWITCH http://www.freeswitch.org/ >>> ClueCon http://www.cluecon.com/ >>> Twitter: http://twitter.com/FreeSWITCH_wire >>> >>> AIM: anthm >>> MSN:anthony_minessale at hotmail.com >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>> IRC: irc.freenode.net #freeswitch >>> >>> FreeSWITCH Developer Conference >>> sip:888 at conference.freeswitch.org >>> googletalk:conf+888 at conference.freeswitch.org >>> pstn:+19193869900 >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> Join Us At ClueCon - Aug 7-9, 2012 >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120519/39a1e949/attachment-0001.html From avi at avimarcus.net Sat May 19 21:42:10 2012 From: avi at avimarcus.net (Avi Marcus) Date: Sat, 19 May 2012 20:42:10 +0300 Subject: [Freeswitch-users] bridge_early_media and bypass_media In-Reply-To: References: Message-ID: The next step would be to pastebin a log -- both the fs_cli on /log 7 and preferably the siptrace, too, so someone could see what's going on. -Avi On Sat, May 19, 2012 at 8:34 PM, Adrian Andrei wrote: > Thx but same result. No codec negotiation. > > > On Sat, May 19, 2012 at 7:46 PM, Vitalie Colosov wrote: > >> Maybe this is the case when it is required to set the variables into the >> b-leg using {...} in the bridge command (or use "export" instead of "set"). >> Try this: (not sure it will work, but worth to test) >> >> >> >> >> >> >> >> >> >> >> 2012/5/19 Adrian Andrei >> >>> Thank you for the fast answer but doesn't work. The dial plan looks: >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> If I set the codec to G711ulaw for endpoints the call is ok. If I change >>> the codec to something FS doesn't know I receive: >>> >>> 2012-05-19 19:50:52.130039 [INFO] mod_dptools.c:2954 Originate Failed. >>> Cause: INCOMPATIBLE_DESTINATION >>> >>> Param inbound_late_negotiation is true in sofia internal profile. >>> >>> TY >>> >>> >>> On Sat, May 19, 2012 at 5:16 PM, Anthony Minessale < >>> anthony.minessale at gmail.com> wrote: >>> >>>> The best you can do is set >>>> >>>> {inherit_codec=true,bypass_media_after_bridge=true,bridge_early_media=true} >>>> and use the inbound-late-negotiation sofia profile param. >>>> >>>> >>>> On Sat, May 19, 2012 at 7:15 AM, Adrian Andrei >>>> wrote: >>>> > Hello, >>>> > >>>> > In my dial plan I use bypass_media set to true because I want the >>>> codec >>>> > negotiation be made by end points. Also I want the early_media to be >>>> bridge >>>> > from Bleg to Aleg. For this setup I design this dialplan: >>>> > >>>> > >>>> > >>>> > >>>> > >>>> > >>>> > >>>> > Setting bypass_media and bridge_early_media to true cause no early >>>> media >>>> > bridge. >>>> > If I set only bridge_early_media=true and uncomment bypass_media line >>>> then >>>> > early media is bridged but no codec negotiation between end points. >>>> > >>>> > How can I have also codec neg. between end points and early media >>>> bridged? >>>> > >>>> > Thank you >>>> > >>>> > >>>> _________________________________________________________________________ >>>> > Professional FreeSWITCH Consulting Services: >>>> > consulting at freeswitch.org >>>> > http://www.freeswitchsolutions.com >>>> > >>>> > >>>> > >>>> > >>>> > Official FreeSWITCH Sites >>>> > http://www.freeswitch.org >>>> > http://wiki.freeswitch.org >>>> > http://www.cluecon.com >>>> > >>>> > Join Us At ClueCon - Aug 7-9, 2012 >>>> > >>>> > FreeSWITCH-users mailing list >>>> > FreeSWITCH-users at lists.freeswitch.org >>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> > UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> > http://www.freeswitch.org >>>> > >>>> >>>> >>>> >>>> -- >>>> Anthony Minessale II >>>> >>>> FreeSWITCH http://www.freeswitch.org/ >>>> ClueCon http://www.cluecon.com/ >>>> Twitter: http://twitter.com/FreeSWITCH_wire >>>> >>>> AIM: anthm >>>> MSN:anthony_minessale at hotmail.com >>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>> IRC: irc.freenode.net #freeswitch >>>> >>>> FreeSWITCH Developer Conference >>>> sip:888 at conference.freeswitch.org >>>> googletalk:conf+888 at conference.freeswitch.org >>>> pstn:+19193869900 >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> Join Us At ClueCon - Aug 7-9, 2012 >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at 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 at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> Join Us At ClueCon - Aug 7-9, 2012 >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120519/4ff0526b/attachment.html From koralu at gmail.com Sat May 19 22:56:34 2012 From: koralu at gmail.com (Adrian Andrei) Date: Sat, 19 May 2012 21:56:34 +0300 Subject: [Freeswitch-users] bridge_early_media and bypass_media In-Reply-To: References: Message-ID: With this dial plan early media is send but no codec negotiation. For this test I take 2 Xlite with iLBC codec only. The calls are canceled with Cause: INCOMPATIBLE_DESTINATION. Here is the pastebin: http://pastebin.freeswitch.org/19112 If the codec is G711alaw for example the calls are ok. Ty On Sat, May 19, 2012 at 8:42 PM, Avi Marcus wrote: > The next step would be to pastebin a log -- both the fs_cli on /log 7 and > preferably the siptrace, too, so someone could see what's going on. > > -Avi > > > On Sat, May 19, 2012 at 8:34 PM, Adrian Andrei wrote: > >> Thx but same result. No codec negotiation. >> >> >> On Sat, May 19, 2012 at 7:46 PM, Vitalie Colosov wrote: >> >>> Maybe this is the case when it is required to set the variables into the >>> b-leg using {...} in the bridge command (or use "export" instead of "set"). >>> Try this: (not sure it will work, but worth to test) >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> 2012/5/19 Adrian Andrei >>> >>>> Thank you for the fast answer but doesn't work. The dial plan looks: >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> If I set the codec to G711ulaw for endpoints the call is ok. If I >>>> change the codec to something FS doesn't know I receive: >>>> >>>> 2012-05-19 19:50:52.130039 [INFO] mod_dptools.c:2954 Originate Failed. >>>> Cause: INCOMPATIBLE_DESTINATION >>>> >>>> Param inbound_late_negotiation is true in sofia internal profile. >>>> >>>> TY >>>> >>>> >>>> On Sat, May 19, 2012 at 5:16 PM, Anthony Minessale < >>>> anthony.minessale at gmail.com> wrote: >>>> >>>>> The best you can do is set >>>>> >>>>> {inherit_codec=true,bypass_media_after_bridge=true,bridge_early_media=true} >>>>> and use the inbound-late-negotiation sofia profile param. >>>>> >>>>> >>>>> On Sat, May 19, 2012 at 7:15 AM, Adrian Andrei >>>>> wrote: >>>>> > Hello, >>>>> > >>>>> > In my dial plan I use bypass_media set to true because I want the >>>>> codec >>>>> > negotiation be made by end points. Also I want the early_media to be >>>>> bridge >>>>> > from Bleg to Aleg. For this setup I design this dialplan: >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > Setting bypass_media and bridge_early_media to true cause no early >>>>> media >>>>> > bridge. >>>>> > If I set only bridge_early_media=true and uncomment bypass_media >>>>> line then >>>>> > early media is bridged but no codec negotiation between end points. >>>>> > >>>>> > How can I have also codec neg. between end points and early media >>>>> bridged? >>>>> > >>>>> > Thank you >>>>> > >>>>> > >>>>> _________________________________________________________________________ >>>>> > Professional FreeSWITCH Consulting Services: >>>>> > consulting at freeswitch.org >>>>> > http://www.freeswitchsolutions.com >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > Official FreeSWITCH Sites >>>>> > http://www.freeswitch.org >>>>> > http://wiki.freeswitch.org >>>>> > http://www.cluecon.com >>>>> > >>>>> > Join Us At ClueCon - Aug 7-9, 2012 >>>>> > >>>>> > FreeSWITCH-users mailing list >>>>> > FreeSWITCH-users at lists.freeswitch.org >>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> > UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> > http://www.freeswitch.org >>>>> > >>>>> >>>>> >>>>> >>>>> -- >>>>> Anthony Minessale II >>>>> >>>>> FreeSWITCH http://www.freeswitch.org/ >>>>> ClueCon http://www.cluecon.com/ >>>>> Twitter: http://twitter.com/FreeSWITCH_wire >>>>> >>>>> AIM: anthm >>>>> MSN:anthony_minessale at hotmail.com >>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>>> IRC: irc.freenode.net #freeswitch >>>>> >>>>> FreeSWITCH Developer Conference >>>>> sip:888 at conference.freeswitch.org >>>>> googletalk:conf+888 at conference.freeswitch.org >>>>> pstn:+19193869900 >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> Join Us At ClueCon - Aug 7-9, 2012 >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at 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 at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> Join Us At ClueCon - Aug 7-9, 2012 >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at 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 at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> Join Us At ClueCon - Aug 7-9, 2012 >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120519/1857a1e3/attachment-0001.html From bdfoster at endigotech.com Sun May 20 01:08:41 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Sat, 19 May 2012 17:08:41 -0400 Subject: [Freeswitch-users] Huawei E169 Group Buy In-Reply-To: References: <4FAF49A5.5000203@coppice.org> <023801cd31e5$10cf3820$326da860$@com> <90D7665C-274B-4DD1-BC8D-A9B97278A005@opencsta.org> <4FB13875.4040907@coppice.org> Message-ID: I'd really like to pull Giovanni aside and have a talk to see if we can come up with a solution for the "ultimate dongle" that is compatible with all bands in the US and has an external antenna jack. I'm not exactly fluent in all of this. If we can find one and support can be added in easily, I'd go as far as buying one somewhere and letting him have access to my machine so he can figure it all out. Giovanni, contact me off list or on IRC of interested. -BDF On May 18, 2012 11:08 AM, "William Suffill" wrote: > Did we figure out what model is best for the US market? Looks quite > interesting. > > -- W > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120519/5dbd75b8/attachment.html From fluixab at bellsouth.net Sun May 20 02:05:58 2012 From: fluixab at bellsouth.net (Bernard Fluixa) Date: Sat, 19 May 2012 18:05:58 -0400 Subject: [Freeswitch-users] LINGER In-Reply-To: References: <72CBCB40-6B65-4C4C-9F26-7409D6E3DA6B@bellsouth.net> Message-ID: Anthony, Yes. It now works. It was called at a wrong place. Thanks again. Bernard On May 19, 2012, at 10:37 AM, Anthony Minessale wrote: > Are you using the C esl lib in tree? The testsetver.c in libs/esl > listens on port 8084 and uses linger properly. > > > The only way it would be saying that is if the connection was not > > On Fri, May 18, 2012 at 10:00 AM, Bernard Fluixa wrote: >> Hello, >> >> I am working on an outbound ESL application in C on Linux. I can send commands and receive events normally. My problem is that the "linger" command returns >> the "-ERR not controlling a session" error message. The socket application is set to be in async full mode. (). >> >> Any idea? >> >> Thank you >> >> B >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From codecomplete at free.fr Mon May 21 00:45:24 2012 From: codecomplete at free.fr (GillesToo) Date: Sun, 20 May 2012 13:45:24 -0700 (PDT) Subject: [Freeswitch-users] Freeswitch stable on embedded devices? Message-ID: <1337546724347-7568340.post@n2.nabble.com> Hello I'd' like to hear from people who run Freeswitch on embedded devices such as the Seagate DockStar, the SheevaPlug, etc. I intend to connect the device to the PSTN through an Obi110 VoIP gateway along with an ITSP over the Net, and support up to 30 internal extensions. Is Freeswitch on such small hardware rock-solid, is it easy to recompile for such non-x86 platforms, or should I get a regular x86 host like the Asus EeeBoxPC (www.asus.com/Eee/)? Thank you. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-stable-on-embedded-devices-tp7568340.html Sent from the freeswitch-users mailing list archive at Nabble.com. From mel0torme at gmail.com Mon May 21 02:17:25 2012 From: mel0torme at gmail.com (Tom C) Date: Sun, 20 May 2012 15:17:25 -0700 Subject: [Freeswitch-users] Freeswitch stable on embedded devices? In-Reply-To: <1337546724347-7568340.post@n2.nabble.com> References: <1337546724347-7568340.post@n2.nabble.com> Message-ID: I've been running FreeSwitch on a Dockstar for about 2 years now, and, yes, it is rock solid running on this tiny device. At one point, I noticed that it had been running un-interrupted for nearly 6 months. I use it with several voip service providers, and Google Voice integration works perfectly. However, this is just my home PBX, with a max of 4 concurrent calls. I haven't done any scalability testing. Also, there are some things that don't work well: a) When I try to use FLITE (the text-to-speech module) on the dockstar or a pogoplug, the CPU usage shoots up to 100%, resulting audio is choppy, and any ongoing calls also get choppy. I haven't taken a close look at the source code, but I suspect FLITE is using floating-point math. The poor little dockstars and pogoplugs don't have floating-point processors, so any floating-point math eats up a whole lot of CPU time. (I just glanced at the source, and I see they define M_PI = 3.14159. So, yes, floating point math.) c) Music-on-hold works fine. But streaming audio files using Session.streamFile() or Session.say() will sometimes have very choppy audio (but without the extreme CPU usage that FLITE causes). I'm trying to figure out this problem as we speak..... So, basically, if you're using FreeSwitch as a simple home PBX, the dockstar and pogoplug are wonderful devices to use. But if you're going to make use of the more interesting features of FreeSwitch, you will definitely want to invest in an x86 or similar platform. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120520/cb20feac/attachment.html From codecomplete at free.fr Mon May 21 04:31:35 2012 From: codecomplete at free.fr (GillesToo) Date: Sun, 20 May 2012 17:31:35 -0700 (PDT) Subject: [Freeswitch-users] Freeswitch stable on embedded devices? In-Reply-To: References: <1337546724347-7568340.post@n2.nabble.com> Message-ID: <1337560295076-7568516.post@n2.nabble.com> Thanks Tom. I won't use FLITE, but I'll have to check which features in Freeswitch require floating-point math to make sure I won't have that problem. Do you use uncompressed files for MoH or MP3, which could explain the chopiness? Do you recompile Freeswitch yourself for the Dockstar? If yes, is that hard to do? I need to make sure I'm able to update the devices in case serious bugs or useful features come up once they're out in the field. Thank you. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-stable-on-embedded-devices-tp7568340p7568516.html Sent from the freeswitch-users mailing list archive at Nabble.com. From ahe.sanath at gmail.com Mon May 21 09:45:18 2012 From: ahe.sanath at gmail.com (Sanath Prasanna) Date: Mon, 21 May 2012 11:15:18 +0530 Subject: [Freeswitch-users] Run LUA script in different server In-Reply-To: References: Message-ID: Hi MC, I did the change according to ure instruction. But error is coming. Here I attached freeswitch.log file I change the confs as follows in BOX A. (Operator connected Freeswitch box) BOX B ip is 10.22.29.253 vi /usr/local/freeswitch/conf/dialplan/default.xml Also add following to ACL file in BOX B Pls advice to solve the problem here. Br, Sanath On Fri, May 18, 2012 at 9:57 AM, Michael Collins wrote: > If I understand your question correctly, yes you can do this. You can send > calls from one FreeSWITCH server to another. Start here: > http://wiki.freeswitch.org/wiki/Connect_Two_FreeSWITCH_Boxes > > Best way to learn is to get the FreeSWITCH books from Packt Publishing and > just start hacking code. > > -MC > > > On Thu, May 17, 2012 at 9:07 PM, Sanath Prasanna wrote: > >> Tx for advice MC & Anita. Can I do work around like this . >> Another freeswitch instant will be start in other server & calls will be >> transfer from operator connected freeswitch instance to this new freeswitch >> instance & vise versa. Pls advice. >> >> >> On Thu, May 17, 2012 at 5:05 PM, Anita Hall wrote: >> >>> You could run a Lua ESL server on a different machine but this will not >>> be the same as running a Lua script. >>> http://wiki.freeswitch.org/wiki/Event_Socket_Library >>> >>> regards, >>> Anita >>> >>> >>> >>> On Thu, May 17, 2012 at 4:37 AM, Michael Collins wrote: >>> >>>> I don't think you can directly do what you are describing. However, you >>>> might be able to use mod_httapi for this. There's some documentation on the >>>> wiki and in the module. Keep in mind that this is a relatively new module >>>> so we don't have lots of examples yet, so you'll probably be doing a fair >>>> amount of research and testing. >>>> >>>> -MC >>>> >>>> >>>> On Wed, May 16, 2012 at 5:59 AM, Sanath Prasanna wrote: >>>> >>>>> Hi all, >>>>> I have 2 servers. One server has SIP GW connection From Operator & IVR >>>>> applications need to build in other server. How to call distributed LUA >>>>> applications with Mysql Databases from the SIP GW server ? Pls advice. >>>>> Main idea is, maintaining SIP connection in one server & all the IVR >>>>> applications in other server. >>>>> Br, >>>>> Sanath >>>>> >>>>> >>>>> >>>> >>>> >>>> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120521/ce4132bc/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: freeswitch.log Type: application/octet-stream Size: 15732 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120521/ce4132bc/attachment-0001.obj From miha at softnet.si Mon May 21 10:51:46 2012 From: miha at softnet.si (Miha) Date: Mon, 21 May 2012 08:51:46 +0200 Subject: [Freeswitch-users] Call_group intercept Message-ID: <4FB9E602.8070708@softnet.si> HI, I read wiki tutorial about call group intercept (http://wiki.freeswitch.org/wiki/Callgroup_intercept) ans also implemented it. (If I make call intercept which is not for callgroup it works) I put this in public dialplan, so that every call UUID will be inserted in mysql database. When call hit public dialplan I can not access variable callgroup. As callgroup variable is not set I can not make right mysql insert. What is the best way to deal with this? THanks! Miha -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120521/bd69cb57/attachment.html From anita.hall at simmortel.com Mon May 21 11:09:48 2012 From: anita.hall at simmortel.com (Anita Hall) Date: Mon, 21 May 2012 12:39:48 +0530 Subject: [Freeswitch-users] Spandsp mulated modems inbound/no /dev/FS* device? In-Reply-To: <4FB61319.6060205@integrafin.co.uk> References: <4FA24100.7040908@integrafin.co.uk> <20120503151946.GA824@eagle.cupis.co.uk> <4FA3EE97.1090908@integrafin.co.uk> <4FA596BF.4090703@integrafin.co.uk> <4FA635AE.5050006@integrafin.co.uk> <4FA66862.1090300@coppice.org> <4FA6871F.5050909@puzzled.xs4all.nl> <4FA696B1.8050706@coppice.org> <4FB553A3.5010807@integrafin.co.uk> <4FB61319.6060205@integrafin.co.uk> Message-ID: Hi Alex I got ptlib, opan and t38modem compile and installed. So far so good. Is there a wiki entry you have made with details of your FS + T38modem + Hylafax set-up, configuration files etc. I was able to dig up some stuff from your mails in the list but some wiki entry will help me and others a lot. My team member has put up a Wiki entry at http://wiki.freeswitch.org/wiki/T38modem It will help us a lot if you could do a quick edit and copy/paste of your config. Many many thanks! regards, Anita On Fri, May 18, 2012 at 2:45 PM, Alex Crow wrote: > On 18/05/12 09:34, Anita Hall wrote: > > Hi Alex > > > > Many thanks for the reply. > > > > Do I need t38modem if I am already receiving the Fax calls over E1 via > > a Sangoma Card in my FS machine ? I read on the list that Hylafax > > takes audio and the only reason you needed t38modem was because your > > FS was getting calls over SIP. > > > > Anita, > > Yes, you do if you want to use Hylafax and need inbound faxing, because > it needs a modem device. T38Modem provides that (eg /dev/ttyT38x). > > > This is my topology > > ~~~~~~~~~ > > Hylafax <-------->|______| <-------> FreeSWITCH <---------> E1 > > Pretty much the same as mine in effect. > > So you would need > > Hylafax <-------->T38Modem<-------> FreeSWITCH/t38gateway <---------> E1 > > So hylafax speaks "Fax data" to T38modem, which speaks T38 to > FreeSwitch, which gateways to/from audio over the E1. You may need to > change my configs a bit to get that last bit working, maybe not. > > > > ~~~~~~~~~ > > > > Since Hylafax is already taking audio calls, may be I do not need the > > t38modem ? If not, can Hylafax take audio calls directly from FreeSWITCH > ? > > Not unless the inbuilt modem can be made to be an endpoint for calls > from coming in from the E1, which I understand at present it cannot. > This is because Hylafax does not send and receive audio itself, it talks > fax control commands to modems and sends and receives data over a serial > port. > > > > > This is your topology (taken from your mails in the list) > > ~~~~~~~~~ > > > Hylafax<---Audio---->t38modem<---T38--->Freeswitch/t38gateway<----G.711/SIP--->Mitel > > 3300(T38 unsupported, grr)<---ISDN30---->PSTN > > ~~~~~~~~~ > > Actually the Mitel does support T38 with a license. The thing I'm not > sure about is how good its gatewaying to ISDN is. > > Cheers > > Alex > > > -- > This message is intended only for the addressee and may contain > confidential information. Unless you are that person, you may not > disclose its contents or use it in any way and are requested to delete > the message along with any attachments and notify us immediately. > > "Transact" is operated by Integrated Financial Arrangements plc > Domain House, 5-7 Singer Street, London EC2A 4BQ > Tel: (020) 7608 4900 Fax: (020) 7608 5300 > (Registered office: as above; Registered in England and Wales under > number: 3727592) > Authorised and regulated by the Financial Services Authority (entered on > the FSA Register; number: 190856) > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120521/5f01ecf6/attachment.html From aftnix at gmail.com Mon May 21 11:26:25 2012 From: aftnix at gmail.com (Arif Hossain) Date: Mon, 21 May 2012 13:26:25 +0600 Subject: [Freeswitch-users] BYE message is not relayed to the UAC In-Reply-To: References: Message-ID: Hi, We have the following network architecture : UAC1------------------------->kamailio-------------------->VoipSwitch----->PSTN---------->Phone1 (Sip Client) Now UAC1 calls Phone1 and everything is ok. If UAC1 hangs up session is terminated cleanly. But if Phone1 hangs up the BYE message which ?comes to kamailio and goes back to VoipSwitch instead of relayed to UAC1 . So The session becomes a zombie one, And UAC1 unfortunately gets billed for a session which should be terminated. Following is the Call flow as seen from VoipSwitch : ?| ? ? ? ? ? ? ? ? ? ? ?| | ? ? ? ? | ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?| ? ? | |134.856 ?| ? ? ? ? INVITE SDP | ? ? ? ? |(7890) ? ------------------> ?(5060) ? | |134.858 ?| ? ? ? ? 407 Proxy Authentication Required | ? ? ? ? |(7890) ? <------------------ ?(5060) ? | |134.902 ?| ? ? ? ? ACK ? ? ? | | ? ? ? ? |(7890) ? ------------------> ?(5060) ? | |135.408 ?| ? ? ? ? INVITE SDP | ? ? ? ? |(7890) ? ------------------> ?(5060) ? | |135.414 ?| ? ? ? ? 100 Trying| | ? ? ? ? |(7890) ? <------------------ ?(5060) ? | |140.121 ?| ? ? ? ? 183 Session Progress SDP | ? ? ? ? |(7890) ? <------------------ ?(5060) ? | |140.184 ?| ? ? ? ? RTP (g729) ? ? ? ? ? ? ? ? ? ?| | ? ? ? ? |(61868) ?<------------------ ?(5136) ? | |141.295 ?| ? ? ? ? RTP (g729) ? ? ? ? ? ? ? ? ? ?| | ? ? ? ? |(61868) ?------------------> ?(5136) ? | |153.701 ?| ? ? ? ? 200 OK SDP | ? ? ? ? |(7890) ? <------------------ ?(5060) ? | |153.713 ?| ? ? ? ? RTP (g729) ? ? ? ? ? ? ? ? ? ?| | ? ? ? ? |(61868) ?------------------> ?(5136) ? | |154.126 ?| ? ? ? ? ACK ? ? ? | | ? ? ? ? |(7890) ? ------------------> ?(5060) ? | |159.988 ?| ? ? ? ? BYE ? ? ? | | ? ? ? ? |(7890) ? <------------------ ?(5060) ? | |160.031 ?| ? ? ? ? BYE ? ? ? | | ? ? ? ? |(7890) ? ------------------> ?(5060) ? | |160.478 ?| ? ? ? ? BYE ? ? ? | | ? ? ? ? |(7890) ? ------------------> ?(5060) ? | |161.412 ?| ? ? ? ? BYE ? ? ? | | ? ? ? ? |(7890) ? ------------------> ?(5060) ? | |163.280 ?| ? ? ? ? BYE ? ? ? | | ? ? ? ? |(7890) ? ------------------> ?(5060) ? | |167.015 ?| ? ? ? ? BYE ? ? ? | | ? ? ? ? |(7890) ? ------------------> ?(5060) ? | |170.750 ?| ? ? ? ? BYE ? ? ? | | ? ? ? ? |(7890) ? ------------------> ?(5060) ? | |174.481 ?| ? ? ? ? BYE ? ? ? | | ? ? ? ? |(7890) ? ------------------> ?(5060) ? | |178.216 ?| ? ? ? ? BYE ? ? ? | | ? ? ? ? |(7890) ? ------------------> ?(5060) ? | |181.952 ?| ? ? ? ? BYE ? ? ? | | ? ? ? ? |(7890) ? ------------------> ?(5060) ? | |185.687 ?| ? ? ? ? BYE ? ? ? | | ? ? ? ? |(7890) ? ------------------> ?(5060) ? | |188.018 ?| ? ? ? ? 408 Request Timeout | ? ? ? ? |(7890) ? ------------------> ?(5060) ? | |211.849 ?| ? ? ? ? BYE ? ? ? | | ? ? ? ? |(7890) ? ------------------> ?(5060) ? | |212.292 ?| ? ? ? ? BYE ? ? ? | Sip Traces : kamailio------>VoipSwitch I'm posting only the offending BYE msg instead of full trace , because of the mail will become difficult to read . If more traces needed, i can post it. The following BYE message is sent by VoipSwitch: BYE sip:ipphone at 205.164.40.74 SIP/2.0 Route: CSeq: 2 BYE Via: SIP/2.0/UDP 205.164.40.74:5060 From: sip:008801673345531 at 205.164.40.74;tag=100528120745985872655137 Call-ID: IqBknV19AuxW0jk.8BjuE4hyx93Ws9qS To: "123456" ;tag=Zopl5lj5YiqyaSR5Le3QnfoR-G0NZAGG Content-Length: 0 Kamailio instead of relaying the message, sends a BYE message towards VoipSwitch: BYE sip:ipphone at 205.164.40.74 SIP/2.0 Max-Forwards: 10 CSeq: 2 BYE Via: SIP/2.0/UDP 108.166.195.189:7890;branch=z9hG4bK4b2b.5d893e95.0 Via: SIP/2.0/UDP 205.164.40.74:5060;rport=5060 From: sip:008801673345531 at 205.164.40.74;tag=100528120745985872655137 Call-ID: IqBknV19AuxW0jk.8BjuE4hyx93Ws9qS To: "123456" ;tag=Zopl5lj5YiqyaSR5Le3QnfoR-G0NZAGG Content-Length: 0 When the first BYE message comes from VoipSwitch , kamailio does the following : May 20 02:25:53 VOS20-108 /usr/local/sbin/kamailio[16442]: DEBUG: [receive.c:289]: receive_msg: cleaning up May 20 02:25:53 VOS20-108 /usr/local/sbin/kamailio[16442]: DEBUG: [parser/sdp/sdp.c:751]: _sdp = 0x831bf10 May 20 02:25:53 VOS20-108 /usr/local/sbin/kamailio[16442]: DEBUG: [parser/sdp/sdp.c:753]: sdp = 0x83043dc May 20 02:25:53 VOS20-108 /usr/local/sbin/kamailio[16442]: DEBUG: [parser/sdp/sdp.c:755]: session = 0x8304504 May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: [parser/msg_parser.c:630]: SIP Request: May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: [parser/msg_parser.c:632]: ?method: ? May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: [parser/msg_parser.c:634]: ?uri: May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: [parser/msg_parser.c:636]: ?version: May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: [parser/msg_parser.c:167]: get_hdr_field: cseq : <1> May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: [parser/parse_via.c:1287]: Found param type 232, = ; state=16 May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: [parser/parse_via.c:2300]: end of header reached, state=5 May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: [parser/msg_parser.c:515]: parse_headers: Via found, flags=2 May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: [parser/msg_parser.c:517]: parse_headers: this is the first via May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: [receive.c:145]: After parse_msg... May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: [receive.c:186]: preparing to run routing scripts... May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: [parser/parse_to.c:174]: DEBUG: add_param: tag=arILprdVR1srJ76HHlt4BEc3XsyaWcZm May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: [parser/parse_to.c:803]: end of header reached, state=29 May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: [parser/msg_parser.c:187]: DEBUG: get_hdr_field: [76]; uri=[sip:ipphone at 205.164.40.74] May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: [parser/msg_parser.c:189]: DEBUG: to body ["ipphone" ] May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: [parser/msg_parser.c:201]: DEBUG: get_hdr_body : content_length=0 May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: [parser/msg_parser.c:103]: found end of header May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: maxfwd [mf_funcs.c:66]: max_forwards header not found! May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: [parser/parse_to.c:174]: DEBUG: add_param: tag=1905251223419334290936029 May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: [parser/parse_to.c:803]: end of header reached, state=29 May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: siputils [checks.c:76]: totag found May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: rr [loose.c:85]: is_preloaded: No May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: [socket_info.c:501]: grep_sock_info - checking if host==us: 13==15 && ?[205.164.40.74] == [108.166.195.189] May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: [socket_info.c:504]: grep_sock_info - checking if port 7890 matches port 5060 May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: [socket_info.c:501]: grep_sock_info - checking if host==us: 13==15 && ?[205.164.40.74] == [108.166.195.189] May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: [socket_info.c:504]: grep_sock_info - checking if port 5060 matches port 5060 May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: [forward.c:448]: check_self: host != me May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: [socket_info.c:501]: grep_sock_info - checking if host==us: 15==15 && ?[108.166.195.189] == [108.166.195.189] May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: [socket_info.c:504]: grep_sock_info - checking if port 7890 matches port 7890 May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: rr [loose.c:792]: Topmost route URI: 'sip:108.166.195.189:7890;lr=on;nat=yes' is me May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: [parser/msg_parser.c:103]: found end of header May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: rr [loose.c:257]: No next Route HF found May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: rr [loose.c:811]: No next URI found May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: rr [loose.c:983]: params are <;lr=on;nat=yes> May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: siputils [checks.c:76]: totag found May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: tm [t_lookup.c:1379]: DEBUG: t_newtran: msg id=2501 , global msg id=2500 , T on entrance=0xffffffff May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: tm [t_lookup.c:528]: t_lookup_request: start searching: hash=29177, isACK=0 May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: tm [t_lookup.c:564]: DEBUG: proceeding to pre-RFC3261 transaction matching May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: tm [t_lookup.c:711]: DEBUG: t_lookup_request: no transaction found May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: tm [t_hooks.c:374]: DBG: trans=0xb61626a4, callback type 1, id 0 entered May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: [msg_translator.c:204]: check_via_address(205.164.40.74, 205.164.40.74, 0) May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: tm [t_funcs.c:388]: SER: new transaction fwd'ed May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: [usr_avp.c:646]: DEBUG:destroy_avp_list: destroying list (nil) May 20 02:26:04 VOS20-108 last message repeated 5 times May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: [receive.c:289]: receive_msg: cleaning up May 20 02:26:22 VOS20-108 /usr/local/sbin/kamailio[16445]: DEBUG: [udp_server.c:486]: udp_rcv_loop: probing packet received from 180.234.62.230 38722 May 20 02:26:34 VOS20-108 /usr/local/sbin/kamailio[16450]: DEBUG: tm [t_reply.c:1134]: ->>>>>>>>> T_code=0, new_code=408 May 20 02:26:34 VOS20-108 /usr/local/sbin/kamailio[16450]: DEBUG: tm [t_reply.c:1636]: DEBUG: relay_reply: branch=0, save=0, relay=0 May 20 02:26:34 VOS20-108 /usr/local/sbin/kamailio[16450]: DEBUG: [msg_translator.c:204]: check_via_address(205.164.40.74, 205.164.40.74, 0) May 20 02:26:34 VOS20-108 /usr/local/sbin/kamailio[16450]: DEBUG: [mem/shm_mem.c:105]: WARNING:vqm_resize: resize(0) called May 20 02:26:34 VOS20-108 /usr/local/sbin/kamailio[16450]: DEBUG: tm [t_hooks.c:288]: DBG: trans=0xb61626a4, callback type 128, id 0 entered May 20 02:26:54 VOS20-108 /usr/local/sbin/kamailio[16451]: WARNING: [timer.c:450]: WARNING: our timer runs faster then real-time (300000 ms / 4800 ticks our time .-> 299923 ms / 4798 ticks real time) OT: How do you guys maintain 80 column mails? i do not use a mail client, use gmail mailbox. -- -aft -- -aft From tarik.bts.gi at gmail.com Mon May 21 14:07:28 2012 From: tarik.bts.gi at gmail.com (ghallab) Date: Mon, 21 May 2012 10:07:28 +0000 Subject: [Freeswitch-users] ignore_display_updates meaning an polycom problem (urgent please) Message-ID: <4FBA13E0.5060400@gmail.com> Hi guys, Can you explain me what well happen if I set the variable "ignore_display_updates=true" before I make bridge action and what will happen if I set it to false? because I have a problem with FS and Polycom 550: when A call from outside of the FS network to B which is inside, then B answer and transfer the call to C which is inside also, when C's phone is ringing he see B's caller ID and when he answer, he see A's caller ID. Me I want that he see always caller ID of A. I tried to set "ignore_display_updates=true" in this case B see just C's caller ID when his is ringing and when he answer. There is any way to fix this issue? I am thinking about to set /"effective_caller_id_number={$origination_caller_id_number}" but I am not sure if this will work. / -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120521/1aae5d3d/attachment.html From peter.olsson at visionutveckling.se Mon May 21 13:22:54 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Mon, 21 May 2012 09:22:54 +0000 Subject: [Freeswitch-users] ignore_display_updates meaning an polycom problem (urgent please) Message-ID: <1FFF97C269757C458224B7C895F35F150D20D7@cantor.std.visionutv.se> I guess it mostly depends on the Polycom device. For instance, it's a big difference if the call is transferred using blind or supervised transfer. You will probably need to collect the SIP traffic, and check what actually happens when doing this scenario. Mvh Peter Olsson Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r ghallab Skickat: den 21 maj 2012 12:07 Till: FreeSWITCH Users Help ?mne: [Freeswitch-users] ignore_display_updates meaning an polycom problem (urgent please) Hi guys, Can you explain me what well happen if I set the variable "ignore_display_updates=true" before I make bridge action and what will happen if I set it to false? because I have a problem with FS and Polycom 550: when A call from outside of the FS network to B which is inside, then B answer and transfer the call to C which is inside also, when C's phone is ringing he see B's caller ID and when he answer, he see A's caller ID. Me I want that he see always caller ID of A. I tried to set "ignore_display_updates=true" in this case B see just C's caller ID when his is ringing and when he answer. There is any way to fix this issue? I am thinking about to set "effective_caller_id_number={$origination_caller_id_number}" but I am not sure if this will work. !DSPAM:4fba049932768390320688! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120521/09b51464/attachment.html From gopalakrishnan.an at gmail.com Mon May 21 14:12:24 2012 From: gopalakrishnan.an at gmail.com (Gopalakrishnan N) Date: Mon, 21 May 2012 15:42:24 +0530 Subject: [Freeswitch-users] FreeTDM with BRI Card Message-ID: Hi all, I am using Openvox BRI card with Freeswitch using FreeTDM. In NT mode I couldn't get dialtone. I got the error like " Unable to get channel 1: -1". Please help me to solve this issue. I have attached my configuration details. I have used FreeTDM module with LibPRI and Freeswitch Regards, Gopal. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120521/6678062e/attachment.html -------------- next part -------------- freeswitch at localhost.localdomain> ftdm list +OK span: 1 (BRI_1) type: isdn physical_status: ok signaling_status: UP chan_count: 3 dialplan: asterisk context: from-intern dial_regex: fail_dial_regex: hold_music: analog_options: none +OK span: 2 (BRI_2) type: isdn physical_status: ok signaling_status: DOWN chan_count: 3 dialplan: XML context: from-bri dial_regex: fail_dial_regex: hold_music: analog_options: none freeswitch at localhost.localdomain> ftdm dump 1 1 span_id: 1 chan_id: 1 physical_span_id: 1 physical_chan_id: 1 physical_status: ok physical_status_red: 0 physical_status_yellow: 0 physical_status_rai: 0 physical_status_blue: 0 physical_status_ais: 0 physical_status_general: 0 signaling_status: UP type: B state: DOWN last_state: RESTART txgain: 0.00 rxgain: 0.00 cid_date: cid_name: cid_num: ani: aniII: dnis: rdnis: cause: NONE session: (none) -- States -- -- Function -- -- Location -- -- Time Offset -- DOWN => RESTART [on_dchan_up] [ftmod_libpri.c:1520] 0ms RESTART => DOWN [state_advance] [ftmod_libpri.c:685] 1ms Time since last state change: 4138429ms [root at localhost bin]# cat ../conf/freetdm.conf [span zt BRI_1] trunk_type => bri_ptmp b-channel => 1-2 d-channel => 3 [span zt BRI_2] trunk_type => bri_ptmp b-channel => 1-2 d-channel => 3 [root at localhost bin]# cat ../conf/autoload_configs/freetdm.conf.xml WHILE OFFHOOK THE NT PHONE freeswitch at internal> 2012-04-27 19:45:25.465395 [DEBUG] ftmod_libpri.c:146 2012-04-27 19:45:25.465395 [DEBUG] ftmod_libpri.c:146 < Protocol Discriminator: Q.931 (8) len=8 2012-04-27 19:45:25.465395 [DEBUG] ftmod_libpri.c:146 < TEI=65 Call Ref: len= 1 (reference 1/0x1) (Sent from originator) 2012-04-27 19:45:25.465395 [DEBUG] ftmod_libpri.c:146 < Message Type: DISCONNECT (69) 2012-04-27 19:45:25.465395 [DEBUG] ftmod_libpri.c:146 < [08 02 80 90] 2012-04-27 19:45:25.465395 [DEBUG] ftmod_libpri.c:146 < Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: User (0) 2012-04-27 19:45:25.465395 [DEBUG] ftmod_libpri.c:146 < Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] 2012-04-27 19:45:25.465395 [DEBUG] ftmod_libpri.c:146 Received message for call 0xb77119a0 on link 0x87cbb08 TEI/SAPI 65/0 2012-04-27 19:45:25.465395 [DEBUG] ftmod_libpri.c:146 -- Processing IE 8 (cs0, Cause) 2012-04-27 19:45:25.465395 [DEBUG] ftmod_libpri.c:146 q931.c:8707 post_handle_q931_message: Call 1 enters state 12 (Disconnect Indication). Hold state: Idle 2012-04-27 19:45:25.465395 [DEBUG] ftmod_libpri.c:146 q931.c:6837 q931_hangup: Hangup other cref:1 2012-04-27 19:45:25.465395 [DEBUG] ftmod_libpri.c:146 q931.c:6594 __q931_hangup: ourstate Disconnect Indication, peerstate Disconnect Request, hold-state Idle 2012-04-27 19:45:25.465395 [DEBUG] ftmod_libpri.c:146 q931.c:5703 q931_release: Call 1 enters state 19 (Release Request). Hold state: Idle 2012-04-27 19:45:25.465395 [DEBUG] ftmod_libpri.c:146 Sending message for call 0xb77119a0 on call->link: 0x87cbb08 with TEI/SAPI 65/0 2012-04-27 19:45:25.465395 [DEBUG] ftmod_libpri.c:146 2012-04-27 19:45:25.465395 [DEBUG] ftmod_libpri.c:146 > DL-DATA request 2012-04-27 19:45:25.465395 [DEBUG] ftmod_libpri.c:146 > Protocol Discriminator: Q.931 (8) len=8 2012-04-27 19:45:25.465395 [DEBUG] ftmod_libpri.c:146 > TEI=65 Call Ref: len= 1 (reference 1/0x1) (Sent to originator) 2012-04-27 19:45:25.465395 [DEBUG] ftmod_libpri.c:146 > Message Type: RELEASE COMPLETE (90) 2012-04-27 19:45:25.465395 [DEBUG] ftmod_libpri.c:146 2012-04-27 19:45:25.465395 [DEBUG] ftmod_libpri.c:146 > Protocol Discriminator: Q.931 (8) len=8 2012-04-27 19:45:25.465395 [DEBUG] ftmod_libpri.c:146 > TEI=65 Call Ref: len= 1 (reference 1/0x1) (Sent to originator) 2012-04-27 19:45:25.465395 [DEBUG] ftmod_libpri.c:146 > Message Type: RELEASE COMPLETE (90) 2012-04-27 19:45:25.465395 [DEBUG] ftmod_libpri.c:146 > [08 02 81 90] 2012-04-27 19:45:25.465395 [DEBUG] ftmod_libpri.c:146 > Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1) 2012-04-27 19:45:25.465395 [DEBUG] ftmod_libpri.c:146 > Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] 2012-04-27 19:45:30.005365 [DEBUG] ftmod_libpri.c:146 2012-04-27 19:45:30.005365 [DEBUG] ftmod_libpri.c:146 < Protocol Discriminator: Q.931 (8) len=17 2012-04-27 19:45:30.005365 [DEBUG] ftmod_libpri.c:146 < TEI=65 Call Ref: len= 1 (reference 1/0x1) (Sent from originator) 2012-04-27 19:45:30.005365 [DEBUG] ftmod_libpri.c:146 < Message Type: SETUP (5) 2012-04-27 19:45:30.005365 [DEBUG] ftmod_libpri.c:146 < [04 03 80 90 a3] 2012-04-27 19:45:30.005365 [DEBUG] ftmod_libpri.c:146 < Bearer Capability (len= 5) [ Ext: 1 Coding-Std: 0 Info transfer capability: Speech (0) 2012-04-27 19:45:30.005365 [DEBUG] ftmod_libpri.c:146 < Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) 2012-04-27 19:45:30.005365 [DEBUG] ftmod_libpri.c:146 < User information layer 1: A-Law (35) 2012-04-27 19:45:30.005365 [DEBUG] ftmod_libpri.c:146 < [6c 02 01 80] 2012-04-27 19:45:30.005365 [DEBUG] ftmod_libpri.c:146 < Calling Number (len= 4) [ Ext: 0 TON: Unknown Number Type (0) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) 2012-04-27 19:45:30.005365 [DEBUG] ftmod_libpri.c:146 < Presentation: Presentation permitted, user number not screened (0) '' ] 2012-04-27 19:45:30.005365 [DEBUG] ftmod_libpri.c:146 < [7d 02 91 81] 2012-04-27 19:45:30.005365 [DEBUG] ftmod_libpri.c:146 < IE: High-layer Compatibility (len = 4) 2012-04-27 19:45:30.005365 [DEBUG] ftmod_libpri.c:146 Received message for call 0xb77119a0 on link 0x87cbb08 TEI/SAPI 65/0 2012-04-27 19:45:30.005365 [DEBUG] ftmod_libpri.c:146 -- Processing Q.931 Call Setup 2012-04-27 19:45:30.005365 [DEBUG] ftmod_libpri.c:146 -- Processing IE 4 (cs0, Bearer Capability) 2012-04-27 19:45:30.005365 [DEBUG] ftmod_libpri.c:146 -- Processing IE 108 (cs0, Calling Party Number) 2012-04-27 19:45:30.005365 [DEBUG] ftmod_libpri.c:146 -- Processing IE 125 (cs0, High-layer Compatibility) 2012-04-27 19:45:30.005365 [DEBUG] ftmod_libpri.c:146 !! No handler for IE 125 (cs0, High-layer Compatibility) From miha at softnet.si Mon May 21 14:45:25 2012 From: miha at softnet.si (Miha) Date: Mon, 21 May 2012 12:45:25 +0200 Subject: [Freeswitch-users] Advance SBC In-Reply-To: <4FBA1CAA.3070001@softnet.si> References: <4FBA1CAA.3070001@softnet.si> Message-ID: <4FBA1CC5.4020907@softnet.si> Hi, I am looking on wiki about implementing SBC with mod_easyroute and mod_lcr (http://wiki.freeswitch.org/wiki/Advance_SBC_with_mod_lcr_and_mod_easyroute). I have configured as is written on wiki in everything works. Just one question. Should mod_lcr automatically be routing to less cost destination if first destination is down (this does not happend)? Regards, Miha From Rob.Moore at Aeriandi.com Mon May 21 15:01:58 2012 From: Rob.Moore at Aeriandi.com (Rob Moore) Date: Mon, 21 May 2012 11:01:58 +0000 Subject: [Freeswitch-users] Eaves Drop In-Reply-To: References: <49C5FCA19A8A114493EBAACA42FE5899105437F7@1AERDCEXCHMBX1.AER.AERCO.local> Message-ID: <49C5FCA19A8A114493EBAACA42FE589910575536@1AERDCEXCHMBX1.AER.AERCO.local> Hi Anthony, r your reply, sorry it's taken some time for me to pick this one up again, we've been dealing with a few more urgent projects away from this at our end that has taken up my attention. So I've attempted looking at this list, although it lists the bugs it doesn't seem to give any functionality to be able to remove the media bug. Looking at the Thanks fo eavesdrop app a little more it seems we can disconnect the bug using DTMF *. This works fine when using a handset but I've tried to recreate this using send_dtmf or uuid_send_dtmf and although you hear the DTMF the app doesn't seem to register this. Tried sending from both channels the eavesdropper and eavesdropped neither seems to actually remove the bug. Would anybody be able to let me know what the * is actually doing within Eavesdrop? I'd rather be able to call this directly from the API rather than send some unnecessary DTMF. Can anyone shed any light on what I need? Thanks Rob From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: 28 April 2012 02:06 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Eaves Drop There are commands to list and clear media bugs from a certian uuid. I forgot the name but you should see it in mod commands. On Apr 27, 2012 6:35 AM, "Rob Moore" > wrote: Hi All, Hopefully looking for a quick answer here. We are using the Eavesdrop function to supply a "call barging" service for a call centre we are running through a web UI but we have a few problems around transfers where we park the call and switching the barge from one agent to another. Basically I need some method of removing the media bug from the agents channel. I've searched the Wiki and done a few googles but I can't seem to find anything that we can call to remove a bug. We're doing all of our control of freeswitch through lua so anything that we can call from the CLI would be great. Anyone have any suggestions? Thanks in advance! Rob _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120521/20dd96b3/attachment-0001.html From tarik.bts.gi at gmail.com Mon May 21 16:18:03 2012 From: tarik.bts.gi at gmail.com (ghallab) Date: Mon, 21 May 2012 12:18:03 +0000 Subject: [Freeswitch-users] ignore_display_updates meaning an polycom problem (urgent please) In-Reply-To: <1FFF97C269757C458224B7C895F35F150D20D7@cantor.std.visionutv.se> References: <1FFF97C269757C458224B7C895F35F150D20D7@cantor.std.visionutv.se> Message-ID: <4FBA327B.9050501@gmail.com> Could you tell me what is the impact of each one (transferred using blind or supervised transfer)? On 05/21/2012 09:22 AM, Peter Olsson wrote: > > I guess it mostly depends on the Polycom device. For instance, it's a > big difference if the call is transferred using blind or supervised > transfer. You will probably need to collect the SIP traffic, and check > what actually happens when doing this scenario. > > Mvh > > Peter Olsson > > *Fr?n:*freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] *F?r *ghallab > *Skickat:* den 21 maj 2012 12:07 > *Till:* FreeSWITCH Users Help > *?mne:* [Freeswitch-users] ignore_display_updates meaning an polycom > problem (urgent please) > > Hi guys, > > Can you explain me what well happen if I set the variable > "ignore_display_updates=true" before I make bridge action and what > will happen if I set it to false? because I have a problem with FS and > Polycom 550: when A call from outside of the FS network to B which is > inside, then B answer and transfer the call to C which is inside > also, when C's phone is ringing he see B's caller ID and when he > answer, he see A's caller ID. Me I want that he see always caller ID > of A. I tried to set "ignore_display_updates=true" in this case B see > just C's caller ID when his is ringing and when he answer. There is > any way to fix this issue? I am thinking about to set > /"effective_caller_id_number={$origination_caller_id_number}" but I am > not sure if this will work. / > > > > > > !DSPAM:4fba049932768390320688! > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120521/60b26a08/attachment.html From peter.olsson at visionutveckling.se Mon May 21 15:54:44 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Mon, 21 May 2012 11:54:44 +0000 Subject: [Freeswitch-users] ignore_display_updates meaning an polycom problem (urgent please) Message-ID: <1FFF97C269757C458224B7C895F35F150D23E4@cantor.std.visionutv.se> I'm not sure how Polycom does their transfers, but usually these are the differences; When doing blind transfer, the phone has one call leg, and then sends a SIP REFER (to FreeSWITCH in this case), telling it to take that call and blindly transfer it to another extension. In this case the original CallerID should be transferred correctly. When doing supervised/consultation transfer, the phone will start by making a new call to the extension you want to transfer to. In this state the call will be from the phone doing the transfer, to the phone that is the destination (and the caller-id on the destination phone will be the transferers phone). When the destination phone picks up, the transferer says something like "Ok, here is a call for you", and then completes the transfer. Now it will probably send a REFER with replaces tags to FreeSWITCH, telling it to drop the two call legs to the transferer, and instead directly bridge the incoming external call to the transfer destination (these two legs already exist, so FS will just start bridging the RTP for these channels). As soon as this has happened, FS will send a display update, since the CallerID actually changed now, from the transferers phone to the original external call. As I said, I'm not sure if this is how Polycom does their transfers, but the basic difference between a blind and supervised transfer is as described above. So in this case, if you want to do a supervised transfer, and let the transfer target get the original callerID directly (before the call are actually transferred), I think you will need to check out the entire SIP call flow, and then take appropiate actions (either change config in FS, and/or change config on the phones). However, most people expect it to work as it does by default... /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r ghallab Skickat: den 21 maj 2012 14:18 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] ignore_display_updates meaning an polycom problem (urgent please) Could you tell me what is the impact of each one ( transferred using blind or supervised transfer)? On 05/21/2012 09:22 AM, Peter Olsson wrote: I guess it mostly depends on the Polycom device. For instance, it's a big difference if the call is transferred using blind or supervised transfer. You will probably need to collect the SIP traffic, and check what actually happens when doing this scenario. Mvh Peter Olsson Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r ghallab Skickat: den 21 maj 2012 12:07 Till: FreeSWITCH Users Help ?mne: [Freeswitch-users] ignore_display_updates meaning an polycom problem (urgent please) Hi guys, Can you explain me what well happen if I set the variable "ignore_display_updates=true" before I make bridge action and what will happen if I set it to false? because I have a problem with FS and Polycom 550: when A call from outside of the FS network to B which is inside, then B answer and transfer the call to C which is inside also, when C's phone is ringing he see B's caller ID and when he answer, he see A's caller ID. Me I want that he see always caller ID of A. I tried to set "ignore_display_updates=true" in this case B see just C's caller ID when his is ringing and when he answer. There is any way to fix this issue? I am thinking about to set "effective_caller_id_number={$origination_caller_id_number}" but I am not sure if this will work. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4fba232f32763870889139! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120521/f551f33c/attachment-0001.html From vipkilla at gmail.com Mon May 21 16:49:03 2012 From: vipkilla at gmail.com (Vik Killa) Date: Mon, 21 May 2012 08:49:03 -0400 Subject: [Freeswitch-users] Polycom blind transfer uses unexpected dialplan context Message-ID: I have a registered UA with the directory variable 'user_context' set to 'the.domain.com' When the registered UA dials a number, it first goes through the dialplan context 'the.domain.com' as expected, depending on the destination number dialed, the call will be transferred to another context, in this example 'routing' context. When the UA performs an attended transfer, the dialed number goes through the contexts as expected (the.domain.com->routing) but when performing a blind transfer the dialed number starts in the 'routing' context. Why wouldn't the dialed number follow the UA's 'user-context' variable and start in the 'the.domain.com' context? This is happening on a Polycom 330 with today's FS GIT. From vipkilla at gmail.com Mon May 21 16:54:28 2012 From: vipkilla at gmail.com (Vik Killa) Date: Mon, 21 May 2012 08:54:28 -0400 Subject: [Freeswitch-users] Polycom blind transfer uses unexpected dialplan context In-Reply-To: References: Message-ID: I found the solution was to set the channel variable like: force_transfer_context=the.domain.com On Mon, May 21, 2012 at 8:49 AM, Vik Killa wrote: > I have a registered UA with the directory variable 'user_context' set > to 'the.domain.com' > When the registered UA dials a number, it first goes through the > dialplan context 'the.domain.com' as expected, depending on the > destination number dialed, the call will be transferred to another > context, in this example 'routing' context. When the UA performs an > attended transfer, the dialed number goes through the contexts as > expected (the.domain.com->routing) but when performing a blind > transfer the dialed number starts in the 'routing' context. Why > wouldn't the dialed number follow the UA's 'user-context' variable and > start in the 'the.domain.com' context? This is happening on a Polycom > 330 with today's FS GIT. From acrow at integrafin.co.uk Mon May 21 19:09:14 2012 From: acrow at integrafin.co.uk (Alex Crow) Date: Mon, 21 May 2012 16:09:14 +0100 Subject: [Freeswitch-users] Spandsp mulated modems inbound/no /dev/FS* device? In-Reply-To: References: <4FA24100.7040908@integrafin.co.uk> <20120503151946.GA824@eagle.cupis.co.uk> <4FA3EE97.1090908@integrafin.co.uk> <4FA596BF.4090703@integrafin.co.uk> <4FA635AE.5050006@integrafin.co.uk> <4FA66862.1090300@coppice.org> <4FA6871F.5050909@puzzled.xs4all.nl> <4FA696B1.8050706@coppice.org> <4FB553A3.5010807@integrafin.co.uk> <4FB61319.6060205@integrafin.co.uk> Message-ID: <4FBA5A9A.4090109@integrafin.co.uk> On 21/05/12 08:09, Anita Hall wrote: > Hi Alex > > I got ptlib, opan and t38modem compile and installed. So far so good. > Is there a wiki entry you have made with details of your FS + T38modem > + Hylafax set-up, configuration files etc. > > I was able to dig up some stuff from your mails in the list but some > wiki entry will help me and others a lot. > > My team member has put up a Wiki entry at > http://wiki.freeswitch.org/wiki/T38modem > > It will help us a lot if you could do a quick edit and copy/paste of > your config. > > Many many thanks! > > regards, > Anita > > Hi, I posted up my inbound and outbound dialplan dialplans on the wiki. I think the absolute_codec_string would not be required in the case of bridging to FreeTDM instead of an IP gateway. Please try it. Cheers Alex -- This message is intended only for the addressee and may contain confidential information. Unless you are that person, you may not disclose its contents or use it in any way and are requested to delete the message along with any attachments and notify us immediately. "Transact" is operated by Integrated Financial Arrangements plc Domain House, 5-7 Singer Street, London EC2A 4BQ Tel: (020) 7608 4900 Fax: (020) 7608 5300 (Registered office: as above; Registered in England and Wales under number: 3727592) Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856) From anthony.minessale at gmail.com Mon May 21 19:16:03 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 21 May 2012 10:16:03 -0500 Subject: [Freeswitch-users] Spandsp mulated modems inbound/no /dev/FS* device? In-Reply-To: <4FB553A3.5010807@integrafin.co.uk> References: <4FA24100.7040908@integrafin.co.uk> <20120503151946.GA824@eagle.cupis.co.uk> <4FA3EE97.1090908@integrafin.co.uk> <4FA596BF.4090703@integrafin.co.uk> <4FA635AE.5050006@integrafin.co.uk> <4FA66862.1090300@coppice.org> <4FA6871F.5050909@puzzled.xs4all.nl> <4FA696B1.8050706@coppice.org> <4FB553A3.5010807@integrafin.co.uk> Message-ID: On Thu, May 17, 2012 at 2:38 PM, Alex Crow wrote: > On 17/05/12 14:51, Anita Hall wrote: >> Hi Alex >> >> Did you get the answer to your first question ? Did you succeed in >> using the emulated modem option for taking fax calls to hylafax ? >> >> I put the following in spandsp.conf.xml and got /dev/FS[0-4] devices >> as soft links to /dev/pts/[4-8]. My freeswitch is running as root user >> so I did not face the issue you did. >> >> >> >> >> >> What next? Could you point me to some doc in hylafax? >> >> (And before Steve lashes out at me again, I must clarify, I do not >> want to just play around, but my boss is clueless and wants me to >> evaluate hylafax :() >> >> regards, >> Anita >> I need to stamp out some FUD 1) It works going both ways. Its been complete for some time and in use. 2) I can't find anything but helpful comments from Steve. >> > > Anita, > > The answer I got was that inbound is not yet supported by that > mechanism, so I'd advise trying the Hylafax/T38Modem option if you need > to use Hylafax. The T38Modem site provides tarballs of the OPAL and > required dependencies to build it. The one main advantage we see for it > is that there are quite a few free and commercial clients for Hylafax > that can behave as a printer in Windows, so the user selects to print > say a .doc file, chooses the fax printer, enters a number and off it > goes. We used WHFC with XP, works great. We even have chaps batch faxing > 100+ faxes a day using it from Excel with a macro. > > The other route for inbound is well documented, it using the inbuilt > FreeSWITCH faxing and sending to email. Outbound is equally simple but > not IMHO as easy for our end users as Hylafax (email-to-fax is simple > but printing to fax from Windows not so simple). But with some work on > your side, the spandsp stuff could probably do anything HF can do. > > As I said, Hylafax is excellent software and has been deployed on major > production sites for probably 15 years+ with billions of total faxes > delivered. We've used it for more than ten years with about 400 faxes > in/out per day (which is piddling compared with some other sites, 10,000 > per day is not difficult). If you want to avoid using t38modem you'd > probably have to set up a dedicated HF box with an ISDN/POTS > multichannel card (and if you want to fax from/to FS use a BRI or T1/E1 > card to send audio to and from it). There are other ways of integrating > it with backoffice systems including FTP, email, etc. > > To try to deploy a business fax system without first evaluating HylaFAX > is in my opinion insane, so maybe your boss has a point. If, however, > your boss is the one driving you to ask all these very diverse questions > on this list (and they are indeed frequent and often completely > unrelated to each other), perhaps you should ask him/her to focus on > those that are his top priorities, and pick the first 3 to 5 and focus > on those first, or in fact just compose a list of required features, > post them up in a single email rather than 50, and ask "is this possible?". > > Just being a bit more targeted with your questions and how they link > together might elicit more responses. Try to pick one solution at a > time, test it, ask questions on that and stick with it for a bit and I > think you'll get more help. > > Cheers > > Alex > > > -- > This message is intended only for the addressee and may contain > confidential information. ?Unless you are that person, you may not > disclose its contents or use it in any way and are requested to delete > the message along with any attachments and notify us immediately. > > "Transact" is operated by Integrated Financial Arrangements plc > Domain House, 5-7 Singer Street, London ?EC2A 4BQ > Tel: (020) 7608 4900 Fax: (020) 7608 5300 > (Registered office: as above; Registered in England and Wales under number: 3727592) > Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856) > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From anthony.minessale at gmail.com Mon May 21 19:52:24 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 21 May 2012 10:52:24 -0500 Subject: [Freeswitch-users] Eaves Drop In-Reply-To: <49C5FCA19A8A114493EBAACA42FE589910575536@1AERDCEXCHMBX1.AER.AERCO.local> References: <49C5FCA19A8A114493EBAACA42FE5899105437F7@1AERDCEXCHMBX1.AER.AERCO.local> <49C5FCA19A8A114493EBAACA42FE589910575536@1AERDCEXCHMBX1.AER.AERCO.local> Message-ID: uuid_broadcast remove_bugs:: On Mon, May 21, 2012 at 6:01 AM, Rob Moore wrote: > Hi Anthony, > > r your reply, sorry it?s taken some time for me to pick this one up again, > we?ve been dealing with a few more urgent projects away from this at our end > that has taken up my attention. > > > > So I?ve attempted looking at this list, although it lists the bugs it > doesn?t seem to give any functionality to be able to remove the media bug. > > > > Looking at the > > Thanks fo > > eavesdrop app a little more it seems we can disconnect the bug using DTMF *. > This works fine when using a handset but I?ve tried to recreate this using > send_dtmf or uuid_send_dtmf? and although you hear the DTMF the app doesn?t > seem to register this. Tried sending from both channels the eavesdropper and > eavesdropped neither seems to actually remove the bug. > > > > Would anybody be able to let me know what the * is actually doing within > Eavesdrop? I?d rather be able to call this directly from the API rather than > send some unnecessary DTMF. > > > > Can anyone shed any light on what I need? > > > > Thanks > > > > Rob > > > > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony > Minessale > Sent: 28 April 2012 02:06 > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Eaves Drop > > > > There are commands to list and clear media bugs from a certian uuid.? I > forgot the name but you should see it in mod commands. > > On Apr 27, 2012 6:35 AM, "Rob Moore" wrote: > > Hi All, > > > > Hopefully looking for a quick answer here. We are using the Eavesdrop > function to supply a ?call barging? service for a call centre we are running > through a web UI but we have a few problems around transfers where we park > the call and switching the barge from one agent to another. > > > > Basically I need some method of removing the media bug from the agents > channel. I?ve searched the Wiki and done a few googles but I can?t seem to > find anything that we can call to remove a bug. > > > > We?re doing all of our control of freeswitch through lua so anything that we > can call from the CLI would be great. > > > > Anyone have any suggestions? > > > > Thanks in advance! > > > > Rob > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From msc at freeswitch.org Mon May 21 20:14:44 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 21 May 2012 09:14:44 -0700 Subject: [Freeswitch-users] Call_group intercept In-Reply-To: <4FB9E602.8070708@softnet.si> References: <4FB9E602.8070708@softnet.si> Message-ID: I think that perhaps you are misunderstanding what the instructions on that wiki page are doing. The var "callgroup" is set in the user's directory entry. Did you do that for each user in your pickup group? -MC On Sun, May 20, 2012 at 11:51 PM, Miha wrote: > HI, > > I read wiki tutorial about call group intercept ( > http://wiki.freeswitch.org/wiki/Callgroup_intercept) ans also implemented > it. (If I make call intercept which is not for callgroup it works) > > I put this in public dialplan, so that every call UUID will be inserted in > mysql database. > > > > > > > > When call hit public dialplan I can not access variable callgroup. As > callgroup variable is not set I can not make right mysql insert. What is > the best way to deal with this? > > THanks! > > Miha > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120521/9a55eb5b/attachment-0001.html From msc at freeswitch.org Mon May 21 20:22:05 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 21 May 2012 09:22:05 -0700 Subject: [Freeswitch-users] BYE message is not relayed to the UAC In-Reply-To: References: Message-ID: Your best bet is to use a pastebin for this sort of thing. The FreeSWITCH project maintains their own at pastebin.freeswitch.org. There are numerous others (pastie, pastebin.com, etc.) After you put all the information in a pastebin you'll get a handy URL that you can include in the email thread. One other advantage to pastebins is that you can refer to a specific line number in paste. ("Look at the BYE message at line xxx - it looks unusual...") -MC On Mon, May 21, 2012 at 12:26 AM, Arif Hossain wrote: > Hi, > > We have the following network architecture : > > > UAC1------------------------->kamailio-------------------->VoipSwitch----->PSTN---------->Phone1 > (Sip Client) > > Now UAC1 calls Phone1 and everything is ok. If UAC1 hangs up session > is terminated cleanly. > But if Phone1 hangs up the BYE message which comes to kamailio and > goes back to VoipSwitch > instead of relayed to UAC1 . > > So The session becomes a zombie one, And UAC1 unfortunately gets > billed for a session > which should be terminated. > > Following is the Call flow as seen from VoipSwitch : > > | | > | | | IP> | > |134.856 | INVITE SDP > | |(7890) ------------------> (5060) | > |134.858 | 407 Proxy Authentication Required > | |(7890) <------------------ (5060) | > |134.902 | ACK | > | |(7890) ------------------> (5060) | > |135.408 | INVITE SDP > | |(7890) ------------------> (5060) | > |135.414 | 100 Trying| > | |(7890) <------------------ (5060) | > |140.121 | 183 Session Progress SDP > | |(7890) <------------------ (5060) | > |140.184 | RTP (g729) | > | |(61868) <------------------ (5136) | > |141.295 | RTP (g729) | > | |(61868) ------------------> (5136) | > |153.701 | 200 OK SDP > | |(7890) <------------------ (5060) | > |153.713 | RTP (g729) | > | |(61868) ------------------> (5136) | > |154.126 | ACK | > | |(7890) ------------------> (5060) | > |159.988 | BYE | > | |(7890) <------------------ (5060) | > |160.031 | BYE | > | |(7890) ------------------> (5060) | > |160.478 | BYE | > | |(7890) ------------------> (5060) | > |161.412 | BYE | > | |(7890) ------------------> (5060) | > |163.280 | BYE | > | |(7890) ------------------> (5060) | > |167.015 | BYE | > | |(7890) ------------------> (5060) | > |170.750 | BYE | > | |(7890) ------------------> (5060) | > |174.481 | BYE | > | |(7890) ------------------> (5060) | > |178.216 | BYE | > | |(7890) ------------------> (5060) | > |181.952 | BYE | > | |(7890) ------------------> (5060) | > |185.687 | BYE | > | |(7890) ------------------> (5060) | > |188.018 | 408 Request Timeout > | |(7890) ------------------> (5060) | > |211.849 | BYE | > | |(7890) ------------------> (5060) | > |212.292 | BYE | > > > Sip Traces : > > kamailio------>VoipSwitch > > I'm posting only the offending BYE msg instead of full trace , because > of the mail will become difficult to read . If more traces needed, i > can post it. > > The following BYE message is sent by VoipSwitch: > > BYE sip:ipphone at 205.164.40.74 SIP/2.0 > Route: > CSeq: 2 BYE > Via: SIP/2.0/UDP 205.164.40.74:5060 > From: sip:008801673345531 at 205.164.40.74;tag=100528120745985872655137 > Call-ID: IqBknV19AuxW0jk.8BjuE4hyx93Ws9qS > To: "123456" >;tag=Zopl5lj5YiqyaSR5Le3QnfoR-G0NZAGG > Content-Length: 0 > > Kamailio instead of relaying the message, sends a BYE message towards > VoipSwitch: > > BYE sip:ipphone at 205.164.40.74 SIP/2.0 > Max-Forwards: 10 > CSeq: 2 BYE > Via: SIP/2.0/UDP 108.166.195.189:7890;branch=z9hG4bK4b2b.5d893e95.0 > Via: SIP/2.0/UDP 205.164.40.74:5060;rport=5060 > From: sip:008801673345531 at 205.164.40.74;tag=100528120745985872655137 > Call-ID: IqBknV19AuxW0jk.8BjuE4hyx93Ws9qS > To: "123456" >;tag=Zopl5lj5YiqyaSR5Le3QnfoR-G0NZAGG > Content-Length: 0 > > When the first BYE message comes from VoipSwitch , kamailio does the > following : > May 20 02:25:53 VOS20-108 /usr/local/sbin/kamailio[16442]: DEBUG: > [receive.c:289]: receive_msg: cleaning up > May 20 02:25:53 VOS20-108 /usr/local/sbin/kamailio[16442]: DEBUG: > [parser/sdp/sdp.c:751]: _sdp = 0x831bf10 > May 20 02:25:53 VOS20-108 /usr/local/sbin/kamailio[16442]: DEBUG: > [parser/sdp/sdp.c:753]: sdp = 0x83043dc > May 20 02:25:53 VOS20-108 /usr/local/sbin/kamailio[16442]: DEBUG: > [parser/sdp/sdp.c:755]: session = 0x8304504 > May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: > [parser/msg_parser.c:630]: SIP Request: > May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: > [parser/msg_parser.c:632]: method: > May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: > [parser/msg_parser.c:634]: uri: > > May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: > [parser/msg_parser.c:636]: version: > May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: > [parser/msg_parser.c:167]: get_hdr_field: cseq : <1> > > May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: > [parser/parse_via.c:1287]: Found param type 232, = > ; state=16 > May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: > [parser/parse_via.c:2300]: end of header reached, state=5 > May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: > [parser/msg_parser.c:515]: parse_headers: Via found, flags=2 > May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: > [parser/msg_parser.c:517]: parse_headers: this is the first via > May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: > [receive.c:145]: After parse_msg... > May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: > [receive.c:186]: preparing to run routing scripts... > May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: > [parser/parse_to.c:174]: DEBUG: add_param: > tag=arILprdVR1srJ76HHlt4BEc3XsyaWcZm > May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: > [parser/parse_to.c:803]: end of header reached, state=29 > May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: > [parser/msg_parser.c:187]: DEBUG: get_hdr_field: [76]; > uri=[sip:ipphone at 205.164.40.74] > May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: > [parser/msg_parser.c:189]: DEBUG: to body ["ipphone" > ] > May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: > [parser/msg_parser.c:201]: DEBUG: get_hdr_body : > content_length=0 > May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: > [parser/msg_parser.c:103]: found end of header > May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: > maxfwd [mf_funcs.c:66]: max_forwards header not found! > May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: > [parser/parse_to.c:174]: DEBUG: add_param: > tag=1905251223419334290936029 > May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: > [parser/parse_to.c:803]: end of header reached, state=29 > May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: > siputils [checks.c:76]: totag found > May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: rr > [loose.c:85]: is_preloaded: No > May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: > [socket_info.c:501]: grep_sock_info - checking if host==us: > 13==15 && [205.164.40.74] == [108.166.195.189] > May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: > [socket_info.c:504]: grep_sock_info - checking if port 7890 > matches port 5060 > May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: > [socket_info.c:501]: grep_sock_info - checking if host==us: > 13==15 && [205.164.40.74] == [108.166.195.189] > May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: > [socket_info.c:504]: grep_sock_info - checking if port 5060 > matches port 5060 > May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: > [forward.c:448]: check_self: host != me > May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: > [socket_info.c:501]: grep_sock_info - checking if host==us: > 15==15 && [108.166.195.189] == [108.166.195.189] > May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: > [socket_info.c:504]: grep_sock_info - checking if port 7890 > matches port 7890 > May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: rr > [loose.c:792]: Topmost route URI: > 'sip:108.166.195.189:7890;lr=on;nat=yes' is me > May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: > [parser/msg_parser.c:103]: found end of header > May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: rr > [loose.c:257]: No next Route HF found > May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: rr > [loose.c:811]: No next URI found > May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: rr > [loose.c:983]: params are <;lr=on;nat=yes> > May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: > siputils [checks.c:76]: totag found > May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: tm > [t_lookup.c:1379]: DEBUG: t_newtran: msg id=2501 , global msg id=2500 > , T on entrance=0xffffffff > May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: tm > [t_lookup.c:528]: t_lookup_request: start searching: hash=29177, > isACK=0 > May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: tm > [t_lookup.c:564]: DEBUG: proceeding to pre-RFC3261 transaction > matching > May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: tm > [t_lookup.c:711]: DEBUG: t_lookup_request: no transaction found > May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: tm > [t_hooks.c:374]: DBG: trans=0xb61626a4, callback type 1, id 0 entered > May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: > [msg_translator.c:204]: check_via_address(205.164.40.74, > 205.164.40.74, 0) > May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: tm > [t_funcs.c:388]: SER: new transaction fwd'ed > May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: > [usr_avp.c:646]: DEBUG:destroy_avp_list: destroying list (nil) > May 20 02:26:04 VOS20-108 last message repeated 5 times > May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: > [receive.c:289]: receive_msg: cleaning up > May 20 02:26:22 VOS20-108 /usr/local/sbin/kamailio[16445]: DEBUG: > [udp_server.c:486]: udp_rcv_loop: probing packet received from > 180.234.62.230 38722 > May 20 02:26:34 VOS20-108 /usr/local/sbin/kamailio[16450]: DEBUG: tm > [t_reply.c:1134]: ->>>>>>>>> T_code=0, new_code=408 > May 20 02:26:34 VOS20-108 /usr/local/sbin/kamailio[16450]: DEBUG: tm > [t_reply.c:1636]: DEBUG: relay_reply: branch=0, save=0, relay=0 > May 20 02:26:34 VOS20-108 /usr/local/sbin/kamailio[16450]: DEBUG: > [msg_translator.c:204]: check_via_address(205.164.40.74, > 205.164.40.74, 0) > May 20 02:26:34 VOS20-108 /usr/local/sbin/kamailio[16450]: DEBUG: > [mem/shm_mem.c:105]: WARNING:vqm_resize: resize(0) called > May 20 02:26:34 VOS20-108 /usr/local/sbin/kamailio[16450]: DEBUG: tm > [t_hooks.c:288]: DBG: trans=0xb61626a4, callback type 128, id 0 > entered > May 20 02:26:54 VOS20-108 /usr/local/sbin/kamailio[16451]: WARNING: > [timer.c:450]: WARNING: our timer runs faster then real-time > (300000 ms / 4800 ticks our time .-> 299923 ms / 4798 ticks real time) > > OT: How do you guys maintain 80 column mails? i do not use a mail > client, use gmail mailbox. > -- > -aft > > > -- > -aft > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120521/c171d15d/attachment-0001.html From babak.freeswitch at gmail.com Mon May 21 21:10:53 2012 From: babak.freeswitch at gmail.com (babak yakhchali) Date: Mon, 21 May 2012 21:40:53 +0430 Subject: [Freeswitch-users] sofia current calls for a user Message-ID: Hi Is there any way to use sofia apis to find out if currently there is a call from or to a user? reading sofia code shows that its using sip_dialogs table to track user presence and call data but there is no api call such as (sofia ...) which shows that data. thanx -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120521/733e20bd/attachment.html From marketing at cluecon.com Mon May 21 21:23:45 2012 From: marketing at cluecon.com (Michael Collins) Date: Mon, 21 May 2012 10:23:45 -0700 Subject: [Freeswitch-users] FreeSWITCH News and Notes Message-ID: Monday, May 21, 2012 We're looking forward this week to a followup on a conversation that we began last Wednesday. After Fred Dixon gave a great demonstration of Big Blue Box we had our very own Ken Rice give an introduction to writing a simple dialplan module in C. This week Ken Rice will follow up on the discussion and answer any questions you may have. If you missed last week's discussion you can download the audio by clicking the "torrent" link on the past callspage. In ClueCon news we are happy to report that Norm Tomlins and Voice Networkare back to sponsor us again in 2012. We look forward to seeing Norm again this August. We have also posted a number of presentations on this year's schedule . More presentations are in the works, so stay tuned for further announcements. Visit the ClueCon site to register, or call us at 877.742.CLUE if you have any questions about speakers, sponsoring, or anything else related to the event. Only 77 days until ClueCon! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE cc12-0521 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120521/ce184f8c/attachment.html From gabe at gundy.org Mon May 21 22:27:34 2012 From: gabe at gundy.org (Gabriel Gunderson) Date: Mon, 21 May 2012 12:27:34 -0600 Subject: [Freeswitch-users] FreeSWITCH News and Notes In-Reply-To: References: Message-ID: On Mon, May 21, 2012 at 11:23 AM, Michael Collins wrote: > More presentations are in the works, so stay tuned for further > announcements. Visit the ClueCon site to register, or call us at > 877.742.CLUE if you have any questions about speakers, sponsoring, or > anything else related to the event. > > Only 77 days until ClueCon! I had a great time last year. Not only I'm going again this year, but I'm trying to get several of the local open source / telephony guys in my area to come out too. It goes without saying that I'm hoping to see many of you from this list there! Best, Gabe From blackc2004 at gmail.com Mon May 21 22:09:58 2012 From: blackc2004 at gmail.com (Cj B) Date: Mon, 21 May 2012 11:09:58 -0700 Subject: [Freeswitch-users] Ring group auto answers and then diconnects Message-ID: <2D163CA6-ABBF-46BA-9C02-26B9358490AC@gmail.com> Hi all, I've setup a pretty simple ring group using fusionpbx and multi-tenant. I have ext's 9001-9004 and a ring group of Ext 8003. All 4 exts are part of the ring group. When I dial the ring group ext it gets answered right away but none of the phones ever ring. Then after 30 seconds (the timeout) I get the greeting "Good bye" and then it disconnects. I've posted all the output that happens here: http://pastebin.com/dZjwKVSA Can someone please help me out, this seems like it should be a pretty simple thing to do, so i'm sure it's just a setup issue that I've done wrong! Thanks. Cj B -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120521/7c7fd661/attachment.html From msc at freeswitch.org Mon May 21 22:49:22 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 21 May 2012 11:49:22 -0700 Subject: [Freeswitch-users] Ring group auto answers and then diconnects In-Reply-To: <2D163CA6-ABBF-46BA-9C02-26B9358490AC@gmail.com> References: <2D163CA6-ABBF-46BA-9C02-26B9358490AC@gmail.com> Message-ID: I see that it is only calling a single extension, in this case x9001. (See line #117 of your pastebin.) Can you call extension 9001 directly? If so, do you see anything different in the console log? -MC On Mon, May 21, 2012 at 11:09 AM, Cj B wrote: > Hi all, > > I've setup a pretty simple ring group using fusionpbx and multi-tenant. I > have ext's 9001-9004 and a ring group of Ext 8003. All 4 exts are part of > the ring group. > > When I dial the ring group ext it gets answered right away but none of the > phones ever ring. Then after 30 seconds (the timeout) I get the greeting > "Good bye" and then it disconnects. > > I've posted all the output that happens here: > http://pastebin.com/dZjwKVSA > > Can someone please help me out, this seems like it should be a pretty > simple thing to do, so i'm sure it's just a setup issue that I've done > wrong! > > Thanks. > Cj B > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120521/4ab2a211/attachment.html From peter at uringme.com Mon May 21 22:57:47 2012 From: peter at uringme.com (peter at uringme.com) Date: Mon, 21 May 2012 11:57:47 -0700 (PDT) Subject: [Freeswitch-users] SAVPF and a=crypto Message-ID: <1337626667.74763.YahooMailClassic@web2819.biz.mail.ne1.yahoo.com> I'm trying out sipML5 ( http://code.google.com/p/sipml5/ ) with FS and I had a problem with their demo. Their code uses a=crypto lines on RTP/SAVPF and FS is bouncing it with: 2012-05-21 13:04:47.754843 [ERR] sofia_glue.c:4478 a=crypto in RTP/AVP, refer to rfc3711 The crypto line is in RTP/SAVPF, though, not RTP/AVP. Is this line coming up because it's not RTP/SAVP? Am I missing a module for RTP/SAVPF, or is this even supported? Following is a trace -- I'm calling x7779 on my server which just plays a voice prompt. INVITE sip:7779 at x.x.x.x SIP/2.0 Via: SIP/2.0/UDP 87.106.69.240:4060;branch=z9hG4bK-524287-1---a56eca302b8d6412;rport Via: SIP/2.0/TCP y.y.y.y:62676;branch=z9hG4bK1P36uFnLdbzHiEzWdVAd0G6nk6kdOdEt;rport=62676;received=y.y.y.y Max-Forwards: 69 Contact: "Peter Krause";+sip.ice To: From: ;tag=gz6bNBeAkHM7eAR2eoJG Call-ID: 434d70e1-5cc1-d096-442c-499c8cd600ad CSeq: 6437 INVITE Content-Type: application/sdp Organization: Doubango Telecom User-Agent: IM-client/OMA1.0 sipML5/v0.0.0000.0 Content-Length: 3483 v=0 o=- 275367574 1 IN IP4 127.0.0.1 s=webrtc (chrome 20.0.1127.0) - Doubango Telecom (sipML5 r000) t=0 0 m=audio 54450 RTP/SAVPF 103 104 0 8 106 105 13 126 c=IN IP4 y.y.y.y a=rtcp:54451 IN IP4 y.y.y.y a=candidate:0 1 udp 2130706432 192.168.1.8 50256 typ host network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username 2C/PMnZyAP2pLbP0 password 0xigYC8ZZibTPCejG5JKwjBN generation 0 a=candidate:0 2 udp 2130706432 192.168.1.8 50257 typ host network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username 2BPqrxt6Vp+wJNL3 password UEWxx9yByE/JlsmFHa0zkbTs generation 0 a=candidate:0 1 udp 1912602624 y.y.y.y 54450 typ srflx network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username 2C/PMnZyAP2pLbP0 password 0xigYC8ZZibTPCejG5JKwjBN generation 0 a=candidate:0 2 udp 1912602624 y.y.y.y 54451 typ srflx network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username 2BPqrxt6Vp+wJNL3 password UEWxx9yByE/JlsmFHa0zkbTs generation 0 a=candidate:0 1 tcp 1694498816 192.168.1.8 62677 typ host network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username 2C/PMnZyAP2pLbP0 password 0xigYC8ZZibTPCejG5JKwjBN generation 0 a=candidate:0 2 tcp 1694498816 192.168.1.8 62678 typ host network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username 2BPqrxt6Vp+wJNL3 password UEWxx9yByE/JlsmFHa0zkbTs generation 0 a=mid:audio a=rtcp-mux a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:33ayFPGAoi+nni8kbYZfBO7tH3qI1SaH7ilFU7GT a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:Hd6rAXtyms1rJgC4OTJk/K4AGpnwLUn9Aqdx1sbd a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=ssrc:2075114348 cname:vcjaT0z7bAvF1TL3 a=ssrc:2075114348 mslabel:TOAV4oLn6QD7CfJieQ90FAEQB06orMEVm2na a=ssrc:2075114348 label:TOAV4oLn6QD7CfJieQ90FAEQB06orMEVm2na0 a=sendrecv m=video 54452 RTP/SAVPF 100 101 102 c=IN IP4 y.y.y.y a=rtcp:54453 IN IP4 y.y.y.y a=candidate:0 1 udp 2130706432 192.168.1.8 50258 typ host network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username PlVs0WI8o1Q05fcc password eakcTlAsBYwpmZvTc58E1thy generation 0 a=candidate:0 2 udp 2130706432 192.168.1.8 50259 typ host network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username gUagDr7SY/5Jxned password 6J+08YaMdKaer/95B9lfxCEM generation 0 a=candidate:0 1 udp 1912602624 y.y.y.y 54452 typ srflx network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username PlVs0WI8o1Q05fcc password eakcTlAsBYwpmZvTc58E1thy generation 0 a=candidate:0 2 udp 1912602624 y.y.y.y 54453 typ srflx network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username gUagDr7SY/5Jxned password 6J+08YaMdKaer/95B9lfxCEM generation 0 a=candidate:0 1 tcp 1694498816 192.168.1.8 62679 typ host network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username PlVs0WI8o1Q05fcc password eakcTlAsBYwpmZvTc58E1thy generation 0 a=candidate:0 2 tcp 1694498816 192.168.1.8 62680 typ host network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username gUagDr7SY/5Jxned password 6J+08YaMdKaer/95B9lfxCEM generation 0 a=mid:video a=rtcp-mux a=crypto:0 AES_CM_128_HMAC_SHA1_80 inline:RIxXHNE/5GtDfkXnAV4CLXBQGxPzlzBg3Xyg3Lfs a=rtpmap:100 VP8/90000 a=rtpmap:101 red/90000 a=rtpmap:102 ulpfec/90000 a=ssrc:2075114348 cname:vcjaT0z7bAvF1TL3 a=ssrc:2075114348 mslabel:TOAV4oLn6QD7CfJieQ90FAEQB06orMEVm2na a=ssrc:2075114348 label:TOAV4oLn6QD7CfJieQ90FAEQB06orMEVm2na0 a=sendrecv ------------------------------------------------------------------------ send 448 bytes to udp/[87.106.69.240]:4060 at 17:04:47.774032: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 87.106.69.240:4060;branch=z9hG4bK-524287-1---a56eca302b8d6412;rport=4060 Via: SIP/2.0/TCP y.y.y.y:62676;branch=z9hG4bK1P36uFnLdbzHiEzWdVAd0G6nk6kdOdEt;rport=62676;received=y.y.y.y From: ;tag=gz6bNBeAkHM7eAR2eoJG To: Call-ID: 434d70e1-5cc1-d096-442c-499c8cd600ad CSeq: 6437 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Content-Length: 0 ------------------------------------------------------------------------ 2012-05-21 13:04:47.754843 [NOTICE] switch_channel.c:816 New Channel sofia/external/7777 at uringme [120f19be-a367-11e1-b22e-57011ca30f08] 2012-05-21 13:04:47.754843 [DEBUG] sofia.c:4770 Channel sofia/external/7777 at uringme entering state [received][100] 2012-05-21 13:04:47.754843 [DEBUG] sofia.c:4781 Remote SDP: v=0 o=- 275367574 1 IN IP4 127.0.0.1 s=webrtc (chrome 20.0.1127.0) - Doubango Telecom (sipML5 r000) t=0 0 m=audio 54450 RTP/SAVPF 103 104 0 8 106 105 13 126 c=IN IP4 y.y.y.y a=rtcp:54451 IN IP4 y.y.y.y a=candidate:0 1 udp 2130706432 192.168.1.8 50256 typ host network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username 2C/PMnZyAP2pLbP0 password 0xigYC8ZZibTPCejG5JKwjBN generation 0 a=candidate:0 2 udp 2130706432 192.168.1.8 50257 typ host network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username 2BPqrxt6Vp+wJNL3 password UEWxx9yByE/JlsmFHa0zkbTs generation 0 a=candidate:0 1 udp 1912602624 y.y.y.y 54450 typ srflx network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username 2C/PMnZyAP2pLbP0 password 0xigYC8ZZibTPCejG5JKwjBN generation 0 a=candidate:0 2 udp 1912602624 y.y.y.y 54451 typ srflx network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username 2BPqrxt6Vp+wJNL3 password UEWxx9yByE/JlsmFHa0zkbTs generation 0 a=candidate:0 1 tcp 1694498816 192.168.1.8 62677 typ host network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username 2C/PMnZyAP2pLbP0 password 0xigYC8ZZibTPCejG5JKwjBN generation 0 a=candidate:0 2 tcp 1694498816 192.168.1.8 62678 typ host network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username 2BPqrxt6Vp+wJNL3 password UEWxx9yByE/JlsmFHa0zkbTs generation 0 a=mid:audio a=rtcp-mux a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:33ayFPGAoi+nni8kbYZfBO7tH3qI1SaH7ilFU7GT a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:Hd6rAXtyms1rJgC4OTJk/K4AGpnwLUn9Aqdx1sbd a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=ssrc:2075114348 cname:vcjaT0z7bAvF1TL3 a=ssrc:2075114348 mslabel:TOAV4oLn6QD7CfJieQ90FAEQB06orMEVm2na a=ssrc:2075114348 label:TOAV4oLn6QD7CfJieQ90FAEQB06orMEVm2na0 m=video 54452 RTP/SAVPF 100 101 102 c=IN IP4 y.y.y.y a=rtcp:54453 IN IP4 y.y.y.y a=candidate:0 1 udp 2130706432 192.168.1.8 50258 typ host network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username PlVs0WI8o1Q05fcc password eakcTlAsBYwpmZvTc58E1thy generation 0 a=candidate:0 2 udp 2130706432 192.168.1.8 50259 typ host network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username gUagDr7SY/5Jxned password 6J+08YaMdKaer/95B9lfxCEM generation 0 a=candidate:0 1 udp 1912602624 y.y.y.y 54452 typ srflx network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username PlVs0WI8o1Q05fcc password eakcTlAsBYwpmZvTc58E1thy generation 0 a=candidate:0 2 udp 1912602624 y.y.y.y 54453 typ srflx network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username gUagDr7SY/5Jxned password 6J+08YaMdKaer/95B9lfxCEM generation 0 a=candidate:0 1 tcp 1694498816 192.168.1.8 62679 typ host network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username PlVs0WI8o1Q05fcc password eakcTlAsBYwpmZvTc58E1thy generation 0 a=candidate:0 2 tcp 1694498816 192.168.1.8 62680 typ host network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username gUagDr7SY/5Jxned password 6J+08YaMdKaer/95B9lfxCEM generation 0 a=mid:video a=rtcp-mux a=crypto:0 AES_CM_128_HMAC_SHA1_80 inline:RIxXHNE/5GtDfkXnAV4CLXBQGxPzlzBg3Xyg3Lfs a=rtpmap:100 VP8/90000 a=rtpmap:101 red/90000 a=rtpmap:102 ulpfec/90000 a=ssrc:2075114348 cname:vcjaT0z7bAvF1TL3 a=ssrc:2075114348 mslabel:TOAV4oLn6QD7CfJieQ90FAEQB06orMEVm2na a=ssrc:2075114348 label:TOAV4oLn6QD7CfJieQ90FAEQB06orMEVm2na0 2012-05-21 13:04:47.754843 [ERR] sofia_glue.c:4478 a=crypto in RTP/AVP, refer to rfc3711 2012-05-21 13:04:47.754843 [DEBUG] switch_channel.c:2592 (sofia/external/7777 at uringme) Callstate Change DOWN -> HANGUP 2012-05-21 13:04:47.754843 [NOTICE] sofia.c:4998 Hangup sofia/external/7777 at uringme [CS_NEW] [INCOMPATIBLE_DESTINATION] From acrow at integrafin.co.uk Mon May 21 23:15:31 2012 From: acrow at integrafin.co.uk (Alex Crow) Date: Mon, 21 May 2012 20:15:31 +0100 Subject: [Freeswitch-users] Spandsp mulated modems inbound/no /dev/FS* device? In-Reply-To: References: <4FA24100.7040908@integrafin.co.uk> <20120503151946.GA824@eagle.cupis.co.uk> <4FA3EE97.1090908@integrafin.co.uk> <4FA596BF.4090703@integrafin.co.uk> <4FA635AE.5050006@integrafin.co.uk> <4FA66862.1090300@coppice.org> <4FA6871F.5050909@puzzled.xs4all.nl> <4FA696B1.8050706@coppice.org> <4FB553A3.5010807@integrafin.co.uk> Message-ID: <4FBA9453.2000601@integrafin.co.uk> On 21/05/12 16:16, Anthony Minessale wrote: > On Thu, May 17, 2012 at 2:38 PM, Alex Crow wrote: >> On 17/05/12 14:51, Anita Hall wrote: >>> Hi Alex >>> >>> Did you get the answer to your first question ? Did you succeed in >>> using the emulated modem option for taking fax calls to hylafax ? >>> >>> I put the following in spandsp.conf.xml and got /dev/FS[0-4] devices >>> as soft links to /dev/pts/[4-8]. My freeswitch is running as root user >>> so I did not face the issue you did. >>> >>> >>> >>> >>> >>> What next? Could you point me to some doc in hylafax? >>> >>> (And before Steve lashes out at me again, I must clarify, I do not >>> want to just play around, but my boss is clueless and wants me to >>> evaluate hylafax :() >>> >>> regards, >>> Anita >>> > I need to stamp out some FUD > > 1) It works going both ways. Its been complete for some time and in use. > 2) I can't find anything but helpful comments from Steve. > > Anthony. I won't comment on point 2 as I think it's unproductive, but I also have no idea how to use the emulated modems for inbound fax. If you could suggest a way that would be great. I was expecting to see a registered endpoint but nothing appeared, so I am lost as to what I should put in my dialplan to, for instance, direct a call from a PSTN gateway or FreeTDM endpoint to the virtual modem (so Hylafax can answer it). Also I have the problem that unless FreeSWITCH is run as root, the device nodes are not created. Unless in the last few weeks some docs have been updated on this, I'm still stuck with T38modem. Cheers Alex From msc at freeswitch.org Mon May 21 23:19:48 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 21 May 2012 12:19:48 -0700 Subject: [Freeswitch-users] sofia current calls for a user In-Reply-To: References: Message-ID: What are you trying to accomplish? It sounds like maybe mod_limit could be used, depending on exactly what you are trying to do. -MC On Mon, May 21, 2012 at 10:10 AM, babak yakhchali < babak.freeswitch at gmail.com> wrote: > Hi > Is there any way to use sofia apis to find out if currently there is a > call from or to a user? reading sofia code shows that its using sip_dialogs > table to track user presence and call data but there is no api call such as > (sofia ...) which shows that data. > thanx > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120521/8cab3c25/attachment.html From msc at freeswitch.org Mon May 21 23:23:39 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 21 May 2012 12:23:39 -0700 Subject: [Freeswitch-users] Run LUA script in different server In-Reply-To: References: Message-ID: Look at line #186 of your trace: 2012-05-21 07:56:49.735660 [ERR] mod_sofia.c:3957 Invalid Profile You need to figure out why your internal profile isn't running. Try "sofia profile internal restart" and see what happens. -MC On Sun, May 20, 2012 at 10:45 PM, Sanath Prasanna wrote: > Hi MC, > I did the change according to ure instruction. But error is coming. Here I > attached freeswitch.log file > > I change the confs as follows in BOX A. (Operator connected Freeswitch box) > BOX B ip is 10.22.29.253 > > vi /usr/local/freeswitch/conf/dialplan/default.xml > > > > > > > > Also add following to ACL file in BOX B > > Pls advice to solve the problem here. > Br, > Sanath > > > On Fri, May 18, 2012 at 9:57 AM, Michael Collins wrote: > >> If I understand your question correctly, yes you can do this. You can >> send calls from one FreeSWITCH server to another. Start here: >> http://wiki.freeswitch.org/wiki/Connect_Two_FreeSWITCH_Boxes >> >> Best way to learn is to get the FreeSWITCH books from Packt Publishing >> and just start hacking code. >> >> -MC >> >> >> On Thu, May 17, 2012 at 9:07 PM, Sanath Prasanna wrote: >> >>> Tx for advice MC & Anita. Can I do work around like this . >>> Another freeswitch instant will be start in other server & calls will be >>> transfer from operator connected freeswitch instance to this new freeswitch >>> instance & vise versa. Pls advice. >>> >>> >>> On Thu, May 17, 2012 at 5:05 PM, Anita Hall wrote: >>> >>>> You could run a Lua ESL server on a different machine but this will not >>>> be the same as running a Lua script. >>>> http://wiki.freeswitch.org/wiki/Event_Socket_Library >>>> >>>> regards, >>>> Anita >>>> >>>> >>>> >>>> On Thu, May 17, 2012 at 4:37 AM, Michael Collins wrote: >>>> >>>>> I don't think you can directly do what you are describing. However, >>>>> you might be able to use mod_httapi for this. There's some documentation on >>>>> the wiki and in the module. Keep in mind that this is a relatively new >>>>> module so we don't have lots of examples yet, so you'll probably be doing a >>>>> fair amount of research and testing. >>>>> >>>>> -MC >>>>> >>>>> >>>>> On Wed, May 16, 2012 at 5:59 AM, Sanath Prasanna >>>> > wrote: >>>>> >>>>>> Hi all, >>>>>> I have 2 servers. One server has SIP GW connection From Operator & >>>>>> IVR applications need to build in other server. How to call distributed LUA >>>>>> applications with Mysql Databases from the SIP GW server ? Pls advice. >>>>>> Main idea is, maintaining SIP connection in one server & all the IVR >>>>>> applications in other server. >>>>>> Br, >>>>>> Sanath >>>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> >> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120521/3b985269/attachment.html From krice at freeswitch.org Mon May 21 23:28:28 2012 From: krice at freeswitch.org (Ken Rice) Date: Mon, 21 May 2012 14:28:28 -0500 Subject: [Freeswitch-users] Spandsp mulated modems inbound/no /dev/FS* device? In-Reply-To: <4FBA9453.2000601@integrafin.co.uk> Message-ID: On 5/21/12 2:15 PM, "Alex Crow" wrote: > > > Also I have the problem that unless FreeSWITCH is run as root, the > device nodes are not created. > > Unless in the last few weeks some docs have been updated on this, I'm > still stuck with T38modem. > If freeswitch running as root and not running as root doesn't work, this should tell you that you have a permissions issue... The FreeSWITCH developers fixing your systems perms issues is beyond the scope of the FreeSWITCH project... Otherwise, the FreeSWITCH modems basically work just like t38modem... The only setting is how many of them to create then the only other settings is configuring your platform to properly allow freeswitch (or the user that freeswitch is running as) create those devices then configure hylafax to use them as normal. The steps to do this can vary from platform to platform (or even versions of the platform. For example how you do this on centos5 is not how you do this on centos6 due to changes in the platform.) K From aftnix at gmail.com Mon May 21 23:33:47 2012 From: aftnix at gmail.com (Arif Hossain) Date: Tue, 22 May 2012 01:33:47 +0600 Subject: [Freeswitch-users] BYE message is not relayed to the UAC In-Reply-To: References: Message-ID: On Mon, May 21, 2012 at 10:22 PM, Michael Collins wrote: > Your best bet is to use a pastebin for this sort of thing. The FreeSWITCH > project maintains their own at pastebin.freeswitch.org. There are numerous > others (pastie, pastebin.com, etc.) After you put all the information in a > pastebin you'll get a handy URL that you can include in the email thread. > One other advantage to pastebins is that you can refer to a specific line > number in paste. ("Look at the BYE message at line xxx - it looks > unusual...") > > -MC > freeswitch's pastebin asks for authentication. > On Mon, May 21, 2012 at 12:26 AM, Arif Hossain wrote: >> >> Hi, >> >> We have the following network architecture : >> >> >> UAC1------------------------->kamailio-------------------->VoipSwitch----->PSTN---------->Phone1 >> (Sip Client) >> >> Now UAC1 calls Phone1 and everything is ok. If UAC1 hangs up session >> is terminated cleanly. >> But if Phone1 hangs up the BYE message which ?comes to kamailio and >> goes back to VoipSwitch >> instead of relayed to UAC1 . >> >> So The session becomes a zombie one, And UAC1 unfortunately gets >> billed for a session >> which should be terminated. >> >> Following is the Call flow as seen from VoipSwitch : >> >> ?| ? ? ? ? ? ? ? ? ? ? ?| >> | ? ? ? ? | ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?| > IP> ? ? | >> |134.856 ?| ? ? ? ? INVITE SDP >> | ? ? ? ? |(7890) ? ------------------> ?(5060) ? | >> |134.858 ?| ? ? ? ? 407 Proxy Authentication Required >> | ? ? ? ? |(7890) ? <------------------ ?(5060) ? | >> |134.902 ?| ? ? ? ? ACK ? ? ? | >> | ? ? ? ? |(7890) ? ------------------> ?(5060) ? | >> |135.408 ?| ? ? ? ? INVITE SDP >> | ? ? ? ? |(7890) ? ------------------> ?(5060) ? | >> |135.414 ?| ? ? ? ? 100 Trying| >> | ? ? ? ? |(7890) ? <------------------ ?(5060) ? | >> |140.121 ?| ? ? ? ? 183 Session Progress SDP >> | ? ? ? ? |(7890) ? <------------------ ?(5060) ? | >> |140.184 ?| ? ? ? ? RTP (g729) ? ? ? ? ? ? ? ? ? ?| >> | ? ? ? ? |(61868) ?<------------------ ?(5136) ? | >> |141.295 ?| ? ? ? ? RTP (g729) ? ? ? ? ? ? ? ? ? ?| >> | ? ? ? ? |(61868) ?------------------> ?(5136) ? | >> |153.701 ?| ? ? ? ? 200 OK SDP >> | ? ? ? ? |(7890) ? <------------------ ?(5060) ? | >> |153.713 ?| ? ? ? ? RTP (g729) ? ? ? ? ? ? ? ? ? ?| >> | ? ? ? ? |(61868) ?------------------> ?(5136) ? | >> |154.126 ?| ? ? ? ? ACK ? ? ? | >> | ? ? ? ? |(7890) ? ------------------> ?(5060) ? | >> |159.988 ?| ? ? ? ? BYE ? ? ? | >> | ? ? ? ? |(7890) ? <------------------ ?(5060) ? | >> |160.031 ?| ? ? ? ? BYE ? ? ? | >> | ? ? ? ? |(7890) ? ------------------> ?(5060) ? | >> |160.478 ?| ? ? ? ? BYE ? ? ? | >> | ? ? ? ? |(7890) ? ------------------> ?(5060) ? | >> |161.412 ?| ? ? ? ? BYE ? ? ? | >> | ? ? ? ? |(7890) ? ------------------> ?(5060) ? | >> |163.280 ?| ? ? ? ? BYE ? ? ? | >> | ? ? ? ? |(7890) ? ------------------> ?(5060) ? | >> |167.015 ?| ? ? ? ? BYE ? ? ? | >> | ? ? ? ? |(7890) ? ------------------> ?(5060) ? | >> |170.750 ?| ? ? ? ? BYE ? ? ? | >> | ? ? ? ? |(7890) ? ------------------> ?(5060) ? | >> |174.481 ?| ? ? ? ? BYE ? ? ? | >> | ? ? ? ? |(7890) ? ------------------> ?(5060) ? | >> |178.216 ?| ? ? ? ? BYE ? ? ? | >> | ? ? ? ? |(7890) ? ------------------> ?(5060) ? | >> |181.952 ?| ? ? ? ? BYE ? ? ? | >> | ? ? ? ? |(7890) ? ------------------> ?(5060) ? | >> |185.687 ?| ? ? ? ? BYE ? ? ? | >> | ? ? ? ? |(7890) ? ------------------> ?(5060) ? | >> |188.018 ?| ? ? ? ? 408 Request Timeout >> | ? ? ? ? |(7890) ? ------------------> ?(5060) ? | >> |211.849 ?| ? ? ? ? BYE ? ? ? | >> | ? ? ? ? |(7890) ? ------------------> ?(5060) ? | >> |212.292 ?| ? ? ? ? BYE ? ? ? | >> >> >> Sip Traces : >> >> kamailio------>VoipSwitch >> >> I'm posting only the offending BYE msg instead of full trace , because >> of the mail will become difficult to read . If more traces needed, i >> can post it. >> >> The following BYE message is sent by VoipSwitch: >> >> BYE sip:ipphone at 205.164.40.74 SIP/2.0 >> Route: >> CSeq: 2 BYE >> Via: SIP/2.0/UDP 205.164.40.74:5060 >> From: sip:008801673345531 at 205.164.40.74;tag=100528120745985872655137 >> Call-ID: IqBknV19AuxW0jk.8BjuE4hyx93Ws9qS >> To: "123456" >> ;tag=Zopl5lj5YiqyaSR5Le3QnfoR-G0NZAGG >> Content-Length: 0 >> >> Kamailio instead of relaying the message, sends a BYE message towards >> VoipSwitch: >> >> BYE sip:ipphone at 205.164.40.74 SIP/2.0 >> Max-Forwards: 10 >> CSeq: 2 BYE >> Via: SIP/2.0/UDP 108.166.195.189:7890;branch=z9hG4bK4b2b.5d893e95.0 >> Via: SIP/2.0/UDP 205.164.40.74:5060;rport=5060 >> From: sip:008801673345531 at 205.164.40.74;tag=100528120745985872655137 >> Call-ID: IqBknV19AuxW0jk.8BjuE4hyx93Ws9qS >> To: "123456" >> ;tag=Zopl5lj5YiqyaSR5Le3QnfoR-G0NZAGG >> Content-Length: 0 >> >> When the first BYE message comes from VoipSwitch , kamailio does the >> following : >> May 20 02:25:53 VOS20-108 /usr/local/sbin/kamailio[16442]: DEBUG: >> [receive.c:289]: receive_msg: cleaning up >> May 20 02:25:53 VOS20-108 /usr/local/sbin/kamailio[16442]: DEBUG: >> [parser/sdp/sdp.c:751]: _sdp = 0x831bf10 >> May 20 02:25:53 VOS20-108 /usr/local/sbin/kamailio[16442]: DEBUG: >> [parser/sdp/sdp.c:753]: sdp = 0x83043dc >> May 20 02:25:53 VOS20-108 /usr/local/sbin/kamailio[16442]: DEBUG: >> [parser/sdp/sdp.c:755]: session = 0x8304504 >> May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: >> [parser/msg_parser.c:630]: SIP Request: >> May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: >> [parser/msg_parser.c:632]: ?method: ? >> May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: >> [parser/msg_parser.c:634]: ?uri: >> >> May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: >> [parser/msg_parser.c:636]: ?version: >> May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: >> [parser/msg_parser.c:167]: get_hdr_field: cseq : <1> >> >> May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: >> [parser/parse_via.c:1287]: Found param type 232, = >> ; state=16 >> May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: >> [parser/parse_via.c:2300]: end of header reached, state=5 >> May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: >> [parser/msg_parser.c:515]: parse_headers: Via found, flags=2 >> May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: >> [parser/msg_parser.c:517]: parse_headers: this is the first via >> May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: >> [receive.c:145]: After parse_msg... >> May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: >> [receive.c:186]: preparing to run routing scripts... >> May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: >> [parser/parse_to.c:174]: DEBUG: add_param: >> tag=arILprdVR1srJ76HHlt4BEc3XsyaWcZm >> May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: >> [parser/parse_to.c:803]: end of header reached, state=29 >> May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: >> [parser/msg_parser.c:187]: DEBUG: get_hdr_field: [76]; >> uri=[sip:ipphone at 205.164.40.74] >> May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: >> [parser/msg_parser.c:189]: DEBUG: to body ["ipphone" >> ] >> May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: >> [parser/msg_parser.c:201]: DEBUG: get_hdr_body : >> content_length=0 >> May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: >> [parser/msg_parser.c:103]: found end of header >> May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: >> maxfwd [mf_funcs.c:66]: max_forwards header not found! >> May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: >> [parser/parse_to.c:174]: DEBUG: add_param: >> tag=1905251223419334290936029 >> May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: >> [parser/parse_to.c:803]: end of header reached, state=29 >> May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: >> siputils [checks.c:76]: totag found >> May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: rr >> [loose.c:85]: is_preloaded: No >> May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: >> [socket_info.c:501]: grep_sock_info - checking if host==us: >> 13==15 && ?[205.164.40.74] == [108.166.195.189] >> May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: >> [socket_info.c:504]: grep_sock_info - checking if port 7890 >> matches port 5060 >> May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: >> [socket_info.c:501]: grep_sock_info - checking if host==us: >> 13==15 && ?[205.164.40.74] == [108.166.195.189] >> May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: >> [socket_info.c:504]: grep_sock_info - checking if port 5060 >> matches port 5060 >> May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: >> [forward.c:448]: check_self: host != me >> May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: >> [socket_info.c:501]: grep_sock_info - checking if host==us: >> 15==15 && ?[108.166.195.189] == [108.166.195.189] >> May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: >> [socket_info.c:504]: grep_sock_info - checking if port 7890 >> matches port 7890 >> May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: rr >> [loose.c:792]: Topmost route URI: >> 'sip:108.166.195.189:7890;lr=on;nat=yes' is me >> May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: >> [parser/msg_parser.c:103]: found end of header >> May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: rr >> [loose.c:257]: No next Route HF found >> May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: rr >> [loose.c:811]: No next URI found >> May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: rr >> [loose.c:983]: params are <;lr=on;nat=yes> >> May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: >> siputils [checks.c:76]: totag found >> May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: tm >> [t_lookup.c:1379]: DEBUG: t_newtran: msg id=2501 , global msg id=2500 >> , T on entrance=0xffffffff >> May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: tm >> [t_lookup.c:528]: t_lookup_request: start searching: hash=29177, >> isACK=0 >> May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: tm >> [t_lookup.c:564]: DEBUG: proceeding to pre-RFC3261 transaction >> matching >> May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: tm >> [t_lookup.c:711]: DEBUG: t_lookup_request: no transaction found >> May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: tm >> [t_hooks.c:374]: DBG: trans=0xb61626a4, callback type 1, id 0 entered >> May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: >> [msg_translator.c:204]: check_via_address(205.164.40.74, >> 205.164.40.74, 0) >> May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: tm >> [t_funcs.c:388]: SER: new transaction fwd'ed >> May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: >> [usr_avp.c:646]: DEBUG:destroy_avp_list: destroying list (nil) >> May 20 02:26:04 VOS20-108 last message repeated 5 times >> May 20 02:26:04 VOS20-108 /usr/local/sbin/kamailio[16443]: DEBUG: >> [receive.c:289]: receive_msg: cleaning up >> May 20 02:26:22 VOS20-108 /usr/local/sbin/kamailio[16445]: DEBUG: >> [udp_server.c:486]: udp_rcv_loop: probing packet received from >> 180.234.62.230 38722 >> May 20 02:26:34 VOS20-108 /usr/local/sbin/kamailio[16450]: DEBUG: tm >> [t_reply.c:1134]: ->>>>>>>>> T_code=0, new_code=408 >> May 20 02:26:34 VOS20-108 /usr/local/sbin/kamailio[16450]: DEBUG: tm >> [t_reply.c:1636]: DEBUG: relay_reply: branch=0, save=0, relay=0 >> May 20 02:26:34 VOS20-108 /usr/local/sbin/kamailio[16450]: DEBUG: >> [msg_translator.c:204]: check_via_address(205.164.40.74, >> 205.164.40.74, 0) >> May 20 02:26:34 VOS20-108 /usr/local/sbin/kamailio[16450]: DEBUG: >> [mem/shm_mem.c:105]: WARNING:vqm_resize: resize(0) called >> May 20 02:26:34 VOS20-108 /usr/local/sbin/kamailio[16450]: DEBUG: tm >> [t_hooks.c:288]: DBG: trans=0xb61626a4, callback type 128, id 0 >> entered >> May 20 02:26:54 VOS20-108 /usr/local/sbin/kamailio[16451]: WARNING: >> [timer.c:450]: WARNING: our timer runs faster then real-time >> (300000 ms / 4800 ticks our time .-> 299923 ms / 4798 ticks real time) >> >> OT: How do you guys maintain 80 column mails? i do not use a mail >> client, use gmail mailbox. >> -- >> -aft >> >> >> -- >> -aft >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- -aft From msc at freeswitch.org Mon May 21 23:35:45 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 21 May 2012 12:35:45 -0700 Subject: [Freeswitch-users] BYE message is not relayed to the UAC In-Reply-To: References: Message-ID: On Mon, May 21, 2012 at 12:33 PM, Arif Hossain wrote: > On Mon, May 21, 2012 at 10:22 PM, Michael Collins > wrote: > > Your best bet is to use a pastebin for this sort of thing. The FreeSWITCH > > project maintains their own at pastebin.freeswitch.org. There are > numerous > > others (pastie, pastebin.com, etc.) After you put all the information > in a > > pastebin you'll get a handy URL that you can include in the email thread. > > One other advantage to pastebins is that you can refer to a specific line > > number in paste. ("Look at the BYE message at line xxx - it looks > > unusual...") > > > > -MC > > > > freeswitch's pastebin asks for authentication. > > Yes, it does. ;) -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120521/04b3dad5/attachment.html From krice at freeswitch.org Mon May 21 23:38:57 2012 From: krice at freeswitch.org (Ken Rice) Date: Mon, 21 May 2012 14:38:57 -0500 Subject: [Freeswitch-users] BYE message is not relayed to the UAC In-Reply-To: Message-ID: Someone else failed the IQ test it seems... I mean really people how hard is it to READ that little popup box that tells you want to put in there K On 5/21/12 2:35 PM, "Michael Collins" wrote: > > > On Mon, May 21, 2012 at 12:33 PM, Arif Hossain wrote: >> On Mon, May 21, 2012 at 10:22 PM, Michael Collins wrote: >>> > Your best bet is to use a pastebin for this sort of thing. The FreeSWITCH >>> > project maintains their own at pastebin.freeswitch.org >>> . There are numerous >>> > others (pastie, pastebin.com , etc.) After you put >>> all the information in a >>> > pastebin you'll get a handy URL that you can include in the email thread. >>> > One other advantage to pastebins is that you can refer to a specific line >>> > number in paste. ("Look at the BYE message at line xxx - it looks >>> > unusual...") >>> > >>> > -MC >>> > >> >> freeswitch's pastebin asks for authentication. >> > > Yes, it does. ;) > > -MC > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120521/69b155c3/attachment.html From aftnix at gmail.com Mon May 21 23:45:44 2012 From: aftnix at gmail.com (Arif Hossain) Date: Tue, 22 May 2012 01:45:44 +0600 Subject: [Freeswitch-users] BYE message is not relayed to the UAC In-Reply-To: References: Message-ID: On Tue, May 22, 2012 at 1:38 AM, Ken Rice wrote: > Someone else failed the IQ test it seems... > > I mean really people how hard is it to READ that little popup box that tells > you want to put in there > Hi Ken, I can hear laughter sounds from here, deep inside asia :) -- -aft From aftnix at gmail.com Mon May 21 23:51:31 2012 From: aftnix at gmail.com (Arif Hossain) Date: Tue, 22 May 2012 01:51:31 +0600 Subject: [Freeswitch-users] BYE message is not relayed to the UAC In-Reply-To: References: Message-ID: On Tue, May 22, 2012 at 1:35 AM, Michael Collins wrote: > Yes, it does. ;) > > -MC > Ok done http://pastebin.freeswitch.org/19136 now we can talk problems :) -- -aft From sos at sokhapkin.dyndns.org Tue May 22 00:00:11 2012 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Mon, 21 May 2012 16:00:11 -0400 Subject: [Freeswitch-users] BYE message is not relayed to the UAC In-Reply-To: References: Message-ID: <4746584.BNOiHJVmYq@sos> Could you post the complete SIP trace, starting from INVITE? Is 205.164.40.74 IP address of voipswitch? On Tuesday 22 May 2012 01:51:31 Arif Hossain wrote: > On Tue, May 22, 2012 at 1:35 AM, Michael Collins wrote: > > Yes, it does. ;) > > > > -MC > > Ok done > http://pastebin.freeswitch.org/19136 > > now we can talk problems :) From acrow at integrafin.co.uk Tue May 22 00:13:03 2012 From: acrow at integrafin.co.uk (Alex Crow) Date: Mon, 21 May 2012 21:13:03 +0100 Subject: [Freeswitch-users] Spandsp mulated modems inbound/no /dev/FS* device? In-Reply-To: References: Message-ID: <4FBAA1CF.5060907@integrafin.co.uk> On 21/05/12 20:28, Ken Rice wrote: > > > On 5/21/12 2:15 PM, "Alex Crow" wrote: >> Also I have the problem that unless FreeSWITCH is run as root, the >> device nodes are not created. >> >> Unless in the last few weeks some docs have been updated on this, I'm >> still stuck with T38modem. >> > If freeswitch running as root and not running as root doesn't work, this > should tell you that you have a permissions issue... The FreeSWITCH > developers fixing your systems perms issues is beyond the scope of the > FreeSWITCH project... > > Otherwise, the FreeSWITCH modems basically work just like t38modem... The > only setting is how many of them to create then the only other settings is > configuring your platform to properly allow freeswitch (or the user that > freeswitch is running as) create those devices then configure hylafax to use > them as normal. > > The steps to do this can vary from platform to platform (or even versions of > the platform. For example how you do this on centos5 is not how you do this > on centos6 due to changes in the platform.) > > K > > Ken, I respectfully disagree with the "like t38modem" bit of this. With t38modem, you create a set of t38modem devices which actually register against FreeSWITCH and therefore appear as registered endpoints, so it's easy to route calls to them (ie transfer to "t38modem0 default XML"). When I did run FS as root, despite the device nodes being created, I saw no endpoint registrations for the softmodems, nor did I find any documentation saying that these endpoints may be addressed as, for instance "spandsp/FSx". I am also used to writing udev rules, but I can't find anything that allows a non-root uid/gid user to create device nodes for a device provided by a userspace program. It'd be great to avoid T38modem, and I've asked these questions before, namely: 1) Does anyone have any examples on any platform of tweaking udev to allow FreeSWITCH to create device nodes at startup when running as, say, user freeswitch or user www-data (without making the primary group of said users "root" or setting setuid root on the executable). I am not prepared to take the risk of running as either effective uid or gid 0. Please share if so. I'm running on debian stable. 2) How does one direct an incoming call, say, from FreeTDM or a SIP/ISDN gateway to one of the softmodems. I cannot find this in the docs. If we can divorce these questions from any association with another user's questions on this list that's fine. I've tried to help that person as much as I can out of my own good nature. Cheers Alex From mario_fs at mgtech.com Tue May 22 00:40:02 2012 From: mario_fs at mgtech.com (Mario G) Date: Mon, 21 May 2012 13:40:02 -0700 Subject: [Freeswitch-users] FreeSWITCH, OSX, Libtool, Macports In-Reply-To: References: <523D8618-21D9-442E-9D23-4321EAFA6558@opencsta.org> <7FFFF20B-D26F-4EDC-A19F-6AE1C48D5048@mgtech.com> <761EBAA8-6CE4-4A9B-BD8F-4A5EB064EEFA@mgtech.com> Message-ID: <7AA59325-D16B-4C20-87AF-87B35B2C763C@mgtech.com> Wish I could help more... this is what I tried today and all worked fine, I am on Snow Leopard. You may want to try to redo the git, may have been an xfr issue? 1, git pull followed by make current - worked 2. clean src fs dir, new git reload, bootstrap.sh, configure with ssl, make install - worked fine On May 21, 2012, at 10:29 AM, Neil Patel wrote: > Hi Mario, > > Thanks for your valuable posts on getting FS up and running on OS X. > > I'm doing so myself and ran into a snag compiling latest Git: > http://jira.freeswitch.org/browse/FS-4230 > > Given your expertise on FS and OSX I thought you may be able to help me get through this issue. Any suggestions? > > Thanks, > Neil > > On Fri, May 11, 2012 at 9:41 AM, Mario G wrote: > Some answers for you: > > On May 10, 2012, at 7:23 PM, Chris Mylonas wrote: > >> That's great Mario. I will update my blog accordingly where I posted some stuff (http://mrvoip.com.au/blog/install-freeswitch-osx-mysql-must-remove-macports) >> >> I just winged it because it used to work, so most of the prerequisites were installed. >> I did not put the --with-openssl flag when doing configure. What functionality do I lose? > > From what I remember, it complained about gnutis or something like that, you have to check the messages but it is really important, used to not be required. It took several days to figure this one out. I think the message was: > configure: error: --with-ssl was given, but GNUTLS is not available. > >> >> >> pkgconfig was already on my system from earlier efforts - so I guess I got lucky with that one. >> To be honest, once I resolved my macports issues (by removing it) it was just like doing it in linux. > > The funny thing is I still see a few pkgconfig missing messages during the make but FS works fine for me, I think it's for things I am not using. I will be looking into this more before I update the wiki. > >> >> I'd like to give it another shot with a fresh OSX install - but I really don't want to go about setting up my machine again! I have an old imac that is not doing much, but I don't think it will run 10.6 :( I will try and give it a shot this afternoon seeing as my girlfriend is out taking photos :) > > FYI, when I have not updated for a long time or make major changes like updating xCode: I backup the Mac Mini running FS as a bootable partition (I use SuperDuper), then I connect it to an iMac and boot using firewire target mode to boot from the backup. Now I can play round all I want. If all goes well, I just restore to the Mini, if not I can start over. For frequent updates I just backup the FS bin and source in the Mini with and do a git pull, etc.... > Mario G >> >> Kind Regards, >> Chris >> >> >> >> >> On 11/05/2012, at 11:48 AM, Mario G wrote: >> >>> Chris, I have been running FS on 10.6,x since 2010 fine and have been updating a lot recently to help testing, I have had no problems. I don't use macport stuff. I wrote the original FS osX wiki but there are some differences since 2010 (I am working on fixing the wiki in the next few weeks). The wiki install instructions work if you add the following. Hope this helps: >>> >>> 1. Add this info: >>> FreeSWITCH? has many functions that invoke PKG-CONFIG, so it must be downloaded and installed separately. >>> # Got to [http://pkgconfig.freedesktop.org/releases/ here] and download the latest pkg-config, by default it is placed into your Downloads folder. MUST USE .25 for OSXdue to glib2 dependencies! >>> # Open Downloads and click the file to uncompress it. >>> # Launch the Terminal application if not already running and issue these commands to move the source to the src directory, build and install: >>> cd ~/Downloads >>> mv pkg-config-0.25 /usr/local/src >>> cd /usr/local/src/pkg-config-0.25 >>> ./configure >>> make >>> sudo make install >> >>> >>> 2. You MUST do this: >>> ./bootstrap.sh >>> ./configure --with-openssl >>> >>> 3. The FLITE fix is no longer needed. >>> >>> On May 10, 2012, at 5:01 PM, Chris Mylonas wrote: >>> >>>> Hi FS Users, >>>> >>>> tl;dr; - removed macports, removed tree, pulled fresh tree = installed FS on OSX. >>>> >>>> >>>> Here is the longer version that had a bit of a pre-emptive whinge about libtool version mismatch. But we never got there - it worked! >>>> >>>> I'm in the process of re-installing FS on OSX (10.6.8). I have removed Macports to try and get this up and running. >>>> I re-bootstrapped and got this error: >>>> >>>> quiet_libtool: Version mismatch error. This is libtool 2.4, but the >>>> quiet_libtool: definition of this LT_INIT comes from libtool 2.2.4. >>>> quiet_libtool: You should recreate aclocal.m4 with macros from libtool 2.4 >>>> quiet_libtool: and run autoconf again. >>>> make[1]: *** [libfreeswitch_la-switch_apr.lo] Error 63 >>>> make: *** [all] Error 2 >>>> >>>> Removed the whole git tree just in case there was some left over junk. Pulled a fresh tree and noticed this remark during a fresh bootstrap >>>> >>>> arakis:freeswitch-git chrismylonas$ rm -Rf freeswitch/ >>>> arakis:freeswitch-git chrismylonas$ git clone git://git.freeswitch.org/freeswitch.git >>>> Cloning into freeswitch... >>>> remote: Counting objects: 185609, done. >>>> remote: Compressing objects: 100% (39208/39208), done. >>>> remote: Total 185609 (delta 143241), reused 181443 (delta 140035) >>>> Receiving objects: 100% (185609/185609), 77.85 MiB | 267 KiB/s, done. >>>> Resolving deltas: 100% (143241/143241), done. >>>> arakis:freeswitch-git chrismylonas$ cd freeswitch/ >>>> arakis:freeswitch chrismylonas$ ./bootstrap.sh >>>> bootstrap: checking installation... >>>> bootstrap: autoconf version 2.61 (ok) >>>> bootstrap: automake version 1.10 (ok) >>>> bootstrap: aclocal version 1.10 (ok) >>>> bootstrap: libtool version 2.2.4 (ok) >>>> Bootstrapping using: >>>> autoconf : /usr/bin/autoconf >>>> automake : /usr/bin/automake >>>> aclocal : /usr/bin/aclocal >>>> libtool : /usr/bin/glibtool (2.2.4.) >>>> libtoolize: /usr/bin/glibtoolize >>>> make : /usr/bin/make (GNU Make 3.81) >>>> awk : () >>>> >>>> It is still reporting a 2.2.4 version of libtool. >>>> ... >>>> ... >>>> ... >>>> In the end, it has been compiled and installed though >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at 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 at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120521/989d702e/attachment-0001.html From anthony.minessale at gmail.com Tue May 22 01:23:59 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 21 May 2012 16:23:59 -0500 Subject: [Freeswitch-users] Spandsp mulated modems inbound/no /dev/FS* device? In-Reply-To: <4FBAA1CF.5060907@integrafin.co.uk> References: <4FBAA1CF.5060907@integrafin.co.uk> Message-ID: IIRC its modem/1 or 2 3 4 etc. Probably a for auto like tdm. Check the code to be sure.... On May 21, 2012 3:13 PM, "Alex Crow" wrote: > On 21/05/12 20:28, Ken Rice wrote: > > > > > > On 5/21/12 2:15 PM, "Alex Crow" wrote: > >> Also I have the problem that unless FreeSWITCH is run as root, the > >> device nodes are not created. > >> > >> Unless in the last few weeks some docs have been updated on this, I'm > >> still stuck with T38modem. > >> > > If freeswitch running as root and not running as root doesn't work, this > > should tell you that you have a permissions issue... The FreeSWITCH > > developers fixing your systems perms issues is beyond the scope of the > > FreeSWITCH project... > > > > Otherwise, the FreeSWITCH modems basically work just like t38modem... The > > only setting is how many of them to create then the only other settings > is > > configuring your platform to properly allow freeswitch (or the user that > > freeswitch is running as) create those devices then configure hylafax to > use > > them as normal. > > > > The steps to do this can vary from platform to platform (or even > versions of > > the platform. For example how you do this on centos5 is not how you do > this > > on centos6 due to changes in the platform.) > > > > K > > > > > > Ken, > > I respectfully disagree with the "like t38modem" bit of this. With > t38modem, you create a set of t38modem devices which actually register > against FreeSWITCH and therefore appear as registered endpoints, so it's > easy to route calls to them (ie transfer to "t38modem0 default XML"). > When I did run FS as root, despite the device nodes being created, I saw > no endpoint registrations for the softmodems, nor did I find any > documentation saying that these endpoints may be addressed as, for > instance "spandsp/FSx". > > I am also used to writing udev rules, but I can't find anything that > allows a non-root uid/gid user to create device nodes for a device > provided by a userspace program. > > It'd be great to avoid T38modem, and I've asked these questions before, > namely: > > 1) Does anyone have any examples on any platform of tweaking udev to > allow FreeSWITCH to create device nodes at startup when running as, say, > user freeswitch or user www-data (without making the primary group of > said users "root" or setting setuid root on the executable). I am not > prepared to take the risk of running as either effective uid or gid 0. > Please share if so. I'm running on debian stable. > > 2) How does one direct an incoming call, say, from FreeTDM or a SIP/ISDN > gateway to one of the softmodems. I cannot find this in the docs. > > If we can divorce these questions from any association with another > user's questions on this list that's fine. I've tried to help that > person as much as I can out of my own good nature. > > Cheers > > Alex > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120521/97066dfe/attachment.html From msc at freeswitch.org Tue May 22 03:28:20 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 21 May 2012 16:28:20 -0700 Subject: [Freeswitch-users] Spandsp mulated modems inbound/no /dev/FS* device? In-Reply-To: References: <4FBAA1CF.5060907@integrafin.co.uk> Message-ID: I looked at the source, just to confirm and I believe Anthony is correct. Line 871 of mod_spandsp_modem.c: switch_snprintf(name, sizeof(name), "modem/%d/%s", modem->slot, number); That would suggest a syntax just like FreeTDM, where the dialstring is "modem/x/y" where x is the "slot" and y is the dialed number. Also, I suspect this block starting at line 856 means you can use a literal "a" for the slot and it will hunt for the next available modem: if (!strcasecmp(modem_id_string, "a")) { modem_id = -1; } else { modem_id = atoi(modem_id_string); } Please try this out and confirm. If possible, add the information to the wiki. If you can't update the wiki then please report back and let us know whether the dialstring format worked or not. Thanks, MC On Mon, May 21, 2012 at 2:23 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > IIRC its modem/1 or 2 3 4 etc. Probably a for auto like tdm. Check the > code to be sure.... > On May 21, 2012 3:13 PM, "Alex Crow" wrote: > >> On 21/05/12 20:28, Ken Rice wrote: >> > >> > >> > On 5/21/12 2:15 PM, "Alex Crow" wrote: >> >> Also I have the problem that unless FreeSWITCH is run as root, the >> >> device nodes are not created. >> >> >> >> Unless in the last few weeks some docs have been updated on this, I'm >> >> still stuck with T38modem. >> >> >> > If freeswitch running as root and not running as root doesn't work, this >> > should tell you that you have a permissions issue... The FreeSWITCH >> > developers fixing your systems perms issues is beyond the scope of the >> > FreeSWITCH project... >> > >> > Otherwise, the FreeSWITCH modems basically work just like t38modem... >> The >> > only setting is how many of them to create then the only other settings >> is >> > configuring your platform to properly allow freeswitch (or the user that >> > freeswitch is running as) create those devices then configure hylafax >> to use >> > them as normal. >> > >> > The steps to do this can vary from platform to platform (or even >> versions of >> > the platform. For example how you do this on centos5 is not how you do >> this >> > on centos6 due to changes in the platform.) >> > >> > K >> > >> > >> >> Ken, >> >> I respectfully disagree with the "like t38modem" bit of this. With >> t38modem, you create a set of t38modem devices which actually register >> against FreeSWITCH and therefore appear as registered endpoints, so it's >> easy to route calls to them (ie transfer to "t38modem0 default XML"). >> When I did run FS as root, despite the device nodes being created, I saw >> no endpoint registrations for the softmodems, nor did I find any >> documentation saying that these endpoints may be addressed as, for >> instance "spandsp/FSx". >> >> I am also used to writing udev rules, but I can't find anything that >> allows a non-root uid/gid user to create device nodes for a device >> provided by a userspace program. >> >> It'd be great to avoid T38modem, and I've asked these questions before, >> namely: >> >> 1) Does anyone have any examples on any platform of tweaking udev to >> allow FreeSWITCH to create device nodes at startup when running as, say, >> user freeswitch or user www-data (without making the primary group of >> said users "root" or setting setuid root on the executable). I am not >> prepared to take the risk of running as either effective uid or gid 0. >> Please share if so. I'm running on debian stable. >> >> 2) How does one direct an incoming call, say, from FreeTDM or a SIP/ISDN >> gateway to one of the softmodems. I cannot find this in the docs. >> >> If we can divorce these questions from any association with another >> user's questions on this list that's fine. I've tried to help that >> person as much as I can out of my own good nature. >> >> Cheers >> >> Alex >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120521/bde56090/attachment.html From neilp at cs.stanford.edu Tue May 22 03:38:46 2012 From: neilp at cs.stanford.edu (Neil Patel) Date: Mon, 21 May 2012 16:38:46 -0700 Subject: [Freeswitch-users] FreeSWITCH, OSX, Libtool, Macports In-Reply-To: <7AA59325-D16B-4C20-87AF-87B35B2C763C@mgtech.com> References: <523D8618-21D9-442E-9D23-4321EAFA6558@opencsta.org> <7FFFF20B-D26F-4EDC-A19F-6AE1C48D5048@mgtech.com> <761EBAA8-6CE4-4A9B-BD8F-4A5EB064EEFA@mgtech.com> <7AA59325-D16B-4C20-87AF-87B35B2C763C@mgtech.com> Message-ID: Thanks Mario, There was an old Jira that reported the same problem as mine, compiling on OSX: http://jira.freeswitch.org/browse/FS-1949 Looks like the fix had to do with setting some config parameters to compile FS in 64-bit mode (I'm on 64-bit macbook pro). Is this still necessary? I remember reading on another thread that nowadays you should not mess with the compile flags when running ./configure since the process auto-detects what you need. But still, is there any reason to force compile FS in 64 bit or 32 bit mode for OSX? Also note that I'm on OSX Lion, though I hope that isn't the cause. Thanks, Neil On Mon, May 21, 2012 at 1:40 PM, Mario G wrote: > Wish I could help more... this is what I tried today and all worked fine, > I am on Snow Leopard. You may want to try to redo the git, may have been an > xfr issue? > 1, git pull followed by make current - worked > 2. clean src fs dir, new git reload, bootstrap.sh, configure with ssl, > make install - worked fine > > > On May 21, 2012, at 10:29 AM, Neil Patel wrote: > > Hi Mario, > > Thanks for your valuable posts on getting FS up and running on OS X. > > I'm doing so myself and ran into a snag compiling latest Git: > http://jira.freeswitch.org/browse/FS-4230 > > Given your expertise on FS and OSX I thought you may be able to help me > get through this issue. Any suggestions? > > Thanks, > Neil > > On Fri, May 11, 2012 at 9:41 AM, Mario G wrote: > >> Some answers for you: >> >> On May 10, 2012, at 7:23 PM, Chris Mylonas wrote: >> >> That's great Mario. I will update my blog accordingly where I posted >> some stuff ( >> http://mrvoip.com.au/blog/install-freeswitch-osx-mysql-must-remove-macports >> ) >> >> I just winged it because it used to work, so most of the prerequisites >> were installed. >> I did not put the --with-openssl flag when doing configure. What >> functionality do I lose? >> >> >> From what I remember, it complained about gnutis or something like that, >> you have to check the messages but it is really important, used to not be >> required. It took several days to figure this one out. I think the message >> was: >> configure: error: --with-ssl was given, but GNUTLS is not available. >> >> >> >> pkgconfig was already on my system from earlier efforts - so I guess I >> got lucky with that one. >> To be honest, once I resolved my macports issues (by removing it) it was >> just like doing it in linux. >> >> >> The funny thing is I still see a few pkgconfig missing messages during >> the make but FS works fine for me, I think it's for things I am not using. >> I will be looking into this more before I update the wiki. >> >> >> I'd like to give it another shot with a fresh OSX install - but I really >> don't want to go about setting up my machine again! I have an old imac >> that is not doing much, but I don't think it will run 10.6 :( I will try >> and give it a shot this afternoon seeing as my girlfriend is out taking >> photos :) >> >> >> FYI, when I have not updated for a long time or make major changes like >> updating xCode: I backup the Mac Mini running FS as a bootable partition (I >> use SuperDuper), then I connect it to an iMac and boot using firewire >> target mode to boot from the backup. Now I can play round all I want. If >> all goes well, I just restore to the Mini, if not I can start over. For >> frequent updates I just backup the FS bin and source in the Mini with and >> do a git pull, etc.... >> Mario G >> >> >> Kind Regards, >> Chris >> >> >> >> >> On 11/05/2012, at 11:48 AM, Mario G wrote: >> >> Chris, I have been running FS on 10.6,x since 2010 fine and have been >> updating a lot recently to help testing, I have had no problems. I don't >> use macport stuff. I wrote the original FS osX wiki but there are some >> differences since 2010 (I am working on fixing the wiki in the next few >> weeks). The wiki install instructions work if you add the following. Hope >> this helps: >> >> 1. Add this info: >> FreeSWITCH? has many functions that invoke PKG-CONFIG, so it must be >> downloaded and installed separately. >> # Got to [http://pkgconfig.freedesktop.org/releases/ here] and download >> the latest pkg-config, by default it is placed into your Downloads folder. >> MUST USE .25 for OSXdue to glib2 dependencies! >> # Open Downloads and click the file to uncompress it. >> # Launch the Terminal application if not already running and issue these >> commands to move the source to the src directory, build and install: >> cd ~/Downloads >> mv pkg-config-0.25 /usr/local/src >> cd /usr/local/src/pkg-config-0.25 >> ./configure >> make >> sudo make install > into /usr/bin >> >> >> 2. You MUST do this: >> ./bootstrap.sh >> ./configure --with-openssl >> >> 3. The FLITE fix is no longer needed. >> >> On May 10, 2012, at 5:01 PM, Chris Mylonas wrote: >> >> Hi FS Users, >> >> tl;dr; - removed macports, removed tree, pulled fresh tree = installed >> FS on OSX. >> >> >> Here is the longer version that had a bit of a pre-emptive whinge about >> libtool version mismatch. But we never got there - it worked! >> >> I'm in the process of re-installing FS on OSX (10.6.8). I have removed >> Macports to try and get this up and running. >> I re-bootstrapped and got this error: >> >> quiet_libtool: Version mismatch error. This is libtool 2.4, but the >> quiet_libtool: definition of this LT_INIT comes from libtool 2.2.4. >> quiet_libtool: You should recreate aclocal.m4 with macros from libtool 2.4 >> quiet_libtool: and run autoconf again. >> make[1]: *** [libfreeswitch_la-switch_apr.lo] Error 63 >> make: *** [all] Error 2 >> >> Removed the whole git tree just in case there was some left over junk. >> Pulled a fresh tree and noticed this remark during a fresh bootstrap >> >> arakis:freeswitch-git chrismylonas$ rm -Rf freeswitch/ >> arakis:freeswitch-git chrismylonas$ git clone >> git://git.freeswitch.org/freeswitch.git >> Cloning into freeswitch... >> remote: Counting objects: 185609, done. >> remote: Compressing objects: 100% (39208/39208), done. >> remote: Total 185609 (delta 143241), reused 181443 (delta 140035) >> Receiving objects: 100% (185609/185609), 77.85 MiB | 267 KiB/s, done. >> Resolving deltas: 100% (143241/143241), done. >> arakis:freeswitch-git chrismylonas$ cd freeswitch/ >> arakis:freeswitch chrismylonas$ ./bootstrap.sh >> bootstrap: checking installation... >> bootstrap: autoconf version 2.61 (ok) >> bootstrap: automake version 1.10 (ok) >> bootstrap: aclocal version 1.10 (ok) >> bootstrap: libtool version 2.2.4 (ok) >> Bootstrapping using: >> autoconf : /usr/bin/autoconf >> automake : /usr/bin/automake >> aclocal : /usr/bin/aclocal >> libtool : /usr/bin/glibtool (2.2.4.) >> libtoolize: /usr/bin/glibtoolize >> make : /usr/bin/make (GNU Make 3.81) >> awk : () >> >> It is still reporting a 2.2.4 version of libtool. >> ... >> ... >> ... >> In the end, it has been compiled and installed though >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120521/09961903/attachment-0001.html From mel0torme at gmail.com Tue May 22 04:53:41 2012 From: mel0torme at gmail.com (Tom C) Date: Mon, 21 May 2012 17:53:41 -0700 Subject: [Freeswitch-users] Freeswitch stable on embedded devices? In-Reply-To: <1337560295076-7568516.post@n2.nabble.com> References: <1337546724347-7568340.post@n2.nabble.com> <1337560295076-7568516.post@n2.nabble.com> Message-ID: Do you use uncompressed files for MoH or MP3, which could explain the > chopiness? > > Nope, just the standard WAV files that come with the git. And I'm using the correct bitrate, so no trans-coding issues. Do you recompile Freeswitch yourself for the Dockstar? If yes, is that hard > to do? I need to make sure I'm able to update the devices in case serious > bugs or useful features come up once they're out in the field. > > Yes, compiling on the dockstar is just as easy as on any other platform, once you have all the requisite packages installed. The smaller Linux distributions made for these devices don't contain every package that the major (desktop) distributions have. A quick web search will turn up a list for you. Everything builds fine on the dockstar with only 128megs of RAM. Except, coincidentally enough, FLITE, which needs about 450megs of addressable memory to build successfully. So to compile that module, you need to create a swap file of at least 350 megs. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120521/7836d5e9/attachment.html From steveu at coppice.org Tue May 22 05:11:10 2012 From: steveu at coppice.org (Steve Underwood) Date: Tue, 22 May 2012 09:11:10 +0800 Subject: [Freeswitch-users] Spandsp mulated modems inbound/no /dev/FS* device? In-Reply-To: References: <4FBAA1CF.5060907@integrafin.co.uk> Message-ID: <4FBAE7AE.1000606@coppice.org> What all this should tell us is a wiki page is badly needed on this topic. Specifically, people need clear information about: - How the modems appear to the user inside FS - How the modems appear to the user in /dev - How the permissions of the modems in /dev can be controlled, so non-root users are OK I still haven't seen anyone give information on the last point. Its easy to set up udev rules to control the generation of the pts devices in /dev, but I am not clear how to control the permissions during the creation of the links in /dev which give these devices meaningful names. Steve On 05/22/2012 07:28 AM, Michael Collins wrote: > I looked at the source, just to confirm and I believe Anthony is correct. > > Line 871 of mod_spandsp_modem.c: > switch_snprintf(name, sizeof(name), "modem/%d/%s", modem->slot, number); > > That would suggest a syntax just like FreeTDM, where the dialstring is > "modem/x/y" where x is the "slot" and y is the dialed number. Also, I > suspect this block starting at line 856 means you can use a literal > "a" for the slot and it will hunt for the next available modem: > > if (!strcasecmp(modem_id_string, "a")) { > modem_id = -1; > } else { > modem_id = atoi(modem_id_string); > } > > Please try this out and confirm. If possible, add the information to > the wiki. If you can't update the wiki then please report back and let > us know whether the dialstring format worked or not. > > Thanks, > MC > > On Mon, May 21, 2012 at 2:23 PM, Anthony Minessale > > wrote: > > IIRC its modem/1 or 2 3 4 etc. Probably a for auto like tdm. > Check the code to be sure.... > > On May 21, 2012 3:13 PM, "Alex Crow" > wrote: > > On 21/05/12 20:28, Ken Rice wrote: > > > > > > On 5/21/12 2:15 PM, "Alex Crow" > wrote: > >> Also I have the problem that unless FreeSWITCH is run as > root, the > >> device nodes are not created. > >> > >> Unless in the last few weeks some docs have been updated on > this, I'm > >> still stuck with T38modem. > >> > > If freeswitch running as root and not running as root > doesn't work, this > > should tell you that you have a permissions issue... The > FreeSWITCH > > developers fixing your systems perms issues is beyond the > scope of the > > FreeSWITCH project... > > > > Otherwise, the FreeSWITCH modems basically work just like > t38modem... The > > only setting is how many of them to create then the only > other settings is > > configuring your platform to properly allow freeswitch (or > the user that > > freeswitch is running as) create those devices then > configure hylafax to use > > them as normal. > > > > The steps to do this can vary from platform to platform (or > even versions of > > the platform. For example how you do this on centos5 is not > how you do this > > on centos6 due to changes in the platform.) > > > > K > > > > > > Ken, > > I respectfully disagree with the "like t38modem" bit of this. With > t38modem, you create a set of t38modem devices which actually > register > against FreeSWITCH and therefore appear as registered > endpoints, so it's > easy to route calls to them (ie transfer to "t38modem0 default > XML"). > When I did run FS as root, despite the device nodes being > created, I saw > no endpoint registrations for the softmodems, nor did I find any > documentation saying that these endpoints may be addressed as, for > instance "spandsp/FSx". > > I am also used to writing udev rules, but I can't find > anything that > allows a non-root uid/gid user to create device nodes for a device > provided by a userspace program. > > It'd be great to avoid T38modem, and I've asked these > questions before, > namely: > > 1) Does anyone have any examples on any platform of tweaking > udev to > allow FreeSWITCH to create device nodes at startup when > running as, say, > user freeswitch or user www-data (without making the primary > group of > said users "root" or setting setuid root on the executable). I > am not > prepared to take the risk of running as either effective uid > or gid 0. > Please share if so. I'm running on debian stable. > > 2) How does one direct an incoming call, say, from FreeTDM or > a SIP/ISDN > gateway to one of the softmodems. I cannot find this in the docs. > > If we can divorce these questions from any association with > another > user's questions on this list that's fine. I've tried to help that > person as much as I can out of my own good nature. > > Cheers > > Alex > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From blackc2004 at gmail.com Tue May 22 06:04:02 2012 From: blackc2004 at gmail.com (Cj B) Date: Mon, 21 May 2012 19:04:02 -0700 Subject: [Freeswitch-users] Ring group auto answers and then diconnects In-Reply-To: References: <2D163CA6-ABBF-46BA-9C02-26B9358490AC@gmail.com> Message-ID: <82CAD599-05A4-48DF-910C-4ACF146DDE4C@gmail.com> Sorry, I forgot that I had removed all the other exts to test it out. So here's a little more information now that I have played with it more. - If I call from EXT to EXT, the phones ring each other, but they never pick up (and there's no ringing sound played to the caller). If I answer the call it just always says "Answering" and "Connecting", but never actually connects. http://pastebin.com/gupzThup - When i dial the ring group it says "connected" right away and then after 30 seconds goes to the goodbye voicemail message. http://pastebin.com/qE5nikri The phones ring, but again when i pick up the call it doesn't actually answer - after you dial the ring group and hang up, then try to use any ext and call any other ext the call never goes through and ends up being a "timeout". It's very bizarre, I was thinking maybe it might be a nat setting? Any more ideas as to what could be going on? Cj B On May 21, 2012, at 11:49 AM, Michael Collins wrote: > I see that it is only calling a single extension, in this case x9001. (See line #117 of your pastebin.) Can you call extension 9001 directly? If so, do you see anything different in the console log? > > -MC > > On Mon, May 21, 2012 at 11:09 AM, Cj B wrote: > Hi all, > > I've setup a pretty simple ring group using fusionpbx and multi-tenant. I have ext's 9001-9004 and a ring group of Ext 8003. All 4 exts are part of the ring group. > > When I dial the ring group ext it gets answered right away but none of the phones ever ring. Then after 30 seconds (the timeout) I get the greeting "Good bye" and then it disconnects. > > I've posted all the output that happens here: > http://pastebin.com/dZjwKVSA > > Can someone please help me out, this seems like it should be a pretty simple thing to do, so i'm sure it's just a setup issue that I've done wrong! > > Thanks. > Cj B > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120521/40aec91a/attachment.html From zhulizhong at live.com Tue May 22 06:29:23 2012 From: zhulizhong at live.com (James zhu) Date: Tue, 22 May 2012 02:29:23 +0000 Subject: [Freeswitch-users] FreeTDM with BRI Card In-Reply-To: References: Message-ID: I do not think the dahdi and libpri can work properly. i think nobody fully test with FW and dahdi,libpri.. Best regards, James.zhu Doing asterisk/PRI/ss7/dahdi, linux, asterisk/sangoma cards, recording device, VOIP gateway. website: www.hiastar.com From: gopalakrishnan.an at gmail.com Date: Mon, 21 May 2012 15:42:24 +0530 To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] FreeTDM with BRI Card Hi all, I am using Openvox BRI card with Freeswitch using FreeTDM. In NT mode I couldn't get dialtone. I got the error like " Unable to get channel 1: -1". Please help me to solve this issue. I have attached my configuration details. I have used FreeTDM module with LibPRI and Freeswitch Regards,Gopal. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120522/7d53e7c3/attachment-0001.html From bdfoster at endigotech.com Tue May 22 08:59:49 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Tue, 22 May 2012 00:59:49 -0400 Subject: [Freeswitch-users] Ring group auto answers and then diconnects In-Reply-To: <2D163CA6-ABBF-46BA-9C02-26B9358490AC@gmail.com> References: <2D163CA6-ABBF-46BA-9C02-26B9358490AC@gmail.com> Message-ID: Pastebin ringgroup.lua. you probably aren't telling it where to go if the bridge fails. -BDF On May 21, 2012 2:33 PM, "Cj B" wrote: > Hi all, > > I've setup a pretty simple ring group using fusionpbx and multi-tenant. I > have ext's 9001-9004 and a ring group of Ext 8003. All 4 exts are part of > the ring group. > > When I dial the ring group ext it gets answered right away but none of the > phones ever ring. Then after 30 seconds (the timeout) I get the greeting > "Good bye" and then it disconnects. > > I've posted all the output that happens here: > http://pastebin.com/dZjwKVSA > > Can someone please help me out, this seems like it should be a pretty > simple thing to do, so i'm sure it's just a setup issue that I've done > wrong! > > Thanks. > Cj B > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120522/8f959944/attachment.html From bdfoster at endigotech.com Tue May 22 09:33:58 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Tue, 22 May 2012 01:33:58 -0400 Subject: [Freeswitch-users] Freeswitch stable on embedded devices? In-Reply-To: References: <1337546724347-7568340.post@n2.nabble.com> <1337560295076-7568516.post@n2.nabble.com> Message-ID: You are still transcoding from .wav to ulaw alaw etc. You can work around that by not transcoding on the fly. Using media bugs is transcoding. I don't know if there's a good work around for that though. -BDF On May 21, 2012 8:55 PM, "Tom C" wrote: > > > Do you use uncompressed files for MoH or MP3, which could explain the >> chopiness? >> >> Nope, just the standard WAV files that come with the git. And I'm using > the correct bitrate, so no trans-coding issues. > > > > Do you recompile Freeswitch yourself for the Dockstar? If yes, is that hard >> to do? I need to make sure I'm able to update the devices in case serious >> bugs or useful features come up once they're out in the field. >> >> > Yes, compiling on the dockstar is just as easy as on any other platform, > once you have all the requisite packages installed. The smaller Linux > distributions made for these devices don't contain every package that the > major (desktop) distributions have. A quick web search will turn up a list > for you. > > Everything builds fine on the dockstar with only 128megs of RAM. Except, > coincidentally enough, FLITE, which needs about 450megs of addressable > memory to build successfully. So to compile that module, you need to > create a swap file of at least 350 megs. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120522/4270f1b5/attachment.html From miha at softnet.si Tue May 22 10:11:59 2012 From: miha at softnet.si (Miha) Date: Tue, 22 May 2012 08:11:59 +0200 Subject: [Freeswitch-users] Call_group intercept In-Reply-To: References: <4FB9E602.8070708@softnet.si> Message-ID: <4FBB2E2F.7070901@softnet.si> On 5/21/2012 6:14 PM, Michael Collins wrote: > I think that perhaps you are misunderstanding what the instructions on > that wiki page are doing. The var "callgroup" is set in the user's > directory entry. Did you do that for each user in your pickup group? > > -MC > > On Sun, May 20, 2012 at 11:51 PM, Miha > wrote: > > HI, > > I read wiki tutorial about call group intercept > (http://wiki.freeswitch.org/wiki/Callgroup_intercept) ans also > implemented it. (If I make call intercept which is not for > callgroup it works) > > I put this in public dialplan, so that every call UUID will be > inserted in mysql database. > > > > > > > When call hit public dialplan I can not access variable callgroup. > As callgroup variable is not set I can not make right mysql > insert. What is the best way to deal with this? > > THanks! > > Miha > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org HI @Michal, I have defined callgroup for users that I would like to have in the same callgroup. After call hit public dialplan I make but in log I can not see that any variable for that user is set. If I make call from default dialplan variables are set and I can use them. Would it be better to redirect call to default dialplan? Regards, Miha -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120522/abee8b62/attachment.html From daniel.sokolowski at danols.com Tue May 22 06:00:22 2012 From: daniel.sokolowski at danols.com (Daniel Sokolowski) Date: Mon, 21 May 2012 22:00:22 -0400 Subject: [Freeswitch-users] Registration Failed with status Service Unavailable [503] Message-ID: <4FBAF336.4060500@danols.com> Hi All, I'm running FreeSWITCH Version 1.2.0 (git-b3b2c37 2012-05-18 13-41-16 -0500) and have been running a FreeSWITCH for over a year now more or less with no problems. I believe though I either have found a bug or a problem in my Debian setup. If I do a system reboot FreeSWITCH fails to connect to my 3 gateways: 2012-05-21 21:15:31.115850 [ERR] sofia_reg.c:1986 freephoneline.ca Registration Failed with status Service Unavailable [503]. failure #14 2012-05-21 21:15:31.115850 [ERR] sofia_reg.c:1986 internetcalls.com Registration Failed with status Service Unavailable [503]. failure #14 2012-05-21 21:15:31.215876 [ERR] sofia_reg.c:1986 voip.ms Registration Failed with status Service Unavailable [503]. failure #14 The work around is to manually call the init.d script '/etc/init.d/freeswitch restart'. I have been able to duplicate this behaviou over a dozen times now. I believe during boot FreeSWITCH somehow detects wrong network settings and keeps using them on retries - only a restart clears this issue. My init.d script was taken from the FS wiki and can be found here: http://dpaste.com/750951/ --- I already modified it to require '$network' service with no effect on the issue I'm having. Does anyone have any idea what might be causing this? -- Daniel Sokolowski Web Engineer Danols Web Engineering http://webdesign.danols.com/ Office: 613-817-6833 Fax: 613-817-4553 Toll Free: 1-855-5DANOLS Kingston, ON K7L 1H3, Canada Notice of Confidentiality: The information transmitted is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Any review re-transmission dissemination or other use of or taking of any action in reliance upon this information by persons or entities other than the intended recipient is prohibited. If you received this in error please contact the sender immediately by return electronic transmission and then immediately delete this transmission including all attachments without copying distributing or disclosing same. From msc at freeswitch.org Tue May 22 11:07:09 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 22 May 2012 00:07:09 -0700 Subject: [Freeswitch-users] Call_group intercept In-Reply-To: <4FBB2E2F.7070901@softnet.si> References: <4FB9E602.8070708@softnet.si> <4FBB2E2F.7070901@softnet.si> Message-ID: HI @Michal, > > I have defined callgroup for users that I would like to have in the same > callgroup. > > After call hit public dialplan I make but in > log I can not see that any variable for that user is set. If I make call > from default dialplan variables are set and I can use them. > Would it be better to redirect call to default dialplan? > > Regards, > Miha > > If your calls are hitting the public context then that suggests to me that you are not authenticating the user when he/she calls. If you don't authenticate then you don't get the user's channel variables. This happens, e.g., when you let callers in via ACL with a rule in the "domains" ACL. If you know which user it is supposed to be then you can cheat and use the set_user app: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_set_user -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120522/2b6e981f/attachment.html From babak.freeswitch at gmail.com Tue May 22 16:55:40 2012 From: babak.freeswitch at gmail.com (babak yakhchali) Date: Tue, 22 May 2012 17:25:40 +0430 Subject: [Freeswitch-users] sofia current calls for a user In-Reply-To: References: Message-ID: For example I need to find out if user 1000 at default is talking to someone or not. Currently I use "show channels like 1000" and then parse it for channels like sofia/internal/1000 or something like this. but as I mentioned, looking at mod_sofia code I see there is a table named sip_dialogs which contains info that I need but there is no api (or I donno about it) to show sip dialogs. I need some way to get that info if it is possible thanx -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120522/80a59883/attachment.html From lists.jj at googlemail.com Tue May 22 12:24:42 2012 From: lists.jj at googlemail.com (John Jacob) Date: Tue, 22 May 2012 10:24:42 +0200 Subject: [Freeswitch-users] default dialplan: signal busy when busy and voicemail disabled Message-ID: Hi, I guess you are all aware of the default dialplan entry for local extensions: [...] So, if a caller calls a callee, which disabled his voicemail and is currently busy/unavailable FreeSWITCH plays a good bye message and hangs up. This is unexpected for most of our clients and they are interpreting it as a malfunctioning PBX. I'd like to change this default behavior and respect the extension's setting for "vm-enabled". If it's enabled, the default behavior is fine, if it's disabled, I'd like FreeSWITCH to signal *busy* to the caller. It looks like I can't test for user parameters with conditions, so I don't have a clue how to implement this? For now I just removed answer,sleep and bridge to voicemail and busy is signalled just fine, but I'm aware that I killed voicemail completely by doing this ;) Any hints or tipps? I can't implement dynamic dialplans with mod_curl though, because those don't fit into the current setup. Thanks, Johannes From peter at uringme.com Tue May 22 17:52:58 2012 From: peter at uringme.com (peter at uringme.com) Date: Tue, 22 May 2012 06:52:58 -0700 (PDT) Subject: [Freeswitch-users] SAVPF and a=crypto In-Reply-To: <1337626667.74763.YahooMailClassic@web2819.biz.mail.ne1.yahoo.com> Message-ID: <1337694778.2980.YahooMailClassic@web2801.biz.mail.ne1.yahoo.com> Is there anyone who knows anything about FreeSwitch and SAVPF? Should this be on the dev group instead? Thanks Peter --- On Mon, 5/21/12, peter at uringme.com wrote: > From: peter at uringme.com > Subject: SAVPF and a=crypto > To: freeswitch-users at lists.freeswitch.org > Date: Monday, May 21, 2012, 2:57 PM > I'm trying out sipML5 ( http://code.google.com/p/sipml5/ ) with FS and I had a > problem with their demo.? Their code uses a=crypto > lines on RTP/SAVPF and FS is bouncing it with: > > 2012-05-21 13:04:47.754843 [ERR] sofia_glue.c:4478 a=crypto > in RTP/AVP, refer to rfc3711 > > The crypto line is in RTP/SAVPF, though, not RTP/AVP.? > Is this line coming up because it's not RTP/SAVP?? Am I > missing a module for RTP/SAVPF, or is this even supported? > > Following is a trace -- I'm calling x7779 on my server which > just plays a voice prompt. > > > ???INVITE sip:7779 at x.x.x.x SIP/2.0 > ???Via: SIP/2.0/UDP > 87.106.69.240:4060;branch=z9hG4bK-524287-1---a56eca302b8d6412;rport > ???Via: SIP/2.0/TCP > y.y.y.y:62676;branch=z9hG4bK1P36uFnLdbzHiEzWdVAd0G6nk6kdOdEt;rport=62676;received=y.y.y.y > ???Max-Forwards: 69 > ???Contact: "Peter > Krause";+sip.ice > ???To: > ???From: > ;tag=gz6bNBeAkHM7eAR2eoJG > ???Call-ID: > 434d70e1-5cc1-d096-442c-499c8cd600ad > ???CSeq: 6437 INVITE > ???Content-Type: application/sdp > ???Organization: Doubango Telecom > ???User-Agent: IM-client/OMA1.0 > sipML5/v0.0.0000.0 > ???Content-Length: 3483 > > ???v=0 > ???o=- 275367574 1 IN IP4 127.0.0.1 > ???s=webrtc (chrome 20.0.1127.0) - Doubango > Telecom (sipML5 r000) > ???t=0 0 > ???m=audio 54450 RTP/SAVPF 103 104 0 8 106 > 105 13 126 > ???c=IN IP4 y.y.y.y > ???a=rtcp:54451 IN IP4 y.y.y.y > ???a=candidate:0 1 udp 2130706432 192.168.1.8 > 50256 typ host network_name > {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username > 2C/PMnZyAP2pLbP0 password 0xigYC8ZZibTPCejG5JKwjBN > generation 0 > ???a=candidate:0 2 udp 2130706432 192.168.1.8 > 50257 typ host network_name > {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username > 2BPqrxt6Vp+wJNL3 password UEWxx9yByE/JlsmFHa0zkbTs > generation 0 > ???a=candidate:0 1 udp 1912602624 y.y.y.y > 54450 typ srflx network_name > {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username > 2C/PMnZyAP2pLbP0 password 0xigYC8ZZibTPCejG5JKwjBN > generation 0 > ???a=candidate:0 2 udp 1912602624 y.y.y.y > 54451 typ srflx network_name > {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username > 2BPqrxt6Vp+wJNL3 password UEWxx9yByE/JlsmFHa0zkbTs > generation 0 > ???a=candidate:0 1 tcp 1694498816 192.168.1.8 > 62677 typ host network_name > {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username > 2C/PMnZyAP2pLbP0 password 0xigYC8ZZibTPCejG5JKwjBN > generation 0 > ???a=candidate:0 2 tcp 1694498816 192.168.1.8 > 62678 typ host network_name > {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username > 2BPqrxt6Vp+wJNL3 password UEWxx9yByE/JlsmFHa0zkbTs > generation 0 > ???a=mid:audio > ???a=rtcp-mux > ???a=crypto:0 AES_CM_128_HMAC_SHA1_32 > inline:33ayFPGAoi+nni8kbYZfBO7tH3qI1SaH7ilFU7GT > ???a=crypto:1 AES_CM_128_HMAC_SHA1_80 > inline:Hd6rAXtyms1rJgC4OTJk/K4AGpnwLUn9Aqdx1sbd > ???a=rtpmap:103 ISAC/16000 > ???a=rtpmap:104 ISAC/32000 > ???a=rtpmap:0 PCMU/8000 > ???a=rtpmap:8 PCMA/8000 > ???a=rtpmap:106 CN/32000 > ???a=rtpmap:105 CN/16000 > ???a=rtpmap:13 CN/8000 > ???a=rtpmap:126 telephone-event/8000 > ???a=ssrc:2075114348 cname:vcjaT0z7bAvF1TL3 > ???a=ssrc:2075114348 > mslabel:TOAV4oLn6QD7CfJieQ90FAEQB06orMEVm2na > ???a=ssrc:2075114348 > label:TOAV4oLn6QD7CfJieQ90FAEQB06orMEVm2na0 > ???a=sendrecv > ???m=video 54452 RTP/SAVPF 100 101 102 > ???c=IN IP4 y.y.y.y > ???a=rtcp:54453 IN IP4 y.y.y.y > ???a=candidate:0 1 udp 2130706432 192.168.1.8 > 50258 typ host network_name > {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username > PlVs0WI8o1Q05fcc password eakcTlAsBYwpmZvTc58E1thy > generation 0 > ???a=candidate:0 2 udp 2130706432 192.168.1.8 > 50259 typ host network_name > {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username > gUagDr7SY/5Jxned password 6J+08YaMdKaer/95B9lfxCEM > generation 0 > ???a=candidate:0 1 udp 1912602624 y.y.y.y > 54452 typ srflx network_name > {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username > PlVs0WI8o1Q05fcc password eakcTlAsBYwpmZvTc58E1thy > generation 0 > ???a=candidate:0 2 udp 1912602624 y.y.y.y > 54453 typ srflx network_name > {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username > gUagDr7SY/5Jxned password 6J+08YaMdKaer/95B9lfxCEM > generation 0 > ???a=candidate:0 1 tcp 1694498816 192.168.1.8 > 62679 typ host network_name > {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username > PlVs0WI8o1Q05fcc password eakcTlAsBYwpmZvTc58E1thy > generation 0 > ???a=candidate:0 2 tcp 1694498816 192.168.1.8 > 62680 typ host network_name > {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username > gUagDr7SY/5Jxned password 6J+08YaMdKaer/95B9lfxCEM > generation 0 > ???a=mid:video > ???a=rtcp-mux > ???a=crypto:0 AES_CM_128_HMAC_SHA1_80 > inline:RIxXHNE/5GtDfkXnAV4CLXBQGxPzlzBg3Xyg3Lfs > ???a=rtpmap:100 VP8/90000 > ???a=rtpmap:101 red/90000 > ???a=rtpmap:102 ulpfec/90000 > ???a=ssrc:2075114348 cname:vcjaT0z7bAvF1TL3 > ???a=ssrc:2075114348 > mslabel:TOAV4oLn6QD7CfJieQ90FAEQB06orMEVm2na > ???a=ssrc:2075114348 > label:TOAV4oLn6QD7CfJieQ90FAEQB06orMEVm2na0 > ???a=sendrecv > ???------------------------------------------------------------------------ > send 448 bytes to udp/[87.106.69.240]:4060 at > 17:04:47.774032: > ???------------------------------------------------------------------------ > ???SIP/2.0 100 Trying > ???Via: SIP/2.0/UDP > 87.106.69.240:4060;branch=z9hG4bK-524287-1---a56eca302b8d6412;rport=4060 > ???Via: SIP/2.0/TCP > y.y.y.y:62676;branch=z9hG4bK1P36uFnLdbzHiEzWdVAd0G6nk6kdOdEt;rport=62676;received=y.y.y.y > ???From: > ;tag=gz6bNBeAkHM7eAR2eoJG > ???To: > ???Call-ID: > 434d70e1-5cc1-d096-442c-499c8cd600ad > ???CSeq: 6437 INVITE > ???User-Agent: > FreeSWITCH-mod_sofia/1.0.head-git- > ???Content-Length: 0 > > ???------------------------------------------------------------------------ > 2012-05-21 13:04:47.754843 [NOTICE] switch_channel.c:816 New > Channel sofia/external/7777 at uringme > [120f19be-a367-11e1-b22e-57011ca30f08] > 2012-05-21 13:04:47.754843 [DEBUG] sofia.c:4770 Channel > sofia/external/7777 at uringme entering state [received][100] > 2012-05-21 13:04:47.754843 [DEBUG] sofia.c:4781 Remote SDP: > v=0 > o=- 275367574 1 IN IP4 127.0.0.1 > s=webrtc (chrome 20.0.1127.0) - Doubango Telecom (sipML5 > r000) > t=0 0 > m=audio 54450 RTP/SAVPF 103 104 0 8 106 105 13 126 > c=IN IP4 y.y.y.y > a=rtcp:54451 IN IP4 y.y.y.y > a=candidate:0 1 udp 2130706432 192.168.1.8 50256 typ host > network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username > 2C/PMnZyAP2pLbP0 password 0xigYC8ZZibTPCejG5JKwjBN > generation 0 > a=candidate:0 2 udp 2130706432 192.168.1.8 50257 typ host > network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username > 2BPqrxt6Vp+wJNL3 password UEWxx9yByE/JlsmFHa0zkbTs > generation 0 > a=candidate:0 1 udp 1912602624 y.y.y.y 54450 typ srflx > network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username > 2C/PMnZyAP2pLbP0 password 0xigYC8ZZibTPCejG5JKwjBN > generation 0 > a=candidate:0 2 udp 1912602624 y.y.y.y 54451 typ srflx > network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username > 2BPqrxt6Vp+wJNL3 password UEWxx9yByE/JlsmFHa0zkbTs > generation 0 > a=candidate:0 1 tcp 1694498816 192.168.1.8 62677 typ host > network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username > 2C/PMnZyAP2pLbP0 password 0xigYC8ZZibTPCejG5JKwjBN > generation 0 > a=candidate:0 2 tcp 1694498816 192.168.1.8 62678 typ host > network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username > 2BPqrxt6Vp+wJNL3 password UEWxx9yByE/JlsmFHa0zkbTs > generation 0 > a=mid:audio > a=rtcp-mux > a=crypto:0 AES_CM_128_HMAC_SHA1_32 > inline:33ayFPGAoi+nni8kbYZfBO7tH3qI1SaH7ilFU7GT > a=crypto:1 AES_CM_128_HMAC_SHA1_80 > inline:Hd6rAXtyms1rJgC4OTJk/K4AGpnwLUn9Aqdx1sbd > a=rtpmap:103 ISAC/16000 > a=rtpmap:104 ISAC/32000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:106 CN/32000 > a=rtpmap:105 CN/16000 > a=rtpmap:13 CN/8000 > a=rtpmap:126 telephone-event/8000 > a=ssrc:2075114348 cname:vcjaT0z7bAvF1TL3 > a=ssrc:2075114348 > mslabel:TOAV4oLn6QD7CfJieQ90FAEQB06orMEVm2na > a=ssrc:2075114348 > label:TOAV4oLn6QD7CfJieQ90FAEQB06orMEVm2na0 > m=video 54452 RTP/SAVPF 100 101 102 > c=IN IP4 y.y.y.y > a=rtcp:54453 IN IP4 y.y.y.y > a=candidate:0 1 udp 2130706432 192.168.1.8 50258 typ host > network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username > PlVs0WI8o1Q05fcc password eakcTlAsBYwpmZvTc58E1thy > generation 0 > a=candidate:0 2 udp 2130706432 192.168.1.8 50259 typ host > network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username > gUagDr7SY/5Jxned password 6J+08YaMdKaer/95B9lfxCEM > generation 0 > a=candidate:0 1 udp 1912602624 y.y.y.y 54452 typ srflx > network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username > PlVs0WI8o1Q05fcc password eakcTlAsBYwpmZvTc58E1thy > generation 0 > a=candidate:0 2 udp 1912602624 y.y.y.y 54453 typ srflx > network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username > gUagDr7SY/5Jxned password 6J+08YaMdKaer/95B9lfxCEM > generation 0 > a=candidate:0 1 tcp 1694498816 192.168.1.8 62679 typ host > network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username > PlVs0WI8o1Q05fcc password eakcTlAsBYwpmZvTc58E1thy > generation 0 > a=candidate:0 2 tcp 1694498816 192.168.1.8 62680 typ host > network_name {D1FD878A-47C6-4B01-B940-ECA38C8D0617} username > gUagDr7SY/5Jxned password 6J+08YaMdKaer/95B9lfxCEM > generation 0 > a=mid:video > a=rtcp-mux > a=crypto:0 AES_CM_128_HMAC_SHA1_80 > inline:RIxXHNE/5GtDfkXnAV4CLXBQGxPzlzBg3Xyg3Lfs > a=rtpmap:100 VP8/90000 > a=rtpmap:101 red/90000 > a=rtpmap:102 ulpfec/90000 > a=ssrc:2075114348 cname:vcjaT0z7bAvF1TL3 > a=ssrc:2075114348 > mslabel:TOAV4oLn6QD7CfJieQ90FAEQB06orMEVm2na > a=ssrc:2075114348 > label:TOAV4oLn6QD7CfJieQ90FAEQB06orMEVm2na0 > > 2012-05-21 13:04:47.754843 [ERR] sofia_glue.c:4478 a=crypto > in RTP/AVP, refer to rfc3711 > 2012-05-21 13:04:47.754843 [DEBUG] switch_channel.c:2592 > (sofia/external/7777 at uringme) Callstate Change DOWN -> > HANGUP > 2012-05-21 13:04:47.754843 [NOTICE] sofia.c:4998 Hangup > sofia/external/7777 at uringme [CS_NEW] > [INCOMPATIBLE_DESTINATION] > > > From bdfoster at endigotech.com Tue May 22 19:00:35 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Tue, 22 May 2012 11:00:35 -0400 Subject: [Freeswitch-users] default dialplan: signal busy when busy and voicemail disabled In-Reply-To: References: Message-ID: http://wiki.freeswitch.org/wiki/Mod_commands#user_data -BDF On Tue, May 22, 2012 at 4:24 AM, John Jacob wrote: > Hi, > > I guess you are all aware of the default dialplan entry for local > extensions: > > > > [...] > data="user/${dialed_extension}@${domain_name}"/> > > > data="loopback/app=voicemail:default ${domain_name} > ${dialed_extension}"/> > > > > > So, if a caller calls a callee, which disabled his voicemail and is > currently busy/unavailable FreeSWITCH plays a good bye message and > hangs up. This is unexpected for most of our clients and they are > interpreting it as a malfunctioning PBX. > > I'd like to change this default behavior and respect the extension's > setting for "vm-enabled". > If it's enabled, the default behavior is fine, if it's disabled, I'd > like FreeSWITCH to signal *busy* to the caller. > > > It looks like I can't test for user parameters with conditions, so I > don't have a clue how to implement this? > > For now I just removed answer,sleep and bridge to voicemail and busy > is signalled just fine, but I'm aware that I killed voicemail > completely by doing this ;) > > > Any hints or tipps? I can't implement dynamic dialplans with mod_curl > though, because those don't fit into the current setup. > > > Thanks, > > Johannes > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Brian D. Foster Endigo Computer LLC Email: bdfoster at endigotech.com Phone: 317-800-7876 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120522/9e000397/attachment-0001.html From mario_fs at mgtech.com Tue May 22 19:16:42 2012 From: mario_fs at mgtech.com (Mario G) Date: Tue, 22 May 2012 08:16:42 -0700 Subject: [Freeswitch-users] FreeSWITCH, OSX, Libtool, Macports In-Reply-To: References: <523D8618-21D9-442E-9D23-4321EAFA6558@opencsta.org> <7FFFF20B-D26F-4EDC-A19F-6AE1C48D5048@mgtech.com> <761EBAA8-6CE4-4A9B-BD8F-4A5EB064EEFA@mgtech.com> <7AA59325-D16B-4C20-87AF-87B35B2C763C@mgtech.com> Message-ID: <9A697AB9-82AE-42F2-99CE-A1E16A5242DC@mgtech.com> That is interesting, FS is running on a Mac Mini and is in 64 bit mode according to Activity Monitor. I did not add anything to the config flags. I am only using tools from Apple (XCODE 4.2) and never had this problem since I installed FS Oct 2010. I had updated 10.6, XCODE and FS several times and no issues. Keep in mind that the Mini was a clean install of 10.6 (from Apple) and it is primarily used for PBX. I never put anything else on it other than the FS prerequisites mentioned in the wiki and this post. I wonder if something was left over from previous attempts as FS installation. I have multiple machines and am stuck on Leopard due to Quicken (still too buggy), since I waited this long I thought I may as well wait for Mountain Lion and do a clean install of everything on that. Mario G On May 21, 2012, at 4:38 PM, Neil Patel wrote: > Thanks Mario, > > There was an old Jira that reported the same problem as mine, compiling on OSX: > http://jira.freeswitch.org/browse/FS-1949 > > Looks like the fix had to do with setting some config parameters to compile FS in 64-bit mode (I'm on 64-bit macbook pro). Is this still necessary? I remember reading on another thread that nowadays you should not mess with the compile flags when running ./configure since the process auto-detects what you need. But still, is there any reason to force compile FS in 64 bit or 32 bit mode for OSX? > > Also note that I'm on OSX Lion, though I hope that isn't the cause. > > Thanks, > Neil > > On Mon, May 21, 2012 at 1:40 PM, Mario G wrote: > Wish I could help more... this is what I tried today and all worked fine, I am on Snow Leopard. You may want to try to redo the git, may have been an xfr issue? > 1, git pull followed by make current - worked > 2. clean src fs dir, new git reload, bootstrap.sh, configure with ssl, make install - worked fine > > > On May 21, 2012, at 10:29 AM, Neil Patel wrote: > >> Hi Mario, >> >> Thanks for your valuable posts on getting FS up and running on OS X. >> >> I'm doing so myself and ran into a snag compiling latest Git: >> http://jira.freeswitch.org/browse/FS-4230 >> >> Given your expertise on FS and OSX I thought you may be able to help me get through this issue. Any suggestions? >> >> Thanks, >> Neil >> >> On Fri, May 11, 2012 at 9:41 AM, Mario G wrote: >> Some answers for you: >> >> On May 10, 2012, at 7:23 PM, Chris Mylonas wrote: >> >>> That's great Mario. I will update my blog accordingly where I posted some stuff (http://mrvoip.com.au/blog/install-freeswitch-osx-mysql-must-remove-macports) >>> >>> I just winged it because it used to work, so most of the prerequisites were installed. >>> I did not put the --with-openssl flag when doing configure. What functionality do I lose? >> >> From what I remember, it complained about gnutis or something like that, you have to check the messages but it is really important, used to not be required. It took several days to figure this one out. I think the message was: >> configure: error: --with-ssl was given, but GNUTLS is not available. >> >>> >>> >>> pkgconfig was already on my system from earlier efforts - so I guess I got lucky with that one. >>> To be honest, once I resolved my macports issues (by removing it) it was just like doing it in linux. >> >> The funny thing is I still see a few pkgconfig missing messages during the make but FS works fine for me, I think it's for things I am not using. I will be looking into this more before I update the wiki. >> >>> >>> I'd like to give it another shot with a fresh OSX install - but I really don't want to go about setting up my machine again! I have an old imac that is not doing much, but I don't think it will run 10.6 :( I will try and give it a shot this afternoon seeing as my girlfriend is out taking photos :) >> >> FYI, when I have not updated for a long time or make major changes like updating xCode: I backup the Mac Mini running FS as a bootable partition (I use SuperDuper), then I connect it to an iMac and boot using firewire target mode to boot from the backup. Now I can play round all I want. If all goes well, I just restore to the Mini, if not I can start over. For frequent updates I just backup the FS bin and source in the Mini with and do a git pull, etc.... >> Mario G >>> >>> Kind Regards, >>> Chris >>> >>> >>> >>> >>> On 11/05/2012, at 11:48 AM, Mario G wrote: >>> >>>> Chris, I have been running FS on 10.6,x since 2010 fine and have been updating a lot recently to help testing, I have had no problems. I don't use macport stuff. I wrote the original FS osX wiki but there are some differences since 2010 (I am working on fixing the wiki in the next few weeks). The wiki install instructions work if you add the following. Hope this helps: >>>> >>>> 1. Add this info: >>>> FreeSWITCH? has many functions that invoke PKG-CONFIG, so it must be downloaded and installed separately. >>>> # Got to [http://pkgconfig.freedesktop.org/releases/ here] and download the latest pkg-config, by default it is placed into your Downloads folder. MUST USE .25 for OSXdue to glib2 dependencies! >>>> # Open Downloads and click the file to uncompress it. >>>> # Launch the Terminal application if not already running and issue these commands to move the source to the src directory, build and install: >>>> cd ~/Downloads >>>> mv pkg-config-0.25 /usr/local/src >>>> cd /usr/local/src/pkg-config-0.25 >>>> ./configure >>>> make >>>> sudo make install >>> >>>> >>>> 2. You MUST do this: >>>> ./bootstrap.sh >>>> ./configure --with-openssl >>>> >>>> 3. The FLITE fix is no longer needed. >>>> >>>> On May 10, 2012, at 5:01 PM, Chris Mylonas wrote: >>>> >>>>> Hi FS Users, >>>>> >>>>> tl;dr; - removed macports, removed tree, pulled fresh tree = installed FS on OSX. >>>>> >>>>> >>>>> Here is the longer version that had a bit of a pre-emptive whinge about libtool version mismatch. But we never got there - it worked! >>>>> >>>>> I'm in the process of re-installing FS on OSX (10.6.8). I have removed Macports to try and get this up and running. >>>>> I re-bootstrapped and got this error: >>>>> >>>>> quiet_libtool: Version mismatch error. This is libtool 2.4, but the >>>>> quiet_libtool: definition of this LT_INIT comes from libtool 2.2.4. >>>>> quiet_libtool: You should recreate aclocal.m4 with macros from libtool 2.4 >>>>> quiet_libtool: and run autoconf again. >>>>> make[1]: *** [libfreeswitch_la-switch_apr.lo] Error 63 >>>>> make: *** [all] Error 2 >>>>> >>>>> Removed the whole git tree just in case there was some left over junk. Pulled a fresh tree and noticed this remark during a fresh bootstrap >>>>> >>>>> arakis:freeswitch-git chrismylonas$ rm -Rf freeswitch/ >>>>> arakis:freeswitch-git chrismylonas$ git clone git://git.freeswitch.org/freeswitch.git >>>>> Cloning into freeswitch... >>>>> remote: Counting objects: 185609, done. >>>>> remote: Compressing objects: 100% (39208/39208), done. >>>>> remote: Total 185609 (delta 143241), reused 181443 (delta 140035) >>>>> Receiving objects: 100% (185609/185609), 77.85 MiB | 267 KiB/s, done. >>>>> Resolving deltas: 100% (143241/143241), done. >>>>> arakis:freeswitch-git chrismylonas$ cd freeswitch/ >>>>> arakis:freeswitch chrismylonas$ ./bootstrap.sh >>>>> bootstrap: checking installation... >>>>> bootstrap: autoconf version 2.61 (ok) >>>>> bootstrap: automake version 1.10 (ok) >>>>> bootstrap: aclocal version 1.10 (ok) >>>>> bootstrap: libtool version 2.2.4 (ok) >>>>> Bootstrapping using: >>>>> autoconf : /usr/bin/autoconf >>>>> automake : /usr/bin/automake >>>>> aclocal : /usr/bin/aclocal >>>>> libtool : /usr/bin/glibtool (2.2.4.) >>>>> libtoolize: /usr/bin/glibtoolize >>>>> make : /usr/bin/make (GNU Make 3.81) >>>>> awk : () >>>>> >>>>> It is still reporting a 2.2.4 version of libtool. >>>>> ... >>>>> ... >>>>> ... >>>>> In the end, it has been compiled and installed though >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at 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 at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at 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 at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120522/255ac9d8/attachment-0001.html From bdfoster at endigotech.com Tue May 22 19:25:46 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Tue, 22 May 2012 11:25:46 -0400 Subject: [Freeswitch-users] sofia current calls for a user In-Reply-To: References: Message-ID: As stated by MC, mod_limit fulfills that role as a side effect. -BDF On May 22, 2012 8:56 AM, "babak yakhchali" wrote: > For example I need to find out if user 1000 at default is talking to someone > or not. Currently I use "show channels like 1000" and then parse it for > channels like sofia/internal/1000 or something like this. but as I > mentioned, looking at mod_sofia code I see there is a table named > sip_dialogs which contains info that I need but there is no api (or I donno > about it) to show sip dialogs. I need some way to get that info if it is > possible > thanx > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120522/06041809/attachment.html From bdfoster at endigotech.com Tue May 22 19:30:06 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Tue, 22 May 2012 11:30:06 -0400 Subject: [Freeswitch-users] sofia current calls for a user In-Reply-To: References: Message-ID: http://wiki.freeswitch.org/wiki/Limit On May 22, 2012 11:25 AM, "Brian Foster" wrote: > As stated by MC, mod_limit fulfills that role as a side effect. > > -BDF > On May 22, 2012 8:56 AM, "babak yakhchali" > wrote: > >> For example I need to find out if user 1000 at default is talking to >> someone or not. Currently I use "show channels like 1000" and then parse it >> for channels like sofia/internal/1000 or something like this. but as I >> mentioned, looking at mod_sofia code I see there is a table named >> sip_dialogs which contains info that I need but there is no api (or I donno >> about it) to show sip dialogs. I need some way to get that info if it is >> possible >> thanx >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120522/2381ae74/attachment.html From curriegrad2004 at gmail.com Tue May 22 20:58:16 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Tue, 22 May 2012 09:58:16 -0700 Subject: [Freeswitch-users] Freeswitch stable on embedded devices? In-Reply-To: References: <1337546724347-7568340.post@n2.nabble.com> <1337560295076-7568516.post@n2.nabble.com> Message-ID: Just a quick note about embedded platforms and FreeSWITCH, git head compiles with uClibc as the libc just fine. There were a few changes that broke successful compilation of FreeSWITCH, but they have since been fixed. This statement is only applicable given if the all available options of uClibc is present/enabled. This means IPv6 support, RPC support and a few other obsolete BSD, SuSv3. SuSv4 legacy functions was enabled in my uClibc build when I performed the testing. On Mon, May 21, 2012 at 10:33 PM, Brian Foster wrote: > You are still transcoding from .wav to ulaw alaw etc. You can work around > that by not transcoding on the fly. > > Using media bugs is transcoding. I don't know if there's a good work around > for that though. > > -BDF > > On May 21, 2012 8:55 PM, "Tom C" wrote: >> >> >> >>> Do you use uncompressed files for MoH or MP3, which could explain the >>> chopiness? >>> >> Nope, just the standard WAV files that come with the git. ?And I'm using >> the correct bitrate, so no trans-coding issues. >> >> >> >>> Do you recompile Freeswitch yourself for the Dockstar? If yes, is that >>> hard >>> to do? I need to make sure I'm able to update the devices in case serious >>> bugs or useful features come up once they're out in the field. >>> >> >> Yes, compiling on the dockstar is just as easy as on any other platform, >> once you have all the requisite packages installed. ?The smaller Linux >> distributions made for these devices don't contain every package that the >> major (desktop) distributions have. ?A quick web search will turn up a list >> for you. >> >> Everything builds fine on the dockstar with only 128megs of RAM. ?Except, >> coincidentally enough, FLITE, which needs about 450megs of addressable >> memory to build successfully. ?So to compile that module, you need to create >> a swap file of at least 350 megs. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From lists.jj at googlemail.com Tue May 22 20:32:00 2012 From: lists.jj at googlemail.com (Johannes Jakob) Date: Tue, 22 May 2012 18:32:00 +0200 Subject: [Freeswitch-users] default dialplan: signal busy when busy and voicemail disabled In-Reply-To: References: Message-ID: <0A15E7B31EC146A194C9E46D43969173@gmail.com> Thanks Brian, that did the trick! Thanks! -- Johannes Jakob Sent with Sparrow (http://www.sparrowmailapp.com/?sig) On Tuesday, 22. May 2012 at 17:00, Brian Foster wrote: > http://wiki.freeswitch.org/wiki/Mod_commands#user_data > > -BDF > > On Tue, May 22, 2012 at 4:24 AM, John Jacob wrote: > > Hi, > > > > I guess you are all aware of the default dialplan entry for local extensions: > > > > > > > > [...] > > > data="user/${dialed_extension}@${domain_name}"/> > > > > > > > data="loopback/app=voicemail:default ${domain_name} > > ${dialed_extension}"/> > > > > > > > > > > So, if a caller calls a callee, which disabled his voicemail and is > > currently busy/unavailable FreeSWITCH plays a good bye message and > > hangs up. This is unexpected for most of our clients and they are > > interpreting it as a malfunctioning PBX. > > > > I'd like to change this default behavior and respect the extension's > > setting for "vm-enabled". > > If it's enabled, the default behavior is fine, if it's disabled, I'd > > like FreeSWITCH to signal *busy* to the caller. > > > > > > It looks like I can't test for user parameters with conditions, so I > > don't have a clue how to implement this? > > > > For now I just removed answer,sleep and bridge to voicemail and busy > > is signalled just fine, but I'm aware that I killed voicemail > > completely by doing this ;) > > > > > > Any hints or tipps? I can't implement dynamic dialplans with mod_curl > > though, because those don't fit into the current setup. > > > > > > Thanks, > > > > Johannes > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > -- > Brian D. Foster > Endigo Computer LLC > Email: bdfoster at endigotech.com (mailto:bdfoster at endigotech.com) > Phone: 317-800-7876 > Indianapolis, Indiana, USA > > This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From jerry.richards at teotech.com Wed May 23 01:20:08 2012 From: jerry.richards at teotech.com (Jerry Richards) Date: Tue, 22 May 2012 21:20:08 +0000 Subject: [Freeswitch-users] Is There a MySQL Equivalent to Postgresql Prefix Module (supporting mod_lcr)? Message-ID: <1545146083A72C4DB7B66584B7E5D98402BD6DEE@BY2PRD0410MB377.namprd04.prod.outlook.com> Hello, Does anyone know if there is a MySQL compatible module that does what the Postgresql prefix module does for mod_lcr? Thanks, Jerry -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120522/2f183c31/attachment-0001.html From krice at freeswitch.org Wed May 23 01:26:07 2012 From: krice at freeswitch.org (Ken Rice) Date: Tue, 22 May 2012 16:26:07 -0500 Subject: [Freeswitch-users] Is There a MySQL Equivalent to Postgresql Prefix Module (supporting mod_lcr)? In-Reply-To: <1545146083A72C4DB7B66584B7E5D98402BD6DEE@BY2PRD0410MB377.namprd04.prod.outlook.com> Message-ID: There is no equivalent to my knowledge... On 5/22/12 4:20 PM, "Jerry Richards" wrote: > Hello, > Does anyone know if there is a MySQL compatible module that does what the > Postgresql prefix module does for mod_lcr? > > Thanks, > Jerry > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120522/cb2576e2/attachment.html From gregor at infomedia.si Wed May 23 02:06:23 2012 From: gregor at infomedia.si (Gregor Nanger) Date: Wed, 23 May 2012 00:06:23 +0200 Subject: [Freeswitch-users] Freswitch ODBC tables Message-ID: Hi! I have FS in linux and moved sofia db via ODBC to MS Sql server and it works ok. Are there any documentation about tables and how they are used by sofia? Thanx, Gregor -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120523/b7875842/attachment.html From bdfoster at endigotech.com Wed May 23 03:16:47 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Tue, 22 May 2012 19:16:47 -0400 Subject: [Freeswitch-users] Freswitch ODBC tables In-Reply-To: References: Message-ID: All over the wiki, my friend. -BDF On May 22, 2012 6:07 PM, "Gregor Nanger" wrote: > Hi! > > I have FS in linux and moved sofia db via ODBC to MS Sql server and it > works ok. > > Are there any documentation about tables and how they are used by sofia? > > Thanx, Gregor > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120522/4089a434/attachment.html From ahe.sanath at gmail.com Wed May 23 06:21:17 2012 From: ahe.sanath at gmail.com (Sanath Prasanna) Date: Wed, 23 May 2012 07:51:17 +0530 Subject: [Freeswitch-users] Run LUA script in different server In-Reply-To: References: Message-ID: Hi MC, Tx a lot for advice. Problem is port. Now I sorted it & call coming to B server. In here diverted call coming to server A. So I used variable_sip_history_info parameter to extarct real B number.(Real called party number) . But in server B, that parameter is not coming. (variable_sip_history_info) Pls help to solve that. Br, Sanath On Tue, May 22, 2012 at 12:53 AM, Michael Collins wrote: > Look at line #186 of your trace: > 2012-05-21 07:56:49.735660 [ERR] mod_sofia.c:3957 Invalid Profile > > You need to figure out why your internal profile isn't running. Try "sofia > profile internal restart" and see what happens. > > -MC > > > On Sun, May 20, 2012 at 10:45 PM, Sanath Prasanna wrote: > >> Hi MC, >> I did the change according to ure instruction. But error is coming. Here >> I attached freeswitch.log file >> >> I change the confs as follows in BOX A. (Operator connected Freeswitch >> box) >> BOX B ip is 10.22.29.253 >> >> vi /usr/local/freeswitch/conf/dialplan/default.xml >> >> >> >> >> >> >> >> Also add following to ACL file in BOX B >> >> Pls advice to solve the problem here. >> Br, >> Sanath >> >> >> On Fri, May 18, 2012 at 9:57 AM, Michael Collins wrote: >> >>> If I understand your question correctly, yes you can do this. You can >>> send calls from one FreeSWITCH server to another. Start here: >>> http://wiki.freeswitch.org/wiki/Connect_Two_FreeSWITCH_Boxes >>> >>> Best way to learn is to get the FreeSWITCH books from Packt Publishing >>> and just start hacking code. >>> >>> -MC >>> >>> >>> On Thu, May 17, 2012 at 9:07 PM, Sanath Prasanna wrote: >>> >>>> Tx for advice MC & Anita. Can I do work around like this . >>>> Another freeswitch instant will be start in other server & calls will >>>> be transfer from operator connected freeswitch instance to this new >>>> freeswitch instance & vise versa. Pls advice. >>>> >>>> >>>> On Thu, May 17, 2012 at 5:05 PM, Anita Hall wrote: >>>> >>>>> You could run a Lua ESL server on a different machine but this will >>>>> not be the same as running a Lua script. >>>>> http://wiki.freeswitch.org/wiki/Event_Socket_Library >>>>> >>>>> regards, >>>>> Anita >>>>> >>>>> >>>>> >>>>> On Thu, May 17, 2012 at 4:37 AM, Michael Collins wrote: >>>>> >>>>>> I don't think you can directly do what you are describing. However, >>>>>> you might be able to use mod_httapi for this. There's some documentation on >>>>>> the wiki and in the module. Keep in mind that this is a relatively new >>>>>> module so we don't have lots of examples yet, so you'll probably be doing a >>>>>> fair amount of research and testing. >>>>>> >>>>>> -MC >>>>>> >>>>>> >>>>>> On Wed, May 16, 2012 at 5:59 AM, Sanath Prasanna < >>>>>> ahe.sanath at gmail.com> wrote: >>>>>> >>>>>>> Hi all, >>>>>>> I have 2 servers. One server has SIP GW connection From Operator & >>>>>>> IVR applications need to build in other server. How to call distributed LUA >>>>>>> applications with Mysql Databases from the SIP GW server ? Pls advice. >>>>>>> Main idea is, maintaining SIP connection in one server & all the IVR >>>>>>> applications in other server. >>>>>>> Br, >>>>>>> Sanath >>>>>>> >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> >>> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120523/23bad967/attachment.html From albert_nguyen16 at hotmail.com Wed May 23 07:01:28 2012 From: albert_nguyen16 at hotmail.com (Albert Nguyen) Date: Wed, 23 May 2012 03:01:28 +0000 Subject: [Freeswitch-users] [WARNING] switch_core.c:1206 can not locate domain 10.239.236.1 In-Reply-To: References: , Message-ID: Hi, Is there any one know how to resolve this? It gives me this error message and it doesn't allow connection to the switch through this IP interface From: albert_nguyen16 at hotmail.com To: freeswitch-users at lists.freeswitch.org Date: Fri, 18 May 2012 09:34:02 +0000 Subject: Re: [Freeswitch-users] [WARNING] switch_core.c:1206 can not locate domain 10.239.236.1 Hi When I start FS, I get a warning message from FS like [WARNING] switch_core.c:1206 can not locate domain 10.239.236.1 but 10.239.236.1 is the local NIC ip address. How can I fix this problem? Thanks in advance. Al _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120523/9635d265/attachment-0001.html From gamar at center.com Wed May 23 04:02:51 2012 From: gamar at center.com (Gilbert Amar) Date: Tue, 22 May 2012 17:02:51 -0700 Subject: [Freeswitch-users] How do you split bridged calls ? Message-ID: <001201cd3877$655ed5b0$301c8110$@center.com> Hello I am trying to split bridged calls. Using mod_event_socket I try this api uuid_transfer c1061126-a429-11e1-9da3-85575e903927 -both playback::phrase:AC_0003: It does not work as FS seems to really need an extension. the other option I could try is : api uuid_transfer c1061126-a429-11e1-9da3-85575e903927 -both park and then for each leg: sendmsg call-command: execute execute-app-name: playback execute-app-arg: phrase:AC_0003: Is there a better way ? I am using FreeSWITCH Version 1.0.head (git-fdaa155 2012-02-10 13-17-54 +0000) Thank you Gilbert -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120522/50473ca8/attachment.html From avi at avimarcus.net Wed May 23 10:31:45 2012 From: avi at avimarcus.net (Avi Marcus) Date: Wed, 23 May 2012 09:31:45 +0300 Subject: [Freeswitch-users] How do you split bridged calls ? In-Reply-To: <001201cd3877$655ed5b0$301c8110$@center.com> References: <001201cd3877$655ed5b0$301c8110$@center.com> Message-ID: You can't transfer to a playback. Playback is a dialplan application, not a destination. You can, however, playback to each leg individually with http://wiki.freeswitch.org/wiki/Mod_commands#uuid_broadcast uuid_broadcast [aleg|bleg|both] here it would be: api uuid_broadcast c1061126-a429-11e1-9da3-85575e903927 phrase::AC_0003 aleg for A leg. I think. Please update the wiki to make it more descriptive after you play with it. Of course, this still leaves the calls together. If you want to split them I suppose you really need to transfer each leg TO somewhere. -Avi On Wed, May 23, 2012 at 3:02 AM, Gilbert Amar wrote: > Hello **** > > ** ** > > I am trying to split bridged calls.**** > > Using mod_event_socket I try this**** > > ** ** > > api uuid_transfer c1061126-a429-11e1-9da3-85575e903927 -both > playback::phrase:AC_0003:**** > > ** ** > > It does not work as FS seems to really need an extension.**** > > ** ** > > the other option I could try is :**** > > ** ** > > api uuid_transfer c1061126-a429-11e1-9da3-85575e903927 -both park > **** > > ** ** > > and then for each leg:**** > > ** ** > > sendmsg**** > > call-command: execute**** > > execute-app-name: playback**** > > execute-app-arg: phrase:AC_0003:**** > > ** ** > > Is there a better way ?**** > > I am using FreeSWITCH Version 1.0.head (git-fdaa155 2012-02-10 13-17-54 > +0000)**** > > ** ** > > Thank you**** > > ** ** > > Gilbert**** > > ** ** > > ** ** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120523/c84dfe76/attachment.html From miha at softnet.si Wed May 23 10:42:57 2012 From: miha at softnet.si (Miha) Date: Wed, 23 May 2012 08:42:57 +0200 Subject: [Freeswitch-users] Call_group intercept In-Reply-To: References: <4FB9E602.8070708@softnet.si> <4FBB2E2F.7070901@softnet.si> Message-ID: <4FBC86F1.5040606@softnet.si> Hi @Michal, thank you for your reply. I noticed that it was different problem. Thanks! Miha On 5/22/2012 9:07 AM, Michael Collins wrote: > HI @Michal, > > > I have defined callgroup for users that I would like to have in > the same callgroup. > > After call hit public dialplan I make /> but in log I can not see that any variable for that user is > set. If I make call from default dialplan variables are set and I > can use them. > Would it be better to redirect call to default dialplan? > > Regards, > Miha > > > If your calls are hitting the public context then that suggests to me > that you are not authenticating the user when he/she calls. If you > don't authenticate then you don't get the user's channel variables. > This happens, e.g., when you let callers in via ACL with a rule in the > "domains" ACL. > > If you know which user it is supposed to be then you can cheat and use > the set_user app: > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_set_user > > -MC > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120523/a7ca5459/attachment-0001.html From gregor at infomedia.si Wed May 23 10:44:01 2012 From: gregor at infomedia.si (Gregor Nanger) Date: Wed, 23 May 2012 08:44:01 +0200 Subject: [Freeswitch-users] Freswitch ODBC tables In-Reply-To: References: Message-ID: Ok Brian, you got me thank you... 2012/5/23 Brian Foster > All over the wiki, my friend. > > -BDF > On May 22, 2012 6:07 PM, "Gregor Nanger" wrote: > >> Hi! >> >> I have FS in linux and moved sofia db via ODBC to MS Sql server and it >> works ok. >> >> Are there any documentation about tables and how they are used by sofia? >> >> Thanx, Gregor >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120523/b771aa32/attachment.html From gregor at infomedia.si Wed May 23 11:12:48 2012 From: gregor at infomedia.si (Gregor Nanger) Date: Wed, 23 May 2012 09:12:48 +0200 Subject: [Freeswitch-users] Freswitch ODBC tables In-Reply-To: References: Message-ID: Please give me hint what to search :-( I tried ODBC tables, sip_registration table, sofia tables, sofia ODBC and tables are mentioned, but no single description for what tables are used. I figure it out that in registration table are stored active sip registrations with expire date etc..., sip_registration tables has same information, but with more fields. In channels table are stored all legs information when call is active... In Calls are stored active calls when bridge happened between legs... Can you give me a hand for other tables? 2012/5/23 Gregor Nanger > Ok Brian, you got me > > thank you... > > > > > 2012/5/23 Brian Foster > >> All over the wiki, my friend. >> >> -BDF >> On May 22, 2012 6:07 PM, "Gregor Nanger" wrote: >> >>> Hi! >>> >>> I have FS in linux and moved sofia db via ODBC to MS Sql server and it >>> works ok. >>> >>> Are there any documentation about tables and how they are used by sofia? >>> >>> Thanx, Gregor >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> Join Us At ClueCon - Aug 7-9, 2012 >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120523/85c3755e/attachment.html From albert_nguyen16 at hotmail.com Wed May 23 12:26:21 2012 From: albert_nguyen16 at hotmail.com (Albert Nguyen) Date: Wed, 23 May 2012 08:26:21 +0000 Subject: [Freeswitch-users] Help on Explaination of the configurations parrametters In-Reply-To: <4FB54B0F.3050000@cupis.co.uk> References: , <4FB54B0F.3050000@cupis.co.uk> Message-ID: Hi Paul, I am new to FS and trying to setup FS as a "dumb" SBC as per example 2 in the wiki website so that I can convert inband DTMF to RFC 8233 and vice versa. This is the windows version of FS. Is it possible that I can get some assistant from you guys to get it up and running? Please advise how we can proceed. Regards, Al > Date: Thu, 17 May 2012 20:01:35 +0100 > From: paul at cupis.co.uk > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Help on Explaination of the configurations parrametters > > On 17/05/12 02:50, Albert Nguyen wrote: > > I am new to FS and trying to setup a SBC using the example 2 in the FS > > wiki website. The link is > > http://wiki.freeswitch.org/wiki/SBC_FreeSWITCH_Configuration_Example_2. > > > > There are parameters in the example that I have to replaces for it to > > work with my own scenarios. However I have problem understanding the > > following lines. Is there anyone able to explain what does this mean > > > > > expression="(((192.168.)|(172.24.)|(10.10.))\d+\.\d+(:\d+)(;dtg=\w+)?)"> > > > > The bits I am not sure is )\d+\.\d+(:\d+)(;dtg=\w+)?). What does this do? > > This will match certain (IPv4) IP addresses, with ports specified: > > 192.168.x.x:y > 172.24.x.x:y > 10.10.x.x:y > > optionally ending in: > > ;dtg=xxxx > > In the given example, the rest of the XML is executed conditionally on > this match. If the sip_redirect_contact_0 matches the expression, then > the are executed, otherwise the is executed. > > Regards, > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120523/e2491f44/attachment.html From andy at fabulous4.co.uk Wed May 23 12:29:25 2012 From: andy at fabulous4.co.uk (Andy Ayers) Date: Wed, 23 May 2012 09:29:25 +0100 Subject: [Freeswitch-users] Best database setup for high volume Message-ID: <005401cd38be$2a9434b0$7fbc9e10$@fabulous4.co.uk> Hi, Can anyone tell me what the best database setup is for dealing with high call volumes? The background: I've been running with the standard SQLite system for about 3 years without issue but recently am getting a lot of database corruption errors ('database disk image is malformed'). Easily solved by deleting the db and restarting but on occasion it brings my switch down. I've tried upgrading to odbc and mysql but hit 2 problems: FLAG_MULTIPLE_STATEMENTS Error in my_thread_global_end() Both of which are mentioned in the user groups but not with any solutions that worked. I've just upgraded to version 1.2 so this may solve the corruption problems but would really like to get my system set up to handle as much traffic as possible. Any advice or suggestions much appreciated. Kind regards Andy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120523/b2b1d704/attachment-0001.html From miha at softnet.si Wed May 23 12:40:59 2012 From: miha at softnet.si (Miha) Date: Wed, 23 May 2012 10:40:59 +0200 Subject: [Freeswitch-users] Freswitch ODBC tables In-Reply-To: References: Message-ID: <4FBCA29B.1000400@softnet.si> I think you are looking for this: http://wiki.freeswitch.org/wiki/Mod_db Regards, Miha On 5/23/2012 9:12 AM, Gregor Nanger wrote: > Please give me hint what to search :-( > > I tried ODBC tables, sip_registration table, sofia tables, sofia ODBC > and tables are mentioned, but no single description for what tables > are used. > I figure it out that in registration table are stored active sip > registrations with expire date etc..., sip_registration tables has > same information, but with more fields. > > In channels table are stored all legs information when call is active... > > In Calls are stored active calls when bridge happened between legs... > > Can you give me a hand for other tables? > > > 2012/5/23 Gregor Nanger > > > Ok Brian, you got me > > thank you... > > > > > 2012/5/23 Brian Foster > > > All over the wiki, my friend. > > -BDF > > On May 22, 2012 6:07 PM, "Gregor Nanger" > wrote: > > Hi! > > I have FS in linux and moved sofia db via ODBC to MS Sql > server and it works ok. > > Are there any documentation about tables and how they are > used by sofia? > > Thanx, Gregor > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120523/bdd8c1f8/attachment.html From avi at avimarcus.net Wed May 23 12:52:03 2012 From: avi at avimarcus.net (Avi Marcus) Date: Wed, 23 May 2012 11:52:03 +0300 Subject: [Freeswitch-users] Best database setup for high volume In-Reply-To: <005401cd38be$2a9434b0$7fbc9e10$@fabulous4.co.uk> References: <005401cd38be$2a9434b0$7fbc9e10$@fabulous4.co.uk> Message-ID: Is your issue the CDRs & Voicemail or session count, current calls, recovery data.. You can make sure track-calls is off... -nosql -- disable internal sql scoreboard I'm not sure if that kills presence or not. Disable presence if you don't need it - it's a real usage hog. I have odbc to mysql but I made calls,channels,sip_dialogs, sip_subscriptions, etc into memory tables a few months ago. (I left sip_registrations as non-memory for persistence of a sort) -Avi On Wed, May 23, 2012 at 11:29 AM, Andy Ayers wrote: > Hi,**** > > ** ** > > Can anyone tell me what the best database setup is for dealing with high > call volumes?**** > > ** ** > > The background:**** > > ** ** > > I?ve been running with the standard SQLite system for about 3 years > without issue but recently am getting a lot of database corruption errors > (?database disk image is malformed?). Easily solved by deleting the db and > restarting but on occasion it brings my switch down.**** > > ** ** > > I?ve tried upgrading to odbc and mysql but hit 2 problems:**** > > ** ** > > FLAG_MULTIPLE_STATEMENTS**** > > Error in my_thread_global_end()**** > > ** ** > > Both of which are mentioned in the user groups but not with any solutions > that worked.**** > > ** ** > > I?ve just upgraded to version 1.2 so this may solve the corruption > problems but would really like to get my system set up to handle as much > traffic as possible.**** > > ** ** > > Any advice or suggestions much appreciated.**** > > ** ** > > Kind regards**** > > Andy**** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120523/2176a532/attachment.html From peter.olsson at visionutveckling.se Wed May 23 12:53:09 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Wed, 23 May 2012 08:53:09 +0000 Subject: [Freeswitch-users] Freswitch ODBC tables Message-ID: <1FFF97C269757C458224B7C895F35F150D5394@cantor.std.visionutv.se> I don't think there is much documentation about this - since it's mostly data used internally by FS. What do you want to do more specifically? I guess most of the tables are quite self-explaining by their names.. /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Gregor Nanger Skickat: den 23 maj 2012 09:13 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] Freswitch ODBC tables Please give me hint what to search :-( I tried ODBC tables, sip_registration table, sofia tables, sofia ODBC and tables are mentioned, but no single description for what tables are used. I figure it out that in registration table are stored active sip registrations with expire date etc..., sip_registration tables has same information, but with more fields. In channels table are stored all legs information when call is active... In Calls are stored active calls when bridge happened between legs... Can you give me a hand for other tables? 2012/5/23 Gregor Nanger > Ok Brian, you got me thank you... 2012/5/23 Brian Foster > All over the wiki, my friend. -BDF On May 22, 2012 6:07 PM, "Gregor Nanger" > wrote: Hi! I have FS in linux and moved sofia db via ODBC to MS Sql server and it works ok. Are there any documentation about tables and how they are used by sofia? Thanx, Gregor _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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 at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4fbc8cbd32761286258325! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120523/0600dfc7/attachment-0001.html From peter.olsson at visionutveckling.se Wed May 23 12:54:13 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Wed, 23 May 2012 08:54:13 +0000 Subject: [Freeswitch-users] [WARNING] switch_core.c:1206 can not locate domain 10.239.236.1 Message-ID: <1FFF97C269757C458224B7C895F35F150D539F@cantor.std.visionutv.se> I would start by using the vanilla configuration from the FS dist (since that usually works out-of-the-box), and go from there. /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Albert Nguyen Skickat: den 23 maj 2012 05:01 Till: freeswitch-users at lists.freeswitch.org ?mne: Re: [Freeswitch-users] [WARNING] switch_core.c:1206 can not locate domain 10.239.236.1 Hi, Is there any one know how to resolve this? It gives me this error message and it doesn't allow connection to the switch through this IP interface ________________________________ From: albert_nguyen16 at hotmail.com To: freeswitch-users at lists.freeswitch.org Date: Fri, 18 May 2012 09:34:02 +0000 Subject: Re: [Freeswitch-users] [WARNING] switch_core.c:1206 can not locate domain 10.239.236.1 ________________________________ Hi When I start FS, I get a warning message from FS like [WARNING] switch_core.c:1206 can not locate domain 10.239.236.1 but 10.239.236.1 is the local NIC ip address. How can I fix this problem? Thanks in advance. Al _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4fbc520932762143213612! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120523/b8e4feae/attachment.html From gregor at infomedia.si Wed May 23 13:28:39 2012 From: gregor at infomedia.si (Gregor Nanger) Date: Wed, 23 May 2012 11:28:39 +0200 Subject: [Freeswitch-users] Freswitch ODBC tables In-Reply-To: <1FFF97C269757C458224B7C895F35F150D5394@cantor.std.visionutv.se> References: <1FFF97C269757C458224B7C895F35F150D5394@cantor.std.visionutv.se> Message-ID: Thank you guys! Just wanted to figure it out. I wanted to get status of extension (registered, oncall, ringing). And I can do this with registrations, channels and calls table. As I searched, I didn't find better way to do this from external application. There are also no API command to show exactly what I wanted. Gregor 2012/5/23 Peter Olsson > I don?t think there is much documentation about this ? since it?s mostly > data used internally by FS.**** > > ** ** > > What do you want to do more specifically? I guess most of the tables are > quite self-explaining by their names..**** > > ** ** > > /Peter**** > > ** ** > > ** ** > > *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *F?r *Gregor Nanger > *Skickat:* den 23 maj 2012 09:13 > *Till:* FreeSWITCH Users Help > *?mne:* Re: [Freeswitch-users] Freswitch ODBC tables**** > > ** ** > > Please give me hint what to search :-( > > I tried ODBC tables, sip_registration table, sofia tables, sofia ODBC and > tables are mentioned, but no single description for what tables are used. > I figure it out that in registration table are stored active sip > registrations with expire date etc..., sip_registration tables has same > information, but with more fields. > > In channels table are stored all legs information when call is active... > > In Calls are stored active calls when bridge happened between legs... > > Can you give me a hand for other tables? > > **** > > 2012/5/23 Gregor Nanger **** > > Ok Brian, you got me > > thank you...**** > > > > > **** > > 2012/5/23 Brian Foster **** > > All over the wiki, my friend.**** > > -BDF**** > > On May 22, 2012 6:07 PM, "Gregor Nanger" wrote:**** > > Hi! > > I have FS in linux and moved sofia db via ODBC to MS Sql server and it > works ok. > > Are there any documentation about tables and how they are used by sofia? > > Thanx, Gregor**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > ** ** > > > !DSPAM:4fbc8cbd32761286258325! **** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120523/d5ad7274/attachment.html From gregor at infomedia.si Wed May 23 13:30:33 2012 From: gregor at infomedia.si (Gregor Nanger) Date: Wed, 23 May 2012 11:30:33 +0200 Subject: [Freeswitch-users] Freswitch ODBC tables In-Reply-To: References: <1FFF97C269757C458224B7C895F35F150D5394@cantor.std.visionutv.se> Message-ID: And for the record if someone will search. FS on linux, sofia via odbc on MS SQL server 2008 and it works ok. Gregor 2012/5/23 Gregor Nanger > Thank you guys! > > Just wanted to figure it out. I wanted to get status of extension > (registered, oncall, ringing). And I can do this with registrations, > channels and calls table. As I searched, I didn't find better way to do > this from external application. There are also no API command to show > exactly what I wanted. > > Gregor > > > > > 2012/5/23 Peter Olsson > >> I don?t think there is much documentation about this ? since it?s >> mostly data used internally by FS.**** >> >> ** ** >> >> What do you want to do more specifically? I guess most of the tables are >> quite self-explaining by their names..**** >> >> ** ** >> >> /Peter**** >> >> ** ** >> >> ** ** >> >> *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *F?r *Gregor Nanger >> *Skickat:* den 23 maj 2012 09:13 >> *Till:* FreeSWITCH Users Help >> *?mne:* Re: [Freeswitch-users] Freswitch ODBC tables**** >> >> ** ** >> >> Please give me hint what to search :-( >> >> I tried ODBC tables, sip_registration table, sofia tables, sofia ODBC and >> tables are mentioned, but no single description for what tables are used. >> I figure it out that in registration table are stored active sip >> registrations with expire date etc..., sip_registration tables has same >> information, but with more fields. >> >> In channels table are stored all legs information when call is active... >> >> In Calls are stored active calls when bridge happened between legs... >> >> Can you give me a hand for other tables? >> >> **** >> >> 2012/5/23 Gregor Nanger **** >> >> Ok Brian, you got me >> >> thank you...**** >> >> >> >> >> **** >> >> 2012/5/23 Brian Foster **** >> >> All over the wiki, my friend.**** >> >> -BDF**** >> >> On May 22, 2012 6:07 PM, "Gregor Nanger" wrote:**** >> >> Hi! >> >> I have FS in linux and moved sofia db via ODBC to MS Sql server and it >> works ok. >> >> Are there any documentation about tables and how they are used by sofia? >> >> Thanx, Gregor**** >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org**** >> >> ** ** >> >> >> !DSPAM:4fbc8cbd32761286258325! **** >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120523/ab7ff750/attachment-0001.html From gopalakrishnan.an at gmail.com Wed May 23 14:04:58 2012 From: gopalakrishnan.an at gmail.com (Gopalakrishnan N) Date: Wed, 23 May 2012 15:34:58 +0530 Subject: [Freeswitch-users] FreeTDM with BRI Card In-Reply-To: References: Message-ID: thanks for your comments. All I need is, I need to make work OpenVox BRI card in Freeswitch with NT mode. Is there any solution for that? regards, On Tue, May 22, 2012 at 7:59 AM, James zhu wrote: > I do not think the dahdi and libpri can work properly. i think nobody > fully test with FW and dahdi,libpri.. > Best regards, > James.zhu > Doing asterisk/PRI/ss7/dahdi, linux, asterisk/sangoma cards, recording > device, VOIP gateway. > website: www.hiastar.com > > > ------------------------------ > From: gopalakrishnan.an at gmail.com > Date: Mon, 21 May 2012 15:42:24 +0530 > To: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] FreeTDM with BRI Card > > > Hi all, > > I am using Openvox BRI card with Freeswitch using FreeTDM. In NT > mode I couldn't get dialtone. I got the error like " Unable to get channel > 1: -1". Please help me to solve this issue. I have attached my > configuration details. > > I have used FreeTDM module with LibPRI and Freeswitch > > Regards, > Gopal. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: consulting at freeswitch.org > http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel > Communication Server Official FreeSWITCH Sites > http://www.freeswitch.org http://wiki.freeswitch.org > http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 > FreeSWITCH-users mailing list FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120523/6cf22aa4/attachment.html From lists.jj at googlemail.com Wed May 23 12:40:13 2012 From: lists.jj at googlemail.com (Johannes Jakob) Date: Wed, 23 May 2012 10:40:13 +0200 Subject: [Freeswitch-users] FS compatible sugarcrm integration? Message-ID: <67900F5CAC194B7380C19E8ABB5DD384@gmail.com> Hi, I'm looking for a FreeSWITCH compatible solution for integrating our existing sugarcrm (currently community edition, but that can be changed if needed) in our call center. We need * support for our own FreeSWITCH servers, no asterisk, no cloud solution * popups with caller information on incoming calls * click to call for outbound calls * automatic call-logging * a quick way to add "call again" reminders A plugin for sugarcrm, or windows software (even a soft phone would be OK) with sugarcrm integration would be fine. Is anybody of you using something that fits our requirements? Most of the plugins listed on sugarforge are either using Asterisk (AMI) or their own cloud services? Thanks again and best Regards, Johannes From coopara at go2.pl Wed May 23 13:17:40 2012 From: coopara at go2.pl (=?UTF-8?Q?coopara?=) Date: Wed, 23 May 2012 11:17:40 +0200 Subject: [Freeswitch-users] =?utf-8?q?FreeSWITCH_logs_to_C=23_application?= Message-ID: Hi, I would like to collect all FS logs to my C# application. Is there any possible way to do that? I use FreeSWITCH.EventBinding.Bind function and I'm looking for event_id which is switch_event_types_t.SWITCH_EVENT_LOG in handler but unfortunately it never occurs. Best regards, Coopara From saimohan at cem-solutions.net Wed May 23 13:25:24 2012 From: saimohan at cem-solutions.net (Sai Mohan) Date: Wed, 23 May 2012 14:55:24 +0530 Subject: [Freeswitch-users] Reg: Freeswitch registration over TLS as client Message-ID: <4FBCAD04.7040107@cem-solutions.net> Hi all, My question is regarding TLS regisrtation. If freeswitch as a client registering to other servers like asterisk over TLS, if the registering server does not support TLS registration, then whether our freeswitch will try to register over udp/tcp or not? Whether any fall-back mechanism is available or not to try to register over udp/tcp? Whether it keep on trying on transport TLS only? Kindly let me know my quires. Thanks & Regards, Sai Mohan From krice at freeswitch.org Wed May 23 18:15:55 2012 From: krice at freeswitch.org (Ken Rice) Date: Wed, 23 May 2012 09:15:55 -0500 Subject: [Freeswitch-users] Freswitch ODBC tables In-Reply-To: Message-ID: You know Gregor, I don?t know if those tables are actually documented on the wiki, but if you look at the tables after FreeSWITCH creates them they should be fairly self explanatory (to an english speaker anyway). Tony is pretty good about using names that make sense for the various bits and pieces of freeswitch... On 5/23/12 4:30 AM, "Gregor Nanger" wrote: > And for the record if someone will search. FS on linux, sofia via odbc on MS > SQL server 2008 and it works ok. > > Gregor > > > > 2012/5/23 Gregor Nanger >> Thank you guys! >> >> Just wanted to figure it out. I wanted to get status of extension >> (registered, oncall, ringing). And I can do this with registrations, >> channels? and calls table. As I searched, I didn't find better way to do this >> from external application. There are also no API command to show exactly what >> I wanted. >> >> Gregor >> >> >> >> >> 2012/5/23 Peter Olsson >>> I don?t think there is much documentation about this ? since it?s mostly >>> data used internally by FS. >>> ? >>> What do you want to do more specifically? I guess most of the tables are >>> quite self-explaining by their names.. >>> ? >>> /Peter >>> ? >>> ? >>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org >>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Gregor Nanger >>> Skickat: den 23 maj 2012 09:13 >>> Till: FreeSWITCH Users Help >>> ?mne: Re: [Freeswitch-users] Freswitch ODBC tables >>> >>> ? >>> Please give me hint what to search :-( >>> >>> I tried ODBC tables, sip_registration table, sofia tables, sofia ODBC and >>> tables are mentioned, but no single description for what tables are? used. >>> I figure it out that in registration table are stored active sip >>> registrations with expire date etc..., sip_registration tables has same >>> information, but with more fields. >>> >>> In channels table are stored all legs information when call is active... >>> >>> In Calls are stored active calls when bridge happened between legs... >>> >>> Can you give me a hand for other tables? >>> >>> 2012/5/23 Gregor Nanger >>> Ok Brian, you got me >>> >>> thank you... >>> >>> >>> >>> >>> 2012/5/23 Brian Foster >>> All over the wiki, my friend. >>> >>> -BDF >>> >>> On May 22, 2012 6:07 PM, "Gregor Nanger" wrote: >>>> >>>> Hi! >>>> >>>> I have FS in linux and? moved sofia db via ODBC to MS Sql server and it >>>> works ok. >>>> >>>> Are there any documentation about tables and how they are used by sofia? >>>> >>>> Thanx, Gregor >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> Join Us At ClueCon - Aug 7-9, 2012 >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at 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 at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> Join Us At ClueCon - Aug 7-9, 2012 >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> ? >>> >>> !DSPAM:4fbc8cbd32761286258325! >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> Join Us At ClueCon - Aug 7-9, 2012 >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120523/a02572d7/attachment-0001.html From mitch.capper at gmail.com Wed May 23 18:29:30 2012 From: mitch.capper at gmail.com (Mitch Capper) Date: Wed, 23 May 2012 07:29:30 -0700 Subject: [Freeswitch-users] Reg: Freeswitch registration over TLS as client In-Reply-To: <4FBCAD04.7040107@cem-solutions.net> References: <4FBCAD04.7040107@cem-solutions.net> Message-ID: Freeswitch registers of the transport you tell it to but does not automatically fallback for security reasons to another transport. You do not know ahead of time what security a remote server has? Another option would be to try TLS if the profile fails then try it again with TCP but you would have to script that. ~Mitch On Wed, May 23, 2012 at 2:25 AM, Sai Mohan wrote: > Hi all, > > My question is regarding TLS regisrtation. > > If freeswitch as a client registering to other servers like asterisk > over TLS, ?if the registering server does not support TLS registration, > then whether our freeswitch will try to register over udp/tcp or not? > Whether any fall-back mechanism is available or not to try to register > over udp/tcp? Whether it keep on trying on transport TLS only? > > Kindly let me know my quires. > > Thanks & Regards, > Sai Mohan > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From david at styleflare.com Wed May 23 18:36:07 2012 From: david at styleflare.com (David) Date: Wed, 23 May 2012 10:36:07 -0400 Subject: [Freeswitch-users] Seek and Mod_shout remote mp3 file. Message-ID: <4FBCF5D7.7090505@styleflare.com> I am using streamFile from javascript. Is rewind/fastfoward supposed to be supported in such a scenario? I cant seem to figure out how to get it to work. The idea is to playback/rewind/fastforward long files stored on a remote server. Thanks in advance for any pointers. David. From lazyvirus at gmx.com Wed May 23 18:53:22 2012 From: lazyvirus at gmx.com (Bzzz) Date: Wed, 23 May 2012 16:53:22 +0200 Subject: [Freeswitch-users] Freswitch ODBC tables In-Reply-To: References: Message-ID: <20120523165322.36331040@anubis.defcon1> On Wed, 23 May 2012 09:15:55 -0500 Ken Rice wrote: > I don?t know if those tables are actually documented on the wiki, > but if you look at the tables after FreeSWITCH creates them they > should be fairly self explanatory (to an english speaker anyway). A relational diagram would be awesome, as tables don't seem to use referential integrity, which could a risk in large installations that encounters a glitch or a failure (despite the overhead introduced by RI). JY -- Lao Tseu say: Man that scratch his butt at night awakes with stinking fingers. From mario_fs at mgtech.com Wed May 23 19:33:45 2012 From: mario_fs at mgtech.com (Mario G) Date: Wed, 23 May 2012 08:33:45 -0700 Subject: [Freeswitch-users] How to set "keep-alives re-use the TCP connection" ? In-Reply-To: References: <5B3DD049-C7AA-4084-9082-DABF85BA9720@mgtech.com> Message-ID: <65A3D180-3A6F-476D-874C-AA7F202181C7@mgtech.com> For anyone following this or future lookups: The nat-options-ping did not resolve this issue, it is still outstanding. WIll update if/when this is resolved. On May 17, 2012, at 9:27 AM, Mario G wrote: > This is what is confusing, there is no NATing involved... So it still applies to non-natted as well? > Mario G > > On May 16, 2012, at 4:59 PM, Brian Foster wrote: > >> http://wiki.freeswitch.org/wiki/Sofia.conf.xml#nat-options-ping >> >> On May 16, 2012 7:54 PM, "Mario G" wrote: >> OK, does anyone know if this is even possible in FS? >> >> On May 15, 2012, at 5:36 PM, Mario G wrote: >> >> > Using Bria on iPad with a TCP connection, all works except after several hours it no longer registers to FreeSwitch. The Bria support says if this happens set the server PBX to use "keep-alives re-use the TCP connection". I could not find this options in the wiki, only thing close dealt with NAT but FS and the phones are all on local lan. Anyone know how to set this in the user definition? Thanks, >> > Mario G >> > _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > Join Us At ClueCon - Aug 7-9, 2012 >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at 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 at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120523/96e66ba8/attachment.html From krice at freeswitch.org Wed May 23 19:37:32 2012 From: krice at freeswitch.org (Ken Rice) Date: Wed, 23 May 2012 10:37:32 -0500 Subject: [Freeswitch-users] Weekly Conference Call Message-ID: Hey Guys, Don?t Forget today 1PM EST5EDT (that?s 10AM Pacific for you guys on the West Coast of the US) FreeSWITCH community conference call. We?ll be reviewing Dialplanning in C and doing an update on FreeSWITCH 1.2 release cycle. How to join you ask? See http://wiki.freeswitch.org/wiki/Weekly_Conference_Call there are several ways you can join from SIP to PSTN or even Skype. K -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120523/3c559ee7/attachment.html From jerry.richards at teotech.com Wed May 23 21:05:56 2012 From: jerry.richards at teotech.com (Jerry Richards) Date: Wed, 23 May 2012 17:05:56 +0000 Subject: [Freeswitch-users] Channel Variable To Disable Processing of 2833 DTMF Messages? Message-ID: <1545146083A72C4DB7B66584B7E5D98402BD7266@BY2PRD0410MB377.namprd04.prod.outlook.com> Is there a channel variable to disable/ignore 2833 DTMF messages that are going from the internal network toward the PRI? I have a scenario where I am getting both inband and out-of-band DTMF (i.e. double DTMF digits). This happens when a call comes in through the PRI and routes back out the PRI. Inband DTMF coming from the caller is converted to out-of-band 2833 internally, so when the audio is routed back out the PRI, it contains both the inband DTMF and the 2833 conversion. In some cases, we don't want to use disable_dtmf, since we may need to detect inband DTMF in FreeSwitch. Thanks, Jerry -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120523/0b045d68/attachment-0001.html From andy at fabulous4.co.uk Wed May 23 21:36:04 2012 From: andy at fabulous4.co.uk (Andy Ayers) Date: Wed, 23 May 2012 18:36:04 +0100 Subject: [Freeswitch-users] Best database setup for high volume In-Reply-To: References: <005401cd38be$2a9434b0$7fbc9e10$@fabulous4.co.uk> Message-ID: <01ef01cd390a$884b60f0$98e222d0$@fabulous4.co.uk> Many thanks for your reply Avi, that's very helpful. I've disabled the sql scoreboard and presence and all seems to be ok so those should help. My traffic consists of relatively high volumes of incoming and outgoing calls but all one sided. i.e. It's the switch taking the caller through an ivr 'form' so there are no 2-way calls, no bridging, forwarding or directing of calls at all. The only time multiple callers are involved is when we use it for conferencing which is only small scale at the moment. It's the database corruption issue I'm really interested in solving so I'd like to get the odbc connection working if possible. Any info you can provide on how you got that to work would be greatly appreciated. Like I say I hit 2 problems: Firstly on load freeswitch complains that it can't run multiple statements. I've tried everything that's recommended in the MyOdbc docs including setting the options in odbc.ini but still get the error. Some posts talk about needing to use to _r version of the driver but I don't have that on my system. I'm running Debian if that's significant. The second issue was a message popping up in the logs every few seconds saying: Error in my_thread_global_end() nn threads didn't exit. Did you encounter either of these problems or find ways round them? Once again many thanks for any help. Cheers Andy From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi Marcus Sent: 23 May 2012 09:52 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Best database setup for high volume Is your issue the CDRs & Voicemail or session count, current calls, recovery data.. You can make sure track-calls is off... -nosql -- disable internal sql scoreboard I'm not sure if that kills presence or not. Disable presence if you don't need it - it's a real usage hog. I have odbc to mysql but I made calls,channels,sip_dialogs, sip_subscriptions, etc into memory tables a few months ago. (I left sip_registrations as non-memory for persistence of a sort) -Avi On Wed, May 23, 2012 at 11:29 AM, Andy Ayers wrote: Hi, Can anyone tell me what the best database setup is for dealing with high call volumes? The background: I've been running with the standard SQLite system for about 3 years without issue but recently am getting a lot of database corruption errors ('database disk image is malformed'). Easily solved by deleting the db and restarting but on occasion it brings my switch down. I've tried upgrading to odbc and mysql but hit 2 problems: FLAG_MULTIPLE_STATEMENTS Error in my_thread_global_end() Both of which are mentioned in the user groups but not with any solutions that worked. I've just upgraded to version 1.2 so this may solve the corruption problems but would really like to get my system set up to handle as much traffic as possible. Any advice or suggestions much appreciated. Kind regards Andy _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120523/4d0efddc/attachment.html From avi at avimarcus.net Wed May 23 22:09:31 2012 From: avi at avimarcus.net (Avi Marcus) Date: Wed, 23 May 2012 21:09:31 +0300 Subject: [Freeswitch-users] Best database setup for high volume In-Reply-To: <01ef01cd390a$884b60f0$98e222d0$@fabulous4.co.uk> References: <005401cd38be$2a9434b0$7fbc9e10$@fabulous4.co.uk> <01ef01cd390a$884b60f0$98e222d0$@fabulous4.co.uk> Message-ID: On Wed, May 23, 2012 at 8:36 PM, Andy Ayers wrote: > Many thanks for your reply Avi, that?s very helpful.**** > > ** ** > > I?ve disabled the sql scoreboard and presence and all seems to be ok so > those should help.**** > > ** ** > > My traffic consists of relatively high volumes of incoming and outgoing > calls but all one sided. i.e. It?s the switch taking the caller through an > ivr ?form? so there are no 2-way calls, no bridging, forwarding or > directing of calls at all. The only time multiple callers are involved is > when we use it for conferencing which is only small scale at the moment.** > ** > > ** ** > > It?s the database corruption issue I?m really interested in solving so I?d > like to get the odbc connection working if possible. Any info you can > provide on how you got that to work would be greatly appreciated. Like I > say I hit 2 problems:**** > > ** ** > > Firstly on load freeswitch complains that it can?t run multiple > statements. I?ve tried everything that?s recommended in the MyOdbc docs > including setting the options in odbc.ini but still get the error. Some > posts talk about needing to use to _r version of the driver but I don?t > have that on my system. I?m running Debian if that?s significant. > Did you set: OPTIONS = 67108864 in the odbc.ini, for FLAG_MULTI_STATEMENTS? (Supposedly it's "OPTION" on centos) http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core -Avi **** > > The second issue was a message popping up in the logs every few seconds > saying: Error in my_thread_global_end() nn threads didn?t exit.**** > > Did you encounter either of these problems or find ways round them?**** > > > ** > > Once again many thanks for any help.**** > > ** ** > > Cheers**** > > Andy**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Avi Marcus > *Sent:* 23 May 2012 09:52 > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Best database setup for high volume**** > > ** ** > > Is your issue the CDRs & Voicemail or session count, current calls, > recovery data..**** > > You can make sure track-calls is off... **** > > ** ** > > ** ** > > -nosql -- disable internal sql scoreboard**** > > I'm not sure if that kills presence or not.**** > > ** ** > > Disable presence if you don't need it - it's a real usage hog.**** > > ** ** > > I have odbc to mysql but I made calls,channels,sip_dialogs, > sip_subscriptions, etc into memory tables a few months ago.**** > > ** ** > > (I left sip_registrations as non-memory for persistence of a sort)**** > > ** ** > > -Avi**** > > ** ** > > On Wed, May 23, 2012 at 11:29 AM, Andy Ayers wrote: > **** > > Hi,**** > > **** > > Can anyone tell me what the best database setup is for dealing with high > call volumes?**** > > **** > > The background:**** > > **** > > I?ve been running with the standard SQLite system for about 3 years > without issue but recently am getting a lot of database corruption errors > (?database disk image is malformed?). Easily solved by deleting the db and > restarting but on occasion it brings my switch down.**** > > **** > > I?ve tried upgrading to odbc and mysql but hit 2 problems:**** > > **** > > FLAG_MULTIPLE_STATEMENTS**** > > Error in my_thread_global_end()**** > > **** > > Both of which are mentioned in the user groups but not with any solutions > that worked.**** > > **** > > I?ve just upgraded to version 1.2 so this may solve the corruption > problems but would really like to get my system set up to handle as much > traffic as possible.**** > > **** > > Any advice or suggestions much appreciated.**** > > **** > > Kind regards**** > > Andy**** > > **** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120523/dfeb5f48/attachment-0001.html From jerry.richards at teotech.com Wed May 23 22:15:10 2012 From: jerry.richards at teotech.com (Jerry Richards) Date: Wed, 23 May 2012 18:15:10 +0000 Subject: [Freeswitch-users] Best database setup for high volume In-Reply-To: <01ef01cd390a$884b60f0$98e222d0$@fabulous4.co.uk> References: <005401cd38be$2a9434b0$7fbc9e10$@fabulous4.co.uk> <01ef01cd390a$884b60f0$98e222d0$@fabulous4.co.uk> Message-ID: <1545146083A72C4DB7B66584B7E5D98402BD72C6@BY2PRD0410MB377.namprd04.prod.outlook.com> To fix the multiple statements issue, try adding this to the odbc.ini:OPTION = 67108864 From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Andy Ayers Sent: Wednesday, May 23, 2012 10:36 AM To: 'FreeSWITCH Users Help' Subject: Re: [Freeswitch-users] Best database setup for high volume Many thanks for your reply Avi, that's very helpful. I've disabled the sql scoreboard and presence and all seems to be ok so those should help. My traffic consists of relatively high volumes of incoming and outgoing calls but all one sided. i.e. It's the switch taking the caller through an ivr 'form' so there are no 2-way calls, no bridging, forwarding or directing of calls at all. The only time multiple callers are involved is when we use it for conferencing which is only small scale at the moment. It's the database corruption issue I'm really interested in solving so I'd like to get the odbc connection working if possible. Any info you can provide on how you got that to work would be greatly appreciated. Like I say I hit 2 problems: Firstly on load freeswitch complains that it can't run multiple statements. I've tried everything that's recommended in the MyOdbc docs including setting the options in odbc.ini but still get the error. Some posts talk about needing to use to _r version of the driver but I don't have that on my system. I'm running Debian if that's significant. The second issue was a message popping up in the logs every few seconds saying: Error in my_thread_global_end() nn threads didn't exit. Did you encounter either of these problems or find ways round them? Once again many thanks for any help. Cheers Andy From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi Marcus Sent: 23 May 2012 09:52 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Best database setup for high volume Is your issue the CDRs & Voicemail or session count, current calls, recovery data.. You can make sure track-calls is off... -nosql -- disable internal sql scoreboard I'm not sure if that kills presence or not. Disable presence if you don't need it - it's a real usage hog. I have odbc to mysql but I made calls,channels,sip_dialogs, sip_subscriptions, etc into memory tables a few months ago. (I left sip_registrations as non-memory for persistence of a sort) -Avi On Wed, May 23, 2012 at 11:29 AM, Andy Ayers > wrote: Hi, Can anyone tell me what the best database setup is for dealing with high call volumes? The background: I've been running with the standard SQLite system for about 3 years without issue but recently am getting a lot of database corruption errors ('database disk image is malformed'). Easily solved by deleting the db and restarting but on occasion it brings my switch down. I've tried upgrading to odbc and mysql but hit 2 problems: FLAG_MULTIPLE_STATEMENTS Error in my_thread_global_end() Both of which are mentioned in the user groups but not with any solutions that worked. I've just upgraded to version 1.2 so this may solve the corruption problems but would really like to get my system set up to handle as much traffic as possible. Any advice or suggestions much appreciated. Kind regards Andy _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120523/5f2c06db/attachment.html From lists.jj at googlemail.com Wed May 23 17:30:48 2012 From: lists.jj at googlemail.com (Johannes Jakob) Date: Wed, 23 May 2012 15:30:48 +0200 Subject: [Freeswitch-users] sofia profile internal fails to restart since enabling odbc Message-ID: <62464934D26248FCBE72D0E05BCA5249@gmail.com> Hi, I recently switched our PBX to ODBC (mysql) driven database for all things freeswitch. Since then, I can't just sofia profile internal restart without it being deleted: 2012-05-18 11:07:46.275244 [NOTICE] sofia.c:2092 Waiting for worker thread nua_destroy(0x7fb73c0088d0): FATAL: nua_shutdown not completed [?] 2012-05-18 11:07:46.275244 [NOTICE] sofia.c:4665 Started Profile internal [sofia_reg_internal] 2012-05-18 11:07:46.275244 [DEBUG] sofia.c:1861 Creating agent for internal 2012-05-18 11:07:46.275244 [ERR] sofia.c:1940 Error Creating SIP UA for profile: internal How can I debug/trace what's keeping sofia UA from stopping? Thanks for your help! Regards, Johannes freeswitch at internal> version FreeSWITCH Version 1.1.beta1 (git-c41a16d 2012-04-05 14-28-01 -0500) autoload_configs# grep odbc * | grep -v "<\!--" cidlookup.conf.xml: db.conf.xml: easyroute.conf.xml: lcr.conf.xml: spidermonkey.conf.xml: voicemail.conf.xml: (cidlookup, easyroute, lcr and spidermonkey aren't used anyways) "full log": 2012-05-18 11:07:46.275244 [NOTICE] sofia.c:2092 Waiting for worker thread nua_destroy(0x7fb73c0088d0): FATAL: nua_shutdown not completed 2012-05-18 11:07:46.275244 [DEBUG] sofia.c:2141 Write lock internal 2012-05-18 11:07:46.275244 [DEBUG] sofia.c:2152 Write unlock internal 2012-05-18 11:07:46.275244 [DEBUG] sofia.c:3770 user-agent-string [FreeSWITCH] 2012-05-18 11:07:46.275244 [DEBUG] sofia.c:3770 debug [0] 2012-05-18 11:07:46.275244 [DEBUG] sofia.c:3770 sip-trace [no] 2012-05-18 11:07:46.275244 [DEBUG] sofia.c:3770 sip-capture [no] 2012-05-18 11:07:46.275244 [DEBUG] sofia.c:3770 watchdog-enabled [no] 2012-05-18 11:07:46.275244 [DEBUG] sofia.c:3770 watchdog-step-timeout [30000] 2012-05-18 11:07:46.275244 [DEBUG] sofia.c:3770 watchdog-event-timeout [30000] 2012-05-18 11:07:46.275244 [DEBUG] sofia.c:3770 log-auth-failures [true] 2012-05-18 11:07:46.275244 [DEBUG] sofia.c:3770 forward-unsolicited-mwi-notify [false] 2012-05-18 11:07:46.275244 [DEBUG] sofia.c:3770 context [public] 2012-05-18 11:07:46.275244 [DEBUG] sofia.c:3770 rfc2833-pt [101] 2012-05-18 11:07:46.275244 [DEBUG] sofia.c:3770 sip-port [5060] 2012-05-18 11:07:46.275244 [DEBUG] sofia.c:3770 dialplan [XML] 2012-05-18 11:07:46.275244 [DEBUG] sofia.c:3770 dtmf-duration [2000] 2012-05-18 11:07:46.275244 [DEBUG] sofia.c:3770 inbound-codec-prefs [PCMA,PCMU,G7221 at 32000h,G7221 at 16000h,G722,iLBC,GSM,H263,H264] 2012-05-18 11:07:46.275244 [DEBUG] sofia.c:3770 outbound-codec-prefs [PCMA,PCMU,G7221 at 32000h,G7221 at 16000h,G722,iLBC,GSM,H263,H264] 2012-05-18 11:07:46.275244 [DEBUG] sofia.c:3770 rtp-timer-name [soft] 2012-05-18 11:07:46.275244 [DEBUG] sofia.c:3770 rtp-ip [94.186.133.69] 2012-05-18 11:07:46.275244 [DEBUG] sofia.c:3770 sip-ip [94.186.133.69] 2012-05-18 11:07:46.275244 [DEBUG] sofia.c:3770 hold-music [local_stream://moh] 2012-05-18 11:07:46.275244 [DEBUG] sofia.c:3770 apply-nat-acl [nat.auto] 2012-05-18 11:07:46.275244 [DEBUG] sofia.c:3770 apply-inbound-acl [domains] 2012-05-18 11:07:46.275244 [DEBUG] sofia.c:3770 local-network-acl [localnet.auto] 2012-05-18 11:07:46.275244 [DEBUG] sofia.c:3770 record-path [/usr/local/freeswitch/recordings] 2012-05-18 11:07:46.275244 [DEBUG] sofia.c:3770 record-template [${caller_id_number}.${target_domain}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav] 2012-05-18 11:07:46.275244 [DEBUG] sofia.c:3770 manage-presence [true] 2012-05-18 11:07:46.275244 [DEBUG] sofia.c:3770 presence-hosts [,94.186.133.69] 2012-05-18 11:07:46.275244 [DEBUG] sofia.c:3770 presence-privacy [] 2012-05-18 11:07:46.275244 [DEBUG] sofia.c:3770 inbound-codec-negotiation [generous] 2012-05-18 11:07:46.275244 [DEBUG] sofia.c:3770 tls [true] 2012-05-18 11:07:46.275244 [DEBUG] sofia.c:3770 tls-only [false] 2012-05-18 11:07:46.275244 [DEBUG] sofia.c:3770 tls-bind-params [transport=tls] 2012-05-18 11:07:46.275244 [DEBUG] sofia.c:3770 tls-sip-port [5061] 2012-05-18 11:07:46.275244 [DEBUG] sofia.c:3770 tls-cert-dir [/usr/local/freeswitch/conf/ssl] 2012-05-18 11:07:46.275244 [DEBUG] sofia.c:3770 tls-passphrase [] 2012-05-18 11:07:46.275244 [DEBUG] sofia.c:3770 tls-verify-date [true] 2012-05-18 11:07:46.275244 [DEBUG] sofia.c:3770 tls-verify-policy [none] 2012-05-18 11:07:46.275244 [DEBUG] sofia.c:3770 tls-verify-depth [2] 2012-05-18 11:07:46.275244 [DEBUG] sofia.c:3770 tls-verify-in-subjects [] 2012-05-18 11:07:46.275244 [DEBUG] sofia.c:3770 tls-version [tlsv1] 2012-05-18 11:07:46.275244 [DEBUG] sofia.c:3770 nonce-ttl [60] 2012-05-18 11:07:46.275244 [DEBUG] sofia.c:3770 auth-calls [true] 2012-05-18 11:07:46.275244 [DEBUG] sofia.c:3770 inbound-reg-force-matching-username [true] 2012-05-18 11:07:46.275244 [DEBUG] sofia.c:3770 auth-all-packets [false] 2012-05-18 11:07:46.275244 [DEBUG] sofia.c:3770 ext-rtp-ip [auto-nat] 2012-05-18 11:07:46.275244 [DEBUG] sofia.c:3770 ext-sip-ip [auto-nat] 2012-05-18 11:07:46.275244 [DEBUG] sofia.c:3770 rtp-timeout-sec [300] 2012-05-18 11:07:46.275244 [DEBUG] sofia.c:3770 rtp-hold-timeout-sec [1800] 2012-05-18 11:07:46.275244 [DEBUG] sofia.c:3770 challenge-realm [auto_from] 2012-05-18 11:07:46.275244 [DEBUG] sofia.c:3770 t38_passthru [true] 2012-05-18 11:07:46.275244 [DEBUG] sofia.c:3770 t38-passthru [true] 2012-05-18 11:07:46.275244 [DEBUG] sofia.c:3770 sip-force-expires [120] 2012-05-18 11:07:46.275244 [DEBUG] sofia.c:3770 multiple-registrations [true] 2012-05-18 11:07:46.275244 [NOTICE] sofia.c:4665 Started Profile internal [sofia_reg_internal] 2012-05-18 11:07:46.275244 [DEBUG] sofia.c:1861 Creating agent for internal 2012-05-18 11:07:46.275244 [ERR] sofia.c:1940 Error Creating SIP UA for profile: internal From drk at drkngs.net Wed May 23 20:35:40 2012 From: drk at drkngs.net (Dave R. Kompel) Date: Wed, 23 May 2012 09:35:40 -0700 Subject: [Freeswitch-users] FreeSWITCH logs to C# application In-Reply-To: Message-ID: <20120523163540.f7f805e7@mail.tritonwest.net> There is an EventConsumer class already there for you in mod_managed. Just new up one, and then on a dedicated thread, in a loop, just call the EventConsumer.Pop method. There is no need to try to do cross app domain calls to the FreeSWITCH... namespace. --Dave _____ From: coopara [mailto:coopara at go2.pl] To: freeswitch-users at lists.freeswitch.org Sent: Wed, 23 May 2012 02:17:40 -0700 Subject: [Freeswitch-users] FreeSWITCH logs to C# application Hi, I would like to collect all FS logs to my C# application. Is there any possible way to do that? I use FreeSWITCH.EventBinding.Bind function and I'm looking for event_id which is switch_event_types_t.SWITCH_EVENT_LOG in handler but unfortunately it never occurs. Best regards, Coopara _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120523/1d18a7ac/attachment.html From drk at drkngs.net Wed May 23 20:39:13 2012 From: drk at drkngs.net (Dave R. Kompel) Date: Wed, 23 May 2012 09:39:13 -0700 Subject: [Freeswitch-users] =?iso-8859-1?q?How_do_you_split_bridged_calls_?= =?iso-8859-1?q?=3F?= In-Reply-To: Message-ID: <20120523163913.b07558d1@mail.tritonwest.net> Or you could use uuid_transfer, such as "uuid_transfer -both park inline". If you want to use dialplan apps in the transfer just use the "inline" dialplan, and not XML. --Dave _____ From: Avi Marcus [mailto:avi at avimarcus.net] To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org] Sent: Tue, 22 May 2012 23:31:45 -0700 Subject: Re: [Freeswitch-users] How do you split bridged calls ? You can't transfer to a playback. Playback is a dialplan application, not a destination. You can, however, playback to each leg individually with http://wiki.freeswitch.org/wiki/Mod_commands#uuid_broadcast uuid_broadcast [aleg|bleg|both] here it would be: api uuid_broadcast c1061126-a429-11e1-9da3-85575e903927 phrase::AC_0003 aleg for A leg. I think. Please update the wiki to make it more descriptive after you play with it. Of course, this still leaves the calls together. If you want to split them I suppose you really need to transfer each leg TO somewhere. -Avi On Wed, May 23, 2012 at 3:02 AM, Gilbert Amar wrote: Hello I am trying to split bridged calls. Using mod_event_socket I try this api uuid_transfer c1061126-a429-11e1-9da3-85575e903927 -both playback::phrase:AC_0003: It does not work as FS seems to really need an extension. the other option I could try is : api uuid_transfer c1061126-a429-11e1-9da3-85575e903927 -both park and then for each leg: sendmsg call-command: execute execute-app-name: playback execute-app-arg: phrase:AC_0003: Is there a better way ? I am using FreeSWITCH Version 1.0.head (git-fdaa155 2012-02-10 13-17-54 +0000) Thank you Gilbert _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120523/6d07469e/attachment.html From ahe.sanath at gmail.com Thu May 24 10:00:03 2012 From: ahe.sanath at gmail.com (Sanath Prasanna) Date: Thu, 24 May 2012 11:30:03 +0530 Subject: [Freeswitch-users] Run LUA script in different server In-Reply-To: References: Message-ID: Hi all, Tx a lot for advice. Problem is port. Now I sorted it & call coming to B server. In here diverted call coming to server A. So I used variable_sip_history_info parameter to extarct real B number.(Real called party number) . But in server B, that parameter is not coming. (variable_sip_history_info) Pls help to solve that. Br, Sanath > On Tue, May 22, 2012 at 12:53 AM, Michael Collins wrote: > >> Look at line #186 of your trace: >> 2012-05-21 07:56:49.735660 [ERR] mod_sofia.c:3957 Invalid Profile >> >> You need to figure out why your internal profile isn't running. Try >> "sofia profile internal restart" and see what happens. >> >> -MC >> >> >> On Sun, May 20, 2012 at 10:45 PM, Sanath Prasanna wrote: >> >>> Hi MC, >>> I did the change according to ure instruction. But error is coming. Here >>> I attached freeswitch.log file >>> >>> I change the confs as follows in BOX A. (Operator connected Freeswitch >>> box) >>> BOX B ip is 10.22.29.253 >>> >>> vi /usr/local/freeswitch/conf/dialplan/default.xml >>> >>> >>> >>> >>> >>> >>> >>> Also add following to ACL file in BOX B >>> >>> Pls advice to solve the problem here. >>> Br, >>> Sanath >>> >>> >>> On Fri, May 18, 2012 at 9:57 AM, Michael Collins wrote: >>> >>>> If I understand your question correctly, yes you can do this. You can >>>> send calls from one FreeSWITCH server to another. Start here: >>>> http://wiki.freeswitch.org/wiki/Connect_Two_FreeSWITCH_Boxes >>>> >>>> Best way to learn is to get the FreeSWITCH books from Packt Publishing >>>> and just start hacking code. >>>> >>>> -MC >>>> >>>> >>>> On Thu, May 17, 2012 at 9:07 PM, Sanath Prasanna wrote: >>>> >>>>> Tx for advice MC & Anita. Can I do work around like this . >>>>> Another freeswitch instant will be start in other server & calls will >>>>> be transfer from operator connected freeswitch instance to this new >>>>> freeswitch instance & vise versa. Pls advice. >>>>> >>>>> >>>>> On Thu, May 17, 2012 at 5:05 PM, Anita Hall wrote: >>>>> >>>>>> You could run a Lua ESL server on a different machine but this will >>>>>> not be the same as running a Lua script. >>>>>> http://wiki.freeswitch.org/wiki/Event_Socket_Library >>>>>> >>>>>> regards, >>>>>> Anita >>>>>> >>>>>> >>>>>> >>>>>> On Thu, May 17, 2012 at 4:37 AM, Michael Collins wrote: >>>>>> >>>>>>> I don't think you can directly do what you are describing. However, >>>>>>> you might be able to use mod_httapi for this. There's some documentation on >>>>>>> the wiki and in the module. Keep in mind that this is a relatively new >>>>>>> module so we don't have lots of examples yet, so you'll probably be doing a >>>>>>> fair amount of research and testing. >>>>>>> >>>>>>> -MC >>>>>>> >>>>>>> >>>>>>> On Wed, May 16, 2012 at 5:59 AM, Sanath Prasanna < >>>>>>> ahe.sanath at gmail.com> wrote: >>>>>>> >>>>>>>> Hi all, >>>>>>>> I have 2 servers. One server has SIP GW connection From Operator & >>>>>>>> IVR applications need to build in other server. How to call distributed LUA >>>>>>>> applications with Mysql Databases from the SIP GW server ? Pls advice. >>>>>>>> Main idea is, maintaining SIP connection in one server & all the IVR >>>>>>>> applications in other server. >>>>>>>> Br, >>>>>>>> Sanath >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120524/7ca10b57/attachment-0001.html From anita.hall at simmortel.com Thu May 24 12:07:10 2012 From: anita.hall at simmortel.com (Anita Hall) Date: Thu, 24 May 2012 13:37:10 +0530 Subject: [Freeswitch-users] Spandsp mulated modems inbound/no /dev/FS* device? In-Reply-To: <4FBAE7AE.1000606@coppice.org> References: <4FBAA1CF.5060907@integrafin.co.uk> <4FBAE7AE.1000606@coppice.org> Message-ID: Many thanks to everybody! I will put this on wiki but I am stuck at the very beginning. I am sorry for being such a noob but this is my first tryst with modems. I have modems appearing as /dev/FS[0-4] which link to /dev/pts/[4-7] # ls -l /dev/FS? lrwxrwxrwx 1 root root 10 2012-05-17 17:49 /dev/FS0 -> /dev/pts/4 lrwxrwxrwx 1 root root 10 2012-05-17 17:49 /dev/FS1 -> /dev/pts/5 lrwxrwxrwx 1 root root 10 2012-05-17 17:49 /dev/FS2 -> /dev/pts/6 lrwxrwxrwx 1 root root 10 2012-05-17 17:49 /dev/FS3 -> /dev/pts/7 lrwxrwxrwx 1 root root 10 2012-05-17 17:49 /dev/FS4 -> /dev/pts/8 But, # cu -l /dev/FS0 cu: open (/dev/FS0): Permission denied cu: /dev/FS0: Line in use I think this is because FreeSWITCH is using this device, but then how does HylaFax or any other program like cu talk to it ? This page says that I should be able to check my modem before I configure it with Hylafax. http://www.hylafax.org/content/Handbook:Basic_Server_Configuration:Checking_your_Modem regards, Anita On Tue, May 22, 2012 at 6:41 AM, Steve Underwood wrote: > What all this should tell us is a wiki page is badly needed on this > topic. Specifically, people need clear information about: > - How the modems appear to the user inside FS > - How the modems appear to the user in /dev > - How the permissions of the modems in /dev can be controlled, so > non-root users are OK > I still haven't seen anyone give information on the last point. Its easy > to set up udev rules to control the generation of the pts devices in > /dev, but I am not clear how to control the permissions during the > creation of the links in /dev which give these devices meaningful names. > > Steve > > > On 05/22/2012 07:28 AM, Michael Collins wrote: > > I looked at the source, just to confirm and I believe Anthony is correct. > > > > Line 871 of mod_spandsp_modem.c: > > switch_snprintf(name, sizeof(name), "modem/%d/%s", modem->slot, number); > > > > That would suggest a syntax just like FreeTDM, where the dialstring is > > "modem/x/y" where x is the "slot" and y is the dialed number. Also, I > > suspect this block starting at line 856 means you can use a literal > > "a" for the slot and it will hunt for the next available modem: > > > > if (!strcasecmp(modem_id_string, "a")) { > > modem_id = -1; > > } else { > > modem_id = atoi(modem_id_string); > > } > > > > Please try this out and confirm. If possible, add the information to > > the wiki. If you can't update the wiki then please report back and let > > us know whether the dialstring format worked or not. > > > > Thanks, > > MC > > > > On Mon, May 21, 2012 at 2:23 PM, Anthony Minessale > > > > wrote: > > > > IIRC its modem/1 or 2 3 4 etc. Probably a for auto like tdm. > > Check the code to be sure.... > > > > On May 21, 2012 3:13 PM, "Alex Crow" > > wrote: > > > > On 21/05/12 20:28, Ken Rice wrote: > > > > > > > > > On 5/21/12 2:15 PM, "Alex Crow" > > wrote: > > >> Also I have the problem that unless FreeSWITCH is run as > > root, the > > >> device nodes are not created. > > >> > > >> Unless in the last few weeks some docs have been updated on > > this, I'm > > >> still stuck with T38modem. > > >> > > > If freeswitch running as root and not running as root > > doesn't work, this > > > should tell you that you have a permissions issue... The > > FreeSWITCH > > > developers fixing your systems perms issues is beyond the > > scope of the > > > FreeSWITCH project... > > > > > > Otherwise, the FreeSWITCH modems basically work just like > > t38modem... The > > > only setting is how many of them to create then the only > > other settings is > > > configuring your platform to properly allow freeswitch (or > > the user that > > > freeswitch is running as) create those devices then > > configure hylafax to use > > > them as normal. > > > > > > The steps to do this can vary from platform to platform (or > > even versions of > > > the platform. For example how you do this on centos5 is not > > how you do this > > > on centos6 due to changes in the platform.) > > > > > > K > > > > > > > > > > Ken, > > > > I respectfully disagree with the "like t38modem" bit of this. > With > > t38modem, you create a set of t38modem devices which actually > > register > > against FreeSWITCH and therefore appear as registered > > endpoints, so it's > > easy to route calls to them (ie transfer to "t38modem0 default > > XML"). > > When I did run FS as root, despite the device nodes being > > created, I saw > > no endpoint registrations for the softmodems, nor did I find any > > documentation saying that these endpoints may be addressed as, > for > > instance "spandsp/FSx". > > > > I am also used to writing udev rules, but I can't find > > anything that > > allows a non-root uid/gid user to create device nodes for a > device > > provided by a userspace program. > > > > It'd be great to avoid T38modem, and I've asked these > > questions before, > > namely: > > > > 1) Does anyone have any examples on any platform of tweaking > > udev to > > allow FreeSWITCH to create device nodes at startup when > > running as, say, > > user freeswitch or user www-data (without making the primary > > group of > > said users "root" or setting setuid root on the executable). I > > am not > > prepared to take the risk of running as either effective uid > > or gid 0. > > Please share if so. I'm running on debian stable. > > > > 2) How does one direct an incoming call, say, from FreeTDM or > > a SIP/ISDN > > gateway to one of the softmodems. I cannot find this in the docs. > > > > If we can divorce these questions from any association with > > another > > user's questions on this list that's fine. I've tried to help > that > > person as much as I can out of my own good nature. > > > > Cheers > > > > Alex > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120524/744d4ad4/attachment.html From andy at fabulous4.co.uk Thu May 24 13:31:50 2012 From: andy at fabulous4.co.uk (Andy Ayers) Date: Thu, 24 May 2012 10:31:50 +0100 Subject: [Freeswitch-users] Best database setup for high volume In-Reply-To: References: <005401cd38be$2a9434b0$7fbc9e10$@fabulous4.co.uk> <01ef01cd390a$884b60f0$98e222d0$@fabulous4.co.uk> Message-ID: <008e01cd3990$0cefac00$26cf0400$@fabulous4.co.uk> Hi Avi, Yes sadly I've been through all those options and am still getting the error. Tried OPTION and OPTIONS but it doesn't seem to make any difference. There seem to be a few other folks on the users list that have had the same problem and not managed to find a solution. Cheers Andy From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi Marcus Sent: 23 May 2012 19:10 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Best database setup for high volume On Wed, May 23, 2012 at 8:36 PM, Andy Ayers wrote: Many thanks for your reply Avi, that's very helpful. I've disabled the sql scoreboard and presence and all seems to be ok so those should help. My traffic consists of relatively high volumes of incoming and outgoing calls but all one sided. i.e. It's the switch taking the caller through an ivr 'form' so there are no 2-way calls, no bridging, forwarding or directing of calls at all. The only time multiple callers are involved is when we use it for conferencing which is only small scale at the moment. It's the database corruption issue I'm really interested in solving so I'd like to get the odbc connection working if possible. Any info you can provide on how you got that to work would be greatly appreciated. Like I say I hit 2 problems: Firstly on load freeswitch complains that it can't run multiple statements. I've tried everything that's recommended in the MyOdbc docs including setting the options in odbc.ini but still get the error. Some posts talk about needing to use to _r version of the driver but I don't have that on my system. I'm running Debian if that's significant. Did you set: OPTIONS = 67108864 in the odbc.ini, for FLAG_MULTI_STATEMENTS? (Supposedly it's "OPTION" on centos) http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core -Avi The second issue was a message popping up in the logs every few seconds saying: Error in my_thread_global_end() nn threads didn't exit. Did you encounter either of these problems or find ways round them? Once again many thanks for any help. Cheers Andy From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi Marcus Sent: 23 May 2012 09:52 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Best database setup for high volume Is your issue the CDRs & Voicemail or session count, current calls, recovery data.. You can make sure track-calls is off... -nosql -- disable internal sql scoreboard I'm not sure if that kills presence or not. Disable presence if you don't need it - it's a real usage hog. I have odbc to mysql but I made calls,channels,sip_dialogs, sip_subscriptions, etc into memory tables a few months ago. (I left sip_registrations as non-memory for persistence of a sort) -Avi On Wed, May 23, 2012 at 11:29 AM, Andy Ayers wrote: Hi, Can anyone tell me what the best database setup is for dealing with high call volumes? The background: I've been running with the standard SQLite system for about 3 years without issue but recently am getting a lot of database corruption errors ('database disk image is malformed'). Easily solved by deleting the db and restarting but on occasion it brings my switch down. I've tried upgrading to odbc and mysql but hit 2 problems: FLAG_MULTIPLE_STATEMENTS Error in my_thread_global_end() Both of which are mentioned in the user groups but not with any solutions that worked. I've just upgraded to version 1.2 so this may solve the corruption problems but would really like to get my system set up to handle as much traffic as possible. Any advice or suggestions much appreciated. Kind regards Andy _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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 at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120524/7f82a569/attachment-0001.html From acrow at integrafin.co.uk Thu May 24 14:32:42 2012 From: acrow at integrafin.co.uk (Alex Crow) Date: Thu, 24 May 2012 11:32:42 +0100 Subject: [Freeswitch-users] Spandsp mulated modems inbound/no /dev/FS* device? In-Reply-To: References: <4FBAA1CF.5060907@integrafin.co.uk> <4FBAE7AE.1000606@coppice.org> Message-ID: <4FBE0E4A.6090408@integrafin.co.uk> On 24/05/12 09:07, Anita Hall wrote: > Many thanks to everybody! > > I will put this on wiki but I am stuck at the very beginning. I am > sorry for being such a noob but this is my first tryst with modems. > > I have modems appearing as /dev/FS[0-4] which link to /dev/pts/[4-7] > > # ls -l /dev/FS? > lrwxrwxrwx 1 root root 10 2012-05-17 17:49 /dev/FS0 -> /dev/pts/4 > lrwxrwxrwx 1 root root 10 2012-05-17 17:49 /dev/FS1 -> /dev/pts/5 > lrwxrwxrwx 1 root root 10 2012-05-17 17:49 /dev/FS2 -> /dev/pts/6 > lrwxrwxrwx 1 root root 10 2012-05-17 17:49 /dev/FS3 -> /dev/pts/7 > lrwxrwxrwx 1 root root 10 2012-05-17 17:49 /dev/FS4 -> /dev/pts/8 > > But, > # cu -l /dev/FS0 > cu: open (/dev/FS0): Permission denied > cu: /dev/FS0: Line in use > > I think this is because FreeSWITCH is using this device, but then how > does HylaFax or any other program like cu talk to it ? > > This page says that I should be able to check my modem before I > configure it with Hylafax. > http://www.hylafax.org/content/Handbook:Basic_Server_Configuration:Checking_your_Modem > > > regards, > Anita > > The permissions on the links are not relevant. It's the permissions on the /dev/pts/* devices that take effect. Alex -- This message is intended only for the addressee and may contain confidential information. Unless you are that person, you may not disclose its contents or use it in any way and are requested to delete the message along with any attachments and notify us immediately. "Transact" is operated by Integrated Financial Arrangements plc Domain House, 5-7 Singer Street, London EC2A 4BQ Tel: (020) 7608 4900 Fax: (020) 7608 5300 (Registered office: as above; Registered in England and Wales under number: 3727592) Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856) From wesleyakio at tuntscorp.com Thu May 24 15:56:36 2012 From: wesleyakio at tuntscorp.com (Wesley Akio) Date: Thu, 24 May 2012 08:56:36 -0300 Subject: [Freeswitch-users] Best database setup for high volume In-Reply-To: <008e01cd3990$0cefac00$26cf0400$@fabulous4.co.uk> References: <005401cd38be$2a9434b0$7fbc9e10$@fabulous4.co.uk> <01ef01cd390a$884b60f0$98e222d0$@fabulous4.co.uk> <008e01cd3990$0cefac00$26cf0400$@fabulous4.co.uk> Message-ID: Sent from mobile, sorry for the typos.... Em 24/05/2012 06:33, "Andy Ayers" escreveu: > Hi Avi,**** > > ** ** > > Yes sadly I?ve been through all those options and am still getting the > error. Tried OPTION and OPTIONS but it doesn?t seem to make any difference. > There seem to be a few other folks on the users list that have had the same > problem and not managed to find a solution.**** > > ** ** > > Cheers**** > > Andy**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Avi Marcus > *Sent:* 23 May 2012 19:10 > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Best database setup for high volume**** > > ** ** > > On Wed, May 23, 2012 at 8:36 PM, Andy Ayers wrote:* > *** > > Many thanks for your reply Avi, that?s very helpful.**** > > **** > > I?ve disabled the sql scoreboard and presence and all seems to be ok so > those should help.**** > > **** > > My traffic consists of relatively high volumes of incoming and outgoing > calls but all one sided. i.e. It?s the switch taking the caller through an > ivr ?form? so there are no 2-way calls, no bridging, forwarding or > directing of calls at all. The only time multiple callers are involved is > when we use it for conferencing which is only small scale at the moment.** > ** > > **** > > It?s the database corruption issue I?m really interested in solving so I?d > like to get the odbc connection working if possible. Any info you can > provide on how you got that to work would be greatly appreciated. Like I > say I hit 2 problems:**** > > **** > > Firstly on load freeswitch complains that it can?t run multiple > statements. I?ve tried everything that?s recommended in the MyOdbc docs > including setting the options in odbc.ini but still get the error. Some > posts talk about needing to use to _r version of the driver but I don?t > have that on my system. I?m running Debian if that?s significant.**** > > Did you set:**** > > OPTIONS = 67108864**** > > in the odbc.ini, for FLAG_MULTI_STATEMENTS? (Supposedly it's "OPTION" on > centos)**** > > http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core**** > > ** ** > > -Avi**** > > ** ** > > The second issue was a message popping up in the logs every few seconds > saying: Error in my_thread_global_end() nn threads didn?t exit.**** > > Did you encounter either of these problems or find ways round them?**** > > **** > > Once again many thanks for any help.**** > > **** > > Cheers**** > > Andy**** > > **** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Avi Marcus > *Sent:* 23 May 2012 09:52 > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Best database setup for high volume**** > > **** > > Is your issue the CDRs & Voicemail or session count, current calls, > recovery data..**** > > You can make sure track-calls is off... **** > > **** > > ** ** > > ** ** > > **** > > -nosql -- disable internal sql scoreboard**** > > I'm not sure if that kills presence or not.**** > > **** > > Disable presence if you don't need it - it's a real usage hog.**** > > **** > > I have odbc to mysql but I made calls,channels,sip_dialogs, > sip_subscriptions, etc into memory tables a few months ago.**** > > **** > > (I left sip_registrations as non-memory for persistence of a sort)**** > > **** > > -Avi**** > > **** > > On Wed, May 23, 2012 at 11:29 AM, Andy Ayers wrote: > **** > > Hi,**** > > **** > > Can anyone tell me what the best database setup is for dealing with high > call volumes?**** > > **** > > The background:**** > > **** > > I?ve been running with the standard SQLite system for about 3 years > without issue but recently am getting a lot of database corruption errors > (?database disk image is malformed?). Easily solved by deleting the db and > restarting but on occasion it brings my switch down.**** > > **** > > I?ve tried upgrading to odbc and mysql but hit 2 problems:**** > > **** > > FLAG_MULTIPLE_STATEMENTS**** > > Error in my_thread_global_end()**** > > **** > > Both of which are mentioned in the user groups but not with any solutions > that worked.**** > > **** > > I?ve just upgraded to version 1.2 so this may solve the corruption > problems but would really like to get my system set up to handle as much > traffic as possible.**** > > **** > > Any advice or suggestions much appreciated.**** > > **** > > Kind regards**** > > Andy**** > > **** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120524/6bebed37/attachment-0001.html From wesleyakio at tuntscorp.com Thu May 24 15:58:02 2012 From: wesleyakio at tuntscorp.com (Wesley Akio) Date: Thu, 24 May 2012 08:58:02 -0300 Subject: [Freeswitch-users] Best database setup for high volume In-Reply-To: <008e01cd3990$0cefac00$26cf0400$@fabulous4.co.uk> References: <005401cd38be$2a9434b0$7fbc9e10$@fabulous4.co.uk> <01ef01cd390a$884b60f0$98e222d0$@fabulous4.co.uk> <008e01cd3990$0cefac00$26cf0400$@fabulous4.co.uk> Message-ID: Out of curiosity, do you run sqlite from memory or disk? Sent from mobile, sorry for the typos.... Em 24/05/2012 06:33, "Andy Ayers" escreveu: > Hi Avi,**** > > ** ** > > Yes sadly I?ve been through all those options and am still getting the > error. Tried OPTION and OPTIONS but it doesn?t seem to make any difference. > There seem to be a few other folks on the users list that have had the same > problem and not managed to find a solution.**** > > ** ** > > Cheers**** > > Andy**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Avi Marcus > *Sent:* 23 May 2012 19:10 > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Best database setup for high volume**** > > ** ** > > On Wed, May 23, 2012 at 8:36 PM, Andy Ayers wrote:* > *** > > Many thanks for your reply Avi, that?s very helpful.**** > > **** > > I?ve disabled the sql scoreboard and presence and all seems to be ok so > those should help.**** > > **** > > My traffic consists of relatively high volumes of incoming and outgoing > calls but all one sided. i.e. It?s the switch taking the caller through an > ivr ?form? so there are no 2-way calls, no bridging, forwarding or > directing of calls at all. The only time multiple callers are involved is > when we use it for conferencing which is only small scale at the moment.** > ** > > **** > > It?s the database corruption issue I?m really interested in solving so I?d > like to get the odbc connection working if possible. Any info you can > provide on how you got that to work would be greatly appreciated. Like I > say I hit 2 problems:**** > > **** > > Firstly on load freeswitch complains that it can?t run multiple > statements. I?ve tried everything that?s recommended in the MyOdbc docs > including setting the options in odbc.ini but still get the error. Some > posts talk about needing to use to _r version of the driver but I don?t > have that on my system. I?m running Debian if that?s significant.**** > > Did you set:**** > > OPTIONS = 67108864**** > > in the odbc.ini, for FLAG_MULTI_STATEMENTS? (Supposedly it's "OPTION" on > centos)**** > > http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core**** > > ** ** > > -Avi**** > > ** ** > > The second issue was a message popping up in the logs every few seconds > saying: Error in my_thread_global_end() nn threads didn?t exit.**** > > Did you encounter either of these problems or find ways round them?**** > > **** > > Once again many thanks for any help.**** > > **** > > Cheers**** > > Andy**** > > **** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Avi Marcus > *Sent:* 23 May 2012 09:52 > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Best database setup for high volume**** > > **** > > Is your issue the CDRs & Voicemail or session count, current calls, > recovery data..**** > > You can make sure track-calls is off... **** > > **** > > ** ** > > ** ** > > **** > > -nosql -- disable internal sql scoreboard**** > > I'm not sure if that kills presence or not.**** > > **** > > Disable presence if you don't need it - it's a real usage hog.**** > > **** > > I have odbc to mysql but I made calls,channels,sip_dialogs, > sip_subscriptions, etc into memory tables a few months ago.**** > > **** > > (I left sip_registrations as non-memory for persistence of a sort)**** > > **** > > -Avi**** > > **** > > On Wed, May 23, 2012 at 11:29 AM, Andy Ayers wrote: > **** > > Hi,**** > > **** > > Can anyone tell me what the best database setup is for dealing with high > call volumes?**** > > **** > > The background:**** > > **** > > I?ve been running with the standard SQLite system for about 3 years > without issue but recently am getting a lot of database corruption errors > (?database disk image is malformed?). Easily solved by deleting the db and > restarting but on occasion it brings my switch down.**** > > **** > > I?ve tried upgrading to odbc and mysql but hit 2 problems:**** > > **** > > FLAG_MULTIPLE_STATEMENTS**** > > Error in my_thread_global_end()**** > > **** > > Both of which are mentioned in the user groups but not with any solutions > that worked.**** > > **** > > I?ve just upgraded to version 1.2 so this may solve the corruption > problems but would really like to get my system set up to handle as much > traffic as possible.**** > > **** > > Any advice or suggestions much appreciated.**** > > **** > > Kind regards**** > > Andy**** > > **** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120524/a24c64ff/attachment.html From andrew at cassidywebservices.co.uk Thu May 24 19:51:58 2012 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Thu, 24 May 2012 16:51:58 +0100 Subject: [Freeswitch-users] mod_callcenter making no attempt to contact agents Message-ID: Here's some pastebin links from earlier on IRC: Cli output and queue config: http://pastebin.freeswitch.org/19154 Call log output: http://pastebin.freeswitch.org/19156 So, here's the thing, queue configured, agents and tiers are in the database only (eventually going to be shared). Set up an extension, dial it, get put in queue, hear MoH. However, there's no sign of any attempt to contact the agent, even though it's registered and set as Available. No outgoing sip packets, nothing logged. I've literally just trid a make current, makes no difference. Attached a pcap from server side, showing no invites to the agent. Any ideas folks? -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T *03300 100 960 *F *03300 100 961 *E *andrew at cassidywebservices.co.uk *W *www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120524/d2d761d7/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: queue.pcap Type: application/octet-stream Size: 15137 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120524/d2d761d7/attachment-0001.obj From sherifomran2000 at yahoo.com Thu May 24 19:53:51 2012 From: sherifomran2000 at yahoo.com (Sherif Omran) Date: Thu, 24 May 2012 08:53:51 -0700 (PDT) Subject: [Freeswitch-users] Freeswitch (SBC + Media) does not know that gateway phone is picked up In-Reply-To: Message-ID: <1337874831.37385.YahooMailClassic@web110810.mail.gq1.yahoo.com> Hi all, My topology is as follows: Kamailio? -> FS (SBS+Media server) I came across an issue with my system as follows.? I have a Hardphone registered. When I do local call inside kamailio, it gets to FS and returns back well and FS understands when I lift the handset. However, I added a gateway (german landline server), when I call my self from another phone, the call gets to FS and then transmits to Kamailio, it rings my extension but when I lift the handset FS does not notice it and keeps ringing. Any body has an Idea? Here is my gateway trunk. ??????? ??????? ??????? ??????? ??????? ????? thanks in advance Sherif Omran -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120524/2e38f92d/attachment.html From kris at kriskinc.com Thu May 24 20:51:51 2012 From: kris at kriskinc.com (Kristian Kielhofner) Date: Thu, 24 May 2012 12:51:51 -0400 Subject: [Freeswitch-users] Freeswitch (SBC + Media) does not know that gateway phone is picked up In-Reply-To: <1337874831.37385.YahooMailClassic@web110810.mail.gq1.yahoo.com> References: <1337874831.37385.YahooMailClassic@web110810.mail.gq1.yahoo.com> Message-ID: Siptrace and logs please. On Thu, May 24, 2012 at 11:53 AM, Sherif Omran wrote: > > Hi all, > > My topology is as follows: > > Kamailio? -> FS (SBS+Media server) > > I came across an issue with my system as follows.? I have a Hardphone registered. When I do local call inside kamailio, it gets to FS and returns back well and FS understands when I lift the handset. However, I added a gateway (german landline server), when I call my self from another phone, the call gets to FS and then transmits to Kamailio, it rings my extension but when I lift the handset FS does not notice it and keeps ringing. > > Any body has an Idea? Here is my gateway trunk. > > > ??????? > ??????? > ??????? > ??????? > ??????? > ????? > > > thanks in advance > Sherif Omran > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Kristian Kielhofner From rnbrady at gmail.com Thu May 24 14:26:27 2012 From: rnbrady at gmail.com (Richard Brady) Date: Thu, 24 May 2012 11:26:27 +0100 Subject: [Freeswitch-users] Can an inbound call be associated with a gateway? Message-ID: Hi folks I'd like to know whether FreeSWITCH is able to associate an inbound call with a gateway. On the one hand I see documentation of variables which can be added to a gateway with the direction="inbound" attribute, which makes me think it's possible. On the other hand, I cannot see how FreeSWITCH binds an incoming call which is allowed by an ACL and therefore accepted but not challenged (so it is IP authenticated) to a gateway. If the binding is by IP address, where in the configuration is the association of the address or CIDR range to the gateway? I get a feeling that gateways are an outbound only concept, but the documentation seems to contradict this. Any clarification would be most helpful. Thanks, Richard -- Richard Brady -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120524/516c599b/attachment.html From paul at cupis.co.uk Thu May 24 23:14:38 2012 From: paul at cupis.co.uk (Paul Cupis) Date: Thu, 24 May 2012 20:14:38 +0100 Subject: [Freeswitch-users] FS as a dumb SBC (was: Re: Help on Explaination of the configurations parrametters) Message-ID: <4FBE889E.6000406@cupis.co.uk> On 23/05/12 09:26, Albert Nguyen wrote: > I am new to FS and trying to setup FS as a "dumb" SBC as per > example 2 in the wiki website so that I can convert inband DTMF to > RFC 8233 and vice versa. This is the windows version of FS. Is it > possible that I can get some assistant from you guys to get it up and > running? Please advise how we can proceed. One way of doing this would be to setup two SIP profiles, one facing the device sending you RFC2833 and once facing the device which will only accept in-band. On the SIP profile facing the in-band device, disable RFC2833 support by setting the varaible "rfc2833-pt" to something <95, and then when you route a call from RFC2833 to in-band, add something like: to your dialplan. Similarly you will want some config for calls going in-band->rfc2833 to do the conversion, perhaps something like: Note that I've not specifically tested the above. If you are going down the above route, you might also want to read: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf_generate http://wiki.freeswitch.org/wiki/DTMF Regards, From sherifomran2000 at yahoo.com Fri May 25 03:40:17 2012 From: sherifomran2000 at yahoo.com (Sherif Omran) Date: Thu, 24 May 2012 16:40:17 -0700 (PDT) Subject: [Freeswitch-users] Freeswitch (SBC + Media) does not know that gateway phone is picked up In-Reply-To: Message-ID: <1337902817.51217.YahooMailClassic@web110816.mail.gq1.yahoo.com> Hi all, here is the siptrace: To figure it out 1- gateway called bluesip.net. It send invide using caller number at bluesip.net 2- This call should go to extension kb-1002. kb means go from freeswitch port 6090 to kamailio port 5060 3- It should go to call extension 1002 in Kamailio 4- Extension 1002 rings but when I reply, it does not notice I replied ./fs_cli ??????????? _____ ____???? ____ _???? ___????????????? ?????????? |? ___/ ___|?? / ___| |?? |_ _|???????????? ?????????? | |_? \___ \? | |?? | |??? | |??????????? ?????????? |? _|? ___) | | |___| |___ | |????????????? ?????????? |_|?? |____/?? \____|_____|___|??????????? ******************************************************* * Anthony Minessale II, Ken Rice,???????????????????? * * Michael Jerris, Travis Cross??????????????????????? * * FreeSWITCH (http://www.freeswitch.org)????????????? * * Paypal Donations Appreciated: paypal at freeswitch.org * * Brought to you by ClueCon http://www.cluecon.com/?? * ******************************************************* Type /help to see a list of commands +OK log level? [7] freeswitch at internal> tracelevel -ERR tracelevel Command not found! freeswitch at internal> sofia global siptrace on +OK Global siptrace on recv 1104 bytes from udp/[217.74.179.29]:5060 at 23:08:45.822962: ?? ------------------------------------------------------------------------ ?? INVITE sip:gw+trunk_1000 at 78.138.90.58:6090;transport=udp;gw=trunk_1000 SIP/2.0 ?? Record-Route: ?? Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 ?? Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 ?? From: "+41793940965" ;tag=as00589402 ?? To: ?? Contact: ?? Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net ?? CSeq: 102 INVITE ?? User-Agent: blueSIP PSTN GW ?? Max-Forwards: 69 ?? Date: Thu, 24 May 2012 23:08:44 GMT ?? Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY ?? Supported: replaces ?? Content-Type: application/sdp ?? Content-Length: 367 ?? P-hint: USRLOC ?? ?? v=0 ?? o=root 20076 20076 IN IP4 217.74.179.28 ?? s=session ?? c=IN IP4 217.74.179.28 ?? t=0 0 ?? m=audio 25626 RTP/AVP 8 0 18 111 3 101 ?? a=rtpmap:8 PCMA/8000 ?? a=rtpmap:0 PCMU/8000 ?? a=rtpmap:18 G729/8000 ?? a=fmtp:18 annexb=no ?? a=rtpmap:111 G726-32/8000 ?? a=rtpmap:3 GSM/8000 ?? a=rtpmap:101 telephone-event/8000 ?? a=fmtp:101 0-16 ?? a=silenceSupp:off - - - - ?? a=ptime:20 ?? a=sendrecv ?? ------------------------------------------------------------------------ send 494 bytes to udp/[217.74.179.29]:5060 at 23:08:45.844486: ?? ------------------------------------------------------------------------ ?? SIP/2.0 100 Trying ?? Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 ?? Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 ?? Record-Route: ?? From: "+41793940965" ;tag=as00589402 ?? To: ?? Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net ?? CSeq: 102 INVITE ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ 2012-05-24 23:08:45.840749 [NOTICE] switch_channel.c:926 New Channel sofia/internal/+41793940965 at bluesip.net [69cd22e6-a5f5-11e1-a833-7b1f73a7ffcf] 2012-05-24 23:08:45.840749 [DEBUG] sofia.c:5532 Channel sofia/internal/+41793940965 at bluesip.net entering state [received][100] 2012-05-24 23:08:45.840749 [DEBUG] sofia.c:5543 Remote SDP: v=0 o=root 20076 20076 IN IP4 217.74.179.28 s=session c=IN IP4 217.74.179.28 t=0 0 m=audio 25626 RTP/AVP 8 0 18 111 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:111 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMA:8:8000:20:64000]/[iLBC:97:8000:30:13330] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMU:0:8000:20:64000]/[iLBC:97:8000:30:13330] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G729:18:8000:20:8000]/[iLBC:97:8000:30:13330] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G726-32:111:8000:20:0]/[iLBC:97:8000:30:13330] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [GSM:3:8000:20:13200]/[iLBC:97:8000:30:13330] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [telephone-event:101:8000:20:0]/[iLBC:97:8000:30:13330] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf send/recv payload to 101 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMA:8:8000:20:64000]/[G7221:115:32000:20:48000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMU:0:8000:20:64000]/[G7221:115:32000:20:48000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G729:18:8000:20:8000]/[G7221:115:32000:20:48000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G726-32:111:8000:20:0]/[G7221:115:32000:20:48000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [GSM:3:8000:20:13200]/[G7221:115:32000:20:48000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [telephone-event:101:8000:20:0]/[G7221:115:32000:20:48000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf send/recv payload to 101 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMA:8:8000:20:64000]/[G7221:107:16000:20:32000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMU:0:8000:20:64000]/[G7221:107:16000:20:32000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G729:18:8000:20:8000]/[G7221:107:16000:20:32000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G726-32:111:8000:20:0]/[G7221:107:16000:20:32000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [GSM:3:8000:20:13200]/[G7221:107:16000:20:32000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [telephone-event:101:8000:20:0]/[G7221:107:16000:20:32000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf send/recv payload to 101 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMA:8:8000:20:64000]/[G722:9:8000:20:64000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G729:18:8000:20:8000]/[G722:9:8000:20:64000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G726-32:111:8000:20:0]/[G722:9:8000:20:64000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [GSM:3:8000:20:13200]/[G722:9:8000:20:64000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [telephone-event:101:8000:20:0]/[G722:9:8000:20:64000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf send/recv payload to 101 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMA:8:8000:20:64000]/[PCMU:0:8000:20:64000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:2996 Set Codec sofia/internal/+41793940965 at bluesip.net PCMU/8000 20 ms 160 samples 64000 bits 2012-05-24 23:08:45.840749 [DEBUG] switch_core_codec.c:111 sofia/internal/+41793940965 at bluesip.net Original read codec set to PCMU:0 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf send/recv payload to 101 2012-05-24 23:08:45.840749 [DEBUG] sofia.c:5757 (sofia/internal/+41793940965 at bluesip.net) State Change CS_NEW -> CS_INIT 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_INIT 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/+41793940965 at bluesip.net) State INIT 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:85 sofia/internal/+41793940965 at bluesip.net SOFIA INIT 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:125 (sofia/internal/+41793940965 at bluesip.net) State Change CS_INIT -> CS_ROUTING 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/+41793940965 at bluesip.net) State INIT going to sleep 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_ROUTING 2012-05-24 23:08:45.840749 [DEBUG] switch_channel.c:1886 (sofia/internal/+41793940965 at bluesip.net) Callstate Change DOWN -> RINGING 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/+41793940965 at bluesip.net) State ROUTING 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:148 sofia/internal/+41793940965 at bluesip.net SOFIA ROUTING 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:104 sofia/internal/+41793940965 at bluesip.net Standard ROUTING 2012-05-24 23:08:45.840749 [INFO] mod_dialplan_xml.c:485 Processing +41793940965 <+41793940965>->kb-1002 in context public Dialplan: sofia/internal/+41793940965 at bluesip.net parsing [public->from_kamailio] continue=false Dialplan: sofia/internal/+41793940965 at bluesip.net Regex (PASS) [from_kamailio] destination_number(kb-1002) =~ /^(.+)$/ break=on-false Dialplan: sofia/internal/+41793940965 at bluesip.net Action transfer(kb-1002 XML default) 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:154 (sofia/internal/+41793940965 at bluesip.net) State Change CS_ROUTING -> CS_EXECUTE 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/+41793940965 at bluesip.net) State ROUTING going to sleep 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_EXECUTE 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/+41793940965 at bluesip.net) State EXECUTE 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:241 sofia/internal/+41793940965 at bluesip.net SOFIA EXECUTE 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:192 sofia/internal/+41793940965 at bluesip.net Standard EXECUTE EXECUTE sofia/internal/+41793940965 at bluesip.net transfer(kb-1002 XML default) 2012-05-24 23:08:45.840749 [DEBUG] switch_ivr.c:1711 (sofia/internal/+41793940965 at bluesip.net) State Change CS_EXECUTE -> CS_ROUTING 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:731 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:08:45.840749 [NOTICE] switch_ivr.c:1717 Transfer sofia/internal/+41793940965 at bluesip.net to XML[kb-1002 at default] 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/+41793940965 at bluesip.net) State EXECUTE going to sleep 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_ROUTING 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/+41793940965 at bluesip.net) State ROUTING 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:148 sofia/internal/+41793940965 at bluesip.net SOFIA ROUTING 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:104 sofia/internal/+41793940965 at bluesip.net Standard ROUTING 2012-05-24 23:08:45.840749 [INFO] mod_dialplan_xml.c:485 Processing +41793940965 <+41793940965>->kb-1002 in context default Dialplan: sofia/internal/+41793940965 at bluesip.net parsing [default->vbox] continue=false Dialplan: sofia/internal/+41793940965 at bluesip.net Regex (FAIL) [vbox] destination_number(kb-1002) =~ /^vb-([0-9][0-9][0-9])$/ break=on-false Dialplan: sofia/internal/+41793940965 at bluesip.net parsing [default->vmenu] continue=false Dialplan: sofia/internal/+41793940965 at bluesip.net Regex (FAIL) [vmenu] destination_number(kb-1002) =~ /^vm-([0-9][0-9][0-9][0-9])$/ break=on-false Dialplan: sofia/internal/+41793940965 at bluesip.net parsing [default->kbridge] continue=false Dialplan: sofia/internal/+41793940965 at bluesip.net Regex (PASS) [kbridge] destination_number(kb-1002) =~ /^kb-(.+)$/ break=on-false Dialplan: sofia/internal/+41793940965 at bluesip.net Action set(proxy_media=true) Dialplan: sofia/internal/+41793940965 at bluesip.net Action set(call_timeout=50) Dialplan: sofia/internal/+41793940965 at bluesip.net Action set(continue_on_fail=true) Dialplan: sofia/internal/+41793940965 at bluesip.net Action set(hangup_after_bridge=true) Dialplan: sofia/internal/+41793940965 at bluesip.net Action set(sip_invite_domain=78.138.90.58) Dialplan: sofia/internal/+41793940965 at bluesip.net Action export(sip_contact_user=ufs) Dialplan: sofia/internal/+41793940965 at bluesip.net Action bridge(sofia/78.138.90.58/1002 at 78.138.90.58:5060) Dialplan: sofia/internal/+41793940965 at bluesip.net Action answer() Dialplan: sofia/internal/+41793940965 at bluesip.net Action voicemail(default ${domain_name} 1002) 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:154 (sofia/internal/+41793940965 at bluesip.net) State Change CS_ROUTING -> CS_EXECUTE 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/+41793940965 at bluesip.net) State ROUTING going to sleep 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_EXECUTE 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/+41793940965 at bluesip.net) State EXECUTE 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:241 sofia/internal/+41793940965 at bluesip.net SOFIA EXECUTE 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:192 sofia/internal/+41793940965 at bluesip.net Standard EXECUTE EXECUTE sofia/internal/+41793940965 at bluesip.net set(proxy_media=true) 2012-05-24 23:08:45.840749 [DEBUG] mod_dptools.c:1281 sofia/internal/+41793940965 at bluesip.net SET [proxy_media]=[true] EXECUTE sofia/internal/+41793940965 at bluesip.net set(call_timeout=50) 2012-05-24 23:08:45.840749 [DEBUG] mod_dptools.c:1281 sofia/internal/+41793940965 at bluesip.net SET [call_timeout]=[50] EXECUTE sofia/internal/+41793940965 at bluesip.net set(continue_on_fail=true) 2012-05-24 23:08:45.840749 [DEBUG] mod_dptools.c:1281 sofia/internal/+41793940965 at bluesip.net SET [continue_on_fail]=[true] EXECUTE sofia/internal/+41793940965 at bluesip.net set(hangup_after_bridge=true) 2012-05-24 23:08:45.840749 [DEBUG] mod_dptools.c:1281 sofia/internal/+41793940965 at bluesip.net SET [hangup_after_bridge]=[true] EXECUTE sofia/internal/+41793940965 at bluesip.net set(sip_invite_domain=78.138.90.58) 2012-05-24 23:08:45.840749 [DEBUG] mod_dptools.c:1281 sofia/internal/+41793940965 at bluesip.net SET [sip_invite_domain]=[78.138.90.58] EXECUTE sofia/internal/+41793940965 at bluesip.net export(sip_contact_user=ufs) 2012-05-24 23:08:45.840749 [DEBUG] switch_channel.c:1093 EXPORT (export_vars) [sip_contact_user]=[ufs] EXECUTE sofia/internal/+41793940965 at bluesip.net bridge(sofia/78.138.90.58/1002 at 78.138.90.58:5060) 2012-05-24 23:08:45.840749 [DEBUG] switch_channel.c:1047 sofia/internal/+41793940965 at bluesip.net EXPORTING[export_vars] [sip_contact_user]=[ufs] to event 2012-05-24 23:08:45.840749 [DEBUG] switch_ivr_originate.c:1884 Parsing global variables 2012-05-24 23:08:45.840749 [NOTICE] switch_channel.c:926 New Channel sofia/internal/1002 at 78.138.90.58:5060 [69ce1f0c-a5f5-11e1-a838-7b1f73a7ffcf] 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:4691 (sofia/internal/1002 at 78.138.90.58:5060) State Change CS_NEW -> CS_INIT 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1002 at 78.138.90.58:5060) Running State Change CS_INIT 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/1002 at 78.138.90.58:5060) State INIT 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:85 sofia/internal/1002 at 78.138.90.58:5060 SOFIA INIT 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:1871 sofia/internal/1002 at 78.138.90.58:5060 Patched SDP --- v=0 o=root 20076 20076 IN IP4 217.74.179.28 s=session c=IN IP4 217.74.179.28 t=0 0 m=audio 25626 RTP/AVP 8 0 18 111 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:111 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 +++ v=0 o=FreeSWITCH 1629510322 1629510323 IN IP4 78.138.90.58 s=FreeSWITCH c=IN IP4 78.138.90.58 t=0 0 m=audio 31178 RTP/AVP 8 0 18 111 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:111 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 send 1340 bytes to udp/[78.138.90.58]:5060 at 23:08:45.852280: ?? ------------------------------------------------------------------------ ?? INVITE sip:1002 at 78.138.90.58:5060 SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKapDgFeegKyHXK ?? Max-Forwards: 67 ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? CSeq: 28614398 INVITE ?? Contact: ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? 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: 372 ?? P-hint: USRLOC ?? X-FS-Support: update_display,send_info ?? Remote-Party-ID: "+41793940965" ;party=calling;screen=yes;privacy=off ?? ?? v=0 ?? o=FreeSWITCH 1629510322 1629510323 IN IP4 78.138.90.58 ?? s=FreeSWITCH ?? c=IN IP4 78.138.90.58 ?? t=0 0 ?? m=audio 31178 RTP/AVP 8 0 18 111 3 101 ?? a=rtpmap:8 PCMA/8000 ?? a=rtpmap:0 PCMU/8000 ?? a=rtpmap:18 G729/8000 ?? a=fmtp:18 annexb=no ?? a=rtpmap:111 G726-32/8000 ?? a=rtpmap:3 GSM/8000 ?? a=rtpmap:101 telephone-event/8000 ?? a=fmtp:101 0-16 ?? a=silenceSupp:off - - - - ?? a=ptime:20 ?? ------------------------------------------------------------------------ 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:125 (sofia/internal/1002 at 78.138.90.58:5060) State Change CS_INIT -> CS_ROUTING 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/1002 at 78.138.90.58:5060) State INIT going to sleep 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1002 at 78.138.90.58:5060) Running State Change CS_ROUTING 2012-05-24 23:08:45.840749 [DEBUG] switch_channel.c:1886 (sofia/internal/1002 at 78.138.90.58:5060) Callstate Change DOWN -> RINGING 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/1002 at 78.138.90.58:5060) State ROUTING 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:148 sofia/internal/1002 at 78.138.90.58:5060 SOFIA ROUTING 2012-05-24 23:08:45.840749 [DEBUG] switch_ivr_originate.c:66 (sofia/internal/1002 at 78.138.90.58:5060) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/1002 at 78.138.90.58:5060) State ROUTING going to sleep 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1002 at 78.138.90.58:5060) Running State Change CS_CONSUME_MEDIA 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:429 (sofia/internal/1002 at 78.138.90.58:5060) State CONSUME_MEDIA 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:429 (sofia/internal/1002 at 78.138.90.58:5060) State CONSUME_MEDIA going to sleep 2012-05-24 23:08:45.840749 [DEBUG] sofia.c:5532 Channel sofia/internal/1002 at 78.138.90.58:5060 entering state [calling][0] recv 365 bytes from udp/[78.138.90.58]:5060 at 23:08:45.964309: ?? ------------------------------------------------------------------------ ?? SIP/2.0 100 trying -- your call is important to us ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKapDgFeegKyHXK ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? CSeq: 28614398 INVITE ?? Server: kamailio (3.1.5 (x86_64/linux)) ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 496 bytes from udp/[78.138.90.58]:5060 at 23:08:47.414766: ?? ------------------------------------------------------------------------ ?? SIP/2.0 180 Ringing ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKapDgFeegKyHXK ?? Record-Route: ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ;tag=qCSaB2YJOAsAr2dJ-RWNMhPoDdn1RFvs ?? CSeq: 28614398 INVITE ?? Contact: "Mama" ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, REGISTER ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ 2012-05-24 23:08:47.400741 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:47.400741 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:47.400741 [DEBUG] sofia.c:5532 Channel sofia/internal/1002 at 78.138.90.58:5060 entering state [proceeding][180] 2012-05-24 23:08:47.400741 [NOTICE] sofia.c:5624 Ring-Ready sofia/internal/1002 at 78.138.90.58:5060! send 995 bytes to udp/[217.74.179.29]:5060 at 23:08:47.416538: ?? ------------------------------------------------------------------------ ?? SIP/2.0 180 Ringing ?? Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 ?? Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 ?? Record-Route: ?? From: "+41793940965" ;tag=as00589402 ?? To: ;tag=S7UZQygFt62Nm ?? Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net ?? CSeq: 102 INVITE ?? Contact: ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Accept: application/sdp ?? 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-Length: 0 ?? Remote-Party-ID: "Outbound Call" ;party=calling;privacy=off;screen=no ?? ?? ------------------------------------------------------------------------ 2012-05-24 23:08:47.400741 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:08:47.400741 [NOTICE] mod_sofia.c:2514 Ring-Ready sofia/internal/+41793940965 at bluesip.net! 2012-05-24 23:08:47.400741 [DEBUG] sofia.c:5532 Channel sofia/internal/+41793940965 at bluesip.net entering state [early][180] 2012-05-24 23:08:47.400741 [DEBUG] switch_core_session.c:731 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:08:47.400741 [NOTICE] switch_ivr_originate.c:483 Ring Ready sofia/internal/+41793940965 at bluesip.net! recv 496 bytes from udp/[78.138.90.58]:5060 at 23:08:47.804967: ?? ------------------------------------------------------------------------ ?? SIP/2.0 180 Ringing ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKapDgFeegKyHXK ?? Record-Route: ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ;tag=549D2DD03BBA7C67 ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? CSeq: 28614398 INVITE ?? Contact: ?? User-Agent: AVM FRITZ!Box Fon WLAN 7170 29.04.80 (Jan 27 2010) ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ 2012-05-24 23:08:47.800746 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:47.800746 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:47.800746 [DEBUG] sofia.c:5532 Channel sofia/internal/1002 at 78.138.90.58:5060 entering state [proceeding][180] recv 496 bytes from udp/[78.138.90.58]:5060 at 23:08:48.182098: ?? ------------------------------------------------------------------------ ?? SIP/2.0 180 Ringing ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKapDgFeegKyHXK ?? Record-Route: ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ;tag=qCSaB2YJOAsAr2dJ-RWNMhPoDdn1RFvs ?? CSeq: 28614398 INVITE ?? Contact: "Mama" ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, REGISTER ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ 2012-05-24 23:08:48.180741 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:48.180741 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:48.180741 [DEBUG] sofia.c:5532 Channel sofia/internal/1002 at 78.138.90.58:5060 entering state [proceeding][180] recv 496 bytes from udp/[78.138.90.58]:5060 at 23:08:48.417403: ?? ------------------------------------------------------------------------ ?? SIP/2.0 180 Ringing ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKapDgFeegKyHXK ?? Record-Route: ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ;tag=549D2DD03BBA7C67 ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? CSeq: 28614398 INVITE ?? Contact: ?? User-Agent: AVM FRITZ!Box Fon WLAN 7170 29.04.80 (Jan 27 2010) ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ 2012-05-24 23:08:48.400744 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:48.400744 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:48.400744 [DEBUG] sofia.c:5532 Channel sofia/internal/1002 at 78.138.90.58:5060 entering state [proceeding][180] recv 496 bytes from udp/[78.138.90.58]:5060 at 23:08:49.292599: ?? ------------------------------------------------------------------------ ?? SIP/2.0 180 Ringing ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKapDgFeegKyHXK ?? Record-Route: ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ;tag=549D2DD03BBA7C67 ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? CSeq: 28614398 INVITE ?? Contact: ?? User-Agent: AVM FRITZ!Box Fon WLAN 7170 29.04.80 (Jan 27 2010) ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ 2012-05-24 23:08:49.280742 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:49.280742 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:49.280742 [DEBUG] sofia.c:5532 Channel sofia/internal/1002 at 78.138.90.58:5060 entering state [proceeding][180] recv 478 bytes from udp/[217.74.179.29]:5060 at 23:09:15.066012: ?? ------------------------------------------------------------------------ ?? CANCEL sip:gw+trunk_1000 at 78.138.90.58:6090;transport=udp;gw=trunk_1000 SIP/2.0 ?? Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 ?? Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 ?? From: "+41793940965" ;tag=as00589402 ?? To: ?? Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net ?? CSeq: 102 CANCEL ?? User-Agent: blueSIP PSTN GW ?? Max-Forwards: 69 ?? Content-Length: 0 ?? P-hint: USRLOC ?? ?? ------------------------------------------------------------------------ send 369 bytes to udp/[217.74.179.29]:5060 at 23:09:15.066176: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 ?? Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 ?? From: "+41793940965" ;tag=as00589402 ?? To: ;tag=S7UZQygFt62Nm ?? Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net ?? CSeq: 102 CANCEL ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ send 766 bytes to udp/[217.74.179.29]:5060 at 23:09:15.066289: ?? ------------------------------------------------------------------------ ?? SIP/2.0 487 Request Terminated ?? Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 ?? Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 ?? From: "+41793940965" ;tag=as00589402 ?? To: ;tag=S7UZQygFt62Nm ?? Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net ?? CSeq: 102 INVITE ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? 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-Length: 0 ?? ?? ------------------------------------------------------------------------ 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:09:15.060742 [DEBUG] sofia.c:5532 Channel sofia/internal/+41793940965 at bluesip.net entering state [terminated][487] 2012-05-24 23:09:15.060742 [DEBUG] switch_channel.c:2848 (sofia/internal/+41793940965 at bluesip.net) Callstate Change RINGING -> HANGUP 2012-05-24 23:09:15.060742 [NOTICE] sofia.c:6301 Hangup sofia/internal/+41793940965 at bluesip.net [CS_EXECUTE] [ORIGINATOR_CANCEL] 2012-05-24 23:09:15.060742 [DEBUG] switch_channel.c:2871 Send signal sofia/internal/+41793940965 at bluesip.net [KILL] 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:09:15.060742 [DEBUG] switch_channel.c:2848 (sofia/internal/1002 at 78.138.90.58:5060) Callstate Change RINGING -> HANGUP 2012-05-24 23:09:15.060742 [NOTICE] switch_ivr_originate.c:3182 Hangup sofia/internal/1002 at 78.138.90.58:5060 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL] 2012-05-24 23:09:15.060742 [DEBUG] switch_channel.c:2871 Send signal sofia/internal/1002 at 78.138.90.58:5060 [KILL] 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1002 at 78.138.90.58:5060) Running State Change CS_HANGUP 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/1002 at 78.138.90.58:5060) State HANGUP 2012-05-24 23:09:15.060742 [DEBUG] mod_sofia.c:469 Channel sofia/internal/1002 at 78.138.90.58:5060 hanging up, cause: ORIGINATOR_CANCEL 2012-05-24 23:09:15.060742 [DEBUG] mod_sofia.c:523 Sending CANCEL to sofia/internal/1002 at 78.138.90.58:5060 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:47 sofia/internal/1002 at 78.138.90.58:5060 Standard HANGUP, cause: ORIGINATOR_CANCEL 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/1002 at 78.138.90.58:5060) State HANGUP going to sleep 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:393 (sofia/internal/1002 at 78.138.90.58:5060) State Change CS_HANGUP -> CS_REPORTING 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1002 at 78.138.90.58:5060) Running State Change CS_REPORTING 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/1002 at 78.138.90.58:5060) State REPORTING 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:79 sofia/internal/1002 at 78.138.90.58:5060 Standard REPORTING, cause: ORIGINATOR_CANCEL 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/1002 at 78.138.90.58:5060) State REPORTING going to sleep 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:387 (sofia/internal/1002 at 78.138.90.58:5060) State Change CS_REPORTING -> CS_DESTROY 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1382 Session 9 (sofia/internal/1002 at 78.138.90.58:5060) Locked, Waiting on external entities 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:09:15.060742 [DEBUG] switch_ivr_originate.c:3358 Originate Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL] 2012-05-24 23:09:15.060742 [NOTICE] switch_core_session.c:1400 Session 9 (sofia/internal/1002 at 78.138.90.58:5060) Ended 2012-05-24 23:09:15.060742 [NOTICE] switch_core_session.c:1402 Close Channel sofia/internal/1002 at 78.138.90.58:5060 [CS_DESTROY] 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:491 (sofia/internal/1002 at 78.138.90.58:5060) Callstate Change HANGUP -> DOWN 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:494 (sofia/internal/1002 at 78.138.90.58:5060) Running State Change CS_DESTROY 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/1002 at 78.138.90.58:5060) State DESTROY 2012-05-24 23:09:15.060742 [DEBUG] mod_sofia.c:374 sofia/internal/1002 at 78.138.90.58:5060 SOFIA DESTROY 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:86 sofia/internal/1002 at 78.138.90.58:5060 Standard DESTROY 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/1002 at 78.138.90.58:5060) State DESTROY going to sleep 2012-05-24 23:09:15.060742 [INFO] mod_dptools.c:2922 Originate Failed.? Cause: ORIGINATOR_CANCEL 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:2287 sofia/internal/+41793940965 at bluesip.net skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/+41793940965 at bluesip.net) State EXECUTE going to sleep 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_HANGUP 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/+41793940965 at bluesip.net) State HANGUP 2012-05-24 23:09:15.060742 [DEBUG] mod_sofia.c:469 Channel sofia/internal/+41793940965 at bluesip.net hanging up, cause: ORIGINATOR_CANCEL 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:47 sofia/internal/+41793940965 at bluesip.net Standard HANGUP, cause: ORIGINATOR_CANCEL 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/+41793940965 at bluesip.net) State HANGUP going to sleep 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:393 (sofia/internal/+41793940965 at bluesip.net) State Change CS_HANGUP -> CS_REPORTING 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_REPORTING 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/+41793940965 at bluesip.net) State REPORTING send 383 bytes to udp/[78.138.90.58]:5060 at 23:09:15.087695: ?? ------------------------------------------------------------------------ ?? CANCEL sip:1002 at 78.138.90.58:5060 SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKapDgFeegKyHXK ?? Max-Forwards: 67 ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? CSeq: 28614398 CANCEL ?? Reason: FreeSWITCH;cause=487;text="ORIGINATOR_CANCEL" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 408 bytes from udp/[217.74.179.29]:5060 at 23:09:15.094670: ?? ------------------------------------------------------------------------ ?? ACK sip:gw+trunk_1000 at 78.138.90.58:6090;transport=udp;gw=trunk_1000 SIP/2.0 ?? Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 ?? From: "+41793940965" ;tag=as00589402 ?? Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net ?? To: ;tag=S7UZQygFt62Nm ?? CSeq: 102 ACK ?? User-Agent: Sip EXpress router(0.9.7 (i386/linux)) ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:79 sofia/internal/+41793940965 at bluesip.net Standard REPORTING, cause: ORIGINATOR_CANCEL 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/+41793940965 at bluesip.net) State REPORTING going to sleep 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:387 (sofia/internal/+41793940965 at bluesip.net) State Change CS_REPORTING -> CS_DESTROY 2012-05-24 23:09:15.120807 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:09:15.120807 [DEBUG] switch_core_session.c:1382 Session 8 (sofia/internal/+41793940965 at bluesip.net) Locked, Waiting on external entities 2012-05-24 23:09:15.120807 [NOTICE] switch_core_session.c:1400 Session 8 (sofia/internal/+41793940965 at bluesip.net) Ended 2012-05-24 23:09:15.120807 [NOTICE] switch_core_session.c:1402 Close Channel sofia/internal/+41793940965 at bluesip.net [CS_DESTROY] 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:491 (sofia/internal/+41793940965 at bluesip.net) Callstate Change HANGUP -> DOWN 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:494 (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_DESTROY 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/+41793940965 at bluesip.net) State DESTROY 2012-05-24 23:09:15.120807 [DEBUG] mod_sofia.c:374 sofia/internal/+41793940965 at bluesip.net SOFIA DESTROY 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:86 sofia/internal/+41793940965 at bluesip.net Standard DESTROY 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/+41793940965 at bluesip.net) State DESTROY going to sleep recv 378 bytes from udp/[78.138.90.58]:5060 at 23:09:15.135235: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 canceling ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKapDgFeegKyHXK ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ;tag=a6a1c5f60faecf035a1ae5b6e96e979a-8c5b ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? CSeq: 28614398 CANCEL ?? Server: kamailio (3.1.5 (x86_64/linux)) ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 446 bytes from udp/[78.138.90.58]:5060 at 23:09:16.419308: ?? ------------------------------------------------------------------------ ?? SIP/2.0 487 Request Terminated ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKapDgFeegKyHXK ?? Record-Route: ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ;tag=qCSaB2YJOAsAr2dJ-RWNMhPoDdn1RFvs ?? CSeq: 28614398 INVITE ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, REGISTER ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 354 bytes to udp/[78.138.90.58]:5060 at 23:09:16.419543: ?? ------------------------------------------------------------------------ ?? ACK sip:1002 at 78.138.90.58:5060 SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKapDgFeegKyHXK ?? Max-Forwards: 67 ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ;tag=qCSaB2YJOAsAr2dJ-RWNMhPoDdn1RFvs ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? CSeq: 28614398 ACK ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 564 bytes from udp/[41.34.123.243]:58881 at 23:13:13.612418: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjHQIpv.ACHGbXD4IHrMr72LwZ8kYdlu-J ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=wOP.FVT4Crk8O5E.bB7BWUhNO6QG91IG ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60926 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 744 bytes to udp/[41.34.123.243]:58881 at 23:13:13.613152: ?? ------------------------------------------------------------------------ ?? SIP/2.0 401 Unauthorized ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjHQIpv.ACHGbXD4IHrMr72LwZ8kYdlu-J;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=wOP.FVT4Crk8O5E.bB7BWUhNO6QG91IG ?? To: "Sherif 1003" ;tag=USeHUmjpmrFUB ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60926 REGISTER ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? WWW-Authenticate: Digest realm="sip.pcfone.com", nonce="09674764-a5f6-11e1-a83c-7b1f73a7ffcf", algorithm=MD5, qop="auth" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 830 bytes from udp/[41.34.123.243]:58881 at 23:13:13.847746: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjFr7rIAdAJPRQd2GmrMrVib3KT3Rup0q4 ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=wOP.FVT4Crk8O5E.bB7BWUhNO6QG91IG ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60927 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Authorization: Digest username="1002", realm="sip.pcfone.com", nonce="09674764-a5f6-11e1-a83c-7b1f73a7ffcf", uri="sip:sip.pcfone.com:6090", response="3bc0bb82a803ded3641704c4eb78d58f", algorithm=MD5, cnonce="sc3AJfhuev18Bs-L-6tORmI.RXlxnPOz", qop=auth, nc=00000001 ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 706 bytes to udp/[41.34.123.243]:58881 at 23:13:13.850197: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjFr7rIAdAJPRQd2GmrMrVib3KT3Rup0q4;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=wOP.FVT4Crk8O5E.bB7BWUhNO6QG91IG ?? To: "Sherif 1003" ;tag=v279vF3SH15DQ ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60927 REGISTER ?? Contact: ;expires=300 ?? Date: Thu, 24 May 2012 23:13:13 GMT ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ send 989 bytes to udp/[41.34.123.243]:58881 at 23:13:13.888717: ?? ------------------------------------------------------------------------ ?? NOTIFY sip:1002 at 192.168.178.26:58881;ob SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKBZ68g9yKg77FF ?? Route: ;ob ?? Max-Forwards: 70 ?? From: ;tag=XB12yamXeav0j ?? To: ?? Call-ID: e0efa252-2098-1230-8985-00163e6bb553 ?? CSeq: 28614532 NOTIFY ?? Contact: ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? 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=noresource ?? Content-Type: application/simple-message-summary ?? Content-Length: 91 ?? ?? Messages-Waiting: yes ?? Message-Account: sip:1002 at 78.138.90.58 ?? Voice-Message: 1/0 (0/0) ?? ?? ------------------------------------------------------------------------ recv 308 bytes from udp/[41.34.123.243]:58881 at 23:13:14.298850: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;received=78.138.90.58;branch=z9hG4bKBZ68g9yKg77FF ?? Call-ID: e0efa252-2098-1230-8985-00163e6bb553 ?? From: ;tag=XB12yamXeav0j ?? To: ;tag=z9hG4bKBZ68g9yKg77FF ?? CSeq: 28614532 NOTIFY ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ recv 564 bytes from udp/[41.34.123.243]:58881 at 23:18:09.067232: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjABTslmZIhiHYRtLqQn2erwmX19MVYHFC ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=JBSXL5WZLWbLzpvRFoTSP5x8qQslI1ST ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60928 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 744 bytes to udp/[41.34.123.243]:58881 at 23:18:09.068121: ?? ------------------------------------------------------------------------ ?? SIP/2.0 401 Unauthorized ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjABTslmZIhiHYRtLqQn2erwmX19MVYHFC;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=JBSXL5WZLWbLzpvRFoTSP5x8qQslI1ST ?? To: "Sherif 1003" ;tag=ymtU0540BKjKe ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60928 REGISTER ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? WWW-Authenticate: Digest realm="sip.pcfone.com", nonce="b9821df4-a5f6-11e1-a83d-7b1f73a7ffcf", algorithm=MD5, qop="auth" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 830 bytes from udp/[41.34.123.243]:58881 at 23:18:09.302944: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjRgtztcy63vMT97Jp5pH9n311li5EFeY- ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=JBSXL5WZLWbLzpvRFoTSP5x8qQslI1ST ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60929 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Authorization: Digest username="1002", realm="sip.pcfone.com", nonce="b9821df4-a5f6-11e1-a83d-7b1f73a7ffcf", uri="sip:sip.pcfone.com:6090", response="519c6301816b51f3bbb98d97a347e92b", algorithm=MD5, cnonce="sc3AJfhuev18Bs-L-6tORmI.RXlxnPOz", qop=auth, nc=00000001 ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 706 bytes to udp/[41.34.123.243]:58881 at 23:18:09.305217: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjRgtztcy63vMT97Jp5pH9n311li5EFeY-;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=JBSXL5WZLWbLzpvRFoTSP5x8qQslI1ST ?? To: "Sherif 1003" ;tag=ZXKm20N48U85S ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60929 REGISTER ?? Contact: ;expires=300 ?? Date: Thu, 24 May 2012 23:18:09 GMT ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ send 989 bytes to udp/[41.34.123.243]:58881 at 23:18:09.379987: ?? ------------------------------------------------------------------------ ?? NOTIFY sip:1002 at 192.168.178.26:58881;ob SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKc8Z1j4FQDgy2a ?? Route: ;ob ?? Max-Forwards: 70 ?? From: ;tag=06cD4U6754yrN ?? To: ?? Call-ID: 91100602-2099-1230-8985-00163e6bb553 ?? CSeq: 28614680 NOTIFY ?? Contact: ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? 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=noresource ?? Content-Type: application/simple-message-summary ?? Content-Length: 91 ?? ?? Messages-Waiting: yes ?? Message-Account: sip:1002 at 78.138.90.58 ?? Voice-Message: 1/0 (0/0) ?? ?? ------------------------------------------------------------------------ recv 308 bytes from udp/[41.34.123.243]:58881 at 23:18:09.773629: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;received=78.138.90.58;branch=z9hG4bKc8Z1j4FQDgy2a ?? Call-ID: 91100602-2099-1230-8985-00163e6bb553 ?? From: ;tag=06cD4U6754yrN ?? To: ;tag=z9hG4bKc8Z1j4FQDgy2a ?? CSeq: 28614680 NOTIFY ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ recv 564 bytes from udp/[41.34.123.243]:58881 at 23:23:04.522455: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjaCuKyaBi6ZqcsHyp5HhH3WthJStom0SK ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=qfUt1fQWuE-Ro3zWld6X0r6WidtD6B1q ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60930 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 744 bytes to udp/[41.34.123.243]:58881 at 23:23:04.523158: ?? ------------------------------------------------------------------------ ?? SIP/2.0 401 Unauthorized ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjaCuKyaBi6ZqcsHyp5HhH3WthJStom0SK;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=qfUt1fQWuE-Ro3zWld6X0r6WidtD6B1q ?? To: "Sherif 1003" ;tag=1F655pQB3DNBH ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60930 REGISTER ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? WWW-Authenticate: Digest realm="sip.pcfone.com", nonce="699cfe2a-a5f7-11e1-a83e-7b1f73a7ffcf", algorithm=MD5, qop="auth" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 830 bytes from udp/[41.34.123.243]:58881 at 23:23:04.757798: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjs8SAufeoyzCLrePaAbYhZuOqQF47XIc- ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=qfUt1fQWuE-Ro3zWld6X0r6WidtD6B1q ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60931 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Authorization: Digest username="1002", realm="sip.pcfone.com", nonce="699cfe2a-a5f7-11e1-a83e-7b1f73a7ffcf", uri="sip:sip.pcfone.com:6090", response="23a886934aaf049f08432b6bb226bd37", algorithm=MD5, cnonce="sc3AJfhuev18Bs-L-6tORmI.RXlxnPOz", qop=auth, nc=00000001 ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 706 bytes to udp/[41.34.123.243]:58881 at 23:23:04.759924: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjs8SAufeoyzCLrePaAbYhZuOqQF47XIc-;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=qfUt1fQWuE-Ro3zWld6X0r6WidtD6B1q ?? To: "Sherif 1003" ;tag=2rZy7H8e0pByc ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60931 REGISTER ?? Contact: ;expires=300 ?? Date: Thu, 24 May 2012 23:23:04 GMT ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ send 989 bytes to udp/[41.34.123.243]:58881 at 23:23:04.872728: ?? ------------------------------------------------------------------------ ?? NOTIFY sip:1002 at 192.168.178.26:58881;ob SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKDHStmZ0taSmNp ?? Route: ;ob ?? Max-Forwards: 70 ?? From: ;tag=31rQ9cSjXZ1gr ?? To: ?? Call-ID: 4130a1ce-209a-1230-8985-00163e6bb553 ?? CSeq: 28614828 NOTIFY ?? Contact: ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? 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=noresource ?? Content-Type: application/simple-message-summary ?? Content-Length: 91 ?? ?? Messages-Waiting: yes ?? Message-Account: sip:1002 at 78.138.90.58 ?? Voice-Message: 1/0 (0/0) ?? ?? ------------------------------------------------------------------------ recv 308 bytes from udp/[41.34.123.243]:58881 at 23:23:05.269063: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;received=78.138.90.58;branch=z9hG4bKDHStmZ0taSmNp ?? Call-ID: 4130a1ce-209a-1230-8985-00163e6bb553 ?? From: ;tag=31rQ9cSjXZ1gr ?? To: ;tag=z9hG4bKDHStmZ0taSmNp ?? CSeq: 28614828 NOTIFY ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 916 bytes to udp/[217.74.179.29]:5060 at 23:25:45.873423: ?? ------------------------------------------------------------------------ ?? REGISTER sip:bluesip.net;transport=udp SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKetjKptHy71a8H ?? Max-Forwards: 70 ?? From: ;tag=6r0vBQZS650Fg ?? To: ?? Call-ID: 2a3e4696-a5b9-11e1-a7ed-7b1f73a7ffcf ?? CSeq: 28601493 REGISTER ?? Contact: ?? Expires: 3600 ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? Authorization: Digest username="bluesip/sherifomran", realm="bluesip.net", nonce="4fbeba4ebfb7414d5c701bd6f994402d551f3acf", algorithm=MD5, uri="sip:bluesip.net;transport=udp", response="c6cdcafe0418e519bc9ee0d8fa3d4d74" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 722 bytes from udp/[217.74.179.29]:5060 at 23:25:45.904176: ?? ------------------------------------------------------------------------ ?? SIP/2.0 401 Unauthorized ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKetjKptHy71a8H ?? From: ;tag=6r0vBQZS650Fg ?? To: ;tag=0354a2e1b960c9cc2279eca4e5f84e20.1b16 ?? Call-ID: 2a3e4696-a5b9-11e1-a7ed-7b1f73a7ffcf ?? CSeq: 28601493 REGISTER ?? WWW-Authenticate: Digest realm="bluesip.net", nonce="4fbec4a4ba82d1f1feaa32ccec01b85be054fcf7", stale=true ?? Server: Sip EXpress router (0.9.7 (i386/linux)) ?? Content-Length: 0 ?? Warning: 392 217.74.179.29:5060 "Noisy feedback tells:? pid=5455 req_src_ip=78.138.90.58 req_src_port=6090 in_uri=sip:bluesip.net;transport=udp out_uri=sip:bluesip.net;transport=udp via_cnt==1" ?? ?? ------------------------------------------------------------------------ send 916 bytes to udp/[217.74.179.29]:5060 at 23:25:45.904416: ?? ------------------------------------------------------------------------ ?? REGISTER sip:bluesip.net;transport=udp SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKF3BcrN214a1tD ?? Max-Forwards: 70 ?? From: ;tag=6r0vBQZS650Fg ?? To: ?? Call-ID: 2a3e4696-a5b9-11e1-a7ed-7b1f73a7ffcf ?? CSeq: 28601494 REGISTER ?? Contact: ?? Expires: 3600 ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? Authorization: Digest username="bluesip/sherifomran", realm="bluesip.net", nonce="4fbec4a4ba82d1f1feaa32ccec01b85be054fcf7", algorithm=MD5, uri="sip:bluesip.net;transport=udp", response="4c09dbe4b9accb52d4104b40dfe20040" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 699 bytes from udp/[217.74.179.29]:5060 at 23:25:45.935046: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKF3BcrN214a1tD ?? From: ;tag=6r0vBQZS650Fg ?? To: ;tag=0354a2e1b960c9cc2279eca4e5f84e20.2017 ?? Call-ID: 2a3e4696-a5b9-11e1-a7ed-7b1f73a7ffcf ?? CSeq: 28601494 REGISTER ?? Contact: ;q=0.5;expires=3600 ?? Server: Sip EXpress router (0.9.7 (i386/linux)) ?? Content-Length: 0 ?? Warning: 392 217.74.179.29:5060 "Noisy feedback tells:? pid=5462 req_src_ip=78.138.90.58 req_src_port=6090 in_uri=sip:bluesip.net;transport=udp out_uri=sip:bluesip.net;transport=udp via_cnt==1" ?? ?? ------------------------------------------------------------------------ recv 564 bytes from udp/[41.34.123.243]:58881 at 23:27:59.977057: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjEOP.a4NoXqoFqC7a.k2tyeX-mYlUyH4H ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60932 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 744 bytes to udp/[41.34.123.243]:58881 at 23:27:59.977960: ?? ------------------------------------------------------------------------ ?? SIP/2.0 401 Unauthorized ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjEOP.a4NoXqoFqC7a.k2tyeX-mYlUyH4H;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K ?? To: "Sherif 1003" ;tag=4ajgB89Nt8Q3K ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60932 REGISTER ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? WWW-Authenticate: Digest realm="sip.pcfone.com", nonce="19b7cc40-a5f8-11e1-a83f-7b1f73a7ffcf", algorithm=MD5, qop="auth" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 564 bytes from udp/[41.34.123.243]:58881 at 23:28:00.468487: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjEOP.a4NoXqoFqC7a.k2tyeX-mYlUyH4H ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60932 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 744 bytes to udp/[41.34.123.243]:58881 at 23:28:00.468679: ?? ------------------------------------------------------------------------ ?? SIP/2.0 401 Unauthorized ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjEOP.a4NoXqoFqC7a.k2tyeX-mYlUyH4H;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K ?? To: "Sherif 1003" ;tag=4ajgB89Nt8Q3K ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60932 REGISTER ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? WWW-Authenticate: Digest realm="sip.pcfone.com", nonce="19b7cc40-a5f8-11e1-a83f-7b1f73a7ffcf", algorithm=MD5, qop="auth" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 830 bytes from udp/[41.34.123.243]:58881 at 23:28:00.717446: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPj0iHd3-XUjnvcBs9DiJUCWzrmYTmWxFLt ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60933 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Authorization: Digest username="1002", realm="sip.pcfone.com", nonce="19b7cc40-a5f8-11e1-a83f-7b1f73a7ffcf", uri="sip:sip.pcfone.com:6090", response="e21c903e6d98b5adcdd2eefa75217157", algorithm=MD5, cnonce="sc3AJfhuev18Bs-L-6tORmI.RXlxnPOz", qop=auth, nc=00000001 ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 706 bytes to udp/[41.34.123.243]:58881 at 23:28:00.719947: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPj0iHd3-XUjnvcBs9DiJUCWzrmYTmWxFLt;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K ?? To: "Sherif 1003" ;tag=5KB9c3tSQHepF ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60933 REGISTER ?? Contact: ;expires=300 ?? Date: Thu, 24 May 2012 23:28:00 GMT ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ send 989 bytes to udp/[41.34.123.243]:58881 at 23:28:00.868368: ?? ------------------------------------------------------------------------ ?? NOTIFY sip:1002 at 192.168.178.26:58881;ob SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKgc54SgK51KQDS ?? Route: ;ob ?? Max-Forwards: 70 ?? From: ;tag=6v41eyBXmt48a ?? To: ?? Call-ID: f19dfbce-209a-1230-8985-00163e6bb553 ?? CSeq: 28614976 NOTIFY ?? Contact: ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? 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=noresource ?? Content-Type: application/simple-message-summary ?? Content-Length: 91 ?? ?? Messages-Waiting: yes ?? Message-Account: sip:1002 at 78.138.90.58 ?? Voice-Message: 1/0 (0/0) ?? ?? ------------------------------------------------------------------------ recv 308 bytes from udp/[41.34.123.243]:58881 at 23:28:01.128279: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;received=78.138.90.58;branch=z9hG4bKgc54SgK51KQDS ?? Call-ID: f19dfbce-209a-1230-8985-00163e6bb553 ?? From: ;tag=6v41eyBXmt48a ?? To: ;tag=z9hG4bKgc54SgK51KQDS ?? CSeq: 28614976 NOTIFY ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 913 bytes to udp/[217.74.179.29]:5060 at 23:28:11.885213: ?? ------------------------------------------------------------------------ ?? REGISTER sip:bluesip.net;transport=udp SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKHNyXUB48yvD0m ?? Max-Forwards: 70 ?? From: ;tag=5F739Uep9vaXm ?? To: ?? Call-ID: 2a3e49c0-a5b9-11e1-a7ee-7b1f73a7ffcf ?? CSeq: 28601493 REGISTER ?? Contact: ?? Expires: 3600 ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? Authorization: Digest username="bluesip/salahomran", realm="bluesip.net", nonce="4fbebeca79a5cff7c417463814cb9054d313bcac", algorithm=MD5, uri="sip:bluesip.net;transport=udp", response="bb1babb90f4ea1dc8fbb9aa45d6038c7" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 720 bytes from udp/[217.74.179.29]:5060 at 23:28:11.915764: ?? ------------------------------------------------------------------------ ?? SIP/2.0 401 Unauthorized ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKHNyXUB48yvD0m ?? From: ;tag=5F739Uep9vaXm ?? To: ;tag=0354a2e1b960c9cc2279eca4e5f84e20.3454 ?? Call-ID: 2a3e49c0-a5b9-11e1-a7ee-7b1f73a7ffcf ?? CSeq: 28601493 REGISTER ?? WWW-Authenticate: Digest realm="bluesip.net", nonce="4fbec5365af3a174f1cc0b8cda770c727d25bf71", stale=true ?? Server: Sip EXpress router (0.9.7 (i386/linux)) ?? Content-Length: 0 ?? Warning: 392 217.74.179.29:5060 "Noisy feedback tells:? pid=5462 req_src_ip=78.138.90.58 req_src_port=6090 in_uri=sip:bluesip.net;transport=udp out_uri=sip:bluesip.net;transport=udp via_cnt==1" ?? ?? ------------------------------------------------------------------------ send 913 bytes to udp/[217.74.179.29]:5060 at 23:28:11.915914: ?? ------------------------------------------------------------------------ ?? REGISTER sip:bluesip.net;transport=udp SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKjyQpX6mcv53jg ?? Max-Forwards: 70 ?? From: ;tag=5F739Uep9vaXm ?? To: ?? Call-ID: 2a3e49c0-a5b9-11e1-a7ee-7b1f73a7ffcf ?? CSeq: 28601494 REGISTER ?? Contact: ?? Expires: 3600 ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? Authorization: Digest username="bluesip/salahomran", realm="bluesip.net", nonce="4fbec5365af3a174f1cc0b8cda770c727d25bf71", algorithm=MD5, uri="sip:bluesip.net;transport=udp", response="e260f6dc9b01ce8bea8c5ffb36bd4ddd" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 697 bytes from udp/[217.74.179.29]:5060 at 23:28:11.946325: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKjyQpX6mcv53jg ?? From: ;tag=5F739Uep9vaXm ?? To: ;tag=0354a2e1b960c9cc2279eca4e5f84e20.4b5c ?? Call-ID: 2a3e49c0-a5b9-11e1-a7ee-7b1f73a7ffcf ?? CSeq: 28601494 REGISTER ?? Contact: ;q=0.5;expires=3600 ?? Server: Sip EXpress router (0.9.7 (i386/linux)) ?? Content-Length: 0 ?? Warning: 392 217.74.179.29:5060 "Noisy feedback tells:? pid=5455 req_src_ip=78.138.90.58 req_src_port=6090 in_uri=sip:bluesip.net;transport=udp out_uri=sip:bluesip.net;transport=udp via_cnt==1" ?? ?? ------------------------------------------------------------------------ recv 564 bytes from udp/[41.34.123.243]:58881 at 23:32:55.930149: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjuk8DZGiaXfN7BmtslNHVjtSg5UXsbHXc ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=Pz.t-0tGXj3saM.ehBiB1EAMmmitze1D ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60934 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 744 bytes to udp/[41.34.123.243]:58881 at 23:32:55.931324: ?? ------------------------------------------------------------------------ ?? SIP/2.0 401 Unauthorized ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjuk8DZGiaXfN7BmtslNHVjtSg5UXsbHXc;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=Pz.t-0tGXj3saM.ehBiB1EAMmmitze1D ?? To: "Sherif 1003" ;tag=75XtgSv0H3tUp ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60934 REGISTER ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? WWW-Authenticate: Digest realm="sip.pcfone.com", nonce="ca1eabee-a5f8-11e1-a840-7b1f73a7ffcf", algorithm=MD5, qop="auth" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 830 bytes from udp/[41.34.123.243]:58881 at 23:32:56.165624: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjKEQ6xXmyzZxA6vCM88vHNYAOPAjUqFsD ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=Pz.t-0tGXj3saM.ehBiB1EAMmmitze1D ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60935 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Authorization: Digest username="1002", realm="sip.pcfone.com", nonce="ca1eabee-a5f8-11e1-a840-7b1f73a7ffcf", uri="sip:sip.pcfone.com:6090", response="ff6d12f4ec7abd4b37187222b3af2358", algorithm=MD5, cnonce="sc3AJfhuev18Bs-L-6tORmI.RXlxnPOz", qop=auth, nc=00000001 ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 706 bytes to udp/[41.34.123.243]:58881 at 23:32:56.168033: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjKEQ6xXmyzZxA6vCM88vHNYAOPAjUqFsD;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=Pz.t-0tGXj3saM.ehBiB1EAMmmitze1D ?? To: "Sherif 1003" ;tag=8eQKjmD4ecHej ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60935 REGISTER ?? Contact: ;expires=300 ?? Date: Thu, 24 May 2012 23:32:56 GMT ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ send 989 bytes to udp/[41.34.123.243]:58881 at 23:32:56.360238: ?? ------------------------------------------------------------------------ ?? NOTIFY sip:1002 at 192.168.178.26:58881;ob SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKK7gFZ15FSet5B ?? Route: ;ob ?? Max-Forwards: 70 ?? From: ;tag=9QgcmFy7BN70D ?? To: ?? Call-ID: a1be7708-209b-1230-8985-00163e6bb553 ?? CSeq: 28615124 NOTIFY ?? Contact: ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? 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=noresource ?? Content-Type: application/simple-message-summary ?? Content-Length: 91 ?? ?? Messages-Waiting: yes ?? Message-Account: sip:1002 at 78.138.90.58 ?? Voice-Message: 1/0 (0/0) ?? ?? ------------------------------------------------------------------------ recv 308 bytes from udp/[41.34.123.243]:58881 at 23:32:56.623554: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;received=78.138.90.58;branch=z9hG4bKK7gFZ15FSet5B ?? Call-ID: a1be7708-209b-1230-8985-00163e6bb553 ?? From: ;tag=9QgcmFy7BN70D ?? To: ;tag=z9hG4bKK7gFZ15FSet5B ?? CSeq: 28615124 NOTIFY ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ freeswitch at internal> --- On Thu, 5/24/12, Kristian Kielhofner wrote: From: Kristian Kielhofner Subject: Re: [Freeswitch-users] Freeswitch (SBC + Media) does not know that gateway phone is picked up To: "FreeSWITCH Users Help" Date: Thursday, May 24, 2012, 7:51 PM Siptrace and logs please. On Thu, May 24, 2012 at 11:53 AM, Sherif Omran wrote: > > Hi all, > > My topology is as follows: > > Kamailio? -> FS (SBS+Media server) > > I came across an issue with my system as follows.? I have a Hardphone registered. When I do local call inside kamailio, it gets to FS and returns back well and FS understands when I lift the handset. However, I added a gateway (german landline server), when I call my self from another phone, the call gets to FS and then transmits to Kamailio, it rings my extension but when I lift the handset FS does not notice it and keeps ringing. > > Any body has an Idea? Here is my gateway trunk. > > > ??????? > ??????? > ??????? > ??????? > ??????? > ????? > > > thanks in advance > Sherif Omran > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Kristian Kielhofner _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120524/682d1664/attachment-0001.html From nathandownes at hotmail.com Fri May 25 04:00:47 2012 From: nathandownes at hotmail.com (Nathan Downes) Date: Fri, 25 May 2012 00:00:47 +0000 Subject: [Freeswitch-users] RTP media issue Message-ID: Hi, I had previous reported an issue with poor voice quality, appearing to stem from occasion wrong timestamps coming from provider, but the end user's experience was much worse than what I could see/hear in the trace. I have finally captured an event inbound and outbound. The thing I don't understand is I thought even though FS proxied the media it didn't touch it or change anything, but it appears it is. The 2 traces are http://www.nortec.com.au/inbound.pcap.gz and http://www.nortec.com.au/outbound.pcap.gz Inbound is from my trunk provider to FS box and outbound is FS box to ATA in FTTH GPON. The event I am talking about, if both traces are open, is in the inbound one inbetween packet 8114 and 8117 the provider drops a packet or I don't receive it. In the corresponding outbound trace, between packet 8144 and 8152, it appears FS misses a whole heap of packets (.1 seconds) between 8146 and 8152 then it increases the timestamp only by 40 rather than 160 on packet 8152. This seems to not affect SIP phones themselves but causes issues with the FTTH GPON ATA. This causes a gap in the audio for the end user, and when they miss a high number of packets even though it sounds good on the inbound trace the end users experience is horrible. This trace is actually a good one, but the wrong timestamp can occur once per second, causing end user to lose 10%+ of incoming audio only. The issue only affects the audio coming from provider to FS to end user. I am chasing it up with the voice provider to try and eliminate the occasional packet loss, but if I could stop/fix FS from doing its adjustment/gap/something the end user wouldn't even notice it. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120525/0b60fb29/attachment.html From djbinter at gmail.com Fri May 25 05:19:46 2012 From: djbinter at gmail.com (Dorn DJBinter) Date: Thu, 24 May 2012 18:19:46 -0700 Subject: [Freeswitch-users] RTP media issue In-Reply-To: References: Message-ID: <6271356323340828380@unknownmsgid> Sent from my iPad On May 24, 2012, at 5:01 PM, Nathan Downes wrote: Hi, I had previous reported an issue with poor voice quality, appearing to stem from occasion wrong timestamps coming from provider, but the end user's experience was much worse than what I could see/hear in the trace. I have finally captured an event inbound and outbound. The thing I don't understand is I thought even though FS proxied the media it didn't touch it or change anything, but it appears it is. The 2 traces are http://www.nortec.com.au/inbound.pcap.gz and http://www.nortec.com.au/outbound.pcap.gz Inbound is from my trunk provider to FS box and outbound is FS box to ATA in FTTH GPON. The event I am talking about, if both traces are open, is in the inbound one inbetween packet 8114 and 8117 the provider drops a packet or I don't receive it. In the corresponding outbound trace, between packet 8144 and 8152, it appears FS misses a whole heap of packets (.1 seconds) between 8146 and 8152 then it increases the timestamp only by 40 rather than 160 on packet 8152. This seems to not affect SIP phones themselves but causes issues with the FTTH GPON ATA. This causes a gap in the audio for the end user, and when they miss a high number of packets even though it sounds good on the inbound trace the end users experience is horrible. This trace is actually a good one, but the wrong timestamp can occur once per second, causing end user to lose 10%+ of incoming audio only. The issue only affects the audio coming from provider to FS to end user. I am chasing it up with the voice provider to try and eliminate the occasional packet loss, but if I could stop/fix FS from doing its adjustment/gap/something the end user wouldn't even notice it. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120524/4bc76f10/attachment.html From nathandownes at hotmail.com Fri May 25 06:51:19 2012 From: nathandownes at hotmail.com (Nathan Downes) Date: Fri, 25 May 2012 02:51:19 +0000 Subject: [Freeswitch-users] RTP media issue In-Reply-To: <02a201cd3a1a$2a658770$7f309650$@gmail.com> References: , <02a201cd3a1a$2a658770$7f309650$@gmail.com> Message-ID: Hi, enable-soa Set the value to "false" to diable SIP SOA from sofia to tell sofia not to touch the exchange of SDP I don't think this is related to the exchange of an SDP message.. Can you elaborate more before I try it? I can't make things worse or change things I don't understand. From: djbinter at gmail.com To: freeswitch-users at lists.freeswitch.org CC: nathan at nortec.com.au Subject: Re: [Freeswitch-users] RTP media issue Date: Fri, 25 May 2012 11:19:46 +1000 Sent from my iPad On May 24, 2012, at 5:01 PM, Nathan Downes wrote: Hi, I had previous reported an issue with poor voice quality, appearing to stem from occasion wrong timestamps coming from provider, but the end user's experience was much worse than what I could see/hear in the trace. I have finally captured an event inbound and outbound. The thing I don't understand is I thought even though FS proxied the media it didn't touch it or change anything, but it appears it is. The 2 traces are http://www.nortec.com.au/inbound.pcap.gz and http://www.nortec.com.au/outbound.pcap.gz Inbound is from my trunk provider to FS box and outbound is FS box to ATA in FTTH GPON. The event I am talking about, if both traces are open, is in the inbound one inbetween packet 8114 and 8117 the provider drops a packet or I don't receive it. In the corresponding outbound trace, between packet 8144 and 8152, it appears FS misses a whole heap of packets (.1 seconds) between 8146 and 8152 then it increases the timestamp only by 40 rather than 160 on packet 8152. This seems to not affect SIP phones themselves but causes issues with the FTTH GPON ATA. This causes a gap in the audio for the end user, and when they miss a high number of packets even though it sounds good on the inbound trace the end users experience is horrible. This trace is actually a good one, but the wrong timestamp can occur once per second, causing end user to lose 10%+ of incoming audio only. The issue only affects the audio coming from provider to FS to end user. I am chasing it up with the voice provider to try and eliminate the occasional packet loss, but if I could stop/fix FS from doing its adjustment/gap/something the end user wouldn't even notice it. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120525/85d4ab97/attachment.html From chaiyawut.so at gmail.com Fri May 25 07:27:33 2012 From: chaiyawut.so at gmail.com (Chaiyawut Sookplang) Date: Fri, 25 May 2012 10:27:33 +0700 Subject: [Freeswitch-users] Why I could not detect DTMF when using Originate command Message-ID: I want to make an automatic IVR call using originate command. I issued command "originate {ignore_early_media=true}sofia/gateway/trunk_1/0860216060 8888" and then I found that DTMF from the receiver side couldn't be detected. On the other hand, if I call to extension 8888, IVR and DTMF work fine. From govoiper at gmail.com Fri May 25 09:08:23 2012 From: govoiper at gmail.com (SamyGo) Date: Fri, 25 May 2012 10:08:23 +0500 Subject: [Freeswitch-users] Freeswitch (SBC + Media) does not know that gateway phone is picked up In-Reply-To: <1337902817.51217.YahooMailClassic@web110816.mail.gq1.yahoo.com> References: <1337902817.51217.YahooMailClassic@web110816.mail.gq1.yahoo.com> Message-ID: Hi, These are SIP traces on FreeSWITCH console, whereas you are saying and it seems that kamailio is not detecting the answering (200 OK)of the call from extension 1002. Please, can you take a sip trace..!! I see you've both kamailio and FS on same server! Please take a pcap from the linux console using the following command. #tcpdump -i any -s 0 -w new-call-trace.pcap -vvv Please be quick on turning the sip trace on and off as quickly as possible to avoid extra packets. Once done open the file in wireshark ; apply filter "sip || rtp" and then save the resulting capture in separate file. Send us the new file to analyse. One more silly question probably, I see REGISTERs coming to your FS as well and the calls to gateways are made from FS too !!, umm...just thinking what are you using kamailio for!!? Thanks BR, Sammy On Fri, May 25, 2012 at 4:40 AM, Sherif Omran wrote: > Hi all, > > here is the siptrace: To figure it out > 1- gateway called bluesip.net. It send invide using caller > number at bluesip.net > 2- This call should go to extension kb-1002. kb means go from freeswitch > port 6090 to kamailio port 5060 > 3- It should go to call extension 1002 in Kamailio > 4- Extension 1002 rings but when I reply, it does not notice I replied > > > ./fs_cli > _____ ____ ____ _ ___ > | ___/ ___| / ___| | |_ _| > | |_ \___ \ | | | | | | > | _| ___) | | |___| |___ | | > |_| |____/ \____|_____|___| > > ******************************************************* > * Anthony Minessale II, Ken Rice, * > * Michael Jerris, Travis Cross * > * FreeSWITCH (http://www.freeswitch.org) * > * Paypal Donations Appreciated: paypal at freeswitch.org * > * Brought to you by ClueCon http://www.cluecon.com/ * > ******************************************************* > > Type /help to see a list of commands > > > +OK log level [7] > freeswitch at internal> tracelevel > -ERR tracelevel Command not found! > > freeswitch at internal> sofia global siptrace on > +OK Global siptrace on > recv 1104 bytes from udp/[217.74.179.29]:5060 at 23:08:45.822962: > ------------------------------------------------------------------------ > INVITE sip:gw+trunk_1000 at 78.138.90.58:6090;transport=udp;gw=trunk_1000 > SIP/2.0 > Record-Route: > Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 > Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 > From: "+41793940965" ;tag=as00589402 > To: > Contact: > Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net > CSeq: 102 INVITE > User-Agent: blueSIP PSTN GW > Max-Forwards: 69 > Date: Thu, 24 May 2012 23:08:44 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Content-Type: application/sdp > Content-Length: 367 > P-hint: USRLOC > > v=0 > o=root 20076 20076 IN IP4 217.74.179.28 > s=session > c=IN IP4 217.74.179.28 > t=0 0 > m=audio 25626 RTP/AVP 8 0 18 111 3 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:111 G726-32/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > a=sendrecv > ------------------------------------------------------------------------ > send 494 bytes to udp/[217.74.179.29]:5060 at 23:08:45.844486: > ------------------------------------------------------------------------ > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 > Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 > Record-Route: > From: "+41793940965" ;tag=as00589402 > To: > Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net > CSeq: 102 INVITE > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Content-Length: 0 > > ------------------------------------------------------------------------ > 2012-05-24 23:08:45.840749 [NOTICE] switch_channel.c:926 New Channel > sofia/internal/+41793940965 at bluesip.net[69cd22e6-a5f5-11e1-a833-7b1f73a7ffcf] > 2012-05-24 23:08:45.840749 [DEBUG] sofia.c:5532 Channel sofia/internal/ > +41793940965 at bluesip.net entering state [received][100] > 2012-05-24 23:08:45.840749 [DEBUG] sofia.c:5543 Remote SDP: > v=0 > o=root 20076 20076 IN IP4 217.74.179.28 > s=session > c=IN IP4 217.74.179.28 > t=0 0 > m=audio 25626 RTP/AVP 8 0 18 111 3 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:111 G726-32/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [PCMA:8:8000:20:64000]/[iLBC:97:8000:30:13330] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [PCMU:0:8000:20:64000]/[iLBC:97:8000:30:13330] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [G729:18:8000:20:8000]/[iLBC:97:8000:30:13330] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [G726-32:111:8000:20:0]/[iLBC:97:8000:30:13330] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [GSM:3:8000:20:13200]/[iLBC:97:8000:30:13330] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [telephone-event:101:8000:20:0]/[iLBC:97:8000:30:13330] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf > send/recv payload to 101 > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [PCMA:8:8000:20:64000]/[G7221:115:32000:20:48000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [PCMU:0:8000:20:64000]/[G7221:115:32000:20:48000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [G729:18:8000:20:8000]/[G7221:115:32000:20:48000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [G726-32:111:8000:20:0]/[G7221:115:32000:20:48000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [GSM:3:8000:20:13200]/[G7221:115:32000:20:48000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [telephone-event:101:8000:20:0]/[G7221:115:32000:20:48000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf > send/recv payload to 101 > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [PCMA:8:8000:20:64000]/[G7221:107:16000:20:32000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [PCMU:0:8000:20:64000]/[G7221:107:16000:20:32000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [G729:18:8000:20:8000]/[G7221:107:16000:20:32000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [G726-32:111:8000:20:0]/[G7221:107:16000:20:32000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [GSM:3:8000:20:13200]/[G7221:107:16000:20:32000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [telephone-event:101:8000:20:0]/[G7221:107:16000:20:32000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf > send/recv payload to 101 > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [PCMA:8:8000:20:64000]/[G722:9:8000:20:64000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [G729:18:8000:20:8000]/[G722:9:8000:20:64000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [G726-32:111:8000:20:0]/[G722:9:8000:20:64000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [GSM:3:8000:20:13200]/[G722:9:8000:20:64000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [telephone-event:101:8000:20:0]/[G722:9:8000:20:64000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf > send/recv payload to 101 > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [PCMA:8:8000:20:64000]/[PCMU:0:8000:20:64000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:2996 Set Codec > sofia/internal/+41793940965 at bluesip.net PCMU/8000 20 ms 160 samples 64000 > bits > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_codec.c:111 sofia/internal/ > +41793940965 at bluesip.net Original read codec set to PCMU:0 > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf > send/recv payload to 101 > 2012-05-24 23:08:45.840749 [DEBUG] sofia.c:5757 (sofia/internal/ > +41793940965 at bluesip.net) State Change CS_NEW -> CS_INIT > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_INIT > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:401 > (sofia/internal/+41793940965 at bluesip.net) State INIT > 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:85 sofia/internal/ > +41793940965 at bluesip.net SOFIA INIT > 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:125 (sofia/internal/ > +41793940965 at bluesip.net) State Change CS_INIT -> CS_ROUTING > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:401 > (sofia/internal/+41793940965 at bluesip.net) State INIT going to sleep > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_ROUTING > 2012-05-24 23:08:45.840749 [DEBUG] switch_channel.c:1886 (sofia/internal/ > +41793940965 at bluesip.net) Callstate Change DOWN -> RINGING > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 > (sofia/internal/+41793940965 at bluesip.net) State ROUTING > 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:148 sofia/internal/ > +41793940965 at bluesip.net SOFIA ROUTING > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:104 > sofia/internal/+41793940965 at bluesip.net Standard ROUTING > 2012-05-24 23:08:45.840749 [INFO] mod_dialplan_xml.c:485 Processing > +41793940965 <+41793940965>->kb-1002 in context public > Dialplan: sofia/internal/+41793940965 at bluesip.net parsing > [public->from_kamailio] continue=false > Dialplan: sofia/internal/+41793940965 at bluesip.net Regex (PASS) > [from_kamailio] destination_number(kb-1002) =~ /^(.+)$/ break=on-false > Dialplan: sofia/internal/+41793940965 at bluesip.net Action transfer(kb-1002 > XML default) > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:154 > (sofia/internal/+41793940965 at bluesip.net) State Change CS_ROUTING -> > CS_EXECUTE > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 > (sofia/internal/+41793940965 at bluesip.net) State ROUTING going to sleep > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_EXECUTE > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:417 > (sofia/internal/+41793940965 at bluesip.net) State EXECUTE > 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:241 sofia/internal/ > +41793940965 at bluesip.net SOFIA EXECUTE > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:192 > sofia/internal/+41793940965 at bluesip.net Standard EXECUTE > EXECUTE sofia/internal/+41793940965 at bluesip.net transfer(kb-1002 XML > default) > 2012-05-24 23:08:45.840749 [DEBUG] switch_ivr.c:1711 (sofia/internal/ > +41793940965 at bluesip.net) State Change CS_EXECUTE -> CS_ROUTING > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:731 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:08:45.840749 [NOTICE] switch_ivr.c:1717 Transfer > sofia/internal/+41793940965 at bluesip.net to XML[kb-1002 at default] > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:417 > (sofia/internal/+41793940965 at bluesip.net) State EXECUTE going to sleep > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_ROUTING > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 > (sofia/internal/+41793940965 at bluesip.net) State ROUTING > 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:148 sofia/internal/ > +41793940965 at bluesip.net SOFIA ROUTING > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:104 > sofia/internal/+41793940965 at bluesip.net Standard ROUTING > 2012-05-24 23:08:45.840749 [INFO] mod_dialplan_xml.c:485 Processing > +41793940965 <+41793940965>->kb-1002 in context default > Dialplan: sofia/internal/+41793940965 at bluesip.net parsing [default->vbox] > continue=false > Dialplan: sofia/internal/+41793940965 at bluesip.net Regex (FAIL) [vbox] > destination_number(kb-1002) =~ /^vb-([0-9][0-9][0-9])$/ break=on-false > Dialplan: sofia/internal/+41793940965 at bluesip.net parsing > [default->vmenu] continue=false > Dialplan: sofia/internal/+41793940965 at bluesip.net Regex (FAIL) [vmenu] > destination_number(kb-1002) =~ /^vm-([0-9][0-9][0-9][0-9])$/ break=on-false > Dialplan: sofia/internal/+41793940965 at bluesip.net parsing > [default->kbridge] continue=false > Dialplan: sofia/internal/+41793940965 at bluesip.net Regex (PASS) [kbridge] > destination_number(kb-1002) =~ /^kb-(.+)$/ break=on-false > Dialplan: sofia/internal/+41793940965 at bluesip.net Action > set(proxy_media=true) > Dialplan: sofia/internal/+41793940965 at bluesip.net Action > set(call_timeout=50) > Dialplan: sofia/internal/+41793940965 at bluesip.net Action > set(continue_on_fail=true) > Dialplan: sofia/internal/+41793940965 at bluesip.net Action > set(hangup_after_bridge=true) > Dialplan: sofia/internal/+41793940965 at bluesip.net Action > set(sip_invite_domain=78.138.90.58) > Dialplan: sofia/internal/+41793940965 at bluesip.net Action > export(sip_contact_user=ufs) > Dialplan: sofia/internal/+41793940965 at bluesip.net Action bridge(sofia/ > 78.138.90.58/1002 at 78.138.90.58:5060) > Dialplan: sofia/internal/+41793940965 at bluesip.net Action answer() > Dialplan: sofia/internal/+41793940965 at bluesip.net Action > voicemail(default ${domain_name} 1002) > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:154 > (sofia/internal/+41793940965 at bluesip.net) State Change CS_ROUTING -> > CS_EXECUTE > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 > (sofia/internal/+41793940965 at bluesip.net) State ROUTING going to sleep > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_EXECUTE > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:417 > (sofia/internal/+41793940965 at bluesip.net) State EXECUTE > 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:241 sofia/internal/ > +41793940965 at bluesip.net SOFIA EXECUTE > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:192 > sofia/internal/+41793940965 at bluesip.net Standard EXECUTE > EXECUTE sofia/internal/+41793940965 at bluesip.net set(proxy_media=true) > 2012-05-24 23:08:45.840749 [DEBUG] mod_dptools.c:1281 sofia/internal/ > +41793940965 at bluesip.net SET [proxy_media]=[true] > EXECUTE sofia/internal/+41793940965 at bluesip.net set(call_timeout=50) > 2012-05-24 23:08:45.840749 [DEBUG] mod_dptools.c:1281 sofia/internal/ > +41793940965 at bluesip.net SET [call_timeout]=[50] > EXECUTE sofia/internal/+41793940965 at bluesip.net set(continue_on_fail=true) > 2012-05-24 23:08:45.840749 [DEBUG] mod_dptools.c:1281 sofia/internal/ > +41793940965 at bluesip.net SET [continue_on_fail]=[true] > EXECUTE sofia/internal/+41793940965 at bluesip.netset(hangup_after_bridge=true) > 2012-05-24 23:08:45.840749 [DEBUG] mod_dptools.c:1281 sofia/internal/ > +41793940965 at bluesip.net SET [hangup_after_bridge]=[true] > EXECUTE sofia/internal/+41793940965 at bluesip.netset(sip_invite_domain=78.138.90.58) > 2012-05-24 23:08:45.840749 [DEBUG] mod_dptools.c:1281 sofia/internal/ > +41793940965 at bluesip.net SET [sip_invite_domain]=[78.138.90.58] > EXECUTE sofia/internal/+41793940965 at bluesip.netexport(sip_contact_user=ufs) > 2012-05-24 23:08:45.840749 [DEBUG] switch_channel.c:1093 EXPORT > (export_vars) [sip_contact_user]=[ufs] > EXECUTE sofia/internal/+41793940965 at bluesip.net bridge(sofia/ > 78.138.90.58/1002 at 78.138.90.58:5060) > 2012-05-24 23:08:45.840749 [DEBUG] switch_channel.c:1047 sofia/internal/ > +41793940965 at bluesip.net EXPORTING[export_vars] [sip_contact_user]=[ufs] > to event > 2012-05-24 23:08:45.840749 [DEBUG] switch_ivr_originate.c:1884 Parsing > global variables > 2012-05-24 23:08:45.840749 [NOTICE] switch_channel.c:926 New Channel > sofia/internal/1002 at 78.138.90.58:5060[69ce1f0c-a5f5-11e1-a838-7b1f73a7ffcf] > 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:4691 (sofia/internal/ > 1002 at 78.138.90.58:5060) State Change CS_NEW -> CS_INIT > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/1002 at 78.138.90.58:5060) Running State Change CS_INIT > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:401 > (sofia/internal/1002 at 78.138.90.58:5060) State INIT > 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:85 sofia/internal/ > 1002 at 78.138.90.58:5060 SOFIA INIT > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:1871 sofia/internal/ > 1002 at 78.138.90.58:5060 Patched SDP > --- > v=0 > o=root 20076 20076 IN IP4 217.74.179.28 > s=session > c=IN IP4 217.74.179.28 > t=0 0 > m=audio 25626 RTP/AVP 8 0 18 111 3 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:111 G726-32/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > > +++ > v=0 > o=FreeSWITCH 1629510322 1629510323 IN IP4 78.138.90.58 > s=FreeSWITCH > c=IN IP4 78.138.90.58 > t=0 0 > m=audio 31178 RTP/AVP 8 0 18 111 3 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:111 G726-32/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > > send 1340 bytes to udp/[78.138.90.58]:5060 at 23:08:45.852280: > ------------------------------------------------------------------------ > INVITE sip:1002 at 78.138.90.58:5060 SIP/2.0 > Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKapDgFeegKyHXK > Max-Forwards: 67 > From: "+41793940965" ;tag=tgNrSS1jQFS8F > To: > Call-ID: 412c8312-2098-1230-8985-00163e6bb553 > CSeq: 28614398 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > 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: 372 > P-hint: USRLOC > X-FS-Support: update_display,send_info > Remote-Party-ID: "+41793940965" >;party=calling;screen=yes;privacy=off > > v=0 > o=FreeSWITCH 1629510322 1629510323 IN IP4 78.138.90.58 > s=FreeSWITCH > c=IN IP4 78.138.90.58 > t=0 0 > m=audio 31178 RTP/AVP 8 0 18 111 3 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:111 G726-32/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > ------------------------------------------------------------------------ > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:125 (sofia/internal/ > 1002 at 78.138.90.58:5060) State Change CS_INIT -> CS_ROUTING > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:401 > (sofia/internal/1002 at 78.138.90.58:5060) State INIT going to sleep > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/1002 at 78.138.90.58:5060) Running State Change CS_ROUTING > 2012-05-24 23:08:45.840749 [DEBUG] switch_channel.c:1886 (sofia/internal/ > 1002 at 78.138.90.58:5060) Callstate Change DOWN -> RINGING > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 > (sofia/internal/1002 at 78.138.90.58:5060) State ROUTING > 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:148 sofia/internal/ > 1002 at 78.138.90.58:5060 SOFIA ROUTING > 2012-05-24 23:08:45.840749 [DEBUG] switch_ivr_originate.c:66 > (sofia/internal/1002 at 78.138.90.58:5060) State Change CS_ROUTING -> > CS_CONSUME_MEDIA > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 > (sofia/internal/1002 at 78.138.90.58:5060) State ROUTING going to sleep > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/1002 at 78.138.90.58:5060) Running State Change > CS_CONSUME_MEDIA > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:429 > (sofia/internal/1002 at 78.138.90.58:5060) State CONSUME_MEDIA > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:429 > (sofia/internal/1002 at 78.138.90.58:5060) State CONSUME_MEDIA going to sleep > 2012-05-24 23:08:45.840749 [DEBUG] sofia.c:5532 Channel sofia/internal/ > 1002 at 78.138.90.58:5060 entering state [calling][0] > recv 365 bytes from udp/[78.138.90.58]:5060 at 23:08:45.964309: > ------------------------------------------------------------------------ > SIP/2.0 100 trying -- your call is important to us > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;branch=z9hG4bKapDgFeegKyHXK > From: "+41793940965" ;tag=tgNrSS1jQFS8F > To: > Call-ID: 412c8312-2098-1230-8985-00163e6bb553 > CSeq: 28614398 INVITE > Server: kamailio (3.1.5 (x86_64/linux)) > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 496 bytes from udp/[78.138.90.58]:5060 at 23:08:47.414766: > ------------------------------------------------------------------------ > SIP/2.0 180 Ringing > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;branch=z9hG4bKapDgFeegKyHXK > Record-Route: > Call-ID: 412c8312-2098-1230-8985-00163e6bb553 > From: "+41793940965" ;tag=tgNrSS1jQFS8F > To: ;tag=qCSaB2YJOAsAr2dJ-RWNMhPoDdn1RFvs > CSeq: 28614398 INVITE > Contact: "Mama" > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, REGISTER > Content-Length: 0 > > ------------------------------------------------------------------------ > 2012-05-24 23:08:47.400741 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:47.400741 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:47.400741 [DEBUG] sofia.c:5532 Channel sofia/internal/ > 1002 at 78.138.90.58:5060 entering state [proceeding][180] > 2012-05-24 23:08:47.400741 [NOTICE] sofia.c:5624 Ring-Ready sofia/internal/ > 1002 at 78.138.90.58:5060! > send 995 bytes to udp/[217.74.179.29]:5060 at 23:08:47.416538: > ------------------------------------------------------------------------ > SIP/2.0 180 Ringing > Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 > Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 > Record-Route: > From: "+41793940965" ;tag=as00589402 > To: ;tag=S7UZQygFt62Nm > Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net > CSeq: 102 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Accept: application/sdp > 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-Length: 0 > Remote-Party-ID: "Outbound Call" >;party=calling;privacy=off;screen=no > > ------------------------------------------------------------------------ > 2012-05-24 23:08:47.400741 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:08:47.400741 [NOTICE] mod_sofia.c:2514 Ring-Ready > sofia/internal/+41793940965 at bluesip.net! > 2012-05-24 23:08:47.400741 [DEBUG] sofia.c:5532 Channel sofia/internal/ > +41793940965 at bluesip.net entering state [early][180] > 2012-05-24 23:08:47.400741 [DEBUG] switch_core_session.c:731 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:08:47.400741 [NOTICE] switch_ivr_originate.c:483 Ring Ready > sofia/internal/+41793940965 at bluesip.net! > recv 496 bytes from udp/[78.138.90.58]:5060 at 23:08:47.804967: > ------------------------------------------------------------------------ > SIP/2.0 180 Ringing > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;branch=z9hG4bKapDgFeegKyHXK > Record-Route: > From: "+41793940965" ;tag=tgNrSS1jQFS8F > To: ;tag=549D2DD03BBA7C67 > Call-ID: 412c8312-2098-1230-8985-00163e6bb553 > CSeq: 28614398 INVITE > Contact: > User-Agent: AVM FRITZ!Box Fon WLAN 7170 29.04.80 (Jan 27 2010) > Content-Length: 0 > > ------------------------------------------------------------------------ > 2012-05-24 23:08:47.800746 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:47.800746 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:47.800746 [DEBUG] sofia.c:5532 Channel sofia/internal/ > 1002 at 78.138.90.58:5060 entering state [proceeding][180] > recv 496 bytes from udp/[78.138.90.58]:5060 at 23:08:48.182098: > ------------------------------------------------------------------------ > SIP/2.0 180 Ringing > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;branch=z9hG4bKapDgFeegKyHXK > Record-Route: > Call-ID: 412c8312-2098-1230-8985-00163e6bb553 > From: "+41793940965" ;tag=tgNrSS1jQFS8F > To: ;tag=qCSaB2YJOAsAr2dJ-RWNMhPoDdn1RFvs > CSeq: 28614398 INVITE > Contact: "Mama" > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, REGISTER > Content-Length: 0 > > ------------------------------------------------------------------------ > 2012-05-24 23:08:48.180741 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:48.180741 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:48.180741 [DEBUG] sofia.c:5532 Channel sofia/internal/ > 1002 at 78.138.90.58:5060 entering state [proceeding][180] > recv 496 bytes from udp/[78.138.90.58]:5060 at 23:08:48.417403: > ------------------------------------------------------------------------ > SIP/2.0 180 Ringing > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;branch=z9hG4bKapDgFeegKyHXK > Record-Route: > From: "+41793940965" ;tag=tgNrSS1jQFS8F > To: ;tag=549D2DD03BBA7C67 > Call-ID: 412c8312-2098-1230-8985-00163e6bb553 > CSeq: 28614398 INVITE > Contact: > User-Agent: AVM FRITZ!Box Fon WLAN 7170 29.04.80 (Jan 27 2010) > Content-Length: 0 > > ------------------------------------------------------------------------ > 2012-05-24 23:08:48.400744 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:48.400744 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:48.400744 [DEBUG] sofia.c:5532 Channel sofia/internal/ > 1002 at 78.138.90.58:5060 entering state [proceeding][180] > recv 496 bytes from udp/[78.138.90.58]:5060 at 23:08:49.292599: > ------------------------------------------------------------------------ > SIP/2.0 180 Ringing > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;branch=z9hG4bKapDgFeegKyHXK > Record-Route: > From: "+41793940965" ;tag=tgNrSS1jQFS8F > To: ;tag=549D2DD03BBA7C67 > Call-ID: 412c8312-2098-1230-8985-00163e6bb553 > CSeq: 28614398 INVITE > Contact: > User-Agent: AVM FRITZ!Box Fon WLAN 7170 29.04.80 (Jan 27 2010) > Content-Length: 0 > > ------------------------------------------------------------------------ > 2012-05-24 23:08:49.280742 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:49.280742 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:49.280742 [DEBUG] sofia.c:5532 Channel sofia/internal/ > 1002 at 78.138.90.58:5060 entering state [proceeding][180] > recv 478 bytes from udp/[217.74.179.29]:5060 at 23:09:15.066012: > ------------------------------------------------------------------------ > CANCEL sip:gw+trunk_1000 at 78.138.90.58:6090;transport=udp;gw=trunk_1000 > SIP/2.0 > Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 > Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 > From: "+41793940965" ;tag=as00589402 > To: > Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net > CSeq: 102 CANCEL > User-Agent: blueSIP PSTN GW > Max-Forwards: 69 > Content-Length: 0 > P-hint: USRLOC > > ------------------------------------------------------------------------ > send 369 bytes to udp/[217.74.179.29]:5060 at 23:09:15.066176: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 > Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 > From: "+41793940965" ;tag=as00589402 > To: ;tag=S7UZQygFt62Nm > Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net > CSeq: 102 CANCEL > Content-Length: 0 > > ------------------------------------------------------------------------ > send 766 bytes to udp/[217.74.179.29]:5060 at 23:09:15.066289: > ------------------------------------------------------------------------ > SIP/2.0 487 Request Terminated > Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 > Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 > From: "+41793940965" ;tag=as00589402 > To: ;tag=S7UZQygFt62Nm > Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net > CSeq: 102 INVITE > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > 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-Length: 0 > > ------------------------------------------------------------------------ > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:09:15.060742 [DEBUG] sofia.c:5532 Channel sofia/internal/ > +41793940965 at bluesip.net entering state [terminated][487] > 2012-05-24 23:09:15.060742 [DEBUG] switch_channel.c:2848 (sofia/internal/ > +41793940965 at bluesip.net) Callstate Change RINGING -> HANGUP > 2012-05-24 23:09:15.060742 [NOTICE] sofia.c:6301 Hangup sofia/internal/ > +41793940965 at bluesip.net [CS_EXECUTE] [ORIGINATOR_CANCEL] > 2012-05-24 23:09:15.060742 [DEBUG] switch_channel.c:2871 Send signal > sofia/internal/+41793940965 at bluesip.net [KILL] > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:09:15.060742 [DEBUG] switch_channel.c:2848 (sofia/internal/ > 1002 at 78.138.90.58:5060) Callstate Change RINGING -> HANGUP > 2012-05-24 23:09:15.060742 [NOTICE] switch_ivr_originate.c:3182 Hangup > sofia/internal/1002 at 78.138.90.58:5060 [CS_CONSUME_MEDIA] > [ORIGINATOR_CANCEL] > 2012-05-24 23:09:15.060742 [DEBUG] switch_channel.c:2871 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [KILL] > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/1002 at 78.138.90.58:5060) Running State Change CS_HANGUP > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:602 > (sofia/internal/1002 at 78.138.90.58:5060) State HANGUP > 2012-05-24 23:09:15.060742 [DEBUG] mod_sofia.c:469 Channel sofia/internal/ > 1002 at 78.138.90.58:5060 hanging up, cause: ORIGINATOR_CANCEL > 2012-05-24 23:09:15.060742 [DEBUG] mod_sofia.c:523 Sending CANCEL to > sofia/internal/1002 at 78.138.90.58:5060 > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:47 > sofia/internal/1002 at 78.138.90.58:5060 Standard HANGUP, cause: > ORIGINATOR_CANCEL > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:602 > (sofia/internal/1002 at 78.138.90.58:5060) State HANGUP going to sleep > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:393 > (sofia/internal/1002 at 78.138.90.58:5060) State Change CS_HANGUP -> > CS_REPORTING > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/1002 at 78.138.90.58:5060) Running State Change CS_REPORTING > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:662 > (sofia/internal/1002 at 78.138.90.58:5060) State REPORTING > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:79 > sofia/internal/1002 at 78.138.90.58:5060 Standard REPORTING, cause: > ORIGINATOR_CANCEL > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:662 > (sofia/internal/1002 at 78.138.90.58:5060) State REPORTING going to sleep > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:387 > (sofia/internal/1002 at 78.138.90.58:5060) State Change CS_REPORTING -> > CS_DESTROY > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1382 Session 9 > (sofia/internal/1002 at 78.138.90.58:5060) Locked, Waiting on external > entities > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:09:15.060742 [DEBUG] switch_ivr_originate.c:3358 Originate > Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL] > 2012-05-24 23:09:15.060742 [NOTICE] switch_core_session.c:1400 Session 9 > (sofia/internal/1002 at 78.138.90.58:5060) Ended > 2012-05-24 23:09:15.060742 [NOTICE] switch_core_session.c:1402 Close > Channel sofia/internal/1002 at 78.138.90.58:5060 [CS_DESTROY] > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:491 > (sofia/internal/1002 at 78.138.90.58:5060) Callstate Change HANGUP -> DOWN > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:494 > (sofia/internal/1002 at 78.138.90.58:5060) Running State Change CS_DESTROY > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:504 > (sofia/internal/1002 at 78.138.90.58:5060) State DESTROY > 2012-05-24 23:09:15.060742 [DEBUG] mod_sofia.c:374 sofia/internal/ > 1002 at 78.138.90.58:5060 SOFIA DESTROY > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:86 > sofia/internal/1002 at 78.138.90.58:5060 Standard DESTROY > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:504 > (sofia/internal/1002 at 78.138.90.58:5060) State DESTROY going to sleep > 2012-05-24 23:09:15.060742 [INFO] mod_dptools.c:2922 Originate Failed. > Cause: ORIGINATOR_CANCEL > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:2287 > sofia/internal/+41793940965 at bluesip.net skip receive message > [APPLICATION_EXEC_COMPLETE] (channel is hungup already) > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:417 > (sofia/internal/+41793940965 at bluesip.net) State EXECUTE going to sleep > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_HANGUP > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:602 > (sofia/internal/+41793940965 at bluesip.net) State HANGUP > 2012-05-24 23:09:15.060742 [DEBUG] mod_sofia.c:469 Channel sofia/internal/ > +41793940965 at bluesip.net hanging up, cause: ORIGINATOR_CANCEL > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:47 > sofia/internal/+41793940965 at bluesip.net Standard HANGUP, cause: > ORIGINATOR_CANCEL > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:602 > (sofia/internal/+41793940965 at bluesip.net) State HANGUP going to sleep > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:393 > (sofia/internal/+41793940965 at bluesip.net) State Change CS_HANGUP -> > CS_REPORTING > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/+41793940965 at bluesip.net) Running State Change > CS_REPORTING > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:662 > (sofia/internal/+41793940965 at bluesip.net) State REPORTING > send 383 bytes to udp/[78.138.90.58]:5060 at 23:09:15.087695: > ------------------------------------------------------------------------ > CANCEL sip:1002 at 78.138.90.58:5060 SIP/2.0 > Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKapDgFeegKyHXK > Max-Forwards: 67 > From: "+41793940965" ;tag=tgNrSS1jQFS8F > To: > Call-ID: 412c8312-2098-1230-8985-00163e6bb553 > CSeq: 28614398 CANCEL > Reason: FreeSWITCH;cause=487;text="ORIGINATOR_CANCEL" > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 408 bytes from udp/[217.74.179.29]:5060 at 23:09:15.094670: > ------------------------------------------------------------------------ > ACK sip:gw+trunk_1000 at 78.138.90.58:6090;transport=udp;gw=trunk_1000 > SIP/2.0 > Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 > From: "+41793940965" ;tag=as00589402 > Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net > To: ;tag=S7UZQygFt62Nm > CSeq: 102 ACK > User-Agent: Sip EXpress router(0.9.7 (i386/linux)) > Content-Length: 0 > > ------------------------------------------------------------------------ > 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:79 > sofia/internal/+41793940965 at bluesip.net Standard REPORTING, cause: > ORIGINATOR_CANCEL > 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:662 > (sofia/internal/+41793940965 at bluesip.net) State REPORTING going to sleep > 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:387 > (sofia/internal/+41793940965 at bluesip.net) State Change CS_REPORTING -> > CS_DESTROY > 2012-05-24 23:09:15.120807 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:09:15.120807 [DEBUG] switch_core_session.c:1382 Session 8 > (sofia/internal/+41793940965 at bluesip.net) Locked, Waiting on external > entities > 2012-05-24 23:09:15.120807 [NOTICE] switch_core_session.c:1400 Session 8 > (sofia/internal/+41793940965 at bluesip.net) Ended > 2012-05-24 23:09:15.120807 [NOTICE] switch_core_session.c:1402 Close > Channel sofia/internal/+41793940965 at bluesip.net [CS_DESTROY] > 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:491 > (sofia/internal/+41793940965 at bluesip.net) Callstate Change HANGUP -> DOWN > 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:494 > (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_DESTROY > 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:504 > (sofia/internal/+41793940965 at bluesip.net) State DESTROY > 2012-05-24 23:09:15.120807 [DEBUG] mod_sofia.c:374 sofia/internal/ > +41793940965 at bluesip.net SOFIA DESTROY > 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:86 > sofia/internal/+41793940965 at bluesip.net Standard DESTROY > 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:504 > (sofia/internal/+41793940965 at bluesip.net) State DESTROY going to sleep > recv 378 bytes from udp/[78.138.90.58]:5060 at 23:09:15.135235: > ------------------------------------------------------------------------ > SIP/2.0 200 canceling > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;branch=z9hG4bKapDgFeegKyHXK > From: "+41793940965" ;tag=tgNrSS1jQFS8F > To: >;tag=a6a1c5f60faecf035a1ae5b6e96e979a-8c5b > Call-ID: 412c8312-2098-1230-8985-00163e6bb553 > CSeq: 28614398 CANCEL > Server: kamailio (3.1.5 (x86_64/linux)) > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 446 bytes from udp/[78.138.90.58]:5060 at 23:09:16.419308: > ------------------------------------------------------------------------ > SIP/2.0 487 Request Terminated > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;branch=z9hG4bKapDgFeegKyHXK > Record-Route: > Call-ID: 412c8312-2098-1230-8985-00163e6bb553 > From: "+41793940965" ;tag=tgNrSS1jQFS8F > To: ;tag=qCSaB2YJOAsAr2dJ-RWNMhPoDdn1RFvs > CSeq: 28614398 INVITE > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, REGISTER > Content-Length: 0 > > ------------------------------------------------------------------------ > send 354 bytes to udp/[78.138.90.58]:5060 at 23:09:16.419543: > ------------------------------------------------------------------------ > ACK sip:1002 at 78.138.90.58:5060 SIP/2.0 > Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKapDgFeegKyHXK > Max-Forwards: 67 > From: "+41793940965" ;tag=tgNrSS1jQFS8F > To: ;tag=qCSaB2YJOAsAr2dJ-RWNMhPoDdn1RFvs > Call-ID: 412c8312-2098-1230-8985-00163e6bb553 > CSeq: 28614398 ACK > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 564 bytes from udp/[41.34.123.243]:58881 at 23:13:13.612418: > ------------------------------------------------------------------------ > REGISTER sip:sip.pcfone.com:6090 SIP/2.0 > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport;branch=z9hG4bKPjHQIpv.ACHGbXD4IHrMr72LwZ8kYdlu-J > Max-Forwards: 70 > From: "Sherif 1003" >;tag=wOP.FVT4Crk8O5E.bB7BWUhNO6QG91IG > To: "Sherif 1003" > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60926 REGISTER > User-Agent: Telephone 1.0.2 > Contact: "Sherif 1003" > Expires: 300 > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, > REFER, MESSAGE, OPTIONS > Content-Length: 0 > > ------------------------------------------------------------------------ > send 744 bytes to udp/[41.34.123.243]:58881 at 23:13:13.613152: > ------------------------------------------------------------------------ > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport=58881;branch=z9hG4bKPjHQIpv.ACHGbXD4IHrMr72LwZ8kYdlu-J;received=41.34.123.243 > From: "Sherif 1003" >;tag=wOP.FVT4Crk8O5E.bB7BWUhNO6QG91IG > To: "Sherif 1003" ;tag=USeHUmjpmrFUB > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60926 REGISTER > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > WWW-Authenticate: Digest realm="sip.pcfone.com", > nonce="09674764-a5f6-11e1-a83c-7b1f73a7ffcf", algorithm=MD5, qop="auth" > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 830 bytes from udp/[41.34.123.243]:58881 at 23:13:13.847746: > ------------------------------------------------------------------------ > REGISTER sip:sip.pcfone.com:6090 SIP/2.0 > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport;branch=z9hG4bKPjFr7rIAdAJPRQd2GmrMrVib3KT3Rup0q4 > Max-Forwards: 70 > From: "Sherif 1003" >;tag=wOP.FVT4Crk8O5E.bB7BWUhNO6QG91IG > To: "Sherif 1003" > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60927 REGISTER > User-Agent: Telephone 1.0.2 > Contact: "Sherif 1003" > Expires: 300 > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, > REFER, MESSAGE, OPTIONS > Authorization: Digest username="1002", realm="sip.pcfone.com", > nonce="09674764-a5f6-11e1-a83c-7b1f73a7ffcf", uri="sip:sip.pcfone.com:6090", > response="3bc0bb82a803ded3641704c4eb78d58f", algorithm=MD5, > cnonce="sc3AJfhuev18Bs-L-6tORmI.RXlxnPOz", qop=auth, nc=00000001 > Content-Length: 0 > > ------------------------------------------------------------------------ > send 706 bytes to udp/[41.34.123.243]:58881 at 23:13:13.850197: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport=58881;branch=z9hG4bKPjFr7rIAdAJPRQd2GmrMrVib3KT3Rup0q4;received=41.34.123.243 > From: "Sherif 1003" >;tag=wOP.FVT4Crk8O5E.bB7BWUhNO6QG91IG > To: "Sherif 1003" ;tag=v279vF3SH15DQ > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60927 REGISTER > Contact: ;expires=300 > Date: Thu, 24 May 2012 23:13:13 GMT > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Content-Length: 0 > > ------------------------------------------------------------------------ > send 989 bytes to udp/[41.34.123.243]:58881 at 23:13:13.888717: > ------------------------------------------------------------------------ > NOTIFY sip:1002 at 192.168.178.26:58881;ob SIP/2.0 > Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKBZ68g9yKg77FF > Route: ;ob > Max-Forwards: 70 > From: ;tag=XB12yamXeav0j > To: > Call-ID: e0efa252-2098-1230-8985-00163e6bb553 > CSeq: 28614532 NOTIFY > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > 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=noresource > Content-Type: application/simple-message-summary > Content-Length: 91 > > Messages-Waiting: yes > Message-Account: sip:1002 at 78.138.90.58 > Voice-Message: 1/0 (0/0) > > ------------------------------------------------------------------------ > recv 308 bytes from udp/[41.34.123.243]:58881 at 23:13:14.298850: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;received=78.138.90.58;branch=z9hG4bKBZ68g9yKg77FF > Call-ID: e0efa252-2098-1230-8985-00163e6bb553 > From: ;tag=XB12yamXeav0j > To: ;tag=z9hG4bKBZ68g9yKg77FF > CSeq: 28614532 NOTIFY > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 564 bytes from udp/[41.34.123.243]:58881 at 23:18:09.067232: > ------------------------------------------------------------------------ > REGISTER sip:sip.pcfone.com:6090 SIP/2.0 > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport;branch=z9hG4bKPjABTslmZIhiHYRtLqQn2erwmX19MVYHFC > Max-Forwards: 70 > From: "Sherif 1003" >;tag=JBSXL5WZLWbLzpvRFoTSP5x8qQslI1ST > To: "Sherif 1003" > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60928 REGISTER > User-Agent: Telephone 1.0.2 > Contact: "Sherif 1003" > Expires: 300 > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, > REFER, MESSAGE, OPTIONS > Content-Length: 0 > > ------------------------------------------------------------------------ > send 744 bytes to udp/[41.34.123.243]:58881 at 23:18:09.068121: > ------------------------------------------------------------------------ > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport=58881;branch=z9hG4bKPjABTslmZIhiHYRtLqQn2erwmX19MVYHFC;received=41.34.123.243 > From: "Sherif 1003" >;tag=JBSXL5WZLWbLzpvRFoTSP5x8qQslI1ST > To: "Sherif 1003" ;tag=ymtU0540BKjKe > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60928 REGISTER > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > WWW-Authenticate: Digest realm="sip.pcfone.com", > nonce="b9821df4-a5f6-11e1-a83d-7b1f73a7ffcf", algorithm=MD5, qop="auth" > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 830 bytes from udp/[41.34.123.243]:58881 at 23:18:09.302944: > ------------------------------------------------------------------------ > REGISTER sip:sip.pcfone.com:6090 SIP/2.0 > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport;branch=z9hG4bKPjRgtztcy63vMT97Jp5pH9n311li5EFeY- > Max-Forwards: 70 > From: "Sherif 1003" >;tag=JBSXL5WZLWbLzpvRFoTSP5x8qQslI1ST > To: "Sherif 1003" > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60929 REGISTER > User-Agent: Telephone 1.0.2 > Contact: "Sherif 1003" > Expires: 300 > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, > REFER, MESSAGE, OPTIONS > Authorization: Digest username="1002", realm="sip.pcfone.com", > nonce="b9821df4-a5f6-11e1-a83d-7b1f73a7ffcf", uri="sip:sip.pcfone.com:6090", > response="519c6301816b51f3bbb98d97a347e92b", algorithm=MD5, > cnonce="sc3AJfhuev18Bs-L-6tORmI.RXlxnPOz", qop=auth, nc=00000001 > Content-Length: 0 > > ------------------------------------------------------------------------ > send 706 bytes to udp/[41.34.123.243]:58881 at 23:18:09.305217: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport=58881;branch=z9hG4bKPjRgtztcy63vMT97Jp5pH9n311li5EFeY-;received=41.34.123.243 > From: "Sherif 1003" >;tag=JBSXL5WZLWbLzpvRFoTSP5x8qQslI1ST > To: "Sherif 1003" ;tag=ZXKm20N48U85S > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60929 REGISTER > Contact: ;expires=300 > Date: Thu, 24 May 2012 23:18:09 GMT > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Content-Length: 0 > > ------------------------------------------------------------------------ > send 989 bytes to udp/[41.34.123.243]:58881 at 23:18:09.379987: > ------------------------------------------------------------------------ > NOTIFY sip:1002 at 192.168.178.26:58881;ob SIP/2.0 > Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKc8Z1j4FQDgy2a > Route: ;ob > Max-Forwards: 70 > From: ;tag=06cD4U6754yrN > To: > Call-ID: 91100602-2099-1230-8985-00163e6bb553 > CSeq: 28614680 NOTIFY > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > 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=noresource > Content-Type: application/simple-message-summary > Content-Length: 91 > > Messages-Waiting: yes > Message-Account: sip:1002 at 78.138.90.58 > Voice-Message: 1/0 (0/0) > > ------------------------------------------------------------------------ > recv 308 bytes from udp/[41.34.123.243]:58881 at 23:18:09.773629: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;received=78.138.90.58;branch=z9hG4bKc8Z1j4FQDgy2a > Call-ID: 91100602-2099-1230-8985-00163e6bb553 > From: ;tag=06cD4U6754yrN > To: ;tag=z9hG4bKc8Z1j4FQDgy2a > CSeq: 28614680 NOTIFY > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 564 bytes from udp/[41.34.123.243]:58881 at 23:23:04.522455: > ------------------------------------------------------------------------ > REGISTER sip:sip.pcfone.com:6090 SIP/2.0 > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport;branch=z9hG4bKPjaCuKyaBi6ZqcsHyp5HhH3WthJStom0SK > Max-Forwards: 70 > From: "Sherif 1003" >;tag=qfUt1fQWuE-Ro3zWld6X0r6WidtD6B1q > To: "Sherif 1003" > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60930 REGISTER > User-Agent: Telephone 1.0.2 > Contact: "Sherif 1003" > Expires: 300 > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, > REFER, MESSAGE, OPTIONS > Content-Length: 0 > > ------------------------------------------------------------------------ > send 744 bytes to udp/[41.34.123.243]:58881 at 23:23:04.523158: > ------------------------------------------------------------------------ > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport=58881;branch=z9hG4bKPjaCuKyaBi6ZqcsHyp5HhH3WthJStom0SK;received=41.34.123.243 > From: "Sherif 1003" >;tag=qfUt1fQWuE-Ro3zWld6X0r6WidtD6B1q > To: "Sherif 1003" ;tag=1F655pQB3DNBH > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60930 REGISTER > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > WWW-Authenticate: Digest realm="sip.pcfone.com", > nonce="699cfe2a-a5f7-11e1-a83e-7b1f73a7ffcf", algorithm=MD5, qop="auth" > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 830 bytes from udp/[41.34.123.243]:58881 at 23:23:04.757798: > ------------------------------------------------------------------------ > REGISTER sip:sip.pcfone.com:6090 SIP/2.0 > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport;branch=z9hG4bKPjs8SAufeoyzCLrePaAbYhZuOqQF47XIc- > Max-Forwards: 70 > From: "Sherif 1003" >;tag=qfUt1fQWuE-Ro3zWld6X0r6WidtD6B1q > To: "Sherif 1003" > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60931 REGISTER > User-Agent: Telephone 1.0.2 > Contact: "Sherif 1003" > Expires: 300 > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, > REFER, MESSAGE, OPTIONS > Authorization: Digest username="1002", realm="sip.pcfone.com", > nonce="699cfe2a-a5f7-11e1-a83e-7b1f73a7ffcf", uri="sip:sip.pcfone.com:6090", > response="23a886934aaf049f08432b6bb226bd37", algorithm=MD5, > cnonce="sc3AJfhuev18Bs-L-6tORmI.RXlxnPOz", qop=auth, nc=00000001 > Content-Length: 0 > > ------------------------------------------------------------------------ > send 706 bytes to udp/[41.34.123.243]:58881 at 23:23:04.759924: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport=58881;branch=z9hG4bKPjs8SAufeoyzCLrePaAbYhZuOqQF47XIc-;received=41.34.123.243 > From: "Sherif 1003" >;tag=qfUt1fQWuE-Ro3zWld6X0r6WidtD6B1q > To: "Sherif 1003" ;tag=2rZy7H8e0pByc > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60931 REGISTER > Contact: ;expires=300 > Date: Thu, 24 May 2012 23:23:04 GMT > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Content-Length: 0 > > ------------------------------------------------------------------------ > send 989 bytes to udp/[41.34.123.243]:58881 at 23:23:04.872728: > ------------------------------------------------------------------------ > NOTIFY sip:1002 at 192.168.178.26:58881;ob SIP/2.0 > Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKDHStmZ0taSmNp > Route: ;ob > Max-Forwards: 70 > From: ;tag=31rQ9cSjXZ1gr > To: > Call-ID: 4130a1ce-209a-1230-8985-00163e6bb553 > CSeq: 28614828 NOTIFY > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > 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=noresource > Content-Type: application/simple-message-summary > Content-Length: 91 > > Messages-Waiting: yes > Message-Account: sip:1002 at 78.138.90.58 > Voice-Message: 1/0 (0/0) > > ------------------------------------------------------------------------ > recv 308 bytes from udp/[41.34.123.243]:58881 at 23:23:05.269063: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;received=78.138.90.58;branch=z9hG4bKDHStmZ0taSmNp > Call-ID: 4130a1ce-209a-1230-8985-00163e6bb553 > From: ;tag=31rQ9cSjXZ1gr > To: ;tag=z9hG4bKDHStmZ0taSmNp > CSeq: 28614828 NOTIFY > Content-Length: 0 > > ------------------------------------------------------------------------ > send 916 bytes to udp/[217.74.179.29]:5060 at 23:25:45.873423: > ------------------------------------------------------------------------ > REGISTER sip:bluesip.net;transport=udp SIP/2.0 > Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKetjKptHy71a8H > Max-Forwards: 70 > From: ;transport=udp>;tag=6r0vBQZS650Fg > To: > Call-ID: 2a3e4696-a5b9-11e1-a7ed-7b1f73a7ffcf > CSeq: 28601493 REGISTER > Contact: ;transport=udp;gw=trunk_1000> > Expires: 3600 > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Authorization: Digest username="bluesip/sherifomran", realm=" > bluesip.net", nonce="4fbeba4ebfb7414d5c701bd6f994402d551f3acf", > algorithm=MD5, uri="sip:bluesip.net;transport=udp", > response="c6cdcafe0418e519bc9ee0d8fa3d4d74" > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 722 bytes from udp/[217.74.179.29]:5060 at 23:25:45.904176: > ------------------------------------------------------------------------ > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;branch=z9hG4bKetjKptHy71a8H > From: ;transport=udp>;tag=6r0vBQZS650Fg > To: ;transport=udp>;tag=0354a2e1b960c9cc2279eca4e5f84e20.1b16 > Call-ID: 2a3e4696-a5b9-11e1-a7ed-7b1f73a7ffcf > CSeq: 28601493 REGISTER > WWW-Authenticate: Digest realm="bluesip.net", > nonce="4fbec4a4ba82d1f1feaa32ccec01b85be054fcf7", stale=true > Server: Sip EXpress router (0.9.7 (i386/linux)) > Content-Length: 0 > Warning: 392 217.74.179.29:5060 "Noisy feedback tells: pid=5455 > req_src_ip=78.138.90.58 req_src_port=6090 in_uri=sip:bluesip.net;transport=udp > out_uri=sip:bluesip.net;transport=udp via_cnt==1" > > ------------------------------------------------------------------------ > send 916 bytes to udp/[217.74.179.29]:5060 at 23:25:45.904416: > ------------------------------------------------------------------------ > REGISTER sip:bluesip.net;transport=udp SIP/2.0 > Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKF3BcrN214a1tD > Max-Forwards: 70 > From: ;transport=udp>;tag=6r0vBQZS650Fg > To: > Call-ID: 2a3e4696-a5b9-11e1-a7ed-7b1f73a7ffcf > CSeq: 28601494 REGISTER > Contact: ;transport=udp;gw=trunk_1000> > Expires: 3600 > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Authorization: Digest username="bluesip/sherifomran", realm=" > bluesip.net", nonce="4fbec4a4ba82d1f1feaa32ccec01b85be054fcf7", > algorithm=MD5, uri="sip:bluesip.net;transport=udp", > response="4c09dbe4b9accb52d4104b40dfe20040" > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 699 bytes from udp/[217.74.179.29]:5060 at 23:25:45.935046: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;branch=z9hG4bKF3BcrN214a1tD > From: ;transport=udp>;tag=6r0vBQZS650Fg > To: ;transport=udp>;tag=0354a2e1b960c9cc2279eca4e5f84e20.2017 > Call-ID: 2a3e4696-a5b9-11e1-a7ed-7b1f73a7ffcf > CSeq: 28601494 REGISTER > Contact: ;transport=udp;gw=trunk_1000>;q=0.5;expires=3600 > Server: Sip EXpress router (0.9.7 (i386/linux)) > Content-Length: 0 > Warning: 392 217.74.179.29:5060 "Noisy feedback tells: pid=5462 > req_src_ip=78.138.90.58 req_src_port=6090 in_uri=sip:bluesip.net;transport=udp > out_uri=sip:bluesip.net;transport=udp via_cnt==1" > > ------------------------------------------------------------------------ > recv 564 bytes from udp/[41.34.123.243]:58881 at 23:27:59.977057: > ------------------------------------------------------------------------ > REGISTER sip:sip.pcfone.com:6090 SIP/2.0 > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport;branch=z9hG4bKPjEOP.a4NoXqoFqC7a.k2tyeX-mYlUyH4H > Max-Forwards: 70 > From: "Sherif 1003" >;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K > To: "Sherif 1003" > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60932 REGISTER > User-Agent: Telephone 1.0.2 > Contact: "Sherif 1003" > Expires: 300 > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, > REFER, MESSAGE, OPTIONS > Content-Length: 0 > > ------------------------------------------------------------------------ > send 744 bytes to udp/[41.34.123.243]:58881 at 23:27:59.977960: > ------------------------------------------------------------------------ > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport=58881;branch=z9hG4bKPjEOP.a4NoXqoFqC7a.k2tyeX-mYlUyH4H;received=41.34.123.243 > From: "Sherif 1003" >;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K > To: "Sherif 1003" ;tag=4ajgB89Nt8Q3K > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60932 REGISTER > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > WWW-Authenticate: Digest realm="sip.pcfone.com", > nonce="19b7cc40-a5f8-11e1-a83f-7b1f73a7ffcf", algorithm=MD5, qop="auth" > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 564 bytes from udp/[41.34.123.243]:58881 at 23:28:00.468487: > ------------------------------------------------------------------------ > REGISTER sip:sip.pcfone.com:6090 SIP/2.0 > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport;branch=z9hG4bKPjEOP.a4NoXqoFqC7a.k2tyeX-mYlUyH4H > Max-Forwards: 70 > From: "Sherif 1003" >;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K > To: "Sherif 1003" > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60932 REGISTER > User-Agent: Telephone 1.0.2 > Contact: "Sherif 1003" > Expires: 300 > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, > REFER, MESSAGE, OPTIONS > Content-Length: 0 > > ------------------------------------------------------------------------ > send 744 bytes to udp/[41.34.123.243]:58881 at 23:28:00.468679: > ------------------------------------------------------------------------ > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport=58881;branch=z9hG4bKPjEOP.a4NoXqoFqC7a.k2tyeX-mYlUyH4H;received=41.34.123.243 > From: "Sherif 1003" >;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K > To: "Sherif 1003" ;tag=4ajgB89Nt8Q3K > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60932 REGISTER > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > WWW-Authenticate: Digest realm="sip.pcfone.com", > nonce="19b7cc40-a5f8-11e1-a83f-7b1f73a7ffcf", algorithm=MD5, qop="auth" > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 830 bytes from udp/[41.34.123.243]:58881 at 23:28:00.717446: > ------------------------------------------------------------------------ > REGISTER sip:sip.pcfone.com:6090 SIP/2.0 > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport;branch=z9hG4bKPj0iHd3-XUjnvcBs9DiJUCWzrmYTmWxFLt > Max-Forwards: 70 > From: "Sherif 1003" >;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K > To: "Sherif 1003" > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60933 REGISTER > User-Agent: Telephone 1.0.2 > Contact: "Sherif 1003" > Expires: 300 > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, > REFER, MESSAGE, OPTIONS > Authorization: Digest username="1002", realm="sip.pcfone.com", > nonce="19b7cc40-a5f8-11e1-a83f-7b1f73a7ffcf", uri="sip:sip.pcfone.com:6090", > response="e21c903e6d98b5adcdd2eefa75217157", algorithm=MD5, > cnonce="sc3AJfhuev18Bs-L-6tORmI.RXlxnPOz", qop=auth, nc=00000001 > Content-Length: 0 > > ------------------------------------------------------------------------ > send 706 bytes to udp/[41.34.123.243]:58881 at 23:28:00.719947: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport=58881;branch=z9hG4bKPj0iHd3-XUjnvcBs9DiJUCWzrmYTmWxFLt;received=41.34.123.243 > From: "Sherif 1003" >;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K > To: "Sherif 1003" ;tag=5KB9c3tSQHepF > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60933 REGISTER > Contact: ;expires=300 > Date: Thu, 24 May 2012 23:28:00 GMT > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Content-Length: 0 > > ------------------------------------------------------------------------ > send 989 bytes to udp/[41.34.123.243]:58881 at 23:28:00.868368: > ------------------------------------------------------------------------ > NOTIFY sip:1002 at 192.168.178.26:58881;ob SIP/2.0 > Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKgc54SgK51KQDS > Route: ;ob > Max-Forwards: 70 > From: ;tag=6v41eyBXmt48a > To: > Call-ID: f19dfbce-209a-1230-8985-00163e6bb553 > CSeq: 28614976 NOTIFY > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > 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=noresource > Content-Type: application/simple-message-summary > Content-Length: 91 > > Messages-Waiting: yes > Message-Account: sip:1002 at 78.138.90.58 > Voice-Message: 1/0 (0/0) > > ------------------------------------------------------------------------ > recv 308 bytes from udp/[41.34.123.243]:58881 at 23:28:01.128279: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;received=78.138.90.58;branch=z9hG4bKgc54SgK51KQDS > Call-ID: f19dfbce-209a-1230-8985-00163e6bb553 > From: ;tag=6v41eyBXmt48a > To: ;tag=z9hG4bKgc54SgK51KQDS > CSeq: 28614976 NOTIFY > Content-Length: 0 > > ------------------------------------------------------------------------ > send 913 bytes to udp/[217.74.179.29]:5060 at 23:28:11.885213: > ------------------------------------------------------------------------ > REGISTER sip:bluesip.net;transport=udp SIP/2.0 > Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKHNyXUB48yvD0m > Max-Forwards: 70 > From: ;transport=udp>;tag=5F739Uep9vaXm > To: > Call-ID: 2a3e49c0-a5b9-11e1-a7ee-7b1f73a7ffcf > CSeq: 28601493 REGISTER > Contact: ;transport=udp;gw=trunk_1002> > Expires: 3600 > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Authorization: Digest username="bluesip/salahomran", realm="bluesip.net", > nonce="4fbebeca79a5cff7c417463814cb9054d313bcac", algorithm=MD5, uri="sip: > bluesip.net;transport=udp", response="bb1babb90f4ea1dc8fbb9aa45d6038c7" > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 720 bytes from udp/[217.74.179.29]:5060 at 23:28:11.915764: > ------------------------------------------------------------------------ > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;branch=z9hG4bKHNyXUB48yvD0m > From: ;transport=udp>;tag=5F739Uep9vaXm > To: ;transport=udp>;tag=0354a2e1b960c9cc2279eca4e5f84e20.3454 > Call-ID: 2a3e49c0-a5b9-11e1-a7ee-7b1f73a7ffcf > CSeq: 28601493 REGISTER > WWW-Authenticate: Digest realm="bluesip.net", > nonce="4fbec5365af3a174f1cc0b8cda770c727d25bf71", stale=true > Server: Sip EXpress router (0.9.7 (i386/linux)) > Content-Length: 0 > Warning: 392 217.74.179.29:5060 "Noisy feedback tells: pid=5462 > req_src_ip=78.138.90.58 req_src_port=6090 in_uri=sip:bluesip.net;transport=udp > out_uri=sip:bluesip.net;transport=udp via_cnt==1" > > ------------------------------------------------------------------------ > send 913 bytes to udp/[217.74.179.29]:5060 at 23:28:11.915914: > ------------------------------------------------------------------------ > REGISTER sip:bluesip.net;transport=udp SIP/2.0 > Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKjyQpX6mcv53jg > Max-Forwards: 70 > From: ;transport=udp>;tag=5F739Uep9vaXm > To: > Call-ID: 2a3e49c0-a5b9-11e1-a7ee-7b1f73a7ffcf > CSeq: 28601494 REGISTER > Contact: ;transport=udp;gw=trunk_1002> > Expires: 3600 > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Authorization: Digest username="bluesip/salahomran", realm="bluesip.net", > nonce="4fbec5365af3a174f1cc0b8cda770c727d25bf71", algorithm=MD5, uri="sip: > bluesip.net;transport=udp", response="e260f6dc9b01ce8bea8c5ffb36bd4ddd" > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 697 bytes from udp/[217.74.179.29]:5060 at 23:28:11.946325: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;branch=z9hG4bKjyQpX6mcv53jg > From: ;transport=udp>;tag=5F739Uep9vaXm > To: ;transport=udp>;tag=0354a2e1b960c9cc2279eca4e5f84e20.4b5c > Call-ID: 2a3e49c0-a5b9-11e1-a7ee-7b1f73a7ffcf > CSeq: 28601494 REGISTER > Contact: ;transport=udp;gw=trunk_1002>;q=0.5;expires=3600 > Server: Sip EXpress router (0.9.7 (i386/linux)) > Content-Length: 0 > Warning: 392 217.74.179.29:5060 "Noisy feedback tells: pid=5455 > req_src_ip=78.138.90.58 req_src_port=6090 in_uri=sip:bluesip.net;transport=udp > out_uri=sip:bluesip.net;transport=udp via_cnt==1" > > ------------------------------------------------------------------------ > recv 564 bytes from udp/[41.34.123.243]:58881 at 23:32:55.930149: > ------------------------------------------------------------------------ > REGISTER sip:sip.pcfone.com:6090 SIP/2.0 > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport;branch=z9hG4bKPjuk8DZGiaXfN7BmtslNHVjtSg5UXsbHXc > Max-Forwards: 70 > From: "Sherif 1003" >;tag=Pz.t-0tGXj3saM.ehBiB1EAMmmitze1D > To: "Sherif 1003" > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60934 REGISTER > User-Agent: Telephone 1.0.2 > Contact: "Sherif 1003" > Expires: 300 > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, > REFER, MESSAGE, OPTIONS > Content-Length: 0 > > ------------------------------------------------------------------------ > send 744 bytes to udp/[41.34.123.243]:58881 at 23:32:55.931324: > ------------------------------------------------------------------------ > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport=58881;branch=z9hG4bKPjuk8DZGiaXfN7BmtslNHVjtSg5UXsbHXc;received=41.34.123.243 > From: "Sherif 1003" >;tag=Pz.t-0tGXj3saM.ehBiB1EAMmmitze1D > To: "Sherif 1003" ;tag=75XtgSv0H3tUp > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60934 REGISTER > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > WWW-Authenticate: Digest realm="sip.pcfone.com", > nonce="ca1eabee-a5f8-11e1-a840-7b1f73a7ffcf", algorithm=MD5, qop="auth" > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 830 bytes from udp/[41.34.123.243]:58881 at 23:32:56.165624: > ------------------------------------------------------------------------ > REGISTER sip:sip.pcfone.com:6090 SIP/2.0 > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport;branch=z9hG4bKPjKEQ6xXmyzZxA6vCM88vHNYAOPAjUqFsD > Max-Forwards: 70 > From: "Sherif 1003" >;tag=Pz.t-0tGXj3saM.ehBiB1EAMmmitze1D > To: "Sherif 1003" > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60935 REGISTER > User-Agent: Telephone 1.0.2 > Contact: "Sherif 1003" > Expires: 300 > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, > REFER, MESSAGE, OPTIONS > Authorization: Digest username="1002", realm="sip.pcfone.com", > nonce="ca1eabee-a5f8-11e1-a840-7b1f73a7ffcf", uri="sip:sip.pcfone.com:6090", > response="ff6d12f4ec7abd4b37187222b3af2358", algorithm=MD5, > cnonce="sc3AJfhuev18Bs-L-6tORmI.RXlxnPOz", qop=auth, nc=00000001 > Content-Length: 0 > > ------------------------------------------------------------------------ > send 706 bytes to udp/[41.34.123.243]:58881 at 23:32:56.168033: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport=58881;branch=z9hG4bKPjKEQ6xXmyzZxA6vCM88vHNYAOPAjUqFsD;received=41.34.123.243 > From: "Sherif 1003" >;tag=Pz.t-0tGXj3saM.ehBiB1EAMmmitze1D > To: "Sherif 1003" ;tag=8eQKjmD4ecHej > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60935 REGISTER > Contact: ;expires=300 > Date: Thu, 24 May 2012 23:32:56 GMT > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Content-Length: 0 > > ------------------------------------------------------------------------ > send 989 bytes to udp/[41.34.123.243]:58881 at 23:32:56.360238: > ------------------------------------------------------------------------ > NOTIFY sip:1002 at 192.168.178.26:58881;ob SIP/2.0 > Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKK7gFZ15FSet5B > Route: ;ob > Max-Forwards: 70 > From: ;tag=9QgcmFy7BN70D > To: > Call-ID: a1be7708-209b-1230-8985-00163e6bb553 > CSeq: 28615124 NOTIFY > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > 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=noresource > Content-Type: application/simple-message-summary > Content-Length: 91 > > Messages-Waiting: yes > Message-Account: sip:1002 at 78.138.90.58 > Voice-Message: 1/0 (0/0) > > ------------------------------------------------------------------------ > recv 308 bytes from udp/[41.34.123.243]:58881 at 23:32:56.623554: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;received=78.138.90.58;branch=z9hG4bKK7gFZ15FSet5B > Call-ID: a1be7708-209b-1230-8985-00163e6bb553 > From: ;tag=9QgcmFy7BN70D > To: ;tag=z9hG4bKK7gFZ15FSet5B > CSeq: 28615124 NOTIFY > Content-Length: 0 > > ------------------------------------------------------------------------ > freeswitch at internal> > > > --- On *Thu, 5/24/12, Kristian Kielhofner * wrote: > > > From: Kristian Kielhofner > Subject: Re: [Freeswitch-users] Freeswitch (SBC + Media) does not know > that gateway phone is picked up > To: "FreeSWITCH Users Help" > Date: Thursday, May 24, 2012, 7:51 PM > > > Siptrace and logs please. > > On Thu, May 24, 2012 at 11:53 AM, Sherif Omran > > > wrote: > > > > Hi all, > > > > My topology is as follows: > > > > Kamailio -> FS (SBS+Media server) > > > > I came across an issue with my system as follows. I have a Hardphone > registered. When I do local call inside kamailio, it gets to FS and returns > back well and FS understands when I lift the handset. However, I added a > gateway (german landline server), when I call my self from another phone, > the call gets to FS and then transmits to Kamailio, it rings my extension > but when I lift the handset FS does not notice it and keeps ringing. > > > > Any body has an Idea? Here is my gateway trunk. > > > > > > > > > > > > > > > > > > > > > > thanks in advance > > Sherif Omran > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > Kristian Kielhofner > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120525/ec12c351/attachment-0001.html From sherifomran2000 at yahoo.com Fri May 25 11:20:32 2012 From: sherifomran2000 at yahoo.com (Sherif Omran) Date: Fri, 25 May 2012 00:20:32 -0700 (PDT) Subject: [Freeswitch-users] Freeswitch (SBC + Media) does not know that gateway phone is picked up In-Reply-To: Message-ID: <1337930432.2677.YahooMailClassic@web110809.mail.gq1.yahoo.com> Dear Sammy, Thank you for your question ... Yes, the GW is registered through FS because I did not know how to register it to kamailio. But it seems better to register it to kamailio. One more information, calls from 1001 to 1002 go to kamailio then to FS then return back to kamailio smoothly. Thus I would suggest that I change registering the gateway from FS to kamailio. but How to? --- On Fri, 5/25/12, SamyGo wrote: From: SamyGo Subject: Re: [Freeswitch-users] Freeswitch (SBC + Media) does not know that gateway phone is picked up To: "FreeSWITCH Users Help" Date: Friday, May 25, 2012, 8:08 AM Hi,These are SIP traces on FreeSWITCH console, whereas you are saying and it seems that kamailio is not detecting the answering (200 OK)of the call from extension 1002. Please, can you take a sip trace..!! I see you've both kamailio and FS on same server! Please take a pcap from the linux console using the following command. #tcpdump -i any -s 0 -w new-call-trace.pcap -vvv Please be quick on turning the sip trace on and off as quickly as possible to avoid extra packets. Once done open the file in wireshark ;?apply filter "sip || rtp" and then save the resulting capture in separate file. Send us the new file to analyse. One more silly question probably, I see REGISTERs coming to your FS as well and the calls to gateways are made from FS too !!, umm...just thinking what are you using kamailio for!!? ThanksBR,Sammy On Fri, May 25, 2012 at 4:40 AM, Sherif Omran wrote: Hi all, here is the siptrace: To figure it out 1- gateway called bluesip.net. It send invide using caller number at bluesip.net 2- This call should go to extension kb-1002. kb means go from freeswitch port 6090 to kamailio port 5060 3- It should go to call extension 1002 in Kamailio 4- Extension 1002 rings but when I reply, it does not notice I replied ./fs_cli ??????????? _____ ____???? ____ _???? ___????????????? ?????????? |? ___/ ___|?? / ___| |?? |_ _|???????????? ?????????? | |_? \___ \? | |?? | |??? | |??????????? ?????????? |? _|? ___) | | |___| |___ | |????????????? ?????????? |_|?? |____/?? \____|_____|___|??????????? ******************************************************* * Anthony Minessale II, Ken Rice,???????????????????? * * Michael Jerris, Travis Cross??????????????????????? * * FreeSWITCH (http://www.freeswitch.org)????????????? * * Paypal Donations Appreciated: paypal at freeswitch.org * * Brought to you by ClueCon http://www.cluecon.com/?? * ******************************************************* Type /help to see a list of commands +OK log level? [7] freeswitch at internal> tracelevel -ERR tracelevel Command not found! freeswitch at internal> sofia global siptrace on +OK Global siptrace on recv 1104 bytes from udp/[217.74.179.29]:5060 at 23:08:45.822962: ?? ------------------------------------------------------------------------ ?? INVITE sip:gw+trunk_1000 at 78.138.90.58:6090;transport=udp;gw=trunk_1000 SIP/2.0 ?? Record-Route: ?? Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 ?? Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 ?? From: "+41793940965" ;tag=as00589402 ?? To: ?? Contact: ?? Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net ?? CSeq: 102 INVITE ?? User-Agent: blueSIP PSTN GW ?? Max-Forwards: 69 ?? Date: Thu, 24 May 2012 23:08:44 GMT ?? Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY ?? Supported: replaces ?? Content-Type: application/sdp ?? Content-Length: 367 ?? P-hint: USRLOC ?? ?? v=0 ?? o=root 20076 20076 IN IP4 217.74.179.28 ?? s=session ?? c=IN IP4 217.74.179.28 ?? t=0 0 ?? m=audio 25626 RTP/AVP 8 0 18 111 3 101 ?? a=rtpmap:8 PCMA/8000 ?? a=rtpmap:0 PCMU/8000 ?? a=rtpmap:18 G729/8000 ?? a=fmtp:18 annexb=no ?? a=rtpmap:111 G726-32/8000 ?? a=rtpmap:3 GSM/8000 ?? a=rtpmap:101 telephone-event/8000 ?? a=fmtp:101 0-16 ?? a=silenceSupp:off - - - - ?? a=ptime:20 ?? a=sendrecv ?? ------------------------------------------------------------------------ send 494 bytes to udp/[217.74.179.29]:5060 at 23:08:45.844486: ?? ------------------------------------------------------------------------ ?? SIP/2.0 100 Trying ?? Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 ?? Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 ?? Record-Route: ?? From: "+41793940965" ;tag=as00589402 ?? To: ?? Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net ?? CSeq: 102 INVITE ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ 2012-05-24 23:08:45.840749 [NOTICE] switch_channel.c:926 New Channel sofia/internal/+41793940965 at bluesip.net [69cd22e6-a5f5-11e1-a833-7b1f73a7ffcf] 2012-05-24 23:08:45.840749 [DEBUG] sofia.c:5532 Channel sofia/internal/+41793940965 at bluesip.net entering state [received][100] 2012-05-24 23:08:45.840749 [DEBUG] sofia.c:5543 Remote SDP: v=0 o=root 20076 20076 IN IP4 217.74.179.28 s=session c=IN IP4 217.74.179.28 t=0 0 m=audio 25626 RTP/AVP 8 0 18 111 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:111 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMA:8:8000:20:64000]/[iLBC:97:8000:30:13330] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMU:0:8000:20:64000]/[iLBC:97:8000:30:13330] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G729:18:8000:20:8000]/[iLBC:97:8000:30:13330] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G726-32:111:8000:20:0]/[iLBC:97:8000:30:13330] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [GSM:3:8000:20:13200]/[iLBC:97:8000:30:13330] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [telephone-event:101:8000:20:0]/[iLBC:97:8000:30:13330] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf send/recv payload to 101 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMA:8:8000:20:64000]/[G7221:115:32000:20:48000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMU:0:8000:20:64000]/[G7221:115:32000:20:48000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G729:18:8000:20:8000]/[G7221:115:32000:20:48000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G726-32:111:8000:20:0]/[G7221:115:32000:20:48000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [GSM:3:8000:20:13200]/[G7221:115:32000:20:48000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [telephone-event:101:8000:20:0]/[G7221:115:32000:20:48000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf send/recv payload to 101 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMA:8:8000:20:64000]/[G7221:107:16000:20:32000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMU:0:8000:20:64000]/[G7221:107:16000:20:32000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G729:18:8000:20:8000]/[G7221:107:16000:20:32000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G726-32:111:8000:20:0]/[G7221:107:16000:20:32000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [GSM:3:8000:20:13200]/[G7221:107:16000:20:32000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [telephone-event:101:8000:20:0]/[G7221:107:16000:20:32000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf send/recv payload to 101 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMA:8:8000:20:64000]/[G722:9:8000:20:64000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G729:18:8000:20:8000]/[G722:9:8000:20:64000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G726-32:111:8000:20:0]/[G722:9:8000:20:64000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [GSM:3:8000:20:13200]/[G722:9:8000:20:64000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [telephone-event:101:8000:20:0]/[G722:9:8000:20:64000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf send/recv payload to 101 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMA:8:8000:20:64000]/[PCMU:0:8000:20:64000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:2996 Set Codec sofia/internal/+41793940965 at bluesip.net PCMU/8000 20 ms 160 samples 64000 bits 2012-05-24 23:08:45.840749 [DEBUG] switch_core_codec.c:111 sofia/internal/+41793940965 at bluesip.net Original read codec set to PCMU:0 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf send/recv payload to 101 2012-05-24 23:08:45.840749 [DEBUG] sofia.c:5757 (sofia/internal/+41793940965 at bluesip.net) State Change CS_NEW -> CS_INIT 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_INIT 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/+41793940965 at bluesip.net) State INIT 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:85 sofia/internal/+41793940965 at bluesip.net SOFIA INIT 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:125 (sofia/internal/+41793940965 at bluesip.net) State Change CS_INIT -> CS_ROUTING 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/+41793940965 at bluesip.net) State INIT going to sleep 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_ROUTING 2012-05-24 23:08:45.840749 [DEBUG] switch_channel.c:1886 (sofia/internal/+41793940965 at bluesip.net) Callstate Change DOWN -> RINGING 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/+41793940965 at bluesip.net) State ROUTING 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:148 sofia/internal/+41793940965 at bluesip.net SOFIA ROUTING 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:104 sofia/internal/+41793940965 at bluesip.net Standard ROUTING 2012-05-24 23:08:45.840749 [INFO] mod_dialplan_xml.c:485 Processing +41793940965 <+41793940965>->kb-1002 in context public Dialplan: sofia/internal/+41793940965 at bluesip.net parsing [public->from_kamailio] continue=false Dialplan: sofia/internal/+41793940965 at bluesip.net Regex (PASS) [from_kamailio] destination_number(kb-1002) =~ /^(.+)$/ break=on-false Dialplan: sofia/internal/+41793940965 at bluesip.net Action transfer(kb-1002 XML default) 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:154 (sofia/internal/+41793940965 at bluesip.net) State Change CS_ROUTING -> CS_EXECUTE 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/+41793940965 at bluesip.net) State ROUTING going to sleep 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_EXECUTE 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/+41793940965 at bluesip.net) State EXECUTE 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:241 sofia/internal/+41793940965 at bluesip.net SOFIA EXECUTE 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:192 sofia/internal/+41793940965 at bluesip.net Standard EXECUTE EXECUTE sofia/internal/+41793940965 at bluesip.net transfer(kb-1002 XML default) 2012-05-24 23:08:45.840749 [DEBUG] switch_ivr.c:1711 (sofia/internal/+41793940965 at bluesip.net) State Change CS_EXECUTE -> CS_ROUTING 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:731 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:08:45.840749 [NOTICE] switch_ivr.c:1717 Transfer sofia/internal/+41793940965 at bluesip.net to XML[kb-1002 at default] 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/+41793940965 at bluesip.net) State EXECUTE going to sleep 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_ROUTING 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/+41793940965 at bluesip.net) State ROUTING 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:148 sofia/internal/+41793940965 at bluesip.net SOFIA ROUTING 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:104 sofia/internal/+41793940965 at bluesip.net Standard ROUTING 2012-05-24 23:08:45.840749 [INFO] mod_dialplan_xml.c:485 Processing +41793940965 <+41793940965>->kb-1002 in context default Dialplan: sofia/internal/+41793940965 at bluesip.net parsing [default->vbox] continue=false Dialplan: sofia/internal/+41793940965 at bluesip.net Regex (FAIL) [vbox] destination_number(kb-1002) =~ /^vb-([0-9][0-9][0-9])$/ break=on-false Dialplan: sofia/internal/+41793940965 at bluesip.net parsing [default->vmenu] continue=false Dialplan: sofia/internal/+41793940965 at bluesip.net Regex (FAIL) [vmenu] destination_number(kb-1002) =~ /^vm-([0-9][0-9][0-9][0-9])$/ break=on-false Dialplan: sofia/internal/+41793940965 at bluesip.net parsing [default->kbridge] continue=false Dialplan: sofia/internal/+41793940965 at bluesip.net Regex (PASS) [kbridge] destination_number(kb-1002) =~ /^kb-(.+)$/ break=on-false Dialplan: sofia/internal/+41793940965 at bluesip.net Action set(proxy_media=true) Dialplan: sofia/internal/+41793940965 at bluesip.net Action set(call_timeout=50) Dialplan: sofia/internal/+41793940965 at bluesip.net Action set(continue_on_fail=true) Dialplan: sofia/internal/+41793940965 at bluesip.net Action set(hangup_after_bridge=true) Dialplan: sofia/internal/+41793940965 at bluesip.net Action set(sip_invite_domain=78.138.90.58) Dialplan: sofia/internal/+41793940965 at bluesip.net Action export(sip_contact_user=ufs) Dialplan: sofia/internal/+41793940965 at bluesip.net Action bridge(sofia/78.138.90.58/1002 at 78.138.90.58:5060) Dialplan: sofia/internal/+41793940965 at bluesip.net Action answer() Dialplan: sofia/internal/+41793940965 at bluesip.net Action voicemail(default ${domain_name} 1002) 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:154 (sofia/internal/+41793940965 at bluesip.net) State Change CS_ROUTING -> CS_EXECUTE 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/+41793940965 at bluesip.net) State ROUTING going to sleep 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_EXECUTE 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/+41793940965 at bluesip.net) State EXECUTE 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:241 sofia/internal/+41793940965 at bluesip.net SOFIA EXECUTE 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:192 sofia/internal/+41793940965 at bluesip.net Standard EXECUTE EXECUTE sofia/internal/+41793940965 at bluesip.net set(proxy_media=true) 2012-05-24 23:08:45.840749 [DEBUG] mod_dptools.c:1281 sofia/internal/+41793940965 at bluesip.net SET [proxy_media]=[true] EXECUTE sofia/internal/+41793940965 at bluesip.net set(call_timeout=50) 2012-05-24 23:08:45.840749 [DEBUG] mod_dptools.c:1281 sofia/internal/+41793940965 at bluesip.net SET [call_timeout]=[50] EXECUTE sofia/internal/+41793940965 at bluesip.net set(continue_on_fail=true) 2012-05-24 23:08:45.840749 [DEBUG] mod_dptools.c:1281 sofia/internal/+41793940965 at bluesip.net SET [continue_on_fail]=[true] EXECUTE sofia/internal/+41793940965 at bluesip.net set(hangup_after_bridge=true) 2012-05-24 23:08:45.840749 [DEBUG] mod_dptools.c:1281 sofia/internal/+41793940965 at bluesip.net SET [hangup_after_bridge]=[true] EXECUTE sofia/internal/+41793940965 at bluesip.net set(sip_invite_domain=78.138.90.58) 2012-05-24 23:08:45.840749 [DEBUG] mod_dptools.c:1281 sofia/internal/+41793940965 at bluesip.net SET [sip_invite_domain]=[78.138.90.58] EXECUTE sofia/internal/+41793940965 at bluesip.net export(sip_contact_user=ufs) 2012-05-24 23:08:45.840749 [DEBUG] switch_channel.c:1093 EXPORT (export_vars) [sip_contact_user]=[ufs] EXECUTE sofia/internal/+41793940965 at bluesip.net bridge(sofia/78.138.90.58/1002 at 78.138.90.58:5060) 2012-05-24 23:08:45.840749 [DEBUG] switch_channel.c:1047 sofia/internal/+41793940965 at bluesip.net EXPORTING[export_vars] [sip_contact_user]=[ufs] to event 2012-05-24 23:08:45.840749 [DEBUG] switch_ivr_originate.c:1884 Parsing global variables 2012-05-24 23:08:45.840749 [NOTICE] switch_channel.c:926 New Channel sofia/internal/1002 at 78.138.90.58:5060 [69ce1f0c-a5f5-11e1-a838-7b1f73a7ffcf] 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:4691 (sofia/internal/1002 at 78.138.90.58:5060) State Change CS_NEW -> CS_INIT 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1002 at 78.138.90.58:5060) Running State Change CS_INIT 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/1002 at 78.138.90.58:5060) State INIT 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:85 sofia/internal/1002 at 78.138.90.58:5060 SOFIA INIT 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:1871 sofia/internal/1002 at 78.138.90.58:5060 Patched SDP --- v=0 o=root 20076 20076 IN IP4 217.74.179.28 s=session c=IN IP4 217.74.179.28 t=0 0 m=audio 25626 RTP/AVP 8 0 18 111 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:111 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 +++ v=0 o=FreeSWITCH 1629510322 1629510323 IN IP4 78.138.90.58 s=FreeSWITCH c=IN IP4 78.138.90.58 t=0 0 m=audio 31178 RTP/AVP 8 0 18 111 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:111 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 send 1340 bytes to udp/[78.138.90.58]:5060 at 23:08:45.852280: ?? ------------------------------------------------------------------------ ?? INVITE sip:1002 at 78.138.90.58:5060 SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKapDgFeegKyHXK ?? Max-Forwards: 67 ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? CSeq: 28614398 INVITE ?? Contact: ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? 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: 372 ?? P-hint: USRLOC ?? X-FS-Support: update_display,send_info ?? Remote-Party-ID: "+41793940965" ;party=calling;screen=yes;privacy=off ?? ?? v=0 ?? o=FreeSWITCH 1629510322 1629510323 IN IP4 78.138.90.58 ?? s=FreeSWITCH ?? c=IN IP4 78.138.90.58 ?? t=0 0 ?? m=audio 31178 RTP/AVP 8 0 18 111 3 101 ?? a=rtpmap:8 PCMA/8000 ?? a=rtpmap:0 PCMU/8000 ?? a=rtpmap:18 G729/8000 ?? a=fmtp:18 annexb=no ?? a=rtpmap:111 G726-32/8000 ?? a=rtpmap:3 GSM/8000 ?? a=rtpmap:101 telephone-event/8000 ?? a=fmtp:101 0-16 ?? a=silenceSupp:off - - - - ?? a=ptime:20 ?? ------------------------------------------------------------------------ 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:125 (sofia/internal/1002 at 78.138.90.58:5060) State Change CS_INIT -> CS_ROUTING 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/1002 at 78.138.90.58:5060) State INIT going to sleep 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1002 at 78.138.90.58:5060) Running State Change CS_ROUTING 2012-05-24 23:08:45.840749 [DEBUG] switch_channel.c:1886 (sofia/internal/1002 at 78.138.90.58:5060) Callstate Change DOWN -> RINGING 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/1002 at 78.138.90.58:5060) State ROUTING 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:148 sofia/internal/1002 at 78.138.90.58:5060 SOFIA ROUTING 2012-05-24 23:08:45.840749 [DEBUG] switch_ivr_originate.c:66 (sofia/internal/1002 at 78.138.90.58:5060) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/1002 at 78.138.90.58:5060) State ROUTING going to sleep 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1002 at 78.138.90.58:5060) Running State Change CS_CONSUME_MEDIA 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:429 (sofia/internal/1002 at 78.138.90.58:5060) State CONSUME_MEDIA 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:429 (sofia/internal/1002 at 78.138.90.58:5060) State CONSUME_MEDIA going to sleep 2012-05-24 23:08:45.840749 [DEBUG] sofia.c:5532 Channel sofia/internal/1002 at 78.138.90.58:5060 entering state [calling][0] recv 365 bytes from udp/[78.138.90.58]:5060 at 23:08:45.964309: ?? ------------------------------------------------------------------------ ?? SIP/2.0 100 trying -- your call is important to us ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKapDgFeegKyHXK ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? CSeq: 28614398 INVITE ?? Server: kamailio (3.1.5 (x86_64/linux)) ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 496 bytes from udp/[78.138.90.58]:5060 at 23:08:47.414766: ?? ------------------------------------------------------------------------ ?? SIP/2.0 180 Ringing ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKapDgFeegKyHXK ?? Record-Route: ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ;tag=qCSaB2YJOAsAr2dJ-RWNMhPoDdn1RFvs ?? CSeq: 28614398 INVITE ?? Contact: "Mama" ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, REGISTER ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ 2012-05-24 23:08:47.400741 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:47.400741 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:47.400741 [DEBUG] sofia.c:5532 Channel sofia/internal/1002 at 78.138.90.58:5060 entering state [proceeding][180] 2012-05-24 23:08:47.400741 [NOTICE] sofia.c:5624 Ring-Ready sofia/internal/1002 at 78.138.90.58:5060! send 995 bytes to udp/[217.74.179.29]:5060 at 23:08:47.416538: ?? ------------------------------------------------------------------------ ?? SIP/2.0 180 Ringing ?? Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 ?? Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 ?? Record-Route: ?? From: "+41793940965" ;tag=as00589402 ?? To: ;tag=S7UZQygFt62Nm ?? Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net ?? CSeq: 102 INVITE ?? Contact: ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Accept: application/sdp ?? 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-Length: 0 ?? Remote-Party-ID: "Outbound Call" ;party=calling;privacy=off;screen=no ?? ?? ------------------------------------------------------------------------ 2012-05-24 23:08:47.400741 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:08:47.400741 [NOTICE] mod_sofia.c:2514 Ring-Ready sofia/internal/+41793940965 at bluesip.net! 2012-05-24 23:08:47.400741 [DEBUG] sofia.c:5532 Channel sofia/internal/+41793940965 at bluesip.net entering state [early][180] 2012-05-24 23:08:47.400741 [DEBUG] switch_core_session.c:731 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:08:47.400741 [NOTICE] switch_ivr_originate.c:483 Ring Ready sofia/internal/+41793940965 at bluesip.net! recv 496 bytes from udp/[78.138.90.58]:5060 at 23:08:47.804967: ?? ------------------------------------------------------------------------ ?? SIP/2.0 180 Ringing ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKapDgFeegKyHXK ?? Record-Route: ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ;tag=549D2DD03BBA7C67 ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? CSeq: 28614398 INVITE ?? Contact: ?? User-Agent: AVM FRITZ!Box Fon WLAN 7170 29.04.80 (Jan 27 2010) ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ 2012-05-24 23:08:47.800746 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:47.800746 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:47.800746 [DEBUG] sofia.c:5532 Channel sofia/internal/1002 at 78.138.90.58:5060 entering state [proceeding][180] recv 496 bytes from udp/[78.138.90.58]:5060 at 23:08:48.182098: ?? ------------------------------------------------------------------------ ?? SIP/2.0 180 Ringing ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKapDgFeegKyHXK ?? Record-Route: ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ;tag=qCSaB2YJOAsAr2dJ-RWNMhPoDdn1RFvs ?? CSeq: 28614398 INVITE ?? Contact: "Mama" ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, REGISTER ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ 2012-05-24 23:08:48.180741 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:48.180741 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:48.180741 [DEBUG] sofia.c:5532 Channel sofia/internal/1002 at 78.138.90.58:5060 entering state [proceeding][180] recv 496 bytes from udp/[78.138.90.58]:5060 at 23:08:48.417403: ?? ------------------------------------------------------------------------ ?? SIP/2.0 180 Ringing ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKapDgFeegKyHXK ?? Record-Route: ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ;tag=549D2DD03BBA7C67 ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? CSeq: 28614398 INVITE ?? Contact: ?? User-Agent: AVM FRITZ!Box Fon WLAN 7170 29.04.80 (Jan 27 2010) ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ 2012-05-24 23:08:48.400744 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:48.400744 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:48.400744 [DEBUG] sofia.c:5532 Channel sofia/internal/1002 at 78.138.90.58:5060 entering state [proceeding][180] recv 496 bytes from udp/[78.138.90.58]:5060 at 23:08:49.292599: ?? ------------------------------------------------------------------------ ?? SIP/2.0 180 Ringing ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKapDgFeegKyHXK ?? Record-Route: ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ;tag=549D2DD03BBA7C67 ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? CSeq: 28614398 INVITE ?? Contact: ?? User-Agent: AVM FRITZ!Box Fon WLAN 7170 29.04.80 (Jan 27 2010) ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ 2012-05-24 23:08:49.280742 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:49.280742 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:49.280742 [DEBUG] sofia.c:5532 Channel sofia/internal/1002 at 78.138.90.58:5060 entering state [proceeding][180] recv 478 bytes from udp/[217.74.179.29]:5060 at 23:09:15.066012: ?? ------------------------------------------------------------------------ ?? CANCEL sip:gw+trunk_1000 at 78.138.90.58:6090;transport=udp;gw=trunk_1000 SIP/2.0 ?? Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 ?? Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 ?? From: "+41793940965" ;tag=as00589402 ?? To: ?? Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net ?? CSeq: 102 CANCEL ?? User-Agent: blueSIP PSTN GW ?? Max-Forwards: 69 ?? Content-Length: 0 ?? P-hint: USRLOC ?? ?? ------------------------------------------------------------------------ send 369 bytes to udp/[217.74.179.29]:5060 at 23:09:15.066176: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 ?? Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 ?? From: "+41793940965" ;tag=as00589402 ?? To: ;tag=S7UZQygFt62Nm ?? Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net ?? CSeq: 102 CANCEL ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ send 766 bytes to udp/[217.74.179.29]:5060 at 23:09:15.066289: ?? ------------------------------------------------------------------------ ?? SIP/2.0 487 Request Terminated ?? Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 ?? Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 ?? From: "+41793940965" ;tag=as00589402 ?? To: ;tag=S7UZQygFt62Nm ?? Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net ?? CSeq: 102 INVITE ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? 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-Length: 0 ?? ?? ------------------------------------------------------------------------ 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:09:15.060742 [DEBUG] sofia.c:5532 Channel sofia/internal/+41793940965 at bluesip.net entering state [terminated][487] 2012-05-24 23:09:15.060742 [DEBUG] switch_channel.c:2848 (sofia/internal/+41793940965 at bluesip.net) Callstate Change RINGING -> HANGUP 2012-05-24 23:09:15.060742 [NOTICE] sofia.c:6301 Hangup sofia/internal/+41793940965 at bluesip.net [CS_EXECUTE] [ORIGINATOR_CANCEL] 2012-05-24 23:09:15.060742 [DEBUG] switch_channel.c:2871 Send signal sofia/internal/+41793940965 at bluesip.net [KILL] 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:09:15.060742 [DEBUG] switch_channel.c:2848 (sofia/internal/1002 at 78.138.90.58:5060) Callstate Change RINGING -> HANGUP 2012-05-24 23:09:15.060742 [NOTICE] switch_ivr_originate.c:3182 Hangup sofia/internal/1002 at 78.138.90.58:5060 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL] 2012-05-24 23:09:15.060742 [DEBUG] switch_channel.c:2871 Send signal sofia/internal/1002 at 78.138.90.58:5060 [KILL] 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1002 at 78.138.90.58:5060) Running State Change CS_HANGUP 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/1002 at 78.138.90.58:5060) State HANGUP 2012-05-24 23:09:15.060742 [DEBUG] mod_sofia.c:469 Channel sofia/internal/1002 at 78.138.90.58:5060 hanging up, cause: ORIGINATOR_CANCEL 2012-05-24 23:09:15.060742 [DEBUG] mod_sofia.c:523 Sending CANCEL to sofia/internal/1002 at 78.138.90.58:5060 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:47 sofia/internal/1002 at 78.138.90.58:5060 Standard HANGUP, cause: ORIGINATOR_CANCEL 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/1002 at 78.138.90.58:5060) State HANGUP going to sleep 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:393 (sofia/internal/1002 at 78.138.90.58:5060) State Change CS_HANGUP -> CS_REPORTING 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1002 at 78.138.90.58:5060) Running State Change CS_REPORTING 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/1002 at 78.138.90.58:5060) State REPORTING 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:79 sofia/internal/1002 at 78.138.90.58:5060 Standard REPORTING, cause: ORIGINATOR_CANCEL 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/1002 at 78.138.90.58:5060) State REPORTING going to sleep 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:387 (sofia/internal/1002 at 78.138.90.58:5060) State Change CS_REPORTING -> CS_DESTROY 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1382 Session 9 (sofia/internal/1002 at 78.138.90.58:5060) Locked, Waiting on external entities 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:09:15.060742 [DEBUG] switch_ivr_originate.c:3358 Originate Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL] 2012-05-24 23:09:15.060742 [NOTICE] switch_core_session.c:1400 Session 9 (sofia/internal/1002 at 78.138.90.58:5060) Ended 2012-05-24 23:09:15.060742 [NOTICE] switch_core_session.c:1402 Close Channel sofia/internal/1002 at 78.138.90.58:5060 [CS_DESTROY] 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:491 (sofia/internal/1002 at 78.138.90.58:5060) Callstate Change HANGUP -> DOWN 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:494 (sofia/internal/1002 at 78.138.90.58:5060) Running State Change CS_DESTROY 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/1002 at 78.138.90.58:5060) State DESTROY 2012-05-24 23:09:15.060742 [DEBUG] mod_sofia.c:374 sofia/internal/1002 at 78.138.90.58:5060 SOFIA DESTROY 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:86 sofia/internal/1002 at 78.138.90.58:5060 Standard DESTROY 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/1002 at 78.138.90.58:5060) State DESTROY going to sleep 2012-05-24 23:09:15.060742 [INFO] mod_dptools.c:2922 Originate Failed.? Cause: ORIGINATOR_CANCEL 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:2287 sofia/internal/+41793940965 at bluesip.net skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/+41793940965 at bluesip.net) State EXECUTE going to sleep 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_HANGUP 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/+41793940965 at bluesip.net) State HANGUP 2012-05-24 23:09:15.060742 [DEBUG] mod_sofia.c:469 Channel sofia/internal/+41793940965 at bluesip.net hanging up, cause: ORIGINATOR_CANCEL 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:47 sofia/internal/+41793940965 at bluesip.net Standard HANGUP, cause: ORIGINATOR_CANCEL 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/+41793940965 at bluesip.net) State HANGUP going to sleep 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:393 (sofia/internal/+41793940965 at bluesip.net) State Change CS_HANGUP -> CS_REPORTING 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_REPORTING 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/+41793940965 at bluesip.net) State REPORTING send 383 bytes to udp/[78.138.90.58]:5060 at 23:09:15.087695: ?? ------------------------------------------------------------------------ ?? CANCEL sip:1002 at 78.138.90.58:5060 SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKapDgFeegKyHXK ?? Max-Forwards: 67 ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? CSeq: 28614398 CANCEL ?? Reason: FreeSWITCH;cause=487;text="ORIGINATOR_CANCEL" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 408 bytes from udp/[217.74.179.29]:5060 at 23:09:15.094670: ?? ------------------------------------------------------------------------ ?? ACK sip:gw+trunk_1000 at 78.138.90.58:6090;transport=udp;gw=trunk_1000 SIP/2.0 ?? Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 ?? From: "+41793940965" ;tag=as00589402 ?? Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net ?? To: ;tag=S7UZQygFt62Nm ?? CSeq: 102 ACK ?? User-Agent: Sip EXpress router(0.9.7 (i386/linux)) ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:79 sofia/internal/+41793940965 at bluesip.net Standard REPORTING, cause: ORIGINATOR_CANCEL 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/+41793940965 at bluesip.net) State REPORTING going to sleep 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:387 (sofia/internal/+41793940965 at bluesip.net) State Change CS_REPORTING -> CS_DESTROY 2012-05-24 23:09:15.120807 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:09:15.120807 [DEBUG] switch_core_session.c:1382 Session 8 (sofia/internal/+41793940965 at bluesip.net) Locked, Waiting on external entities 2012-05-24 23:09:15.120807 [NOTICE] switch_core_session.c:1400 Session 8 (sofia/internal/+41793940965 at bluesip.net) Ended 2012-05-24 23:09:15.120807 [NOTICE] switch_core_session.c:1402 Close Channel sofia/internal/+41793940965 at bluesip.net [CS_DESTROY] 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:491 (sofia/internal/+41793940965 at bluesip.net) Callstate Change HANGUP -> DOWN 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:494 (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_DESTROY 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/+41793940965 at bluesip.net) State DESTROY 2012-05-24 23:09:15.120807 [DEBUG] mod_sofia.c:374 sofia/internal/+41793940965 at bluesip.net SOFIA DESTROY 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:86 sofia/internal/+41793940965 at bluesip.net Standard DESTROY 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/+41793940965 at bluesip.net) State DESTROY going to sleep recv 378 bytes from udp/[78.138.90.58]:5060 at 23:09:15.135235: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 canceling ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKapDgFeegKyHXK ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ;tag=a6a1c5f60faecf035a1ae5b6e96e979a-8c5b ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? CSeq: 28614398 CANCEL ?? Server: kamailio (3.1.5 (x86_64/linux)) ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 446 bytes from udp/[78.138.90.58]:5060 at 23:09:16.419308: ?? ------------------------------------------------------------------------ ?? SIP/2.0 487 Request Terminated ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKapDgFeegKyHXK ?? Record-Route: ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ;tag=qCSaB2YJOAsAr2dJ-RWNMhPoDdn1RFvs ?? CSeq: 28614398 INVITE ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, REGISTER ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 354 bytes to udp/[78.138.90.58]:5060 at 23:09:16.419543: ?? ------------------------------------------------------------------------ ?? ACK sip:1002 at 78.138.90.58:5060 SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKapDgFeegKyHXK ?? Max-Forwards: 67 ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ;tag=qCSaB2YJOAsAr2dJ-RWNMhPoDdn1RFvs ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? CSeq: 28614398 ACK ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 564 bytes from udp/[41.34.123.243]:58881 at 23:13:13.612418: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjHQIpv.ACHGbXD4IHrMr72LwZ8kYdlu-J ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=wOP.FVT4Crk8O5E.bB7BWUhNO6QG91IG ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60926 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 744 bytes to udp/[41.34.123.243]:58881 at 23:13:13.613152: ?? ------------------------------------------------------------------------ ?? SIP/2.0 401 Unauthorized ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjHQIpv.ACHGbXD4IHrMr72LwZ8kYdlu-J;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=wOP.FVT4Crk8O5E.bB7BWUhNO6QG91IG ?? To: "Sherif 1003" ;tag=USeHUmjpmrFUB ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60926 REGISTER ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? WWW-Authenticate: Digest realm="sip.pcfone.com", nonce="09674764-a5f6-11e1-a83c-7b1f73a7ffcf", algorithm=MD5, qop="auth" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 830 bytes from udp/[41.34.123.243]:58881 at 23:13:13.847746: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjFr7rIAdAJPRQd2GmrMrVib3KT3Rup0q4 ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=wOP.FVT4Crk8O5E.bB7BWUhNO6QG91IG ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60927 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Authorization: Digest username="1002", realm="sip.pcfone.com", nonce="09674764-a5f6-11e1-a83c-7b1f73a7ffcf", uri="sip:sip.pcfone.com:6090", response="3bc0bb82a803ded3641704c4eb78d58f", algorithm=MD5, cnonce="sc3AJfhuev18Bs-L-6tORmI.RXlxnPOz", qop=auth, nc=00000001 ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 706 bytes to udp/[41.34.123.243]:58881 at 23:13:13.850197: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjFr7rIAdAJPRQd2GmrMrVib3KT3Rup0q4;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=wOP.FVT4Crk8O5E.bB7BWUhNO6QG91IG ?? To: "Sherif 1003" ;tag=v279vF3SH15DQ ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60927 REGISTER ?? Contact: ;expires=300 ?? Date: Thu, 24 May 2012 23:13:13 GMT ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ send 989 bytes to udp/[41.34.123.243]:58881 at 23:13:13.888717: ?? ------------------------------------------------------------------------ ?? NOTIFY sip:1002 at 192.168.178.26:58881;ob SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKBZ68g9yKg77FF ?? Route: ;ob ?? Max-Forwards: 70 ?? From: ;tag=XB12yamXeav0j ?? To: ?? Call-ID: e0efa252-2098-1230-8985-00163e6bb553 ?? CSeq: 28614532 NOTIFY ?? Contact: ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? 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=noresource ?? Content-Type: application/simple-message-summary ?? Content-Length: 91 ?? ?? Messages-Waiting: yes ?? Message-Account: sip:1002 at 78.138.90.58 ?? Voice-Message: 1/0 (0/0) ?? ?? ------------------------------------------------------------------------ recv 308 bytes from udp/[41.34.123.243]:58881 at 23:13:14.298850: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;received=78.138.90.58;branch=z9hG4bKBZ68g9yKg77FF ?? Call-ID: e0efa252-2098-1230-8985-00163e6bb553 ?? From: ;tag=XB12yamXeav0j ?? To: ;tag=z9hG4bKBZ68g9yKg77FF ?? CSeq: 28614532 NOTIFY ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ recv 564 bytes from udp/[41.34.123.243]:58881 at 23:18:09.067232: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjABTslmZIhiHYRtLqQn2erwmX19MVYHFC ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=JBSXL5WZLWbLzpvRFoTSP5x8qQslI1ST ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60928 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 744 bytes to udp/[41.34.123.243]:58881 at 23:18:09.068121: ?? ------------------------------------------------------------------------ ?? SIP/2.0 401 Unauthorized ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjABTslmZIhiHYRtLqQn2erwmX19MVYHFC;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=JBSXL5WZLWbLzpvRFoTSP5x8qQslI1ST ?? To: "Sherif 1003" ;tag=ymtU0540BKjKe ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60928 REGISTER ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? WWW-Authenticate: Digest realm="sip.pcfone.com", nonce="b9821df4-a5f6-11e1-a83d-7b1f73a7ffcf", algorithm=MD5, qop="auth" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 830 bytes from udp/[41.34.123.243]:58881 at 23:18:09.302944: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjRgtztcy63vMT97Jp5pH9n311li5EFeY- ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=JBSXL5WZLWbLzpvRFoTSP5x8qQslI1ST ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60929 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Authorization: Digest username="1002", realm="sip.pcfone.com", nonce="b9821df4-a5f6-11e1-a83d-7b1f73a7ffcf", uri="sip:sip.pcfone.com:6090", response="519c6301816b51f3bbb98d97a347e92b", algorithm=MD5, cnonce="sc3AJfhuev18Bs-L-6tORmI.RXlxnPOz", qop=auth, nc=00000001 ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 706 bytes to udp/[41.34.123.243]:58881 at 23:18:09.305217: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjRgtztcy63vMT97Jp5pH9n311li5EFeY-;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=JBSXL5WZLWbLzpvRFoTSP5x8qQslI1ST ?? To: "Sherif 1003" ;tag=ZXKm20N48U85S ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60929 REGISTER ?? Contact: ;expires=300 ?? Date: Thu, 24 May 2012 23:18:09 GMT ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ send 989 bytes to udp/[41.34.123.243]:58881 at 23:18:09.379987: ?? ------------------------------------------------------------------------ ?? NOTIFY sip:1002 at 192.168.178.26:58881;ob SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKc8Z1j4FQDgy2a ?? Route: ;ob ?? Max-Forwards: 70 ?? From: ;tag=06cD4U6754yrN ?? To: ?? Call-ID: 91100602-2099-1230-8985-00163e6bb553 ?? CSeq: 28614680 NOTIFY ?? Contact: ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? 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=noresource ?? Content-Type: application/simple-message-summary ?? Content-Length: 91 ?? ?? Messages-Waiting: yes ?? Message-Account: sip:1002 at 78.138.90.58 ?? Voice-Message: 1/0 (0/0) ?? ?? ------------------------------------------------------------------------ recv 308 bytes from udp/[41.34.123.243]:58881 at 23:18:09.773629: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;received=78.138.90.58;branch=z9hG4bKc8Z1j4FQDgy2a ?? Call-ID: 91100602-2099-1230-8985-00163e6bb553 ?? From: ;tag=06cD4U6754yrN ?? To: ;tag=z9hG4bKc8Z1j4FQDgy2a ?? CSeq: 28614680 NOTIFY ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ recv 564 bytes from udp/[41.34.123.243]:58881 at 23:23:04.522455: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjaCuKyaBi6ZqcsHyp5HhH3WthJStom0SK ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=qfUt1fQWuE-Ro3zWld6X0r6WidtD6B1q ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60930 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 744 bytes to udp/[41.34.123.243]:58881 at 23:23:04.523158: ?? ------------------------------------------------------------------------ ?? SIP/2.0 401 Unauthorized ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjaCuKyaBi6ZqcsHyp5HhH3WthJStom0SK;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=qfUt1fQWuE-Ro3zWld6X0r6WidtD6B1q ?? To: "Sherif 1003" ;tag=1F655pQB3DNBH ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60930 REGISTER ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? WWW-Authenticate: Digest realm="sip.pcfone.com", nonce="699cfe2a-a5f7-11e1-a83e-7b1f73a7ffcf", algorithm=MD5, qop="auth" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 830 bytes from udp/[41.34.123.243]:58881 at 23:23:04.757798: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjs8SAufeoyzCLrePaAbYhZuOqQF47XIc- ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=qfUt1fQWuE-Ro3zWld6X0r6WidtD6B1q ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60931 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Authorization: Digest username="1002", realm="sip.pcfone.com", nonce="699cfe2a-a5f7-11e1-a83e-7b1f73a7ffcf", uri="sip:sip.pcfone.com:6090", response="23a886934aaf049f08432b6bb226bd37", algorithm=MD5, cnonce="sc3AJfhuev18Bs-L-6tORmI.RXlxnPOz", qop=auth, nc=00000001 ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 706 bytes to udp/[41.34.123.243]:58881 at 23:23:04.759924: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjs8SAufeoyzCLrePaAbYhZuOqQF47XIc-;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=qfUt1fQWuE-Ro3zWld6X0r6WidtD6B1q ?? To: "Sherif 1003" ;tag=2rZy7H8e0pByc ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60931 REGISTER ?? Contact: ;expires=300 ?? Date: Thu, 24 May 2012 23:23:04 GMT ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ send 989 bytes to udp/[41.34.123.243]:58881 at 23:23:04.872728: ?? ------------------------------------------------------------------------ ?? NOTIFY sip:1002 at 192.168.178.26:58881;ob SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKDHStmZ0taSmNp ?? Route: ;ob ?? Max-Forwards: 70 ?? From: ;tag=31rQ9cSjXZ1gr ?? To: ?? Call-ID: 4130a1ce-209a-1230-8985-00163e6bb553 ?? CSeq: 28614828 NOTIFY ?? Contact: ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? 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=noresource ?? Content-Type: application/simple-message-summary ?? Content-Length: 91 ?? ?? Messages-Waiting: yes ?? Message-Account: sip:1002 at 78.138.90.58 ?? Voice-Message: 1/0 (0/0) ?? ?? ------------------------------------------------------------------------ recv 308 bytes from udp/[41.34.123.243]:58881 at 23:23:05.269063: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;received=78.138.90.58;branch=z9hG4bKDHStmZ0taSmNp ?? Call-ID: 4130a1ce-209a-1230-8985-00163e6bb553 ?? From: ;tag=31rQ9cSjXZ1gr ?? To: ;tag=z9hG4bKDHStmZ0taSmNp ?? CSeq: 28614828 NOTIFY ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 916 bytes to udp/[217.74.179.29]:5060 at 23:25:45.873423: ?? ------------------------------------------------------------------------ ?? REGISTER sip:bluesip.net;transport=udp SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKetjKptHy71a8H ?? Max-Forwards: 70 ?? From: ;tag=6r0vBQZS650Fg ?? To: ?? Call-ID: 2a3e4696-a5b9-11e1-a7ed-7b1f73a7ffcf ?? CSeq: 28601493 REGISTER ?? Contact: ?? Expires: 3600 ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? Authorization: Digest username="bluesip/sherifomran", realm="bluesip.net", nonce="4fbeba4ebfb7414d5c701bd6f994402d551f3acf", algorithm=MD5, uri="sip:bluesip.net;transport=udp", response="c6cdcafe0418e519bc9ee0d8fa3d4d74" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 722 bytes from udp/[217.74.179.29]:5060 at 23:25:45.904176: ?? ------------------------------------------------------------------------ ?? SIP/2.0 401 Unauthorized ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKetjKptHy71a8H ?? From: ;tag=6r0vBQZS650Fg ?? To: ;tag=0354a2e1b960c9cc2279eca4e5f84e20.1b16 ?? Call-ID: 2a3e4696-a5b9-11e1-a7ed-7b1f73a7ffcf ?? CSeq: 28601493 REGISTER ?? WWW-Authenticate: Digest realm="bluesip.net", nonce="4fbec4a4ba82d1f1feaa32ccec01b85be054fcf7", stale=true ?? Server: Sip EXpress router (0.9.7 (i386/linux)) ?? Content-Length: 0 ?? Warning: 392 217.74.179.29:5060 "Noisy feedback tells:? pid=5455 req_src_ip=78.138.90.58 req_src_port=6090 in_uri=sip:bluesip.net;transport=udp out_uri=sip:bluesip.net;transport=udp via_cnt==1" ?? ?? ------------------------------------------------------------------------ send 916 bytes to udp/[217.74.179.29]:5060 at 23:25:45.904416: ?? ------------------------------------------------------------------------ ?? REGISTER sip:bluesip.net;transport=udp SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKF3BcrN214a1tD ?? Max-Forwards: 70 ?? From: ;tag=6r0vBQZS650Fg ?? To: ?? Call-ID: 2a3e4696-a5b9-11e1-a7ed-7b1f73a7ffcf ?? CSeq: 28601494 REGISTER ?? Contact: ?? Expires: 3600 ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? Authorization: Digest username="bluesip/sherifomran", realm="bluesip.net", nonce="4fbec4a4ba82d1f1feaa32ccec01b85be054fcf7", algorithm=MD5, uri="sip:bluesip.net;transport=udp", response="4c09dbe4b9accb52d4104b40dfe20040" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 699 bytes from udp/[217.74.179.29]:5060 at 23:25:45.935046: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKF3BcrN214a1tD ?? From: ;tag=6r0vBQZS650Fg ?? To: ;tag=0354a2e1b960c9cc2279eca4e5f84e20.2017 ?? Call-ID: 2a3e4696-a5b9-11e1-a7ed-7b1f73a7ffcf ?? CSeq: 28601494 REGISTER ?? Contact: ;q=0.5;expires=3600 ?? Server: Sip EXpress router (0.9.7 (i386/linux)) ?? Content-Length: 0 ?? Warning: 392 217.74.179.29:5060 "Noisy feedback tells:? pid=5462 req_src_ip=78.138.90.58 req_src_port=6090 in_uri=sip:bluesip.net;transport=udp out_uri=sip:bluesip.net;transport=udp via_cnt==1" ?? ?? ------------------------------------------------------------------------ recv 564 bytes from udp/[41.34.123.243]:58881 at 23:27:59.977057: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjEOP.a4NoXqoFqC7a.k2tyeX-mYlUyH4H ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60932 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 744 bytes to udp/[41.34.123.243]:58881 at 23:27:59.977960: ?? ------------------------------------------------------------------------ ?? SIP/2.0 401 Unauthorized ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjEOP.a4NoXqoFqC7a.k2tyeX-mYlUyH4H;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K ?? To: "Sherif 1003" ;tag=4ajgB89Nt8Q3K ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60932 REGISTER ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? WWW-Authenticate: Digest realm="sip.pcfone.com", nonce="19b7cc40-a5f8-11e1-a83f-7b1f73a7ffcf", algorithm=MD5, qop="auth" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 564 bytes from udp/[41.34.123.243]:58881 at 23:28:00.468487: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjEOP.a4NoXqoFqC7a.k2tyeX-mYlUyH4H ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60932 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 744 bytes to udp/[41.34.123.243]:58881 at 23:28:00.468679: ?? ------------------------------------------------------------------------ ?? SIP/2.0 401 Unauthorized ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjEOP.a4NoXqoFqC7a.k2tyeX-mYlUyH4H;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K ?? To: "Sherif 1003" ;tag=4ajgB89Nt8Q3K ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60932 REGISTER ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? WWW-Authenticate: Digest realm="sip.pcfone.com", nonce="19b7cc40-a5f8-11e1-a83f-7b1f73a7ffcf", algorithm=MD5, qop="auth" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 830 bytes from udp/[41.34.123.243]:58881 at 23:28:00.717446: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPj0iHd3-XUjnvcBs9DiJUCWzrmYTmWxFLt ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60933 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Authorization: Digest username="1002", realm="sip.pcfone.com", nonce="19b7cc40-a5f8-11e1-a83f-7b1f73a7ffcf", uri="sip:sip.pcfone.com:6090", response="e21c903e6d98b5adcdd2eefa75217157", algorithm=MD5, cnonce="sc3AJfhuev18Bs-L-6tORmI.RXlxnPOz", qop=auth, nc=00000001 ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 706 bytes to udp/[41.34.123.243]:58881 at 23:28:00.719947: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPj0iHd3-XUjnvcBs9DiJUCWzrmYTmWxFLt;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K ?? To: "Sherif 1003" ;tag=5KB9c3tSQHepF ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60933 REGISTER ?? Contact: ;expires=300 ?? Date: Thu, 24 May 2012 23:28:00 GMT ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ send 989 bytes to udp/[41.34.123.243]:58881 at 23:28:00.868368: ?? ------------------------------------------------------------------------ ?? NOTIFY sip:1002 at 192.168.178.26:58881;ob SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKgc54SgK51KQDS ?? Route: ;ob ?? Max-Forwards: 70 ?? From: ;tag=6v41eyBXmt48a ?? To: ?? Call-ID: f19dfbce-209a-1230-8985-00163e6bb553 ?? CSeq: 28614976 NOTIFY ?? Contact: ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? 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=noresource ?? Content-Type: application/simple-message-summary ?? Content-Length: 91 ?? ?? Messages-Waiting: yes ?? Message-Account: sip:1002 at 78.138.90.58 ?? Voice-Message: 1/0 (0/0) ?? ?? ------------------------------------------------------------------------ recv 308 bytes from udp/[41.34.123.243]:58881 at 23:28:01.128279: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;received=78.138.90.58;branch=z9hG4bKgc54SgK51KQDS ?? Call-ID: f19dfbce-209a-1230-8985-00163e6bb553 ?? From: ;tag=6v41eyBXmt48a ?? To: ;tag=z9hG4bKgc54SgK51KQDS ?? CSeq: 28614976 NOTIFY ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 913 bytes to udp/[217.74.179.29]:5060 at 23:28:11.885213: ?? ------------------------------------------------------------------------ ?? REGISTER sip:bluesip.net;transport=udp SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKHNyXUB48yvD0m ?? Max-Forwards: 70 ?? From: ;tag=5F739Uep9vaXm ?? To: ?? Call-ID: 2a3e49c0-a5b9-11e1-a7ee-7b1f73a7ffcf ?? CSeq: 28601493 REGISTER ?? Contact: ?? Expires: 3600 ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? Authorization: Digest username="bluesip/salahomran", realm="bluesip.net", nonce="4fbebeca79a5cff7c417463814cb9054d313bcac", algorithm=MD5, uri="sip:bluesip.net;transport=udp", response="bb1babb90f4ea1dc8fbb9aa45d6038c7" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 720 bytes from udp/[217.74.179.29]:5060 at 23:28:11.915764: ?? ------------------------------------------------------------------------ ?? SIP/2.0 401 Unauthorized ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKHNyXUB48yvD0m ?? From: ;tag=5F739Uep9vaXm ?? To: ;tag=0354a2e1b960c9cc2279eca4e5f84e20.3454 ?? Call-ID: 2a3e49c0-a5b9-11e1-a7ee-7b1f73a7ffcf ?? CSeq: 28601493 REGISTER ?? WWW-Authenticate: Digest realm="bluesip.net", nonce="4fbec5365af3a174f1cc0b8cda770c727d25bf71", stale=true ?? Server: Sip EXpress router (0.9.7 (i386/linux)) ?? Content-Length: 0 ?? Warning: 392 217.74.179.29:5060 "Noisy feedback tells:? pid=5462 req_src_ip=78.138.90.58 req_src_port=6090 in_uri=sip:bluesip.net;transport=udp out_uri=sip:bluesip.net;transport=udp via_cnt==1" ?? ?? ------------------------------------------------------------------------ send 913 bytes to udp/[217.74.179.29]:5060 at 23:28:11.915914: ?? ------------------------------------------------------------------------ ?? REGISTER sip:bluesip.net;transport=udp SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKjyQpX6mcv53jg ?? Max-Forwards: 70 ?? From: ;tag=5F739Uep9vaXm ?? To: ?? Call-ID: 2a3e49c0-a5b9-11e1-a7ee-7b1f73a7ffcf ?? CSeq: 28601494 REGISTER ?? Contact: ?? Expires: 3600 ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? Authorization: Digest username="bluesip/salahomran", realm="bluesip.net", nonce="4fbec5365af3a174f1cc0b8cda770c727d25bf71", algorithm=MD5, uri="sip:bluesip.net;transport=udp", response="e260f6dc9b01ce8bea8c5ffb36bd4ddd" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 697 bytes from udp/[217.74.179.29]:5060 at 23:28:11.946325: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKjyQpX6mcv53jg ?? From: ;tag=5F739Uep9vaXm ?? To: ;tag=0354a2e1b960c9cc2279eca4e5f84e20.4b5c ?? Call-ID: 2a3e49c0-a5b9-11e1-a7ee-7b1f73a7ffcf ?? CSeq: 28601494 REGISTER ?? Contact: ;q=0.5;expires=3600 ?? Server: Sip EXpress router (0.9.7 (i386/linux)) ?? Content-Length: 0 ?? Warning: 392 217.74.179.29:5060 "Noisy feedback tells:? pid=5455 req_src_ip=78.138.90.58 req_src_port=6090 in_uri=sip:bluesip.net;transport=udp out_uri=sip:bluesip.net;transport=udp via_cnt==1" ?? ?? ------------------------------------------------------------------------ recv 564 bytes from udp/[41.34.123.243]:58881 at 23:32:55.930149: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjuk8DZGiaXfN7BmtslNHVjtSg5UXsbHXc ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=Pz.t-0tGXj3saM.ehBiB1EAMmmitze1D ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60934 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 744 bytes to udp/[41.34.123.243]:58881 at 23:32:55.931324: ?? ------------------------------------------------------------------------ ?? SIP/2.0 401 Unauthorized ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjuk8DZGiaXfN7BmtslNHVjtSg5UXsbHXc;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=Pz.t-0tGXj3saM.ehBiB1EAMmmitze1D ?? To: "Sherif 1003" ;tag=75XtgSv0H3tUp ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60934 REGISTER ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? WWW-Authenticate: Digest realm="sip.pcfone.com", nonce="ca1eabee-a5f8-11e1-a840-7b1f73a7ffcf", algorithm=MD5, qop="auth" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 830 bytes from udp/[41.34.123.243]:58881 at 23:32:56.165624: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjKEQ6xXmyzZxA6vCM88vHNYAOPAjUqFsD ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=Pz.t-0tGXj3saM.ehBiB1EAMmmitze1D ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60935 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Authorization: Digest username="1002", realm="sip.pcfone.com", nonce="ca1eabee-a5f8-11e1-a840-7b1f73a7ffcf", uri="sip:sip.pcfone.com:6090", response="ff6d12f4ec7abd4b37187222b3af2358", algorithm=MD5, cnonce="sc3AJfhuev18Bs-L-6tORmI.RXlxnPOz", qop=auth, nc=00000001 ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 706 bytes to udp/[41.34.123.243]:58881 at 23:32:56.168033: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjKEQ6xXmyzZxA6vCM88vHNYAOPAjUqFsD;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=Pz.t-0tGXj3saM.ehBiB1EAMmmitze1D ?? To: "Sherif 1003" ;tag=8eQKjmD4ecHej ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60935 REGISTER ?? Contact: ;expires=300 ?? Date: Thu, 24 May 2012 23:32:56 GMT ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ send 989 bytes to udp/[41.34.123.243]:58881 at 23:32:56.360238: ?? ------------------------------------------------------------------------ ?? NOTIFY sip:1002 at 192.168.178.26:58881;ob SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKK7gFZ15FSet5B ?? Route: ;ob ?? Max-Forwards: 70 ?? From: ;tag=9QgcmFy7BN70D ?? To: ?? Call-ID: a1be7708-209b-1230-8985-00163e6bb553 ?? CSeq: 28615124 NOTIFY ?? Contact: ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? 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=noresource ?? Content-Type: application/simple-message-summary ?? Content-Length: 91 ?? ?? Messages-Waiting: yes ?? Message-Account: sip:1002 at 78.138.90.58 ?? Voice-Message: 1/0 (0/0) ?? ?? ------------------------------------------------------------------------ recv 308 bytes from udp/[41.34.123.243]:58881 at 23:32:56.623554: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;received=78.138.90.58;branch=z9hG4bKK7gFZ15FSet5B ?? Call-ID: a1be7708-209b-1230-8985-00163e6bb553 ?? From: ;tag=9QgcmFy7BN70D ?? To: ;tag=z9hG4bKK7gFZ15FSet5B ?? CSeq: 28615124 NOTIFY ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ freeswitch at internal> --- On Thu, 5/24/12, Kristian Kielhofner wrote: From: Kristian Kielhofner Subject: Re: [Freeswitch-users] Freeswitch (SBC + Media) does not know that gateway phone is picked up To: "FreeSWITCH Users Help" Date: Thursday, May 24, 2012, 7:51 PM Siptrace and logs please. On Thu, May 24, 2012 at 11:53 AM, Sherif Omran wrote: > > Hi all, > > My topology is as follows: > > Kamailio? -> FS (SBS+Media server) > > I came across an issue with my system as follows.? I have a Hardphone registered. When I do local call inside kamailio, it gets to FS and returns back well and FS understands when I lift the handset. However, I added a gateway (german landline server), when I call my self from another phone, the call gets to FS and then transmits to Kamailio, it rings my extension but when I lift the handset FS does not notice it and keeps ringing. > > Any body has an Idea? Here is my gateway trunk. > > > ??????? > ??????? > ??????? > ??????? > ??????? > ????? > > > thanks in advance > Sherif Omran > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Kristian Kielhofner _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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 at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -----Inline Attachment Follows----- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120525/264dffc0/attachment-0001.html From ocset at the800group.com Fri May 25 11:41:54 2012 From: ocset at the800group.com (ocset) Date: Fri, 25 May 2012 15:41:54 +0800 Subject: [Freeswitch-users] Bridge incoming call to external number Message-ID: <4FBF37C2.3090401@the800group.com> Hi Setup Windows 7 Freeswitch GXW4104 FXO 2 x POTS lines I have setup Freeswitch to bridge an incoming call to an external mobile phone using a GXW4104 FXO device. The bridge works great but I am having a real headache with the connection not being disconnected when the users hang up. There is no disconnection issue when a call comes in on either line and is connected to an internal extension (like ext 1001). It is when one line is bride to the second that the issue occurs. 1. Would this be a limitation of the GXW4104 2. Is it possible to write a "lua" script that could manage the disconnect some how? 3. If I enable "Polarity Reversal" under "FXO Termination" then the disconnect works but I have issues with sound and the system not registering when someone picks up. Please help me fix/diagnose this problem. Thanks O From shaheryarkh at googlemail.com Fri May 25 11:45:35 2012 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Fri, 25 May 2012 09:45:35 +0200 Subject: [Freeswitch-users] Freeswitch (SBC + Media) does not know that gateway phone is picked up In-Reply-To: <1337930432.2677.YahooMailClassic@web110809.mail.gq1.yahoo.com> References: <1337930432.2677.YahooMailClassic@web110809.mail.gq1.yahoo.com> Message-ID: For registering GW to Kamailio you can use registrar module with some appropriate auth module, e.g. auth_db. If you need to do the reverse i.e. register kamailio to some GW use uac module. Having said that, i don't think its an issue on FS side, are you sure you are calling record_route method for initial INVITE. because call path seems to be broken in reply route so kamailio does not seem to be able to transmit answer packet (200 OK) to calling party. Anyways, its just a guess, perhaps we can help you more if we have sip traces from kamailio side. Thank you. On Fri, May 25, 2012 at 9:20 AM, Sherif Omran wrote: > Dear Sammy, > > Thank you for your question ... Yes, the GW is registered through FS > because I did not know how to register it to kamailio. But it seems better > to register it to kamailio. > One more information, calls from 1001 to 1002 go to kamailio then to FS > then return back to kamailio smoothly. Thus I would suggest that I change > registering the gateway from FS to kamailio. but How to? > > > > --- On *Fri, 5/25/12, SamyGo * wrote: > > > From: SamyGo > Subject: Re: [Freeswitch-users] Freeswitch (SBC + Media) does not know > that gateway phone is picked up > To: "FreeSWITCH Users Help" > Date: Friday, May 25, 2012, 8:08 AM > > Hi, > These are SIP traces on FreeSWITCH console, whereas you are saying and it > seems that kamailio is not detecting the answering (200 OK)of the call from > extension 1002. Please, can you take a sip trace..!! I see you've both > kamailio and FS on same server! Please take a pcap from the linux console > using the following command. > > #tcpdump -i any -s 0 -w new-call-trace.pcap -vvv > > Please be quick on turning the sip trace on and off as quickly as possible > to avoid extra packets. Once done open the file in wireshark > ; apply filter "sip || rtp" and then save the > resulting capture in separate file. Send us the new file to analyse. > > One more silly question probably, I see REGISTERs coming to your FS as > well and the calls to gateways are made from FS too !!, umm...just thinking > what are you using kamailio for!!? > > Thanks > BR, > Sammy > > > On Fri, May 25, 2012 at 4:40 AM, Sherif Omran > > wrote: > > Hi all, > > here is the siptrace: To figure it out > 1- gateway called bluesip.net. It send invide using caller > number at bluesip.net > 2- This call should go to extension kb-1002. kb means go from freeswitch > port 6090 to kamailio port 5060 > 3- It should go to call extension 1002 in Kamailio > 4- Extension 1002 rings but when I reply, it does not notice I replied > > > ./fs_cli > _____ ____ ____ _ ___ > | ___/ ___| / ___| | |_ _| > | |_ \___ \ | | | | | | > | _| ___) | | |___| |___ | | > |_| |____/ \____|_____|___| > > ******************************************************* > * Anthony Minessale II, Ken Rice, * > * Michael Jerris, Travis Cross * > * FreeSWITCH (http://www.freeswitch.org) * > * Paypal Donations Appreciated: paypal at freeswitch.org* > * Brought to you by ClueCon http://www.cluecon.com/ * > ******************************************************* > > Type /help to see a list of commands > > > +OK log level [7] > freeswitch at internal> tracelevel > -ERR tracelevel Command not found! > > freeswitch at internal> sofia global siptrace on > +OK Global siptrace on > recv 1104 bytes from udp/[217.74.179.29]:5060 at 23:08:45.822962: > ------------------------------------------------------------------------ > INVITE sip:gw+trunk_1000 at 78.138.90.58:6090;transport=udp;gw=trunk_1000 > SIP/2.0 > Record-Route: > Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 > Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 > From: "+41793940965" ;tag=as00589402 > To: > Contact: > Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net > CSeq: 102 INVITE > User-Agent: blueSIP PSTN GW > Max-Forwards: 69 > Date: Thu, 24 May 2012 23:08:44 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Content-Type: application/sdp > Content-Length: 367 > P-hint: USRLOC > > v=0 > o=root 20076 20076 IN IP4 217.74.179.28 > s=session > c=IN IP4 217.74.179.28 > t=0 0 > m=audio 25626 RTP/AVP 8 0 18 111 3 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:111 G726-32/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > a=sendrecv > ------------------------------------------------------------------------ > send 494 bytes to udp/[217.74.179.29]:5060 at 23:08:45.844486: > ------------------------------------------------------------------------ > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 > Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 > Record-Route: > From: "+41793940965" ;tag=as00589402 > To: > Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net > CSeq: 102 INVITE > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Content-Length: 0 > > ------------------------------------------------------------------------ > 2012-05-24 23:08:45.840749 [NOTICE] switch_channel.c:926 New Channel > sofia/internal/+41793940965 at bluesip.net[69cd22e6-a5f5-11e1-a833-7b1f73a7ffcf] > 2012-05-24 23:08:45.840749 [DEBUG] sofia.c:5532 Channel sofia/internal/ > +41793940965 at bluesip.net entering state [received][100] > 2012-05-24 23:08:45.840749 [DEBUG] sofia.c:5543 Remote SDP: > v=0 > o=root 20076 20076 IN IP4 217.74.179.28 > s=session > c=IN IP4 217.74.179.28 > t=0 0 > m=audio 25626 RTP/AVP 8 0 18 111 3 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:111 G726-32/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [PCMA:8:8000:20:64000]/[iLBC:97:8000:30:13330] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [PCMU:0:8000:20:64000]/[iLBC:97:8000:30:13330] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [G729:18:8000:20:8000]/[iLBC:97:8000:30:13330] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [G726-32:111:8000:20:0]/[iLBC:97:8000:30:13330] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [GSM:3:8000:20:13200]/[iLBC:97:8000:30:13330] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [telephone-event:101:8000:20:0]/[iLBC:97:8000:30:13330] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf > send/recv payload to 101 > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [PCMA:8:8000:20:64000]/[G7221:115:32000:20:48000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [PCMU:0:8000:20:64000]/[G7221:115:32000:20:48000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [G729:18:8000:20:8000]/[G7221:115:32000:20:48000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [G726-32:111:8000:20:0]/[G7221:115:32000:20:48000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [GSM:3:8000:20:13200]/[G7221:115:32000:20:48000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [telephone-event:101:8000:20:0]/[G7221:115:32000:20:48000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf > send/recv payload to 101 > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [PCMA:8:8000:20:64000]/[G7221:107:16000:20:32000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [PCMU:0:8000:20:64000]/[G7221:107:16000:20:32000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [G729:18:8000:20:8000]/[G7221:107:16000:20:32000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [G726-32:111:8000:20:0]/[G7221:107:16000:20:32000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [GSM:3:8000:20:13200]/[G7221:107:16000:20:32000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [telephone-event:101:8000:20:0]/[G7221:107:16000:20:32000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf > send/recv payload to 101 > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [PCMA:8:8000:20:64000]/[G722:9:8000:20:64000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [G729:18:8000:20:8000]/[G722:9:8000:20:64000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [G726-32:111:8000:20:0]/[G722:9:8000:20:64000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [GSM:3:8000:20:13200]/[G722:9:8000:20:64000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [telephone-event:101:8000:20:0]/[G722:9:8000:20:64000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf > send/recv payload to 101 > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [PCMA:8:8000:20:64000]/[PCMU:0:8000:20:64000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:2996 Set Codec > sofia/internal/+41793940965 at bluesip.net PCMU/8000 20 ms 160 samples 64000 > bits > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_codec.c:111 sofia/internal/ > +41793940965 at bluesip.net Original read codec set to PCMU:0 > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf > send/recv payload to 101 > 2012-05-24 23:08:45.840749 [DEBUG] sofia.c:5757 (sofia/internal/ > +41793940965 at bluesip.net) State Change CS_NEW -> CS_INIT > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_INIT > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:401 > (sofia/internal/+41793940965 at bluesip.net) State INIT > 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:85 sofia/internal/ > +41793940965 at bluesip.net SOFIA INIT > 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:125 (sofia/internal/ > +41793940965 at bluesip.net) State Change CS_INIT -> CS_ROUTING > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:401 > (sofia/internal/+41793940965 at bluesip.net) State INIT going to sleep > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_ROUTING > 2012-05-24 23:08:45.840749 [DEBUG] switch_channel.c:1886 (sofia/internal/ > +41793940965 at bluesip.net) Callstate Change DOWN -> RINGING > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 > (sofia/internal/+41793940965 at bluesip.net) State ROUTING > 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:148 sofia/internal/ > +41793940965 at bluesip.net SOFIA ROUTING > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:104 > sofia/internal/+41793940965 at bluesip.net Standard ROUTING > 2012-05-24 23:08:45.840749 [INFO] mod_dialplan_xml.c:485 Processing > +41793940965 <+41793940965>->kb-1002 in context public > Dialplan: sofia/internal/+41793940965 at bluesip.net parsing > [public->from_kamailio] continue=false > Dialplan: sofia/internal/+41793940965 at bluesip.net Regex (PASS) > [from_kamailio] destination_number(kb-1002) =~ /^(.+)$/ break=on-false > Dialplan: sofia/internal/+41793940965 at bluesip.net Action transfer(kb-1002 > XML default) > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:154 > (sofia/internal/+41793940965 at bluesip.net) State Change CS_ROUTING -> > CS_EXECUTE > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 > (sofia/internal/+41793940965 at bluesip.net) State ROUTING going to sleep > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_EXECUTE > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:417 > (sofia/internal/+41793940965 at bluesip.net) State EXECUTE > 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:241 sofia/internal/ > +41793940965 at bluesip.net SOFIA EXECUTE > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:192 > sofia/internal/+41793940965 at bluesip.net Standard EXECUTE > EXECUTE sofia/internal/+41793940965 at bluesip.net transfer(kb-1002 XML > default) > 2012-05-24 23:08:45.840749 [DEBUG] switch_ivr.c:1711 (sofia/internal/ > +41793940965 at bluesip.net) State Change CS_EXECUTE -> CS_ROUTING > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:731 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:08:45.840749 [NOTICE] switch_ivr.c:1717 Transfer > sofia/internal/+41793940965 at bluesip.net to XML[kb-1002 at default] > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:417 > (sofia/internal/+41793940965 at bluesip.net) State EXECUTE going to sleep > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_ROUTING > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 > (sofia/internal/+41793940965 at bluesip.net) State ROUTING > 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:148 sofia/internal/ > +41793940965 at bluesip.net SOFIA ROUTING > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:104 > sofia/internal/+41793940965 at bluesip.net Standard ROUTING > 2012-05-24 23:08:45.840749 [INFO] mod_dialplan_xml.c:485 Processing > +41793940965 <+41793940965>->kb-1002 in context default > Dialplan: sofia/internal/+41793940965 at bluesip.net parsing [default->vbox] > continue=false > Dialplan: sofia/internal/+41793940965 at bluesip.net Regex (FAIL) [vbox] > destination_number(kb-1002) =~ /^vb-([0-9][0-9][0-9])$/ break=on-false > Dialplan: sofia/internal/+41793940965 at bluesip.net parsing > [default->vmenu] continue=false > Dialplan: sofia/internal/+41793940965 at bluesip.net Regex (FAIL) [vmenu] > destination_number(kb-1002) =~ /^vm-([0-9][0-9][0-9][0-9])$/ break=on-false > Dialplan: sofia/internal/+41793940965 at bluesip.net parsing > [default->kbridge] continue=false > Dialplan: sofia/internal/+41793940965 at bluesip.net Regex (PASS) [kbridge] > destination_number(kb-1002) =~ /^kb-(.+)$/ break=on-false > Dialplan: sofia/internal/+41793940965 at bluesip.net Action > set(proxy_media=true) > Dialplan: sofia/internal/+41793940965 at bluesip.net Action > set(call_timeout=50) > Dialplan: sofia/internal/+41793940965 at bluesip.net Action > set(continue_on_fail=true) > Dialplan: sofia/internal/+41793940965 at bluesip.net Action > set(hangup_after_bridge=true) > Dialplan: sofia/internal/+41793940965 at bluesip.net Action > set(sip_invite_domain=78.138.90.58) > Dialplan: sofia/internal/+41793940965 at bluesip.net Action > export(sip_contact_user=ufs) > Dialplan: sofia/internal/+41793940965 at bluesip.net Action bridge(sofia/ > 78.138.90.58/1002 at 78.138.90.58:5060) > Dialplan: sofia/internal/+41793940965 at bluesip.net Action answer() > Dialplan: sofia/internal/+41793940965 at bluesip.net Action > voicemail(default ${domain_name} 1002) > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:154 > (sofia/internal/+41793940965 at bluesip.net) State Change CS_ROUTING -> > CS_EXECUTE > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 > (sofia/internal/+41793940965 at bluesip.net) State ROUTING going to sleep > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_EXECUTE > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:417 > (sofia/internal/+41793940965 at bluesip.net) State EXECUTE > 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:241 sofia/internal/ > +41793940965 at bluesip.net SOFIA EXECUTE > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:192 > sofia/internal/+41793940965 at bluesip.net Standard EXECUTE > EXECUTE sofia/internal/+41793940965 at bluesip.net set(proxy_media=true) > 2012-05-24 23:08:45.840749 [DEBUG] mod_dptools.c:1281 sofia/internal/ > +41793940965 at bluesip.net SET [proxy_media]=[true] > EXECUTE sofia/internal/+41793940965 at bluesip.net set(call_timeout=50) > 2012-05-24 23:08:45.840749 [DEBUG] mod_dptools.c:1281 sofia/internal/ > +41793940965 at bluesip.net SET [call_timeout]=[50] > EXECUTE sofia/internal/+41793940965 at bluesip.net set(continue_on_fail=true) > 2012-05-24 23:08:45.840749 [DEBUG] mod_dptools.c:1281 sofia/internal/ > +41793940965 at bluesip.net SET [continue_on_fail]=[true] > EXECUTE sofia/internal/+41793940965 at bluesip.netset(hangup_after_bridge=true) > 2012-05-24 23:08:45.840749 [DEBUG] mod_dptools.c:1281 sofia/internal/ > +41793940965 at bluesip.net SET [hangup_after_bridge]=[true] > EXECUTE sofia/internal/+41793940965 at bluesip.netset(sip_invite_domain=78.138.90.58) > 2012-05-24 23:08:45.840749 [DEBUG] mod_dptools.c:1281 sofia/internal/ > +41793940965 at bluesip.net SET [sip_invite_domain]=[78.138.90.58] > EXECUTE sofia/internal/+41793940965 at bluesip.netexport(sip_contact_user=ufs) > 2012-05-24 23:08:45.840749 [DEBUG] switch_channel.c:1093 EXPORT > (export_vars) [sip_contact_user]=[ufs] > EXECUTE sofia/internal/+41793940965 at bluesip.net bridge(sofia/ > 78.138.90.58/1002 at 78.138.90.58:5060) > 2012-05-24 23:08:45.840749 [DEBUG] switch_channel.c:1047 sofia/internal/ > +41793940965 at bluesip.net EXPORTING[export_vars] [sip_contact_user]=[ufs] > to event > 2012-05-24 23:08:45.840749 [DEBUG] switch_ivr_originate.c:1884 Parsing > global variables > 2012-05-24 23:08:45.840749 [NOTICE] switch_channel.c:926 New Channel > sofia/internal/1002 at 78.138.90.58:5060[69ce1f0c-a5f5-11e1-a838-7b1f73a7ffcf] > 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:4691 (sofia/internal/ > 1002 at 78.138.90.58:5060) State Change CS_NEW -> CS_INIT > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/1002 at 78.138.90.58:5060) Running State Change CS_INIT > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:401 > (sofia/internal/1002 at 78.138.90.58:5060) State INIT > 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:85 sofia/internal/ > 1002 at 78.138.90.58:5060 SOFIA INIT > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:1871 sofia/internal/ > 1002 at 78.138.90.58:5060 Patched SDP > --- > v=0 > o=root 20076 20076 IN IP4 217.74.179.28 > s=session > c=IN IP4 217.74.179.28 > t=0 0 > m=audio 25626 RTP/AVP 8 0 18 111 3 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:111 G726-32/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > > +++ > v=0 > o=FreeSWITCH 1629510322 1629510323 IN IP4 78.138.90.58 > s=FreeSWITCH > c=IN IP4 78.138.90.58 > t=0 0 > m=audio 31178 RTP/AVP 8 0 18 111 3 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:111 G726-32/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > > send 1340 bytes to udp/[78.138.90.58]:5060 at 23:08:45.852280: > ------------------------------------------------------------------------ > INVITE sip:1002 at 78.138.90.58:5060 SIP/2.0 > Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKapDgFeegKyHXK > Max-Forwards: 67 > From: "+41793940965" ;tag=tgNrSS1jQFS8F > To: > Call-ID: 412c8312-2098-1230-8985-00163e6bb553 > CSeq: 28614398 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > 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: 372 > P-hint: USRLOC > X-FS-Support: update_display,send_info > Remote-Party-ID: "+41793940965" >;party=calling;screen=yes;privacy=off > > v=0 > o=FreeSWITCH 1629510322 1629510323 IN IP4 78.138.90.58 > s=FreeSWITCH > c=IN IP4 78.138.90.58 > t=0 0 > m=audio 31178 RTP/AVP 8 0 18 111 3 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:111 G726-32/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > ------------------------------------------------------------------------ > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:125 (sofia/internal/ > 1002 at 78.138.90.58:5060) State Change CS_INIT -> CS_ROUTING > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:401 > (sofia/internal/1002 at 78.138.90.58:5060) State INIT going to sleep > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/1002 at 78.138.90.58:5060) Running State Change CS_ROUTING > 2012-05-24 23:08:45.840749 [DEBUG] switch_channel.c:1886 (sofia/internal/ > 1002 at 78.138.90.58:5060) Callstate Change DOWN -> RINGING > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 > (sofia/internal/1002 at 78.138.90.58:5060) State ROUTING > 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:148 sofia/internal/ > 1002 at 78.138.90.58:5060 SOFIA ROUTING > 2012-05-24 23:08:45.840749 [DEBUG] switch_ivr_originate.c:66 > (sofia/internal/1002 at 78.138.90.58:5060) State Change CS_ROUTING -> > CS_CONSUME_MEDIA > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 > (sofia/internal/1002 at 78.138.90.58:5060) State ROUTING going to sleep > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/1002 at 78.138.90.58:5060) Running State Change > CS_CONSUME_MEDIA > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:429 > (sofia/internal/1002 at 78.138.90.58:5060) State CONSUME_MEDIA > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:429 > (sofia/internal/1002 at 78.138.90.58:5060) State CONSUME_MEDIA going to sleep > 2012-05-24 23:08:45.840749 [DEBUG] sofia.c:5532 Channel sofia/internal/ > 1002 at 78.138.90.58:5060 entering state [calling][0] > recv 365 bytes from udp/[78.138.90.58]:5060 at 23:08:45.964309: > ------------------------------------------------------------------------ > SIP/2.0 100 trying -- your call is important to us > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;branch=z9hG4bKapDgFeegKyHXK > From: "+41793940965" ;tag=tgNrSS1jQFS8F > To: > Call-ID: 412c8312-2098-1230-8985-00163e6bb553 > CSeq: 28614398 INVITE > Server: kamailio (3.1.5 (x86_64/linux)) > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 496 bytes from udp/[78.138.90.58]:5060 at 23:08:47.414766: > ------------------------------------------------------------------------ > SIP/2.0 180 Ringing > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;branch=z9hG4bKapDgFeegKyHXK > Record-Route: > Call-ID: 412c8312-2098-1230-8985-00163e6bb553 > From: "+41793940965" ;tag=tgNrSS1jQFS8F > To: > >;tag=qCSaB2YJOAsAr2dJ-RWNMhPoDdn1RFvs > CSeq: 28614398 INVITE > Contact: "Mama" > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, REGISTER > Content-Length: 0 > > ------------------------------------------------------------------------ > 2012-05-24 23:08:47.400741 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:47.400741 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:47.400741 [DEBUG] sofia.c:5532 Channel sofia/internal/ > 1002 at 78.138.90.58:5060 entering state [proceeding][180] > 2012-05-24 23:08:47.400741 [NOTICE] sofia.c:5624 Ring-Ready sofia/internal/ > 1002 at 78.138.90.58:5060! > send 995 bytes to udp/[217.74.179.29]:5060 at 23:08:47.416538: > ------------------------------------------------------------------------ > SIP/2.0 180 Ringing > Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 > Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 > Record-Route: > From: "+41793940965" ;tag=as00589402 > To: ;tag=S7UZQygFt62Nm > Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net > CSeq: 102 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Accept: application/sdp > 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-Length: 0 > Remote-Party-ID: "Outbound Call" > >;party=calling;privacy=off;screen=no > > ------------------------------------------------------------------------ > 2012-05-24 23:08:47.400741 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:08:47.400741 [NOTICE] mod_sofia.c:2514 Ring-Ready > sofia/internal/+41793940965 at bluesip.net! > 2012-05-24 23:08:47.400741 [DEBUG] sofia.c:5532 Channel sofia/internal/ > +41793940965 at bluesip.net entering state [early][180] > 2012-05-24 23:08:47.400741 [DEBUG] switch_core_session.c:731 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:08:47.400741 [NOTICE] switch_ivr_originate.c:483 Ring Ready > sofia/internal/+41793940965 at bluesip.net! > recv 496 bytes from udp/[78.138.90.58]:5060 at 23:08:47.804967: > ------------------------------------------------------------------------ > SIP/2.0 180 Ringing > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;branch=z9hG4bKapDgFeegKyHXK > Record-Route: > From: "+41793940965" ;tag=tgNrSS1jQFS8F > To: ;tag=549D2DD03BBA7C67 > Call-ID: 412c8312-2098-1230-8985-00163e6bb553 > CSeq: 28614398 INVITE > Contact: > ;uniq=6AC0DF4D2E498C8ACA82CB22226B9> > User-Agent: AVM FRITZ!Box Fon WLAN 7170 29.04.80 (Jan 27 2010) > Content-Length: 0 > > ------------------------------------------------------------------------ > 2012-05-24 23:08:47.800746 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:47.800746 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:47.800746 [DEBUG] sofia.c:5532 Channel sofia/internal/ > 1002 at 78.138.90.58:5060 entering state [proceeding][180] > recv 496 bytes from udp/[78.138.90.58]:5060 at 23:08:48.182098: > ------------------------------------------------------------------------ > SIP/2.0 180 Ringing > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;branch=z9hG4bKapDgFeegKyHXK > Record-Route: > Call-ID: 412c8312-2098-1230-8985-00163e6bb553 > From: "+41793940965" ;tag=tgNrSS1jQFS8F > To: > >;tag=qCSaB2YJOAsAr2dJ-RWNMhPoDdn1RFvs > CSeq: 28614398 INVITE > Contact: "Mama" > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, REGISTER > Content-Length: 0 > > ------------------------------------------------------------------------ > 2012-05-24 23:08:48.180741 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:48.180741 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:48.180741 [DEBUG] sofia.c:5532 Channel sofia/internal/ > 1002 at 78.138.90.58:5060 entering state [proceeding][180] > recv 496 bytes from udp/[78.138.90.58]:5060 at 23:08:48.417403: > ------------------------------------------------------------------------ > SIP/2.0 180 Ringing > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;branch=z9hG4bKapDgFeegKyHXK > Record-Route: > From: "+41793940965" ;tag=tgNrSS1jQFS8F > To: ;tag=549D2DD03BBA7C67 > Call-ID: 412c8312-2098-1230-8985-00163e6bb553 > CSeq: 28614398 INVITE > Contact: > ;uniq=6AC0DF4D2E498C8ACA82CB22226B9> > User-Agent: AVM FRITZ!Box Fon WLAN 7170 29.04.80 (Jan 27 2010) > Content-Length: 0 > > ------------------------------------------------------------------------ > 2012-05-24 23:08:48.400744 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:48.400744 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:48.400744 [DEBUG] sofia.c:5532 Channel sofia/internal/ > 1002 at 78.138.90.58:5060 entering state [proceeding][180] > recv 496 bytes from udp/[78.138.90.58]:5060 at 23:08:49.292599: > ------------------------------------------------------------------------ > SIP/2.0 180 Ringing > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;branch=z9hG4bKapDgFeegKyHXK > Record-Route: > From: "+41793940965" ;tag=tgNrSS1jQFS8F > To: ;tag=549D2DD03BBA7C67 > Call-ID: 412c8312-2098-1230-8985-00163e6bb553 > CSeq: 28614398 INVITE > Contact: > ;uniq=6AC0DF4D2E498C8ACA82CB22226B9> > User-Agent: AVM FRITZ!Box Fon WLAN 7170 29.04.80 (Jan 27 2010) > Content-Length: 0 > > ------------------------------------------------------------------------ > 2012-05-24 23:08:49.280742 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:49.280742 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:49.280742 [DEBUG] sofia.c:5532 Channel sofia/internal/ > 1002 at 78.138.90.58:5060 entering state [proceeding][180] > recv 478 bytes from udp/[217.74.179.29]:5060 at 23:09:15.066012: > ------------------------------------------------------------------------ > CANCEL sip:gw+trunk_1000 at 78.138.90.58:6090;transport=udp;gw=trunk_1000 > SIP/2.0 > Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 > Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 > From: "+41793940965" ;tag=as00589402 > To: > Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net > CSeq: 102 CANCEL > User-Agent: blueSIP PSTN GW > Max-Forwards: 69 > Content-Length: 0 > P-hint: USRLOC > > ------------------------------------------------------------------------ > send 369 bytes to udp/[217.74.179.29]:5060 at 23:09:15.066176: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 > Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 > From: "+41793940965" ;tag=as00589402 > To: ;tag=S7UZQygFt62Nm > Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net > CSeq: 102 CANCEL > Content-Length: 0 > > ------------------------------------------------------------------------ > send 766 bytes to udp/[217.74.179.29]:5060 at 23:09:15.066289: > ------------------------------------------------------------------------ > SIP/2.0 487 Request Terminated > Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 > Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 > From: "+41793940965" ;tag=as00589402 > To: ;tag=S7UZQygFt62Nm > Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net > CSeq: 102 INVITE > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > 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-Length: 0 > > ------------------------------------------------------------------------ > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:09:15.060742 [DEBUG] sofia.c:5532 Channel sofia/internal/ > +41793940965 at bluesip.net entering state [terminated][487] > 2012-05-24 23:09:15.060742 [DEBUG] switch_channel.c:2848 (sofia/internal/ > +41793940965 at bluesip.net) Callstate Change RINGING -> HANGUP > 2012-05-24 23:09:15.060742 [NOTICE] sofia.c:6301 Hangup sofia/internal/ > +41793940965 at bluesip.net [CS_EXECUTE] [ORIGINATOR_CANCEL] > 2012-05-24 23:09:15.060742 [DEBUG] switch_channel.c:2871 Send signal > sofia/internal/+41793940965 at bluesip.net [KILL] > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:09:15.060742 [DEBUG] switch_channel.c:2848 (sofia/internal/ > 1002 at 78.138.90.58:5060) Callstate Change RINGING -> HANGUP > 2012-05-24 23:09:15.060742 [NOTICE] switch_ivr_originate.c:3182 Hangup > sofia/internal/1002 at 78.138.90.58:5060 [CS_CONSUME_MEDIA] > [ORIGINATOR_CANCEL] > 2012-05-24 23:09:15.060742 [DEBUG] switch_channel.c:2871 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [KILL] > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/1002 at 78.138.90.58:5060) Running State Change CS_HANGUP > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:602 > (sofia/internal/1002 at 78.138.90.58:5060) State HANGUP > 2012-05-24 23:09:15.060742 [DEBUG] mod_sofia.c:469 Channel sofia/internal/ > 1002 at 78.138.90.58:5060 hanging up, cause: ORIGINATOR_CANCEL > 2012-05-24 23:09:15.060742 [DEBUG] mod_sofia.c:523 Sending CANCEL to > sofia/internal/1002 at 78.138.90.58:5060 > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:47 > sofia/internal/1002 at 78.138.90.58:5060 Standard HANGUP, cause: > ORIGINATOR_CANCEL > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:602 > (sofia/internal/1002 at 78.138.90.58:5060) State HANGUP going to sleep > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:393 > (sofia/internal/1002 at 78.138.90.58:5060) State Change CS_HANGUP -> > CS_REPORTING > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/1002 at 78.138.90.58:5060) Running State Change CS_REPORTING > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:662 > (sofia/internal/1002 at 78.138.90.58:5060) State REPORTING > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:79 > sofia/internal/1002 at 78.138.90.58:5060 Standard REPORTING, cause: > ORIGINATOR_CANCEL > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:662 > (sofia/internal/1002 at 78.138.90.58:5060) State REPORTING going to sleep > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:387 > (sofia/internal/1002 at 78.138.90.58:5060) State Change CS_REPORTING -> > CS_DESTROY > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1382 Session 9 > (sofia/internal/1002 at 78.138.90.58:5060) Locked, Waiting on external > entities > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:09:15.060742 [DEBUG] switch_ivr_originate.c:3358 Originate > Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL] > 2012-05-24 23:09:15.060742 [NOTICE] switch_core_session.c:1400 Session 9 > (sofia/internal/1002 at 78.138.90.58:5060) Ended > 2012-05-24 23:09:15.060742 [NOTICE] switch_core_session.c:1402 Close > Channel sofia/internal/1002 at 78.138.90.58:5060 [CS_DESTROY] > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:491 > (sofia/internal/1002 at 78.138.90.58:5060) Callstate Change HANGUP -> DOWN > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:494 > (sofia/internal/1002 at 78.138.90.58:5060) Running State Change CS_DESTROY > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:504 > (sofia/internal/1002 at 78.138.90.58:5060) State DESTROY > 2012-05-24 23:09:15.060742 [DEBUG] mod_sofia.c:374 sofia/internal/ > 1002 at 78.138.90.58:5060 SOFIA DESTROY > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:86 > sofia/internal/1002 at 78.138.90.58:5060 Standard DESTROY > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:504 > (sofia/internal/1002 at 78.138.90.58:5060) State DESTROY going to sleep > 2012-05-24 23:09:15.060742 [INFO] mod_dptools.c:2922 Originate Failed. > Cause: ORIGINATOR_CANCEL > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:2287 > sofia/internal/+41793940965 at bluesip.net skip receive message > [APPLICATION_EXEC_COMPLETE] (channel is hungup already) > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:417 > (sofia/internal/+41793940965 at bluesip.net) State EXECUTE going to sleep > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_HANGUP > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:602 > (sofia/internal/+41793940965 at bluesip.net) State HANGUP > 2012-05-24 23:09:15.060742 [DEBUG] mod_sofia.c:469 Channel sofia/internal/ > +41793940965 at bluesip.net hanging up, cause: ORIGINATOR_CANCEL > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:47 > sofia/internal/+41793940965 at bluesip.net Standard HANGUP, cause: > ORIGINATOR_CANCEL > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:602 > (sofia/internal/+41793940965 at bluesip.net) State HANGUP going to sleep > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:393 > (sofia/internal/+41793940965 at bluesip.net) State Change CS_HANGUP -> > CS_REPORTING > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/+41793940965 at bluesip.net) Running State Change > CS_REPORTING > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:662 > (sofia/internal/+41793940965 at bluesip.net) State REPORTING > send 383 bytes to udp/[78.138.90.58]:5060 at 23:09:15.087695: > ------------------------------------------------------------------------ > CANCEL sip:1002 at 78.138.90.58:5060 SIP/2.0 > Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKapDgFeegKyHXK > Max-Forwards: 67 > From: "+41793940965" ;tag=tgNrSS1jQFS8F > To: > Call-ID: 412c8312-2098-1230-8985-00163e6bb553 > CSeq: 28614398 CANCEL > Reason: FreeSWITCH;cause=487;text="ORIGINATOR_CANCEL" > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 408 bytes from udp/[217.74.179.29]:5060 at 23:09:15.094670: > ------------------------------------------------------------------------ > ACK sip:gw+trunk_1000 at 78.138.90.58:6090;transport=udp;gw=trunk_1000 > SIP/2.0 > Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 > From: "+41793940965" ;tag=as00589402 > Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net > To: ;tag=S7UZQygFt62Nm > CSeq: 102 ACK > User-Agent: Sip EXpress router(0.9.7 (i386/linux)) > Content-Length: 0 > > ------------------------------------------------------------------------ > 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:79 > sofia/internal/+41793940965 at bluesip.net Standard REPORTING, cause: > ORIGINATOR_CANCEL > 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:662 > (sofia/internal/+41793940965 at bluesip.net) State REPORTING going to sleep > 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:387 > (sofia/internal/+41793940965 at bluesip.net) State Change CS_REPORTING -> > CS_DESTROY > 2012-05-24 23:09:15.120807 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:09:15.120807 [DEBUG] switch_core_session.c:1382 Session 8 > (sofia/internal/+41793940965 at bluesip.net) Locked, Waiting on external > entities > 2012-05-24 23:09:15.120807 [NOTICE] switch_core_session.c:1400 Session 8 > (sofia/internal/+41793940965 at bluesip.net) Ended > 2012-05-24 23:09:15.120807 [NOTICE] switch_core_session.c:1402 Close > Channel sofia/internal/+41793940965 at bluesip.net [CS_DESTROY] > 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:491 > (sofia/internal/+41793940965 at bluesip.net) Callstate Change HANGUP -> DOWN > 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:494 > (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_DESTROY > 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:504 > (sofia/internal/+41793940965 at bluesip.net) State DESTROY > 2012-05-24 23:09:15.120807 [DEBUG] mod_sofia.c:374 sofia/internal/ > +41793940965 at bluesip.net SOFIA DESTROY > 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:86 > sofia/internal/+41793940965 at bluesip.net Standard DESTROY > 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:504 > (sofia/internal/+41793940965 at bluesip.net) State DESTROY going to sleep > recv 378 bytes from udp/[78.138.90.58]:5060 at 23:09:15.135235: > ------------------------------------------------------------------------ > SIP/2.0 200 canceling > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;branch=z9hG4bKapDgFeegKyHXK > From: "+41793940965" ;tag=tgNrSS1jQFS8F > To: >;tag=a6a1c5f60faecf035a1ae5b6e96e979a-8c5b > Call-ID: 412c8312-2098-1230-8985-00163e6bb553 > CSeq: 28614398 CANCEL > Server: kamailio (3.1.5 (x86_64/linux)) > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 446 bytes from udp/[78.138.90.58]:5060 at 23:09:16.419308: > ------------------------------------------------------------------------ > SIP/2.0 487 Request Terminated > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;branch=z9hG4bKapDgFeegKyHXK > Record-Route: > Call-ID: 412c8312-2098-1230-8985-00163e6bb553 > From: "+41793940965" ;tag=tgNrSS1jQFS8F > To: > >;tag=qCSaB2YJOAsAr2dJ-RWNMhPoDdn1RFvs > CSeq: 28614398 INVITE > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, REGISTER > Content-Length: 0 > > ------------------------------------------------------------------------ > send 354 bytes to udp/[78.138.90.58]:5060 at 23:09:16.419543: > ------------------------------------------------------------------------ > ACK sip:1002 at 78.138.90.58:5060 SIP/2.0 > Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKapDgFeegKyHXK > Max-Forwards: 67 > From: "+41793940965" ;tag=tgNrSS1jQFS8F > To: > >;tag=qCSaB2YJOAsAr2dJ-RWNMhPoDdn1RFvs > Call-ID: 412c8312-2098-1230-8985-00163e6bb553 > CSeq: 28614398 ACK > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 564 bytes from udp/[41.34.123.243]:58881 at 23:13:13.612418: > ------------------------------------------------------------------------ > REGISTER sip:sip.pcfone.com:6090 SIP/2.0 > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport;branch=z9hG4bKPjHQIpv.ACHGbXD4IHrMr72LwZ8kYdlu-J > Max-Forwards: 70 > From: "Sherif 1003" > >;tag=wOP.FVT4Crk8O5E.bB7BWUhNO6QG91IG > To: "Sherif 1003" > > > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60926 REGISTER > User-Agent: Telephone 1.0.2 > Contact: "Sherif 1003" > Expires: 300 > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, > REFER, MESSAGE, OPTIONS > Content-Length: 0 > > ------------------------------------------------------------------------ > send 744 bytes to udp/[41.34.123.243]:58881 at 23:13:13.613152: > ------------------------------------------------------------------------ > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport=58881;branch=z9hG4bKPjHQIpv.ACHGbXD4IHrMr72LwZ8kYdlu-J;received= > 41.34.123.243 > From: "Sherif 1003" > >;tag=wOP.FVT4Crk8O5E.bB7BWUhNO6QG91IG > To: "Sherif 1003" > >;tag=USeHUmjpmrFUB > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60926 REGISTER > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > WWW-Authenticate: Digest realm="sip.pcfone.com", > nonce="09674764-a5f6-11e1-a83c-7b1f73a7ffcf", algorithm=MD5, qop="auth" > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 830 bytes from udp/[41.34.123.243]:58881 at 23:13:13.847746: > ------------------------------------------------------------------------ > REGISTER sip:sip.pcfone.com:6090 SIP/2.0 > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport;branch=z9hG4bKPjFr7rIAdAJPRQd2GmrMrVib3KT3Rup0q4 > Max-Forwards: 70 > From: "Sherif 1003" > >;tag=wOP.FVT4Crk8O5E.bB7BWUhNO6QG91IG > To: "Sherif 1003" > > > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60927 REGISTER > User-Agent: Telephone 1.0.2 > Contact: "Sherif 1003" > Expires: 300 > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, > REFER, MESSAGE, OPTIONS > Authorization: Digest username="1002", realm="sip.pcfone.com", > nonce="09674764-a5f6-11e1-a83c-7b1f73a7ffcf", uri="sip:sip.pcfone.com:6090", > response="3bc0bb82a803ded3641704c4eb78d58f", algorithm=MD5, > cnonce="sc3AJfhuev18Bs-L-6tORmI.RXlxnPOz", qop=auth, nc=00000001 > Content-Length: 0 > > ------------------------------------------------------------------------ > send 706 bytes to udp/[41.34.123.243]:58881 at 23:13:13.850197: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport=58881;branch=z9hG4bKPjFr7rIAdAJPRQd2GmrMrVib3KT3Rup0q4;received= > 41.34.123.243 > From: "Sherif 1003" > >;tag=wOP.FVT4Crk8O5E.bB7BWUhNO6QG91IG > To: "Sherif 1003" > >;tag=v279vF3SH15DQ > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60927 REGISTER > Contact: ;expires=300 > Date: Thu, 24 May 2012 23:13:13 GMT > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Content-Length: 0 > > ------------------------------------------------------------------------ > send 989 bytes to udp/[41.34.123.243]:58881 at 23:13:13.888717: > ------------------------------------------------------------------------ > NOTIFY sip:1002 at 192.168.178.26:58881;ob SIP/2.0 > Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKBZ68g9yKg77FF > Route: ;ob > Max-Forwards: 70 > From: > >;tag=XB12yamXeav0j > To: > > > Call-ID: e0efa252-2098-1230-8985-00163e6bb553 > CSeq: 28614532 NOTIFY > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > 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=noresource > Content-Type: application/simple-message-summary > Content-Length: 91 > > Messages-Waiting: yes > Message-Account: sip:1002 at 78.138.90.58 > Voice-Message: 1/0 (0/0) > > ------------------------------------------------------------------------ > recv 308 bytes from udp/[41.34.123.243]:58881 at 23:13:14.298850: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;received=78.138.90.58;branch=z9hG4bKBZ68g9yKg77FF > Call-ID: e0efa252-2098-1230-8985-00163e6bb553 > From: > >;tag=XB12yamXeav0j > To: > >;tag=z9hG4bKBZ68g9yKg77FF > CSeq: 28614532 NOTIFY > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 564 bytes from udp/[41.34.123.243]:58881 at 23:18:09.067232: > ------------------------------------------------------------------------ > REGISTER sip:sip.pcfone.com:6090 SIP/2.0 > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport;branch=z9hG4bKPjABTslmZIhiHYRtLqQn2erwmX19MVYHFC > Max-Forwards: 70 > From: "Sherif 1003" > >;tag=JBSXL5WZLWbLzpvRFoTSP5x8qQslI1ST > To: "Sherif 1003" > > > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60928 REGISTER > User-Agent: Telephone 1.0.2 > Contact: "Sherif 1003" > Expires: 300 > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, > REFER, MESSAGE, OPTIONS > Content-Length: 0 > > ------------------------------------------------------------------------ > send 744 bytes to udp/[41.34.123.243]:58881 at 23:18:09.068121: > ------------------------------------------------------------------------ > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport=58881;branch=z9hG4bKPjABTslmZIhiHYRtLqQn2erwmX19MVYHFC;received= > 41.34.123.243 > From: "Sherif 1003" > >;tag=JBSXL5WZLWbLzpvRFoTSP5x8qQslI1ST > To: "Sherif 1003" > >;tag=ymtU0540BKjKe > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60928 REGISTER > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > WWW-Authenticate: Digest realm="sip.pcfone.com", > nonce="b9821df4-a5f6-11e1-a83d-7b1f73a7ffcf", algorithm=MD5, qop="auth" > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 830 bytes from udp/[41.34.123.243]:58881 at 23:18:09.302944: > ------------------------------------------------------------------------ > REGISTER sip:sip.pcfone.com:6090 SIP/2.0 > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport;branch=z9hG4bKPjRgtztcy63vMT97Jp5pH9n311li5EFeY- > Max-Forwards: 70 > From: "Sherif 1003" > >;tag=JBSXL5WZLWbLzpvRFoTSP5x8qQslI1ST > To: "Sherif 1003" > > > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60929 REGISTER > User-Agent: Telephone 1.0.2 > Contact: "Sherif 1003" > Expires: 300 > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, > REFER, MESSAGE, OPTIONS > Authorization: Digest username="1002", realm="sip.pcfone.com", > nonce="b9821df4-a5f6-11e1-a83d-7b1f73a7ffcf", uri="sip:sip.pcfone.com:6090", > response="519c6301816b51f3bbb98d97a347e92b", algorithm=MD5, > cnonce="sc3AJfhuev18Bs-L-6tORmI.RXlxnPOz", qop=auth, nc=00000001 > Content-Length: 0 > > ------------------------------------------------------------------------ > send 706 bytes to udp/[41.34.123.243]:58881 at 23:18:09.305217: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport=58881;branch=z9hG4bKPjRgtztcy63vMT97Jp5pH9n311li5EFeY-;received= > 41.34.123.243 > From: "Sherif 1003" > >;tag=JBSXL5WZLWbLzpvRFoTSP5x8qQslI1ST > To: "Sherif 1003" > >;tag=ZXKm20N48U85S > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60929 REGISTER > Contact: ;expires=300 > Date: Thu, 24 May 2012 23:18:09 GMT > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Content-Length: 0 > > ------------------------------------------------------------------------ > send 989 bytes to udp/[41.34.123.243]:58881 at 23:18:09.379987: > ------------------------------------------------------------------------ > NOTIFY sip:1002 at 192.168.178.26:58881;ob SIP/2.0 > Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKc8Z1j4FQDgy2a > Route: ;ob > Max-Forwards: 70 > From: > >;tag=06cD4U6754yrN > To: > > > Call-ID: 91100602-2099-1230-8985-00163e6bb553 > CSeq: 28614680 NOTIFY > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > 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=noresource > Content-Type: application/simple-message-summary > Content-Length: 91 > > Messages-Waiting: yes > Message-Account: sip:1002 at 78.138.90.58 > Voice-Message: 1/0 (0/0) > > ------------------------------------------------------------------------ > recv 308 bytes from udp/[41.34.123.243]:58881 at 23:18:09.773629: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;received=78.138.90.58;branch=z9hG4bKc8Z1j4FQDgy2a > Call-ID: 91100602-2099-1230-8985-00163e6bb553 > From: > >;tag=06cD4U6754yrN > To: > >;tag=z9hG4bKc8Z1j4FQDgy2a > CSeq: 28614680 NOTIFY > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 564 bytes from udp/[41.34.123.243]:58881 at 23:23:04.522455: > ------------------------------------------------------------------------ > REGISTER sip:sip.pcfone.com:6090 SIP/2.0 > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport;branch=z9hG4bKPjaCuKyaBi6ZqcsHyp5HhH3WthJStom0SK > Max-Forwards: 70 > From: "Sherif 1003" > >;tag=qfUt1fQWuE-Ro3zWld6X0r6WidtD6B1q > To: "Sherif 1003" > > > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60930 REGISTER > User-Agent: Telephone 1.0.2 > Contact: "Sherif 1003" > Expires: 300 > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, > REFER, MESSAGE, OPTIONS > Content-Length: 0 > > ------------------------------------------------------------------------ > send 744 bytes to udp/[41.34.123.243]:58881 at 23:23:04.523158: > ------------------------------------------------------------------------ > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport=58881;branch=z9hG4bKPjaCuKyaBi6ZqcsHyp5HhH3WthJStom0SK;received= > 41.34.123.243 > From: "Sherif 1003" > >;tag=qfUt1fQWuE-Ro3zWld6X0r6WidtD6B1q > To: "Sherif 1003" > >;tag=1F655pQB3DNBH > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60930 REGISTER > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > WWW-Authenticate: Digest realm="sip.pcfone.com", > nonce="699cfe2a-a5f7-11e1-a83e-7b1f73a7ffcf", algorithm=MD5, qop="auth" > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 830 bytes from udp/[41.34.123.243]:58881 at 23:23:04.757798: > ------------------------------------------------------------------------ > REGISTER sip:sip.pcfone.com:6090 SIP/2.0 > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport;branch=z9hG4bKPjs8SAufeoyzCLrePaAbYhZuOqQF47XIc- > Max-Forwards: 70 > From: "Sherif 1003" > >;tag=qfUt1fQWuE-Ro3zWld6X0r6WidtD6B1q > To: "Sherif 1003" > > > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60931 REGISTER > User-Agent: Telephone 1.0.2 > Contact: "Sherif 1003" > Expires: 300 > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, > REFER, MESSAGE, OPTIONS > Authorization: Digest username="1002", realm="sip.pcfone.com", > nonce="699cfe2a-a5f7-11e1-a83e-7b1f73a7ffcf", uri="sip:sip.pcfone.com:6090", > response="23a886934aaf049f08432b6bb226bd37", algorithm=MD5, > cnonce="sc3AJfhuev18Bs-L-6tORmI.RXlxnPOz", qop=auth, nc=00000001 > Content-Length: 0 > > ------------------------------------------------------------------------ > send 706 bytes to udp/[41.34.123.243]:58881 at 23:23:04.759924: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport=58881;branch=z9hG4bKPjs8SAufeoyzCLrePaAbYhZuOqQF47XIc-;received= > 41.34.123.243 > From: "Sherif 1003" > >;tag=qfUt1fQWuE-Ro3zWld6X0r6WidtD6B1q > To: "Sherif 1003" > >;tag=2rZy7H8e0pByc > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60931 REGISTER > Contact: ;expires=300 > Date: Thu, 24 May 2012 23:23:04 GMT > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Content-Length: 0 > > ------------------------------------------------------------------------ > send 989 bytes to udp/[41.34.123.243]:58881 at 23:23:04.872728: > ------------------------------------------------------------------------ > NOTIFY sip:1002 at 192.168.178.26:58881;ob SIP/2.0 > Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKDHStmZ0taSmNp > Route: ;ob > Max-Forwards: 70 > From: > >;tag=31rQ9cSjXZ1gr > To: > > > Call-ID: 4130a1ce-209a-1230-8985-00163e6bb553 > CSeq: 28614828 NOTIFY > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > 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=noresource > Content-Type: application/simple-message-summary > Content-Length: 91 > > Messages-Waiting: yes > Message-Account: sip:1002 at 78.138.90.58 > Voice-Message: 1/0 (0/0) > > ------------------------------------------------------------------------ > recv 308 bytes from udp/[41.34.123.243]:58881 at 23:23:05.269063: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;received=78.138.90.58;branch=z9hG4bKDHStmZ0taSmNp > Call-ID: 4130a1ce-209a-1230-8985-00163e6bb553 > From: > >;tag=31rQ9cSjXZ1gr > To: > >;tag=z9hG4bKDHStmZ0taSmNp > CSeq: 28614828 NOTIFY > Content-Length: 0 > > ------------------------------------------------------------------------ > send 916 bytes to udp/[217.74.179.29]:5060 at 23:25:45.873423: > ------------------------------------------------------------------------ > REGISTER sip:bluesip.net;transport=udp SIP/2.0 > Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKetjKptHy71a8H > Max-Forwards: 70 > From: > ;transport=udp>;tag=6r0vBQZS650Fg > To: > ;transport=udp> > Call-ID: 2a3e4696-a5b9-11e1-a7ed-7b1f73a7ffcf > CSeq: 28601493 REGISTER > Contact: ;transport=udp;gw=trunk_1000> > Expires: 3600 > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Authorization: Digest username="bluesip/sherifomran", realm=" > bluesip.net", nonce="4fbeba4ebfb7414d5c701bd6f994402d551f3acf", > algorithm=MD5, uri="sip:bluesip.net;transport=udp", > response="c6cdcafe0418e519bc9ee0d8fa3d4d74" > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 722 bytes from udp/[217.74.179.29]:5060 at 23:25:45.904176: > ------------------------------------------------------------------------ > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;branch=z9hG4bKetjKptHy71a8H > From: > ;transport=udp>;tag=6r0vBQZS650Fg > To: > ;transport=udp>;tag=0354a2e1b960c9cc2279eca4e5f84e20.1b16 > Call-ID: 2a3e4696-a5b9-11e1-a7ed-7b1f73a7ffcf > CSeq: 28601493 REGISTER > WWW-Authenticate: Digest realm="bluesip.net", > nonce="4fbec4a4ba82d1f1feaa32ccec01b85be054fcf7", stale=true > Server: Sip EXpress router (0.9.7 (i386/linux)) > Content-Length: 0 > Warning: 392 217.74.179.29:5060 "Noisy feedback tells: pid=5455 > req_src_ip=78.138.90.58 req_src_port=6090 in_uri=sip:bluesip.net;transport=udp > out_uri=sip:bluesip.net;transport=udp via_cnt==1" > > ------------------------------------------------------------------------ > send 916 bytes to udp/[217.74.179.29]:5060 at 23:25:45.904416: > ------------------------------------------------------------------------ > REGISTER sip:bluesip.net;transport=udp SIP/2.0 > Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKF3BcrN214a1tD > Max-Forwards: 70 > From: > ;transport=udp>;tag=6r0vBQZS650Fg > To: > ;transport=udp> > Call-ID: 2a3e4696-a5b9-11e1-a7ed-7b1f73a7ffcf > CSeq: 28601494 REGISTER > Contact: ;transport=udp;gw=trunk_1000> > Expires: 3600 > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Authorization: Digest username="bluesip/sherifomran", realm=" > bluesip.net", nonce="4fbec4a4ba82d1f1feaa32ccec01b85be054fcf7", > algorithm=MD5, uri="sip:bluesip.net;transport=udp", > response="4c09dbe4b9accb52d4104b40dfe20040" > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 699 bytes from udp/[217.74.179.29]:5060 at 23:25:45.935046: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;branch=z9hG4bKF3BcrN214a1tD > From: > ;transport=udp>;tag=6r0vBQZS650Fg > To: > ;transport=udp>;tag=0354a2e1b960c9cc2279eca4e5f84e20.2017 > Call-ID: 2a3e4696-a5b9-11e1-a7ed-7b1f73a7ffcf > CSeq: 28601494 REGISTER > Contact: ;transport=udp;gw=trunk_1000>;q=0.5;expires=3600 > Server: Sip EXpress router (0.9.7 (i386/linux)) > Content-Length: 0 > Warning: 392 217.74.179.29:5060 "Noisy feedback tells: pid=5462 > req_src_ip=78.138.90.58 req_src_port=6090 in_uri=sip:bluesip.net;transport=udp > out_uri=sip:bluesip.net;transport=udp via_cnt==1" > > ------------------------------------------------------------------------ > recv 564 bytes from udp/[41.34.123.243]:58881 at 23:27:59.977057: > ------------------------------------------------------------------------ > REGISTER sip:sip.pcfone.com:6090 SIP/2.0 > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport;branch=z9hG4bKPjEOP.a4NoXqoFqC7a.k2tyeX-mYlUyH4H > Max-Forwards: 70 > From: "Sherif 1003" > >;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K > To: "Sherif 1003" > > > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60932 REGISTER > User-Agent: Telephone 1.0.2 > Contact: "Sherif 1003" > Expires: 300 > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, > REFER, MESSAGE, OPTIONS > Content-Length: 0 > > ------------------------------------------------------------------------ > send 744 bytes to udp/[41.34.123.243]:58881 at 23:27:59.977960: > ------------------------------------------------------------------------ > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport=58881;branch=z9hG4bKPjEOP.a4NoXqoFqC7a.k2tyeX-mYlUyH4H;received= > 41.34.123.243 > From: "Sherif 1003" > >;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K > To: "Sherif 1003" > >;tag=4ajgB89Nt8Q3K > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60932 REGISTER > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > WWW-Authenticate: Digest realm="sip.pcfone.com", > nonce="19b7cc40-a5f8-11e1-a83f-7b1f73a7ffcf", algorithm=MD5, qop="auth" > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 564 bytes from udp/[41.34.123.243]:58881 at 23:28:00.468487: > ------------------------------------------------------------------------ > REGISTER sip:sip.pcfone.com:6090 SIP/2.0 > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport;branch=z9hG4bKPjEOP.a4NoXqoFqC7a.k2tyeX-mYlUyH4H > Max-Forwards: 70 > From: "Sherif 1003" > >;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K > To: "Sherif 1003" > > > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60932 REGISTER > User-Agent: Telephone 1.0.2 > Contact: "Sherif 1003" > Expires: 300 > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, > REFER, MESSAGE, OPTIONS > Content-Length: 0 > > ------------------------------------------------------------------------ > send 744 bytes to udp/[41.34.123.243]:58881 at 23:28:00.468679: > ------------------------------------------------------------------------ > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport=58881;branch=z9hG4bKPjEOP.a4NoXqoFqC7a.k2tyeX-mYlUyH4H;received= > 41.34.123.243 > From: "Sherif 1003" > >;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K > To: "Sherif 1003" > >;tag=4ajgB89Nt8Q3K > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60932 REGISTER > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > WWW-Authenticate: Digest realm="sip.pcfone.com", > nonce="19b7cc40-a5f8-11e1-a83f-7b1f73a7ffcf", algorithm=MD5, qop="auth" > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 830 bytes from udp/[41.34.123.243]:58881 at 23:28:00.717446: > ------------------------------------------------------------------------ > REGISTER sip:sip.pcfone.com:6090 SIP/2.0 > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport;branch=z9hG4bKPj0iHd3-XUjnvcBs9DiJUCWzrmYTmWxFLt > Max-Forwards: 70 > From: "Sherif 1003" > >;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K > To: "Sherif 1003" > > > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60933 REGISTER > User-Agent: Telephone 1.0.2 > Contact: "Sherif 1003" > Expires: 300 > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, > REFER, MESSAGE, OPTIONS > Authorization: Digest username="1002", realm="sip.pcfone.com", > nonce="19b7cc40-a5f8-11e1-a83f-7b1f73a7ffcf", uri="sip:sip.pcfone.com:6090", > response="e21c903e6d98b5adcdd2eefa75217157", algorithm=MD5, > cnonce="sc3AJfhuev18Bs-L-6tORmI.RXlxnPOz", qop=auth, nc=00000001 > Content-Length: 0 > > ------------------------------------------------------------------------ > send 706 bytes to udp/[41.34.123.243]:58881 at 23:28:00.719947: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport=58881;branch=z9hG4bKPj0iHd3-XUjnvcBs9DiJUCWzrmYTmWxFLt;received= > 41.34.123.243 > From: "Sherif 1003" > >;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K > To: "Sherif 1003" > >;tag=5KB9c3tSQHepF > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60933 REGISTER > Contact: ;expires=300 > Date: Thu, 24 May 2012 23:28:00 GMT > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Content-Length: 0 > > ------------------------------------------------------------------------ > send 989 bytes to udp/[41.34.123.243]:58881 at 23:28:00.868368: > ------------------------------------------------------------------------ > NOTIFY sip:1002 at 192.168.178.26:58881;ob SIP/2.0 > Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKgc54SgK51KQDS > Route: ;ob > Max-Forwards: 70 > From: > >;tag=6v41eyBXmt48a > To: > > > Call-ID: f19dfbce-209a-1230-8985-00163e6bb553 > CSeq: 28614976 NOTIFY > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > 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=noresource > Content-Type: application/simple-message-summary > Content-Length: 91 > > Messages-Waiting: yes > Message-Account: sip:1002 at 78.138.90.58 > Voice-Message: 1/0 (0/0) > > ------------------------------------------------------------------------ > recv 308 bytes from udp/[41.34.123.243]:58881 at 23:28:01.128279: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;received=78.138.90.58;branch=z9hG4bKgc54SgK51KQDS > Call-ID: f19dfbce-209a-1230-8985-00163e6bb553 > From: > >;tag=6v41eyBXmt48a > To: > >;tag=z9hG4bKgc54SgK51KQDS > CSeq: 28614976 NOTIFY > Content-Length: 0 > > ------------------------------------------------------------------------ > send 913 bytes to udp/[217.74.179.29]:5060 at 23:28:11.885213: > ------------------------------------------------------------------------ > REGISTER sip:bluesip.net;transport=udp SIP/2.0 > Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKHNyXUB48yvD0m > Max-Forwards: 70 > From: > ;transport=udp>;tag=5F739Uep9vaXm > To: > ;transport=udp> > Call-ID: 2a3e49c0-a5b9-11e1-a7ee-7b1f73a7ffcf > CSeq: 28601493 REGISTER > Contact: ;transport=udp;gw=trunk_1002> > Expires: 3600 > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Authorization: Digest username="bluesip/salahomran", realm="bluesip.net", > nonce="4fbebeca79a5cff7c417463814cb9054d313bcac", algorithm=MD5, uri="sip: > bluesip.net;transport=udp", response="bb1babb90f4ea1dc8fbb9aa45d6038c7" > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 720 bytes from udp/[217.74.179.29]:5060 at 23:28:11.915764: > ------------------------------------------------------------------------ > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;branch=z9hG4bKHNyXUB48yvD0m > From: > ;transport=udp>;tag=5F739Uep9vaXm > To: > ;transport=udp>;tag=0354a2e1b960c9cc2279eca4e5f84e20.3454 > Call-ID: 2a3e49c0-a5b9-11e1-a7ee-7b1f73a7ffcf > CSeq: 28601493 REGISTER > WWW-Authenticate: Digest realm="bluesip.net", > nonce="4fbec5365af3a174f1cc0b8cda770c727d25bf71", stale=true > Server: Sip EXpress router (0.9.7 (i386/linux)) > Content-Length: 0 > Warning: 392 217.74.179.29:5060 "Noisy feedback tells: pid=5462 > req_src_ip=78.138.90.58 req_src_port=6090 in_uri=sip:bluesip.net;transport=udp > out_uri=sip:bluesip.net;transport=udp via_cnt==1" > > ------------------------------------------------------------------------ > send 913 bytes to udp/[217.74.179.29]:5060 at 23:28:11.915914: > ------------------------------------------------------------------------ > REGISTER sip:bluesip.net;transport=udp SIP/2.0 > Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKjyQpX6mcv53jg > Max-Forwards: 70 > From: > ;transport=udp>;tag=5F739Uep9vaXm > To: > ;transport=udp> > Call-ID: 2a3e49c0-a5b9-11e1-a7ee-7b1f73a7ffcf > CSeq: 28601494 REGISTER > Contact: ;transport=udp;gw=trunk_1002> > Expires: 3600 > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Authorization: Digest username="bluesip/salahomran", realm="bluesip.net", > nonce="4fbec5365af3a174f1cc0b8cda770c727d25bf71", algorithm=MD5, uri="sip: > bluesip.net;transport=udp", response="e260f6dc9b01ce8bea8c5ffb36bd4ddd" > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 697 bytes from udp/[217.74.179.29]:5060 at 23:28:11.946325: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;branch=z9hG4bKjyQpX6mcv53jg > From: > ;transport=udp>;tag=5F739Uep9vaXm > To: > ;transport=udp>;tag=0354a2e1b960c9cc2279eca4e5f84e20.4b5c > Call-ID: 2a3e49c0-a5b9-11e1-a7ee-7b1f73a7ffcf > CSeq: 28601494 REGISTER > Contact: ;transport=udp;gw=trunk_1002>;q=0.5;expires=3600 > Server: Sip EXpress router (0.9.7 (i386/linux)) > Content-Length: 0 > Warning: 392 217.74.179.29:5060 "Noisy feedback tells: pid=5455 > req_src_ip=78.138.90.58 req_src_port=6090 in_uri=sip:bluesip.net;transport=udp > out_uri=sip:bluesip.net;transport=udp via_cnt==1" > > ------------------------------------------------------------------------ > recv 564 bytes from udp/[41.34.123.243]:58881 at 23:32:55.930149: > ------------------------------------------------------------------------ > REGISTER sip:sip.pcfone.com:6090 SIP/2.0 > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport;branch=z9hG4bKPjuk8DZGiaXfN7BmtslNHVjtSg5UXsbHXc > Max-Forwards: 70 > From: "Sherif 1003" > >;tag=Pz.t-0tGXj3saM.ehBiB1EAMmmitze1D > To: "Sherif 1003" > > > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60934 REGISTER > User-Agent: Telephone 1.0.2 > Contact: "Sherif 1003" > Expires: 300 > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, > REFER, MESSAGE, OPTIONS > Content-Length: 0 > > ------------------------------------------------------------------------ > send 744 bytes to udp/[41.34.123.243]:58881 at 23:32:55.931324: > ------------------------------------------------------------------------ > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport=58881;branch=z9hG4bKPjuk8DZGiaXfN7BmtslNHVjtSg5UXsbHXc;received= > 41.34.123.243 > From: "Sherif 1003" > >;tag=Pz.t-0tGXj3saM.ehBiB1EAMmmitze1D > To: "Sherif 1003" > >;tag=75XtgSv0H3tUp > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60934 REGISTER > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > WWW-Authenticate: Digest realm="sip.pcfone.com", > nonce="ca1eabee-a5f8-11e1-a840-7b1f73a7ffcf", algorithm=MD5, qop="auth" > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 830 bytes from udp/[41.34.123.243]:58881 at 23:32:56.165624: > ------------------------------------------------------------------------ > REGISTER sip:sip.pcfone.com:6090 SIP/2.0 > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport;branch=z9hG4bKPjKEQ6xXmyzZxA6vCM88vHNYAOPAjUqFsD > Max-Forwards: 70 > From: "Sherif 1003" > >;tag=Pz.t-0tGXj3saM.ehBiB1EAMmmitze1D > To: "Sherif 1003" > > > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60935 REGISTER > User-Agent: Telephone 1.0.2 > Contact: "Sherif 1003" > Expires: 300 > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, > REFER, MESSAGE, OPTIONS > Authorization: Digest username="1002", realm="sip.pcfone.com", > nonce="ca1eabee-a5f8-11e1-a840-7b1f73a7ffcf", uri="sip:sip.pcfone.com:6090", > response="ff6d12f4ec7abd4b37187222b3af2358", algorithm=MD5, > cnonce="sc3AJfhuev18Bs-L-6tORmI.RXlxnPOz", qop=auth, nc=00000001 > Content-Length: 0 > > ------------------------------------------------------------------------ > send 706 bytes to udp/[41.34.123.243]:58881 at 23:32:56.168033: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport=58881;branch=z9hG4bKPjKEQ6xXmyzZxA6vCM88vHNYAOPAjUqFsD;received= > 41.34.123.243 > From: "Sherif 1003" > >;tag=Pz.t-0tGXj3saM.ehBiB1EAMmmitze1D > To: "Sherif 1003" > >;tag=8eQKjmD4ecHej > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60935 REGISTER > Contact: ;expires=300 > Date: Thu, 24 May 2012 23:32:56 GMT > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Content-Length: 0 > > ------------------------------------------------------------------------ > send 989 bytes to udp/[41.34.123.243]:58881 at 23:32:56.360238: > ------------------------------------------------------------------------ > NOTIFY sip:1002 at 192.168.178.26:58881;ob SIP/2.0 > Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKK7gFZ15FSet5B > Route: ;ob > Max-Forwards: 70 > From: > >;tag=9QgcmFy7BN70D > To: > > > Call-ID: a1be7708-209b-1230-8985-00163e6bb553 > CSeq: 28615124 NOTIFY > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > 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=noresource > Content-Type: application/simple-message-summary > Content-Length: 91 > > Messages-Waiting: yes > Message-Account: sip:1002 at 78.138.90.58 > Voice-Message: 1/0 (0/0) > > ------------------------------------------------------------------------ > recv 308 bytes from udp/[41.34.123.243]:58881 at 23:32:56.623554: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;received=78.138.90.58;branch=z9hG4bKK7gFZ15FSet5B > Call-ID: a1be7708-209b-1230-8985-00163e6bb553 > From: > >;tag=9QgcmFy7BN70D > To: > >;tag=z9hG4bKK7gFZ15FSet5B > CSeq: 28615124 NOTIFY > Content-Length: 0 > > ------------------------------------------------------------------------ > freeswitch at internal> > > > --- On *Thu, 5/24/12, Kristian Kielhofner > >* wrote: > > > From: Kristian Kielhofner > > > Subject: Re: [Freeswitch-users] Freeswitch (SBC + Media) does not know > that gateway phone is picked up > To: "FreeSWITCH Users Help" > > > Date: Thursday, May 24, 2012, 7:51 PM > > > Siptrace and logs please. > > On Thu, May 24, 2012 at 11:53 AM, Sherif Omran > > > wrote: > > > > Hi all, > > > > My topology is as follows: > > > > Kamailio -> FS (SBS+Media server) > > > > I came across an issue with my system as follows. I have a Hardphone > registered. When I do local call inside kamailio, it gets to FS and returns > back well and FS understands when I lift the handset. However, I added a > gateway (german landline server), when I call my self from another phone, > the call gets to FS and then transmits to Kamailio, it rings my extension > but when I lift the handset FS does not notice it and keeps ringing. > > > > Any body has an Idea? Here is my gateway trunk. > > > > > > > > > > > > > > > > > > > > > > thanks in advance > > Sherif Omran > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > Kristian Kielhofner > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -----Inline Attachment Follows----- > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120525/6748271c/attachment-0001.html From govoiper at gmail.com Fri May 25 11:46:16 2012 From: govoiper at gmail.com (SamyGo) Date: Fri, 25 May 2012 12:46:16 +0500 Subject: [Freeswitch-users] Freeswitch (SBC + Media) does not know that gateway phone is picked up In-Reply-To: <1337930432.2677.YahooMailClassic@web110809.mail.gq1.yahoo.com> References: <1337930432.2677.YahooMailClassic@web110809.mail.gq1.yahoo.com> Message-ID: Hi again, If you want kamailio register to the provider then use UAC module. Kamailio will use the username/password and register with the provider. Regards, Sammy On Fri, May 25, 2012 at 12:20 PM, Sherif Omran wrote: > Dear Sammy, > > Thank you for your question ... Yes, the GW is registered through FS > because I did not know how to register it to kamailio. But it seems better > to register it to kamailio. > One more information, calls from 1001 to 1002 go to kamailio then to FS > then return back to kamailio smoothly. Thus I would suggest that I change > registering the gateway from FS to kamailio. but How to? > > > > --- On *Fri, 5/25/12, SamyGo * wrote: > > > From: SamyGo > Subject: Re: [Freeswitch-users] Freeswitch (SBC + Media) does not know > that gateway phone is picked up > To: "FreeSWITCH Users Help" > Date: Friday, May 25, 2012, 8:08 AM > > Hi, > These are SIP traces on FreeSWITCH console, whereas you are saying and it > seems that kamailio is not detecting the answering (200 OK)of the call from > extension 1002. Please, can you take a sip trace..!! I see you've both > kamailio and FS on same server! Please take a pcap from the linux console > using the following command. > > #tcpdump -i any -s 0 -w new-call-trace.pcap -vvv > > Please be quick on turning the sip trace on and off as quickly as possible > to avoid extra packets. Once done open the file in wireshark > ; apply filter "sip || rtp" and then save the > resulting capture in separate file. Send us the new file to analyse. > > One more silly question probably, I see REGISTERs coming to your FS as > well and the calls to gateways are made from FS too !!, umm...just thinking > what are you using kamailio for!!? > > Thanks > BR, > Sammy > > > On Fri, May 25, 2012 at 4:40 AM, Sherif Omran > > wrote: > > Hi all, > > here is the siptrace: To figure it out > 1- gateway called bluesip.net. It send invide using caller > number at bluesip.net > 2- This call should go to extension kb-1002. kb means go from freeswitch > port 6090 to kamailio port 5060 > 3- It should go to call extension 1002 in Kamailio > 4- Extension 1002 rings but when I reply, it does not notice I replied > > > ./fs_cli > _____ ____ ____ _ ___ > | ___/ ___| / ___| | |_ _| > | |_ \___ \ | | | | | | > | _| ___) | | |___| |___ | | > |_| |____/ \____|_____|___| > > ******************************************************* > * Anthony Minessale II, Ken Rice, * > * Michael Jerris, Travis Cross * > * FreeSWITCH (http://www.freeswitch.org) * > * Paypal Donations Appreciated: paypal at freeswitch.org* > * Brought to you by ClueCon http://www.cluecon.com/ * > ******************************************************* > > Type /help to see a list of commands > > > +OK log level [7] > freeswitch at internal> tracelevel > -ERR tracelevel Command not found! > > freeswitch at internal> sofia global siptrace on > +OK Global siptrace on > recv 1104 bytes from udp/[217.74.179.29]:5060 at 23:08:45.822962: > ------------------------------------------------------------------------ > INVITE sip:gw+trunk_1000 at 78.138.90.58:6090;transport=udp;gw=trunk_1000 > SIP/2.0 > Record-Route: > Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 > Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 > From: "+41793940965" ;tag=as00589402 > To: > Contact: > Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net > CSeq: 102 INVITE > User-Agent: blueSIP PSTN GW > Max-Forwards: 69 > Date: Thu, 24 May 2012 23:08:44 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Content-Type: application/sdp > Content-Length: 367 > P-hint: USRLOC > > v=0 > o=root 20076 20076 IN IP4 217.74.179.28 > s=session > c=IN IP4 217.74.179.28 > t=0 0 > m=audio 25626 RTP/AVP 8 0 18 111 3 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:111 G726-32/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > a=sendrecv > ------------------------------------------------------------------------ > send 494 bytes to udp/[217.74.179.29]:5060 at 23:08:45.844486: > ------------------------------------------------------------------------ > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 > Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 > Record-Route: > From: "+41793940965" ;tag=as00589402 > To: > Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net > CSeq: 102 INVITE > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Content-Length: 0 > > ------------------------------------------------------------------------ > 2012-05-24 23:08:45.840749 [NOTICE] switch_channel.c:926 New Channel > sofia/internal/+41793940965 at bluesip.net[69cd22e6-a5f5-11e1-a833-7b1f73a7ffcf] > 2012-05-24 23:08:45.840749 [DEBUG] sofia.c:5532 Channel sofia/internal/ > +41793940965 at bluesip.net entering state [received][100] > 2012-05-24 23:08:45.840749 [DEBUG] sofia.c:5543 Remote SDP: > v=0 > o=root 20076 20076 IN IP4 217.74.179.28 > s=session > c=IN IP4 217.74.179.28 > t=0 0 > m=audio 25626 RTP/AVP 8 0 18 111 3 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:111 G726-32/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [PCMA:8:8000:20:64000]/[iLBC:97:8000:30:13330] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [PCMU:0:8000:20:64000]/[iLBC:97:8000:30:13330] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [G729:18:8000:20:8000]/[iLBC:97:8000:30:13330] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [G726-32:111:8000:20:0]/[iLBC:97:8000:30:13330] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [GSM:3:8000:20:13200]/[iLBC:97:8000:30:13330] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [telephone-event:101:8000:20:0]/[iLBC:97:8000:30:13330] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf > send/recv payload to 101 > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [PCMA:8:8000:20:64000]/[G7221:115:32000:20:48000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [PCMU:0:8000:20:64000]/[G7221:115:32000:20:48000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [G729:18:8000:20:8000]/[G7221:115:32000:20:48000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [G726-32:111:8000:20:0]/[G7221:115:32000:20:48000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [GSM:3:8000:20:13200]/[G7221:115:32000:20:48000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [telephone-event:101:8000:20:0]/[G7221:115:32000:20:48000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf > send/recv payload to 101 > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [PCMA:8:8000:20:64000]/[G7221:107:16000:20:32000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [PCMU:0:8000:20:64000]/[G7221:107:16000:20:32000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [G729:18:8000:20:8000]/[G7221:107:16000:20:32000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [G726-32:111:8000:20:0]/[G7221:107:16000:20:32000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [GSM:3:8000:20:13200]/[G7221:107:16000:20:32000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [telephone-event:101:8000:20:0]/[G7221:107:16000:20:32000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf > send/recv payload to 101 > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [PCMA:8:8000:20:64000]/[G722:9:8000:20:64000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [G729:18:8000:20:8000]/[G722:9:8000:20:64000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [G726-32:111:8000:20:0]/[G722:9:8000:20:64000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [GSM:3:8000:20:13200]/[G722:9:8000:20:64000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [telephone-event:101:8000:20:0]/[G722:9:8000:20:64000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf > send/recv payload to 101 > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [PCMA:8:8000:20:64000]/[PCMU:0:8000:20:64000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:2996 Set Codec > sofia/internal/+41793940965 at bluesip.net PCMU/8000 20 ms 160 samples 64000 > bits > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_codec.c:111 sofia/internal/ > +41793940965 at bluesip.net Original read codec set to PCMU:0 > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf > send/recv payload to 101 > 2012-05-24 23:08:45.840749 [DEBUG] sofia.c:5757 (sofia/internal/ > +41793940965 at bluesip.net) State Change CS_NEW -> CS_INIT > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_INIT > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:401 > (sofia/internal/+41793940965 at bluesip.net) State INIT > 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:85 sofia/internal/ > +41793940965 at bluesip.net SOFIA INIT > 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:125 (sofia/internal/ > +41793940965 at bluesip.net) State Change CS_INIT -> CS_ROUTING > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:401 > (sofia/internal/+41793940965 at bluesip.net) State INIT going to sleep > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_ROUTING > 2012-05-24 23:08:45.840749 [DEBUG] switch_channel.c:1886 (sofia/internal/ > +41793940965 at bluesip.net) Callstate Change DOWN -> RINGING > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 > (sofia/internal/+41793940965 at bluesip.net) State ROUTING > 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:148 sofia/internal/ > +41793940965 at bluesip.net SOFIA ROUTING > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:104 > sofia/internal/+41793940965 at bluesip.net Standard ROUTING > 2012-05-24 23:08:45.840749 [INFO] mod_dialplan_xml.c:485 Processing > +41793940965 <+41793940965>->kb-1002 in context public > Dialplan: sofia/internal/+41793940965 at bluesip.net parsing > [public->from_kamailio] continue=false > Dialplan: sofia/internal/+41793940965 at bluesip.net Regex (PASS) > [from_kamailio] destination_number(kb-1002) =~ /^(.+)$/ break=on-false > Dialplan: sofia/internal/+41793940965 at bluesip.net Action transfer(kb-1002 > XML default) > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:154 > (sofia/internal/+41793940965 at bluesip.net) State Change CS_ROUTING -> > CS_EXECUTE > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 > (sofia/internal/+41793940965 at bluesip.net) State ROUTING going to sleep > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_EXECUTE > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:417 > (sofia/internal/+41793940965 at bluesip.net) State EXECUTE > 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:241 sofia/internal/ > +41793940965 at bluesip.net SOFIA EXECUTE > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:192 > sofia/internal/+41793940965 at bluesip.net Standard EXECUTE > EXECUTE sofia/internal/+41793940965 at bluesip.net transfer(kb-1002 XML > default) > 2012-05-24 23:08:45.840749 [DEBUG] switch_ivr.c:1711 (sofia/internal/ > +41793940965 at bluesip.net) State Change CS_EXECUTE -> CS_ROUTING > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:731 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:08:45.840749 [NOTICE] switch_ivr.c:1717 Transfer > sofia/internal/+41793940965 at bluesip.net to XML[kb-1002 at default] > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:417 > (sofia/internal/+41793940965 at bluesip.net) State EXECUTE going to sleep > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_ROUTING > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 > (sofia/internal/+41793940965 at bluesip.net) State ROUTING > 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:148 sofia/internal/ > +41793940965 at bluesip.net SOFIA ROUTING > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:104 > sofia/internal/+41793940965 at bluesip.net Standard ROUTING > 2012-05-24 23:08:45.840749 [INFO] mod_dialplan_xml.c:485 Processing > +41793940965 <+41793940965>->kb-1002 in context default > Dialplan: sofia/internal/+41793940965 at bluesip.net parsing [default->vbox] > continue=false > Dialplan: sofia/internal/+41793940965 at bluesip.net Regex (FAIL) [vbox] > destination_number(kb-1002) =~ /^vb-([0-9][0-9][0-9])$/ break=on-false > Dialplan: sofia/internal/+41793940965 at bluesip.net parsing > [default->vmenu] continue=false > Dialplan: sofia/internal/+41793940965 at bluesip.net Regex (FAIL) [vmenu] > destination_number(kb-1002) =~ /^vm-([0-9][0-9][0-9][0-9])$/ break=on-false > Dialplan: sofia/internal/+41793940965 at bluesip.net parsing > [default->kbridge] continue=false > Dialplan: sofia/internal/+41793940965 at bluesip.net Regex (PASS) [kbridge] > destination_number(kb-1002) =~ /^kb-(.+)$/ break=on-false > Dialplan: sofia/internal/+41793940965 at bluesip.net Action > set(proxy_media=true) > Dialplan: sofia/internal/+41793940965 at bluesip.net Action > set(call_timeout=50) > Dialplan: sofia/internal/+41793940965 at bluesip.net Action > set(continue_on_fail=true) > Dialplan: sofia/internal/+41793940965 at bluesip.net Action > set(hangup_after_bridge=true) > Dialplan: sofia/internal/+41793940965 at bluesip.net Action > set(sip_invite_domain=78.138.90.58) > Dialplan: sofia/internal/+41793940965 at bluesip.net Action > export(sip_contact_user=ufs) > Dialplan: sofia/internal/+41793940965 at bluesip.net Action bridge(sofia/ > 78.138.90.58/1002 at 78.138.90.58:5060) > Dialplan: sofia/internal/+41793940965 at bluesip.net Action answer() > Dialplan: sofia/internal/+41793940965 at bluesip.net Action > voicemail(default ${domain_name} 1002) > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:154 > (sofia/internal/+41793940965 at bluesip.net) State Change CS_ROUTING -> > CS_EXECUTE > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 > (sofia/internal/+41793940965 at bluesip.net) State ROUTING going to sleep > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_EXECUTE > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:417 > (sofia/internal/+41793940965 at bluesip.net) State EXECUTE > 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:241 sofia/internal/ > +41793940965 at bluesip.net SOFIA EXECUTE > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:192 > sofia/internal/+41793940965 at bluesip.net Standard EXECUTE > EXECUTE sofia/internal/+41793940965 at bluesip.net set(proxy_media=true) > 2012-05-24 23:08:45.840749 [DEBUG] mod_dptools.c:1281 sofia/internal/ > +41793940965 at bluesip.net SET [proxy_media]=[true] > EXECUTE sofia/internal/+41793940965 at bluesip.net set(call_timeout=50) > 2012-05-24 23:08:45.840749 [DEBUG] mod_dptools.c:1281 sofia/internal/ > +41793940965 at bluesip.net SET [call_timeout]=[50] > EXECUTE sofia/internal/+41793940965 at bluesip.net set(continue_on_fail=true) > 2012-05-24 23:08:45.840749 [DEBUG] mod_dptools.c:1281 sofia/internal/ > +41793940965 at bluesip.net SET [continue_on_fail]=[true] > EXECUTE sofia/internal/+41793940965 at bluesip.netset(hangup_after_bridge=true) > 2012-05-24 23:08:45.840749 [DEBUG] mod_dptools.c:1281 sofia/internal/ > +41793940965 at bluesip.net SET [hangup_after_bridge]=[true] > EXECUTE sofia/internal/+41793940965 at bluesip.netset(sip_invite_domain=78.138.90.58) > 2012-05-24 23:08:45.840749 [DEBUG] mod_dptools.c:1281 sofia/internal/ > +41793940965 at bluesip.net SET [sip_invite_domain]=[78.138.90.58] > EXECUTE sofia/internal/+41793940965 at bluesip.netexport(sip_contact_user=ufs) > 2012-05-24 23:08:45.840749 [DEBUG] switch_channel.c:1093 EXPORT > (export_vars) [sip_contact_user]=[ufs] > EXECUTE sofia/internal/+41793940965 at bluesip.net bridge(sofia/ > 78.138.90.58/1002 at 78.138.90.58:5060) > 2012-05-24 23:08:45.840749 [DEBUG] switch_channel.c:1047 sofia/internal/ > +41793940965 at bluesip.net EXPORTING[export_vars] [sip_contact_user]=[ufs] > to event > 2012-05-24 23:08:45.840749 [DEBUG] switch_ivr_originate.c:1884 Parsing > global variables > 2012-05-24 23:08:45.840749 [NOTICE] switch_channel.c:926 New Channel > sofia/internal/1002 at 78.138.90.58:5060[69ce1f0c-a5f5-11e1-a838-7b1f73a7ffcf] > 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:4691 (sofia/internal/ > 1002 at 78.138.90.58:5060) State Change CS_NEW -> CS_INIT > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/1002 at 78.138.90.58:5060) Running State Change CS_INIT > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:401 > (sofia/internal/1002 at 78.138.90.58:5060) State INIT > 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:85 sofia/internal/ > 1002 at 78.138.90.58:5060 SOFIA INIT > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:1871 sofia/internal/ > 1002 at 78.138.90.58:5060 Patched SDP > --- > v=0 > o=root 20076 20076 IN IP4 217.74.179.28 > s=session > c=IN IP4 217.74.179.28 > t=0 0 > m=audio 25626 RTP/AVP 8 0 18 111 3 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:111 G726-32/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > > +++ > v=0 > o=FreeSWITCH 1629510322 1629510323 IN IP4 78.138.90.58 > s=FreeSWITCH > c=IN IP4 78.138.90.58 > t=0 0 > m=audio 31178 RTP/AVP 8 0 18 111 3 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:111 G726-32/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > > send 1340 bytes to udp/[78.138.90.58]:5060 at 23:08:45.852280: > ------------------------------------------------------------------------ > INVITE sip:1002 at 78.138.90.58:5060 SIP/2.0 > Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKapDgFeegKyHXK > Max-Forwards: 67 > From: "+41793940965" ;tag=tgNrSS1jQFS8F > To: > Call-ID: 412c8312-2098-1230-8985-00163e6bb553 > CSeq: 28614398 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > 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: 372 > P-hint: USRLOC > X-FS-Support: update_display,send_info > Remote-Party-ID: "+41793940965" >;party=calling;screen=yes;privacy=off > > v=0 > o=FreeSWITCH 1629510322 1629510323 IN IP4 78.138.90.58 > s=FreeSWITCH > c=IN IP4 78.138.90.58 > t=0 0 > m=audio 31178 RTP/AVP 8 0 18 111 3 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:111 G726-32/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > ------------------------------------------------------------------------ > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:125 (sofia/internal/ > 1002 at 78.138.90.58:5060) State Change CS_INIT -> CS_ROUTING > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:401 > (sofia/internal/1002 at 78.138.90.58:5060) State INIT going to sleep > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/1002 at 78.138.90.58:5060) Running State Change CS_ROUTING > 2012-05-24 23:08:45.840749 [DEBUG] switch_channel.c:1886 (sofia/internal/ > 1002 at 78.138.90.58:5060) Callstate Change DOWN -> RINGING > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 > (sofia/internal/1002 at 78.138.90.58:5060) State ROUTING > 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:148 sofia/internal/ > 1002 at 78.138.90.58:5060 SOFIA ROUTING > 2012-05-24 23:08:45.840749 [DEBUG] switch_ivr_originate.c:66 > (sofia/internal/1002 at 78.138.90.58:5060) State Change CS_ROUTING -> > CS_CONSUME_MEDIA > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 > (sofia/internal/1002 at 78.138.90.58:5060) State ROUTING going to sleep > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/1002 at 78.138.90.58:5060) Running State Change > CS_CONSUME_MEDIA > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:429 > (sofia/internal/1002 at 78.138.90.58:5060) State CONSUME_MEDIA > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:429 > (sofia/internal/1002 at 78.138.90.58:5060) State CONSUME_MEDIA going to sleep > 2012-05-24 23:08:45.840749 [DEBUG] sofia.c:5532 Channel sofia/internal/ > 1002 at 78.138.90.58:5060 entering state [calling][0] > recv 365 bytes from udp/[78.138.90.58]:5060 at 23:08:45.964309: > ------------------------------------------------------------------------ > SIP/2.0 100 trying -- your call is important to us > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;branch=z9hG4bKapDgFeegKyHXK > From: "+41793940965" ;tag=tgNrSS1jQFS8F > To: > Call-ID: 412c8312-2098-1230-8985-00163e6bb553 > CSeq: 28614398 INVITE > Server: kamailio (3.1.5 (x86_64/linux)) > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 496 bytes from udp/[78.138.90.58]:5060 at 23:08:47.414766: > ------------------------------------------------------------------------ > SIP/2.0 180 Ringing > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;branch=z9hG4bKapDgFeegKyHXK > Record-Route: > Call-ID: 412c8312-2098-1230-8985-00163e6bb553 > From: "+41793940965" ;tag=tgNrSS1jQFS8F > To: > >;tag=qCSaB2YJOAsAr2dJ-RWNMhPoDdn1RFvs > CSeq: 28614398 INVITE > Contact: "Mama" > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, REGISTER > Content-Length: 0 > > ------------------------------------------------------------------------ > 2012-05-24 23:08:47.400741 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:47.400741 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:47.400741 [DEBUG] sofia.c:5532 Channel sofia/internal/ > 1002 at 78.138.90.58:5060 entering state [proceeding][180] > 2012-05-24 23:08:47.400741 [NOTICE] sofia.c:5624 Ring-Ready sofia/internal/ > 1002 at 78.138.90.58:5060! > send 995 bytes to udp/[217.74.179.29]:5060 at 23:08:47.416538: > ------------------------------------------------------------------------ > SIP/2.0 180 Ringing > Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 > Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 > Record-Route: > From: "+41793940965" ;tag=as00589402 > To: ;tag=S7UZQygFt62Nm > Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net > CSeq: 102 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Accept: application/sdp > 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-Length: 0 > Remote-Party-ID: "Outbound Call" > >;party=calling;privacy=off;screen=no > > ------------------------------------------------------------------------ > 2012-05-24 23:08:47.400741 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:08:47.400741 [NOTICE] mod_sofia.c:2514 Ring-Ready > sofia/internal/+41793940965 at bluesip.net! > 2012-05-24 23:08:47.400741 [DEBUG] sofia.c:5532 Channel sofia/internal/ > +41793940965 at bluesip.net entering state [early][180] > 2012-05-24 23:08:47.400741 [DEBUG] switch_core_session.c:731 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:08:47.400741 [NOTICE] switch_ivr_originate.c:483 Ring Ready > sofia/internal/+41793940965 at bluesip.net! > recv 496 bytes from udp/[78.138.90.58]:5060 at 23:08:47.804967: > ------------------------------------------------------------------------ > SIP/2.0 180 Ringing > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;branch=z9hG4bKapDgFeegKyHXK > Record-Route: > From: "+41793940965" ;tag=tgNrSS1jQFS8F > To: ;tag=549D2DD03BBA7C67 > Call-ID: 412c8312-2098-1230-8985-00163e6bb553 > CSeq: 28614398 INVITE > Contact: > ;uniq=6AC0DF4D2E498C8ACA82CB22226B9> > User-Agent: AVM FRITZ!Box Fon WLAN 7170 29.04.80 (Jan 27 2010) > Content-Length: 0 > > ------------------------------------------------------------------------ > 2012-05-24 23:08:47.800746 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:47.800746 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:47.800746 [DEBUG] sofia.c:5532 Channel sofia/internal/ > 1002 at 78.138.90.58:5060 entering state [proceeding][180] > recv 496 bytes from udp/[78.138.90.58]:5060 at 23:08:48.182098: > ------------------------------------------------------------------------ > SIP/2.0 180 Ringing > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;branch=z9hG4bKapDgFeegKyHXK > Record-Route: > Call-ID: 412c8312-2098-1230-8985-00163e6bb553 > From: "+41793940965" ;tag=tgNrSS1jQFS8F > To: > >;tag=qCSaB2YJOAsAr2dJ-RWNMhPoDdn1RFvs > CSeq: 28614398 INVITE > Contact: "Mama" > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, REGISTER > Content-Length: 0 > > ------------------------------------------------------------------------ > 2012-05-24 23:08:48.180741 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:48.180741 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:48.180741 [DEBUG] sofia.c:5532 Channel sofia/internal/ > 1002 at 78.138.90.58:5060 entering state [proceeding][180] > recv 496 bytes from udp/[78.138.90.58]:5060 at 23:08:48.417403: > ------------------------------------------------------------------------ > SIP/2.0 180 Ringing > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;branch=z9hG4bKapDgFeegKyHXK > Record-Route: > From: "+41793940965" ;tag=tgNrSS1jQFS8F > To: ;tag=549D2DD03BBA7C67 > Call-ID: 412c8312-2098-1230-8985-00163e6bb553 > CSeq: 28614398 INVITE > Contact: > ;uniq=6AC0DF4D2E498C8ACA82CB22226B9> > User-Agent: AVM FRITZ!Box Fon WLAN 7170 29.04.80 (Jan 27 2010) > Content-Length: 0 > > ------------------------------------------------------------------------ > 2012-05-24 23:08:48.400744 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:48.400744 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:48.400744 [DEBUG] sofia.c:5532 Channel sofia/internal/ > 1002 at 78.138.90.58:5060 entering state [proceeding][180] > recv 496 bytes from udp/[78.138.90.58]:5060 at 23:08:49.292599: > ------------------------------------------------------------------------ > SIP/2.0 180 Ringing > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;branch=z9hG4bKapDgFeegKyHXK > Record-Route: > From: "+41793940965" ;tag=tgNrSS1jQFS8F > To: ;tag=549D2DD03BBA7C67 > Call-ID: 412c8312-2098-1230-8985-00163e6bb553 > CSeq: 28614398 INVITE > Contact: > ;uniq=6AC0DF4D2E498C8ACA82CB22226B9> > User-Agent: AVM FRITZ!Box Fon WLAN 7170 29.04.80 (Jan 27 2010) > Content-Length: 0 > > ------------------------------------------------------------------------ > 2012-05-24 23:08:49.280742 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:49.280742 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:49.280742 [DEBUG] sofia.c:5532 Channel sofia/internal/ > 1002 at 78.138.90.58:5060 entering state [proceeding][180] > recv 478 bytes from udp/[217.74.179.29]:5060 at 23:09:15.066012: > ------------------------------------------------------------------------ > CANCEL sip:gw+trunk_1000 at 78.138.90.58:6090;transport=udp;gw=trunk_1000 > SIP/2.0 > Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 > Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 > From: "+41793940965" ;tag=as00589402 > To: > Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net > CSeq: 102 CANCEL > User-Agent: blueSIP PSTN GW > Max-Forwards: 69 > Content-Length: 0 > P-hint: USRLOC > > ------------------------------------------------------------------------ > send 369 bytes to udp/[217.74.179.29]:5060 at 23:09:15.066176: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 > Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 > From: "+41793940965" ;tag=as00589402 > To: ;tag=S7UZQygFt62Nm > Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net > CSeq: 102 CANCEL > Content-Length: 0 > > ------------------------------------------------------------------------ > send 766 bytes to udp/[217.74.179.29]:5060 at 23:09:15.066289: > ------------------------------------------------------------------------ > SIP/2.0 487 Request Terminated > Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 > Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 > From: "+41793940965" ;tag=as00589402 > To: ;tag=S7UZQygFt62Nm > Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net > CSeq: 102 INVITE > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > 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-Length: 0 > > ------------------------------------------------------------------------ > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:09:15.060742 [DEBUG] sofia.c:5532 Channel sofia/internal/ > +41793940965 at bluesip.net entering state [terminated][487] > 2012-05-24 23:09:15.060742 [DEBUG] switch_channel.c:2848 (sofia/internal/ > +41793940965 at bluesip.net) Callstate Change RINGING -> HANGUP > 2012-05-24 23:09:15.060742 [NOTICE] sofia.c:6301 Hangup sofia/internal/ > +41793940965 at bluesip.net [CS_EXECUTE] [ORIGINATOR_CANCEL] > 2012-05-24 23:09:15.060742 [DEBUG] switch_channel.c:2871 Send signal > sofia/internal/+41793940965 at bluesip.net [KILL] > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:09:15.060742 [DEBUG] switch_channel.c:2848 (sofia/internal/ > 1002 at 78.138.90.58:5060) Callstate Change RINGING -> HANGUP > 2012-05-24 23:09:15.060742 [NOTICE] switch_ivr_originate.c:3182 Hangup > sofia/internal/1002 at 78.138.90.58:5060 [CS_CONSUME_MEDIA] > [ORIGINATOR_CANCEL] > 2012-05-24 23:09:15.060742 [DEBUG] switch_channel.c:2871 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [KILL] > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/1002 at 78.138.90.58:5060) Running State Change CS_HANGUP > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:602 > (sofia/internal/1002 at 78.138.90.58:5060) State HANGUP > 2012-05-24 23:09:15.060742 [DEBUG] mod_sofia.c:469 Channel sofia/internal/ > 1002 at 78.138.90.58:5060 hanging up, cause: ORIGINATOR_CANCEL > 2012-05-24 23:09:15.060742 [DEBUG] mod_sofia.c:523 Sending CANCEL to > sofia/internal/1002 at 78.138.90.58:5060 > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:47 > sofia/internal/1002 at 78.138.90.58:5060 Standard HANGUP, cause: > ORIGINATOR_CANCEL > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:602 > (sofia/internal/1002 at 78.138.90.58:5060) State HANGUP going to sleep > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:393 > (sofia/internal/1002 at 78.138.90.58:5060) State Change CS_HANGUP -> > CS_REPORTING > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/1002 at 78.138.90.58:5060) Running State Change CS_REPORTING > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:662 > (sofia/internal/1002 at 78.138.90.58:5060) State REPORTING > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:79 > sofia/internal/1002 at 78.138.90.58:5060 Standard REPORTING, cause: > ORIGINATOR_CANCEL > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:662 > (sofia/internal/1002 at 78.138.90.58:5060) State REPORTING going to sleep > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:387 > (sofia/internal/1002 at 78.138.90.58:5060) State Change CS_REPORTING -> > CS_DESTROY > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1382 Session 9 > (sofia/internal/1002 at 78.138.90.58:5060) Locked, Waiting on external > entities > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:09:15.060742 [DEBUG] switch_ivr_originate.c:3358 Originate > Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL] > 2012-05-24 23:09:15.060742 [NOTICE] switch_core_session.c:1400 Session 9 > (sofia/internal/1002 at 78.138.90.58:5060) Ended > 2012-05-24 23:09:15.060742 [NOTICE] switch_core_session.c:1402 Close > Channel sofia/internal/1002 at 78.138.90.58:5060 [CS_DESTROY] > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:491 > (sofia/internal/1002 at 78.138.90.58:5060) Callstate Change HANGUP -> DOWN > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:494 > (sofia/internal/1002 at 78.138.90.58:5060) Running State Change CS_DESTROY > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:504 > (sofia/internal/1002 at 78.138.90.58:5060) State DESTROY > 2012-05-24 23:09:15.060742 [DEBUG] mod_sofia.c:374 sofia/internal/ > 1002 at 78.138.90.58:5060 SOFIA DESTROY > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:86 > sofia/internal/1002 at 78.138.90.58:5060 Standard DESTROY > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:504 > (sofia/internal/1002 at 78.138.90.58:5060) State DESTROY going to sleep > 2012-05-24 23:09:15.060742 [INFO] mod_dptools.c:2922 Originate Failed. > Cause: ORIGINATOR_CANCEL > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:2287 > sofia/internal/+41793940965 at bluesip.net skip receive message > [APPLICATION_EXEC_COMPLETE] (channel is hungup already) > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:417 > (sofia/internal/+41793940965 at bluesip.net) State EXECUTE going to sleep > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_HANGUP > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:602 > (sofia/internal/+41793940965 at bluesip.net) State HANGUP > 2012-05-24 23:09:15.060742 [DEBUG] mod_sofia.c:469 Channel sofia/internal/ > +41793940965 at bluesip.net hanging up, cause: ORIGINATOR_CANCEL > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:47 > sofia/internal/+41793940965 at bluesip.net Standard HANGUP, cause: > ORIGINATOR_CANCEL > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:602 > (sofia/internal/+41793940965 at bluesip.net) State HANGUP going to sleep > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:393 > (sofia/internal/+41793940965 at bluesip.net) State Change CS_HANGUP -> > CS_REPORTING > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/+41793940965 at bluesip.net) Running State Change > CS_REPORTING > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:662 > (sofia/internal/+41793940965 at bluesip.net) State REPORTING > send 383 bytes to udp/[78.138.90.58]:5060 at 23:09:15.087695: > ------------------------------------------------------------------------ > CANCEL sip:1002 at 78.138.90.58:5060 SIP/2.0 > Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKapDgFeegKyHXK > Max-Forwards: 67 > From: "+41793940965" ;tag=tgNrSS1jQFS8F > To: > Call-ID: 412c8312-2098-1230-8985-00163e6bb553 > CSeq: 28614398 CANCEL > Reason: FreeSWITCH;cause=487;text="ORIGINATOR_CANCEL" > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 408 bytes from udp/[217.74.179.29]:5060 at 23:09:15.094670: > ------------------------------------------------------------------------ > ACK sip:gw+trunk_1000 at 78.138.90.58:6090;transport=udp;gw=trunk_1000 > SIP/2.0 > Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 > From: "+41793940965" ;tag=as00589402 > Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net > To: ;tag=S7UZQygFt62Nm > CSeq: 102 ACK > User-Agent: Sip EXpress router(0.9.7 (i386/linux)) > Content-Length: 0 > > ------------------------------------------------------------------------ > 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:79 > sofia/internal/+41793940965 at bluesip.net Standard REPORTING, cause: > ORIGINATOR_CANCEL > 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:662 > (sofia/internal/+41793940965 at bluesip.net) State REPORTING going to sleep > 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:387 > (sofia/internal/+41793940965 at bluesip.net) State Change CS_REPORTING -> > CS_DESTROY > 2012-05-24 23:09:15.120807 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:09:15.120807 [DEBUG] switch_core_session.c:1382 Session 8 > (sofia/internal/+41793940965 at bluesip.net) Locked, Waiting on external > entities > 2012-05-24 23:09:15.120807 [NOTICE] switch_core_session.c:1400 Session 8 > (sofia/internal/+41793940965 at bluesip.net) Ended > 2012-05-24 23:09:15.120807 [NOTICE] switch_core_session.c:1402 Close > Channel sofia/internal/+41793940965 at bluesip.net [CS_DESTROY] > 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:491 > (sofia/internal/+41793940965 at bluesip.net) Callstate Change HANGUP -> DOWN > 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:494 > (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_DESTROY > 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:504 > (sofia/internal/+41793940965 at bluesip.net) State DESTROY > 2012-05-24 23:09:15.120807 [DEBUG] mod_sofia.c:374 sofia/internal/ > +41793940965 at bluesip.net SOFIA DESTROY > 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:86 > sofia/internal/+41793940965 at bluesip.net Standard DESTROY > 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:504 > (sofia/internal/+41793940965 at bluesip.net) State DESTROY going to sleep > recv 378 bytes from udp/[78.138.90.58]:5060 at 23:09:15.135235: > ------------------------------------------------------------------------ > SIP/2.0 200 canceling > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;branch=z9hG4bKapDgFeegKyHXK > From: "+41793940965" ;tag=tgNrSS1jQFS8F > To: >;tag=a6a1c5f60faecf035a1ae5b6e96e979a-8c5b > Call-ID: 412c8312-2098-1230-8985-00163e6bb553 > CSeq: 28614398 CANCEL > Server: kamailio (3.1.5 (x86_64/linux)) > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 446 bytes from udp/[78.138.90.58]:5060 at 23:09:16.419308: > ------------------------------------------------------------------------ > SIP/2.0 487 Request Terminated > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;branch=z9hG4bKapDgFeegKyHXK > Record-Route: > Call-ID: 412c8312-2098-1230-8985-00163e6bb553 > From: "+41793940965" ;tag=tgNrSS1jQFS8F > To: > >;tag=qCSaB2YJOAsAr2dJ-RWNMhPoDdn1RFvs > CSeq: 28614398 INVITE > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, REGISTER > Content-Length: 0 > > ------------------------------------------------------------------------ > send 354 bytes to udp/[78.138.90.58]:5060 at 23:09:16.419543: > ------------------------------------------------------------------------ > ACK sip:1002 at 78.138.90.58:5060 SIP/2.0 > Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKapDgFeegKyHXK > Max-Forwards: 67 > From: "+41793940965" ;tag=tgNrSS1jQFS8F > To: > >;tag=qCSaB2YJOAsAr2dJ-RWNMhPoDdn1RFvs > Call-ID: 412c8312-2098-1230-8985-00163e6bb553 > CSeq: 28614398 ACK > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 564 bytes from udp/[41.34.123.243]:58881 at 23:13:13.612418: > ------------------------------------------------------------------------ > REGISTER sip:sip.pcfone.com:6090 SIP/2.0 > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport;branch=z9hG4bKPjHQIpv.ACHGbXD4IHrMr72LwZ8kYdlu-J > Max-Forwards: 70 > From: "Sherif 1003" > >;tag=wOP.FVT4Crk8O5E.bB7BWUhNO6QG91IG > To: "Sherif 1003" > > > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60926 REGISTER > User-Agent: Telephone 1.0.2 > Contact: "Sherif 1003" > Expires: 300 > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, > REFER, MESSAGE, OPTIONS > Content-Length: 0 > > ------------------------------------------------------------------------ > send 744 bytes to udp/[41.34.123.243]:58881 at 23:13:13.613152: > ------------------------------------------------------------------------ > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport=58881;branch=z9hG4bKPjHQIpv.ACHGbXD4IHrMr72LwZ8kYdlu-J;received=41.34.123.243 > From: "Sherif 1003" > >;tag=wOP.FVT4Crk8O5E.bB7BWUhNO6QG91IG > To: "Sherif 1003" > >;tag=USeHUmjpmrFUB > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60926 REGISTER > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > WWW-Authenticate: Digest realm="sip.pcfone.com", > nonce="09674764-a5f6-11e1-a83c-7b1f73a7ffcf", algorithm=MD5, qop="auth" > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 830 bytes from udp/[41.34.123.243]:58881 at 23:13:13.847746: > ------------------------------------------------------------------------ > REGISTER sip:sip.pcfone.com:6090 SIP/2.0 > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport;branch=z9hG4bKPjFr7rIAdAJPRQd2GmrMrVib3KT3Rup0q4 > Max-Forwards: 70 > From: "Sherif 1003" > >;tag=wOP.FVT4Crk8O5E.bB7BWUhNO6QG91IG > To: "Sherif 1003" > > > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60927 REGISTER > User-Agent: Telephone 1.0.2 > Contact: "Sherif 1003" > Expires: 300 > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, > REFER, MESSAGE, OPTIONS > Authorization: Digest username="1002", realm="sip.pcfone.com", > nonce="09674764-a5f6-11e1-a83c-7b1f73a7ffcf", uri="sip:sip.pcfone.com:6090", > response="3bc0bb82a803ded3641704c4eb78d58f", algorithm=MD5, > cnonce="sc3AJfhuev18Bs-L-6tORmI.RXlxnPOz", qop=auth, nc=00000001 > Content-Length: 0 > > ------------------------------------------------------------------------ > send 706 bytes to udp/[41.34.123.243]:58881 at 23:13:13.850197: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport=58881;branch=z9hG4bKPjFr7rIAdAJPRQd2GmrMrVib3KT3Rup0q4;received=41.34.123.243 > From: "Sherif 1003" > >;tag=wOP.FVT4Crk8O5E.bB7BWUhNO6QG91IG > To: "Sherif 1003" > >;tag=v279vF3SH15DQ > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60927 REGISTER > Contact: ;expires=300 > Date: Thu, 24 May 2012 23:13:13 GMT > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Content-Length: 0 > > ------------------------------------------------------------------------ > send 989 bytes to udp/[41.34.123.243]:58881 at 23:13:13.888717: > ------------------------------------------------------------------------ > NOTIFY sip:1002 at 192.168.178.26:58881;ob SIP/2.0 > Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKBZ68g9yKg77FF > Route: ;ob > Max-Forwards: 70 > From: > >;tag=XB12yamXeav0j > To: > > > Call-ID: e0efa252-2098-1230-8985-00163e6bb553 > CSeq: 28614532 NOTIFY > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > 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=noresource > Content-Type: application/simple-message-summary > Content-Length: 91 > > Messages-Waiting: yes > Message-Account: sip:1002 at 78.138.90.58 > Voice-Message: 1/0 (0/0) > > ------------------------------------------------------------------------ > recv 308 bytes from udp/[41.34.123.243]:58881 at 23:13:14.298850: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;received=78.138.90.58;branch=z9hG4bKBZ68g9yKg77FF > Call-ID: e0efa252-2098-1230-8985-00163e6bb553 > From: > >;tag=XB12yamXeav0j > To: > >;tag=z9hG4bKBZ68g9yKg77FF > CSeq: 28614532 NOTIFY > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 564 bytes from udp/[41.34.123.243]:58881 at 23:18:09.067232: > ------------------------------------------------------------------------ > REGISTER sip:sip.pcfone.com:6090 SIP/2.0 > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport;branch=z9hG4bKPjABTslmZIhiHYRtLqQn2erwmX19MVYHFC > Max-Forwards: 70 > From: "Sherif 1003" > >;tag=JBSXL5WZLWbLzpvRFoTSP5x8qQslI1ST > To: "Sherif 1003" > > > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60928 REGISTER > User-Agent: Telephone 1.0.2 > Contact: "Sherif 1003" > Expires: 300 > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, > REFER, MESSAGE, OPTIONS > Content-Length: 0 > > ------------------------------------------------------------------------ > send 744 bytes to udp/[41.34.123.243]:58881 at 23:18:09.068121: > ------------------------------------------------------------------------ > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport=58881;branch=z9hG4bKPjABTslmZIhiHYRtLqQn2erwmX19MVYHFC;received=41.34.123.243 > From: "Sherif 1003" > >;tag=JBSXL5WZLWbLzpvRFoTSP5x8qQslI1ST > To: "Sherif 1003" > >;tag=ymtU0540BKjKe > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60928 REGISTER > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > WWW-Authenticate: Digest realm="sip.pcfone.com", > nonce="b9821df4-a5f6-11e1-a83d-7b1f73a7ffcf", algorithm=MD5, qop="auth" > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 830 bytes from udp/[41.34.123.243]:58881 at 23:18:09.302944: > ------------------------------------------------------------------------ > REGISTER sip:sip.pcfone.com:6090 SIP/2.0 > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport;branch=z9hG4bKPjRgtztcy63vMT97Jp5pH9n311li5EFeY- > Max-Forwards: 70 > From: "Sherif 1003" > >;tag=JBSXL5WZLWbLzpvRFoTSP5x8qQslI1ST > To: "Sherif 1003" > > > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60929 REGISTER > User-Agent: Telephone 1.0.2 > Contact: "Sherif 1003" > Expires: 300 > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, > REFER, MESSAGE, OPTIONS > Authorization: Digest username="1002", realm="sip.pcfone.com", > nonce="b9821df4-a5f6-11e1-a83d-7b1f73a7ffcf", uri="sip:sip.pcfone.com:6090", > response="519c6301816b51f3bbb98d97a347e92b", algorithm=MD5, > cnonce="sc3AJfhuev18Bs-L-6tORmI.RXlxnPOz", qop=auth, nc=00000001 > Content-Length: 0 > > ------------------------------------------------------------------------ > send 706 bytes to udp/[41.34.123.243]:58881 at 23:18:09.305217: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport=58881;branch=z9hG4bKPjRgtztcy63vMT97Jp5pH9n311li5EFeY-;received=41.34.123.243 > From: "Sherif 1003" > >;tag=JBSXL5WZLWbLzpvRFoTSP5x8qQslI1ST > To: "Sherif 1003" > >;tag=ZXKm20N48U85S > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60929 REGISTER > Contact: ;expires=300 > Date: Thu, 24 May 2012 23:18:09 GMT > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Content-Length: 0 > > ------------------------------------------------------------------------ > send 989 bytes to udp/[41.34.123.243]:58881 at 23:18:09.379987: > ------------------------------------------------------------------------ > NOTIFY sip:1002 at 192.168.178.26:58881;ob SIP/2.0 > Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKc8Z1j4FQDgy2a > Route: ;ob > Max-Forwards: 70 > From: > >;tag=06cD4U6754yrN > To: > > > Call-ID: 91100602-2099-1230-8985-00163e6bb553 > CSeq: 28614680 NOTIFY > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > 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=noresource > Content-Type: application/simple-message-summary > Content-Length: 91 > > Messages-Waiting: yes > Message-Account: sip:1002 at 78.138.90.58 > Voice-Message: 1/0 (0/0) > > ------------------------------------------------------------------------ > recv 308 bytes from udp/[41.34.123.243]:58881 at 23:18:09.773629: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;received=78.138.90.58;branch=z9hG4bKc8Z1j4FQDgy2a > Call-ID: 91100602-2099-1230-8985-00163e6bb553 > From: > >;tag=06cD4U6754yrN > To: > >;tag=z9hG4bKc8Z1j4FQDgy2a > CSeq: 28614680 NOTIFY > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 564 bytes from udp/[41.34.123.243]:58881 at 23:23:04.522455: > ------------------------------------------------------------------------ > REGISTER sip:sip.pcfone.com:6090 SIP/2.0 > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport;branch=z9hG4bKPjaCuKyaBi6ZqcsHyp5HhH3WthJStom0SK > Max-Forwards: 70 > From: "Sherif 1003" > >;tag=qfUt1fQWuE-Ro3zWld6X0r6WidtD6B1q > To: "Sherif 1003" > > > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60930 REGISTER > User-Agent: Telephone 1.0.2 > Contact: "Sherif 1003" > Expires: 300 > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, > REFER, MESSAGE, OPTIONS > Content-Length: 0 > > ------------------------------------------------------------------------ > send 744 bytes to udp/[41.34.123.243]:58881 at 23:23:04.523158: > ------------------------------------------------------------------------ > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport=58881;branch=z9hG4bKPjaCuKyaBi6ZqcsHyp5HhH3WthJStom0SK;received=41.34.123.243 > From: "Sherif 1003" > >;tag=qfUt1fQWuE-Ro3zWld6X0r6WidtD6B1q > To: "Sherif 1003" > >;tag=1F655pQB3DNBH > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60930 REGISTER > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > WWW-Authenticate: Digest realm="sip.pcfone.com", > nonce="699cfe2a-a5f7-11e1-a83e-7b1f73a7ffcf", algorithm=MD5, qop="auth" > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 830 bytes from udp/[41.34.123.243]:58881 at 23:23:04.757798: > ------------------------------------------------------------------------ > REGISTER sip:sip.pcfone.com:6090 SIP/2.0 > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport;branch=z9hG4bKPjs8SAufeoyzCLrePaAbYhZuOqQF47XIc- > Max-Forwards: 70 > From: "Sherif 1003" > >;tag=qfUt1fQWuE-Ro3zWld6X0r6WidtD6B1q > To: "Sherif 1003" > > > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60931 REGISTER > User-Agent: Telephone 1.0.2 > Contact: "Sherif 1003" > Expires: 300 > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, > REFER, MESSAGE, OPTIONS > Authorization: Digest username="1002", realm="sip.pcfone.com", > nonce="699cfe2a-a5f7-11e1-a83e-7b1f73a7ffcf", uri="sip:sip.pcfone.com:6090", > response="23a886934aaf049f08432b6bb226bd37", algorithm=MD5, > cnonce="sc3AJfhuev18Bs-L-6tORmI.RXlxnPOz", qop=auth, nc=00000001 > Content-Length: 0 > > ------------------------------------------------------------------------ > send 706 bytes to udp/[41.34.123.243]:58881 at 23:23:04.759924: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport=58881;branch=z9hG4bKPjs8SAufeoyzCLrePaAbYhZuOqQF47XIc-;received=41.34.123.243 > From: "Sherif 1003" > >;tag=qfUt1fQWuE-Ro3zWld6X0r6WidtD6B1q > To: "Sherif 1003" > >;tag=2rZy7H8e0pByc > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60931 REGISTER > Contact: ;expires=300 > Date: Thu, 24 May 2012 23:23:04 GMT > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Content-Length: 0 > > ------------------------------------------------------------------------ > send 989 bytes to udp/[41.34.123.243]:58881 at 23:23:04.872728: > ------------------------------------------------------------------------ > NOTIFY sip:1002 at 192.168.178.26:58881;ob SIP/2.0 > Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKDHStmZ0taSmNp > Route: ;ob > Max-Forwards: 70 > From: > >;tag=31rQ9cSjXZ1gr > To: > > > Call-ID: 4130a1ce-209a-1230-8985-00163e6bb553 > CSeq: 28614828 NOTIFY > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > 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=noresource > Content-Type: application/simple-message-summary > Content-Length: 91 > > Messages-Waiting: yes > Message-Account: sip:1002 at 78.138.90.58 > Voice-Message: 1/0 (0/0) > > ------------------------------------------------------------------------ > recv 308 bytes from udp/[41.34.123.243]:58881 at 23:23:05.269063: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;received=78.138.90.58;branch=z9hG4bKDHStmZ0taSmNp > Call-ID: 4130a1ce-209a-1230-8985-00163e6bb553 > From: > >;tag=31rQ9cSjXZ1gr > To: > >;tag=z9hG4bKDHStmZ0taSmNp > CSeq: 28614828 NOTIFY > Content-Length: 0 > > ------------------------------------------------------------------------ > send 916 bytes to udp/[217.74.179.29]:5060 at 23:25:45.873423: > ------------------------------------------------------------------------ > REGISTER sip:bluesip.net;transport=udp SIP/2.0 > Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKetjKptHy71a8H > Max-Forwards: 70 > From: > ;transport=udp>;tag=6r0vBQZS650Fg > To: > ;transport=udp> > Call-ID: 2a3e4696-a5b9-11e1-a7ed-7b1f73a7ffcf > CSeq: 28601493 REGISTER > Contact: ;transport=udp;gw=trunk_1000> > Expires: 3600 > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Authorization: Digest username="bluesip/sherifomran", realm=" > bluesip.net", nonce="4fbeba4ebfb7414d5c701bd6f994402d551f3acf", > algorithm=MD5, uri="sip:bluesip.net;transport=udp", > response="c6cdcafe0418e519bc9ee0d8fa3d4d74" > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 722 bytes from udp/[217.74.179.29]:5060 at 23:25:45.904176: > ------------------------------------------------------------------------ > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;branch=z9hG4bKetjKptHy71a8H > From: > ;transport=udp>;tag=6r0vBQZS650Fg > To: > ;transport=udp>;tag=0354a2e1b960c9cc2279eca4e5f84e20.1b16 > Call-ID: 2a3e4696-a5b9-11e1-a7ed-7b1f73a7ffcf > CSeq: 28601493 REGISTER > WWW-Authenticate: Digest realm="bluesip.net", > nonce="4fbec4a4ba82d1f1feaa32ccec01b85be054fcf7", stale=true > Server: Sip EXpress router (0.9.7 (i386/linux)) > Content-Length: 0 > Warning: 392 217.74.179.29:5060 "Noisy feedback tells: pid=5455 > req_src_ip=78.138.90.58 req_src_port=6090 in_uri=sip:bluesip.net;transport=udp > out_uri=sip:bluesip.net;transport=udp via_cnt==1" > > ------------------------------------------------------------------------ > send 916 bytes to udp/[217.74.179.29]:5060 at 23:25:45.904416: > ------------------------------------------------------------------------ > REGISTER sip:bluesip.net;transport=udp SIP/2.0 > Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKF3BcrN214a1tD > Max-Forwards: 70 > From: > ;transport=udp>;tag=6r0vBQZS650Fg > To: > ;transport=udp> > Call-ID: 2a3e4696-a5b9-11e1-a7ed-7b1f73a7ffcf > CSeq: 28601494 REGISTER > Contact: ;transport=udp;gw=trunk_1000> > Expires: 3600 > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Authorization: Digest username="bluesip/sherifomran", realm=" > bluesip.net", nonce="4fbec4a4ba82d1f1feaa32ccec01b85be054fcf7", > algorithm=MD5, uri="sip:bluesip.net;transport=udp", > response="4c09dbe4b9accb52d4104b40dfe20040" > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 699 bytes from udp/[217.74.179.29]:5060 at 23:25:45.935046: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;branch=z9hG4bKF3BcrN214a1tD > From: > ;transport=udp>;tag=6r0vBQZS650Fg > To: > ;transport=udp>;tag=0354a2e1b960c9cc2279eca4e5f84e20.2017 > Call-ID: 2a3e4696-a5b9-11e1-a7ed-7b1f73a7ffcf > CSeq: 28601494 REGISTER > Contact: ;transport=udp;gw=trunk_1000>;q=0.5;expires=3600 > Server: Sip EXpress router (0.9.7 (i386/linux)) > Content-Length: 0 > Warning: 392 217.74.179.29:5060 "Noisy feedback tells: pid=5462 > req_src_ip=78.138.90.58 req_src_port=6090 in_uri=sip:bluesip.net;transport=udp > out_uri=sip:bluesip.net;transport=udp via_cnt==1" > > ------------------------------------------------------------------------ > recv 564 bytes from udp/[41.34.123.243]:58881 at 23:27:59.977057: > ------------------------------------------------------------------------ > REGISTER sip:sip.pcfone.com:6090 SIP/2.0 > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport;branch=z9hG4bKPjEOP.a4NoXqoFqC7a.k2tyeX-mYlUyH4H > Max-Forwards: 70 > From: "Sherif 1003" > >;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K > To: "Sherif 1003" > > > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60932 REGISTER > User-Agent: Telephone 1.0.2 > Contact: "Sherif 1003" > Expires: 300 > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, > REFER, MESSAGE, OPTIONS > Content-Length: 0 > > ------------------------------------------------------------------------ > send 744 bytes to udp/[41.34.123.243]:58881 at 23:27:59.977960: > ------------------------------------------------------------------------ > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport=58881;branch=z9hG4bKPjEOP.a4NoXqoFqC7a.k2tyeX-mYlUyH4H;received=41.34.123.243 > From: "Sherif 1003" > >;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K > To: "Sherif 1003" > >;tag=4ajgB89Nt8Q3K > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60932 REGISTER > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > WWW-Authenticate: Digest realm="sip.pcfone.com", > nonce="19b7cc40-a5f8-11e1-a83f-7b1f73a7ffcf", algorithm=MD5, qop="auth" > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 564 bytes from udp/[41.34.123.243]:58881 at 23:28:00.468487: > ------------------------------------------------------------------------ > REGISTER sip:sip.pcfone.com:6090 SIP/2.0 > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport;branch=z9hG4bKPjEOP.a4NoXqoFqC7a.k2tyeX-mYlUyH4H > Max-Forwards: 70 > From: "Sherif 1003" > >;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K > To: "Sherif 1003" > > > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60932 REGISTER > User-Agent: Telephone 1.0.2 > Contact: "Sherif 1003" > Expires: 300 > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, > REFER, MESSAGE, OPTIONS > Content-Length: 0 > > ------------------------------------------------------------------------ > send 744 bytes to udp/[41.34.123.243]:58881 at 23:28:00.468679: > ------------------------------------------------------------------------ > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport=58881;branch=z9hG4bKPjEOP.a4NoXqoFqC7a.k2tyeX-mYlUyH4H;received=41.34.123.243 > From: "Sherif 1003" > >;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K > To: "Sherif 1003" > >;tag=4ajgB89Nt8Q3K > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60932 REGISTER > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > WWW-Authenticate: Digest realm="sip.pcfone.com", > nonce="19b7cc40-a5f8-11e1-a83f-7b1f73a7ffcf", algorithm=MD5, qop="auth" > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 830 bytes from udp/[41.34.123.243]:58881 at 23:28:00.717446: > ------------------------------------------------------------------------ > REGISTER sip:sip.pcfone.com:6090 SIP/2.0 > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport;branch=z9hG4bKPj0iHd3-XUjnvcBs9DiJUCWzrmYTmWxFLt > Max-Forwards: 70 > From: "Sherif 1003" > >;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K > To: "Sherif 1003" > > > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60933 REGISTER > User-Agent: Telephone 1.0.2 > Contact: "Sherif 1003" > Expires: 300 > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, > REFER, MESSAGE, OPTIONS > Authorization: Digest username="1002", realm="sip.pcfone.com", > nonce="19b7cc40-a5f8-11e1-a83f-7b1f73a7ffcf", uri="sip:sip.pcfone.com:6090", > response="e21c903e6d98b5adcdd2eefa75217157", algorithm=MD5, > cnonce="sc3AJfhuev18Bs-L-6tORmI.RXlxnPOz", qop=auth, nc=00000001 > Content-Length: 0 > > ------------------------------------------------------------------------ > send 706 bytes to udp/[41.34.123.243]:58881 at 23:28:00.719947: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport=58881;branch=z9hG4bKPj0iHd3-XUjnvcBs9DiJUCWzrmYTmWxFLt;received=41.34.123.243 > From: "Sherif 1003" > >;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K > To: "Sherif 1003" > >;tag=5KB9c3tSQHepF > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60933 REGISTER > Contact: ;expires=300 > Date: Thu, 24 May 2012 23:28:00 GMT > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Content-Length: 0 > > ------------------------------------------------------------------------ > send 989 bytes to udp/[41.34.123.243]:58881 at 23:28:00.868368: > ------------------------------------------------------------------------ > NOTIFY sip:1002 at 192.168.178.26:58881;ob SIP/2.0 > Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKgc54SgK51KQDS > Route: ;ob > Max-Forwards: 70 > From: > >;tag=6v41eyBXmt48a > To: > > > Call-ID: f19dfbce-209a-1230-8985-00163e6bb553 > CSeq: 28614976 NOTIFY > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > 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=noresource > Content-Type: application/simple-message-summary > Content-Length: 91 > > Messages-Waiting: yes > Message-Account: sip:1002 at 78.138.90.58 > Voice-Message: 1/0 (0/0) > > ------------------------------------------------------------------------ > recv 308 bytes from udp/[41.34.123.243]:58881 at 23:28:01.128279: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;received=78.138.90.58;branch=z9hG4bKgc54SgK51KQDS > Call-ID: f19dfbce-209a-1230-8985-00163e6bb553 > From: > >;tag=6v41eyBXmt48a > To: > >;tag=z9hG4bKgc54SgK51KQDS > CSeq: 28614976 NOTIFY > Content-Length: 0 > > ------------------------------------------------------------------------ > send 913 bytes to udp/[217.74.179.29]:5060 at 23:28:11.885213: > ------------------------------------------------------------------------ > REGISTER sip:bluesip.net;transport=udp SIP/2.0 > Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKHNyXUB48yvD0m > Max-Forwards: 70 > From: > ;transport=udp>;tag=5F739Uep9vaXm > To: > ;transport=udp> > Call-ID: 2a3e49c0-a5b9-11e1-a7ee-7b1f73a7ffcf > CSeq: 28601493 REGISTER > Contact: ;transport=udp;gw=trunk_1002> > Expires: 3600 > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Authorization: Digest username="bluesip/salahomran", realm="bluesip.net", > nonce="4fbebeca79a5cff7c417463814cb9054d313bcac", algorithm=MD5, uri="sip: > bluesip.net;transport=udp", response="bb1babb90f4ea1dc8fbb9aa45d6038c7" > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 720 bytes from udp/[217.74.179.29]:5060 at 23:28:11.915764: > ------------------------------------------------------------------------ > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;branch=z9hG4bKHNyXUB48yvD0m > From: > ;transport=udp>;tag=5F739Uep9vaXm > To: > ;transport=udp>;tag=0354a2e1b960c9cc2279eca4e5f84e20.3454 > Call-ID: 2a3e49c0-a5b9-11e1-a7ee-7b1f73a7ffcf > CSeq: 28601493 REGISTER > WWW-Authenticate: Digest realm="bluesip.net", > nonce="4fbec5365af3a174f1cc0b8cda770c727d25bf71", stale=true > Server: Sip EXpress router (0.9.7 (i386/linux)) > Content-Length: 0 > Warning: 392 217.74.179.29:5060 "Noisy feedback tells: pid=5462 > req_src_ip=78.138.90.58 req_src_port=6090 in_uri=sip:bluesip.net;transport=udp > out_uri=sip:bluesip.net;transport=udp via_cnt==1" > > ------------------------------------------------------------------------ > send 913 bytes to udp/[217.74.179.29]:5060 at 23:28:11.915914: > ------------------------------------------------------------------------ > REGISTER sip:bluesip.net;transport=udp SIP/2.0 > Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKjyQpX6mcv53jg > Max-Forwards: 70 > From: > ;transport=udp>;tag=5F739Uep9vaXm > To: > ;transport=udp> > Call-ID: 2a3e49c0-a5b9-11e1-a7ee-7b1f73a7ffcf > CSeq: 28601494 REGISTER > Contact: ;transport=udp;gw=trunk_1002> > Expires: 3600 > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Authorization: Digest username="bluesip/salahomran", realm="bluesip.net", > nonce="4fbec5365af3a174f1cc0b8cda770c727d25bf71", algorithm=MD5, uri="sip: > bluesip.net;transport=udp", response="e260f6dc9b01ce8bea8c5ffb36bd4ddd" > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 697 bytes from udp/[217.74.179.29]:5060 at 23:28:11.946325: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;branch=z9hG4bKjyQpX6mcv53jg > From: > ;transport=udp>;tag=5F739Uep9vaXm > To: > ;transport=udp>;tag=0354a2e1b960c9cc2279eca4e5f84e20.4b5c > Call-ID: 2a3e49c0-a5b9-11e1-a7ee-7b1f73a7ffcf > CSeq: 28601494 REGISTER > Contact: ;transport=udp;gw=trunk_1002>;q=0.5;expires=3600 > Server: Sip EXpress router (0.9.7 (i386/linux)) > Content-Length: 0 > Warning: 392 217.74.179.29:5060 "Noisy feedback tells: pid=5455 > req_src_ip=78.138.90.58 req_src_port=6090 in_uri=sip:bluesip.net;transport=udp > out_uri=sip:bluesip.net;transport=udp via_cnt==1" > > ------------------------------------------------------------------------ > recv 564 bytes from udp/[41.34.123.243]:58881 at 23:32:55.930149: > ------------------------------------------------------------------------ > REGISTER sip:sip.pcfone.com:6090 SIP/2.0 > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport;branch=z9hG4bKPjuk8DZGiaXfN7BmtslNHVjtSg5UXsbHXc > Max-Forwards: 70 > From: "Sherif 1003" > >;tag=Pz.t-0tGXj3saM.ehBiB1EAMmmitze1D > To: "Sherif 1003" > > > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60934 REGISTER > User-Agent: Telephone 1.0.2 > Contact: "Sherif 1003" > Expires: 300 > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, > REFER, MESSAGE, OPTIONS > Content-Length: 0 > > ------------------------------------------------------------------------ > send 744 bytes to udp/[41.34.123.243]:58881 at 23:32:55.931324: > ------------------------------------------------------------------------ > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport=58881;branch=z9hG4bKPjuk8DZGiaXfN7BmtslNHVjtSg5UXsbHXc;received=41.34.123.243 > From: "Sherif 1003" > >;tag=Pz.t-0tGXj3saM.ehBiB1EAMmmitze1D > To: "Sherif 1003" > >;tag=75XtgSv0H3tUp > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60934 REGISTER > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > WWW-Authenticate: Digest realm="sip.pcfone.com", > nonce="ca1eabee-a5f8-11e1-a840-7b1f73a7ffcf", algorithm=MD5, qop="auth" > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 830 bytes from udp/[41.34.123.243]:58881 at 23:32:56.165624: > ------------------------------------------------------------------------ > REGISTER sip:sip.pcfone.com:6090 SIP/2.0 > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport;branch=z9hG4bKPjKEQ6xXmyzZxA6vCM88vHNYAOPAjUqFsD > Max-Forwards: 70 > From: "Sherif 1003" > >;tag=Pz.t-0tGXj3saM.ehBiB1EAMmmitze1D > To: "Sherif 1003" > > > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60935 REGISTER > User-Agent: Telephone 1.0.2 > Contact: "Sherif 1003" > Expires: 300 > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, > REFER, MESSAGE, OPTIONS > Authorization: Digest username="1002", realm="sip.pcfone.com", > nonce="ca1eabee-a5f8-11e1-a840-7b1f73a7ffcf", uri="sip:sip.pcfone.com:6090", > response="ff6d12f4ec7abd4b37187222b3af2358", algorithm=MD5, > cnonce="sc3AJfhuev18Bs-L-6tORmI.RXlxnPOz", qop=auth, nc=00000001 > Content-Length: 0 > > ------------------------------------------------------------------------ > send 706 bytes to udp/[41.34.123.243]:58881 at 23:32:56.168033: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport=58881;branch=z9hG4bKPjKEQ6xXmyzZxA6vCM88vHNYAOPAjUqFsD;received=41.34.123.243 > From: "Sherif 1003" > >;tag=Pz.t-0tGXj3saM.ehBiB1EAMmmitze1D > To: "Sherif 1003" > >;tag=8eQKjmD4ecHej > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60935 REGISTER > Contact: ;expires=300 > Date: Thu, 24 May 2012 23:32:56 GMT > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Content-Length: 0 > > ------------------------------------------------------------------------ > send 989 bytes to udp/[41.34.123.243]:58881 at 23:32:56.360238: > ------------------------------------------------------------------------ > NOTIFY sip:1002 at 192.168.178.26:58881;ob SIP/2.0 > Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKK7gFZ15FSet5B > Route: ;ob > Max-Forwards: 70 > From: > >;tag=9QgcmFy7BN70D > To: > > > Call-ID: a1be7708-209b-1230-8985-00163e6bb553 > CSeq: 28615124 NOTIFY > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > 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=noresource > Content-Type: application/simple-message-summary > Content-Length: 91 > > Messages-Waiting: yes > Message-Account: sip:1002 at 78.138.90.58 > Voice-Message: 1/0 (0/0) > > ------------------------------------------------------------------------ > recv 308 bytes from udp/[41.34.123.243]:58881 at 23:32:56.623554: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;received=78.138.90.58;branch=z9hG4bKK7gFZ15FSet5B > Call-ID: a1be7708-209b-1230-8985-00163e6bb553 > From: > >;tag=9QgcmFy7BN70D > To: > >;tag=z9hG4bKK7gFZ15FSet5B > CSeq: 28615124 NOTIFY > Content-Length: 0 > > ------------------------------------------------------------------------ > freeswitch at internal> > > > --- On *Thu, 5/24/12, Kristian Kielhofner > >* wrote: > > > From: Kristian Kielhofner > > > Subject: Re: [Freeswitch-users] Freeswitch (SBC + Media) does not know > that gateway phone is picked up > To: "FreeSWITCH Users Help" > > > Date: Thursday, May 24, 2012, 7:51 PM > > > Siptrace and logs please. > > On Thu, May 24, 2012 at 11:53 AM, Sherif Omran > > > wrote: > > > > Hi all, > > > > My topology is as follows: > > > > Kamailio -> FS (SBS+Media server) > > > > I came across an issue with my system as follows. I have a Hardphone > registered. When I do local call inside kamailio, it gets to FS and returns > back well and FS understands when I lift the handset. However, I added a > gateway (german landline server), when I call my self from another phone, > the call gets to FS and then transmits to Kamailio, it rings my extension > but when I lift the handset FS does not notice it and keeps ringing. > > > > Any body has an Idea? Here is my gateway trunk. > > > > > > > > > > > > > > > > > > > > > > thanks in advance > > Sherif Omran > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > Kristian Kielhofner > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -----Inline Attachment Follows----- > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120525/9678e5c4/attachment-0001.html From albert_nguyen16 at hotmail.com Fri May 25 12:54:31 2012 From: albert_nguyen16 at hotmail.com (Albert Nguyen) Date: Fri, 25 May 2012 08:54:31 +0000 Subject: [Freeswitch-users] FS as a dumb SBC (was: Re: Help on Explaination of the configurations parrametters) In-Reply-To: <4FBE889E.6000406@cupis.co.uk> References: <4FBE889E.6000406@cupis.co.uk> Message-ID: Hi Paul, Thanks very much for your suggestion. I'll give it a try to see how it goes Regards, Al > Date: Thu, 24 May 2012 20:14:38 +0100 > From: paul at cupis.co.uk > To: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] FS as a dumb SBC (was: Re: Help on Explaination of the configurations parrametters) > > On 23/05/12 09:26, Albert Nguyen wrote: > > I am new to FS and trying to setup FS as a "dumb" SBC as per > > example 2 in the wiki website so that I can convert inband DTMF to > > RFC 8233 and vice versa. This is the windows version of FS. Is it > > possible that I can get some assistant from you guys to get it up and > > running? Please advise how we can proceed. > > One way of doing this would be to setup two SIP profiles, one facing the > device sending you RFC2833 and once facing the device which will only > accept in-band. > > On the SIP profile facing the in-band device, disable RFC2833 support by > setting the varaible "rfc2833-pt" to something <95, and then when you > route a call from RFC2833 to in-band, add something like: > > data="execute_on_media=start_dtmf_generate"/> > > to your dialplan. > > Similarly you will want some config for calls going in-band->rfc2833 to > do the conversion, perhaps something like: > > > > Note that I've not specifically tested the above. > > If you are going down the above route, you might also want to read: > > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf_generate > > http://wiki.freeswitch.org/wiki/DTMF > > > Regards, > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120525/75fa6b82/attachment.html From sherifomran2000 at yahoo.com Fri May 25 12:52:26 2012 From: sherifomran2000 at yahoo.com (Sherif Omran) Date: Fri, 25 May 2012 01:52:26 -0700 (PDT) Subject: [Freeswitch-users] Freeswitch (SBC + Media) does not know that gateway phone is picked up Message-ID: <1337935946.46011.YahooMailClassic@web110806.mail.gq1.yahoo.com> Hello guys, here is the trace (see the attachment) kind regards Sherif --- On Fri, 5/25/12, SamyGo wrote: From: SamyGo Subject: Re: [Freeswitch-users] Freeswitch (SBC + Media) does not know that gateway phone is picked up To: "FreeSWITCH Users Help" Date: Friday, May 25, 2012, 10:46 AM Hi again,If you want kamailio register to the provider then use UAC module. Kamailio will use the username/password and register with the provider. Regards,Sammy On Fri, May 25, 2012 at 12:20 PM, Sherif Omran wrote: Dear Sammy, Thank you for your question ... Yes, the GW is registered through FS because I did not know how to register it to kamailio. But it seems better to register it to kamailio. One more information, calls from 1001 to 1002 go to kamailio then to FS then return back to kamailio smoothly. Thus I would suggest that I change registering the gateway from FS to kamailio. but How to? --- On Fri, 5/25/12, SamyGo wrote: From: SamyGo Subject: Re: [Freeswitch-users] Freeswitch (SBC + Media) does not know that gateway phone is picked up To: "FreeSWITCH Users Help" Date: Friday, May 25, 2012, 8:08 AM Hi,These are SIP traces on FreeSWITCH console, whereas you are saying and it seems that kamailio is not detecting the answering (200 OK)of the call from extension 1002. Please, can you take a sip trace..!! I see you've both kamailio and FS on same server! Please take a pcap from the linux console using the following command. #tcpdump -i any -s 0 -w new-call-trace.pcap -vvv Please be quick on turning the sip trace on and off as quickly as possible to avoid extra packets. Once done open the file in wireshark ;?apply filter "sip || rtp" and then save the resulting capture in separate file. Send us the new file to analyse. One more silly question probably, I see REGISTERs coming to your FS as well and the calls to gateways are made from FS too !!, umm...just thinking what are you using kamailio for!!? ThanksBR,Sammy On Fri, May 25, 2012 at 4:40 AM, Sherif Omran wrote: Hi all, here is the siptrace: To figure it out 1- gateway called bluesip.net. It send invide using caller number at bluesip.net 2- This call should go to extension kb-1002. kb means go from freeswitch port 6090 to kamailio port 5060 3- It should go to call extension 1002 in Kamailio 4- Extension 1002 rings but when I reply, it does not notice I replied ./fs_cli ??????????? _____ ____???? ____ _???? ___????????????? ?????????? |? ___/ ___|?? / ___| |?? |_ _|???????????? ?????????? | |_? \___ \? | |?? | |??? | |??????????? ?????????? |? _|? ___) | | |___| |___ | |????????????? ?????????? |_|?? |____/?? \____|_____|___|??????????? ******************************************************* * Anthony Minessale II, Ken Rice,???????????????????? * * Michael Jerris, Travis Cross??????????????????????? * * FreeSWITCH (http://www.freeswitch.org)????????????? * * Paypal Donations Appreciated: paypal at freeswitch.org * * Brought to you by ClueCon http://www.cluecon.com/?? * ******************************************************* Type /help to see a list of commands +OK log level? [7] freeswitch at internal> tracelevel -ERR tracelevel Command not found! freeswitch at internal> sofia global siptrace on +OK Global siptrace on recv 1104 bytes from udp/[217.74.179.29]:5060 at 23:08:45.822962: ?? ------------------------------------------------------------------------ ?? INVITE sip:gw+trunk_1000 at 78.138.90.58:6090;transport=udp;gw=trunk_1000 SIP/2.0 ?? Record-Route: ?? Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 ?? Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 ?? From: "+41793940965" ;tag=as00589402 ?? To: ?? Contact: ?? Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net ?? CSeq: 102 INVITE ?? User-Agent: blueSIP PSTN GW ?? Max-Forwards: 69 ?? Date: Thu, 24 May 2012 23:08:44 GMT ?? Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY ?? Supported: replaces ?? Content-Type: application/sdp ?? Content-Length: 367 ?? P-hint: USRLOC ?? ?? v=0 ?? o=root 20076 20076 IN IP4 217.74.179.28 ?? s=session ?? c=IN IP4 217.74.179.28 ?? t=0 0 ?? m=audio 25626 RTP/AVP 8 0 18 111 3 101 ?? a=rtpmap:8 PCMA/8000 ?? a=rtpmap:0 PCMU/8000 ?? a=rtpmap:18 G729/8000 ?? a=fmtp:18 annexb=no ?? a=rtpmap:111 G726-32/8000 ?? a=rtpmap:3 GSM/8000 ?? a=rtpmap:101 telephone-event/8000 ?? a=fmtp:101 0-16 ?? a=silenceSupp:off - - - - ?? a=ptime:20 ?? a=sendrecv ?? ------------------------------------------------------------------------ send 494 bytes to udp/[217.74.179.29]:5060 at 23:08:45.844486: ?? ------------------------------------------------------------------------ ?? SIP/2.0 100 Trying ?? Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 ?? Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 ?? Record-Route: ?? From: "+41793940965" ;tag=as00589402 ?? To: ?? Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net ?? CSeq: 102 INVITE ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ 2012-05-24 23:08:45.840749 [NOTICE] switch_channel.c:926 New Channel sofia/internal/+41793940965 at bluesip.net [69cd22e6-a5f5-11e1-a833-7b1f73a7ffcf] 2012-05-24 23:08:45.840749 [DEBUG] sofia.c:5532 Channel sofia/internal/+41793940965 at bluesip.net entering state [received][100] 2012-05-24 23:08:45.840749 [DEBUG] sofia.c:5543 Remote SDP: v=0 o=root 20076 20076 IN IP4 217.74.179.28 s=session c=IN IP4 217.74.179.28 t=0 0 m=audio 25626 RTP/AVP 8 0 18 111 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:111 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMA:8:8000:20:64000]/[iLBC:97:8000:30:13330] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMU:0:8000:20:64000]/[iLBC:97:8000:30:13330] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G729:18:8000:20:8000]/[iLBC:97:8000:30:13330] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G726-32:111:8000:20:0]/[iLBC:97:8000:30:13330] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [GSM:3:8000:20:13200]/[iLBC:97:8000:30:13330] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [telephone-event:101:8000:20:0]/[iLBC:97:8000:30:13330] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf send/recv payload to 101 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMA:8:8000:20:64000]/[G7221:115:32000:20:48000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMU:0:8000:20:64000]/[G7221:115:32000:20:48000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G729:18:8000:20:8000]/[G7221:115:32000:20:48000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G726-32:111:8000:20:0]/[G7221:115:32000:20:48000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [GSM:3:8000:20:13200]/[G7221:115:32000:20:48000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [telephone-event:101:8000:20:0]/[G7221:115:32000:20:48000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf send/recv payload to 101 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMA:8:8000:20:64000]/[G7221:107:16000:20:32000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMU:0:8000:20:64000]/[G7221:107:16000:20:32000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G729:18:8000:20:8000]/[G7221:107:16000:20:32000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G726-32:111:8000:20:0]/[G7221:107:16000:20:32000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [GSM:3:8000:20:13200]/[G7221:107:16000:20:32000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [telephone-event:101:8000:20:0]/[G7221:107:16000:20:32000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf send/recv payload to 101 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMA:8:8000:20:64000]/[G722:9:8000:20:64000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G729:18:8000:20:8000]/[G722:9:8000:20:64000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G726-32:111:8000:20:0]/[G722:9:8000:20:64000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [GSM:3:8000:20:13200]/[G722:9:8000:20:64000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [telephone-event:101:8000:20:0]/[G722:9:8000:20:64000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf send/recv payload to 101 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMA:8:8000:20:64000]/[PCMU:0:8000:20:64000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:2996 Set Codec sofia/internal/+41793940965 at bluesip.net PCMU/8000 20 ms 160 samples 64000 bits 2012-05-24 23:08:45.840749 [DEBUG] switch_core_codec.c:111 sofia/internal/+41793940965 at bluesip.net Original read codec set to PCMU:0 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf send/recv payload to 101 2012-05-24 23:08:45.840749 [DEBUG] sofia.c:5757 (sofia/internal/+41793940965 at bluesip.net) State Change CS_NEW -> CS_INIT 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_INIT 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/+41793940965 at bluesip.net) State INIT 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:85 sofia/internal/+41793940965 at bluesip.net SOFIA INIT 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:125 (sofia/internal/+41793940965 at bluesip.net) State Change CS_INIT -> CS_ROUTING 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/+41793940965 at bluesip.net) State INIT going to sleep 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_ROUTING 2012-05-24 23:08:45.840749 [DEBUG] switch_channel.c:1886 (sofia/internal/+41793940965 at bluesip.net) Callstate Change DOWN -> RINGING 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/+41793940965 at bluesip.net) State ROUTING 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:148 sofia/internal/+41793940965 at bluesip.net SOFIA ROUTING 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:104 sofia/internal/+41793940965 at bluesip.net Standard ROUTING 2012-05-24 23:08:45.840749 [INFO] mod_dialplan_xml.c:485 Processing +41793940965 <+41793940965>->kb-1002 in context public Dialplan: sofia/internal/+41793940965 at bluesip.net parsing [public->from_kamailio] continue=false Dialplan: sofia/internal/+41793940965 at bluesip.net Regex (PASS) [from_kamailio] destination_number(kb-1002) =~ /^(.+)$/ break=on-false Dialplan: sofia/internal/+41793940965 at bluesip.net Action transfer(kb-1002 XML default) 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:154 (sofia/internal/+41793940965 at bluesip.net) State Change CS_ROUTING -> CS_EXECUTE 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/+41793940965 at bluesip.net) State ROUTING going to sleep 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_EXECUTE 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/+41793940965 at bluesip.net) State EXECUTE 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:241 sofia/internal/+41793940965 at bluesip.net SOFIA EXECUTE 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:192 sofia/internal/+41793940965 at bluesip.net Standard EXECUTE EXECUTE sofia/internal/+41793940965 at bluesip.net transfer(kb-1002 XML default) 2012-05-24 23:08:45.840749 [DEBUG] switch_ivr.c:1711 (sofia/internal/+41793940965 at bluesip.net) State Change CS_EXECUTE -> CS_ROUTING 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:731 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:08:45.840749 [NOTICE] switch_ivr.c:1717 Transfer sofia/internal/+41793940965 at bluesip.net to XML[kb-1002 at default] 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/+41793940965 at bluesip.net) State EXECUTE going to sleep 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_ROUTING 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/+41793940965 at bluesip.net) State ROUTING 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:148 sofia/internal/+41793940965 at bluesip.net SOFIA ROUTING 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:104 sofia/internal/+41793940965 at bluesip.net Standard ROUTING 2012-05-24 23:08:45.840749 [INFO] mod_dialplan_xml.c:485 Processing +41793940965 <+41793940965>->kb-1002 in context default Dialplan: sofia/internal/+41793940965 at bluesip.net parsing [default->vbox] continue=false Dialplan: sofia/internal/+41793940965 at bluesip.net Regex (FAIL) [vbox] destination_number(kb-1002) =~ /^vb-([0-9][0-9][0-9])$/ break=on-false Dialplan: sofia/internal/+41793940965 at bluesip.net parsing [default->vmenu] continue=false Dialplan: sofia/internal/+41793940965 at bluesip.net Regex (FAIL) [vmenu] destination_number(kb-1002) =~ /^vm-([0-9][0-9][0-9][0-9])$/ break=on-false Dialplan: sofia/internal/+41793940965 at bluesip.net parsing [default->kbridge] continue=false Dialplan: sofia/internal/+41793940965 at bluesip.net Regex (PASS) [kbridge] destination_number(kb-1002) =~ /^kb-(.+)$/ break=on-false Dialplan: sofia/internal/+41793940965 at bluesip.net Action set(proxy_media=true) Dialplan: sofia/internal/+41793940965 at bluesip.net Action set(call_timeout=50) Dialplan: sofia/internal/+41793940965 at bluesip.net Action set(continue_on_fail=true) Dialplan: sofia/internal/+41793940965 at bluesip.net Action set(hangup_after_bridge=true) Dialplan: sofia/internal/+41793940965 at bluesip.net Action set(sip_invite_domain=78.138.90.58) Dialplan: sofia/internal/+41793940965 at bluesip.net Action export(sip_contact_user=ufs) Dialplan: sofia/internal/+41793940965 at bluesip.net Action bridge(sofia/78.138.90.58/1002 at 78.138.90.58:5060) Dialplan: sofia/internal/+41793940965 at bluesip.net Action answer() Dialplan: sofia/internal/+41793940965 at bluesip.net Action voicemail(default ${domain_name} 1002) 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:154 (sofia/internal/+41793940965 at bluesip.net) State Change CS_ROUTING -> CS_EXECUTE 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/+41793940965 at bluesip.net) State ROUTING going to sleep 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_EXECUTE 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/+41793940965 at bluesip.net) State EXECUTE 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:241 sofia/internal/+41793940965 at bluesip.net SOFIA EXECUTE 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:192 sofia/internal/+41793940965 at bluesip.net Standard EXECUTE EXECUTE sofia/internal/+41793940965 at bluesip.net set(proxy_media=true) 2012-05-24 23:08:45.840749 [DEBUG] mod_dptools.c:1281 sofia/internal/+41793940965 at bluesip.net SET [proxy_media]=[true] EXECUTE sofia/internal/+41793940965 at bluesip.net set(call_timeout=50) 2012-05-24 23:08:45.840749 [DEBUG] mod_dptools.c:1281 sofia/internal/+41793940965 at bluesip.net SET [call_timeout]=[50] EXECUTE sofia/internal/+41793940965 at bluesip.net set(continue_on_fail=true) 2012-05-24 23:08:45.840749 [DEBUG] mod_dptools.c:1281 sofia/internal/+41793940965 at bluesip.net SET [continue_on_fail]=[true] EXECUTE sofia/internal/+41793940965 at bluesip.net set(hangup_after_bridge=true) 2012-05-24 23:08:45.840749 [DEBUG] mod_dptools.c:1281 sofia/internal/+41793940965 at bluesip.net SET [hangup_after_bridge]=[true] EXECUTE sofia/internal/+41793940965 at bluesip.net set(sip_invite_domain=78.138.90.58) 2012-05-24 23:08:45.840749 [DEBUG] mod_dptools.c:1281 sofia/internal/+41793940965 at bluesip.net SET [sip_invite_domain]=[78.138.90.58] EXECUTE sofia/internal/+41793940965 at bluesip.net export(sip_contact_user=ufs) 2012-05-24 23:08:45.840749 [DEBUG] switch_channel.c:1093 EXPORT (export_vars) [sip_contact_user]=[ufs] EXECUTE sofia/internal/+41793940965 at bluesip.net bridge(sofia/78.138.90.58/1002 at 78.138.90.58:5060) 2012-05-24 23:08:45.840749 [DEBUG] switch_channel.c:1047 sofia/internal/+41793940965 at bluesip.net EXPORTING[export_vars] [sip_contact_user]=[ufs] to event 2012-05-24 23:08:45.840749 [DEBUG] switch_ivr_originate.c:1884 Parsing global variables 2012-05-24 23:08:45.840749 [NOTICE] switch_channel.c:926 New Channel sofia/internal/1002 at 78.138.90.58:5060 [69ce1f0c-a5f5-11e1-a838-7b1f73a7ffcf] 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:4691 (sofia/internal/1002 at 78.138.90.58:5060) State Change CS_NEW -> CS_INIT 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1002 at 78.138.90.58:5060) Running State Change CS_INIT 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/1002 at 78.138.90.58:5060) State INIT 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:85 sofia/internal/1002 at 78.138.90.58:5060 SOFIA INIT 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:1871 sofia/internal/1002 at 78.138.90.58:5060 Patched SDP --- v=0 o=root 20076 20076 IN IP4 217.74.179.28 s=session c=IN IP4 217.74.179.28 t=0 0 m=audio 25626 RTP/AVP 8 0 18 111 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:111 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 +++ v=0 o=FreeSWITCH 1629510322 1629510323 IN IP4 78.138.90.58 s=FreeSWITCH c=IN IP4 78.138.90.58 t=0 0 m=audio 31178 RTP/AVP 8 0 18 111 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:111 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 send 1340 bytes to udp/[78.138.90.58]:5060 at 23:08:45.852280: ?? ------------------------------------------------------------------------ ?? INVITE sip:1002 at 78.138.90.58:5060 SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKapDgFeegKyHXK ?? Max-Forwards: 67 ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? CSeq: 28614398 INVITE ?? Contact: ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? 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: 372 ?? P-hint: USRLOC ?? X-FS-Support: update_display,send_info ?? Remote-Party-ID: "+41793940965" ;party=calling;screen=yes;privacy=off ?? ?? v=0 ?? o=FreeSWITCH 1629510322 1629510323 IN IP4 78.138.90.58 ?? s=FreeSWITCH ?? c=IN IP4 78.138.90.58 ?? t=0 0 ?? m=audio 31178 RTP/AVP 8 0 18 111 3 101 ?? a=rtpmap:8 PCMA/8000 ?? a=rtpmap:0 PCMU/8000 ?? a=rtpmap:18 G729/8000 ?? a=fmtp:18 annexb=no ?? a=rtpmap:111 G726-32/8000 ?? a=rtpmap:3 GSM/8000 ?? a=rtpmap:101 telephone-event/8000 ?? a=fmtp:101 0-16 ?? a=silenceSupp:off - - - - ?? a=ptime:20 ?? ------------------------------------------------------------------------ 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:125 (sofia/internal/1002 at 78.138.90.58:5060) State Change CS_INIT -> CS_ROUTING 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/1002 at 78.138.90.58:5060) State INIT going to sleep 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1002 at 78.138.90.58:5060) Running State Change CS_ROUTING 2012-05-24 23:08:45.840749 [DEBUG] switch_channel.c:1886 (sofia/internal/1002 at 78.138.90.58:5060) Callstate Change DOWN -> RINGING 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/1002 at 78.138.90.58:5060) State ROUTING 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:148 sofia/internal/1002 at 78.138.90.58:5060 SOFIA ROUTING 2012-05-24 23:08:45.840749 [DEBUG] switch_ivr_originate.c:66 (sofia/internal/1002 at 78.138.90.58:5060) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/1002 at 78.138.90.58:5060) State ROUTING going to sleep 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1002 at 78.138.90.58:5060) Running State Change CS_CONSUME_MEDIA 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:429 (sofia/internal/1002 at 78.138.90.58:5060) State CONSUME_MEDIA 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:429 (sofia/internal/1002 at 78.138.90.58:5060) State CONSUME_MEDIA going to sleep 2012-05-24 23:08:45.840749 [DEBUG] sofia.c:5532 Channel sofia/internal/1002 at 78.138.90.58:5060 entering state [calling][0] recv 365 bytes from udp/[78.138.90.58]:5060 at 23:08:45.964309: ?? ------------------------------------------------------------------------ ?? SIP/2.0 100 trying -- your call is important to us ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKapDgFeegKyHXK ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? CSeq: 28614398 INVITE ?? Server: kamailio (3.1.5 (x86_64/linux)) ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 496 bytes from udp/[78.138.90.58]:5060 at 23:08:47.414766: ?? ------------------------------------------------------------------------ ?? SIP/2.0 180 Ringing ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKapDgFeegKyHXK ?? Record-Route: ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ;tag=qCSaB2YJOAsAr2dJ-RWNMhPoDdn1RFvs ?? CSeq: 28614398 INVITE ?? Contact: "Mama" ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, REGISTER ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ 2012-05-24 23:08:47.400741 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:47.400741 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:47.400741 [DEBUG] sofia.c:5532 Channel sofia/internal/1002 at 78.138.90.58:5060 entering state [proceeding][180] 2012-05-24 23:08:47.400741 [NOTICE] sofia.c:5624 Ring-Ready sofia/internal/1002 at 78.138.90.58:5060! send 995 bytes to udp/[217.74.179.29]:5060 at 23:08:47.416538: ?? ------------------------------------------------------------------------ ?? SIP/2.0 180 Ringing ?? Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 ?? Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 ?? Record-Route: ?? From: "+41793940965" ;tag=as00589402 ?? To: ;tag=S7UZQygFt62Nm ?? Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net ?? CSeq: 102 INVITE ?? Contact: ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Accept: application/sdp ?? 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-Length: 0 ?? Remote-Party-ID: "Outbound Call" ;party=calling;privacy=off;screen=no ?? ?? ------------------------------------------------------------------------ 2012-05-24 23:08:47.400741 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:08:47.400741 [NOTICE] mod_sofia.c:2514 Ring-Ready sofia/internal/+41793940965 at bluesip.net! 2012-05-24 23:08:47.400741 [DEBUG] sofia.c:5532 Channel sofia/internal/+41793940965 at bluesip.net entering state [early][180] 2012-05-24 23:08:47.400741 [DEBUG] switch_core_session.c:731 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:08:47.400741 [NOTICE] switch_ivr_originate.c:483 Ring Ready sofia/internal/+41793940965 at bluesip.net! recv 496 bytes from udp/[78.138.90.58]:5060 at 23:08:47.804967: ?? ------------------------------------------------------------------------ ?? SIP/2.0 180 Ringing ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKapDgFeegKyHXK ?? Record-Route: ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ;tag=549D2DD03BBA7C67 ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? CSeq: 28614398 INVITE ?? Contact: ?? User-Agent: AVM FRITZ!Box Fon WLAN 7170 29.04.80 (Jan 27 2010) ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ 2012-05-24 23:08:47.800746 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:47.800746 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:47.800746 [DEBUG] sofia.c:5532 Channel sofia/internal/1002 at 78.138.90.58:5060 entering state [proceeding][180] recv 496 bytes from udp/[78.138.90.58]:5060 at 23:08:48.182098: ?? ------------------------------------------------------------------------ ?? SIP/2.0 180 Ringing ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKapDgFeegKyHXK ?? Record-Route: ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ;tag=qCSaB2YJOAsAr2dJ-RWNMhPoDdn1RFvs ?? CSeq: 28614398 INVITE ?? Contact: "Mama" ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, REGISTER ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ 2012-05-24 23:08:48.180741 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:48.180741 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:48.180741 [DEBUG] sofia.c:5532 Channel sofia/internal/1002 at 78.138.90.58:5060 entering state [proceeding][180] recv 496 bytes from udp/[78.138.90.58]:5060 at 23:08:48.417403: ?? ------------------------------------------------------------------------ ?? SIP/2.0 180 Ringing ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKapDgFeegKyHXK ?? Record-Route: ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ;tag=549D2DD03BBA7C67 ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? CSeq: 28614398 INVITE ?? Contact: ?? User-Agent: AVM FRITZ!Box Fon WLAN 7170 29.04.80 (Jan 27 2010) ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ 2012-05-24 23:08:48.400744 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:48.400744 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:48.400744 [DEBUG] sofia.c:5532 Channel sofia/internal/1002 at 78.138.90.58:5060 entering state [proceeding][180] recv 496 bytes from udp/[78.138.90.58]:5060 at 23:08:49.292599: ?? ------------------------------------------------------------------------ ?? SIP/2.0 180 Ringing ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKapDgFeegKyHXK ?? Record-Route: ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ;tag=549D2DD03BBA7C67 ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? CSeq: 28614398 INVITE ?? Contact: ?? User-Agent: AVM FRITZ!Box Fon WLAN 7170 29.04.80 (Jan 27 2010) ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ 2012-05-24 23:08:49.280742 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:49.280742 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:49.280742 [DEBUG] sofia.c:5532 Channel sofia/internal/1002 at 78.138.90.58:5060 entering state [proceeding][180] recv 478 bytes from udp/[217.74.179.29]:5060 at 23:09:15.066012: ?? ------------------------------------------------------------------------ ?? CANCEL sip:gw+trunk_1000 at 78.138.90.58:6090;transport=udp;gw=trunk_1000 SIP/2.0 ?? Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 ?? Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 ?? From: "+41793940965" ;tag=as00589402 ?? To: ?? Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net ?? CSeq: 102 CANCEL ?? User-Agent: blueSIP PSTN GW ?? Max-Forwards: 69 ?? Content-Length: 0 ?? P-hint: USRLOC ?? ?? ------------------------------------------------------------------------ send 369 bytes to udp/[217.74.179.29]:5060 at 23:09:15.066176: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 ?? Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 ?? From: "+41793940965" ;tag=as00589402 ?? To: ;tag=S7UZQygFt62Nm ?? Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net ?? CSeq: 102 CANCEL ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ send 766 bytes to udp/[217.74.179.29]:5060 at 23:09:15.066289: ?? ------------------------------------------------------------------------ ?? SIP/2.0 487 Request Terminated ?? Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 ?? Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 ?? From: "+41793940965" ;tag=as00589402 ?? To: ;tag=S7UZQygFt62Nm ?? Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net ?? CSeq: 102 INVITE ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? 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-Length: 0 ?? ?? ------------------------------------------------------------------------ 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:09:15.060742 [DEBUG] sofia.c:5532 Channel sofia/internal/+41793940965 at bluesip.net entering state [terminated][487] 2012-05-24 23:09:15.060742 [DEBUG] switch_channel.c:2848 (sofia/internal/+41793940965 at bluesip.net) Callstate Change RINGING -> HANGUP 2012-05-24 23:09:15.060742 [NOTICE] sofia.c:6301 Hangup sofia/internal/+41793940965 at bluesip.net [CS_EXECUTE] [ORIGINATOR_CANCEL] 2012-05-24 23:09:15.060742 [DEBUG] switch_channel.c:2871 Send signal sofia/internal/+41793940965 at bluesip.net [KILL] 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:09:15.060742 [DEBUG] switch_channel.c:2848 (sofia/internal/1002 at 78.138.90.58:5060) Callstate Change RINGING -> HANGUP 2012-05-24 23:09:15.060742 [NOTICE] switch_ivr_originate.c:3182 Hangup sofia/internal/1002 at 78.138.90.58:5060 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL] 2012-05-24 23:09:15.060742 [DEBUG] switch_channel.c:2871 Send signal sofia/internal/1002 at 78.138.90.58:5060 [KILL] 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1002 at 78.138.90.58:5060) Running State Change CS_HANGUP 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/1002 at 78.138.90.58:5060) State HANGUP 2012-05-24 23:09:15.060742 [DEBUG] mod_sofia.c:469 Channel sofia/internal/1002 at 78.138.90.58:5060 hanging up, cause: ORIGINATOR_CANCEL 2012-05-24 23:09:15.060742 [DEBUG] mod_sofia.c:523 Sending CANCEL to sofia/internal/1002 at 78.138.90.58:5060 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:47 sofia/internal/1002 at 78.138.90.58:5060 Standard HANGUP, cause: ORIGINATOR_CANCEL 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/1002 at 78.138.90.58:5060) State HANGUP going to sleep 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:393 (sofia/internal/1002 at 78.138.90.58:5060) State Change CS_HANGUP -> CS_REPORTING 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1002 at 78.138.90.58:5060) Running State Change CS_REPORTING 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/1002 at 78.138.90.58:5060) State REPORTING 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:79 sofia/internal/1002 at 78.138.90.58:5060 Standard REPORTING, cause: ORIGINATOR_CANCEL 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/1002 at 78.138.90.58:5060) State REPORTING going to sleep 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:387 (sofia/internal/1002 at 78.138.90.58:5060) State Change CS_REPORTING -> CS_DESTROY 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1382 Session 9 (sofia/internal/1002 at 78.138.90.58:5060) Locked, Waiting on external entities 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:09:15.060742 [DEBUG] switch_ivr_originate.c:3358 Originate Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL] 2012-05-24 23:09:15.060742 [NOTICE] switch_core_session.c:1400 Session 9 (sofia/internal/1002 at 78.138.90.58:5060) Ended 2012-05-24 23:09:15.060742 [NOTICE] switch_core_session.c:1402 Close Channel sofia/internal/1002 at 78.138.90.58:5060 [CS_DESTROY] 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:491 (sofia/internal/1002 at 78.138.90.58:5060) Callstate Change HANGUP -> DOWN 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:494 (sofia/internal/1002 at 78.138.90.58:5060) Running State Change CS_DESTROY 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/1002 at 78.138.90.58:5060) State DESTROY 2012-05-24 23:09:15.060742 [DEBUG] mod_sofia.c:374 sofia/internal/1002 at 78.138.90.58:5060 SOFIA DESTROY 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:86 sofia/internal/1002 at 78.138.90.58:5060 Standard DESTROY 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/1002 at 78.138.90.58:5060) State DESTROY going to sleep 2012-05-24 23:09:15.060742 [INFO] mod_dptools.c:2922 Originate Failed.? Cause: ORIGINATOR_CANCEL 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:2287 sofia/internal/+41793940965 at bluesip.net skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/+41793940965 at bluesip.net) State EXECUTE going to sleep 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_HANGUP 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/+41793940965 at bluesip.net) State HANGUP 2012-05-24 23:09:15.060742 [DEBUG] mod_sofia.c:469 Channel sofia/internal/+41793940965 at bluesip.net hanging up, cause: ORIGINATOR_CANCEL 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:47 sofia/internal/+41793940965 at bluesip.net Standard HANGUP, cause: ORIGINATOR_CANCEL 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/+41793940965 at bluesip.net) State HANGUP going to sleep 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:393 (sofia/internal/+41793940965 at bluesip.net) State Change CS_HANGUP -> CS_REPORTING 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_REPORTING 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/+41793940965 at bluesip.net) State REPORTING send 383 bytes to udp/[78.138.90.58]:5060 at 23:09:15.087695: ?? ------------------------------------------------------------------------ ?? CANCEL sip:1002 at 78.138.90.58:5060 SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKapDgFeegKyHXK ?? Max-Forwards: 67 ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? CSeq: 28614398 CANCEL ?? Reason: FreeSWITCH;cause=487;text="ORIGINATOR_CANCEL" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 408 bytes from udp/[217.74.179.29]:5060 at 23:09:15.094670: ?? ------------------------------------------------------------------------ ?? ACK sip:gw+trunk_1000 at 78.138.90.58:6090;transport=udp;gw=trunk_1000 SIP/2.0 ?? Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 ?? From: "+41793940965" ;tag=as00589402 ?? Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net ?? To: ;tag=S7UZQygFt62Nm ?? CSeq: 102 ACK ?? User-Agent: Sip EXpress router(0.9.7 (i386/linux)) ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:79 sofia/internal/+41793940965 at bluesip.net Standard REPORTING, cause: ORIGINATOR_CANCEL 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/+41793940965 at bluesip.net) State REPORTING going to sleep 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:387 (sofia/internal/+41793940965 at bluesip.net) State Change CS_REPORTING -> CS_DESTROY 2012-05-24 23:09:15.120807 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:09:15.120807 [DEBUG] switch_core_session.c:1382 Session 8 (sofia/internal/+41793940965 at bluesip.net) Locked, Waiting on external entities 2012-05-24 23:09:15.120807 [NOTICE] switch_core_session.c:1400 Session 8 (sofia/internal/+41793940965 at bluesip.net) Ended 2012-05-24 23:09:15.120807 [NOTICE] switch_core_session.c:1402 Close Channel sofia/internal/+41793940965 at bluesip.net [CS_DESTROY] 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:491 (sofia/internal/+41793940965 at bluesip.net) Callstate Change HANGUP -> DOWN 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:494 (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_DESTROY 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/+41793940965 at bluesip.net) State DESTROY 2012-05-24 23:09:15.120807 [DEBUG] mod_sofia.c:374 sofia/internal/+41793940965 at bluesip.net SOFIA DESTROY 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:86 sofia/internal/+41793940965 at bluesip.net Standard DESTROY 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/+41793940965 at bluesip.net) State DESTROY going to sleep recv 378 bytes from udp/[78.138.90.58]:5060 at 23:09:15.135235: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 canceling ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKapDgFeegKyHXK ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ;tag=a6a1c5f60faecf035a1ae5b6e96e979a-8c5b ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? CSeq: 28614398 CANCEL ?? Server: kamailio (3.1.5 (x86_64/linux)) ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 446 bytes from udp/[78.138.90.58]:5060 at 23:09:16.419308: ?? ------------------------------------------------------------------------ ?? SIP/2.0 487 Request Terminated ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKapDgFeegKyHXK ?? Record-Route: ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ;tag=qCSaB2YJOAsAr2dJ-RWNMhPoDdn1RFvs ?? CSeq: 28614398 INVITE ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, REGISTER ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 354 bytes to udp/[78.138.90.58]:5060 at 23:09:16.419543: ?? ------------------------------------------------------------------------ ?? ACK sip:1002 at 78.138.90.58:5060 SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKapDgFeegKyHXK ?? Max-Forwards: 67 ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ;tag=qCSaB2YJOAsAr2dJ-RWNMhPoDdn1RFvs ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? CSeq: 28614398 ACK ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 564 bytes from udp/[41.34.123.243]:58881 at 23:13:13.612418: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjHQIpv.ACHGbXD4IHrMr72LwZ8kYdlu-J ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=wOP.FVT4Crk8O5E.bB7BWUhNO6QG91IG ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60926 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 744 bytes to udp/[41.34.123.243]:58881 at 23:13:13.613152: ?? ------------------------------------------------------------------------ ?? SIP/2.0 401 Unauthorized ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjHQIpv.ACHGbXD4IHrMr72LwZ8kYdlu-J;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=wOP.FVT4Crk8O5E.bB7BWUhNO6QG91IG ?? To: "Sherif 1003" ;tag=USeHUmjpmrFUB ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60926 REGISTER ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? WWW-Authenticate: Digest realm="sip.pcfone.com", nonce="09674764-a5f6-11e1-a83c-7b1f73a7ffcf", algorithm=MD5, qop="auth" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 830 bytes from udp/[41.34.123.243]:58881 at 23:13:13.847746: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjFr7rIAdAJPRQd2GmrMrVib3KT3Rup0q4 ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=wOP.FVT4Crk8O5E.bB7BWUhNO6QG91IG ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60927 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Authorization: Digest username="1002", realm="sip.pcfone.com", nonce="09674764-a5f6-11e1-a83c-7b1f73a7ffcf", uri="sip:sip.pcfone.com:6090", response="3bc0bb82a803ded3641704c4eb78d58f", algorithm=MD5, cnonce="sc3AJfhuev18Bs-L-6tORmI.RXlxnPOz", qop=auth, nc=00000001 ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 706 bytes to udp/[41.34.123.243]:58881 at 23:13:13.850197: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjFr7rIAdAJPRQd2GmrMrVib3KT3Rup0q4;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=wOP.FVT4Crk8O5E.bB7BWUhNO6QG91IG ?? To: "Sherif 1003" ;tag=v279vF3SH15DQ ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60927 REGISTER ?? Contact: ;expires=300 ?? Date: Thu, 24 May 2012 23:13:13 GMT ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ send 989 bytes to udp/[41.34.123.243]:58881 at 23:13:13.888717: ?? ------------------------------------------------------------------------ ?? NOTIFY sip:1002 at 192.168.178.26:58881;ob SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKBZ68g9yKg77FF ?? Route: ;ob ?? Max-Forwards: 70 ?? From: ;tag=XB12yamXeav0j ?? To: ?? Call-ID: e0efa252-2098-1230-8985-00163e6bb553 ?? CSeq: 28614532 NOTIFY ?? Contact: ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? 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=noresource ?? Content-Type: application/simple-message-summary ?? Content-Length: 91 ?? ?? Messages-Waiting: yes ?? Message-Account: sip:1002 at 78.138.90.58 ?? Voice-Message: 1/0 (0/0) ?? ?? ------------------------------------------------------------------------ recv 308 bytes from udp/[41.34.123.243]:58881 at 23:13:14.298850: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;received=78.138.90.58;branch=z9hG4bKBZ68g9yKg77FF ?? Call-ID: e0efa252-2098-1230-8985-00163e6bb553 ?? From: ;tag=XB12yamXeav0j ?? To: ;tag=z9hG4bKBZ68g9yKg77FF ?? CSeq: 28614532 NOTIFY ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ recv 564 bytes from udp/[41.34.123.243]:58881 at 23:18:09.067232: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjABTslmZIhiHYRtLqQn2erwmX19MVYHFC ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=JBSXL5WZLWbLzpvRFoTSP5x8qQslI1ST ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60928 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 744 bytes to udp/[41.34.123.243]:58881 at 23:18:09.068121: ?? ------------------------------------------------------------------------ ?? SIP/2.0 401 Unauthorized ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjABTslmZIhiHYRtLqQn2erwmX19MVYHFC;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=JBSXL5WZLWbLzpvRFoTSP5x8qQslI1ST ?? To: "Sherif 1003" ;tag=ymtU0540BKjKe ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60928 REGISTER ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? WWW-Authenticate: Digest realm="sip.pcfone.com", nonce="b9821df4-a5f6-11e1-a83d-7b1f73a7ffcf", algorithm=MD5, qop="auth" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 830 bytes from udp/[41.34.123.243]:58881 at 23:18:09.302944: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjRgtztcy63vMT97Jp5pH9n311li5EFeY- ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=JBSXL5WZLWbLzpvRFoTSP5x8qQslI1ST ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60929 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Authorization: Digest username="1002", realm="sip.pcfone.com", nonce="b9821df4-a5f6-11e1-a83d-7b1f73a7ffcf", uri="sip:sip.pcfone.com:6090", response="519c6301816b51f3bbb98d97a347e92b", algorithm=MD5, cnonce="sc3AJfhuev18Bs-L-6tORmI.RXlxnPOz", qop=auth, nc=00000001 ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 706 bytes to udp/[41.34.123.243]:58881 at 23:18:09.305217: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjRgtztcy63vMT97Jp5pH9n311li5EFeY-;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=JBSXL5WZLWbLzpvRFoTSP5x8qQslI1ST ?? To: "Sherif 1003" ;tag=ZXKm20N48U85S ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60929 REGISTER ?? Contact: ;expires=300 ?? Date: Thu, 24 May 2012 23:18:09 GMT ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ send 989 bytes to udp/[41.34.123.243]:58881 at 23:18:09.379987: ?? ------------------------------------------------------------------------ ?? NOTIFY sip:1002 at 192.168.178.26:58881;ob SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKc8Z1j4FQDgy2a ?? Route: ;ob ?? Max-Forwards: 70 ?? From: ;tag=06cD4U6754yrN ?? To: ?? Call-ID: 91100602-2099-1230-8985-00163e6bb553 ?? CSeq: 28614680 NOTIFY ?? Contact: ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? 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=noresource ?? Content-Type: application/simple-message-summary ?? Content-Length: 91 ?? ?? Messages-Waiting: yes ?? Message-Account: sip:1002 at 78.138.90.58 ?? Voice-Message: 1/0 (0/0) ?? ?? ------------------------------------------------------------------------ recv 308 bytes from udp/[41.34.123.243]:58881 at 23:18:09.773629: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;received=78.138.90.58;branch=z9hG4bKc8Z1j4FQDgy2a ?? Call-ID: 91100602-2099-1230-8985-00163e6bb553 ?? From: ;tag=06cD4U6754yrN ?? To: ;tag=z9hG4bKc8Z1j4FQDgy2a ?? CSeq: 28614680 NOTIFY ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ recv 564 bytes from udp/[41.34.123.243]:58881 at 23:23:04.522455: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjaCuKyaBi6ZqcsHyp5HhH3WthJStom0SK ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=qfUt1fQWuE-Ro3zWld6X0r6WidtD6B1q ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60930 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 744 bytes to udp/[41.34.123.243]:58881 at 23:23:04.523158: ?? ------------------------------------------------------------------------ ?? SIP/2.0 401 Unauthorized ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjaCuKyaBi6ZqcsHyp5HhH3WthJStom0SK;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=qfUt1fQWuE-Ro3zWld6X0r6WidtD6B1q ?? To: "Sherif 1003" ;tag=1F655pQB3DNBH ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60930 REGISTER ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? WWW-Authenticate: Digest realm="sip.pcfone.com", nonce="699cfe2a-a5f7-11e1-a83e-7b1f73a7ffcf", algorithm=MD5, qop="auth" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 830 bytes from udp/[41.34.123.243]:58881 at 23:23:04.757798: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjs8SAufeoyzCLrePaAbYhZuOqQF47XIc- ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=qfUt1fQWuE-Ro3zWld6X0r6WidtD6B1q ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60931 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Authorization: Digest username="1002", realm="sip.pcfone.com", nonce="699cfe2a-a5f7-11e1-a83e-7b1f73a7ffcf", uri="sip:sip.pcfone.com:6090", response="23a886934aaf049f08432b6bb226bd37", algorithm=MD5, cnonce="sc3AJfhuev18Bs-L-6tORmI.RXlxnPOz", qop=auth, nc=00000001 ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 706 bytes to udp/[41.34.123.243]:58881 at 23:23:04.759924: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjs8SAufeoyzCLrePaAbYhZuOqQF47XIc-;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=qfUt1fQWuE-Ro3zWld6X0r6WidtD6B1q ?? To: "Sherif 1003" ;tag=2rZy7H8e0pByc ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60931 REGISTER ?? Contact: ;expires=300 ?? Date: Thu, 24 May 2012 23:23:04 GMT ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ send 989 bytes to udp/[41.34.123.243]:58881 at 23:23:04.872728: ?? ------------------------------------------------------------------------ ?? NOTIFY sip:1002 at 192.168.178.26:58881;ob SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKDHStmZ0taSmNp ?? Route: ;ob ?? Max-Forwards: 70 ?? From: ;tag=31rQ9cSjXZ1gr ?? To: ?? Call-ID: 4130a1ce-209a-1230-8985-00163e6bb553 ?? CSeq: 28614828 NOTIFY ?? Contact: ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? 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=noresource ?? Content-Type: application/simple-message-summary ?? Content-Length: 91 ?? ?? Messages-Waiting: yes ?? Message-Account: sip:1002 at 78.138.90.58 ?? Voice-Message: 1/0 (0/0) ?? ?? ------------------------------------------------------------------------ recv 308 bytes from udp/[41.34.123.243]:58881 at 23:23:05.269063: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;received=78.138.90.58;branch=z9hG4bKDHStmZ0taSmNp ?? Call-ID: 4130a1ce-209a-1230-8985-00163e6bb553 ?? From: ;tag=31rQ9cSjXZ1gr ?? To: ;tag=z9hG4bKDHStmZ0taSmNp ?? CSeq: 28614828 NOTIFY ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 916 bytes to udp/[217.74.179.29]:5060 at 23:25:45.873423: ?? ------------------------------------------------------------------------ ?? REGISTER sip:bluesip.net;transport=udp SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKetjKptHy71a8H ?? Max-Forwards: 70 ?? From: ;tag=6r0vBQZS650Fg ?? To: ?? Call-ID: 2a3e4696-a5b9-11e1-a7ed-7b1f73a7ffcf ?? CSeq: 28601493 REGISTER ?? Contact: ?? Expires: 3600 ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? Authorization: Digest username="bluesip/sherifomran", realm="bluesip.net", nonce="4fbeba4ebfb7414d5c701bd6f994402d551f3acf", algorithm=MD5, uri="sip:bluesip.net;transport=udp", response="c6cdcafe0418e519bc9ee0d8fa3d4d74" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 722 bytes from udp/[217.74.179.29]:5060 at 23:25:45.904176: ?? ------------------------------------------------------------------------ ?? SIP/2.0 401 Unauthorized ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKetjKptHy71a8H ?? From: ;tag=6r0vBQZS650Fg ?? To: ;tag=0354a2e1b960c9cc2279eca4e5f84e20.1b16 ?? Call-ID: 2a3e4696-a5b9-11e1-a7ed-7b1f73a7ffcf ?? CSeq: 28601493 REGISTER ?? WWW-Authenticate: Digest realm="bluesip.net", nonce="4fbec4a4ba82d1f1feaa32ccec01b85be054fcf7", stale=true ?? Server: Sip EXpress router (0.9.7 (i386/linux)) ?? Content-Length: 0 ?? Warning: 392 217.74.179.29:5060 "Noisy feedback tells:? pid=5455 req_src_ip=78.138.90.58 req_src_port=6090 in_uri=sip:bluesip.net;transport=udp out_uri=sip:bluesip.net;transport=udp via_cnt==1" ?? ?? ------------------------------------------------------------------------ send 916 bytes to udp/[217.74.179.29]:5060 at 23:25:45.904416: ?? ------------------------------------------------------------------------ ?? REGISTER sip:bluesip.net;transport=udp SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKF3BcrN214a1tD ?? Max-Forwards: 70 ?? From: ;tag=6r0vBQZS650Fg ?? To: ?? Call-ID: 2a3e4696-a5b9-11e1-a7ed-7b1f73a7ffcf ?? CSeq: 28601494 REGISTER ?? Contact: ?? Expires: 3600 ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? Authorization: Digest username="bluesip/sherifomran", realm="bluesip.net", nonce="4fbec4a4ba82d1f1feaa32ccec01b85be054fcf7", algorithm=MD5, uri="sip:bluesip.net;transport=udp", response="4c09dbe4b9accb52d4104b40dfe20040" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 699 bytes from udp/[217.74.179.29]:5060 at 23:25:45.935046: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKF3BcrN214a1tD ?? From: ;tag=6r0vBQZS650Fg ?? To: ;tag=0354a2e1b960c9cc2279eca4e5f84e20.2017 ?? Call-ID: 2a3e4696-a5b9-11e1-a7ed-7b1f73a7ffcf ?? CSeq: 28601494 REGISTER ?? Contact: ;q=0.5;expires=3600 ?? Server: Sip EXpress router (0.9.7 (i386/linux)) ?? Content-Length: 0 ?? Warning: 392 217.74.179.29:5060 "Noisy feedback tells:? pid=5462 req_src_ip=78.138.90.58 req_src_port=6090 in_uri=sip:bluesip.net;transport=udp out_uri=sip:bluesip.net;transport=udp via_cnt==1" ?? ?? ------------------------------------------------------------------------ recv 564 bytes from udp/[41.34.123.243]:58881 at 23:27:59.977057: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjEOP.a4NoXqoFqC7a.k2tyeX-mYlUyH4H ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60932 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 744 bytes to udp/[41.34.123.243]:58881 at 23:27:59.977960: ?? ------------------------------------------------------------------------ ?? SIP/2.0 401 Unauthorized ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjEOP.a4NoXqoFqC7a.k2tyeX-mYlUyH4H;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K ?? To: "Sherif 1003" ;tag=4ajgB89Nt8Q3K ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60932 REGISTER ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? WWW-Authenticate: Digest realm="sip.pcfone.com", nonce="19b7cc40-a5f8-11e1-a83f-7b1f73a7ffcf", algorithm=MD5, qop="auth" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 564 bytes from udp/[41.34.123.243]:58881 at 23:28:00.468487: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjEOP.a4NoXqoFqC7a.k2tyeX-mYlUyH4H ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60932 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 744 bytes to udp/[41.34.123.243]:58881 at 23:28:00.468679: ?? ------------------------------------------------------------------------ ?? SIP/2.0 401 Unauthorized ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjEOP.a4NoXqoFqC7a.k2tyeX-mYlUyH4H;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K ?? To: "Sherif 1003" ;tag=4ajgB89Nt8Q3K ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60932 REGISTER ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? WWW-Authenticate: Digest realm="sip.pcfone.com", nonce="19b7cc40-a5f8-11e1-a83f-7b1f73a7ffcf", algorithm=MD5, qop="auth" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 830 bytes from udp/[41.34.123.243]:58881 at 23:28:00.717446: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPj0iHd3-XUjnvcBs9DiJUCWzrmYTmWxFLt ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60933 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Authorization: Digest username="1002", realm="sip.pcfone.com", nonce="19b7cc40-a5f8-11e1-a83f-7b1f73a7ffcf", uri="sip:sip.pcfone.com:6090", response="e21c903e6d98b5adcdd2eefa75217157", algorithm=MD5, cnonce="sc3AJfhuev18Bs-L-6tORmI.RXlxnPOz", qop=auth, nc=00000001 ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 706 bytes to udp/[41.34.123.243]:58881 at 23:28:00.719947: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPj0iHd3-XUjnvcBs9DiJUCWzrmYTmWxFLt;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K ?? To: "Sherif 1003" ;tag=5KB9c3tSQHepF ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60933 REGISTER ?? Contact: ;expires=300 ?? Date: Thu, 24 May 2012 23:28:00 GMT ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ send 989 bytes to udp/[41.34.123.243]:58881 at 23:28:00.868368: ?? ------------------------------------------------------------------------ ?? NOTIFY sip:1002 at 192.168.178.26:58881;ob SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKgc54SgK51KQDS ?? Route: ;ob ?? Max-Forwards: 70 ?? From: ;tag=6v41eyBXmt48a ?? To: ?? Call-ID: f19dfbce-209a-1230-8985-00163e6bb553 ?? CSeq: 28614976 NOTIFY ?? Contact: ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? 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=noresource ?? Content-Type: application/simple-message-summary ?? Content-Length: 91 ?? ?? Messages-Waiting: yes ?? Message-Account: sip:1002 at 78.138.90.58 ?? Voice-Message: 1/0 (0/0) ?? ?? ------------------------------------------------------------------------ recv 308 bytes from udp/[41.34.123.243]:58881 at 23:28:01.128279: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;received=78.138.90.58;branch=z9hG4bKgc54SgK51KQDS ?? Call-ID: f19dfbce-209a-1230-8985-00163e6bb553 ?? From: ;tag=6v41eyBXmt48a ?? To: ;tag=z9hG4bKgc54SgK51KQDS ?? CSeq: 28614976 NOTIFY ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 913 bytes to udp/[217.74.179.29]:5060 at 23:28:11.885213: ?? ------------------------------------------------------------------------ ?? REGISTER sip:bluesip.net;transport=udp SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKHNyXUB48yvD0m ?? Max-Forwards: 70 ?? From: ;tag=5F739Uep9vaXm ?? To: ?? Call-ID: 2a3e49c0-a5b9-11e1-a7ee-7b1f73a7ffcf ?? CSeq: 28601493 REGISTER ?? Contact: ?? Expires: 3600 ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? Authorization: Digest username="bluesip/salahomran", realm="bluesip.net", nonce="4fbebeca79a5cff7c417463814cb9054d313bcac", algorithm=MD5, uri="sip:bluesip.net;transport=udp", response="bb1babb90f4ea1dc8fbb9aa45d6038c7" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 720 bytes from udp/[217.74.179.29]:5060 at 23:28:11.915764: ?? ------------------------------------------------------------------------ ?? SIP/2.0 401 Unauthorized ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKHNyXUB48yvD0m ?? From: ;tag=5F739Uep9vaXm ?? To: ;tag=0354a2e1b960c9cc2279eca4e5f84e20.3454 ?? Call-ID: 2a3e49c0-a5b9-11e1-a7ee-7b1f73a7ffcf ?? CSeq: 28601493 REGISTER ?? WWW-Authenticate: Digest realm="bluesip.net", nonce="4fbec5365af3a174f1cc0b8cda770c727d25bf71", stale=true ?? Server: Sip EXpress router (0.9.7 (i386/linux)) ?? Content-Length: 0 ?? Warning: 392 217.74.179.29:5060 "Noisy feedback tells:? pid=5462 req_src_ip=78.138.90.58 req_src_port=6090 in_uri=sip:bluesip.net;transport=udp out_uri=sip:bluesip.net;transport=udp via_cnt==1" ?? ?? ------------------------------------------------------------------------ send 913 bytes to udp/[217.74.179.29]:5060 at 23:28:11.915914: ?? ------------------------------------------------------------------------ ?? REGISTER sip:bluesip.net;transport=udp SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKjyQpX6mcv53jg ?? Max-Forwards: 70 ?? From: ;tag=5F739Uep9vaXm ?? To: ?? Call-ID: 2a3e49c0-a5b9-11e1-a7ee-7b1f73a7ffcf ?? CSeq: 28601494 REGISTER ?? Contact: ?? Expires: 3600 ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? Authorization: Digest username="bluesip/salahomran", realm="bluesip.net", nonce="4fbec5365af3a174f1cc0b8cda770c727d25bf71", algorithm=MD5, uri="sip:bluesip.net;transport=udp", response="e260f6dc9b01ce8bea8c5ffb36bd4ddd" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 697 bytes from udp/[217.74.179.29]:5060 at 23:28:11.946325: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKjyQpX6mcv53jg ?? From: ;tag=5F739Uep9vaXm ?? To: ;tag=0354a2e1b960c9cc2279eca4e5f84e20.4b5c ?? Call-ID: 2a3e49c0-a5b9-11e1-a7ee-7b1f73a7ffcf ?? CSeq: 28601494 REGISTER ?? Contact: ;q=0.5;expires=3600 ?? Server: Sip EXpress router (0.9.7 (i386/linux)) ?? Content-Length: 0 ?? Warning: 392 217.74.179.29:5060 "Noisy feedback tells:? pid=5455 req_src_ip=78.138.90.58 req_src_port=6090 in_uri=sip:bluesip.net;transport=udp out_uri=sip:bluesip.net;transport=udp via_cnt==1" ?? ?? ------------------------------------------------------------------------ recv 564 bytes from udp/[41.34.123.243]:58881 at 23:32:55.930149: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjuk8DZGiaXfN7BmtslNHVjtSg5UXsbHXc ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=Pz.t-0tGXj3saM.ehBiB1EAMmmitze1D ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60934 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 744 bytes to udp/[41.34.123.243]:58881 at 23:32:55.931324: ?? ------------------------------------------------------------------------ ?? SIP/2.0 401 Unauthorized ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjuk8DZGiaXfN7BmtslNHVjtSg5UXsbHXc;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=Pz.t-0tGXj3saM.ehBiB1EAMmmitze1D ?? To: "Sherif 1003" ;tag=75XtgSv0H3tUp ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60934 REGISTER ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? WWW-Authenticate: Digest realm="sip.pcfone.com", nonce="ca1eabee-a5f8-11e1-a840-7b1f73a7ffcf", algorithm=MD5, qop="auth" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 830 bytes from udp/[41.34.123.243]:58881 at 23:32:56.165624: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjKEQ6xXmyzZxA6vCM88vHNYAOPAjUqFsD ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=Pz.t-0tGXj3saM.ehBiB1EAMmmitze1D ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60935 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Authorization: Digest username="1002", realm="sip.pcfone.com", nonce="ca1eabee-a5f8-11e1-a840-7b1f73a7ffcf", uri="sip:sip.pcfone.com:6090", response="ff6d12f4ec7abd4b37187222b3af2358", algorithm=MD5, cnonce="sc3AJfhuev18Bs-L-6tORmI.RXlxnPOz", qop=auth, nc=00000001 ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 706 bytes to udp/[41.34.123.243]:58881 at 23:32:56.168033: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjKEQ6xXmyzZxA6vCM88vHNYAOPAjUqFsD;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=Pz.t-0tGXj3saM.ehBiB1EAMmmitze1D ?? To: "Sherif 1003" ;tag=8eQKjmD4ecHej ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60935 REGISTER ?? Contact: ;expires=300 ?? Date: Thu, 24 May 2012 23:32:56 GMT ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ send 989 bytes to udp/[41.34.123.243]:58881 at 23:32:56.360238: ?? ------------------------------------------------------------------------ ?? NOTIFY sip:1002 at 192.168.178.26:58881;ob SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKK7gFZ15FSet5B ?? Route: ;ob ?? Max-Forwards: 70 ?? From: ;tag=9QgcmFy7BN70D ?? To: ?? Call-ID: a1be7708-209b-1230-8985-00163e6bb553 ?? CSeq: 28615124 NOTIFY ?? Contact: ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? 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=noresource ?? Content-Type: application/simple-message-summary ?? Content-Length: 91 ?? ?? Messages-Waiting: yes ?? Message-Account: sip:1002 at 78.138.90.58 ?? Voice-Message: 1/0 (0/0) ?? ?? ------------------------------------------------------------------------ recv 308 bytes from udp/[41.34.123.243]:58881 at 23:32:56.623554: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;received=78.138.90.58;branch=z9hG4bKK7gFZ15FSet5B ?? Call-ID: a1be7708-209b-1230-8985-00163e6bb553 ?? From: ;tag=9QgcmFy7BN70D ?? To: ;tag=z9hG4bKK7gFZ15FSet5B ?? CSeq: 28615124 NOTIFY ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ freeswitch at internal> --- On Thu, 5/24/12, Kristian Kielhofner wrote: From: Kristian Kielhofner Subject: Re: [Freeswitch-users] Freeswitch (SBC + Media) does not know that gateway phone is picked up To: "FreeSWITCH Users Help" Date: Thursday, May 24, 2012, 7:51 PM Siptrace and logs please. On Thu, May 24, 2012 at 11:53 AM, Sherif Omran wrote: > > Hi all, > > My topology is as follows: > > Kamailio? -> FS (SBS+Media server) > > I came across an issue with my system as follows.? I have a Hardphone registered. When I do local call inside kamailio, it gets to FS and returns back well and FS understands when I lift the handset. However, I added a gateway (german landline server), when I call my self from another phone, the call gets to FS and then transmits to Kamailio, it rings my extension but when I lift the handset FS does not notice it and keeps ringing. > > Any body has an Idea? Here is my gateway trunk. > > > ??????? > ??????? > ??????? > ??????? > ??????? > ????? > > > thanks in advance > Sherif Omran > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Kristian Kielhofner _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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 at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -----Inline Attachment Follows----- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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 at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -----Inline Attachment Follows----- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120525/914945d5/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: call.pcap.zip Type: application/x-zip-compressed Size: 69987 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120525/914945d5/attachment-0001.bin From ivanov.evtim at gmail.com Fri May 25 16:22:07 2012 From: ivanov.evtim at gmail.com (Evtim Ivanov) Date: Fri, 25 May 2012 13:22:07 +0100 Subject: [Freeswitch-users] destination sip port ? In-Reply-To: References: Message-ID: Hello All > > I don't want to waste your time , so I'll be as short as possible. > My question is : How to change the destination sip port to one of the > gateways (providers). This provider listen not on standard sip port 5060 > but on 5069. > I tried to set : but it > doesn't work this way. Again trying to send register on 5060 (I dumped). I > also searched in the web but didn't find the answer till now. > So, what is the proper way? > > Thanks in advance > and Best regards > Evtim > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120525/80a314db/attachment.html From peter.olsson at visionutveckling.se Fri May 25 16:56:22 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Fri, 25 May 2012 12:56:22 +0000 Subject: [Freeswitch-users] destination sip port ? Message-ID: <1FFF97C269757C458224B7C895F35F150D8B70@cantor.std.visionutv.se> Try setting "register-proxy" instead. /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Evtim Ivanov Skickat: den 25 maj 2012 14:22 Till: freeswitch-users at lists.freeswitch.org ?mne: [Freeswitch-users] destination sip port ? Hello All I don't want to waste your time , so I'll be as short as possible. My question is : How to change the destination sip port to one of the gateways (providers). This provider listen not on standard sip port 5060 but on 5069. I tried to set : but it doesn't work this way. Again trying to send register on 5060 (I dumped). I also searched in the web but didn't find the answer till now. So, what is the proper way? Thanks in advance and Best regards Evtim !DSPAM:4fbf7d7a32761808049375! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120525/0a409a6a/attachment.html From william.suffill at gmail.com Fri May 25 19:00:54 2012 From: william.suffill at gmail.com (William Suffill) Date: Fri, 25 May 2012 11:00:54 -0400 Subject: [Freeswitch-users] Weekly Conference Call In-Reply-To: References: Message-ID: Was 5-23 recorded? The torrent link 404's. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120525/64d11107/attachment.html From krice at freeswitch.org Fri May 25 19:11:59 2012 From: krice at freeswitch.org (Ken Rice) Date: Fri, 25 May 2012 10:11:59 -0500 Subject: [Freeswitch-users] Weekly Conference Call In-Reply-To: Message-ID: Yeah it was recorded but Collins who posts the recordings is on vacation (or was) so give it a minute On 5/25/12 10:00 AM, "William Suffill" wrote: > Was 5-23 recorded? The torrent link 404's. > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120525/21ec543a/attachment.html From rico-freeswitch at ricozome.net Fri May 25 19:27:34 2012 From: rico-freeswitch at ricozome.net (rico-freeswitch at ricozome.net) Date: Fri, 25 May 2012 17:27:34 +0200 Subject: [Freeswitch-users] My SIP provider is stripping phone numbers ! Message-ID: <4FBFA4E6.10707@ricozome.net> Hi list, I have a stupid issue with my SIP provider "SCT Telecom" (in France) : it strip the first digit of the caller p?one number ! After many support requests they finally reconised the problem is at their side, but their answer is they'll not fix at their side and I must fix it at my side :-/ So I had a look on variables on public context and I see this : Channel-Name: [sofia/external/661864068 at sct-voip.fr] Caller-Username: [661864068] Caller-Caller-ID-Name: [661864068] Caller-Caller-ID-Number: [661864068] Caller-ANI: [661864068] Caller-Channel-Name: [sofia/external/661864068 at sct-voip.fr] variable_sip_from_user: [661864068] variable_sip_from_uri: [661864068 at sct-voip.fr] variable_sip_from_user_stripped: [661864068] variable_sip_full_from: [;tag=21b05c16] variable_sip_contact_user: [661864068] variable_sip_contact_uri: [661864068 at 86.64.146.248] variable_channel_name: [sofia/external/661864068 at sct-voip.fr] variable_caller_id_number: [661864068] The expected caller number is 0661864068, and as you can see is stripped to 661864068 So I added this in the begining of my public dialplan : It works as expected as the variable caller_id_number is updated : variable_caller_id_number: [0661864068] But when it starts to the default context dialplan, the caller number is still truncated as 661864068. So I wonder I handle the problem the wrong way... Any help would be appreciated :) Thanks and regards, -- Rico From william.suffill at gmail.com Fri May 25 19:35:41 2012 From: william.suffill at gmail.com (William Suffill) Date: Fri, 25 May 2012 11:35:41 -0400 Subject: [Freeswitch-users] Weekly Conference Call In-Reply-To: References: Message-ID: He gets to take a vacation? What's that? =) Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120525/f4542ed9/attachment.html From peter.olsson at visionutveckling.se Fri May 25 19:46:30 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Fri, 25 May 2012 15:46:30 +0000 Subject: [Freeswitch-users] My SIP provider is stripping phone numbers ! In-Reply-To: <4FBFA4E6.10707@ricozome.net> References: <4FBFA4E6.10707@ricozome.net> Message-ID: <95AE4DCD-8A97-471E-BA44-D47FFF227D0A@visionutveckling.se> Try setting effective_caller_id_number before you bridge the call. /Peter 25 maj 2012 kl. 17:34 skrev "rico-freeswitch at ricozome.net" : > Hi list, > > I have a stupid issue with my SIP provider "SCT Telecom" (in France) : > it strip the first digit of the caller p?one number ! > After many support requests they finally reconised the problem is at > their side, but their answer is they'll not fix at their side and I must > fix it at my side :-/ > > So I had a look on variables on public context and I see this : > > Channel-Name: [sofia/external/661864068 at sct-voip.fr] > Caller-Username: [661864068] > Caller-Caller-ID-Name: [661864068] > Caller-Caller-ID-Number: [661864068] > Caller-ANI: [661864068] > Caller-Channel-Name: [sofia/external/661864068 at sct-voip.fr] > variable_sip_from_user: [661864068] > variable_sip_from_uri: [661864068 at sct-voip.fr] > variable_sip_from_user_stripped: [661864068] > variable_sip_full_from: > [;tag=21b05c16] > variable_sip_contact_user: [661864068] > variable_sip_contact_uri: [661864068 at 86.64.146.248] > variable_channel_name: [sofia/external/661864068 at sct-voip.fr] > variable_caller_id_number: [661864068] > > The expected caller number is 0661864068, and as you can see is stripped > to 661864068 > > So I added this in the begining of my public dialplan : > > > expression="^01645327|^01645380|^01714170"> > > > > data="caller_id_number=0${caller_id_number}"/> > > > > It works as expected as the variable caller_id_number is updated : > variable_caller_id_number: [0661864068] > But when it starts to the default context dialplan, the caller number is > still truncated as 661864068. > > So I wonder I handle the problem the wrong way... Any help would be > appreciated :) > > Thanks and regards, > > -- > Rico > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > !DSPAM:4fbfa38b32765983520329! > From paul at cupis.co.uk Fri May 25 19:50:29 2012 From: paul at cupis.co.uk (Paul Cupis) Date: Fri, 25 May 2012 16:50:29 +0100 Subject: [Freeswitch-users] My SIP provider is stripping phone numbers ! In-Reply-To: <4FBFA4E6.10707@ricozome.net> References: <4FBFA4E6.10707@ricozome.net> Message-ID: <20120525155029.GA29643@eagle.cupis.co.uk> On Fri, May 25, 2012 at 05:27:34PM +0200, rico-freeswitch at ricozome.net wrote: > I have a stupid issue with my SIP provider "SCT Telecom" (in France) : > it strip the first digit of the caller p??one number ! This is not abnormal - the leading zero is not really present in the CLI on the providers network, just added for presentation to end-users. > The expected caller number is 0661864068, and as you can see is stripped > to 661864068 > > So I added this in the begining of my public dialplan : > > > expression="^01645327|^01645380|^01714170"> > > > > data="caller_id_number=0${caller_id_number}"/> > > > > It works as expected as the variable caller_id_number is updated : > variable_caller_id_number: [0661864068] > But when it starts to the default context dialplan, the caller number is > still truncated as 661864068. You probably want to look at using effective_caller_id_name/number rather than trying to change caller_id_name/number. http://wiki.freeswitch.org/wiki/Channel_Variables#Caller_ID_Related Regards, From anthony.minessale at gmail.com Fri May 25 20:06:06 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 25 May 2012 11:06:06 -0500 Subject: [Freeswitch-users] RTP media issue In-Reply-To: References: <02a201cd3a1a$2a658770$7f309650$@gmail.com> Message-ID: What version of FS are you running? Do you have the debug logs of those calls? you could try using the jitterbuffer. in the inbound DP to FS *before* you answer. Also it looks a little odd to me in this trace if this is the same call, it seems like you answered the call before placing the call to the phone and that phone never answers.... On Thu, May 24, 2012 at 9:51 PM, Nathan Downes wrote: > Hi, > > enable-soa > > > > Set the value to "false" to diable SIP SOA from sofia to tell sofia not to > touch the exchange of SDP > > I don't think this is related to the exchange of an SDP message..? Can you > elaborate more before I try it? I can't make things worse or change things I > don't understand. > > ________________________________ > From: djbinter at gmail.com > To: freeswitch-users at lists.freeswitch.org > CC: nathan at nortec.com.au > Subject: Re: [Freeswitch-users] RTP media issue > Date: Fri, 25 May 2012 11:19:46 +1000 > > > > > > Sent from my iPad > > On May 24, 2012, at 5:01 PM, Nathan Downes wrote: > > Hi, > > I had previous reported an issue with poor voice quality, appearing to stem > from occasion wrong timestamps coming from provider, but the end user's > experience was much worse than what I could see/hear in the trace. > > I have finally captured an event inbound and outbound.? The thing I don't > understand is I thought even though FS proxied the media it didn't touch it > or change anything, but it appears it is. > > The 2 traces are http://www.nortec.com.au/inbound.pcap.gz and > http://www.nortec.com.au/outbound.pcap.gz > > Inbound is from my trunk provider to FS box and outbound is FS box to ATA in > FTTH GPON. > > The event I am talking about, if both traces are open, is in the inbound one > inbetween packet 8114 and 8117 the provider drops a packet or I don't > receive it.? In the corresponding outbound trace, between packet 8144 and > 8152,? it appears FS misses a whole heap of packets (.1 seconds) between > 8146 and 8152 then it increases the timestamp only by 40 rather than 160 on > packet 8152.? This seems to not affect SIP phones themselves but causes > issues with the FTTH GPON ATA. > > This causes a gap in the audio for the end user, and when they miss a high > number of packets even though it sounds good on the inbound trace the end > users experience is horrible.?? This trace is actually a good one, but the > wrong timestamp can occur once per second, causing end user to lose 10%+ of > incoming audio only.? The issue only affects the audio coming from provider > to FS to end user. > > I am chasing it up with the voice provider to try and eliminate the > occasional packet loss, but if I could stop/fix FS from doing its > adjustment/gap/something the end user wouldn't even notice it. > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From ivanov.evtim at gmail.com Fri May 25 18:12:49 2012 From: ivanov.evtim at gmail.com (Evtim Ivanov) Date: Fri, 25 May 2012 15:12:49 +0100 Subject: [Freeswitch-users] destination sip port ? In-Reply-To: <1FFF97C269757C458224B7C895F35F150D8B70@cantor.std.visionutv.se> References: <1FFF97C269757C458224B7C895F35F150D8B70@cantor.std.visionutv.se> Message-ID: Thanks, this way works! 2012/5/25 Peter Olsson > Try setting ?register-proxy? instead.**** > > ** ** > > /Peter**** > > ** ** > > *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *F?r *Evtim Ivanov > *Skickat:* den 25 maj 2012 14:22 > *Till:* freeswitch-users at lists.freeswitch.org > *?mne:* [Freeswitch-users] destination sip port ?**** > > ** ** > > Hello All**** > > > I don't want to waste your time , so I'll be as short as possible. > My question is : How to change the destination sip port to one of the > gateways (providers). This provider listen not on standard sip port 5060 > but on 5069. > I tried to set : but it > doesn't work this way. Again trying to send register on 5060 (I dumped). I > also searched in the web but didn't find the answer till now. > So, what is the proper way? > > Thanks in advance > and Best regards > Evtim**** > > > !DSPAM:4fbf7d7a32761808049375! **** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120525/b40f8e53/attachment.html From sdevoy at bizfocused.com Fri May 25 22:26:32 2012 From: sdevoy at bizfocused.com (Sean Devoy) Date: Fri, 25 May 2012 14:26:32 -0400 Subject: [Freeswitch-users] SCA not working inbound - Multi Domain Message-ID: <040301cd3aa3$e93705f0$bba511d0$@bizfocused.com> Hi all, I have a muti-tennnant configuration that is working nicely except for Shared Call Appearance. The desktop devices are CISCO 504Gs and they are configured as described in the FS Wiki as well as Cisco Documentation. The SCA works perfectly for outbound calls - if either phone pickups like 220, the other phones indicator light flashes red. However, inbound calls will go to only one of the phones (which one has changed a few times) and the other phones line still just stays green and does not ring. Here is the sip interfaces config: The directory entry which both phones connect using: " And the dial plan for ext 220: I did see this in the wiki (http://wiki.freeswitch.org/wiki/Shared_Line_Appearance): If SLA works for outgoing calls and SLA does not work for inbound calls to the SLA phones, you may have some presence problem related to mixed IP and domain names. When using ODBC you may issue the following SQL statement select sip_to_host,sip_from_user,sip_from_host,hostname,presence_id,call_info from sip_dialogs; But I don't have ODBC on this server, so I am a little lost. I have the phones login to domain names, not addresses. I never refer to IP addresses in my xml (except gateways addresses). I am not trying SLA across domain, only within the same domain. I hope someone can spot something. Thanks for your help. Sean -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120525/608f1a60/attachment-0001.html From anthony.minessale at gmail.com Fri May 25 22:35:27 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 25 May 2012 13:35:27 -0500 Subject: [Freeswitch-users] SCA not working inbound - Multi Domain In-Reply-To: <040301cd3aa3$e93705f0$bba511d0$@bizfocused.com> References: <040301cd3aa3$e93705f0$bba511d0$@bizfocused.com> Message-ID: What are the phones putting in the subscribe ? sofia global siptrace on sofia global debug presence|sla then watch for SUBSCRIBE also when you are not using odbc you can get the sql with this app sqlute3 /usr/local/freeswitch/db/sofia_reg_internal.db also try "select * from sip_subscriptions" its all about using the right host name across the board, IP's count as hostnames, they do not magically resolve any dns with SIP On Fri, May 25, 2012 at 1:26 PM, Sean Devoy wrote: > Hi all, > > > > I have a muti-tennnant configuration that is working nicely except for > Shared Call Appearance.? The desktop devices are CISCO 504Gs and they are > configured as described in the FS Wiki as well as Cisco Documentation. > > > > The SCA works perfectly for outbound calls ? if either phone pickups like > 220, the other phones indicator light flashes red.? However, inbound calls > will go to only one of the phones (which one has changed a few times) and > the other phones line still just stays green and does not ring. > > > > Here is the sip interfaces config: > > > > ??? > > ????? > > ????? > > ????? > > ????? > > ????? > > ????? > > ????? > > ????? > > ????? > > ????? > > ????? > > ????? > > ????? > > ????? > > ????? > > ????? > > ????? > > ????? > > ????? > > ??? > > ? > > > > The directory entry which both phones connect using: > > ??? > > ????? > > ??????? > > ??????? > > ??????? > > ??????? > > ??????? > > ??????? > > ??????? > > ??????? > > ??????? > > ????? > > ????? > > ??????? " > > ??????? > > ??????? value="{presence_id=220 at mydomainname.com}${sofia_contact(220 at mydomainname.com)}"/> > > ??????? > > ????? > > ??? > > > > And the dial plan for ext 220: > > ? > > ??? > > ????? data="effective_caller_id_number=${internal_caller_id_number}"/> > > ????? data="effective_caller_id_name=${internal_caller_id_name}"/> > > ????? > > ????? > > ????? > > ????? data="{sip_invite_domain=mydomainname.com}user/220 at mydomainname.com"? /> > > ?????? > > ????? > > ????? > > ??? > > ? > > > > > > > > I did see this in the wiki > (http://wiki.freeswitch.org/wiki/Shared_Line_Appearance): > > If SLA works for outgoing calls and SLA does not work for inbound calls to > the SLA phones, you may have some presence problem related to mixed IP and > domain names. When using ODBC you may issue the following SQL statement > > select > sip_to_host,sip_from_user,sip_from_host,hostname,presence_id,call_info from > sip_dialogs; > > But I don?t have ODBC on this server, so I am a little lost. > > > > I have the phones login to domain names, not addresses.? I never refer to IP > addresses in my xml (except gateways addresses).? I am not trying SLA across > domain, only within the same domain. > > > > I hope someone can spot something.? Thanks for your help. > > > > Sean > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From jalsot at gmail.com Fri May 25 23:38:13 2012 From: jalsot at gmail.com (Tamas Jalsovszky) Date: Fri, 25 May 2012 21:38:13 +0200 Subject: [Freeswitch-users] fax_header question Message-ID: Hello, I'm playing with fax sending and I need to set the fax header for my own line. I can set fax_header and that works nearly perfectly for me. What I need to figure out is, how to set the whole header line, as it seems, spandsp adds the date and time to the left side and ident and page number to the right side. I tried to fill the whole fax_ident variable, but found from the codes, that it has a 50 character limit. The best would be to have fully configurable the header and maybe define the page number as a pattern (like efax or hylafax has if I recall well). Any idea? Kind regards, jalsot -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120525/2d9c2bbb/attachment.html From sherifomran2000 at yahoo.com Sat May 26 00:00:22 2012 From: sherifomran2000 at yahoo.com (Sherif Omran) Date: Fri, 25 May 2012 13:00:22 -0700 (PDT) Subject: [Freeswitch-users] Freeswitch (SBC + Media) does not know that gateway phone is picked up In-Reply-To: <1337935946.46011.YahooMailClassic@web110806.mail.gq1.yahoo.com> Message-ID: <1337976022.10963.YahooMailClassic@web110814.mail.gq1.yahoo.com> Hello guys, here is a pastebin of the call /log 0 http://pastebin.freeswitch.org/19175 detailed is here http://pastebin.freeswitch.org/19174 Any body has an idea? thanks Sherif Omran --- On Fri, 5/25/12, Sherif Omran wrote: From: Sherif Omran Subject: Re: [Freeswitch-users] Freeswitch (SBC + Media) does not know that gateway phone is picked up To: "FreeSWITCH Users Help" Date: Friday, May 25, 2012, 11:52 AM Hello guys, here is the trace (see the attachment) kind regards Sherif --- On Fri, 5/25/12, SamyGo wrote: From: SamyGo Subject: Re: [Freeswitch-users] Freeswitch (SBC + Media) does not know that gateway phone is picked up To: "FreeSWITCH Users Help" Date: Friday, May 25, 2012, 10:46 AM Hi again,If you want kamailio register to the provider then use UAC module. Kamailio will use the username/password and register with the provider. Regards,Sammy On Fri, May 25, 2012 at 12:20 PM, Sherif Omran wrote: Dear Sammy, Thank you for your question ... Yes, the GW is registered through FS because I did not know how to register it to kamailio. But it seems better to register it to kamailio. One more information, calls from 1001 to 1002 go to kamailio then to FS then return back to kamailio smoothly. Thus I would suggest that I change registering the gateway from FS to kamailio. but How to? --- On Fri, 5/25/12, SamyGo wrote: From: SamyGo Subject: Re: [Freeswitch-users] Freeswitch (SBC + Media) does not know that gateway phone is picked up To: "FreeSWITCH Users Help" Date: Friday, May 25, 2012, 8:08 AM Hi,These are SIP traces on FreeSWITCH console, whereas you are saying and it seems that kamailio is not detecting the answering (200 OK)of the call from extension 1002. Please, can you take a sip trace..!! I see you've both kamailio and FS on same server! Please take a pcap from the linux console using the following command. #tcpdump -i any -s 0 -w new-call-trace.pcap -vvv Please be quick on turning the sip trace on and off as quickly as possible to avoid extra packets. Once done open the file in wireshark ;?apply filter "sip || rtp" and then save the resulting capture in separate file. Send us the new file to analyse. One more silly question probably, I see REGISTERs coming to your FS as well and the calls to gateways are made from FS too !!, umm...just thinking what are you using kamailio for!!? ThanksBR,Sammy On Fri, May 25, 2012 at 4:40 AM, Sherif Omran wrote: Hi all, here is the siptrace: To figure it out 1- gateway called bluesip.net. It send invide using caller number at bluesip.net 2- This call should go to extension kb-1002. kb means go from freeswitch port 6090 to kamailio port 5060 3- It should go to call extension 1002 in Kamailio 4- Extension 1002 rings but when I reply, it does not notice I replied ./fs_cli ??????????? _____ ____???? ____ _???? ___????????????? ?????????? |? ___/ ___|?? / ___| |?? |_ _|???????????? ?????????? | |_? \___ \? | |?? | |??? | |??????????? ?????????? |? _|? ___) | | |___| |___ | |????????????? ?????????? |_|?? |____/?? \____|_____|___|??????????? ******************************************************* * Anthony Minessale II, Ken Rice,???????????????????? * * Michael Jerris, Travis Cross??????????????????????? * * FreeSWITCH (http://www.freeswitch.org)????????????? * * Paypal Donations Appreciated: paypal at freeswitch.org * * Brought to you by ClueCon http://www.cluecon.com/?? * ******************************************************* Type /help to see a list of commands +OK log level? [7] freeswitch at internal> tracelevel -ERR tracelevel Command not found! freeswitch at internal> sofia global siptrace on +OK Global siptrace on recv 1104 bytes from udp/[217.74.179.29]:5060 at 23:08:45.822962: ?? ------------------------------------------------------------------------ ?? INVITE sip:gw+trunk_1000 at 78.138.90.58:6090;transport=udp;gw=trunk_1000 SIP/2.0 ?? Record-Route: ?? Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 ?? Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 ?? From: "+41793940965" ;tag=as00589402 ?? To: ?? Contact: ?? Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net ?? CSeq: 102 INVITE ?? User-Agent: blueSIP PSTN GW ?? Max-Forwards: 69 ?? Date: Thu, 24 May 2012 23:08:44 GMT ?? Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY ?? Supported: replaces ?? Content-Type: application/sdp ?? Content-Length: 367 ?? P-hint: USRLOC ?? ?? v=0 ?? o=root 20076 20076 IN IP4 217.74.179.28 ?? s=session ?? c=IN IP4 217.74.179.28 ?? t=0 0 ?? m=audio 25626 RTP/AVP 8 0 18 111 3 101 ?? a=rtpmap:8 PCMA/8000 ?? a=rtpmap:0 PCMU/8000 ?? a=rtpmap:18 G729/8000 ?? a=fmtp:18 annexb=no ?? a=rtpmap:111 G726-32/8000 ?? a=rtpmap:3 GSM/8000 ?? a=rtpmap:101 telephone-event/8000 ?? a=fmtp:101 0-16 ?? a=silenceSupp:off - - - - ?? a=ptime:20 ?? a=sendrecv ?? ------------------------------------------------------------------------ send 494 bytes to udp/[217.74.179.29]:5060 at 23:08:45.844486: ?? ------------------------------------------------------------------------ ?? SIP/2.0 100 Trying ?? Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 ?? Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 ?? Record-Route: ?? From: "+41793940965" ;tag=as00589402 ?? To: ?? Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net ?? CSeq: 102 INVITE ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ 2012-05-24 23:08:45.840749 [NOTICE] switch_channel.c:926 New Channel sofia/internal/+41793940965 at bluesip.net [69cd22e6-a5f5-11e1-a833-7b1f73a7ffcf] 2012-05-24 23:08:45.840749 [DEBUG] sofia.c:5532 Channel sofia/internal/+41793940965 at bluesip.net entering state [received][100] 2012-05-24 23:08:45.840749 [DEBUG] sofia.c:5543 Remote SDP: v=0 o=root 20076 20076 IN IP4 217.74.179.28 s=session c=IN IP4 217.74.179.28 t=0 0 m=audio 25626 RTP/AVP 8 0 18 111 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:111 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMA:8:8000:20:64000]/[iLBC:97:8000:30:13330] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMU:0:8000:20:64000]/[iLBC:97:8000:30:13330] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G729:18:8000:20:8000]/[iLBC:97:8000:30:13330] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G726-32:111:8000:20:0]/[iLBC:97:8000:30:13330] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [GSM:3:8000:20:13200]/[iLBC:97:8000:30:13330] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [telephone-event:101:8000:20:0]/[iLBC:97:8000:30:13330] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf send/recv payload to 101 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMA:8:8000:20:64000]/[G7221:115:32000:20:48000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMU:0:8000:20:64000]/[G7221:115:32000:20:48000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G729:18:8000:20:8000]/[G7221:115:32000:20:48000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G726-32:111:8000:20:0]/[G7221:115:32000:20:48000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [GSM:3:8000:20:13200]/[G7221:115:32000:20:48000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [telephone-event:101:8000:20:0]/[G7221:115:32000:20:48000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf send/recv payload to 101 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMA:8:8000:20:64000]/[G7221:107:16000:20:32000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMU:0:8000:20:64000]/[G7221:107:16000:20:32000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G729:18:8000:20:8000]/[G7221:107:16000:20:32000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G726-32:111:8000:20:0]/[G7221:107:16000:20:32000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [GSM:3:8000:20:13200]/[G7221:107:16000:20:32000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [telephone-event:101:8000:20:0]/[G7221:107:16000:20:32000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf send/recv payload to 101 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMA:8:8000:20:64000]/[G722:9:8000:20:64000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G729:18:8000:20:8000]/[G722:9:8000:20:64000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G726-32:111:8000:20:0]/[G722:9:8000:20:64000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [GSM:3:8000:20:13200]/[G722:9:8000:20:64000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [telephone-event:101:8000:20:0]/[G722:9:8000:20:64000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf send/recv payload to 101 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMA:8:8000:20:64000]/[PCMU:0:8000:20:64000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:2996 Set Codec sofia/internal/+41793940965 at bluesip.net PCMU/8000 20 ms 160 samples 64000 bits 2012-05-24 23:08:45.840749 [DEBUG] switch_core_codec.c:111 sofia/internal/+41793940965 at bluesip.net Original read codec set to PCMU:0 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf send/recv payload to 101 2012-05-24 23:08:45.840749 [DEBUG] sofia.c:5757 (sofia/internal/+41793940965 at bluesip.net) State Change CS_NEW -> CS_INIT 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_INIT 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/+41793940965 at bluesip.net) State INIT 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:85 sofia/internal/+41793940965 at bluesip.net SOFIA INIT 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:125 (sofia/internal/+41793940965 at bluesip.net) State Change CS_INIT -> CS_ROUTING 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/+41793940965 at bluesip.net) State INIT going to sleep 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_ROUTING 2012-05-24 23:08:45.840749 [DEBUG] switch_channel.c:1886 (sofia/internal/+41793940965 at bluesip.net) Callstate Change DOWN -> RINGING 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/+41793940965 at bluesip.net) State ROUTING 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:148 sofia/internal/+41793940965 at bluesip.net SOFIA ROUTING 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:104 sofia/internal/+41793940965 at bluesip.net Standard ROUTING 2012-05-24 23:08:45.840749 [INFO] mod_dialplan_xml.c:485 Processing +41793940965 <+41793940965>->kb-1002 in context public Dialplan: sofia/internal/+41793940965 at bluesip.net parsing [public->from_kamailio] continue=false Dialplan: sofia/internal/+41793940965 at bluesip.net Regex (PASS) [from_kamailio] destination_number(kb-1002) =~ /^(.+)$/ break=on-false Dialplan: sofia/internal/+41793940965 at bluesip.net Action transfer(kb-1002 XML default) 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:154 (sofia/internal/+41793940965 at bluesip.net) State Change CS_ROUTING -> CS_EXECUTE 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/+41793940965 at bluesip.net) State ROUTING going to sleep 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_EXECUTE 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/+41793940965 at bluesip.net) State EXECUTE 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:241 sofia/internal/+41793940965 at bluesip.net SOFIA EXECUTE 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:192 sofia/internal/+41793940965 at bluesip.net Standard EXECUTE EXECUTE sofia/internal/+41793940965 at bluesip.net transfer(kb-1002 XML default) 2012-05-24 23:08:45.840749 [DEBUG] switch_ivr.c:1711 (sofia/internal/+41793940965 at bluesip.net) State Change CS_EXECUTE -> CS_ROUTING 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:731 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:08:45.840749 [NOTICE] switch_ivr.c:1717 Transfer sofia/internal/+41793940965 at bluesip.net to XML[kb-1002 at default] 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/+41793940965 at bluesip.net) State EXECUTE going to sleep 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_ROUTING 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/+41793940965 at bluesip.net) State ROUTING 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:148 sofia/internal/+41793940965 at bluesip.net SOFIA ROUTING 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:104 sofia/internal/+41793940965 at bluesip.net Standard ROUTING 2012-05-24 23:08:45.840749 [INFO] mod_dialplan_xml.c:485 Processing +41793940965 <+41793940965>->kb-1002 in context default Dialplan: sofia/internal/+41793940965 at bluesip.net parsing [default->vbox] continue=false Dialplan: sofia/internal/+41793940965 at bluesip.net Regex (FAIL) [vbox] destination_number(kb-1002) =~ /^vb-([0-9][0-9][0-9])$/ break=on-false Dialplan: sofia/internal/+41793940965 at bluesip.net parsing [default->vmenu] continue=false Dialplan: sofia/internal/+41793940965 at bluesip.net Regex (FAIL) [vmenu] destination_number(kb-1002) =~ /^vm-([0-9][0-9][0-9][0-9])$/ break=on-false Dialplan: sofia/internal/+41793940965 at bluesip.net parsing [default->kbridge] continue=false Dialplan: sofia/internal/+41793940965 at bluesip.net Regex (PASS) [kbridge] destination_number(kb-1002) =~ /^kb-(.+)$/ break=on-false Dialplan: sofia/internal/+41793940965 at bluesip.net Action set(proxy_media=true) Dialplan: sofia/internal/+41793940965 at bluesip.net Action set(call_timeout=50) Dialplan: sofia/internal/+41793940965 at bluesip.net Action set(continue_on_fail=true) Dialplan: sofia/internal/+41793940965 at bluesip.net Action set(hangup_after_bridge=true) Dialplan: sofia/internal/+41793940965 at bluesip.net Action set(sip_invite_domain=78.138.90.58) Dialplan: sofia/internal/+41793940965 at bluesip.net Action export(sip_contact_user=ufs) Dialplan: sofia/internal/+41793940965 at bluesip.net Action bridge(sofia/78.138.90.58/1002 at 78.138.90.58:5060) Dialplan: sofia/internal/+41793940965 at bluesip.net Action answer() Dialplan: sofia/internal/+41793940965 at bluesip.net Action voicemail(default ${domain_name} 1002) 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:154 (sofia/internal/+41793940965 at bluesip.net) State Change CS_ROUTING -> CS_EXECUTE 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/+41793940965 at bluesip.net) State ROUTING going to sleep 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_EXECUTE 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/+41793940965 at bluesip.net) State EXECUTE 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:241 sofia/internal/+41793940965 at bluesip.net SOFIA EXECUTE 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:192 sofia/internal/+41793940965 at bluesip.net Standard EXECUTE EXECUTE sofia/internal/+41793940965 at bluesip.net set(proxy_media=true) 2012-05-24 23:08:45.840749 [DEBUG] mod_dptools.c:1281 sofia/internal/+41793940965 at bluesip.net SET [proxy_media]=[true] EXECUTE sofia/internal/+41793940965 at bluesip.net set(call_timeout=50) 2012-05-24 23:08:45.840749 [DEBUG] mod_dptools.c:1281 sofia/internal/+41793940965 at bluesip.net SET [call_timeout]=[50] EXECUTE sofia/internal/+41793940965 at bluesip.net set(continue_on_fail=true) 2012-05-24 23:08:45.840749 [DEBUG] mod_dptools.c:1281 sofia/internal/+41793940965 at bluesip.net SET [continue_on_fail]=[true] EXECUTE sofia/internal/+41793940965 at bluesip.net set(hangup_after_bridge=true) 2012-05-24 23:08:45.840749 [DEBUG] mod_dptools.c:1281 sofia/internal/+41793940965 at bluesip.net SET [hangup_after_bridge]=[true] EXECUTE sofia/internal/+41793940965 at bluesip.net set(sip_invite_domain=78.138.90.58) 2012-05-24 23:08:45.840749 [DEBUG] mod_dptools.c:1281 sofia/internal/+41793940965 at bluesip.net SET [sip_invite_domain]=[78.138.90.58] EXECUTE sofia/internal/+41793940965 at bluesip.net export(sip_contact_user=ufs) 2012-05-24 23:08:45.840749 [DEBUG] switch_channel.c:1093 EXPORT (export_vars) [sip_contact_user]=[ufs] EXECUTE sofia/internal/+41793940965 at bluesip.net bridge(sofia/78.138.90.58/1002 at 78.138.90.58:5060) 2012-05-24 23:08:45.840749 [DEBUG] switch_channel.c:1047 sofia/internal/+41793940965 at bluesip.net EXPORTING[export_vars] [sip_contact_user]=[ufs] to event 2012-05-24 23:08:45.840749 [DEBUG] switch_ivr_originate.c:1884 Parsing global variables 2012-05-24 23:08:45.840749 [NOTICE] switch_channel.c:926 New Channel sofia/internal/1002 at 78.138.90.58:5060 [69ce1f0c-a5f5-11e1-a838-7b1f73a7ffcf] 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:4691 (sofia/internal/1002 at 78.138.90.58:5060) State Change CS_NEW -> CS_INIT 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1002 at 78.138.90.58:5060) Running State Change CS_INIT 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/1002 at 78.138.90.58:5060) State INIT 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:85 sofia/internal/1002 at 78.138.90.58:5060 SOFIA INIT 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:1871 sofia/internal/1002 at 78.138.90.58:5060 Patched SDP --- v=0 o=root 20076 20076 IN IP4 217.74.179.28 s=session c=IN IP4 217.74.179.28 t=0 0 m=audio 25626 RTP/AVP 8 0 18 111 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:111 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 +++ v=0 o=FreeSWITCH 1629510322 1629510323 IN IP4 78.138.90.58 s=FreeSWITCH c=IN IP4 78.138.90.58 t=0 0 m=audio 31178 RTP/AVP 8 0 18 111 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:111 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 send 1340 bytes to udp/[78.138.90.58]:5060 at 23:08:45.852280: ?? ------------------------------------------------------------------------ ?? INVITE sip:1002 at 78.138.90.58:5060 SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKapDgFeegKyHXK ?? Max-Forwards: 67 ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? CSeq: 28614398 INVITE ?? Contact: ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? 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: 372 ?? P-hint: USRLOC ?? X-FS-Support: update_display,send_info ?? Remote-Party-ID: "+41793940965" ;party=calling;screen=yes;privacy=off ?? ?? v=0 ?? o=FreeSWITCH 1629510322 1629510323 IN IP4 78.138.90.58 ?? s=FreeSWITCH ?? c=IN IP4 78.138.90.58 ?? t=0 0 ?? m=audio 31178 RTP/AVP 8 0 18 111 3 101 ?? a=rtpmap:8 PCMA/8000 ?? a=rtpmap:0 PCMU/8000 ?? a=rtpmap:18 G729/8000 ?? a=fmtp:18 annexb=no ?? a=rtpmap:111 G726-32/8000 ?? a=rtpmap:3 GSM/8000 ?? a=rtpmap:101 telephone-event/8000 ?? a=fmtp:101 0-16 ?? a=silenceSupp:off - - - - ?? a=ptime:20 ?? ------------------------------------------------------------------------ 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:125 (sofia/internal/1002 at 78.138.90.58:5060) State Change CS_INIT -> CS_ROUTING 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/1002 at 78.138.90.58:5060) State INIT going to sleep 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1002 at 78.138.90.58:5060) Running State Change CS_ROUTING 2012-05-24 23:08:45.840749 [DEBUG] switch_channel.c:1886 (sofia/internal/1002 at 78.138.90.58:5060) Callstate Change DOWN -> RINGING 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/1002 at 78.138.90.58:5060) State ROUTING 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:148 sofia/internal/1002 at 78.138.90.58:5060 SOFIA ROUTING 2012-05-24 23:08:45.840749 [DEBUG] switch_ivr_originate.c:66 (sofia/internal/1002 at 78.138.90.58:5060) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/1002 at 78.138.90.58:5060) State ROUTING going to sleep 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1002 at 78.138.90.58:5060) Running State Change CS_CONSUME_MEDIA 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:429 (sofia/internal/1002 at 78.138.90.58:5060) State CONSUME_MEDIA 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:429 (sofia/internal/1002 at 78.138.90.58:5060) State CONSUME_MEDIA going to sleep 2012-05-24 23:08:45.840749 [DEBUG] sofia.c:5532 Channel sofia/internal/1002 at 78.138.90.58:5060 entering state [calling][0] recv 365 bytes from udp/[78.138.90.58]:5060 at 23:08:45.964309: ?? ------------------------------------------------------------------------ ?? SIP/2.0 100 trying -- your call is important to us ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKapDgFeegKyHXK ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? CSeq: 28614398 INVITE ?? Server: kamailio (3.1.5 (x86_64/linux)) ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 496 bytes from udp/[78.138.90.58]:5060 at 23:08:47.414766: ?? ------------------------------------------------------------------------ ?? SIP/2.0 180 Ringing ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKapDgFeegKyHXK ?? Record-Route: ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ;tag=qCSaB2YJOAsAr2dJ-RWNMhPoDdn1RFvs ?? CSeq: 28614398 INVITE ?? Contact: "Mama" ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, REGISTER ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ 2012-05-24 23:08:47.400741 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:47.400741 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:47.400741 [DEBUG] sofia.c:5532 Channel sofia/internal/1002 at 78.138.90.58:5060 entering state [proceeding][180] 2012-05-24 23:08:47.400741 [NOTICE] sofia.c:5624 Ring-Ready sofia/internal/1002 at 78.138.90.58:5060! send 995 bytes to udp/[217.74.179.29]:5060 at 23:08:47.416538: ?? ------------------------------------------------------------------------ ?? SIP/2.0 180 Ringing ?? Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 ?? Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 ?? Record-Route: ?? From: "+41793940965" ;tag=as00589402 ?? To: ;tag=S7UZQygFt62Nm ?? Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net ?? CSeq: 102 INVITE ?? Contact: ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Accept: application/sdp ?? 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-Length: 0 ?? Remote-Party-ID: "Outbound Call" ;party=calling;privacy=off;screen=no ?? ?? ------------------------------------------------------------------------ 2012-05-24 23:08:47.400741 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:08:47.400741 [NOTICE] mod_sofia.c:2514 Ring-Ready sofia/internal/+41793940965 at bluesip.net! 2012-05-24 23:08:47.400741 [DEBUG] sofia.c:5532 Channel sofia/internal/+41793940965 at bluesip.net entering state [early][180] 2012-05-24 23:08:47.400741 [DEBUG] switch_core_session.c:731 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:08:47.400741 [NOTICE] switch_ivr_originate.c:483 Ring Ready sofia/internal/+41793940965 at bluesip.net! recv 496 bytes from udp/[78.138.90.58]:5060 at 23:08:47.804967: ?? ------------------------------------------------------------------------ ?? SIP/2.0 180 Ringing ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKapDgFeegKyHXK ?? Record-Route: ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ;tag=549D2DD03BBA7C67 ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? CSeq: 28614398 INVITE ?? Contact: ?? User-Agent: AVM FRITZ!Box Fon WLAN 7170 29.04.80 (Jan 27 2010) ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ 2012-05-24 23:08:47.800746 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:47.800746 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:47.800746 [DEBUG] sofia.c:5532 Channel sofia/internal/1002 at 78.138.90.58:5060 entering state [proceeding][180] recv 496 bytes from udp/[78.138.90.58]:5060 at 23:08:48.182098: ?? ------------------------------------------------------------------------ ?? SIP/2.0 180 Ringing ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKapDgFeegKyHXK ?? Record-Route: ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ;tag=qCSaB2YJOAsAr2dJ-RWNMhPoDdn1RFvs ?? CSeq: 28614398 INVITE ?? Contact: "Mama" ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, REGISTER ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ 2012-05-24 23:08:48.180741 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:48.180741 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:48.180741 [DEBUG] sofia.c:5532 Channel sofia/internal/1002 at 78.138.90.58:5060 entering state [proceeding][180] recv 496 bytes from udp/[78.138.90.58]:5060 at 23:08:48.417403: ?? ------------------------------------------------------------------------ ?? SIP/2.0 180 Ringing ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKapDgFeegKyHXK ?? Record-Route: ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ;tag=549D2DD03BBA7C67 ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? CSeq: 28614398 INVITE ?? Contact: ?? User-Agent: AVM FRITZ!Box Fon WLAN 7170 29.04.80 (Jan 27 2010) ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ 2012-05-24 23:08:48.400744 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:48.400744 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:48.400744 [DEBUG] sofia.c:5532 Channel sofia/internal/1002 at 78.138.90.58:5060 entering state [proceeding][180] recv 496 bytes from udp/[78.138.90.58]:5060 at 23:08:49.292599: ?? ------------------------------------------------------------------------ ?? SIP/2.0 180 Ringing ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKapDgFeegKyHXK ?? Record-Route: ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ;tag=549D2DD03BBA7C67 ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? CSeq: 28614398 INVITE ?? Contact: ?? User-Agent: AVM FRITZ!Box Fon WLAN 7170 29.04.80 (Jan 27 2010) ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ 2012-05-24 23:08:49.280742 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:49.280742 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:49.280742 [DEBUG] sofia.c:5532 Channel sofia/internal/1002 at 78.138.90.58:5060 entering state [proceeding][180] recv 478 bytes from udp/[217.74.179.29]:5060 at 23:09:15.066012: ?? ------------------------------------------------------------------------ ?? CANCEL sip:gw+trunk_1000 at 78.138.90.58:6090;transport=udp;gw=trunk_1000 SIP/2.0 ?? Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 ?? Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 ?? From: "+41793940965" ;tag=as00589402 ?? To: ?? Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net ?? CSeq: 102 CANCEL ?? User-Agent: blueSIP PSTN GW ?? Max-Forwards: 69 ?? Content-Length: 0 ?? P-hint: USRLOC ?? ?? ------------------------------------------------------------------------ send 369 bytes to udp/[217.74.179.29]:5060 at 23:09:15.066176: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 ?? Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 ?? From: "+41793940965" ;tag=as00589402 ?? To: ;tag=S7UZQygFt62Nm ?? Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net ?? CSeq: 102 CANCEL ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ send 766 bytes to udp/[217.74.179.29]:5060 at 23:09:15.066289: ?? ------------------------------------------------------------------------ ?? SIP/2.0 487 Request Terminated ?? Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 ?? Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 ?? From: "+41793940965" ;tag=as00589402 ?? To: ;tag=S7UZQygFt62Nm ?? Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net ?? CSeq: 102 INVITE ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? 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-Length: 0 ?? ?? ------------------------------------------------------------------------ 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:09:15.060742 [DEBUG] sofia.c:5532 Channel sofia/internal/+41793940965 at bluesip.net entering state [terminated][487] 2012-05-24 23:09:15.060742 [DEBUG] switch_channel.c:2848 (sofia/internal/+41793940965 at bluesip.net) Callstate Change RINGING -> HANGUP 2012-05-24 23:09:15.060742 [NOTICE] sofia.c:6301 Hangup sofia/internal/+41793940965 at bluesip.net [CS_EXECUTE] [ORIGINATOR_CANCEL] 2012-05-24 23:09:15.060742 [DEBUG] switch_channel.c:2871 Send signal sofia/internal/+41793940965 at bluesip.net [KILL] 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:09:15.060742 [DEBUG] switch_channel.c:2848 (sofia/internal/1002 at 78.138.90.58:5060) Callstate Change RINGING -> HANGUP 2012-05-24 23:09:15.060742 [NOTICE] switch_ivr_originate.c:3182 Hangup sofia/internal/1002 at 78.138.90.58:5060 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL] 2012-05-24 23:09:15.060742 [DEBUG] switch_channel.c:2871 Send signal sofia/internal/1002 at 78.138.90.58:5060 [KILL] 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1002 at 78.138.90.58:5060) Running State Change CS_HANGUP 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/1002 at 78.138.90.58:5060) State HANGUP 2012-05-24 23:09:15.060742 [DEBUG] mod_sofia.c:469 Channel sofia/internal/1002 at 78.138.90.58:5060 hanging up, cause: ORIGINATOR_CANCEL 2012-05-24 23:09:15.060742 [DEBUG] mod_sofia.c:523 Sending CANCEL to sofia/internal/1002 at 78.138.90.58:5060 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:47 sofia/internal/1002 at 78.138.90.58:5060 Standard HANGUP, cause: ORIGINATOR_CANCEL 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/1002 at 78.138.90.58:5060) State HANGUP going to sleep 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:393 (sofia/internal/1002 at 78.138.90.58:5060) State Change CS_HANGUP -> CS_REPORTING 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1002 at 78.138.90.58:5060) Running State Change CS_REPORTING 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/1002 at 78.138.90.58:5060) State REPORTING 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:79 sofia/internal/1002 at 78.138.90.58:5060 Standard REPORTING, cause: ORIGINATOR_CANCEL 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/1002 at 78.138.90.58:5060) State REPORTING going to sleep 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:387 (sofia/internal/1002 at 78.138.90.58:5060) State Change CS_REPORTING -> CS_DESTROY 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1382 Session 9 (sofia/internal/1002 at 78.138.90.58:5060) Locked, Waiting on external entities 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:09:15.060742 [DEBUG] switch_ivr_originate.c:3358 Originate Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL] 2012-05-24 23:09:15.060742 [NOTICE] switch_core_session.c:1400 Session 9 (sofia/internal/1002 at 78.138.90.58:5060) Ended 2012-05-24 23:09:15.060742 [NOTICE] switch_core_session.c:1402 Close Channel sofia/internal/1002 at 78.138.90.58:5060 [CS_DESTROY] 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:491 (sofia/internal/1002 at 78.138.90.58:5060) Callstate Change HANGUP -> DOWN 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:494 (sofia/internal/1002 at 78.138.90.58:5060) Running State Change CS_DESTROY 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/1002 at 78.138.90.58:5060) State DESTROY 2012-05-24 23:09:15.060742 [DEBUG] mod_sofia.c:374 sofia/internal/1002 at 78.138.90.58:5060 SOFIA DESTROY 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:86 sofia/internal/1002 at 78.138.90.58:5060 Standard DESTROY 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/1002 at 78.138.90.58:5060) State DESTROY going to sleep 2012-05-24 23:09:15.060742 [INFO] mod_dptools.c:2922 Originate Failed.? Cause: ORIGINATOR_CANCEL 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:2287 sofia/internal/+41793940965 at bluesip.net skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/+41793940965 at bluesip.net) State EXECUTE going to sleep 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_HANGUP 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/+41793940965 at bluesip.net) State HANGUP 2012-05-24 23:09:15.060742 [DEBUG] mod_sofia.c:469 Channel sofia/internal/+41793940965 at bluesip.net hanging up, cause: ORIGINATOR_CANCEL 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:47 sofia/internal/+41793940965 at bluesip.net Standard HANGUP, cause: ORIGINATOR_CANCEL 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/+41793940965 at bluesip.net) State HANGUP going to sleep 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:393 (sofia/internal/+41793940965 at bluesip.net) State Change CS_HANGUP -> CS_REPORTING 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_REPORTING 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/+41793940965 at bluesip.net) State REPORTING send 383 bytes to udp/[78.138.90.58]:5060 at 23:09:15.087695: ?? ------------------------------------------------------------------------ ?? CANCEL sip:1002 at 78.138.90.58:5060 SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKapDgFeegKyHXK ?? Max-Forwards: 67 ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? CSeq: 28614398 CANCEL ?? Reason: FreeSWITCH;cause=487;text="ORIGINATOR_CANCEL" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 408 bytes from udp/[217.74.179.29]:5060 at 23:09:15.094670: ?? ------------------------------------------------------------------------ ?? ACK sip:gw+trunk_1000 at 78.138.90.58:6090;transport=udp;gw=trunk_1000 SIP/2.0 ?? Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 ?? From: "+41793940965" ;tag=as00589402 ?? Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net ?? To: ;tag=S7UZQygFt62Nm ?? CSeq: 102 ACK ?? User-Agent: Sip EXpress router(0.9.7 (i386/linux)) ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:79 sofia/internal/+41793940965 at bluesip.net Standard REPORTING, cause: ORIGINATOR_CANCEL 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/+41793940965 at bluesip.net) State REPORTING going to sleep 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:387 (sofia/internal/+41793940965 at bluesip.net) State Change CS_REPORTING -> CS_DESTROY 2012-05-24 23:09:15.120807 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:09:15.120807 [DEBUG] switch_core_session.c:1382 Session 8 (sofia/internal/+41793940965 at bluesip.net) Locked, Waiting on external entities 2012-05-24 23:09:15.120807 [NOTICE] switch_core_session.c:1400 Session 8 (sofia/internal/+41793940965 at bluesip.net) Ended 2012-05-24 23:09:15.120807 [NOTICE] switch_core_session.c:1402 Close Channel sofia/internal/+41793940965 at bluesip.net [CS_DESTROY] 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:491 (sofia/internal/+41793940965 at bluesip.net) Callstate Change HANGUP -> DOWN 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:494 (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_DESTROY 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/+41793940965 at bluesip.net) State DESTROY 2012-05-24 23:09:15.120807 [DEBUG] mod_sofia.c:374 sofia/internal/+41793940965 at bluesip.net SOFIA DESTROY 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:86 sofia/internal/+41793940965 at bluesip.net Standard DESTROY 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/+41793940965 at bluesip.net) State DESTROY going to sleep recv 378 bytes from udp/[78.138.90.58]:5060 at 23:09:15.135235: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 canceling ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKapDgFeegKyHXK ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ;tag=a6a1c5f60faecf035a1ae5b6e96e979a-8c5b ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? CSeq: 28614398 CANCEL ?? Server: kamailio (3.1.5 (x86_64/linux)) ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 446 bytes from udp/[78.138.90.58]:5060 at 23:09:16.419308: ?? ------------------------------------------------------------------------ ?? SIP/2.0 487 Request Terminated ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKapDgFeegKyHXK ?? Record-Route: ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ;tag=qCSaB2YJOAsAr2dJ-RWNMhPoDdn1RFvs ?? CSeq: 28614398 INVITE ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, REGISTER ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 354 bytes to udp/[78.138.90.58]:5060 at 23:09:16.419543: ?? ------------------------------------------------------------------------ ?? ACK sip:1002 at 78.138.90.58:5060 SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKapDgFeegKyHXK ?? Max-Forwards: 67 ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ;tag=qCSaB2YJOAsAr2dJ-RWNMhPoDdn1RFvs ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? CSeq: 28614398 ACK ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 564 bytes from udp/[41.34.123.243]:58881 at 23:13:13.612418: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjHQIpv.ACHGbXD4IHrMr72LwZ8kYdlu-J ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=wOP.FVT4Crk8O5E.bB7BWUhNO6QG91IG ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60926 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 744 bytes to udp/[41.34.123.243]:58881 at 23:13:13.613152: ?? ------------------------------------------------------------------------ ?? SIP/2.0 401 Unauthorized ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjHQIpv.ACHGbXD4IHrMr72LwZ8kYdlu-J;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=wOP.FVT4Crk8O5E.bB7BWUhNO6QG91IG ?? To: "Sherif 1003" ;tag=USeHUmjpmrFUB ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60926 REGISTER ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? WWW-Authenticate: Digest realm="sip.pcfone.com", nonce="09674764-a5f6-11e1-a83c-7b1f73a7ffcf", algorithm=MD5, qop="auth" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 830 bytes from udp/[41.34.123.243]:58881 at 23:13:13.847746: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjFr7rIAdAJPRQd2GmrMrVib3KT3Rup0q4 ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=wOP.FVT4Crk8O5E.bB7BWUhNO6QG91IG ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60927 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Authorization: Digest username="1002", realm="sip.pcfone.com", nonce="09674764-a5f6-11e1-a83c-7b1f73a7ffcf", uri="sip:sip.pcfone.com:6090", response="3bc0bb82a803ded3641704c4eb78d58f", algorithm=MD5, cnonce="sc3AJfhuev18Bs-L-6tORmI.RXlxnPOz", qop=auth, nc=00000001 ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 706 bytes to udp/[41.34.123.243]:58881 at 23:13:13.850197: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjFr7rIAdAJPRQd2GmrMrVib3KT3Rup0q4;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=wOP.FVT4Crk8O5E.bB7BWUhNO6QG91IG ?? To: "Sherif 1003" ;tag=v279vF3SH15DQ ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60927 REGISTER ?? Contact: ;expires=300 ?? Date: Thu, 24 May 2012 23:13:13 GMT ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ send 989 bytes to udp/[41.34.123.243]:58881 at 23:13:13.888717: ?? ------------------------------------------------------------------------ ?? NOTIFY sip:1002 at 192.168.178.26:58881;ob SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKBZ68g9yKg77FF ?? Route: ;ob ?? Max-Forwards: 70 ?? From: ;tag=XB12yamXeav0j ?? To: ?? Call-ID: e0efa252-2098-1230-8985-00163e6bb553 ?? CSeq: 28614532 NOTIFY ?? Contact: ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? 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=noresource ?? Content-Type: application/simple-message-summary ?? Content-Length: 91 ?? ?? Messages-Waiting: yes ?? Message-Account: sip:1002 at 78.138.90.58 ?? Voice-Message: 1/0 (0/0) ?? ?? ------------------------------------------------------------------------ recv 308 bytes from udp/[41.34.123.243]:58881 at 23:13:14.298850: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;received=78.138.90.58;branch=z9hG4bKBZ68g9yKg77FF ?? Call-ID: e0efa252-2098-1230-8985-00163e6bb553 ?? From: ;tag=XB12yamXeav0j ?? To: ;tag=z9hG4bKBZ68g9yKg77FF ?? CSeq: 28614532 NOTIFY ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ recv 564 bytes from udp/[41.34.123.243]:58881 at 23:18:09.067232: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjABTslmZIhiHYRtLqQn2erwmX19MVYHFC ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=JBSXL5WZLWbLzpvRFoTSP5x8qQslI1ST ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60928 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 744 bytes to udp/[41.34.123.243]:58881 at 23:18:09.068121: ?? ------------------------------------------------------------------------ ?? SIP/2.0 401 Unauthorized ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjABTslmZIhiHYRtLqQn2erwmX19MVYHFC;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=JBSXL5WZLWbLzpvRFoTSP5x8qQslI1ST ?? To: "Sherif 1003" ;tag=ymtU0540BKjKe ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60928 REGISTER ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? WWW-Authenticate: Digest realm="sip.pcfone.com", nonce="b9821df4-a5f6-11e1-a83d-7b1f73a7ffcf", algorithm=MD5, qop="auth" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 830 bytes from udp/[41.34.123.243]:58881 at 23:18:09.302944: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjRgtztcy63vMT97Jp5pH9n311li5EFeY- ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=JBSXL5WZLWbLzpvRFoTSP5x8qQslI1ST ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60929 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Authorization: Digest username="1002", realm="sip.pcfone.com", nonce="b9821df4-a5f6-11e1-a83d-7b1f73a7ffcf", uri="sip:sip.pcfone.com:6090", response="519c6301816b51f3bbb98d97a347e92b", algorithm=MD5, cnonce="sc3AJfhuev18Bs-L-6tORmI.RXlxnPOz", qop=auth, nc=00000001 ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 706 bytes to udp/[41.34.123.243]:58881 at 23:18:09.305217: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjRgtztcy63vMT97Jp5pH9n311li5EFeY-;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=JBSXL5WZLWbLzpvRFoTSP5x8qQslI1ST ?? To: "Sherif 1003" ;tag=ZXKm20N48U85S ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60929 REGISTER ?? Contact: ;expires=300 ?? Date: Thu, 24 May 2012 23:18:09 GMT ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ send 989 bytes to udp/[41.34.123.243]:58881 at 23:18:09.379987: ?? ------------------------------------------------------------------------ ?? NOTIFY sip:1002 at 192.168.178.26:58881;ob SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKc8Z1j4FQDgy2a ?? Route: ;ob ?? Max-Forwards: 70 ?? From: ;tag=06cD4U6754yrN ?? To: ?? Call-ID: 91100602-2099-1230-8985-00163e6bb553 ?? CSeq: 28614680 NOTIFY ?? Contact: ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? 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=noresource ?? Content-Type: application/simple-message-summary ?? Content-Length: 91 ?? ?? Messages-Waiting: yes ?? Message-Account: sip:1002 at 78.138.90.58 ?? Voice-Message: 1/0 (0/0) ?? ?? ------------------------------------------------------------------------ recv 308 bytes from udp/[41.34.123.243]:58881 at 23:18:09.773629: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;received=78.138.90.58;branch=z9hG4bKc8Z1j4FQDgy2a ?? Call-ID: 91100602-2099-1230-8985-00163e6bb553 ?? From: ;tag=06cD4U6754yrN ?? To: ;tag=z9hG4bKc8Z1j4FQDgy2a ?? CSeq: 28614680 NOTIFY ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ recv 564 bytes from udp/[41.34.123.243]:58881 at 23:23:04.522455: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjaCuKyaBi6ZqcsHyp5HhH3WthJStom0SK ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=qfUt1fQWuE-Ro3zWld6X0r6WidtD6B1q ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60930 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 744 bytes to udp/[41.34.123.243]:58881 at 23:23:04.523158: ?? ------------------------------------------------------------------------ ?? SIP/2.0 401 Unauthorized ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjaCuKyaBi6ZqcsHyp5HhH3WthJStom0SK;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=qfUt1fQWuE-Ro3zWld6X0r6WidtD6B1q ?? To: "Sherif 1003" ;tag=1F655pQB3DNBH ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60930 REGISTER ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? WWW-Authenticate: Digest realm="sip.pcfone.com", nonce="699cfe2a-a5f7-11e1-a83e-7b1f73a7ffcf", algorithm=MD5, qop="auth" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 830 bytes from udp/[41.34.123.243]:58881 at 23:23:04.757798: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjs8SAufeoyzCLrePaAbYhZuOqQF47XIc- ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=qfUt1fQWuE-Ro3zWld6X0r6WidtD6B1q ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60931 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Authorization: Digest username="1002", realm="sip.pcfone.com", nonce="699cfe2a-a5f7-11e1-a83e-7b1f73a7ffcf", uri="sip:sip.pcfone.com:6090", response="23a886934aaf049f08432b6bb226bd37", algorithm=MD5, cnonce="sc3AJfhuev18Bs-L-6tORmI.RXlxnPOz", qop=auth, nc=00000001 ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 706 bytes to udp/[41.34.123.243]:58881 at 23:23:04.759924: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjs8SAufeoyzCLrePaAbYhZuOqQF47XIc-;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=qfUt1fQWuE-Ro3zWld6X0r6WidtD6B1q ?? To: "Sherif 1003" ;tag=2rZy7H8e0pByc ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60931 REGISTER ?? Contact: ;expires=300 ?? Date: Thu, 24 May 2012 23:23:04 GMT ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ send 989 bytes to udp/[41.34.123.243]:58881 at 23:23:04.872728: ?? ------------------------------------------------------------------------ ?? NOTIFY sip:1002 at 192.168.178.26:58881;ob SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKDHStmZ0taSmNp ?? Route: ;ob ?? Max-Forwards: 70 ?? From: ;tag=31rQ9cSjXZ1gr ?? To: ?? Call-ID: 4130a1ce-209a-1230-8985-00163e6bb553 ?? CSeq: 28614828 NOTIFY ?? Contact: ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? 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=noresource ?? Content-Type: application/simple-message-summary ?? Content-Length: 91 ?? ?? Messages-Waiting: yes ?? Message-Account: sip:1002 at 78.138.90.58 ?? Voice-Message: 1/0 (0/0) ?? ?? ------------------------------------------------------------------------ recv 308 bytes from udp/[41.34.123.243]:58881 at 23:23:05.269063: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;received=78.138.90.58;branch=z9hG4bKDHStmZ0taSmNp ?? Call-ID: 4130a1ce-209a-1230-8985-00163e6bb553 ?? From: ;tag=31rQ9cSjXZ1gr ?? To: ;tag=z9hG4bKDHStmZ0taSmNp ?? CSeq: 28614828 NOTIFY ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 916 bytes to udp/[217.74.179.29]:5060 at 23:25:45.873423: ?? ------------------------------------------------------------------------ ?? REGISTER sip:bluesip.net;transport=udp SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKetjKptHy71a8H ?? Max-Forwards: 70 ?? From: ;tag=6r0vBQZS650Fg ?? To: ?? Call-ID: 2a3e4696-a5b9-11e1-a7ed-7b1f73a7ffcf ?? CSeq: 28601493 REGISTER ?? Contact: ?? Expires: 3600 ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? Authorization: Digest username="bluesip/sherifomran", realm="bluesip.net", nonce="4fbeba4ebfb7414d5c701bd6f994402d551f3acf", algorithm=MD5, uri="sip:bluesip.net;transport=udp", response="c6cdcafe0418e519bc9ee0d8fa3d4d74" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 722 bytes from udp/[217.74.179.29]:5060 at 23:25:45.904176: ?? ------------------------------------------------------------------------ ?? SIP/2.0 401 Unauthorized ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKetjKptHy71a8H ?? From: ;tag=6r0vBQZS650Fg ?? To: ;tag=0354a2e1b960c9cc2279eca4e5f84e20.1b16 ?? Call-ID: 2a3e4696-a5b9-11e1-a7ed-7b1f73a7ffcf ?? CSeq: 28601493 REGISTER ?? WWW-Authenticate: Digest realm="bluesip.net", nonce="4fbec4a4ba82d1f1feaa32ccec01b85be054fcf7", stale=true ?? Server: Sip EXpress router (0.9.7 (i386/linux)) ?? Content-Length: 0 ?? Warning: 392 217.74.179.29:5060 "Noisy feedback tells:? pid=5455 req_src_ip=78.138.90.58 req_src_port=6090 in_uri=sip:bluesip.net;transport=udp out_uri=sip:bluesip.net;transport=udp via_cnt==1" ?? ?? ------------------------------------------------------------------------ send 916 bytes to udp/[217.74.179.29]:5060 at 23:25:45.904416: ?? ------------------------------------------------------------------------ ?? REGISTER sip:bluesip.net;transport=udp SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKF3BcrN214a1tD ?? Max-Forwards: 70 ?? From: ;tag=6r0vBQZS650Fg ?? To: ?? Call-ID: 2a3e4696-a5b9-11e1-a7ed-7b1f73a7ffcf ?? CSeq: 28601494 REGISTER ?? Contact: ?? Expires: 3600 ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? Authorization: Digest username="bluesip/sherifomran", realm="bluesip.net", nonce="4fbec4a4ba82d1f1feaa32ccec01b85be054fcf7", algorithm=MD5, uri="sip:bluesip.net;transport=udp", response="4c09dbe4b9accb52d4104b40dfe20040" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 699 bytes from udp/[217.74.179.29]:5060 at 23:25:45.935046: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKF3BcrN214a1tD ?? From: ;tag=6r0vBQZS650Fg ?? To: ;tag=0354a2e1b960c9cc2279eca4e5f84e20.2017 ?? Call-ID: 2a3e4696-a5b9-11e1-a7ed-7b1f73a7ffcf ?? CSeq: 28601494 REGISTER ?? Contact: ;q=0.5;expires=3600 ?? Server: Sip EXpress router (0.9.7 (i386/linux)) ?? Content-Length: 0 ?? Warning: 392 217.74.179.29:5060 "Noisy feedback tells:? pid=5462 req_src_ip=78.138.90.58 req_src_port=6090 in_uri=sip:bluesip.net;transport=udp out_uri=sip:bluesip.net;transport=udp via_cnt==1" ?? ?? ------------------------------------------------------------------------ recv 564 bytes from udp/[41.34.123.243]:58881 at 23:27:59.977057: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjEOP.a4NoXqoFqC7a.k2tyeX-mYlUyH4H ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60932 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 744 bytes to udp/[41.34.123.243]:58881 at 23:27:59.977960: ?? ------------------------------------------------------------------------ ?? SIP/2.0 401 Unauthorized ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjEOP.a4NoXqoFqC7a.k2tyeX-mYlUyH4H;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K ?? To: "Sherif 1003" ;tag=4ajgB89Nt8Q3K ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60932 REGISTER ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? WWW-Authenticate: Digest realm="sip.pcfone.com", nonce="19b7cc40-a5f8-11e1-a83f-7b1f73a7ffcf", algorithm=MD5, qop="auth" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 564 bytes from udp/[41.34.123.243]:58881 at 23:28:00.468487: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjEOP.a4NoXqoFqC7a.k2tyeX-mYlUyH4H ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60932 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 744 bytes to udp/[41.34.123.243]:58881 at 23:28:00.468679: ?? ------------------------------------------------------------------------ ?? SIP/2.0 401 Unauthorized ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjEOP.a4NoXqoFqC7a.k2tyeX-mYlUyH4H;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K ?? To: "Sherif 1003" ;tag=4ajgB89Nt8Q3K ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60932 REGISTER ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? WWW-Authenticate: Digest realm="sip.pcfone.com", nonce="19b7cc40-a5f8-11e1-a83f-7b1f73a7ffcf", algorithm=MD5, qop="auth" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 830 bytes from udp/[41.34.123.243]:58881 at 23:28:00.717446: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPj0iHd3-XUjnvcBs9DiJUCWzrmYTmWxFLt ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60933 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Authorization: Digest username="1002", realm="sip.pcfone.com", nonce="19b7cc40-a5f8-11e1-a83f-7b1f73a7ffcf", uri="sip:sip.pcfone.com:6090", response="e21c903e6d98b5adcdd2eefa75217157", algorithm=MD5, cnonce="sc3AJfhuev18Bs-L-6tORmI.RXlxnPOz", qop=auth, nc=00000001 ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 706 bytes to udp/[41.34.123.243]:58881 at 23:28:00.719947: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPj0iHd3-XUjnvcBs9DiJUCWzrmYTmWxFLt;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K ?? To: "Sherif 1003" ;tag=5KB9c3tSQHepF ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60933 REGISTER ?? Contact: ;expires=300 ?? Date: Thu, 24 May 2012 23:28:00 GMT ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ send 989 bytes to udp/[41.34.123.243]:58881 at 23:28:00.868368: ?? ------------------------------------------------------------------------ ?? NOTIFY sip:1002 at 192.168.178.26:58881;ob SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKgc54SgK51KQDS ?? Route: ;ob ?? Max-Forwards: 70 ?? From: ;tag=6v41eyBXmt48a ?? To: ?? Call-ID: f19dfbce-209a-1230-8985-00163e6bb553 ?? CSeq: 28614976 NOTIFY ?? Contact: ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? 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=noresource ?? Content-Type: application/simple-message-summary ?? Content-Length: 91 ?? ?? Messages-Waiting: yes ?? Message-Account: sip:1002 at 78.138.90.58 ?? Voice-Message: 1/0 (0/0) ?? ?? ------------------------------------------------------------------------ recv 308 bytes from udp/[41.34.123.243]:58881 at 23:28:01.128279: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;received=78.138.90.58;branch=z9hG4bKgc54SgK51KQDS ?? Call-ID: f19dfbce-209a-1230-8985-00163e6bb553 ?? From: ;tag=6v41eyBXmt48a ?? To: ;tag=z9hG4bKgc54SgK51KQDS ?? CSeq: 28614976 NOTIFY ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 913 bytes to udp/[217.74.179.29]:5060 at 23:28:11.885213: ?? ------------------------------------------------------------------------ ?? REGISTER sip:bluesip.net;transport=udp SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKHNyXUB48yvD0m ?? Max-Forwards: 70 ?? From: ;tag=5F739Uep9vaXm ?? To: ?? Call-ID: 2a3e49c0-a5b9-11e1-a7ee-7b1f73a7ffcf ?? CSeq: 28601493 REGISTER ?? Contact: ?? Expires: 3600 ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? Authorization: Digest username="bluesip/salahomran", realm="bluesip.net", nonce="4fbebeca79a5cff7c417463814cb9054d313bcac", algorithm=MD5, uri="sip:bluesip.net;transport=udp", response="bb1babb90f4ea1dc8fbb9aa45d6038c7" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 720 bytes from udp/[217.74.179.29]:5060 at 23:28:11.915764: ?? ------------------------------------------------------------------------ ?? SIP/2.0 401 Unauthorized ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKHNyXUB48yvD0m ?? From: ;tag=5F739Uep9vaXm ?? To: ;tag=0354a2e1b960c9cc2279eca4e5f84e20.3454 ?? Call-ID: 2a3e49c0-a5b9-11e1-a7ee-7b1f73a7ffcf ?? CSeq: 28601493 REGISTER ?? WWW-Authenticate: Digest realm="bluesip.net", nonce="4fbec5365af3a174f1cc0b8cda770c727d25bf71", stale=true ?? Server: Sip EXpress router (0.9.7 (i386/linux)) ?? Content-Length: 0 ?? Warning: 392 217.74.179.29:5060 "Noisy feedback tells:? pid=5462 req_src_ip=78.138.90.58 req_src_port=6090 in_uri=sip:bluesip.net;transport=udp out_uri=sip:bluesip.net;transport=udp via_cnt==1" ?? ?? ------------------------------------------------------------------------ send 913 bytes to udp/[217.74.179.29]:5060 at 23:28:11.915914: ?? ------------------------------------------------------------------------ ?? REGISTER sip:bluesip.net;transport=udp SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKjyQpX6mcv53jg ?? Max-Forwards: 70 ?? From: ;tag=5F739Uep9vaXm ?? To: ?? Call-ID: 2a3e49c0-a5b9-11e1-a7ee-7b1f73a7ffcf ?? CSeq: 28601494 REGISTER ?? Contact: ?? Expires: 3600 ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? Authorization: Digest username="bluesip/salahomran", realm="bluesip.net", nonce="4fbec5365af3a174f1cc0b8cda770c727d25bf71", algorithm=MD5, uri="sip:bluesip.net;transport=udp", response="e260f6dc9b01ce8bea8c5ffb36bd4ddd" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 697 bytes from udp/[217.74.179.29]:5060 at 23:28:11.946325: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKjyQpX6mcv53jg ?? From: ;tag=5F739Uep9vaXm ?? To: ;tag=0354a2e1b960c9cc2279eca4e5f84e20.4b5c ?? Call-ID: 2a3e49c0-a5b9-11e1-a7ee-7b1f73a7ffcf ?? CSeq: 28601494 REGISTER ?? Contact: ;q=0.5;expires=3600 ?? Server: Sip EXpress router (0.9.7 (i386/linux)) ?? Content-Length: 0 ?? Warning: 392 217.74.179.29:5060 "Noisy feedback tells:? pid=5455 req_src_ip=78.138.90.58 req_src_port=6090 in_uri=sip:bluesip.net;transport=udp out_uri=sip:bluesip.net;transport=udp via_cnt==1" ?? ?? ------------------------------------------------------------------------ recv 564 bytes from udp/[41.34.123.243]:58881 at 23:32:55.930149: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjuk8DZGiaXfN7BmtslNHVjtSg5UXsbHXc ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=Pz.t-0tGXj3saM.ehBiB1EAMmmitze1D ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60934 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 744 bytes to udp/[41.34.123.243]:58881 at 23:32:55.931324: ?? ------------------------------------------------------------------------ ?? SIP/2.0 401 Unauthorized ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjuk8DZGiaXfN7BmtslNHVjtSg5UXsbHXc;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=Pz.t-0tGXj3saM.ehBiB1EAMmmitze1D ?? To: "Sherif 1003" ;tag=75XtgSv0H3tUp ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60934 REGISTER ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? WWW-Authenticate: Digest realm="sip.pcfone.com", nonce="ca1eabee-a5f8-11e1-a840-7b1f73a7ffcf", algorithm=MD5, qop="auth" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 830 bytes from udp/[41.34.123.243]:58881 at 23:32:56.165624: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjKEQ6xXmyzZxA6vCM88vHNYAOPAjUqFsD ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=Pz.t-0tGXj3saM.ehBiB1EAMmmitze1D ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60935 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Authorization: Digest username="1002", realm="sip.pcfone.com", nonce="ca1eabee-a5f8-11e1-a840-7b1f73a7ffcf", uri="sip:sip.pcfone.com:6090", response="ff6d12f4ec7abd4b37187222b3af2358", algorithm=MD5, cnonce="sc3AJfhuev18Bs-L-6tORmI.RXlxnPOz", qop=auth, nc=00000001 ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 706 bytes to udp/[41.34.123.243]:58881 at 23:32:56.168033: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjKEQ6xXmyzZxA6vCM88vHNYAOPAjUqFsD;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=Pz.t-0tGXj3saM.ehBiB1EAMmmitze1D ?? To: "Sherif 1003" ;tag=8eQKjmD4ecHej ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60935 REGISTER ?? Contact: ;expires=300 ?? Date: Thu, 24 May 2012 23:32:56 GMT ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ send 989 bytes to udp/[41.34.123.243]:58881 at 23:32:56.360238: ?? ------------------------------------------------------------------------ ?? NOTIFY sip:1002 at 192.168.178.26:58881;ob SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKK7gFZ15FSet5B ?? Route: ;ob ?? Max-Forwards: 70 ?? From: ;tag=9QgcmFy7BN70D ?? To: ?? Call-ID: a1be7708-209b-1230-8985-00163e6bb553 ?? CSeq: 28615124 NOTIFY ?? Contact: ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? 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=noresource ?? Content-Type: application/simple-message-summary ?? Content-Length: 91 ?? ?? Messages-Waiting: yes ?? Message-Account: sip:1002 at 78.138.90.58 ?? Voice-Message: 1/0 (0/0) ?? ?? ------------------------------------------------------------------------ recv 308 bytes from udp/[41.34.123.243]:58881 at 23:32:56.623554: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;received=78.138.90.58;branch=z9hG4bKK7gFZ15FSet5B ?? Call-ID: a1be7708-209b-1230-8985-00163e6bb553 ?? From: ;tag=9QgcmFy7BN70D ?? To: ;tag=z9hG4bKK7gFZ15FSet5B ?? CSeq: 28615124 NOTIFY ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ freeswitch at internal> --- On Thu, 5/24/12, Kristian Kielhofner wrote: From: Kristian Kielhofner Subject: Re: [Freeswitch-users] Freeswitch (SBC + Media) does not know that gateway phone is picked up To: "FreeSWITCH Users Help" Date: Thursday, May 24, 2012, 7:51 PM Siptrace and logs please. On Thu, May 24, 2012 at 11:53 AM, Sherif Omran wrote: > > Hi all, > > My topology is as follows: > > Kamailio? -> FS (SBS+Media server) > > I came across an issue with my system as follows.? I have a Hardphone registered. When I do local call inside kamailio, it gets to FS and returns back well and FS understands when I lift the handset. However, I added a gateway (german landline server), when I call my self from another phone, the call gets to FS and then transmits to Kamailio, it rings my extension but when I lift the handset FS does not notice it and keeps ringing. > > Any body has an Idea? Here is my gateway trunk. > > > ??????? > ??????? > ??????? > ??????? > ??????? > ????? > > > thanks in advance > Sherif Omran > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Kristian Kielhofner _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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 at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -----Inline Attachment Follows----- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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 at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -----Inline Attachment Follows----- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -----Inline Attachment Follows----- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120525/22b607e5/attachment-0001.html From sherifomran2000 at yahoo.com Sat May 26 00:05:59 2012 From: sherifomran2000 at yahoo.com (Sherif Omran) Date: Fri, 25 May 2012 13:05:59 -0700 (PDT) Subject: [Freeswitch-users] Freeswitch (SBC + Media) does not know that gateway phone is picked up In-Reply-To: Message-ID: <1337976359.52140.YahooMailClassic@web110804.mail.gq1.yahoo.com> Hello Sammy FS uses port 6090 and registers the GW. When a call comes, it rings the extension but then gives a busy signal and FS keeps ringing till I cancel the call from the GW. recv 477 bytes from udp/[217.74.179.29]:5060 at 19:52:05.662344:? ?------------------------------------------------------------------------? ?CANCEL sip:gw+trunk_1000 at 78.138.90.58:6090;transport=udp;gw=trunk_1000 SIP/2.0? ?Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bKe567.13b0b9e.0? ?Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK16d1a257;rport=5060? ?From: "+41793940965" ;tag=as1dccab06? ?To: ? ?Call-ID: 5cd37edb776a6b3a35e9713a453a3425 at bluesip.net? ?CSeq: 102 CANCEL? ?User-Agent: blueSIP PSTN GW? ?Max-Forwards: 69? ?Content-Length: 0? ?P-hint: USRLOC --- On Fri, 5/25/12, SamyGo wrote: From: SamyGo Subject: Re: [Freeswitch-users] Freeswitch (SBC + Media) does not know that gateway phone is picked up To: "FreeSWITCH Users Help" Date: Friday, May 25, 2012, 10:46 AM Hi again,If you want kamailio register to the provider then use UAC module. Kamailio will use the username/password and register with the provider. Regards,Sammy On Fri, May 25, 2012 at 12:20 PM, Sherif Omran wrote: Dear Sammy, Thank you for your question ... Yes, the GW is registered through FS because I did not know how to register it to kamailio. But it seems better to register it to kamailio. One more information, calls from 1001 to 1002 go to kamailio then to FS then return back to kamailio smoothly. Thus I would suggest that I change registering the gateway from FS to kamailio. but How to? --- On Fri, 5/25/12, SamyGo wrote: From: SamyGo Subject: Re: [Freeswitch-users] Freeswitch (SBC + Media) does not know that gateway phone is picked up To: "FreeSWITCH Users Help" Date: Friday, May 25, 2012, 8:08 AM Hi,These are SIP traces on FreeSWITCH console, whereas you are saying and it seems that kamailio is not detecting the answering (200 OK)of the call from extension 1002. Please, can you take a sip trace..!! I see you've both kamailio and FS on same server! Please take a pcap from the linux console using the following command. #tcpdump -i any -s 0 -w new-call-trace.pcap -vvv Please be quick on turning the sip trace on and off as quickly as possible to avoid extra packets. Once done open the file in wireshark ;?apply filter "sip || rtp" and then save the resulting capture in separate file. Send us the new file to analyse. One more silly question probably, I see REGISTERs coming to your FS as well and the calls to gateways are made from FS too !!, umm...just thinking what are you using kamailio for!!? ThanksBR,Sammy On Fri, May 25, 2012 at 4:40 AM, Sherif Omran wrote: Hi all, here is the siptrace: To figure it out 1- gateway called bluesip.net. It send invide using caller number at bluesip.net 2- This call should go to extension kb-1002. kb means go from freeswitch port 6090 to kamailio port 5060 3- It should go to call extension 1002 in Kamailio 4- Extension 1002 rings but when I reply, it does not notice I replied ./fs_cli ??????????? _____ ____???? ____ _???? ___????????????? ?????????? |? ___/ ___|?? / ___| |?? |_ _|???????????? ?????????? | |_? \___ \? | |?? | |??? | |??????????? ?????????? |? _|? ___) | | |___| |___ | |????????????? ?????????? |_|?? |____/?? \____|_____|___|??????????? ******************************************************* * Anthony Minessale II, Ken Rice,???????????????????? * * Michael Jerris, Travis Cross??????????????????????? * * FreeSWITCH (http://www.freeswitch.org)????????????? * * Paypal Donations Appreciated: paypal at freeswitch.org * * Brought to you by ClueCon http://www.cluecon.com/?? * ******************************************************* Type /help to see a list of commands +OK log level? [7] freeswitch at internal> tracelevel -ERR tracelevel Command not found! freeswitch at internal> sofia global siptrace on +OK Global siptrace on recv 1104 bytes from udp/[217.74.179.29]:5060 at 23:08:45.822962: ?? ------------------------------------------------------------------------ ?? INVITE sip:gw+trunk_1000 at 78.138.90.58:6090;transport=udp;gw=trunk_1000 SIP/2.0 ?? Record-Route: ?? Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 ?? Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 ?? From: "+41793940965" ;tag=as00589402 ?? To: ?? Contact: ?? Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net ?? CSeq: 102 INVITE ?? User-Agent: blueSIP PSTN GW ?? Max-Forwards: 69 ?? Date: Thu, 24 May 2012 23:08:44 GMT ?? Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY ?? Supported: replaces ?? Content-Type: application/sdp ?? Content-Length: 367 ?? P-hint: USRLOC ?? ?? v=0 ?? o=root 20076 20076 IN IP4 217.74.179.28 ?? s=session ?? c=IN IP4 217.74.179.28 ?? t=0 0 ?? m=audio 25626 RTP/AVP 8 0 18 111 3 101 ?? a=rtpmap:8 PCMA/8000 ?? a=rtpmap:0 PCMU/8000 ?? a=rtpmap:18 G729/8000 ?? a=fmtp:18 annexb=no ?? a=rtpmap:111 G726-32/8000 ?? a=rtpmap:3 GSM/8000 ?? a=rtpmap:101 telephone-event/8000 ?? a=fmtp:101 0-16 ?? a=silenceSupp:off - - - - ?? a=ptime:20 ?? a=sendrecv ?? ------------------------------------------------------------------------ send 494 bytes to udp/[217.74.179.29]:5060 at 23:08:45.844486: ?? ------------------------------------------------------------------------ ?? SIP/2.0 100 Trying ?? Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 ?? Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 ?? Record-Route: ?? From: "+41793940965" ;tag=as00589402 ?? To: ?? Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net ?? CSeq: 102 INVITE ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ 2012-05-24 23:08:45.840749 [NOTICE] switch_channel.c:926 New Channel sofia/internal/+41793940965 at bluesip.net [69cd22e6-a5f5-11e1-a833-7b1f73a7ffcf] 2012-05-24 23:08:45.840749 [DEBUG] sofia.c:5532 Channel sofia/internal/+41793940965 at bluesip.net entering state [received][100] 2012-05-24 23:08:45.840749 [DEBUG] sofia.c:5543 Remote SDP: v=0 o=root 20076 20076 IN IP4 217.74.179.28 s=session c=IN IP4 217.74.179.28 t=0 0 m=audio 25626 RTP/AVP 8 0 18 111 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:111 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMA:8:8000:20:64000]/[iLBC:97:8000:30:13330] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMU:0:8000:20:64000]/[iLBC:97:8000:30:13330] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G729:18:8000:20:8000]/[iLBC:97:8000:30:13330] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G726-32:111:8000:20:0]/[iLBC:97:8000:30:13330] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [GSM:3:8000:20:13200]/[iLBC:97:8000:30:13330] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [telephone-event:101:8000:20:0]/[iLBC:97:8000:30:13330] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf send/recv payload to 101 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMA:8:8000:20:64000]/[G7221:115:32000:20:48000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMU:0:8000:20:64000]/[G7221:115:32000:20:48000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G729:18:8000:20:8000]/[G7221:115:32000:20:48000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G726-32:111:8000:20:0]/[G7221:115:32000:20:48000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [GSM:3:8000:20:13200]/[G7221:115:32000:20:48000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [telephone-event:101:8000:20:0]/[G7221:115:32000:20:48000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf send/recv payload to 101 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMA:8:8000:20:64000]/[G7221:107:16000:20:32000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMU:0:8000:20:64000]/[G7221:107:16000:20:32000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G729:18:8000:20:8000]/[G7221:107:16000:20:32000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G726-32:111:8000:20:0]/[G7221:107:16000:20:32000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [GSM:3:8000:20:13200]/[G7221:107:16000:20:32000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [telephone-event:101:8000:20:0]/[G7221:107:16000:20:32000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf send/recv payload to 101 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMA:8:8000:20:64000]/[G722:9:8000:20:64000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G729:18:8000:20:8000]/[G722:9:8000:20:64000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G726-32:111:8000:20:0]/[G722:9:8000:20:64000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [GSM:3:8000:20:13200]/[G722:9:8000:20:64000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [telephone-event:101:8000:20:0]/[G722:9:8000:20:64000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf send/recv payload to 101 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMA:8:8000:20:64000]/[PCMU:0:8000:20:64000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:2996 Set Codec sofia/internal/+41793940965 at bluesip.net PCMU/8000 20 ms 160 samples 64000 bits 2012-05-24 23:08:45.840749 [DEBUG] switch_core_codec.c:111 sofia/internal/+41793940965 at bluesip.net Original read codec set to PCMU:0 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf send/recv payload to 101 2012-05-24 23:08:45.840749 [DEBUG] sofia.c:5757 (sofia/internal/+41793940965 at bluesip.net) State Change CS_NEW -> CS_INIT 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_INIT 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/+41793940965 at bluesip.net) State INIT 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:85 sofia/internal/+41793940965 at bluesip.net SOFIA INIT 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:125 (sofia/internal/+41793940965 at bluesip.net) State Change CS_INIT -> CS_ROUTING 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/+41793940965 at bluesip.net) State INIT going to sleep 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_ROUTING 2012-05-24 23:08:45.840749 [DEBUG] switch_channel.c:1886 (sofia/internal/+41793940965 at bluesip.net) Callstate Change DOWN -> RINGING 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/+41793940965 at bluesip.net) State ROUTING 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:148 sofia/internal/+41793940965 at bluesip.net SOFIA ROUTING 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:104 sofia/internal/+41793940965 at bluesip.net Standard ROUTING 2012-05-24 23:08:45.840749 [INFO] mod_dialplan_xml.c:485 Processing +41793940965 <+41793940965>->kb-1002 in context public Dialplan: sofia/internal/+41793940965 at bluesip.net parsing [public->from_kamailio] continue=false Dialplan: sofia/internal/+41793940965 at bluesip.net Regex (PASS) [from_kamailio] destination_number(kb-1002) =~ /^(.+)$/ break=on-false Dialplan: sofia/internal/+41793940965 at bluesip.net Action transfer(kb-1002 XML default) 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:154 (sofia/internal/+41793940965 at bluesip.net) State Change CS_ROUTING -> CS_EXECUTE 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/+41793940965 at bluesip.net) State ROUTING going to sleep 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_EXECUTE 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/+41793940965 at bluesip.net) State EXECUTE 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:241 sofia/internal/+41793940965 at bluesip.net SOFIA EXECUTE 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:192 sofia/internal/+41793940965 at bluesip.net Standard EXECUTE EXECUTE sofia/internal/+41793940965 at bluesip.net transfer(kb-1002 XML default) 2012-05-24 23:08:45.840749 [DEBUG] switch_ivr.c:1711 (sofia/internal/+41793940965 at bluesip.net) State Change CS_EXECUTE -> CS_ROUTING 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:731 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:08:45.840749 [NOTICE] switch_ivr.c:1717 Transfer sofia/internal/+41793940965 at bluesip.net to XML[kb-1002 at default] 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/+41793940965 at bluesip.net) State EXECUTE going to sleep 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_ROUTING 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/+41793940965 at bluesip.net) State ROUTING 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:148 sofia/internal/+41793940965 at bluesip.net SOFIA ROUTING 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:104 sofia/internal/+41793940965 at bluesip.net Standard ROUTING 2012-05-24 23:08:45.840749 [INFO] mod_dialplan_xml.c:485 Processing +41793940965 <+41793940965>->kb-1002 in context default Dialplan: sofia/internal/+41793940965 at bluesip.net parsing [default->vbox] continue=false Dialplan: sofia/internal/+41793940965 at bluesip.net Regex (FAIL) [vbox] destination_number(kb-1002) =~ /^vb-([0-9][0-9][0-9])$/ break=on-false Dialplan: sofia/internal/+41793940965 at bluesip.net parsing [default->vmenu] continue=false Dialplan: sofia/internal/+41793940965 at bluesip.net Regex (FAIL) [vmenu] destination_number(kb-1002) =~ /^vm-([0-9][0-9][0-9][0-9])$/ break=on-false Dialplan: sofia/internal/+41793940965 at bluesip.net parsing [default->kbridge] continue=false Dialplan: sofia/internal/+41793940965 at bluesip.net Regex (PASS) [kbridge] destination_number(kb-1002) =~ /^kb-(.+)$/ break=on-false Dialplan: sofia/internal/+41793940965 at bluesip.net Action set(proxy_media=true) Dialplan: sofia/internal/+41793940965 at bluesip.net Action set(call_timeout=50) Dialplan: sofia/internal/+41793940965 at bluesip.net Action set(continue_on_fail=true) Dialplan: sofia/internal/+41793940965 at bluesip.net Action set(hangup_after_bridge=true) Dialplan: sofia/internal/+41793940965 at bluesip.net Action set(sip_invite_domain=78.138.90.58) Dialplan: sofia/internal/+41793940965 at bluesip.net Action export(sip_contact_user=ufs) Dialplan: sofia/internal/+41793940965 at bluesip.net Action bridge(sofia/78.138.90.58/1002 at 78.138.90.58:5060) Dialplan: sofia/internal/+41793940965 at bluesip.net Action answer() Dialplan: sofia/internal/+41793940965 at bluesip.net Action voicemail(default ${domain_name} 1002) 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:154 (sofia/internal/+41793940965 at bluesip.net) State Change CS_ROUTING -> CS_EXECUTE 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/+41793940965 at bluesip.net) State ROUTING going to sleep 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_EXECUTE 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/+41793940965 at bluesip.net) State EXECUTE 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:241 sofia/internal/+41793940965 at bluesip.net SOFIA EXECUTE 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:192 sofia/internal/+41793940965 at bluesip.net Standard EXECUTE EXECUTE sofia/internal/+41793940965 at bluesip.net set(proxy_media=true) 2012-05-24 23:08:45.840749 [DEBUG] mod_dptools.c:1281 sofia/internal/+41793940965 at bluesip.net SET [proxy_media]=[true] EXECUTE sofia/internal/+41793940965 at bluesip.net set(call_timeout=50) 2012-05-24 23:08:45.840749 [DEBUG] mod_dptools.c:1281 sofia/internal/+41793940965 at bluesip.net SET [call_timeout]=[50] EXECUTE sofia/internal/+41793940965 at bluesip.net set(continue_on_fail=true) 2012-05-24 23:08:45.840749 [DEBUG] mod_dptools.c:1281 sofia/internal/+41793940965 at bluesip.net SET [continue_on_fail]=[true] EXECUTE sofia/internal/+41793940965 at bluesip.net set(hangup_after_bridge=true) 2012-05-24 23:08:45.840749 [DEBUG] mod_dptools.c:1281 sofia/internal/+41793940965 at bluesip.net SET [hangup_after_bridge]=[true] EXECUTE sofia/internal/+41793940965 at bluesip.net set(sip_invite_domain=78.138.90.58) 2012-05-24 23:08:45.840749 [DEBUG] mod_dptools.c:1281 sofia/internal/+41793940965 at bluesip.net SET [sip_invite_domain]=[78.138.90.58] EXECUTE sofia/internal/+41793940965 at bluesip.net export(sip_contact_user=ufs) 2012-05-24 23:08:45.840749 [DEBUG] switch_channel.c:1093 EXPORT (export_vars) [sip_contact_user]=[ufs] EXECUTE sofia/internal/+41793940965 at bluesip.net bridge(sofia/78.138.90.58/1002 at 78.138.90.58:5060) 2012-05-24 23:08:45.840749 [DEBUG] switch_channel.c:1047 sofia/internal/+41793940965 at bluesip.net EXPORTING[export_vars] [sip_contact_user]=[ufs] to event 2012-05-24 23:08:45.840749 [DEBUG] switch_ivr_originate.c:1884 Parsing global variables 2012-05-24 23:08:45.840749 [NOTICE] switch_channel.c:926 New Channel sofia/internal/1002 at 78.138.90.58:5060 [69ce1f0c-a5f5-11e1-a838-7b1f73a7ffcf] 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:4691 (sofia/internal/1002 at 78.138.90.58:5060) State Change CS_NEW -> CS_INIT 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1002 at 78.138.90.58:5060) Running State Change CS_INIT 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/1002 at 78.138.90.58:5060) State INIT 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:85 sofia/internal/1002 at 78.138.90.58:5060 SOFIA INIT 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:1871 sofia/internal/1002 at 78.138.90.58:5060 Patched SDP --- v=0 o=root 20076 20076 IN IP4 217.74.179.28 s=session c=IN IP4 217.74.179.28 t=0 0 m=audio 25626 RTP/AVP 8 0 18 111 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:111 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 +++ v=0 o=FreeSWITCH 1629510322 1629510323 IN IP4 78.138.90.58 s=FreeSWITCH c=IN IP4 78.138.90.58 t=0 0 m=audio 31178 RTP/AVP 8 0 18 111 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:111 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 send 1340 bytes to udp/[78.138.90.58]:5060 at 23:08:45.852280: ?? ------------------------------------------------------------------------ ?? INVITE sip:1002 at 78.138.90.58:5060 SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKapDgFeegKyHXK ?? Max-Forwards: 67 ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? CSeq: 28614398 INVITE ?? Contact: ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? 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: 372 ?? P-hint: USRLOC ?? X-FS-Support: update_display,send_info ?? Remote-Party-ID: "+41793940965" ;party=calling;screen=yes;privacy=off ?? ?? v=0 ?? o=FreeSWITCH 1629510322 1629510323 IN IP4 78.138.90.58 ?? s=FreeSWITCH ?? c=IN IP4 78.138.90.58 ?? t=0 0 ?? m=audio 31178 RTP/AVP 8 0 18 111 3 101 ?? a=rtpmap:8 PCMA/8000 ?? a=rtpmap:0 PCMU/8000 ?? a=rtpmap:18 G729/8000 ?? a=fmtp:18 annexb=no ?? a=rtpmap:111 G726-32/8000 ?? a=rtpmap:3 GSM/8000 ?? a=rtpmap:101 telephone-event/8000 ?? a=fmtp:101 0-16 ?? a=silenceSupp:off - - - - ?? a=ptime:20 ?? ------------------------------------------------------------------------ 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:125 (sofia/internal/1002 at 78.138.90.58:5060) State Change CS_INIT -> CS_ROUTING 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/1002 at 78.138.90.58:5060) State INIT going to sleep 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1002 at 78.138.90.58:5060) Running State Change CS_ROUTING 2012-05-24 23:08:45.840749 [DEBUG] switch_channel.c:1886 (sofia/internal/1002 at 78.138.90.58:5060) Callstate Change DOWN -> RINGING 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/1002 at 78.138.90.58:5060) State ROUTING 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:148 sofia/internal/1002 at 78.138.90.58:5060 SOFIA ROUTING 2012-05-24 23:08:45.840749 [DEBUG] switch_ivr_originate.c:66 (sofia/internal/1002 at 78.138.90.58:5060) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/1002 at 78.138.90.58:5060) State ROUTING going to sleep 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1002 at 78.138.90.58:5060) Running State Change CS_CONSUME_MEDIA 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:429 (sofia/internal/1002 at 78.138.90.58:5060) State CONSUME_MEDIA 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:429 (sofia/internal/1002 at 78.138.90.58:5060) State CONSUME_MEDIA going to sleep 2012-05-24 23:08:45.840749 [DEBUG] sofia.c:5532 Channel sofia/internal/1002 at 78.138.90.58:5060 entering state [calling][0] recv 365 bytes from udp/[78.138.90.58]:5060 at 23:08:45.964309: ?? ------------------------------------------------------------------------ ?? SIP/2.0 100 trying -- your call is important to us ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKapDgFeegKyHXK ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? CSeq: 28614398 INVITE ?? Server: kamailio (3.1.5 (x86_64/linux)) ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 496 bytes from udp/[78.138.90.58]:5060 at 23:08:47.414766: ?? ------------------------------------------------------------------------ ?? SIP/2.0 180 Ringing ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKapDgFeegKyHXK ?? Record-Route: ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ;tag=qCSaB2YJOAsAr2dJ-RWNMhPoDdn1RFvs ?? CSeq: 28614398 INVITE ?? Contact: "Mama" ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, REGISTER ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ 2012-05-24 23:08:47.400741 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:47.400741 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:47.400741 [DEBUG] sofia.c:5532 Channel sofia/internal/1002 at 78.138.90.58:5060 entering state [proceeding][180] 2012-05-24 23:08:47.400741 [NOTICE] sofia.c:5624 Ring-Ready sofia/internal/1002 at 78.138.90.58:5060! send 995 bytes to udp/[217.74.179.29]:5060 at 23:08:47.416538: ?? ------------------------------------------------------------------------ ?? SIP/2.0 180 Ringing ?? Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 ?? Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 ?? Record-Route: ?? From: "+41793940965" ;tag=as00589402 ?? To: ;tag=S7UZQygFt62Nm ?? Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net ?? CSeq: 102 INVITE ?? Contact: ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Accept: application/sdp ?? 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-Length: 0 ?? Remote-Party-ID: "Outbound Call" ;party=calling;privacy=off;screen=no ?? ?? ------------------------------------------------------------------------ 2012-05-24 23:08:47.400741 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:08:47.400741 [NOTICE] mod_sofia.c:2514 Ring-Ready sofia/internal/+41793940965 at bluesip.net! 2012-05-24 23:08:47.400741 [DEBUG] sofia.c:5532 Channel sofia/internal/+41793940965 at bluesip.net entering state [early][180] 2012-05-24 23:08:47.400741 [DEBUG] switch_core_session.c:731 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:08:47.400741 [NOTICE] switch_ivr_originate.c:483 Ring Ready sofia/internal/+41793940965 at bluesip.net! recv 496 bytes from udp/[78.138.90.58]:5060 at 23:08:47.804967: ?? ------------------------------------------------------------------------ ?? SIP/2.0 180 Ringing ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKapDgFeegKyHXK ?? Record-Route: ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ;tag=549D2DD03BBA7C67 ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? CSeq: 28614398 INVITE ?? Contact: ?? User-Agent: AVM FRITZ!Box Fon WLAN 7170 29.04.80 (Jan 27 2010) ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ 2012-05-24 23:08:47.800746 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:47.800746 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:47.800746 [DEBUG] sofia.c:5532 Channel sofia/internal/1002 at 78.138.90.58:5060 entering state [proceeding][180] recv 496 bytes from udp/[78.138.90.58]:5060 at 23:08:48.182098: ?? ------------------------------------------------------------------------ ?? SIP/2.0 180 Ringing ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKapDgFeegKyHXK ?? Record-Route: ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ;tag=qCSaB2YJOAsAr2dJ-RWNMhPoDdn1RFvs ?? CSeq: 28614398 INVITE ?? Contact: "Mama" ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, REGISTER ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ 2012-05-24 23:08:48.180741 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:48.180741 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:48.180741 [DEBUG] sofia.c:5532 Channel sofia/internal/1002 at 78.138.90.58:5060 entering state [proceeding][180] recv 496 bytes from udp/[78.138.90.58]:5060 at 23:08:48.417403: ?? ------------------------------------------------------------------------ ?? SIP/2.0 180 Ringing ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKapDgFeegKyHXK ?? Record-Route: ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ;tag=549D2DD03BBA7C67 ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? CSeq: 28614398 INVITE ?? Contact: ?? User-Agent: AVM FRITZ!Box Fon WLAN 7170 29.04.80 (Jan 27 2010) ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ 2012-05-24 23:08:48.400744 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:48.400744 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:48.400744 [DEBUG] sofia.c:5532 Channel sofia/internal/1002 at 78.138.90.58:5060 entering state [proceeding][180] recv 496 bytes from udp/[78.138.90.58]:5060 at 23:08:49.292599: ?? ------------------------------------------------------------------------ ?? SIP/2.0 180 Ringing ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKapDgFeegKyHXK ?? Record-Route: ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ;tag=549D2DD03BBA7C67 ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? CSeq: 28614398 INVITE ?? Contact: ?? User-Agent: AVM FRITZ!Box Fon WLAN 7170 29.04.80 (Jan 27 2010) ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ 2012-05-24 23:08:49.280742 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:49.280742 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:49.280742 [DEBUG] sofia.c:5532 Channel sofia/internal/1002 at 78.138.90.58:5060 entering state [proceeding][180] recv 478 bytes from udp/[217.74.179.29]:5060 at 23:09:15.066012: ?? ------------------------------------------------------------------------ ?? CANCEL sip:gw+trunk_1000 at 78.138.90.58:6090;transport=udp;gw=trunk_1000 SIP/2.0 ?? Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 ?? Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 ?? From: "+41793940965" ;tag=as00589402 ?? To: ?? Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net ?? CSeq: 102 CANCEL ?? User-Agent: blueSIP PSTN GW ?? Max-Forwards: 69 ?? Content-Length: 0 ?? P-hint: USRLOC ?? ?? ------------------------------------------------------------------------ send 369 bytes to udp/[217.74.179.29]:5060 at 23:09:15.066176: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 ?? Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 ?? From: "+41793940965" ;tag=as00589402 ?? To: ;tag=S7UZQygFt62Nm ?? Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net ?? CSeq: 102 CANCEL ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ send 766 bytes to udp/[217.74.179.29]:5060 at 23:09:15.066289: ?? ------------------------------------------------------------------------ ?? SIP/2.0 487 Request Terminated ?? Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 ?? Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 ?? From: "+41793940965" ;tag=as00589402 ?? To: ;tag=S7UZQygFt62Nm ?? Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net ?? CSeq: 102 INVITE ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? 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-Length: 0 ?? ?? ------------------------------------------------------------------------ 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:09:15.060742 [DEBUG] sofia.c:5532 Channel sofia/internal/+41793940965 at bluesip.net entering state [terminated][487] 2012-05-24 23:09:15.060742 [DEBUG] switch_channel.c:2848 (sofia/internal/+41793940965 at bluesip.net) Callstate Change RINGING -> HANGUP 2012-05-24 23:09:15.060742 [NOTICE] sofia.c:6301 Hangup sofia/internal/+41793940965 at bluesip.net [CS_EXECUTE] [ORIGINATOR_CANCEL] 2012-05-24 23:09:15.060742 [DEBUG] switch_channel.c:2871 Send signal sofia/internal/+41793940965 at bluesip.net [KILL] 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:09:15.060742 [DEBUG] switch_channel.c:2848 (sofia/internal/1002 at 78.138.90.58:5060) Callstate Change RINGING -> HANGUP 2012-05-24 23:09:15.060742 [NOTICE] switch_ivr_originate.c:3182 Hangup sofia/internal/1002 at 78.138.90.58:5060 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL] 2012-05-24 23:09:15.060742 [DEBUG] switch_channel.c:2871 Send signal sofia/internal/1002 at 78.138.90.58:5060 [KILL] 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1002 at 78.138.90.58:5060) Running State Change CS_HANGUP 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/1002 at 78.138.90.58:5060) State HANGUP 2012-05-24 23:09:15.060742 [DEBUG] mod_sofia.c:469 Channel sofia/internal/1002 at 78.138.90.58:5060 hanging up, cause: ORIGINATOR_CANCEL 2012-05-24 23:09:15.060742 [DEBUG] mod_sofia.c:523 Sending CANCEL to sofia/internal/1002 at 78.138.90.58:5060 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:47 sofia/internal/1002 at 78.138.90.58:5060 Standard HANGUP, cause: ORIGINATOR_CANCEL 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/1002 at 78.138.90.58:5060) State HANGUP going to sleep 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:393 (sofia/internal/1002 at 78.138.90.58:5060) State Change CS_HANGUP -> CS_REPORTING 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1002 at 78.138.90.58:5060) Running State Change CS_REPORTING 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/1002 at 78.138.90.58:5060) State REPORTING 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:79 sofia/internal/1002 at 78.138.90.58:5060 Standard REPORTING, cause: ORIGINATOR_CANCEL 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/1002 at 78.138.90.58:5060) State REPORTING going to sleep 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:387 (sofia/internal/1002 at 78.138.90.58:5060) State Change CS_REPORTING -> CS_DESTROY 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1382 Session 9 (sofia/internal/1002 at 78.138.90.58:5060) Locked, Waiting on external entities 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:09:15.060742 [DEBUG] switch_ivr_originate.c:3358 Originate Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL] 2012-05-24 23:09:15.060742 [NOTICE] switch_core_session.c:1400 Session 9 (sofia/internal/1002 at 78.138.90.58:5060) Ended 2012-05-24 23:09:15.060742 [NOTICE] switch_core_session.c:1402 Close Channel sofia/internal/1002 at 78.138.90.58:5060 [CS_DESTROY] 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:491 (sofia/internal/1002 at 78.138.90.58:5060) Callstate Change HANGUP -> DOWN 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:494 (sofia/internal/1002 at 78.138.90.58:5060) Running State Change CS_DESTROY 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/1002 at 78.138.90.58:5060) State DESTROY 2012-05-24 23:09:15.060742 [DEBUG] mod_sofia.c:374 sofia/internal/1002 at 78.138.90.58:5060 SOFIA DESTROY 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:86 sofia/internal/1002 at 78.138.90.58:5060 Standard DESTROY 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/1002 at 78.138.90.58:5060) State DESTROY going to sleep 2012-05-24 23:09:15.060742 [INFO] mod_dptools.c:2922 Originate Failed.? Cause: ORIGINATOR_CANCEL 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:2287 sofia/internal/+41793940965 at bluesip.net skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/+41793940965 at bluesip.net) State EXECUTE going to sleep 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_HANGUP 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/+41793940965 at bluesip.net) State HANGUP 2012-05-24 23:09:15.060742 [DEBUG] mod_sofia.c:469 Channel sofia/internal/+41793940965 at bluesip.net hanging up, cause: ORIGINATOR_CANCEL 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:47 sofia/internal/+41793940965 at bluesip.net Standard HANGUP, cause: ORIGINATOR_CANCEL 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/+41793940965 at bluesip.net) State HANGUP going to sleep 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:393 (sofia/internal/+41793940965 at bluesip.net) State Change CS_HANGUP -> CS_REPORTING 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_REPORTING 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/+41793940965 at bluesip.net) State REPORTING send 383 bytes to udp/[78.138.90.58]:5060 at 23:09:15.087695: ?? ------------------------------------------------------------------------ ?? CANCEL sip:1002 at 78.138.90.58:5060 SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKapDgFeegKyHXK ?? Max-Forwards: 67 ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? CSeq: 28614398 CANCEL ?? Reason: FreeSWITCH;cause=487;text="ORIGINATOR_CANCEL" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 408 bytes from udp/[217.74.179.29]:5060 at 23:09:15.094670: ?? ------------------------------------------------------------------------ ?? ACK sip:gw+trunk_1000 at 78.138.90.58:6090;transport=udp;gw=trunk_1000 SIP/2.0 ?? Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 ?? From: "+41793940965" ;tag=as00589402 ?? Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net ?? To: ;tag=S7UZQygFt62Nm ?? CSeq: 102 ACK ?? User-Agent: Sip EXpress router(0.9.7 (i386/linux)) ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:79 sofia/internal/+41793940965 at bluesip.net Standard REPORTING, cause: ORIGINATOR_CANCEL 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/+41793940965 at bluesip.net) State REPORTING going to sleep 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:387 (sofia/internal/+41793940965 at bluesip.net) State Change CS_REPORTING -> CS_DESTROY 2012-05-24 23:09:15.120807 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:09:15.120807 [DEBUG] switch_core_session.c:1382 Session 8 (sofia/internal/+41793940965 at bluesip.net) Locked, Waiting on external entities 2012-05-24 23:09:15.120807 [NOTICE] switch_core_session.c:1400 Session 8 (sofia/internal/+41793940965 at bluesip.net) Ended 2012-05-24 23:09:15.120807 [NOTICE] switch_core_session.c:1402 Close Channel sofia/internal/+41793940965 at bluesip.net [CS_DESTROY] 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:491 (sofia/internal/+41793940965 at bluesip.net) Callstate Change HANGUP -> DOWN 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:494 (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_DESTROY 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/+41793940965 at bluesip.net) State DESTROY 2012-05-24 23:09:15.120807 [DEBUG] mod_sofia.c:374 sofia/internal/+41793940965 at bluesip.net SOFIA DESTROY 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:86 sofia/internal/+41793940965 at bluesip.net Standard DESTROY 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/+41793940965 at bluesip.net) State DESTROY going to sleep recv 378 bytes from udp/[78.138.90.58]:5060 at 23:09:15.135235: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 canceling ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKapDgFeegKyHXK ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ;tag=a6a1c5f60faecf035a1ae5b6e96e979a-8c5b ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? CSeq: 28614398 CANCEL ?? Server: kamailio (3.1.5 (x86_64/linux)) ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 446 bytes from udp/[78.138.90.58]:5060 at 23:09:16.419308: ?? ------------------------------------------------------------------------ ?? SIP/2.0 487 Request Terminated ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKapDgFeegKyHXK ?? Record-Route: ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ;tag=qCSaB2YJOAsAr2dJ-RWNMhPoDdn1RFvs ?? CSeq: 28614398 INVITE ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, REGISTER ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 354 bytes to udp/[78.138.90.58]:5060 at 23:09:16.419543: ?? ------------------------------------------------------------------------ ?? ACK sip:1002 at 78.138.90.58:5060 SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKapDgFeegKyHXK ?? Max-Forwards: 67 ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ;tag=qCSaB2YJOAsAr2dJ-RWNMhPoDdn1RFvs ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? CSeq: 28614398 ACK ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 564 bytes from udp/[41.34.123.243]:58881 at 23:13:13.612418: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjHQIpv.ACHGbXD4IHrMr72LwZ8kYdlu-J ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=wOP.FVT4Crk8O5E.bB7BWUhNO6QG91IG ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60926 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 744 bytes to udp/[41.34.123.243]:58881 at 23:13:13.613152: ?? ------------------------------------------------------------------------ ?? SIP/2.0 401 Unauthorized ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjHQIpv.ACHGbXD4IHrMr72LwZ8kYdlu-J;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=wOP.FVT4Crk8O5E.bB7BWUhNO6QG91IG ?? To: "Sherif 1003" ;tag=USeHUmjpmrFUB ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60926 REGISTER ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? WWW-Authenticate: Digest realm="sip.pcfone.com", nonce="09674764-a5f6-11e1-a83c-7b1f73a7ffcf", algorithm=MD5, qop="auth" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 830 bytes from udp/[41.34.123.243]:58881 at 23:13:13.847746: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjFr7rIAdAJPRQd2GmrMrVib3KT3Rup0q4 ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=wOP.FVT4Crk8O5E.bB7BWUhNO6QG91IG ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60927 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Authorization: Digest username="1002", realm="sip.pcfone.com", nonce="09674764-a5f6-11e1-a83c-7b1f73a7ffcf", uri="sip:sip.pcfone.com:6090", response="3bc0bb82a803ded3641704c4eb78d58f", algorithm=MD5, cnonce="sc3AJfhuev18Bs-L-6tORmI.RXlxnPOz", qop=auth, nc=00000001 ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 706 bytes to udp/[41.34.123.243]:58881 at 23:13:13.850197: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjFr7rIAdAJPRQd2GmrMrVib3KT3Rup0q4;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=wOP.FVT4Crk8O5E.bB7BWUhNO6QG91IG ?? To: "Sherif 1003" ;tag=v279vF3SH15DQ ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60927 REGISTER ?? Contact: ;expires=300 ?? Date: Thu, 24 May 2012 23:13:13 GMT ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ send 989 bytes to udp/[41.34.123.243]:58881 at 23:13:13.888717: ?? ------------------------------------------------------------------------ ?? NOTIFY sip:1002 at 192.168.178.26:58881;ob SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKBZ68g9yKg77FF ?? Route: ;ob ?? Max-Forwards: 70 ?? From: ;tag=XB12yamXeav0j ?? To: ?? Call-ID: e0efa252-2098-1230-8985-00163e6bb553 ?? CSeq: 28614532 NOTIFY ?? Contact: ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? 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=noresource ?? Content-Type: application/simple-message-summary ?? Content-Length: 91 ?? ?? Messages-Waiting: yes ?? Message-Account: sip:1002 at 78.138.90.58 ?? Voice-Message: 1/0 (0/0) ?? ?? ------------------------------------------------------------------------ recv 308 bytes from udp/[41.34.123.243]:58881 at 23:13:14.298850: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;received=78.138.90.58;branch=z9hG4bKBZ68g9yKg77FF ?? Call-ID: e0efa252-2098-1230-8985-00163e6bb553 ?? From: ;tag=XB12yamXeav0j ?? To: ;tag=z9hG4bKBZ68g9yKg77FF ?? CSeq: 28614532 NOTIFY ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ recv 564 bytes from udp/[41.34.123.243]:58881 at 23:18:09.067232: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjABTslmZIhiHYRtLqQn2erwmX19MVYHFC ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=JBSXL5WZLWbLzpvRFoTSP5x8qQslI1ST ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60928 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 744 bytes to udp/[41.34.123.243]:58881 at 23:18:09.068121: ?? ------------------------------------------------------------------------ ?? SIP/2.0 401 Unauthorized ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjABTslmZIhiHYRtLqQn2erwmX19MVYHFC;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=JBSXL5WZLWbLzpvRFoTSP5x8qQslI1ST ?? To: "Sherif 1003" ;tag=ymtU0540BKjKe ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60928 REGISTER ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? WWW-Authenticate: Digest realm="sip.pcfone.com", nonce="b9821df4-a5f6-11e1-a83d-7b1f73a7ffcf", algorithm=MD5, qop="auth" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 830 bytes from udp/[41.34.123.243]:58881 at 23:18:09.302944: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjRgtztcy63vMT97Jp5pH9n311li5EFeY- ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=JBSXL5WZLWbLzpvRFoTSP5x8qQslI1ST ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60929 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Authorization: Digest username="1002", realm="sip.pcfone.com", nonce="b9821df4-a5f6-11e1-a83d-7b1f73a7ffcf", uri="sip:sip.pcfone.com:6090", response="519c6301816b51f3bbb98d97a347e92b", algorithm=MD5, cnonce="sc3AJfhuev18Bs-L-6tORmI.RXlxnPOz", qop=auth, nc=00000001 ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 706 bytes to udp/[41.34.123.243]:58881 at 23:18:09.305217: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjRgtztcy63vMT97Jp5pH9n311li5EFeY-;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=JBSXL5WZLWbLzpvRFoTSP5x8qQslI1ST ?? To: "Sherif 1003" ;tag=ZXKm20N48U85S ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60929 REGISTER ?? Contact: ;expires=300 ?? Date: Thu, 24 May 2012 23:18:09 GMT ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ send 989 bytes to udp/[41.34.123.243]:58881 at 23:18:09.379987: ?? ------------------------------------------------------------------------ ?? NOTIFY sip:1002 at 192.168.178.26:58881;ob SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKc8Z1j4FQDgy2a ?? Route: ;ob ?? Max-Forwards: 70 ?? From: ;tag=06cD4U6754yrN ?? To: ?? Call-ID: 91100602-2099-1230-8985-00163e6bb553 ?? CSeq: 28614680 NOTIFY ?? Contact: ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? 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=noresource ?? Content-Type: application/simple-message-summary ?? Content-Length: 91 ?? ?? Messages-Waiting: yes ?? Message-Account: sip:1002 at 78.138.90.58 ?? Voice-Message: 1/0 (0/0) ?? ?? ------------------------------------------------------------------------ recv 308 bytes from udp/[41.34.123.243]:58881 at 23:18:09.773629: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;received=78.138.90.58;branch=z9hG4bKc8Z1j4FQDgy2a ?? Call-ID: 91100602-2099-1230-8985-00163e6bb553 ?? From: ;tag=06cD4U6754yrN ?? To: ;tag=z9hG4bKc8Z1j4FQDgy2a ?? CSeq: 28614680 NOTIFY ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ recv 564 bytes from udp/[41.34.123.243]:58881 at 23:23:04.522455: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjaCuKyaBi6ZqcsHyp5HhH3WthJStom0SK ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=qfUt1fQWuE-Ro3zWld6X0r6WidtD6B1q ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60930 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 744 bytes to udp/[41.34.123.243]:58881 at 23:23:04.523158: ?? ------------------------------------------------------------------------ ?? SIP/2.0 401 Unauthorized ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjaCuKyaBi6ZqcsHyp5HhH3WthJStom0SK;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=qfUt1fQWuE-Ro3zWld6X0r6WidtD6B1q ?? To: "Sherif 1003" ;tag=1F655pQB3DNBH ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60930 REGISTER ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? WWW-Authenticate: Digest realm="sip.pcfone.com", nonce="699cfe2a-a5f7-11e1-a83e-7b1f73a7ffcf", algorithm=MD5, qop="auth" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 830 bytes from udp/[41.34.123.243]:58881 at 23:23:04.757798: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjs8SAufeoyzCLrePaAbYhZuOqQF47XIc- ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=qfUt1fQWuE-Ro3zWld6X0r6WidtD6B1q ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60931 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Authorization: Digest username="1002", realm="sip.pcfone.com", nonce="699cfe2a-a5f7-11e1-a83e-7b1f73a7ffcf", uri="sip:sip.pcfone.com:6090", response="23a886934aaf049f08432b6bb226bd37", algorithm=MD5, cnonce="sc3AJfhuev18Bs-L-6tORmI.RXlxnPOz", qop=auth, nc=00000001 ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 706 bytes to udp/[41.34.123.243]:58881 at 23:23:04.759924: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjs8SAufeoyzCLrePaAbYhZuOqQF47XIc-;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=qfUt1fQWuE-Ro3zWld6X0r6WidtD6B1q ?? To: "Sherif 1003" ;tag=2rZy7H8e0pByc ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60931 REGISTER ?? Contact: ;expires=300 ?? Date: Thu, 24 May 2012 23:23:04 GMT ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ send 989 bytes to udp/[41.34.123.243]:58881 at 23:23:04.872728: ?? ------------------------------------------------------------------------ ?? NOTIFY sip:1002 at 192.168.178.26:58881;ob SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKDHStmZ0taSmNp ?? Route: ;ob ?? Max-Forwards: 70 ?? From: ;tag=31rQ9cSjXZ1gr ?? To: ?? Call-ID: 4130a1ce-209a-1230-8985-00163e6bb553 ?? CSeq: 28614828 NOTIFY ?? Contact: ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? 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=noresource ?? Content-Type: application/simple-message-summary ?? Content-Length: 91 ?? ?? Messages-Waiting: yes ?? Message-Account: sip:1002 at 78.138.90.58 ?? Voice-Message: 1/0 (0/0) ?? ?? ------------------------------------------------------------------------ recv 308 bytes from udp/[41.34.123.243]:58881 at 23:23:05.269063: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;received=78.138.90.58;branch=z9hG4bKDHStmZ0taSmNp ?? Call-ID: 4130a1ce-209a-1230-8985-00163e6bb553 ?? From: ;tag=31rQ9cSjXZ1gr ?? To: ;tag=z9hG4bKDHStmZ0taSmNp ?? CSeq: 28614828 NOTIFY ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 916 bytes to udp/[217.74.179.29]:5060 at 23:25:45.873423: ?? ------------------------------------------------------------------------ ?? REGISTER sip:bluesip.net;transport=udp SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKetjKptHy71a8H ?? Max-Forwards: 70 ?? From: ;tag=6r0vBQZS650Fg ?? To: ?? Call-ID: 2a3e4696-a5b9-11e1-a7ed-7b1f73a7ffcf ?? CSeq: 28601493 REGISTER ?? Contact: ?? Expires: 3600 ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? Authorization: Digest username="bluesip/sherifomran", realm="bluesip.net", nonce="4fbeba4ebfb7414d5c701bd6f994402d551f3acf", algorithm=MD5, uri="sip:bluesip.net;transport=udp", response="c6cdcafe0418e519bc9ee0d8fa3d4d74" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 722 bytes from udp/[217.74.179.29]:5060 at 23:25:45.904176: ?? ------------------------------------------------------------------------ ?? SIP/2.0 401 Unauthorized ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKetjKptHy71a8H ?? From: ;tag=6r0vBQZS650Fg ?? To: ;tag=0354a2e1b960c9cc2279eca4e5f84e20.1b16 ?? Call-ID: 2a3e4696-a5b9-11e1-a7ed-7b1f73a7ffcf ?? CSeq: 28601493 REGISTER ?? WWW-Authenticate: Digest realm="bluesip.net", nonce="4fbec4a4ba82d1f1feaa32ccec01b85be054fcf7", stale=true ?? Server: Sip EXpress router (0.9.7 (i386/linux)) ?? Content-Length: 0 ?? Warning: 392 217.74.179.29:5060 "Noisy feedback tells:? pid=5455 req_src_ip=78.138.90.58 req_src_port=6090 in_uri=sip:bluesip.net;transport=udp out_uri=sip:bluesip.net;transport=udp via_cnt==1" ?? ?? ------------------------------------------------------------------------ send 916 bytes to udp/[217.74.179.29]:5060 at 23:25:45.904416: ?? ------------------------------------------------------------------------ ?? REGISTER sip:bluesip.net;transport=udp SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKF3BcrN214a1tD ?? Max-Forwards: 70 ?? From: ;tag=6r0vBQZS650Fg ?? To: ?? Call-ID: 2a3e4696-a5b9-11e1-a7ed-7b1f73a7ffcf ?? CSeq: 28601494 REGISTER ?? Contact: ?? Expires: 3600 ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? Authorization: Digest username="bluesip/sherifomran", realm="bluesip.net", nonce="4fbec4a4ba82d1f1feaa32ccec01b85be054fcf7", algorithm=MD5, uri="sip:bluesip.net;transport=udp", response="4c09dbe4b9accb52d4104b40dfe20040" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 699 bytes from udp/[217.74.179.29]:5060 at 23:25:45.935046: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKF3BcrN214a1tD ?? From: ;tag=6r0vBQZS650Fg ?? To: ;tag=0354a2e1b960c9cc2279eca4e5f84e20.2017 ?? Call-ID: 2a3e4696-a5b9-11e1-a7ed-7b1f73a7ffcf ?? CSeq: 28601494 REGISTER ?? Contact: ;q=0.5;expires=3600 ?? Server: Sip EXpress router (0.9.7 (i386/linux)) ?? Content-Length: 0 ?? Warning: 392 217.74.179.29:5060 "Noisy feedback tells:? pid=5462 req_src_ip=78.138.90.58 req_src_port=6090 in_uri=sip:bluesip.net;transport=udp out_uri=sip:bluesip.net;transport=udp via_cnt==1" ?? ?? ------------------------------------------------------------------------ recv 564 bytes from udp/[41.34.123.243]:58881 at 23:27:59.977057: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjEOP.a4NoXqoFqC7a.k2tyeX-mYlUyH4H ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60932 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 744 bytes to udp/[41.34.123.243]:58881 at 23:27:59.977960: ?? ------------------------------------------------------------------------ ?? SIP/2.0 401 Unauthorized ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjEOP.a4NoXqoFqC7a.k2tyeX-mYlUyH4H;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K ?? To: "Sherif 1003" ;tag=4ajgB89Nt8Q3K ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60932 REGISTER ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? WWW-Authenticate: Digest realm="sip.pcfone.com", nonce="19b7cc40-a5f8-11e1-a83f-7b1f73a7ffcf", algorithm=MD5, qop="auth" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 564 bytes from udp/[41.34.123.243]:58881 at 23:28:00.468487: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjEOP.a4NoXqoFqC7a.k2tyeX-mYlUyH4H ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60932 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 744 bytes to udp/[41.34.123.243]:58881 at 23:28:00.468679: ?? ------------------------------------------------------------------------ ?? SIP/2.0 401 Unauthorized ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjEOP.a4NoXqoFqC7a.k2tyeX-mYlUyH4H;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K ?? To: "Sherif 1003" ;tag=4ajgB89Nt8Q3K ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60932 REGISTER ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? WWW-Authenticate: Digest realm="sip.pcfone.com", nonce="19b7cc40-a5f8-11e1-a83f-7b1f73a7ffcf", algorithm=MD5, qop="auth" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 830 bytes from udp/[41.34.123.243]:58881 at 23:28:00.717446: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPj0iHd3-XUjnvcBs9DiJUCWzrmYTmWxFLt ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60933 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Authorization: Digest username="1002", realm="sip.pcfone.com", nonce="19b7cc40-a5f8-11e1-a83f-7b1f73a7ffcf", uri="sip:sip.pcfone.com:6090", response="e21c903e6d98b5adcdd2eefa75217157", algorithm=MD5, cnonce="sc3AJfhuev18Bs-L-6tORmI.RXlxnPOz", qop=auth, nc=00000001 ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 706 bytes to udp/[41.34.123.243]:58881 at 23:28:00.719947: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPj0iHd3-XUjnvcBs9DiJUCWzrmYTmWxFLt;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K ?? To: "Sherif 1003" ;tag=5KB9c3tSQHepF ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60933 REGISTER ?? Contact: ;expires=300 ?? Date: Thu, 24 May 2012 23:28:00 GMT ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ send 989 bytes to udp/[41.34.123.243]:58881 at 23:28:00.868368: ?? ------------------------------------------------------------------------ ?? NOTIFY sip:1002 at 192.168.178.26:58881;ob SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKgc54SgK51KQDS ?? Route: ;ob ?? Max-Forwards: 70 ?? From: ;tag=6v41eyBXmt48a ?? To: ?? Call-ID: f19dfbce-209a-1230-8985-00163e6bb553 ?? CSeq: 28614976 NOTIFY ?? Contact: ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? 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=noresource ?? Content-Type: application/simple-message-summary ?? Content-Length: 91 ?? ?? Messages-Waiting: yes ?? Message-Account: sip:1002 at 78.138.90.58 ?? Voice-Message: 1/0 (0/0) ?? ?? ------------------------------------------------------------------------ recv 308 bytes from udp/[41.34.123.243]:58881 at 23:28:01.128279: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;received=78.138.90.58;branch=z9hG4bKgc54SgK51KQDS ?? Call-ID: f19dfbce-209a-1230-8985-00163e6bb553 ?? From: ;tag=6v41eyBXmt48a ?? To: ;tag=z9hG4bKgc54SgK51KQDS ?? CSeq: 28614976 NOTIFY ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 913 bytes to udp/[217.74.179.29]:5060 at 23:28:11.885213: ?? ------------------------------------------------------------------------ ?? REGISTER sip:bluesip.net;transport=udp SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKHNyXUB48yvD0m ?? Max-Forwards: 70 ?? From: ;tag=5F739Uep9vaXm ?? To: ?? Call-ID: 2a3e49c0-a5b9-11e1-a7ee-7b1f73a7ffcf ?? CSeq: 28601493 REGISTER ?? Contact: ?? Expires: 3600 ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? Authorization: Digest username="bluesip/salahomran", realm="bluesip.net", nonce="4fbebeca79a5cff7c417463814cb9054d313bcac", algorithm=MD5, uri="sip:bluesip.net;transport=udp", response="bb1babb90f4ea1dc8fbb9aa45d6038c7" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 720 bytes from udp/[217.74.179.29]:5060 at 23:28:11.915764: ?? ------------------------------------------------------------------------ ?? SIP/2.0 401 Unauthorized ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKHNyXUB48yvD0m ?? From: ;tag=5F739Uep9vaXm ?? To: ;tag=0354a2e1b960c9cc2279eca4e5f84e20.3454 ?? Call-ID: 2a3e49c0-a5b9-11e1-a7ee-7b1f73a7ffcf ?? CSeq: 28601493 REGISTER ?? WWW-Authenticate: Digest realm="bluesip.net", nonce="4fbec5365af3a174f1cc0b8cda770c727d25bf71", stale=true ?? Server: Sip EXpress router (0.9.7 (i386/linux)) ?? Content-Length: 0 ?? Warning: 392 217.74.179.29:5060 "Noisy feedback tells:? pid=5462 req_src_ip=78.138.90.58 req_src_port=6090 in_uri=sip:bluesip.net;transport=udp out_uri=sip:bluesip.net;transport=udp via_cnt==1" ?? ?? ------------------------------------------------------------------------ send 913 bytes to udp/[217.74.179.29]:5060 at 23:28:11.915914: ?? ------------------------------------------------------------------------ ?? REGISTER sip:bluesip.net;transport=udp SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKjyQpX6mcv53jg ?? Max-Forwards: 70 ?? From: ;tag=5F739Uep9vaXm ?? To: ?? Call-ID: 2a3e49c0-a5b9-11e1-a7ee-7b1f73a7ffcf ?? CSeq: 28601494 REGISTER ?? Contact: ?? Expires: 3600 ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? Authorization: Digest username="bluesip/salahomran", realm="bluesip.net", nonce="4fbec5365af3a174f1cc0b8cda770c727d25bf71", algorithm=MD5, uri="sip:bluesip.net;transport=udp", response="e260f6dc9b01ce8bea8c5ffb36bd4ddd" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 697 bytes from udp/[217.74.179.29]:5060 at 23:28:11.946325: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKjyQpX6mcv53jg ?? From: ;tag=5F739Uep9vaXm ?? To: ;tag=0354a2e1b960c9cc2279eca4e5f84e20.4b5c ?? Call-ID: 2a3e49c0-a5b9-11e1-a7ee-7b1f73a7ffcf ?? CSeq: 28601494 REGISTER ?? Contact: ;q=0.5;expires=3600 ?? Server: Sip EXpress router (0.9.7 (i386/linux)) ?? Content-Length: 0 ?? Warning: 392 217.74.179.29:5060 "Noisy feedback tells:? pid=5455 req_src_ip=78.138.90.58 req_src_port=6090 in_uri=sip:bluesip.net;transport=udp out_uri=sip:bluesip.net;transport=udp via_cnt==1" ?? ?? ------------------------------------------------------------------------ recv 564 bytes from udp/[41.34.123.243]:58881 at 23:32:55.930149: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjuk8DZGiaXfN7BmtslNHVjtSg5UXsbHXc ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=Pz.t-0tGXj3saM.ehBiB1EAMmmitze1D ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60934 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 744 bytes to udp/[41.34.123.243]:58881 at 23:32:55.931324: ?? ------------------------------------------------------------------------ ?? SIP/2.0 401 Unauthorized ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjuk8DZGiaXfN7BmtslNHVjtSg5UXsbHXc;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=Pz.t-0tGXj3saM.ehBiB1EAMmmitze1D ?? To: "Sherif 1003" ;tag=75XtgSv0H3tUp ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60934 REGISTER ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? WWW-Authenticate: Digest realm="sip.pcfone.com", nonce="ca1eabee-a5f8-11e1-a840-7b1f73a7ffcf", algorithm=MD5, qop="auth" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 830 bytes from udp/[41.34.123.243]:58881 at 23:32:56.165624: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjKEQ6xXmyzZxA6vCM88vHNYAOPAjUqFsD ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=Pz.t-0tGXj3saM.ehBiB1EAMmmitze1D ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60935 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Authorization: Digest username="1002", realm="sip.pcfone.com", nonce="ca1eabee-a5f8-11e1-a840-7b1f73a7ffcf", uri="sip:sip.pcfone.com:6090", response="ff6d12f4ec7abd4b37187222b3af2358", algorithm=MD5, cnonce="sc3AJfhuev18Bs-L-6tORmI.RXlxnPOz", qop=auth, nc=00000001 ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 706 bytes to udp/[41.34.123.243]:58881 at 23:32:56.168033: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjKEQ6xXmyzZxA6vCM88vHNYAOPAjUqFsD;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=Pz.t-0tGXj3saM.ehBiB1EAMmmitze1D ?? To: "Sherif 1003" ;tag=8eQKjmD4ecHej ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60935 REGISTER ?? Contact: ;expires=300 ?? Date: Thu, 24 May 2012 23:32:56 GMT ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ send 989 bytes to udp/[41.34.123.243]:58881 at 23:32:56.360238: ?? ------------------------------------------------------------------------ ?? NOTIFY sip:1002 at 192.168.178.26:58881;ob SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKK7gFZ15FSet5B ?? Route: ;ob ?? Max-Forwards: 70 ?? From: ;tag=9QgcmFy7BN70D ?? To: ?? Call-ID: a1be7708-209b-1230-8985-00163e6bb553 ?? CSeq: 28615124 NOTIFY ?? Contact: ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? 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=noresource ?? Content-Type: application/simple-message-summary ?? Content-Length: 91 ?? ?? Messages-Waiting: yes ?? Message-Account: sip:1002 at 78.138.90.58 ?? Voice-Message: 1/0 (0/0) ?? ?? ------------------------------------------------------------------------ recv 308 bytes from udp/[41.34.123.243]:58881 at 23:32:56.623554: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;received=78.138.90.58;branch=z9hG4bKK7gFZ15FSet5B ?? Call-ID: a1be7708-209b-1230-8985-00163e6bb553 ?? From: ;tag=9QgcmFy7BN70D ?? To: ;tag=z9hG4bKK7gFZ15FSet5B ?? CSeq: 28615124 NOTIFY ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ freeswitch at internal> --- On Thu, 5/24/12, Kristian Kielhofner wrote: From: Kristian Kielhofner Subject: Re: [Freeswitch-users] Freeswitch (SBC + Media) does not know that gateway phone is picked up To: "FreeSWITCH Users Help" Date: Thursday, May 24, 2012, 7:51 PM Siptrace and logs please. On Thu, May 24, 2012 at 11:53 AM, Sherif Omran wrote: > > Hi all, > > My topology is as follows: > > Kamailio? -> FS (SBS+Media server) > > I came across an issue with my system as follows.? I have a Hardphone registered. When I do local call inside kamailio, it gets to FS and returns back well and FS understands when I lift the handset. However, I added a gateway (german landline server), when I call my self from another phone, the call gets to FS and then transmits to Kamailio, it rings my extension but when I lift the handset FS does not notice it and keeps ringing. > > Any body has an Idea? Here is my gateway trunk. > > > ??????? > ??????? > ??????? > ??????? > ??????? > ????? > > > thanks in advance > Sherif Omran > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Kristian Kielhofner _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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 at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -----Inline Attachment Follows----- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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 at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -----Inline Attachment Follows----- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120525/ccdb4dd9/attachment-0001.html From drk at drkngs.net Fri May 25 22:50:02 2012 From: drk at drkngs.net (Dave R. Kompel) Date: Fri, 25 May 2012 11:50:02 -0700 Subject: [Freeswitch-users] Bridge incoming call to external number In-Reply-To: <4FBF37C2.3090401@the800group.com> Message-ID: <20120525185002.0bc97db6@mail.tritonwest.net> This is not a FreeSWITCH issue. A lot of FXO terminal adaptors do not detect CPC right. In some cases they don't even do it on outgoing calls. Make sure your device is set to detect CPC (Calling Party Control) signalls form the phone line. Make sure the device you are using cal do that too. --Dave _____ From: ocset [mailto:ocset at the800group.com] To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org] Sent: Fri, 25 May 2012 00:41:54 -0700 Subject: [Freeswitch-users] Bridge incoming call to external number Hi Setup Windows 7 Freeswitch GXW4104 FXO 2 x POTS lines I have setup Freeswitch to bridge an incoming call to an external mobile phone using a GXW4104 FXO device. The bridge works great but I am having a real headache with the connection not being disconnected when the users hang up. There is no disconnection issue when a call comes in on either line and is connected to an internal extension (like ext 1001). It is when one line is bride to the second that the issue occurs. 1. Would this be a limitation of the GXW4104 2. Is it possible to write a "lua" script that could manage the disconnect some how? 3. If I enable "Polarity Reversal" under "FXO Termination" then the disconnect works but I have issues with sound and the system not registering when someone picks up. Please help me fix/diagnose this problem. Thanks O _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120525/31efdd8d/attachment.html From neilp at cs.stanford.edu Sat May 26 01:53:34 2012 From: neilp at cs.stanford.edu (Neil Patel) Date: Fri, 25 May 2012 14:53:34 -0700 Subject: [Freeswitch-users] Unable to build Git tree on Mac OSX Lion Message-ID: Hi All, I am getting this error when trying to build latest git on Mac OSX Lion, after fresh checkout and running ./boostraph.sh and ./configure: making all mod_amr Creating mod_amr.so... i686-apple-darwin11-llvm-gcc-4.2: -bundle not allowed with -dynamiclib gcc -I/usr/local/src/freeswitch/libs/curl/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DMACOSX -DHAVE_OPENSSL -g -O2 -Wall -std=c99 -pedantic -Wdeclaration-after-statement -D_GNU_SOURCE -DAMR_PASSTHROUGH -shared -o .libs/mod_amr.so -dynamic -bundle -force-flat-namespace .libs/mod_amr.o /usr/local/src/freeswitch/.libs/libfreeswitch.dylib -lodbc -lresolv -L/usr/local/src/freeswitch/libs/apr-util/xml/expat/lib -lpq /usr/local/src/freeswitch/libs/apr-util/xml/expat/lib/.libs/libexpat.a /usr/local/src/freeswitch/libs/apr/.libs/libapr-1.a -lpthread -lm -L/usr/local/src/freeswitch/libs/srtp -ldl -lssl -lcrypto -lz -lncurses /usr/local/lib/libjpeg.dylib /usr/local/lib/libodbc.dylib -liconv make[5]: *** [mod_amr.so] Error 1 make[4]: *** [all] Error 1 make[3]: *** [mod_amr-all] Error 1 make[2]: *** [all-recursive] Error 1 make[1]: *** [all-recursive] Error 1 make: *** [all] Error 2 Heeding Mario's advice, I tried making with open-ssl flag, but that didn't help. If I go about commenting out offending modules, the same "bundle not allowed with -dynamiclib" appears for any number of other mods. I have installed all the prerequisites for Mac OSX (to my knowledge), including Apple and Unix dev tools. I am 100% macports/fink/brew free, AFAIK. Jira filed, but no response for a while: http://jira.freeswitch.org/browse/FS-4240 My sense this is a missing tool or config issue with my setup (pretty much fresh Lion install). Thanks in advance for your help! Neil -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120525/e93ca73d/attachment.html From nathandownes at hotmail.com Sat May 26 02:03:32 2012 From: nathandownes at hotmail.com (Mr Nathan Downes) Date: Sat, 26 May 2012 08:03:32 +1000 Subject: [Freeswitch-users] RTP media issue In-Reply-To: <007301cd3a96$d106ed90$7314c8b0$@gmail.com> References: <02a201cd3a1a$2a658770$7f309650$@gmail.com> <007301cd3a96$d106ed90$7314c8b0$@gmail.com> Message-ID: Hi Anthony, FS version = FreeSWITCH Version 1.1.beta1 (git-f1b5044 2012-04-26 11-28-47 -0500) I don't have a debug log, but I could probably get it with another trace of both sides of the call, but it would be hard to capture as there is constant calls to this, unless there is a way to do it on a per call basis? I can also only do testing onsite as we don't have the same fibre equipment. I already have jitterbuffer set in both profiles in an attempt to try and stop it using , is there a way to set the cng_plc in the profile itself rather than diaplpan as there are 70 or so numbers in it. In the outbound dialplans I also added because I kept seeing PAUSE JITTERBUFFER in the FS logs when calls were made outbound so I wasn't sure it was doing something and read somewhere it pauses it when it bridges the call The inbound dialplan for all of those people consists of It doesn't affect SIP phones or normal ATA devices we have connected and only affects these FTTH GPON ATA's, but with almost 100 residents in this retirement village and them constantly complaining we have been given til Wednesday to come up with a solution or risk losing our position as the internet/phone provider for that retirement village. It appeared to me that what happens in that trace isn't normal behaviour, I did try rewriting timestamps last week, but as you suggested that appeared to mask the issue but not stop it from happening. That was when I was losing a packet from them each second or so, which by the time it arrived to end user sounded horrible. It has settled down a lot now and maybe 1 or 2 packets per call, but if what is in this trace is the cause each time, that would explain the poor end users experience. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Saturday, 26 May 2012 2:06 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] RTP media issue What version of FS are you running? Do you have the debug logs of those calls? you could try using the jitterbuffer. in the inbound DP to FS *before* you answer. Also it looks a little odd to me in this trace if this is the same call, it seems like you answered the call before placing the call to the phone and that phone never answers.... On Thu, May 24, 2012 at 9:51 PM, Nathan Downes wrote: > Hi, > > enable-soa > > > > Set the value to "false" to diable SIP SOA from sofia to tell sofia > not to touch the exchange of SDP > > I don't think this is related to the exchange of an SDP message..? Can > you elaborate more before I try it? I can't make things worse or > change things I don't understand. > > ________________________________ > From: djbinter at gmail.com > To: freeswitch-users at lists.freeswitch.org > CC: nathan at nortec.com.au > Subject: Re: [Freeswitch-users] RTP media issue > Date: Fri, 25 May 2012 11:19:46 +1000 > > > > > > Sent from my iPad > > On May 24, 2012, at 5:01 PM, Nathan Downes wrote: > > Hi, > > I had previous reported an issue with poor voice quality, appearing to > stem from occasion wrong timestamps coming from provider, but the end > user's experience was much worse than what I could see/hear in the trace. > > I have finally captured an event inbound and outbound.? The thing I > don't understand is I thought even though FS proxied the media it > didn't touch it or change anything, but it appears it is. > > The 2 traces are http://www.nortec.com.au/inbound.pcap.gz and > http://www.nortec.com.au/outbound.pcap.gz > > Inbound is from my trunk provider to FS box and outbound is FS box to > ATA in FTTH GPON. > > The event I am talking about, if both traces are open, is in the > inbound one inbetween packet 8114 and 8117 the provider drops a packet > or I don't receive it.? In the corresponding outbound trace, between > packet 8144 and 8152,? it appears FS misses a whole heap of packets > (.1 seconds) between > 8146 and 8152 then it increases the timestamp only by 40 rather than > 160 on packet 8152.? This seems to not affect SIP phones themselves > but causes issues with the FTTH GPON ATA. > > This causes a gap in the audio for the end user, and when they miss a > high number of packets even though it sounds good on the inbound trace > the end users experience is horrible.?? This trace is actually a good > one, but the wrong timestamp can occur once per second, causing end > user to lose 10%+ of incoming audio only.? The issue only affects the > audio coming from provider to FS to end user. > > I am chasing it up with the voice provider to try and eliminate the > occasional packet loss, but if I could stop/fix FS from doing its > adjustment/gap/something the end user wouldn't even notice it. > > > > ______________________________________________________________________ > ___ > > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org > > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From sdevoy at bizfocused.com Sat May 26 02:05:47 2012 From: sdevoy at bizfocused.com (Sean Devoy) Date: Fri, 25 May 2012 18:05:47 -0400 Subject: [Freeswitch-users] SCA not working inbound - Multi Domain In-Reply-To: References: <040301cd3aa3$e93705f0$bba511d0$@bizfocused.com> Message-ID: <051501cd3ac2$8a187ee0$9e497ca0$@bizfocused.com> Here is the result of select * from sip_subscriptions;" sip|220|fs_lan.bizfocused.com|220|fs_lan.bizfocused.com||call-info|"user" |38e107ab-6cde6635 at 10.10.40.30|"220" ;tag=fd508933c5f5924d|SIP/2.0/UDP 10.10.40.30:5060;branch=z9hG4bK-f5598bfb|1337980638|Cisco/SPA504G-7.4.8a||ex ternal|FreeSwitch1.sumdomain.com|5060|10.10.40.30|-1||"220" ;tag=VrGrXQaOH22R sip|220|fs_lan.bizfocused.com|220|fs_lan.bizfocused.com||call-info|"user" |c08f0c6a-c46e90d2 at 10.10.40.20|"Sean" ;tag=f22a978ae8838032|SIP/2.0/UDP 10.10.40.20:5064;branch=z9hG4bK-905b6faa|1337980643|Cisco/SPA504G-7.4.9c||ex ternal|FreeSwitch1.sumdomain.com|5064|10.10.40.20|-1||"Sean" ;tag=y5VtigPlIghD But it was in: sofia_reg_external.db not internal. I have sorted out all the sip trace data into 2 txt files for the 2 phones involved. They are zipped up at: http://www.bizfocused.com/sip_trace.zip Thank you again for your help. I am way over my head now. -----Original Message----- From: Anthony Minessale [mailto:anthony.minessale at gmail.com] Sent: Friday, May 25, 2012 2:35 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SCA not working inbound - Multi Domain What are the phones putting in the subscribe ? sofia global siptrace on sofia global debug presence|sla then watch for SUBSCRIBE also when you are not using odbc you can get the sql with this app sqlute3 /usr/local/freeswitch/db/sofia_reg_internal.db also try "select * from sip_subscriptions" its all about using the right host name across the board, IP's count as hostnames, they do not magically resolve any dns with SIP On Fri, May 25, 2012 at 1:26 PM, Sean Devoy wrote: > Hi all, > > > > I have a muti-tennnant configuration that is working nicely except for > Shared Call Appearance.? The desktop devices are CISCO 504Gs and they > are configured as described in the FS Wiki as well as Cisco Documentation. > > > > The SCA works perfectly for outbound calls ? if either phone pickups > like 220, the other phones indicator light flashes red.? However, > inbound calls will go to only one of the phones (which one has changed > a few times) and the other phones line still just stays green and does not ring. > > > > Here is the sip interfaces config: > > > > ??? > > ????? > > ????? > > ????? > > ????? > > ????? > > ????? > > ????? > > ????? > > ????? > > ????? > > ????? > > ????? > > ????? > > ????? > > ????? > > ????? > > ????? > > ????? > > ????? > > ??? > > ? > > > > The directory entry which both phones connect using: > > ??? > > ????? > > ??????? > > ??????? value="410420BLEEP"/> > > ??????? > > ??????? > > ??????? > > ??????? > > ??????? > > ??????? > > ??????? > > ????? > > ????? > > ??????? " > > ??????? > > ??????? value="{presence_id=220 at mydomainname.com}${sofia_contact(220 at mydomainn > ame.com)}"/> > > ??????? > > ????? > > ??? > > > > And the dial plan for ext 220: > > ? > > ??? > > ????? data="effective_caller_id_number=${internal_caller_id_number}"/> > > ????? data="effective_caller_id_name=${internal_caller_id_name}"/> > > ????? > > ????? > > ????? > > ????? data="{sip_invite_domain=mydomainname.com}user/220 at mydomainname.com"? > /> > > ?????? > > ????? > > ????? > > ??? > > ? > > > > > > > > I did see this in the wiki > (http://wiki.freeswitch.org/wiki/Shared_Line_Appearance): > > If SLA works for outgoing calls and SLA does not work for inbound > calls to the SLA phones, you may have some presence problem related to > mixed IP and domain names. When using ODBC you may issue the following > SQL statement > > select > sip_to_host,sip_from_user,sip_from_host,hostname,presence_id,call_info > from sip_dialogs; > > But I don?t have ODBC on this server, so I am a little lost. > > > > I have the phones login to domain names, not addresses.? I never refer > to IP addresses in my xml (except gateways addresses).? I am not > trying SLA across domain, only within the same domain. > > > > I hope someone can spot something.? Thanks for your help. > > > > Sean > > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From mario_fs at mgtech.com Sat May 26 02:53:32 2012 From: mario_fs at mgtech.com (Mario G) Date: Fri, 25 May 2012 15:53:32 -0700 Subject: [Freeswitch-users] Unable to build Git tree on Mac OSX Lion In-Reply-To: References: Message-ID: <9C746911-552E-4E7A-9C0C-DB0F5676A108@mgtech.com> The impression I get from the wiki is that mod_amr is optional. I would try to comment it out in modules.conf in the FS source dir and see what happens. I wish I had time to futz with FS on Lion but I am trying to resolve 2 other FS issues for over a month with no luck. This is why I put off updating the wiki, I won't until everything is perfect. On May 25, 2012, at 2:53 PM, Neil Patel wrote: > Hi All, I am getting this error when trying to build latest git on Mac OSX Lion, after fresh checkout and running ./boostraph.sh and ./configure: > > making all mod_amr > Creating mod_amr.so... > i686-apple-darwin11-llvm-gcc-4.2: -bundle not allowed with -dynamiclib > gcc -I/usr/local/src/freeswitch/libs/curl/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DMACOSX -DHAVE_OPENSSL -g -O2 -Wall -std=c99 -pedantic -Wdeclaration-after-statement -D_GNU_SOURCE -DAMR_PASSTHROUGH -shared -o .libs/mod_amr.so -dynamic -bundle -force-flat-namespace .libs/mod_amr.o /usr/local/src/freeswitch/.libs/libfreeswitch.dylib -lodbc -lresolv -L/usr/local/src/freeswitch/libs/apr-util/xml/expat/lib -lpq /usr/local/src/freeswitch/libs/apr-util/xml/expat/lib/.libs/libexpat.a /usr/local/src/freeswitch/libs/apr/.libs/libapr-1.a -lpthread -lm -L/usr/local/src/freeswitch/libs/srtp -ldl -lssl -lcrypto -lz -lncurses /usr/local/lib/libjpeg.dylib /usr/local/lib/libodbc.dylib -liconv > make[5]: *** [mod_amr.so] Error 1 > make[4]: *** [all] Error 1 > make[3]: *** [mod_amr-all] Error 1 > make[2]: *** [all-recursive] Error 1 > make[1]: *** [all-recursive] Error 1 > make: *** [all] Error 2 > > Heeding Mario's advice, I tried making with open-ssl flag, but that didn't help. If I go about commenting out offending modules, the same "bundle not allowed with -dynamiclib" appears for any number of other mods. I have installed all the prerequisites for Mac OSX (to my knowledge), including Apple and Unix dev tools. I am 100% macports/fink/brew free, AFAIK. Jira filed, but no response for a while: > > http://jira.freeswitch.org/browse/FS-4240 > > My sense this is a missing tool or config issue with my setup (pretty much fresh Lion install). Thanks in advance for your help! > Neil > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120525/d004d7ca/attachment.html From neilp at cs.stanford.edu Sat May 26 03:03:15 2012 From: neilp at cs.stanford.edu (Neil Patel) Date: Fri, 25 May 2012 16:03:15 -0700 Subject: [Freeswitch-users] Unable to build Git tree on Mac OSX Lion In-Reply-To: <9C746911-552E-4E7A-9C0C-DB0F5676A108@mgtech.com> References: <9C746911-552E-4E7A-9C0C-DB0F5676A108@mgtech.com> Message-ID: Hi Mario, This looks like a deeper problem. I tried commenting out mod_amr but then it broke on other codecs. The error says -bundle not allowed with -dyanmiclib. Seems like these flags are mutually exclusive. Is there a problem in how the FS is configuring the build? Thanks, Neil On Fri, May 25, 2012 at 3:53 PM, Mario G wrote: > The impression I get from the wiki is that mod_amr is optional. I would > try to comment it out in modules.conf in the FS source dir and see what > happens. I wish I had time to futz with FS on Lion but I am trying to > resolve 2 other FS issues for over a month with no luck. This is why I put > off updating the wiki, I won't until everything is perfect. > > On May 25, 2012, at 2:53 PM, Neil Patel wrote: > > Hi All, I am getting this error when trying to build latest git on Mac OSX > Lion, after fresh checkout and running ./boostraph.sh and ./configure: > > making all mod_amr > Creating mod_amr.so... > i686-apple-darwin11-llvm-gcc-4.2: -bundle not allowed with -dynamiclib > gcc -I/usr/local/src/freeswitch/libs/curl/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/libs/libteletone/src -Werror > -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb > -DMACOSX -DHAVE_OPENSSL -g -O2 -Wall -std=c99 -pedantic > -Wdeclaration-after-statement -D_GNU_SOURCE -DAMR_PASSTHROUGH -shared -o > .libs/mod_amr.so -dynamic -bundle -force-flat-namespace .libs/mod_amr.o > /usr/local/src/freeswitch/.libs/libfreeswitch.dylib -lodbc -lresolv > -L/usr/local/src/freeswitch/libs/apr-util/xml/expat/lib -lpq > /usr/local/src/freeswitch/libs/apr-util/xml/expat/lib/.libs/libexpat.a > /usr/local/src/freeswitch/libs/apr/.libs/libapr-1.a -lpthread -lm > -L/usr/local/src/freeswitch/libs/srtp -ldl -lssl -lcrypto -lz -lncurses > /usr/local/lib/libjpeg.dylib /usr/local/lib/libodbc.dylib -liconv > make[5]: *** [mod_amr.so] Error 1 > make[4]: *** [all] Error 1 > make[3]: *** [mod_amr-all] Error 1 > make[2]: *** [all-recursive] Error 1 > make[1]: *** [all-recursive] Error 1 > make: *** [all] Error 2 > > Heeding Mario's advice, I tried making with open-ssl flag, but that didn't > help. If I go about commenting out offending modules, the same "bundle not > allowed with -dynamiclib" appears for any number of other mods. I have > installed all the prerequisites for Mac OSX (to my knowledge), including > Apple and Unix dev tools. I am 100% macports/fink/brew free, AFAIK. Jira > filed, but no response for a while: > > http://jira.freeswitch.org/browse/FS-4240 > > My sense this is a missing tool or config issue with my setup (pretty much > fresh Lion install). Thanks in advance for your help! > Neil > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120525/6f3fd0fa/attachment.html From mario_fs at mgtech.com Sat May 26 03:50:50 2012 From: mario_fs at mgtech.com (Mario G) Date: Fri, 25 May 2012 16:50:50 -0700 Subject: [Freeswitch-users] Unable to build Git tree on Mac OSX Lion In-Reply-To: References: <9C746911-552E-4E7A-9C0C-DB0F5676A108@mgtech.com> Message-ID: This is not much help but I just updated with the current git and the make current went fine. However, I am on 10.6.8. Just out of curiosity I did a FIND on -bundle and found it was only used once with mod_lua, -dynamiclib was found only once but it looked to be referenced by several modules. I am not a linux developer so have little idea what the messages mean exactly. I found a few hits on the web, all involved source fixes, I know, not much help... Mario G On May 25, 2012, at 4:03 PM, Neil Patel wrote: > Hi Mario, > > This looks like a deeper problem. I tried commenting out mod_amr but then it broke on other codecs. > > The error says -bundle not allowed with -dyanmiclib. Seems like these flags are mutually exclusive. Is there a problem in how the FS is configuring the build? > > Thanks, > Neil > > On Fri, May 25, 2012 at 3:53 PM, Mario G wrote: > The impression I get from the wiki is that mod_amr is optional. I would try to comment it out in modules.conf in the FS source dir and see what happens. I wish I had time to futz with FS on Lion but I am trying to resolve 2 other FS issues for over a month with no luck. This is why I put off updating the wiki, I won't until everything is perfect. > > On May 25, 2012, at 2:53 PM, Neil Patel wrote: > >> Hi All, I am getting this error when trying to build latest git on Mac OSX Lion, after fresh checkout and running ./boostraph.sh and ./configure: >> >> making all mod_amr >> Creating mod_amr.so... >> i686-apple-darwin11-llvm-gcc-4.2: -bundle not allowed with -dynamiclib >> gcc -I/usr/local/src/freeswitch/libs/curl/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DMACOSX -DHAVE_OPENSSL -g -O2 -Wall -std=c99 -pedantic -Wdeclaration-after-statement -D_GNU_SOURCE -DAMR_PASSTHROUGH -shared -o .libs/mod_amr.so -dynamic -bundle -force-flat-namespace .libs/mod_amr.o /usr/local/src/freeswitch/.libs/libfreeswitch.dylib -lodbc -lresolv -L/usr/local/src/freeswitch/libs/apr-util/xml/expat/lib -lpq /usr/local/src/freeswitch/libs/apr-util/xml/expat/lib/.libs/libexpat.a /usr/local/src/freeswitch/libs/apr/.libs/libapr-1.a -lpthread -lm -L/usr/local/src/freeswitch/libs/srtp -ldl -lssl -lcrypto -lz -lncurses /usr/local/lib/libjpeg.dylib /usr/local/lib/libodbc.dylib -liconv >> make[5]: *** [mod_amr.so] Error 1 >> make[4]: *** [all] Error 1 >> make[3]: *** [mod_amr-all] Error 1 >> make[2]: *** [all-recursive] Error 1 >> make[1]: *** [all-recursive] Error 1 >> make: *** [all] Error 2 >> >> Heeding Mario's advice, I tried making with open-ssl flag, but that didn't help. If I go about commenting out offending modules, the same "bundle not allowed with -dynamiclib" appears for any number of other mods. I have installed all the prerequisites for Mac OSX (to my knowledge), including Apple and Unix dev tools. I am 100% macports/fink/brew free, AFAIK. Jira filed, but no response for a while: >> >> http://jira.freeswitch.org/browse/FS-4240 >> >> My sense this is a missing tool or config issue with my setup (pretty much fresh Lion install). Thanks in advance for your help! >> Neil >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120525/1f1bbe64/attachment-0001.html From krice at freeswitch.org Sat May 26 05:44:10 2012 From: krice at freeswitch.org (Ken Rice) Date: Fri, 25 May 2012 20:44:10 -0500 Subject: [Freeswitch-users] Unable to build Git tree on Mac OSX Lion In-Reply-To: Message-ID: Can you do a git bisect on this and figure out where it broke? This should build just fine... K On 5/25/12 6:50 PM, "Mario G" wrote: > This is not much help but I just updated with the current git and the make > current went fine. However, I am on 10.6.8. Just out of curiosity I did a FIND > on -bundle and found it was only used once with mod_lua, -dynamiclib was found > only once but it looked to be referenced by several modules. I am not a linux > developer so have little idea what the messages mean exactly. I found a few > hits on the web, all involved source fixes, I know, not much help... > Mario G > > On May 25, 2012, at 4:03 PM, Neil Patel wrote: > >> Hi Mario, >> >> This looks like a deeper problem. I tried commenting out mod_amr but then it >> broke on other codecs. >> >> The error says -bundle not allowed with -dyanmiclib. Seems like these flags >> are mutually exclusive. Is there a problem in how the FS is configuring the >> build? >> >> Thanks, >> Neil >> >> On Fri, May 25, 2012 at 3:53 PM, Mario G wrote: >>> The impression I get from the wiki is that mod_amr is optional. I would try >>> to comment it out in modules.conf in the FS source dir and see what happens. >>> I wish I had time to futz with FS on Lion but I am trying to resolve 2 other >>> FS issues for over a month with no luck. This is why I put off updating the >>> wiki, I won't until everything is perfect. >>> >>> On May 25, 2012, at 2:53 PM, Neil Patel wrote: >>> >>>> Hi All, I am getting this error when trying to build latest git on Mac OSX >>>> Lion, after fresh checkout and running ./boostraph.sh and ./configure: >>>> >>>> making all mod_amr >>>> Creating mod_amr.so... >>>> i686-apple-darwin11-llvm-gcc-4.2: -bundle not allowed with -dynamiclib >>>> gcc -I/usr/local/src/freeswitch/libs/curl/include >>>> -I/usr/local/src/freeswitch/src/include >>>> -I/usr/local/src/freeswitch/src/include >>>> -I/usr/local/src/freeswitch/libs/libteletone/src -Werror >>>> -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb >>>> -DMACOSX -DHAVE_OPENSSL -g -O2 -Wall -std=c99 -pedantic >>>> -Wdeclaration-after-statement -D_GNU_SOURCE -DAMR_PASSTHROUGH -shared -o >>>> .libs/mod_amr.so -dynamic -bundle -force-flat-namespace .libs/mod_amr.o >>>> /usr/local/src/freeswitch/.libs/libfreeswitch.dylib -lodbc -lresolv >>>> -L/usr/local/src/freeswitch/libs/apr-util/xml/expat/lib -lpq >>>> /usr/local/src/freeswitch/libs/apr-util/xml/expat/lib/.libs/libexpat.a >>>> /usr/local/src/freeswitch/libs/apr/.libs/libapr-1.a -lpthread -lm >>>> -L/usr/local/src/freeswitch/libs/srtp -ldl -lssl -lcrypto -lz -lncurses >>>> /usr/local/lib/libjpeg.dylib /usr/local/lib/libodbc.dylib -liconv >>>> make[5]: *** [mod_amr.so] Error 1 >>>> make[4]: *** [all] Error 1 >>>> make[3]: *** [mod_amr-all] Error 1 >>>> make[2]: *** [all-recursive] Error 1 >>>> make[1]: *** [all-recursive] Error 1 >>>> make: *** [all] Error 2 >>>> >>>> Heeding Mario's advice, I tried making with open-ssl flag, but that didn't >>>> help. If I go about commenting out offending modules, the same "bundle not >>>> allowed with -dynamiclib" appears for any number of other mods. I have >>>> installed all the prerequisites for Mac OSX (to my knowledge), including >>>> Apple and Unix dev tools. I am 100% macports/fink/brew free, AFAIK. Jira >>>> filed, but no response for a while: >>>> >>>> http://jira.freeswitch.org/browse/FS-4240 >>>> >>>> My sense this is a missing tool or config issue with my setup (pretty much >>>> fresh Lion install). Thanks in advance for your help! >>>> Neil >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> Join Us At ClueCon - Aug 7-9, 2012 >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at 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 at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> Join Us At ClueCon - Aug 7-9, 2012 >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120525/77ee38a5/attachment.html From dave at copycall.com Sat May 26 09:29:52 2012 From: dave at copycall.com (copycall) Date: Fri, 25 May 2012 22:29:52 -0700 Subject: [Freeswitch-users] freeswitch - sangoma CAS T1 - a101 configuration Message-ID: hi, i'm having trouble getting instructions to configure freeswitch with a sangoma a101 T1 card for CAS, B8ZS, ESF. the first application i seek to solve is using this setup as a gateway (replacing a dialogic dmg 2000) for sip from my itsp to the tdm endpoint: a legacy centigram - mitel voicemail server. any help will be greatly appreciated. thank you, dave -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120525/2333a691/attachment.html From peter.olsson at visionutveckling.se Sat May 26 16:41:36 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Sat, 26 May 2012 12:41:36 +0000 Subject: [Freeswitch-users] RTP media issue In-Reply-To: References: <02a201cd3a1a$2a658770$7f309650$@gmail.com> <007301cd3a96$d106ed90$7314c8b0$@gmail.com>, Message-ID: <1FFF97C269757C458224B7C895F35F150D9857@cantor.std.visionutv.se> I think the first step is to check the actual signalling. Just as Anthony mentions, the call setup looks weird. Between FS and the ATA (if I understood the different IP's/peers correctly) it seems FS sends early media (183), and it takes the ATA 10 seconds to ACK this. During the time the call is answered on the other end, which dosn't seem to be passed to the ATA (probably because it's so late with the ACK - I'm not sure about that though). So, first check this - it seems to me that FS might still be in "early media" state during the entire call. About the actual RTP, I see one problem here, it's packet 8146 in outbound.pcap. This is a packet that is probably generated by FS (since by that exact time, you're missing one packet from the provider), the problem here is that the timestamp if way off (I'm not sure where FS gets this ts from). It's because of this faulty timestamp that a few packets after this is dropped, since FS won't send packets with a lower timestamp than before. This is also why packet 8152 seems to have a strange timestamp, but it's actually just passed from the other leg, and is the first timestamp that is greater then packet 8146. One possible solution would be to enable rewrite of timestamps (rtp-rewrite-timestamps in the sofia config), also mentioned here: http://wiki.freeswitch.org/wiki/RTP_Issues#Dropped_Audio However, I think you also need to check the strangness in the SIP signalling. /Peter ________________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] för Mr Nathan Downes [nathandownes at hotmail.com] Skickat: den 26 maj 2012 00:03 Till: 'FreeSWITCH Users Help' ?mne: Re: [Freeswitch-users] RTP media issue Hi Anthony, FS version = FreeSWITCH Version 1.1.beta1 (git-f1b5044 2012-04-26 11-28-47 -0500) I don't have a debug log, but I could probably get it with another trace of both sides of the call, but it would be hard to capture as there is constant calls to this, unless there is a way to do it on a per call basis? I can also only do testing onsite as we don't have the same fibre equipment. I already have jitterbuffer set in both profiles in an attempt to try and stop it using , is there a way to set the cng_plc in the profile itself rather than diaplpan as there are 70 or so numbers in it. In the outbound dialplans I also added because I kept seeing PAUSE JITTERBUFFER in the FS logs when calls were made outbound so I wasn't sure it was doing something and read somewhere it pauses it when it bridges the call The inbound dialplan for all of those people consists of It doesn't affect SIP phones or normal ATA devices we have connected and only affects these FTTH GPON ATA's, but with almost 100 residents in this retirement village and them constantly complaining we have been given til Wednesday to come up with a solution or risk losing our position as the internet/phone provider for that retirement village. It appeared to me that what happens in that trace isn't normal behaviour, I did try rewriting timestamps last week, but as you suggested that appeared to mask the issue but not stop it from happening. That was when I was losing a packet from them each second or so, which by the time it arrived to end user sounded horrible. It has settled down a lot now and maybe 1 or 2 packets per call, but if what is in this trace is the cause each time, that would explain the poor end users experience. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Saturday, 26 May 2012 2:06 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] RTP media issue What version of FS are you running? Do you have the debug logs of those calls? you could try using the jitterbuffer. in the inbound DP to FS *before* you answer. Also it looks a little odd to me in this trace if this is the same call, it seems like you answered the call before placing the call to the phone and that phone never answers.... On Thu, May 24, 2012 at 9:51 PM, Nathan Downes wrote: > Hi, > > enable-soa > > > > Set the value to "false" to diable SIP SOA from sofia to tell sofia > not to touch the exchange of SDP > > I don't think this is related to the exchange of an SDP message.. Can > you elaborate more before I try it? I can't make things worse or > change things I don't understand. > > ________________________________ > From: djbinter at gmail.com > To: freeswitch-users at lists.freeswitch.org > CC: nathan at nortec.com.au > Subject: Re: [Freeswitch-users] RTP media issue > Date: Fri, 25 May 2012 11:19:46 +1000 > > > > > > Sent from my iPad > > On May 24, 2012, at 5:01 PM, Nathan Downes wrote: > > Hi, > > I had previous reported an issue with poor voice quality, appearing to > stem from occasion wrong timestamps coming from provider, but the end > user's experience was much worse than what I could see/hear in the trace. > > I have finally captured an event inbound and outbound. The thing I > don't understand is I thought even though FS proxied the media it > didn't touch it or change anything, but it appears it is. > > The 2 traces are http://www.nortec.com.au/inbound.pcap.gz and > http://www.nortec.com.au/outbound.pcap.gz > > Inbound is from my trunk provider to FS box and outbound is FS box to > ATA in FTTH GPON. > > The event I am talking about, if both traces are open, is in the > inbound one inbetween packet 8114 and 8117 the provider drops a packet > or I don't receive it. In the corresponding outbound trace, between > packet 8144 and 8152, it appears FS misses a whole heap of packets > (.1 seconds) between > 8146 and 8152 then it increases the timestamp only by 40 rather than > 160 on packet 8152. This seems to not affect SIP phones themselves > but causes issues with the FTTH GPON ATA. > > This causes a gap in the audio for the end user, and when they miss a > high number of packets even though it sounds good on the inbound trace > the end users experience is horrible. This trace is actually a good > one, but the wrong timestamp can occur once per second, causing end > user to lose 10%+ of incoming audio only. The issue only affects the > audio coming from provider to FS to end user. > > I am chasing it up with the voice provider to try and eliminate the > occasional packet loss, but if I could stop/fix FS from doing its > adjustment/gap/something the end user wouldn't even notice it. > > > > ______________________________________________________________________ > ___ > > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org > > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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 at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4fbfffef32764666710318! From bfmtl at hotmail.com Sat May 26 17:05:26 2012 From: bfmtl at hotmail.com (BF) Date: Sat, 26 May 2012 09:05:26 -0400 Subject: [Freeswitch-users] mod_stress Message-ID: Hello, I'd like to know what the "stress write" from mod_stress module does. I have hard time finding documentation or an example about mod_stress module. Anyone? Thank you. Eric From govoiper at gmail.com Sat May 26 18:52:19 2012 From: govoiper at gmail.com (SamyGo) Date: Sat, 26 May 2012 19:52:19 +0500 Subject: [Freeswitch-users] Freeswitch (SBC + Media) does not know that gateway phone is picked up In-Reply-To: <1337976359.52140.YahooMailClassic@web110804.mail.gq1.yahoo.com> References: <1337976359.52140.YahooMailClassic@web110804.mail.gq1.yahoo.com> Message-ID: Hey Sherrif, You might wanna resend the pcap file as I couldn;t find any single INVITE or any call in that capture. Please review. BR SaGo. On Sat, May 26, 2012 at 1:05 AM, Sherif Omran wrote: > Hello Sammy > > FS uses port 6090 and registers the GW. When a call comes, it rings the > extension but then gives a busy signal and FS keeps ringing till I cancel > the call from the GW. > > > > 1. recv 477 bytes from udp/[217.74.179.29]:5060 at 19:52:05.662344: > 2. > ------------------------------------------------------------------------ > 3. CANCEL sip:gw+trunk_1000 at 78.138.90.58:6090;transport=udp;gw=trunk_1000 > SIP/2.0 > 4. Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bKe567.13b0b9e.0 > 5. Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK16d1a257;rport= > 5060 > 6. From: "+41793940965" >;tag=as1dccab06 > 7. To: > 8. Call-ID: 5cd37edb776a6b3a35e9713a453a3425 at bluesip.net > 9. CSeq: 102 CANCEL > 10. User-Agent: blueSIP PSTN GW > 11. Max-Forwards: 69 > 12. Content-Length: 0 > 13. P-hint: USRLOC > > > > > --- On *Fri, 5/25/12, SamyGo * wrote: > > > From: SamyGo > Subject: Re: [Freeswitch-users] Freeswitch (SBC + Media) does not know > that gateway phone is picked up > To: "FreeSWITCH Users Help" > Date: Friday, May 25, 2012, 10:46 AM > > Hi again, > If you want kamailio register to the provider then use UAC > module. > Kamailio will use the username/password and register with the provider. > > Regards, > Sammy > > > On Fri, May 25, 2012 at 12:20 PM, Sherif Omran > > wrote: > > Dear Sammy, > > Thank you for your question ... Yes, the GW is registered through FS > because I did not know how to register it to kamailio. But it seems better > to register it to kamailio. > One more information, calls from 1001 to 1002 go to kamailio then to FS > then return back to kamailio smoothly. Thus I would suggest that I change > registering the gateway from FS to kamailio. but How to? > > > > --- On *Fri, 5/25/12, SamyGo > >* wrote: > > > From: SamyGo > > > Subject: Re: [Freeswitch-users] Freeswitch (SBC + Media) does not know > that gateway phone is picked up > To: "FreeSWITCH Users Help" > > > Date: Friday, May 25, 2012, 8:08 AM > > Hi, > These are SIP traces on FreeSWITCH console, whereas you are saying and it > seems that kamailio is not detecting the answering (200 OK)of the call from > extension 1002. Please, can you take a sip trace..!! I see you've both > kamailio and FS on same server! Please take a pcap from the linux console > using the following command. > > #tcpdump -i any -s 0 -w new-call-trace.pcap -vvv > > Please be quick on turning the sip trace on and off as quickly as possible > to avoid extra packets. Once done open the file in wireshark > ; apply filter "sip || rtp" and then save the > resulting capture in separate file. Send us the new file to analyse. > > One more silly question probably, I see REGISTERs coming to your FS as > well and the calls to gateways are made from FS too !!, umm...just thinking > what are you using kamailio for!!? > > Thanks > BR, > Sammy > > > On Fri, May 25, 2012 at 4:40 AM, Sherif Omran > > wrote: > > Hi all, > > here is the siptrace: To figure it out > 1- gateway called bluesip.net. It send invide using caller > number at bluesip.net > 2- This call should go to extension kb-1002. kb means go from freeswitch > port 6090 to kamailio port 5060 > 3- It should go to call extension 1002 in Kamailio > 4- Extension 1002 rings but when I reply, it does not notice I replied > > > ./fs_cli > _____ ____ ____ _ ___ > | ___/ ___| / ___| | |_ _| > | |_ \___ \ | | | | | | > | _| ___) | | |___| |___ | | > |_| |____/ \____|_____|___| > > ******************************************************* > * Anthony Minessale II, Ken Rice, * > * Michael Jerris, Travis Cross * > * FreeSWITCH (http://www.freeswitch.org) * > * Paypal Donations Appreciated: paypal at freeswitch.org* > * Brought to you by ClueCon http://www.cluecon.com/ * > ******************************************************* > > Type /help to see a list of commands > > > +OK log level [7] > freeswitch at internal> tracelevel > -ERR tracelevel Command not found! > > freeswitch at internal> sofia global siptrace on > +OK Global siptrace on > recv 1104 bytes from udp/[217.74.179.29]:5060 at 23:08:45.822962: > ------------------------------------------------------------------------ > INVITE sip:gw+trunk_1000 at 78.138.90.58:6090;transport=udp;gw=trunk_1000 > SIP/2.0 > Record-Route: > Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 > Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 > From: "+41793940965" ;tag=as00589402 > To: > Contact: > Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net > CSeq: 102 INVITE > User-Agent: blueSIP PSTN GW > Max-Forwards: 69 > Date: Thu, 24 May 2012 23:08:44 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Content-Type: application/sdp > Content-Length: 367 > P-hint: USRLOC > > v=0 > o=root 20076 20076 IN IP4 217.74.179.28 > s=session > c=IN IP4 217.74.179.28 > t=0 0 > m=audio 25626 RTP/AVP 8 0 18 111 3 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:111 G726-32/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > a=sendrecv > ------------------------------------------------------------------------ > send 494 bytes to udp/[217.74.179.29]:5060 at 23:08:45.844486: > ------------------------------------------------------------------------ > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 > Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 > Record-Route: > From: "+41793940965" ;tag=as00589402 > To: > Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net > CSeq: 102 INVITE > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Content-Length: 0 > > ------------------------------------------------------------------------ > 2012-05-24 23:08:45.840749 [NOTICE] switch_channel.c:926 New Channel > sofia/internal/+41793940965 at bluesip.net[69cd22e6-a5f5-11e1-a833-7b1f73a7ffcf] > 2012-05-24 23:08:45.840749 [DEBUG] sofia.c:5532 Channel sofia/internal/ > +41793940965 at bluesip.net entering state [received][100] > 2012-05-24 23:08:45.840749 [DEBUG] sofia.c:5543 Remote SDP: > v=0 > o=root 20076 20076 IN IP4 217.74.179.28 > s=session > c=IN IP4 217.74.179.28 > t=0 0 > m=audio 25626 RTP/AVP 8 0 18 111 3 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:111 G726-32/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [PCMA:8:8000:20:64000]/[iLBC:97:8000:30:13330] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [PCMU:0:8000:20:64000]/[iLBC:97:8000:30:13330] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [G729:18:8000:20:8000]/[iLBC:97:8000:30:13330] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [G726-32:111:8000:20:0]/[iLBC:97:8000:30:13330] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [GSM:3:8000:20:13200]/[iLBC:97:8000:30:13330] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [telephone-event:101:8000:20:0]/[iLBC:97:8000:30:13330] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf > send/recv payload to 101 > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [PCMA:8:8000:20:64000]/[G7221:115:32000:20:48000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [PCMU:0:8000:20:64000]/[G7221:115:32000:20:48000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [G729:18:8000:20:8000]/[G7221:115:32000:20:48000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [G726-32:111:8000:20:0]/[G7221:115:32000:20:48000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [GSM:3:8000:20:13200]/[G7221:115:32000:20:48000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [telephone-event:101:8000:20:0]/[G7221:115:32000:20:48000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf > send/recv payload to 101 > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [PCMA:8:8000:20:64000]/[G7221:107:16000:20:32000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [PCMU:0:8000:20:64000]/[G7221:107:16000:20:32000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [G729:18:8000:20:8000]/[G7221:107:16000:20:32000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [G726-32:111:8000:20:0]/[G7221:107:16000:20:32000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [GSM:3:8000:20:13200]/[G7221:107:16000:20:32000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [telephone-event:101:8000:20:0]/[G7221:107:16000:20:32000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf > send/recv payload to 101 > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [PCMA:8:8000:20:64000]/[G722:9:8000:20:64000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [G729:18:8000:20:8000]/[G722:9:8000:20:64000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [G726-32:111:8000:20:0]/[G722:9:8000:20:64000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [GSM:3:8000:20:13200]/[G722:9:8000:20:64000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [telephone-event:101:8000:20:0]/[G722:9:8000:20:64000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf > send/recv payload to 101 > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [PCMA:8:8000:20:64000]/[PCMU:0:8000:20:64000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:2996 Set Codec > sofia/internal/+41793940965 at bluesip.net PCMU/8000 20 ms 160 samples 64000 > bits > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_codec.c:111 sofia/internal/ > +41793940965 at bluesip.net Original read codec set to PCMU:0 > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf > send/recv payload to 101 > 2012-05-24 23:08:45.840749 [DEBUG] sofia.c:5757 (sofia/internal/ > +41793940965 at bluesip.net) State Change CS_NEW -> CS_INIT > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_INIT > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:401 > (sofia/internal/+41793940965 at bluesip.net) State INIT > 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:85 sofia/internal/ > +41793940965 at bluesip.net SOFIA INIT > 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:125 (sofia/internal/ > +41793940965 at bluesip.net) State Change CS_INIT -> CS_ROUTING > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:401 > (sofia/internal/+41793940965 at bluesip.net) State INIT going to sleep > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_ROUTING > 2012-05-24 23:08:45.840749 [DEBUG] switch_channel.c:1886 (sofia/internal/ > +41793940965 at bluesip.net) Callstate Change DOWN -> RINGING > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 > (sofia/internal/+41793940965 at bluesip.net) State ROUTING > 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:148 sofia/internal/ > +41793940965 at bluesip.net SOFIA ROUTING > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:104 > sofia/internal/+41793940965 at bluesip.net Standard ROUTING > 2012-05-24 23:08:45.840749 [INFO] mod_dialplan_xml.c:485 Processing > +41793940965 <+41793940965>->kb-1002 in context public > Dialplan: sofia/internal/+41793940965 at bluesip.net parsing > [public->from_kamailio] continue=false > Dialplan: sofia/internal/+41793940965 at bluesip.net Regex (PASS) > [from_kamailio] destination_number(kb-1002) =~ /^(.+)$/ break=on-false > Dialplan: sofia/internal/+41793940965 at bluesip.net Action transfer(kb-1002 > XML default) > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:154 > (sofia/internal/+41793940965 at bluesip.net) State Change CS_ROUTING -> > CS_EXECUTE > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 > (sofia/internal/+41793940965 at bluesip.net) State ROUTING going to sleep > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_EXECUTE > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:417 > (sofia/internal/+41793940965 at bluesip.net) State EXECUTE > 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:241 sofia/internal/ > +41793940965 at bluesip.net SOFIA EXECUTE > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:192 > sofia/internal/+41793940965 at bluesip.net Standard EXECUTE > EXECUTE sofia/internal/+41793940965 at bluesip.net transfer(kb-1002 XML > default) > 2012-05-24 23:08:45.840749 [DEBUG] switch_ivr.c:1711 (sofia/internal/ > +41793940965 at bluesip.net) State Change CS_EXECUTE -> CS_ROUTING > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:731 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:08:45.840749 [NOTICE] switch_ivr.c:1717 Transfer > sofia/internal/+41793940965 at bluesip.net to XML[kb-1002 at default] > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:417 > (sofia/internal/+41793940965 at bluesip.net) State EXECUTE going to sleep > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_ROUTING > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 > (sofia/internal/+41793940965 at bluesip.net) State ROUTING > 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:148 sofia/internal/ > +41793940965 at bluesip.net SOFIA ROUTING > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:104 > sofia/internal/+41793940965 at bluesip.net Standard ROUTING > 2012-05-24 23:08:45.840749 [INFO] mod_dialplan_xml.c:485 Processing > +41793940965 <+41793940965>->kb-1002 in context default > Dialplan: sofia/internal/+41793940965 at bluesip.net parsing [default->vbox] > continue=false > Dialplan: sofia/internal/+41793940965 at bluesip.net Regex (FAIL) [vbox] > destination_number(kb-1002) =~ /^vb-([0-9][0-9][0-9])$/ break=on-false > Dialplan: sofia/internal/+41793940965 at bluesip.net parsing > [default->vmenu] continue=false > Dialplan: sofia/internal/+41793940965 at bluesip.net Regex (FAIL) [vmenu] > destination_number(kb-1002) =~ /^vm-([0-9][0-9][0-9][0-9])$/ break=on-false > Dialplan: sofia/internal/+41793940965 at bluesip.net parsing > [default->kbridge] continue=false > Dialplan: sofia/internal/+41793940965 at bluesip.net Regex (PASS) [kbridge] > destination_number(kb-1002) =~ /^kb-(.+)$/ break=on-false > Dialplan: sofia/internal/+41793940965 at bluesip.net Action > set(proxy_media=true) > Dialplan: sofia/internal/+41793940965 at bluesip.net Action > set(call_timeout=50) > Dialplan: sofia/internal/+41793940965 at bluesip.net Action > set(continue_on_fail=true) > Dialplan: sofia/internal/+41793940965 at bluesip.net Action > set(hangup_after_bridge=true) > Dialplan: sofia/internal/+41793940965 at bluesip.net Action > set(sip_invite_domain=78.138.90.58) > Dialplan: sofia/internal/+41793940965 at bluesip.net Action > export(sip_contact_user=ufs) > Dialplan: sofia/internal/+41793940965 at bluesip.net Action bridge(sofia/ > 78.138.90.58/1002 at 78.138.90.58:5060) > Dialplan: sofia/internal/+41793940965 at bluesip.net Action answer() > Dialplan: sofia/internal/+41793940965 at bluesip.net Action > voicemail(default ${domain_name} 1002) > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:154 > (sofia/internal/+41793940965 at bluesip.net) State Change CS_ROUTING -> > CS_EXECUTE > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 > (sofia/internal/+41793940965 at bluesip.net) State ROUTING going to sleep > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_EXECUTE > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:417 > (sofia/internal/+41793940965 at bluesip.net) State EXECUTE > 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:241 sofia/internal/ > +41793940965 at bluesip.net SOFIA EXECUTE > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:192 > sofia/internal/+41793940965 at bluesip.net Standard EXECUTE > EXECUTE sofia/internal/+41793940965 at bluesip.net set(proxy_media=true) > 2012-05-24 23:08:45.840749 [DEBUG] mod_dptools.c:1281 sofia/internal/ > +41793940965 at bluesip.net SET [proxy_media]=[true] > EXECUTE sofia/internal/+41793940965 at bluesip.net set(call_timeout=50) > 2012-05-24 23:08:45.840749 [DEBUG] mod_dptools.c:1281 sofia/internal/ > +41793940965 at bluesip.net SET [call_timeout]=[50] > EXECUTE sofia/internal/+41793940965 at bluesip.net set(continue_on_fail=true) > 2012-05-24 23:08:45.840749 [DEBUG] mod_dptools.c:1281 sofia/internal/ > +41793940965 at bluesip.net SET [continue_on_fail]=[true] > EXECUTE sofia/internal/+41793940965 at bluesip.netset(hangup_after_bridge=true) > 2012-05-24 23:08:45.840749 [DEBUG] mod_dptools.c:1281 sofia/internal/ > +41793940965 at bluesip.net SET [hangup_after_bridge]=[true] > EXECUTE sofia/internal/+41793940965 at bluesip.netset(sip_invite_domain=78.138.90.58) > 2012-05-24 23:08:45.840749 [DEBUG] mod_dptools.c:1281 sofia/internal/ > +41793940965 at bluesip.net SET [sip_invite_domain]=[78.138.90.58] > EXECUTE sofia/internal/+41793940965 at bluesip.netexport(sip_contact_user=ufs) > 2012-05-24 23:08:45.840749 [DEBUG] switch_channel.c:1093 EXPORT > (export_vars) [sip_contact_user]=[ufs] > EXECUTE sofia/internal/+41793940965 at bluesip.net bridge(sofia/ > 78.138.90.58/1002 at 78.138.90.58:5060) > 2012-05-24 23:08:45.840749 [DEBUG] switch_channel.c:1047 sofia/internal/ > +41793940965 at bluesip.net EXPORTING[export_vars] [sip_contact_user]=[ufs] > to event > 2012-05-24 23:08:45.840749 [DEBUG] switch_ivr_originate.c:1884 Parsing > global variables > 2012-05-24 23:08:45.840749 [NOTICE] switch_channel.c:926 New Channel > sofia/internal/1002 at 78.138.90.58:5060[69ce1f0c-a5f5-11e1-a838-7b1f73a7ffcf] > 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:4691 (sofia/internal/ > 1002 at 78.138.90.58:5060) State Change CS_NEW -> CS_INIT > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/1002 at 78.138.90.58:5060) Running State Change CS_INIT > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:401 > (sofia/internal/1002 at 78.138.90.58:5060) State INIT > 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:85 sofia/internal/ > 1002 at 78.138.90.58:5060 SOFIA INIT > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:1871 sofia/internal/ > 1002 at 78.138.90.58:5060 Patched SDP > --- > v=0 > o=root 20076 20076 IN IP4 217.74.179.28 > s=session > c=IN IP4 217.74.179.28 > t=0 0 > m=audio 25626 RTP/AVP 8 0 18 111 3 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:111 G726-32/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > > +++ > v=0 > o=FreeSWITCH 1629510322 1629510323 IN IP4 78.138.90.58 > s=FreeSWITCH > c=IN IP4 78.138.90.58 > t=0 0 > m=audio 31178 RTP/AVP 8 0 18 111 3 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:111 G726-32/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > > send 1340 bytes to udp/[78.138.90.58]:5060 at 23:08:45.852280: > ------------------------------------------------------------------------ > INVITE sip:1002 at 78.138.90.58:5060 SIP/2.0 > Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKapDgFeegKyHXK > Max-Forwards: 67 > From: "+41793940965" ;tag=tgNrSS1jQFS8F > To: > Call-ID: 412c8312-2098-1230-8985-00163e6bb553 > CSeq: 28614398 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > 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: 372 > P-hint: USRLOC > X-FS-Support: update_display,send_info > Remote-Party-ID: "+41793940965" >;party=calling;screen=yes;privacy=off > > v=0 > o=FreeSWITCH 1629510322 1629510323 IN IP4 78.138.90.58 > s=FreeSWITCH > c=IN IP4 78.138.90.58 > t=0 0 > m=audio 31178 RTP/AVP 8 0 18 111 3 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:111 G726-32/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > ------------------------------------------------------------------------ > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:125 (sofia/internal/ > 1002 at 78.138.90.58:5060) State Change CS_INIT -> CS_ROUTING > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:401 > (sofia/internal/1002 at 78.138.90.58:5060) State INIT going to sleep > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/1002 at 78.138.90.58:5060) Running State Change CS_ROUTING > 2012-05-24 23:08:45.840749 [DEBUG] switch_channel.c:1886 (sofia/internal/ > 1002 at 78.138.90.58:5060) Callstate Change DOWN -> RINGING > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 > (sofia/internal/1002 at 78.138.90.58:5060) State ROUTING > 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:148 sofia/internal/ > 1002 at 78.138.90.58:5060 SOFIA ROUTING > 2012-05-24 23:08:45.840749 [DEBUG] switch_ivr_originate.c:66 > (sofia/internal/1002 at 78.138.90.58:5060) State Change CS_ROUTING -> > CS_CONSUME_MEDIA > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 > (sofia/internal/1002 at 78.138.90.58:5060) State ROUTING going to sleep > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/1002 at 78.138.90.58:5060) Running State Change > CS_CONSUME_MEDIA > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:429 > (sofia/internal/1002 at 78.138.90.58:5060) State CONSUME_MEDIA > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:429 > (sofia/internal/1002 at 78.138.90.58:5060) State CONSUME_MEDIA going to sleep > 2012-05-24 23:08:45.840749 [DEBUG] sofia.c:5532 Channel sofia/internal/ > 1002 at 78.138.90.58:5060 entering state [calling][0] > recv 365 bytes from udp/[78.138.90.58]:5060 at 23:08:45.964309: > ------------------------------------------------------------------------ > SIP/2.0 100 trying -- your call is important to us > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;branch=z9hG4bKapDgFeegKyHXK > From: "+41793940965" ;tag=tgNrSS1jQFS8F > To: > Call-ID: 412c8312-2098-1230-8985-00163e6bb553 > CSeq: 28614398 INVITE > Server: kamailio (3.1.5 (x86_64/linux)) > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 496 bytes from udp/[78.138.90.58]:5060 at 23:08:47.414766: > ------------------------------------------------------------------------ > SIP/2.0 180 Ringing > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;branch=z9hG4bKapDgFeegKyHXK > Record-Route: > Call-ID: 412c8312-2098-1230-8985-00163e6bb553 > From: "+41793940965" ;tag=tgNrSS1jQFS8F > To: > >;tag=qCSaB2YJOAsAr2dJ-RWNMhPoDdn1RFvs > CSeq: 28614398 INVITE > Contact: "Mama" > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, REGISTER > Content-Length: 0 > > ------------------------------------------------------------------------ > 2012-05-24 23:08:47.400741 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:47.400741 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:47.400741 [DEBUG] sofia.c:5532 Channel sofia/internal/ > 1002 at 78.138.90.58:5060 entering state [proceeding][180] > 2012-05-24 23:08:47.400741 [NOTICE] sofia.c:5624 Ring-Ready sofia/internal/ > 1002 at 78.138.90.58:5060! > send 995 bytes to udp/[217.74.179.29]:5060 at 23:08:47.416538: > ------------------------------------------------------------------------ > SIP/2.0 180 Ringing > Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 > Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 > Record-Route: > From: "+41793940965" ;tag=as00589402 > To: ;tag=S7UZQygFt62Nm > Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net > CSeq: 102 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Accept: application/sdp > 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-Length: 0 > Remote-Party-ID: "Outbound Call" > >;party=calling;privacy=off;screen=no > > ------------------------------------------------------------------------ > 2012-05-24 23:08:47.400741 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:08:47.400741 [NOTICE] mod_sofia.c:2514 Ring-Ready > sofia/internal/+41793940965 at bluesip.net! > 2012-05-24 23:08:47.400741 [DEBUG] sofia.c:5532 Channel sofia/internal/ > +41793940965 at bluesip.net entering state [early][180] > 2012-05-24 23:08:47.400741 [DEBUG] switch_core_session.c:731 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:08:47.400741 [NOTICE] switch_ivr_originate.c:483 Ring Ready > sofia/internal/+41793940965 at bluesip.net! > recv 496 bytes from udp/[78.138.90.58]:5060 at 23:08:47.804967: > ------------------------------------------------------------------------ > SIP/2.0 180 Ringing > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;branch=z9hG4bKapDgFeegKyHXK > Record-Route: > From: "+41793940965" ;tag=tgNrSS1jQFS8F > To: ;tag=549D2DD03BBA7C67 > Call-ID: 412c8312-2098-1230-8985-00163e6bb553 > CSeq: 28614398 INVITE > Contact: > ;uniq=6AC0DF4D2E498C8ACA82CB22226B9> > User-Agent: AVM FRITZ!Box Fon WLAN 7170 29.04.80 (Jan 27 2010) > Content-Length: 0 > > ------------------------------------------------------------------------ > 2012-05-24 23:08:47.800746 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:47.800746 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:47.800746 [DEBUG] sofia.c:5532 Channel sofia/internal/ > 1002 at 78.138.90.58:5060 entering state [proceeding][180] > recv 496 bytes from udp/[78.138.90.58]:5060 at 23:08:48.182098: > ------------------------------------------------------------------------ > SIP/2.0 180 Ringing > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;branch=z9hG4bKapDgFeegKyHXK > Record-Route: > Call-ID: 412c8312-2098-1230-8985-00163e6bb553 > From: "+41793940965" ;tag=tgNrSS1jQFS8F > To: > >;tag=qCSaB2YJOAsAr2dJ-RWNMhPoDdn1RFvs > CSeq: 28614398 INVITE > Contact: "Mama" > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, REGISTER > Content-Length: 0 > > ------------------------------------------------------------------------ > 2012-05-24 23:08:48.180741 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:48.180741 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:48.180741 [DEBUG] sofia.c:5532 Channel sofia/internal/ > 1002 at 78.138.90.58:5060 entering state [proceeding][180] > recv 496 bytes from udp/[78.138.90.58]:5060 at 23:08:48.417403: > ------------------------------------------------------------------------ > SIP/2.0 180 Ringing > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;branch=z9hG4bKapDgFeegKyHXK > Record-Route: > From: "+41793940965" ;tag=tgNrSS1jQFS8F > To: ;tag=549D2DD03BBA7C67 > Call-ID: 412c8312-2098-1230-8985-00163e6bb553 > CSeq: 28614398 INVITE > Contact: > ;uniq=6AC0DF4D2E498C8ACA82CB22226B9> > User-Agent: AVM FRITZ!Box Fon WLAN 7170 29.04.80 (Jan 27 2010) > Content-Length: 0 > > ------------------------------------------------------------------------ > 2012-05-24 23:08:48.400744 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:48.400744 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:48.400744 [DEBUG] sofia.c:5532 Channel sofia/internal/ > 1002 at 78.138.90.58:5060 entering state [proceeding][180] > recv 496 bytes from udp/[78.138.90.58]:5060 at 23:08:49.292599: > ------------------------------------------------------------------------ > SIP/2.0 180 Ringing > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;branch=z9hG4bKapDgFeegKyHXK > Record-Route: > From: "+41793940965" ;tag=tgNrSS1jQFS8F > To: ;tag=549D2DD03BBA7C67 > Call-ID: 412c8312-2098-1230-8985-00163e6bb553 > CSeq: 28614398 INVITE > Contact: > ;uniq=6AC0DF4D2E498C8ACA82CB22226B9> > User-Agent: AVM FRITZ!Box Fon WLAN 7170 29.04.80 (Jan 27 2010) > Content-Length: 0 > > ------------------------------------------------------------------------ > 2012-05-24 23:08:49.280742 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:49.280742 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:49.280742 [DEBUG] sofia.c:5532 Channel sofia/internal/ > 1002 at 78.138.90.58:5060 entering state [proceeding][180] > recv 478 bytes from udp/[217.74.179.29]:5060 at 23:09:15.066012: > ------------------------------------------------------------------------ > CANCEL sip:gw+trunk_1000 at 78.138.90.58:6090;transport=udp;gw=trunk_1000 > SIP/2.0 > Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 > Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 > From: "+41793940965" ;tag=as00589402 > To: > Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net > CSeq: 102 CANCEL > User-Agent: blueSIP PSTN GW > Max-Forwards: 69 > Content-Length: 0 > P-hint: USRLOC > > ------------------------------------------------------------------------ > send 369 bytes to udp/[217.74.179.29]:5060 at 23:09:15.066176: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 > Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 > From: "+41793940965" ;tag=as00589402 > To: ;tag=S7UZQygFt62Nm > Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net > CSeq: 102 CANCEL > Content-Length: 0 > > ------------------------------------------------------------------------ > send 766 bytes to udp/[217.74.179.29]:5060 at 23:09:15.066289: > ------------------------------------------------------------------------ > SIP/2.0 487 Request Terminated > Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 > Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 > From: "+41793940965" ;tag=as00589402 > To: ;tag=S7UZQygFt62Nm > Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net > CSeq: 102 INVITE > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > 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-Length: 0 > > ------------------------------------------------------------------------ > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:09:15.060742 [DEBUG] sofia.c:5532 Channel sofia/internal/ > +41793940965 at bluesip.net entering state [terminated][487] > 2012-05-24 23:09:15.060742 [DEBUG] switch_channel.c:2848 (sofia/internal/ > +41793940965 at bluesip.net) Callstate Change RINGING -> HANGUP > 2012-05-24 23:09:15.060742 [NOTICE] sofia.c:6301 Hangup sofia/internal/ > +41793940965 at bluesip.net [CS_EXECUTE] [ORIGINATOR_CANCEL] > 2012-05-24 23:09:15.060742 [DEBUG] switch_channel.c:2871 Send signal > sofia/internal/+41793940965 at bluesip.net [KILL] > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:09:15.060742 [DEBUG] switch_channel.c:2848 (sofia/internal/ > 1002 at 78.138.90.58:5060) Callstate Change RINGING -> HANGUP > 2012-05-24 23:09:15.060742 [NOTICE] switch_ivr_originate.c:3182 Hangup > sofia/internal/1002 at 78.138.90.58:5060 [CS_CONSUME_MEDIA] > [ORIGINATOR_CANCEL] > 2012-05-24 23:09:15.060742 [DEBUG] switch_channel.c:2871 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [KILL] > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/1002 at 78.138.90.58:5060) Running State Change CS_HANGUP > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:602 > (sofia/internal/1002 at 78.138.90.58:5060) State HANGUP > 2012-05-24 23:09:15.060742 [DEBUG] mod_sofia.c:469 Channel sofia/internal/ > 1002 at 78.138.90.58:5060 hanging up, cause: ORIGINATOR_CANCEL > 2012-05-24 23:09:15.060742 [DEBUG] mod_sofia.c:523 Sending CANCEL to > sofia/internal/1002 at 78.138.90.58:5060 > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:47 > sofia/internal/1002 at 78.138.90.58:5060 Standard HANGUP, cause: > ORIGINATOR_CANCEL > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:602 > (sofia/internal/1002 at 78.138.90.58:5060) State HANGUP going to sleep > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:393 > (sofia/internal/1002 at 78.138.90.58:5060) State Change CS_HANGUP -> > CS_REPORTING > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/1002 at 78.138.90.58:5060) Running State Change CS_REPORTING > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:662 > (sofia/internal/1002 at 78.138.90.58:5060) State REPORTING > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:79 > sofia/internal/1002 at 78.138.90.58:5060 Standard REPORTING, cause: > ORIGINATOR_CANCEL > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:662 > (sofia/internal/1002 at 78.138.90.58:5060) State REPORTING going to sleep > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:387 > (sofia/internal/1002 at 78.138.90.58:5060) State Change CS_REPORTING -> > CS_DESTROY > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1382 Session 9 > (sofia/internal/1002 at 78.138.90.58:5060) Locked, Waiting on external > entities > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:09:15.060742 [DEBUG] switch_ivr_originate.c:3358 Originate > Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL] > 2012-05-24 23:09:15.060742 [NOTICE] switch_core_session.c:1400 Session 9 > (sofia/internal/1002 at 78.138.90.58:5060) Ended > 2012-05-24 23:09:15.060742 [NOTICE] switch_core_session.c:1402 Close > Channel sofia/internal/1002 at 78.138.90.58:5060 [CS_DESTROY] > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:491 > (sofia/internal/1002 at 78.138.90.58:5060) Callstate Change HANGUP -> DOWN > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:494 > (sofia/internal/1002 at 78.138.90.58:5060) Running State Change CS_DESTROY > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:504 > (sofia/internal/1002 at 78.138.90.58:5060) State DESTROY > 2012-05-24 23:09:15.060742 [DEBUG] mod_sofia.c:374 sofia/internal/ > 1002 at 78.138.90.58:5060 SOFIA DESTROY > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:86 > sofia/internal/1002 at 78.138.90.58:5060 Standard DESTROY > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:504 > (sofia/internal/1002 at 78.138.90.58:5060) State DESTROY going to sleep > 2012-05-24 23:09:15.060742 [INFO] mod_dptools.c:2922 Originate Failed. > Cause: ORIGINATOR_CANCEL > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:2287 > sofia/internal/+41793940965 at bluesip.net skip receive message > [APPLICATION_EXEC_COMPLETE] (channel is hungup already) > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:417 > (sofia/internal/+41793940965 at bluesip.net) State EXECUTE going to sleep > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_HANGUP > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:602 > (sofia/internal/+41793940965 at bluesip.net) State HANGUP > 2012-05-24 23:09:15.060742 [DEBUG] mod_sofia.c:469 Channel sofia/internal/ > +41793940965 at bluesip.net hanging up, cause: ORIGINATOR_CANCEL > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:47 > sofia/internal/+41793940965 at bluesip.net Standard HANGUP, cause: > ORIGINATOR_CANCEL > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:602 > (sofia/internal/+41793940965 at bluesip.net) State HANGUP going to sleep > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:393 > (sofia/internal/+41793940965 at bluesip.net) State Change CS_HANGUP -> > CS_REPORTING > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/+41793940965 at bluesip.net) Running State Change > CS_REPORTING > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:662 > (sofia/internal/+41793940965 at bluesip.net) State REPORTING > send 383 bytes to udp/[78.138.90.58]:5060 at 23:09:15.087695: > ------------------------------------------------------------------------ > CANCEL sip:1002 at 78.138.90.58:5060 SIP/2.0 > Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKapDgFeegKyHXK > Max-Forwards: 67 > From: "+41793940965" ;tag=tgNrSS1jQFS8F > To: > Call-ID: 412c8312-2098-1230-8985-00163e6bb553 > CSeq: 28614398 CANCEL > Reason: FreeSWITCH;cause=487;text="ORIGINATOR_CANCEL" > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 408 bytes from udp/[217.74.179.29]:5060 at 23:09:15.094670: > ------------------------------------------------------------------------ > ACK sip:gw+trunk_1000 at 78.138.90.58:6090;transport=udp;gw=trunk_1000 > SIP/2.0 > Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 > From: "+41793940965" ;tag=as00589402 > Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net > To: ;tag=S7UZQygFt62Nm > CSeq: 102 ACK > User-Agent: Sip EXpress router(0.9.7 (i386/linux)) > Content-Length: 0 > > ------------------------------------------------------------------------ > 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:79 > sofia/internal/+41793940965 at bluesip.net Standard REPORTING, cause: > ORIGINATOR_CANCEL > 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:662 > (sofia/internal/+41793940965 at bluesip.net) State REPORTING going to sleep > 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:387 > (sofia/internal/+41793940965 at bluesip.net) State Change CS_REPORTING -> > CS_DESTROY > 2012-05-24 23:09:15.120807 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:09:15.120807 [DEBUG] switch_core_session.c:1382 Session 8 > (sofia/internal/+41793940965 at bluesip.net) Locked, Waiting on external > entities > 2012-05-24 23:09:15.120807 [NOTICE] switch_core_session.c:1400 Session 8 > (sofia/internal/+41793940965 at bluesip.net) Ended > 2012-05-24 23:09:15.120807 [NOTICE] switch_core_session.c:1402 Close > Channel sofia/internal/+41793940965 at bluesip.net [CS_DESTROY] > 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:491 > (sofia/internal/+41793940965 at bluesip.net) Callstate Change HANGUP -> DOWN > 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:494 > (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_DESTROY > 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:504 > (sofia/internal/+41793940965 at bluesip.net) State DESTROY > 2012-05-24 23:09:15.120807 [DEBUG] mod_sofia.c:374 sofia/internal/ > +41793940965 at bluesip.net SOFIA DESTROY > 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:86 > sofia/internal/+41793940965 at bluesip.net Standard DESTROY > 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:504 > (sofia/internal/+41793940965 at bluesip.net) State DESTROY going to sleep > recv 378 bytes from udp/[78.138.90.58]:5060 at 23:09:15.135235: > ------------------------------------------------------------------------ > SIP/2.0 200 canceling > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;branch=z9hG4bKapDgFeegKyHXK > From: "+41793940965" ;tag=tgNrSS1jQFS8F > To: >;tag=a6a1c5f60faecf035a1ae5b6e96e979a-8c5b > Call-ID: 412c8312-2098-1230-8985-00163e6bb553 > CSeq: 28614398 CANCEL > Server: kamailio (3.1.5 (x86_64/linux)) > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 446 bytes from udp/[78.138.90.58]:5060 at 23:09:16.419308: > ------------------------------------------------------------------------ > SIP/2.0 487 Request Terminated > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;branch=z9hG4bKapDgFeegKyHXK > Record-Route: > Call-ID: 412c8312-2098-1230-8985-00163e6bb553 > From: "+41793940965" ;tag=tgNrSS1jQFS8F > To: > >;tag=qCSaB2YJOAsAr2dJ-RWNMhPoDdn1RFvs > CSeq: 28614398 INVITE > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, REGISTER > Content-Length: 0 > > ------------------------------------------------------------------------ > send 354 bytes to udp/[78.138.90.58]:5060 at 23:09:16.419543: > ------------------------------------------------------------------------ > ACK sip:1002 at 78.138.90.58:5060 SIP/2.0 > Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKapDgFeegKyHXK > Max-Forwards: 67 > From: "+41793940965" ;tag=tgNrSS1jQFS8F > To: > >;tag=qCSaB2YJOAsAr2dJ-RWNMhPoDdn1RFvs > Call-ID: 412c8312-2098-1230-8985-00163e6bb553 > CSeq: 28614398 ACK > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 564 bytes from udp/[41.34.123.243]:58881 at 23:13:13.612418: > ------------------------------------------------------------------------ > REGISTER sip:sip.pcfone.com:6090 SIP/2.0 > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport;branch=z9hG4bKPjHQIpv.ACHGbXD4IHrMr72LwZ8kYdlu-J > Max-Forwards: 70 > From: "Sherif 1003" > >;tag=wOP.FVT4Crk8O5E.bB7BWUhNO6QG91IG > To: "Sherif 1003" > > > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60926 REGISTER > User-Agent: Telephone 1.0.2 > Contact: "Sherif 1003" > Expires: 300 > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, > REFER, MESSAGE, OPTIONS > Content-Length: 0 > > ------------------------------------------------------------------------ > send 744 bytes to udp/[41.34.123.243]:58881 at 23:13:13.613152: > ------------------------------------------------------------------------ > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport=58881;branch=z9hG4bKPjHQIpv.ACHGbXD4IHrMr72LwZ8kYdlu-J;received=41.34.123.243 > From: "Sherif 1003" > >;tag=wOP.FVT4Crk8O5E.bB7BWUhNO6QG91IG > To: "Sherif 1003" > >;tag=USeHUmjpmrFUB > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60926 REGISTER > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > WWW-Authenticate: Digest realm="sip.pcfone.com", > nonce="09674764-a5f6-11e1-a83c-7b1f73a7ffcf", algorithm=MD5, qop="auth" > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 830 bytes from udp/[41.34.123.243]:58881 at 23:13:13.847746: > ------------------------------------------------------------------------ > REGISTER sip:sip.pcfone.com:6090 SIP/2.0 > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport;branch=z9hG4bKPjFr7rIAdAJPRQd2GmrMrVib3KT3Rup0q4 > Max-Forwards: 70 > From: "Sherif 1003" > >;tag=wOP.FVT4Crk8O5E.bB7BWUhNO6QG91IG > To: "Sherif 1003" > > > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60927 REGISTER > User-Agent: Telephone 1.0.2 > Contact: "Sherif 1003" > Expires: 300 > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, > REFER, MESSAGE, OPTIONS > Authorization: Digest username="1002", realm="sip.pcfone.com", > nonce="09674764-a5f6-11e1-a83c-7b1f73a7ffcf", uri="sip:sip.pcfone.com:6090", > response="3bc0bb82a803ded3641704c4eb78d58f", algorithm=MD5, > cnonce="sc3AJfhuev18Bs-L-6tORmI.RXlxnPOz", qop=auth, nc=00000001 > Content-Length: 0 > > ------------------------------------------------------------------------ > send 706 bytes to udp/[41.34.123.243]:58881 at 23:13:13.850197: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport=58881;branch=z9hG4bKPjFr7rIAdAJPRQd2GmrMrVib3KT3Rup0q4;received=41.34.123.243 > From: "Sherif 1003" > >;tag=wOP.FVT4Crk8O5E.bB7BWUhNO6QG91IG > To: "Sherif 1003" > >;tag=v279vF3SH15DQ > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60927 REGISTER > Contact: ;expires=300 > Date: Thu, 24 May 2012 23:13:13 GMT > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Content-Length: 0 > > ------------------------------------------------------------------------ > send 989 bytes to udp/[41.34.123.243]:58881 at 23:13:13.888717: > ------------------------------------------------------------------------ > NOTIFY sip:1002 at 192.168.178.26:58881;ob SIP/2.0 > Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKBZ68g9yKg77FF > Route: ;ob > Max-Forwards: 70 > From: > >;tag=XB12yamXeav0j > To: > > > Call-ID: e0efa252-2098-1230-8985-00163e6bb553 > CSeq: 28614532 NOTIFY > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > 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=noresource > Content-Type: application/simple-message-summary > Content-Length: 91 > > Messages-Waiting: yes > Message-Account: sip:1002 at 78.138.90.58 > Voice-Message: 1/0 (0/0) > > ------------------------------------------------------------------------ > recv 308 bytes from udp/[41.34.123.243]:58881 at 23:13:14.298850: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;received=78.138.90.58;branch=z9hG4bKBZ68g9yKg77FF > Call-ID: e0efa252-2098-1230-8985-00163e6bb553 > From: > >;tag=XB12yamXeav0j > To: > >;tag=z9hG4bKBZ68g9yKg77FF > CSeq: 28614532 NOTIFY > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 564 bytes from udp/[41.34.123.243]:58881 at 23:18:09.067232: > ------------------------------------------------------------------------ > REGISTER sip:sip.pcfone.com:6090 SIP/2.0 > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport;branch=z9hG4bKPjABTslmZIhiHYRtLqQn2erwmX19MVYHFC > Max-Forwards: 70 > From: "Sherif 1003" > >;tag=JBSXL5WZLWbLzpvRFoTSP5x8qQslI1ST > To: "Sherif 1003" > > > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60928 REGISTER > User-Agent: Telephone 1.0.2 > Contact: "Sherif 1003" > Expires: 300 > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, > REFER, MESSAGE, OPTIONS > Content-Length: 0 > > ------------------------------------------------------------------------ > send 744 bytes to udp/[41.34.123.243]:58881 at 23:18:09.068121: > ------------------------------------------------------------------------ > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport=58881;branch=z9hG4bKPjABTslmZIhiHYRtLqQn2erwmX19MVYHFC;received=41.34.123.243 > From: "Sherif 1003" > >;tag=JBSXL5WZLWbLzpvRFoTSP5x8qQslI1ST > To: "Sherif 1003" > >;tag=ymtU0540BKjKe > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60928 REGISTER > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > WWW-Authenticate: Digest realm="sip.pcfone.com", > nonce="b9821df4-a5f6-11e1-a83d-7b1f73a7ffcf", algorithm=MD5, qop="auth" > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 830 bytes from udp/[41.34.123.243]:58881 at 23:18:09.302944: > ------------------------------------------------------------------------ > REGISTER sip:sip.pcfone.com:6090 SIP/2.0 > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport;branch=z9hG4bKPjRgtztcy63vMT97Jp5pH9n311li5EFeY- > Max-Forwards: 70 > From: "Sherif 1003" > >;tag=JBSXL5WZLWbLzpvRFoTSP5x8qQslI1ST > To: "Sherif 1003" > > > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60929 REGISTER > User-Agent: Telephone 1.0.2 > Contact: "Sherif 1003" > Expires: 300 > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, > REFER, MESSAGE, OPTIONS > Authorization: Digest username="1002", realm="sip.pcfone.com", > nonce="b9821df4-a5f6-11e1-a83d-7b1f73a7ffcf", uri="sip:sip.pcfone.com:6090", > response="519c6301816b51f3bbb98d97a347e92b", algorithm=MD5, > cnonce="sc3AJfhuev18Bs-L-6tORmI.RXlxnPOz", qop=auth, nc=00000001 > Content-Length: 0 > > ------------------------------------------------------------------------ > send 706 bytes to udp/[41.34.123.243]:58881 at 23:18:09.305217: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport=58881;branch=z9hG4bKPjRgtztcy63vMT97Jp5pH9n311li5EFeY-;received=41.34.123.243 > From: "Sherif 1003" > >;tag=JBSXL5WZLWbLzpvRFoTSP5x8qQslI1ST > To: "Sherif 1003" > >;tag=ZXKm20N48U85S > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60929 REGISTER > Contact: ;expires=300 > Date: Thu, 24 May 2012 23:18:09 GMT > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Content-Length: 0 > > ------------------------------------------------------------------------ > send 989 bytes to udp/[41.34.123.243]:58881 at 23:18:09.379987: > ------------------------------------------------------------------------ > NOTIFY sip:1002 at 192.168.178.26:58881;ob SIP/2.0 > Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKc8Z1j4FQDgy2a > Route: ;ob > Max-Forwards: 70 > From: > >;tag=06cD4U6754yrN > To: > > > Call-ID: 91100602-2099-1230-8985-00163e6bb553 > CSeq: 28614680 NOTIFY > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > 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=noresource > Content-Type: application/simple-message-summary > Content-Length: 91 > > Messages-Waiting: yes > Message-Account: sip:1002 at 78.138.90.58 > Voice-Message: 1/0 (0/0) > > ------------------------------------------------------------------------ > recv 308 bytes from udp/[41.34.123.243]:58881 at 23:18:09.773629: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;received=78.138.90.58;branch=z9hG4bKc8Z1j4FQDgy2a > Call-ID: 91100602-2099-1230-8985-00163e6bb553 > From: > >;tag=06cD4U6754yrN > To: > >;tag=z9hG4bKc8Z1j4FQDgy2a > CSeq: 28614680 NOTIFY > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 564 bytes from udp/[41.34.123.243]:58881 at 23:23:04.522455: > ------------------------------------------------------------------------ > REGISTER sip:sip.pcfone.com:6090 SIP/2.0 > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport;branch=z9hG4bKPjaCuKyaBi6ZqcsHyp5HhH3WthJStom0SK > Max-Forwards: 70 > From: "Sherif 1003" > >;tag=qfUt1fQWuE-Ro3zWld6X0r6WidtD6B1q > To: "Sherif 1003" > > > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60930 REGISTER > User-Agent: Telephone 1.0.2 > Contact: "Sherif 1003" > Expires: 300 > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, > REFER, MESSAGE, OPTIONS > Content-Length: 0 > > ------------------------------------------------------------------------ > send 744 bytes to udp/[41.34.123.243]:58881 at 23:23:04.523158: > ------------------------------------------------------------------------ > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport=58881;branch=z9hG4bKPjaCuKyaBi6ZqcsHyp5HhH3WthJStom0SK;received=41.34.123.243 > From: "Sherif 1003" > >;tag=qfUt1fQWuE-Ro3zWld6X0r6WidtD6B1q > To: "Sherif 1003" > >;tag=1F655pQB3DNBH > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60930 REGISTER > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > WWW-Authenticate: Digest realm="sip.pcfone.com", > nonce="699cfe2a-a5f7-11e1-a83e-7b1f73a7ffcf", algorithm=MD5, qop="auth" > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 830 bytes from udp/[41.34.123.243]:58881 at 23:23:04.757798: > ------------------------------------------------------------------------ > REGISTER sip:sip.pcfone.com:6090 SIP/2.0 > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport;branch=z9hG4bKPjs8SAufeoyzCLrePaAbYhZuOqQF47XIc- > Max-Forwards: 70 > From: "Sherif 1003" > >;tag=qfUt1fQWuE-Ro3zWld6X0r6WidtD6B1q > To: "Sherif 1003" > > > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60931 REGISTER > User-Agent: Telephone 1.0.2 > Contact: "Sherif 1003" > Expires: 300 > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, > REFER, MESSAGE, OPTIONS > Authorization: Digest username="1002", realm="sip.pcfone.com", > nonce="699cfe2a-a5f7-11e1-a83e-7b1f73a7ffcf", uri="sip:sip.pcfone.com:6090", > response="23a886934aaf049f08432b6bb226bd37", algorithm=MD5, > cnonce="sc3AJfhuev18Bs-L-6tORmI.RXlxnPOz", qop=auth, nc=00000001 > Content-Length: 0 > > ------------------------------------------------------------------------ > send 706 bytes to udp/[41.34.123.243]:58881 at 23:23:04.759924: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport=58881;branch=z9hG4bKPjs8SAufeoyzCLrePaAbYhZuOqQF47XIc-;received=41.34.123.243 > From: "Sherif 1003" > >;tag=qfUt1fQWuE-Ro3zWld6X0r6WidtD6B1q > To: "Sherif 1003" > >;tag=2rZy7H8e0pByc > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60931 REGISTER > Contact: ;expires=300 > Date: Thu, 24 May 2012 23:23:04 GMT > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Content-Length: 0 > > ------------------------------------------------------------------------ > send 989 bytes to udp/[41.34.123.243]:58881 at 23:23:04.872728: > ------------------------------------------------------------------------ > NOTIFY sip:1002 at 192.168.178.26:58881;ob SIP/2.0 > Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKDHStmZ0taSmNp > Route: ;ob > Max-Forwards: 70 > From: > >;tag=31rQ9cSjXZ1gr > To: > > > Call-ID: 4130a1ce-209a-1230-8985-00163e6bb553 > CSeq: 28614828 NOTIFY > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > 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=noresource > Content-Type: application/simple-message-summary > Content-Length: 91 > > Messages-Waiting: yes > Message-Account: sip:1002 at 78.138.90.58 > Voice-Message: 1/0 (0/0) > > ------------------------------------------------------------------------ > recv 308 bytes from udp/[41.34.123.243]:58881 at 23:23:05.269063: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;received=78.138.90.58;branch=z9hG4bKDHStmZ0taSmNp > Call-ID: 4130a1ce-209a-1230-8985-00163e6bb553 > From: > >;tag=31rQ9cSjXZ1gr > To: > >;tag=z9hG4bKDHStmZ0taSmNp > CSeq: 28614828 NOTIFY > Content-Length: 0 > > ------------------------------------------------------------------------ > send 916 bytes to udp/[217.74.179.29]:5060 at 23:25:45.873423: > ------------------------------------------------------------------------ > REGISTER sip:bluesip.net;transport=udp SIP/2.0 > Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKetjKptHy71a8H > Max-Forwards: 70 > From: > ;transport=udp>;tag=6r0vBQZS650Fg > To: > ;transport=udp> > Call-ID: 2a3e4696-a5b9-11e1-a7ed-7b1f73a7ffcf > CSeq: 28601493 REGISTER > Contact: ;transport=udp;gw=trunk_1000> > Expires: 3600 > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Authorization: Digest username="bluesip/sherifomran", realm=" > bluesip.net", nonce="4fbeba4ebfb7414d5c701bd6f994402d551f3acf", > algorithm=MD5, uri="sip:bluesip.net;transport=udp", > response="c6cdcafe0418e519bc9ee0d8fa3d4d74" > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 722 bytes from udp/[217.74.179.29]:5060 at 23:25:45.904176: > ------------------------------------------------------------------------ > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;branch=z9hG4bKetjKptHy71a8H > From: > ;transport=udp>;tag=6r0vBQZS650Fg > To: > ;transport=udp>;tag=0354a2e1b960c9cc2279eca4e5f84e20.1b16 > Call-ID: 2a3e4696-a5b9-11e1-a7ed-7b1f73a7ffcf > CSeq: 28601493 REGISTER > WWW-Authenticate: Digest realm="bluesip.net", > nonce="4fbec4a4ba82d1f1feaa32ccec01b85be054fcf7", stale=true > Server: Sip EXpress router (0.9.7 (i386/linux)) > Content-Length: 0 > Warning: 392 217.74.179.29:5060 "Noisy feedback tells: pid=5455 > req_src_ip=78.138.90.58 req_src_port=6090 in_uri=sip:bluesip.net;transport=udp > out_uri=sip:bluesip.net;transport=udp via_cnt==1" > > ------------------------------------------------------------------------ > send 916 bytes to udp/[217.74.179.29]:5060 at 23:25:45.904416: > ------------------------------------------------------------------------ > REGISTER sip:bluesip.net;transport=udp SIP/2.0 > Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKF3BcrN214a1tD > Max-Forwards: 70 > From: > ;transport=udp>;tag=6r0vBQZS650Fg > To: > ;transport=udp> > Call-ID: 2a3e4696-a5b9-11e1-a7ed-7b1f73a7ffcf > CSeq: 28601494 REGISTER > Contact: ;transport=udp;gw=trunk_1000> > Expires: 3600 > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Authorization: Digest username="bluesip/sherifomran", realm=" > bluesip.net", nonce="4fbec4a4ba82d1f1feaa32ccec01b85be054fcf7", > algorithm=MD5, uri="sip:bluesip.net;transport=udp", > response="4c09dbe4b9accb52d4104b40dfe20040" > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 699 bytes from udp/[217.74.179.29]:5060 at 23:25:45.935046: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;branch=z9hG4bKF3BcrN214a1tD > From: > ;transport=udp>;tag=6r0vBQZS650Fg > To: > ;transport=udp>;tag=0354a2e1b960c9cc2279eca4e5f84e20.2017 > Call-ID: 2a3e4696-a5b9-11e1-a7ed-7b1f73a7ffcf > CSeq: 28601494 REGISTER > Contact: ;transport=udp;gw=trunk_1000>;q=0.5;expires=3600 > Server: Sip EXpress router (0.9.7 (i386/linux)) > Content-Length: 0 > Warning: 392 217.74.179.29:5060 "Noisy feedback tells: pid=5462 > req_src_ip=78.138.90.58 req_src_port=6090 in_uri=sip:bluesip.net;transport=udp > out_uri=sip:bluesip.net;transport=udp via_cnt==1" > > ------------------------------------------------------------------------ > recv 564 bytes from udp/[41.34.123.243]:58881 at 23:27:59.977057: > ------------------------------------------------------------------------ > REGISTER sip:sip.pcfone.com:6090 SIP/2.0 > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport;branch=z9hG4bKPjEOP.a4NoXqoFqC7a.k2tyeX-mYlUyH4H > Max-Forwards: 70 > From: "Sherif 1003" > >;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K > To: "Sherif 1003" > > > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60932 REGISTER > User-Agent: Telephone 1.0.2 > Contact: "Sherif 1003" > Expires: 300 > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, > REFER, MESSAGE, OPTIONS > Content-Length: 0 > > ------------------------------------------------------------------------ > send 744 bytes to udp/[41.34.123.243]:58881 at 23:27:59.977960: > ------------------------------------------------------------------------ > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport=58881;branch=z9hG4bKPjEOP.a4NoXqoFqC7a.k2tyeX-mYlUyH4H;received=41.34.123.243 > From: "Sherif 1003" > >;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K > To: "Sherif 1003" > >;tag=4ajgB89Nt8Q3K > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60932 REGISTER > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > WWW-Authenticate: Digest realm="sip.pcfone.com", > nonce="19b7cc40-a5f8-11e1-a83f-7b1f73a7ffcf", algorithm=MD5, qop="auth" > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 564 bytes from udp/[41.34.123.243]:58881 at 23:28:00.468487: > ------------------------------------------------------------------------ > REGISTER sip:sip.pcfone.com:6090 SIP/2.0 > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport;branch=z9hG4bKPjEOP.a4NoXqoFqC7a.k2tyeX-mYlUyH4H > Max-Forwards: 70 > From: "Sherif 1003" > >;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K > To: "Sherif 1003" > > > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60932 REGISTER > User-Agent: Telephone 1.0.2 > Contact: "Sherif 1003" > Expires: 300 > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, > REFER, MESSAGE, OPTIONS > Content-Length: 0 > > ------------------------------------------------------------------------ > send 744 bytes to udp/[41.34.123.243]:58881 at 23:28:00.468679: > ------------------------------------------------------------------------ > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport=58881;branch=z9hG4bKPjEOP.a4NoXqoFqC7a.k2tyeX-mYlUyH4H;received=41.34.123.243 > From: "Sherif 1003" > >;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K > To: "Sherif 1003" > >;tag=4ajgB89Nt8Q3K > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60932 REGISTER > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > WWW-Authenticate: Digest realm="sip.pcfone.com", > nonce="19b7cc40-a5f8-11e1-a83f-7b1f73a7ffcf", algorithm=MD5, qop="auth" > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 830 bytes from udp/[41.34.123.243]:58881 at 23:28:00.717446: > ------------------------------------------------------------------------ > REGISTER sip:sip.pcfone.com:6090 SIP/2.0 > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport;branch=z9hG4bKPj0iHd3-XUjnvcBs9DiJUCWzrmYTmWxFLt > Max-Forwards: 70 > From: "Sherif 1003" > >;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K > To: "Sherif 1003" > > > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60933 REGISTER > User-Agent: Telephone 1.0.2 > Contact: "Sherif 1003" > Expires: 300 > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, > REFER, MESSAGE, OPTIONS > Authorization: Digest username="1002", realm="sip.pcfone.com", > nonce="19b7cc40-a5f8-11e1-a83f-7b1f73a7ffcf", uri="sip:sip.pcfone.com:6090", > response="e21c903e6d98b5adcdd2eefa75217157", algorithm=MD5, > cnonce="sc3AJfhuev18Bs-L-6tORmI.RXlxnPOz", qop=auth, nc=00000001 > Content-Length: 0 > > ------------------------------------------------------------------------ > send 706 bytes to udp/[41.34.123.243]:58881 at 23:28:00.719947: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport=58881;branch=z9hG4bKPj0iHd3-XUjnvcBs9DiJUCWzrmYTmWxFLt;received=41.34.123.243 > From: "Sherif 1003" > >;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K > To: "Sherif 1003" > >;tag=5KB9c3tSQHepF > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60933 REGISTER > Contact: ;expires=300 > Date: Thu, 24 May 2012 23:28:00 GMT > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Content-Length: 0 > > ------------------------------------------------------------------------ > send 989 bytes to udp/[41.34.123.243]:58881 at 23:28:00.868368: > ------------------------------------------------------------------------ > NOTIFY sip:1002 at 192.168.178.26:58881;ob SIP/2.0 > Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKgc54SgK51KQDS > Route: ;ob > Max-Forwards: 70 > From: > >;tag=6v41eyBXmt48a > To: > > > Call-ID: f19dfbce-209a-1230-8985-00163e6bb553 > CSeq: 28614976 NOTIFY > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > 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=noresource > Content-Type: application/simple-message-summary > Content-Length: 91 > > Messages-Waiting: yes > Message-Account: sip:1002 at 78.138.90.58 > Voice-Message: 1/0 (0/0) > > ------------------------------------------------------------------------ > recv 308 bytes from udp/[41.34.123.243]:58881 at 23:28:01.128279: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;received=78.138.90.58;branch=z9hG4bKgc54SgK51KQDS > Call-ID: f19dfbce-209a-1230-8985-00163e6bb553 > From: > >;tag=6v41eyBXmt48a > To: > >;tag=z9hG4bKgc54SgK51KQDS > CSeq: 28614976 NOTIFY > Content-Length: 0 > > ------------------------------------------------------------------------ > send 913 bytes to udp/[217.74.179.29]:5060 at 23:28:11.885213: > ------------------------------------------------------------------------ > REGISTER sip:bluesip.net;transport=udp SIP/2.0 > Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKHNyXUB48yvD0m > Max-Forwards: 70 > From: > ;transport=udp>;tag=5F739Uep9vaXm > To: > ;transport=udp> > Call-ID: 2a3e49c0-a5b9-11e1-a7ee-7b1f73a7ffcf > CSeq: 28601493 REGISTER > Contact: ;transport=udp;gw=trunk_1002> > Expires: 3600 > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Authorization: Digest username="bluesip/salahomran", realm="bluesip.net", > nonce="4fbebeca79a5cff7c417463814cb9054d313bcac", algorithm=MD5, uri="sip: > bluesip.net;transport=udp", response="bb1babb90f4ea1dc8fbb9aa45d6038c7" > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 720 bytes from udp/[217.74.179.29]:5060 at 23:28:11.915764: > ------------------------------------------------------------------------ > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;branch=z9hG4bKHNyXUB48yvD0m > From: > ;transport=udp>;tag=5F739Uep9vaXm > To: > ;transport=udp>;tag=0354a2e1b960c9cc2279eca4e5f84e20.3454 > Call-ID: 2a3e49c0-a5b9-11e1-a7ee-7b1f73a7ffcf > CSeq: 28601493 REGISTER > WWW-Authenticate: Digest realm="bluesip.net", > nonce="4fbec5365af3a174f1cc0b8cda770c727d25bf71", stale=true > Server: Sip EXpress router (0.9.7 (i386/linux)) > Content-Length: 0 > Warning: 392 217.74.179.29:5060 "Noisy feedback tells: pid=5462 > req_src_ip=78.138.90.58 req_src_port=6090 in_uri=sip:bluesip.net;transport=udp > out_uri=sip:bluesip.net;transport=udp via_cnt==1" > > ------------------------------------------------------------------------ > send 913 bytes to udp/[217.74.179.29]:5060 at 23:28:11.915914: > ------------------------------------------------------------------------ > REGISTER sip:bluesip.net;transport=udp SIP/2.0 > Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKjyQpX6mcv53jg > Max-Forwards: 70 > From: > ;transport=udp>;tag=5F739Uep9vaXm > To: > ;transport=udp> > Call-ID: 2a3e49c0-a5b9-11e1-a7ee-7b1f73a7ffcf > CSeq: 28601494 REGISTER > Contact: ;transport=udp;gw=trunk_1002> > Expires: 3600 > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Authorization: Digest username="bluesip/salahomran", realm="bluesip.net", > nonce="4fbec5365af3a174f1cc0b8cda770c727d25bf71", algorithm=MD5, uri="sip: > bluesip.net;transport=udp", response="e260f6dc9b01ce8bea8c5ffb36bd4ddd" > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 697 bytes from udp/[217.74.179.29]:5060 at 23:28:11.946325: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;branch=z9hG4bKjyQpX6mcv53jg > From: > ;transport=udp>;tag=5F739Uep9vaXm > To: > ;transport=udp>;tag=0354a2e1b960c9cc2279eca4e5f84e20.4b5c > Call-ID: 2a3e49c0-a5b9-11e1-a7ee-7b1f73a7ffcf > CSeq: 28601494 REGISTER > Contact: ;transport=udp;gw=trunk_1002>;q=0.5;expires=3600 > Server: Sip EXpress router (0.9.7 (i386/linux)) > Content-Length: 0 > Warning: 392 217.74.179.29:5060 "Noisy feedback tells: pid=5455 > req_src_ip=78.138.90.58 req_src_port=6090 in_uri=sip:bluesip.net;transport=udp > out_uri=sip:bluesip.net;transport=udp via_cnt==1" > > ------------------------------------------------------------------------ > recv 564 bytes from udp/[41.34.123.243]:58881 at 23:32:55.930149: > ------------------------------------------------------------------------ > REGISTER sip:sip.pcfone.com:6090 SIP/2.0 > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport;branch=z9hG4bKPjuk8DZGiaXfN7BmtslNHVjtSg5UXsbHXc > Max-Forwards: 70 > From: "Sherif 1003" > >;tag=Pz.t-0tGXj3saM.ehBiB1EAMmmitze1D > To: "Sherif 1003" > > > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60934 REGISTER > User-Agent: Telephone 1.0.2 > Contact: "Sherif 1003" > Expires: 300 > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, > REFER, MESSAGE, OPTIONS > Content-Length: 0 > > ------------------------------------------------------------------------ > send 744 bytes to udp/[41.34.123.243]:58881 at 23:32:55.931324: > ------------------------------------------------------------------------ > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport=58881;branch=z9hG4bKPjuk8DZGiaXfN7BmtslNHVjtSg5UXsbHXc;received=41.34.123.243 > From: "Sherif 1003" > >;tag=Pz.t-0tGXj3saM.ehBiB1EAMmmitze1D > To: "Sherif 1003" > >;tag=75XtgSv0H3tUp > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60934 REGISTER > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > WWW-Authenticate: Digest realm="sip.pcfone.com", > nonce="ca1eabee-a5f8-11e1-a840-7b1f73a7ffcf", algorithm=MD5, qop="auth" > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 830 bytes from udp/[41.34.123.243]:58881 at 23:32:56.165624: > ------------------------------------------------------------------------ > REGISTER sip:sip.pcfone.com:6090 SIP/2.0 > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport;branch=z9hG4bKPjKEQ6xXmyzZxA6vCM88vHNYAOPAjUqFsD > Max-Forwards: 70 > From: "Sherif 1003" > >;tag=Pz.t-0tGXj3saM.ehBiB1EAMmmitze1D > To: "Sherif 1003" > > > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60935 REGISTER > User-Agent: Telephone 1.0.2 > Contact: "Sherif 1003" > Expires: 300 > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, > REFER, MESSAGE, OPTIONS > Authorization: Digest username="1002", realm="sip.pcfone.com", > nonce="ca1eabee-a5f8-11e1-a840-7b1f73a7ffcf", uri="sip:sip.pcfone.com:6090", > response="ff6d12f4ec7abd4b37187222b3af2358", algorithm=MD5, > cnonce="sc3AJfhuev18Bs-L-6tORmI.RXlxnPOz", qop=auth, nc=00000001 > Content-Length: 0 > > ------------------------------------------------------------------------ > send 706 bytes to udp/[41.34.123.243]:58881 at 23:32:56.168033: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport=58881;branch=z9hG4bKPjKEQ6xXmyzZxA6vCM88vHNYAOPAjUqFsD;received=41.34.123.243 > From: "Sherif 1003" > >;tag=Pz.t-0tGXj3saM.ehBiB1EAMmmitze1D > To: "Sherif 1003" > >;tag=8eQKjmD4ecHej > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60935 REGISTER > Contact: ;expires=300 > Date: Thu, 24 May 2012 23:32:56 GMT > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Content-Length: 0 > > ------------------------------------------------------------------------ > send 989 bytes to udp/[41.34.123.243]:58881 at 23:32:56.360238: > ------------------------------------------------------------------------ > NOTIFY sip:1002 at 192.168.178.26:58881;ob SIP/2.0 > Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKK7gFZ15FSet5B > Route: ;ob > Max-Forwards: 70 > From: > >;tag=9QgcmFy7BN70D > To: > > > Call-ID: a1be7708-209b-1230-8985-00163e6bb553 > CSeq: 28615124 NOTIFY > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > 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=noresource > Content-Type: application/simple-message-summary > Content-Length: 91 > > Messages-Waiting: yes > Message-Account: sip:1002 at 78.138.90.58 > Voice-Message: 1/0 (0/0) > > ------------------------------------------------------------------------ > recv 308 bytes from udp/[41.34.123.243]:58881 at 23:32:56.623554: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;received=78.138.90.58;branch=z9hG4bKK7gFZ15FSet5B > Call-ID: a1be7708-209b-1230-8985-00163e6bb553 > From: > >;tag=9QgcmFy7BN70D > To: > >;tag=z9hG4bKK7gFZ15FSet5B > CSeq: 28615124 NOTIFY > Content-Length: 0 > > ------------------------------------------------------------------------ > freeswitch at internal> > > > --- On *Thu, 5/24/12, Kristian Kielhofner > >* wrote: > > > From: Kristian Kielhofner > > > Subject: Re: [Freeswitch-users] Freeswitch (SBC + Media) does not know > that gateway phone is picked up > To: "FreeSWITCH Users Help" > > > Date: Thursday, May 24, 2012, 7:51 PM > > > Siptrace and logs please. > > On Thu, May 24, 2012 at 11:53 AM, Sherif Omran > > > wrote: > > > > Hi all, > > > > My topology is as follows: > > > > Kamailio -> FS (SBS+Media server) > > > > I came across an issue with my system as follows. I have a Hardphone > registered. When I do local call inside kamailio, it gets to FS and returns > back well and FS understands when I lift the handset. However, I added a > gateway (german landline server), when I call my self from another phone, > the call gets to FS and then transmits to Kamailio, it rings my extension > but when I lift the handset FS does not notice it and keeps ringing. > > > > Any body has an Idea? Here is my gateway trunk. > > > > > > > > > > > > > > > > > > > > > > thanks in advance > > Sherif Omran > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > Kristian Kielhofner > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -----Inline Attachment Follows----- > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -----Inline Attachment Follows----- > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120526/8a7e9479/attachment-0001.html From sherifomran2000 at yahoo.com Sat May 26 19:06:24 2012 From: sherifomran2000 at yahoo.com (Sherif Omran) Date: Sat, 26 May 2012 08:06:24 -0700 (PDT) Subject: [Freeswitch-users] Freeswitch (SBC + Media) does not know that gateway phone is picked up In-Reply-To: Message-ID: <1338044784.74842.YahooMailClassic@web110809.mail.gq1.yahoo.com> Hi Sanny, Please see the following links http://pastebin.freeswitch.org/19175 http://pastebin.freeswitch.org/19174 regards, Sherif --- On Sat, 5/26/12, SamyGo wrote: From: SamyGo Subject: Re: [Freeswitch-users] Freeswitch (SBC + Media) does not know that gateway phone is picked up To: "FreeSWITCH Users Help" Date: Saturday, May 26, 2012, 5:52 PM Hey Sherrif,You might wanna resend the pcap file as I couldn;t find any single INVITE or any call in that capture. Please review. BRSaGo. On Sat, May 26, 2012 at 1:05 AM, Sherif Omran wrote: Hello Sammy FS uses port 6090 and registers the GW. When a call comes, it rings the extension but then gives a busy signal and FS keeps ringing till I cancel the call from the GW. recv 477 bytes from udp/[217.74.179.29]:5060 at 19:52:05.662344: ? ?------------------------------------------------------------------------? ?CANCEL sip:gw+trunk_1000 at 78.138.90.58:6090;transport=udp;gw=trunk_1000 SIP/2.0 ? ?Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bKe567.13b0b9e.0? ?Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK16d1a257;rport=5060 ? ?From: "+41793940965" ;tag=as1dccab06 ? ?To: ? ?Call-ID: 5cd37edb776a6b3a35e9713a453a3425 at bluesip.net? ?CSeq: 102 CANCEL? ?User-Agent: blueSIP PSTN GW ? ?Max-Forwards: 69? ?Content-Length: 0? ?P-hint: USRLOC --- On Fri, 5/25/12, SamyGo wrote: From: SamyGo Subject: Re: [Freeswitch-users] Freeswitch (SBC + Media) does not know that gateway phone is picked up To: "FreeSWITCH Users Help" Date: Friday, May 25, 2012, 10:46 AM Hi again,If you want kamailio register to the provider then use UAC module. Kamailio will use the username/password and register with the provider. Regards,Sammy On Fri, May 25, 2012 at 12:20 PM, Sherif Omran wrote: Dear Sammy, Thank you for your question ... Yes, the GW is registered through FS because I did not know how to register it to kamailio. But it seems better to register it to kamailio. One more information, calls from 1001 to 1002 go to kamailio then to FS then return back to kamailio smoothly. Thus I would suggest that I change registering the gateway from FS to kamailio. but How to? --- On Fri, 5/25/12, SamyGo wrote: From: SamyGo Subject: Re: [Freeswitch-users] Freeswitch (SBC + Media) does not know that gateway phone is picked up To: "FreeSWITCH Users Help" Date: Friday, May 25, 2012, 8:08 AM Hi,These are SIP traces on FreeSWITCH console, whereas you are saying and it seems that kamailio is not detecting the answering (200 OK)of the call from extension 1002. Please, can you take a sip trace..!! I see you've both kamailio and FS on same server! Please take a pcap from the linux console using the following command. #tcpdump -i any -s 0 -w new-call-trace.pcap -vvv Please be quick on turning the sip trace on and off as quickly as possible to avoid extra packets. Once done open the file in wireshark ;?apply filter "sip || rtp" and then save the resulting capture in separate file. Send us the new file to analyse. One more silly question probably, I see REGISTERs coming to your FS as well and the calls to gateways are made from FS too !!, umm...just thinking what are you using kamailio for!!? ThanksBR,Sammy On Fri, May 25, 2012 at 4:40 AM, Sherif Omran wrote: Hi all, here is the siptrace: To figure it out 1- gateway called bluesip.net. It send invide using caller number at bluesip.net 2- This call should go to extension kb-1002. kb means go from freeswitch port 6090 to kamailio port 5060 3- It should go to call extension 1002 in Kamailio 4- Extension 1002 rings but when I reply, it does not notice I replied ./fs_cli ??????????? _____ ____???? ____ _???? ___????????????? ?????????? |? ___/ ___|?? / ___| |?? |_ _|???????????? ?????????? | |_? \___ \? | |?? | |??? | |??????????? ?????????? |? _|? ___) | | |___| |___ | |????????????? ?????????? |_|?? |____/?? \____|_____|___|??????????? ******************************************************* * Anthony Minessale II, Ken Rice,???????????????????? * * Michael Jerris, Travis Cross??????????????????????? * * FreeSWITCH (http://www.freeswitch.org)????????????? * * Paypal Donations Appreciated: paypal at freeswitch.org * * Brought to you by ClueCon http://www.cluecon.com/?? * ******************************************************* Type /help to see a list of commands +OK log level? [7] freeswitch at internal> tracelevel -ERR tracelevel Command not found! freeswitch at internal> sofia global siptrace on +OK Global siptrace on recv 1104 bytes from udp/[217.74.179.29]:5060 at 23:08:45.822962: ?? ------------------------------------------------------------------------ ?? INVITE sip:gw+trunk_1000 at 78.138.90.58:6090;transport=udp;gw=trunk_1000 SIP/2.0 ?? Record-Route: ?? Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 ?? Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 ?? From: "+41793940965" ;tag=as00589402 ?? To: ?? Contact: ?? Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net ?? CSeq: 102 INVITE ?? User-Agent: blueSIP PSTN GW ?? Max-Forwards: 69 ?? Date: Thu, 24 May 2012 23:08:44 GMT ?? Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY ?? Supported: replaces ?? Content-Type: application/sdp ?? Content-Length: 367 ?? P-hint: USRLOC ?? ?? v=0 ?? o=root 20076 20076 IN IP4 217.74.179.28 ?? s=session ?? c=IN IP4 217.74.179.28 ?? t=0 0 ?? m=audio 25626 RTP/AVP 8 0 18 111 3 101 ?? a=rtpmap:8 PCMA/8000 ?? a=rtpmap:0 PCMU/8000 ?? a=rtpmap:18 G729/8000 ?? a=fmtp:18 annexb=no ?? a=rtpmap:111 G726-32/8000 ?? a=rtpmap:3 GSM/8000 ?? a=rtpmap:101 telephone-event/8000 ?? a=fmtp:101 0-16 ?? a=silenceSupp:off - - - - ?? a=ptime:20 ?? a=sendrecv ?? ------------------------------------------------------------------------ send 494 bytes to udp/[217.74.179.29]:5060 at 23:08:45.844486: ?? ------------------------------------------------------------------------ ?? SIP/2.0 100 Trying ?? Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 ?? Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 ?? Record-Route: ?? From: "+41793940965" ;tag=as00589402 ?? To: ?? Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net ?? CSeq: 102 INVITE ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ 2012-05-24 23:08:45.840749 [NOTICE] switch_channel.c:926 New Channel sofia/internal/+41793940965 at bluesip.net [69cd22e6-a5f5-11e1-a833-7b1f73a7ffcf] 2012-05-24 23:08:45.840749 [DEBUG] sofia.c:5532 Channel sofia/internal/+41793940965 at bluesip.net entering state [received][100] 2012-05-24 23:08:45.840749 [DEBUG] sofia.c:5543 Remote SDP: v=0 o=root 20076 20076 IN IP4 217.74.179.28 s=session c=IN IP4 217.74.179.28 t=0 0 m=audio 25626 RTP/AVP 8 0 18 111 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:111 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMA:8:8000:20:64000]/[iLBC:97:8000:30:13330] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMU:0:8000:20:64000]/[iLBC:97:8000:30:13330] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G729:18:8000:20:8000]/[iLBC:97:8000:30:13330] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G726-32:111:8000:20:0]/[iLBC:97:8000:30:13330] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [GSM:3:8000:20:13200]/[iLBC:97:8000:30:13330] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [telephone-event:101:8000:20:0]/[iLBC:97:8000:30:13330] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf send/recv payload to 101 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMA:8:8000:20:64000]/[G7221:115:32000:20:48000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMU:0:8000:20:64000]/[G7221:115:32000:20:48000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G729:18:8000:20:8000]/[G7221:115:32000:20:48000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G726-32:111:8000:20:0]/[G7221:115:32000:20:48000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [GSM:3:8000:20:13200]/[G7221:115:32000:20:48000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [telephone-event:101:8000:20:0]/[G7221:115:32000:20:48000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf send/recv payload to 101 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMA:8:8000:20:64000]/[G7221:107:16000:20:32000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMU:0:8000:20:64000]/[G7221:107:16000:20:32000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G729:18:8000:20:8000]/[G7221:107:16000:20:32000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G726-32:111:8000:20:0]/[G7221:107:16000:20:32000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [GSM:3:8000:20:13200]/[G7221:107:16000:20:32000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [telephone-event:101:8000:20:0]/[G7221:107:16000:20:32000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf send/recv payload to 101 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMA:8:8000:20:64000]/[G722:9:8000:20:64000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G729:18:8000:20:8000]/[G722:9:8000:20:64000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G726-32:111:8000:20:0]/[G722:9:8000:20:64000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [GSM:3:8000:20:13200]/[G722:9:8000:20:64000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [telephone-event:101:8000:20:0]/[G722:9:8000:20:64000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf send/recv payload to 101 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMA:8:8000:20:64000]/[PCMU:0:8000:20:64000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:2996 Set Codec sofia/internal/+41793940965 at bluesip.net PCMU/8000 20 ms 160 samples 64000 bits 2012-05-24 23:08:45.840749 [DEBUG] switch_core_codec.c:111 sofia/internal/+41793940965 at bluesip.net Original read codec set to PCMU:0 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf send/recv payload to 101 2012-05-24 23:08:45.840749 [DEBUG] sofia.c:5757 (sofia/internal/+41793940965 at bluesip.net) State Change CS_NEW -> CS_INIT 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_INIT 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/+41793940965 at bluesip.net) State INIT 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:85 sofia/internal/+41793940965 at bluesip.net SOFIA INIT 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:125 (sofia/internal/+41793940965 at bluesip.net) State Change CS_INIT -> CS_ROUTING 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/+41793940965 at bluesip.net) State INIT going to sleep 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_ROUTING 2012-05-24 23:08:45.840749 [DEBUG] switch_channel.c:1886 (sofia/internal/+41793940965 at bluesip.net) Callstate Change DOWN -> RINGING 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/+41793940965 at bluesip.net) State ROUTING 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:148 sofia/internal/+41793940965 at bluesip.net SOFIA ROUTING 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:104 sofia/internal/+41793940965 at bluesip.net Standard ROUTING 2012-05-24 23:08:45.840749 [INFO] mod_dialplan_xml.c:485 Processing +41793940965 <+41793940965>->kb-1002 in context public Dialplan: sofia/internal/+41793940965 at bluesip.net parsing [public->from_kamailio] continue=false Dialplan: sofia/internal/+41793940965 at bluesip.net Regex (PASS) [from_kamailio] destination_number(kb-1002) =~ /^(.+)$/ break=on-false Dialplan: sofia/internal/+41793940965 at bluesip.net Action transfer(kb-1002 XML default) 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:154 (sofia/internal/+41793940965 at bluesip.net) State Change CS_ROUTING -> CS_EXECUTE 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/+41793940965 at bluesip.net) State ROUTING going to sleep 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_EXECUTE 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/+41793940965 at bluesip.net) State EXECUTE 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:241 sofia/internal/+41793940965 at bluesip.net SOFIA EXECUTE 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:192 sofia/internal/+41793940965 at bluesip.net Standard EXECUTE EXECUTE sofia/internal/+41793940965 at bluesip.net transfer(kb-1002 XML default) 2012-05-24 23:08:45.840749 [DEBUG] switch_ivr.c:1711 (sofia/internal/+41793940965 at bluesip.net) State Change CS_EXECUTE -> CS_ROUTING 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:731 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:08:45.840749 [NOTICE] switch_ivr.c:1717 Transfer sofia/internal/+41793940965 at bluesip.net to XML[kb-1002 at default] 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/+41793940965 at bluesip.net) State EXECUTE going to sleep 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_ROUTING 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/+41793940965 at bluesip.net) State ROUTING 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:148 sofia/internal/+41793940965 at bluesip.net SOFIA ROUTING 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:104 sofia/internal/+41793940965 at bluesip.net Standard ROUTING 2012-05-24 23:08:45.840749 [INFO] mod_dialplan_xml.c:485 Processing +41793940965 <+41793940965>->kb-1002 in context default Dialplan: sofia/internal/+41793940965 at bluesip.net parsing [default->vbox] continue=false Dialplan: sofia/internal/+41793940965 at bluesip.net Regex (FAIL) [vbox] destination_number(kb-1002) =~ /^vb-([0-9][0-9][0-9])$/ break=on-false Dialplan: sofia/internal/+41793940965 at bluesip.net parsing [default->vmenu] continue=false Dialplan: sofia/internal/+41793940965 at bluesip.net Regex (FAIL) [vmenu] destination_number(kb-1002) =~ /^vm-([0-9][0-9][0-9][0-9])$/ break=on-false Dialplan: sofia/internal/+41793940965 at bluesip.net parsing [default->kbridge] continue=false Dialplan: sofia/internal/+41793940965 at bluesip.net Regex (PASS) [kbridge] destination_number(kb-1002) =~ /^kb-(.+)$/ break=on-false Dialplan: sofia/internal/+41793940965 at bluesip.net Action set(proxy_media=true) Dialplan: sofia/internal/+41793940965 at bluesip.net Action set(call_timeout=50) Dialplan: sofia/internal/+41793940965 at bluesip.net Action set(continue_on_fail=true) Dialplan: sofia/internal/+41793940965 at bluesip.net Action set(hangup_after_bridge=true) Dialplan: sofia/internal/+41793940965 at bluesip.net Action set(sip_invite_domain=78.138.90.58) Dialplan: sofia/internal/+41793940965 at bluesip.net Action export(sip_contact_user=ufs) Dialplan: sofia/internal/+41793940965 at bluesip.net Action bridge(sofia/78.138.90.58/1002 at 78.138.90.58:5060) Dialplan: sofia/internal/+41793940965 at bluesip.net Action answer() Dialplan: sofia/internal/+41793940965 at bluesip.net Action voicemail(default ${domain_name} 1002) 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:154 (sofia/internal/+41793940965 at bluesip.net) State Change CS_ROUTING -> CS_EXECUTE 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/+41793940965 at bluesip.net) State ROUTING going to sleep 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_EXECUTE 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/+41793940965 at bluesip.net) State EXECUTE 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:241 sofia/internal/+41793940965 at bluesip.net SOFIA EXECUTE 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:192 sofia/internal/+41793940965 at bluesip.net Standard EXECUTE EXECUTE sofia/internal/+41793940965 at bluesip.net set(proxy_media=true) 2012-05-24 23:08:45.840749 [DEBUG] mod_dptools.c:1281 sofia/internal/+41793940965 at bluesip.net SET [proxy_media]=[true] EXECUTE sofia/internal/+41793940965 at bluesip.net set(call_timeout=50) 2012-05-24 23:08:45.840749 [DEBUG] mod_dptools.c:1281 sofia/internal/+41793940965 at bluesip.net SET [call_timeout]=[50] EXECUTE sofia/internal/+41793940965 at bluesip.net set(continue_on_fail=true) 2012-05-24 23:08:45.840749 [DEBUG] mod_dptools.c:1281 sofia/internal/+41793940965 at bluesip.net SET [continue_on_fail]=[true] EXECUTE sofia/internal/+41793940965 at bluesip.net set(hangup_after_bridge=true) 2012-05-24 23:08:45.840749 [DEBUG] mod_dptools.c:1281 sofia/internal/+41793940965 at bluesip.net SET [hangup_after_bridge]=[true] EXECUTE sofia/internal/+41793940965 at bluesip.net set(sip_invite_domain=78.138.90.58) 2012-05-24 23:08:45.840749 [DEBUG] mod_dptools.c:1281 sofia/internal/+41793940965 at bluesip.net SET [sip_invite_domain]=[78.138.90.58] EXECUTE sofia/internal/+41793940965 at bluesip.net export(sip_contact_user=ufs) 2012-05-24 23:08:45.840749 [DEBUG] switch_channel.c:1093 EXPORT (export_vars) [sip_contact_user]=[ufs] EXECUTE sofia/internal/+41793940965 at bluesip.net bridge(sofia/78.138.90.58/1002 at 78.138.90.58:5060) 2012-05-24 23:08:45.840749 [DEBUG] switch_channel.c:1047 sofia/internal/+41793940965 at bluesip.net EXPORTING[export_vars] [sip_contact_user]=[ufs] to event 2012-05-24 23:08:45.840749 [DEBUG] switch_ivr_originate.c:1884 Parsing global variables 2012-05-24 23:08:45.840749 [NOTICE] switch_channel.c:926 New Channel sofia/internal/1002 at 78.138.90.58:5060 [69ce1f0c-a5f5-11e1-a838-7b1f73a7ffcf] 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:4691 (sofia/internal/1002 at 78.138.90.58:5060) State Change CS_NEW -> CS_INIT 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1002 at 78.138.90.58:5060) Running State Change CS_INIT 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/1002 at 78.138.90.58:5060) State INIT 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:85 sofia/internal/1002 at 78.138.90.58:5060 SOFIA INIT 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:1871 sofia/internal/1002 at 78.138.90.58:5060 Patched SDP --- v=0 o=root 20076 20076 IN IP4 217.74.179.28 s=session c=IN IP4 217.74.179.28 t=0 0 m=audio 25626 RTP/AVP 8 0 18 111 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:111 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 +++ v=0 o=FreeSWITCH 1629510322 1629510323 IN IP4 78.138.90.58 s=FreeSWITCH c=IN IP4 78.138.90.58 t=0 0 m=audio 31178 RTP/AVP 8 0 18 111 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:111 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 send 1340 bytes to udp/[78.138.90.58]:5060 at 23:08:45.852280: ?? ------------------------------------------------------------------------ ?? INVITE sip:1002 at 78.138.90.58:5060 SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKapDgFeegKyHXK ?? Max-Forwards: 67 ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? CSeq: 28614398 INVITE ?? Contact: ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? 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: 372 ?? P-hint: USRLOC ?? X-FS-Support: update_display,send_info ?? Remote-Party-ID: "+41793940965" ;party=calling;screen=yes;privacy=off ?? ?? v=0 ?? o=FreeSWITCH 1629510322 1629510323 IN IP4 78.138.90.58 ?? s=FreeSWITCH ?? c=IN IP4 78.138.90.58 ?? t=0 0 ?? m=audio 31178 RTP/AVP 8 0 18 111 3 101 ?? a=rtpmap:8 PCMA/8000 ?? a=rtpmap:0 PCMU/8000 ?? a=rtpmap:18 G729/8000 ?? a=fmtp:18 annexb=no ?? a=rtpmap:111 G726-32/8000 ?? a=rtpmap:3 GSM/8000 ?? a=rtpmap:101 telephone-event/8000 ?? a=fmtp:101 0-16 ?? a=silenceSupp:off - - - - ?? a=ptime:20 ?? ------------------------------------------------------------------------ 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:125 (sofia/internal/1002 at 78.138.90.58:5060) State Change CS_INIT -> CS_ROUTING 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/1002 at 78.138.90.58:5060) State INIT going to sleep 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1002 at 78.138.90.58:5060) Running State Change CS_ROUTING 2012-05-24 23:08:45.840749 [DEBUG] switch_channel.c:1886 (sofia/internal/1002 at 78.138.90.58:5060) Callstate Change DOWN -> RINGING 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/1002 at 78.138.90.58:5060) State ROUTING 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:148 sofia/internal/1002 at 78.138.90.58:5060 SOFIA ROUTING 2012-05-24 23:08:45.840749 [DEBUG] switch_ivr_originate.c:66 (sofia/internal/1002 at 78.138.90.58:5060) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/1002 at 78.138.90.58:5060) State ROUTING going to sleep 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1002 at 78.138.90.58:5060) Running State Change CS_CONSUME_MEDIA 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:429 (sofia/internal/1002 at 78.138.90.58:5060) State CONSUME_MEDIA 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:429 (sofia/internal/1002 at 78.138.90.58:5060) State CONSUME_MEDIA going to sleep 2012-05-24 23:08:45.840749 [DEBUG] sofia.c:5532 Channel sofia/internal/1002 at 78.138.90.58:5060 entering state [calling][0] recv 365 bytes from udp/[78.138.90.58]:5060 at 23:08:45.964309: ?? ------------------------------------------------------------------------ ?? SIP/2.0 100 trying -- your call is important to us ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKapDgFeegKyHXK ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? CSeq: 28614398 INVITE ?? Server: kamailio (3.1.5 (x86_64/linux)) ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 496 bytes from udp/[78.138.90.58]:5060 at 23:08:47.414766: ?? ------------------------------------------------------------------------ ?? SIP/2.0 180 Ringing ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKapDgFeegKyHXK ?? Record-Route: ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ;tag=qCSaB2YJOAsAr2dJ-RWNMhPoDdn1RFvs ?? CSeq: 28614398 INVITE ?? Contact: "Mama" ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, REGISTER ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ 2012-05-24 23:08:47.400741 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:47.400741 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:47.400741 [DEBUG] sofia.c:5532 Channel sofia/internal/1002 at 78.138.90.58:5060 entering state [proceeding][180] 2012-05-24 23:08:47.400741 [NOTICE] sofia.c:5624 Ring-Ready sofia/internal/1002 at 78.138.90.58:5060! send 995 bytes to udp/[217.74.179.29]:5060 at 23:08:47.416538: ?? ------------------------------------------------------------------------ ?? SIP/2.0 180 Ringing ?? Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 ?? Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 ?? Record-Route: ?? From: "+41793940965" ;tag=as00589402 ?? To: ;tag=S7UZQygFt62Nm ?? Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net ?? CSeq: 102 INVITE ?? Contact: ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Accept: application/sdp ?? 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-Length: 0 ?? Remote-Party-ID: "Outbound Call" ;party=calling;privacy=off;screen=no ?? ?? ------------------------------------------------------------------------ 2012-05-24 23:08:47.400741 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:08:47.400741 [NOTICE] mod_sofia.c:2514 Ring-Ready sofia/internal/+41793940965 at bluesip.net! 2012-05-24 23:08:47.400741 [DEBUG] sofia.c:5532 Channel sofia/internal/+41793940965 at bluesip.net entering state [early][180] 2012-05-24 23:08:47.400741 [DEBUG] switch_core_session.c:731 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:08:47.400741 [NOTICE] switch_ivr_originate.c:483 Ring Ready sofia/internal/+41793940965 at bluesip.net! recv 496 bytes from udp/[78.138.90.58]:5060 at 23:08:47.804967: ?? ------------------------------------------------------------------------ ?? SIP/2.0 180 Ringing ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKapDgFeegKyHXK ?? Record-Route: ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ;tag=549D2DD03BBA7C67 ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? CSeq: 28614398 INVITE ?? Contact: ?? User-Agent: AVM FRITZ!Box Fon WLAN 7170 29.04.80 (Jan 27 2010) ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ 2012-05-24 23:08:47.800746 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:47.800746 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:47.800746 [DEBUG] sofia.c:5532 Channel sofia/internal/1002 at 78.138.90.58:5060 entering state [proceeding][180] recv 496 bytes from udp/[78.138.90.58]:5060 at 23:08:48.182098: ?? ------------------------------------------------------------------------ ?? SIP/2.0 180 Ringing ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKapDgFeegKyHXK ?? Record-Route: ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ;tag=qCSaB2YJOAsAr2dJ-RWNMhPoDdn1RFvs ?? CSeq: 28614398 INVITE ?? Contact: "Mama" ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, REGISTER ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ 2012-05-24 23:08:48.180741 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:48.180741 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:48.180741 [DEBUG] sofia.c:5532 Channel sofia/internal/1002 at 78.138.90.58:5060 entering state [proceeding][180] recv 496 bytes from udp/[78.138.90.58]:5060 at 23:08:48.417403: ?? ------------------------------------------------------------------------ ?? SIP/2.0 180 Ringing ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKapDgFeegKyHXK ?? Record-Route: ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ;tag=549D2DD03BBA7C67 ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? CSeq: 28614398 INVITE ?? Contact: ?? User-Agent: AVM FRITZ!Box Fon WLAN 7170 29.04.80 (Jan 27 2010) ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ 2012-05-24 23:08:48.400744 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:48.400744 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:48.400744 [DEBUG] sofia.c:5532 Channel sofia/internal/1002 at 78.138.90.58:5060 entering state [proceeding][180] recv 496 bytes from udp/[78.138.90.58]:5060 at 23:08:49.292599: ?? ------------------------------------------------------------------------ ?? SIP/2.0 180 Ringing ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKapDgFeegKyHXK ?? Record-Route: ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ;tag=549D2DD03BBA7C67 ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? CSeq: 28614398 INVITE ?? Contact: ?? User-Agent: AVM FRITZ!Box Fon WLAN 7170 29.04.80 (Jan 27 2010) ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ 2012-05-24 23:08:49.280742 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:49.280742 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:49.280742 [DEBUG] sofia.c:5532 Channel sofia/internal/1002 at 78.138.90.58:5060 entering state [proceeding][180] recv 478 bytes from udp/[217.74.179.29]:5060 at 23:09:15.066012: ?? ------------------------------------------------------------------------ ?? CANCEL sip:gw+trunk_1000 at 78.138.90.58:6090;transport=udp;gw=trunk_1000 SIP/2.0 ?? Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 ?? Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 ?? From: "+41793940965" ;tag=as00589402 ?? To: ?? Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net ?? CSeq: 102 CANCEL ?? User-Agent: blueSIP PSTN GW ?? Max-Forwards: 69 ?? Content-Length: 0 ?? P-hint: USRLOC ?? ?? ------------------------------------------------------------------------ send 369 bytes to udp/[217.74.179.29]:5060 at 23:09:15.066176: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 ?? Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 ?? From: "+41793940965" ;tag=as00589402 ?? To: ;tag=S7UZQygFt62Nm ?? Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net ?? CSeq: 102 CANCEL ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ send 766 bytes to udp/[217.74.179.29]:5060 at 23:09:15.066289: ?? ------------------------------------------------------------------------ ?? SIP/2.0 487 Request Terminated ?? Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 ?? Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 ?? From: "+41793940965" ;tag=as00589402 ?? To: ;tag=S7UZQygFt62Nm ?? Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net ?? CSeq: 102 INVITE ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? 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-Length: 0 ?? ?? ------------------------------------------------------------------------ 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:09:15.060742 [DEBUG] sofia.c:5532 Channel sofia/internal/+41793940965 at bluesip.net entering state [terminated][487] 2012-05-24 23:09:15.060742 [DEBUG] switch_channel.c:2848 (sofia/internal/+41793940965 at bluesip.net) Callstate Change RINGING -> HANGUP 2012-05-24 23:09:15.060742 [NOTICE] sofia.c:6301 Hangup sofia/internal/+41793940965 at bluesip.net [CS_EXECUTE] [ORIGINATOR_CANCEL] 2012-05-24 23:09:15.060742 [DEBUG] switch_channel.c:2871 Send signal sofia/internal/+41793940965 at bluesip.net [KILL] 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:09:15.060742 [DEBUG] switch_channel.c:2848 (sofia/internal/1002 at 78.138.90.58:5060) Callstate Change RINGING -> HANGUP 2012-05-24 23:09:15.060742 [NOTICE] switch_ivr_originate.c:3182 Hangup sofia/internal/1002 at 78.138.90.58:5060 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL] 2012-05-24 23:09:15.060742 [DEBUG] switch_channel.c:2871 Send signal sofia/internal/1002 at 78.138.90.58:5060 [KILL] 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1002 at 78.138.90.58:5060) Running State Change CS_HANGUP 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/1002 at 78.138.90.58:5060) State HANGUP 2012-05-24 23:09:15.060742 [DEBUG] mod_sofia.c:469 Channel sofia/internal/1002 at 78.138.90.58:5060 hanging up, cause: ORIGINATOR_CANCEL 2012-05-24 23:09:15.060742 [DEBUG] mod_sofia.c:523 Sending CANCEL to sofia/internal/1002 at 78.138.90.58:5060 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:47 sofia/internal/1002 at 78.138.90.58:5060 Standard HANGUP, cause: ORIGINATOR_CANCEL 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/1002 at 78.138.90.58:5060) State HANGUP going to sleep 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:393 (sofia/internal/1002 at 78.138.90.58:5060) State Change CS_HANGUP -> CS_REPORTING 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1002 at 78.138.90.58:5060) Running State Change CS_REPORTING 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/1002 at 78.138.90.58:5060) State REPORTING 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:79 sofia/internal/1002 at 78.138.90.58:5060 Standard REPORTING, cause: ORIGINATOR_CANCEL 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/1002 at 78.138.90.58:5060) State REPORTING going to sleep 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:387 (sofia/internal/1002 at 78.138.90.58:5060) State Change CS_REPORTING -> CS_DESTROY 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1382 Session 9 (sofia/internal/1002 at 78.138.90.58:5060) Locked, Waiting on external entities 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:09:15.060742 [DEBUG] switch_ivr_originate.c:3358 Originate Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL] 2012-05-24 23:09:15.060742 [NOTICE] switch_core_session.c:1400 Session 9 (sofia/internal/1002 at 78.138.90.58:5060) Ended 2012-05-24 23:09:15.060742 [NOTICE] switch_core_session.c:1402 Close Channel sofia/internal/1002 at 78.138.90.58:5060 [CS_DESTROY] 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:491 (sofia/internal/1002 at 78.138.90.58:5060) Callstate Change HANGUP -> DOWN 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:494 (sofia/internal/1002 at 78.138.90.58:5060) Running State Change CS_DESTROY 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/1002 at 78.138.90.58:5060) State DESTROY 2012-05-24 23:09:15.060742 [DEBUG] mod_sofia.c:374 sofia/internal/1002 at 78.138.90.58:5060 SOFIA DESTROY 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:86 sofia/internal/1002 at 78.138.90.58:5060 Standard DESTROY 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/1002 at 78.138.90.58:5060) State DESTROY going to sleep 2012-05-24 23:09:15.060742 [INFO] mod_dptools.c:2922 Originate Failed.? Cause: ORIGINATOR_CANCEL 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:2287 sofia/internal/+41793940965 at bluesip.net skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/+41793940965 at bluesip.net) State EXECUTE going to sleep 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_HANGUP 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/+41793940965 at bluesip.net) State HANGUP 2012-05-24 23:09:15.060742 [DEBUG] mod_sofia.c:469 Channel sofia/internal/+41793940965 at bluesip.net hanging up, cause: ORIGINATOR_CANCEL 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:47 sofia/internal/+41793940965 at bluesip.net Standard HANGUP, cause: ORIGINATOR_CANCEL 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/+41793940965 at bluesip.net) State HANGUP going to sleep 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:393 (sofia/internal/+41793940965 at bluesip.net) State Change CS_HANGUP -> CS_REPORTING 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_REPORTING 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/+41793940965 at bluesip.net) State REPORTING send 383 bytes to udp/[78.138.90.58]:5060 at 23:09:15.087695: ?? ------------------------------------------------------------------------ ?? CANCEL sip:1002 at 78.138.90.58:5060 SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKapDgFeegKyHXK ?? Max-Forwards: 67 ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? CSeq: 28614398 CANCEL ?? Reason: FreeSWITCH;cause=487;text="ORIGINATOR_CANCEL" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 408 bytes from udp/[217.74.179.29]:5060 at 23:09:15.094670: ?? ------------------------------------------------------------------------ ?? ACK sip:gw+trunk_1000 at 78.138.90.58:6090;transport=udp;gw=trunk_1000 SIP/2.0 ?? Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 ?? From: "+41793940965" ;tag=as00589402 ?? Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net ?? To: ;tag=S7UZQygFt62Nm ?? CSeq: 102 ACK ?? User-Agent: Sip EXpress router(0.9.7 (i386/linux)) ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:79 sofia/internal/+41793940965 at bluesip.net Standard REPORTING, cause: ORIGINATOR_CANCEL 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/+41793940965 at bluesip.net) State REPORTING going to sleep 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:387 (sofia/internal/+41793940965 at bluesip.net) State Change CS_REPORTING -> CS_DESTROY 2012-05-24 23:09:15.120807 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:09:15.120807 [DEBUG] switch_core_session.c:1382 Session 8 (sofia/internal/+41793940965 at bluesip.net) Locked, Waiting on external entities 2012-05-24 23:09:15.120807 [NOTICE] switch_core_session.c:1400 Session 8 (sofia/internal/+41793940965 at bluesip.net) Ended 2012-05-24 23:09:15.120807 [NOTICE] switch_core_session.c:1402 Close Channel sofia/internal/+41793940965 at bluesip.net [CS_DESTROY] 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:491 (sofia/internal/+41793940965 at bluesip.net) Callstate Change HANGUP -> DOWN 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:494 (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_DESTROY 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/+41793940965 at bluesip.net) State DESTROY 2012-05-24 23:09:15.120807 [DEBUG] mod_sofia.c:374 sofia/internal/+41793940965 at bluesip.net SOFIA DESTROY 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:86 sofia/internal/+41793940965 at bluesip.net Standard DESTROY 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/+41793940965 at bluesip.net) State DESTROY going to sleep recv 378 bytes from udp/[78.138.90.58]:5060 at 23:09:15.135235: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 canceling ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKapDgFeegKyHXK ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ;tag=a6a1c5f60faecf035a1ae5b6e96e979a-8c5b ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? CSeq: 28614398 CANCEL ?? Server: kamailio (3.1.5 (x86_64/linux)) ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 446 bytes from udp/[78.138.90.58]:5060 at 23:09:16.419308: ?? ------------------------------------------------------------------------ ?? SIP/2.0 487 Request Terminated ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKapDgFeegKyHXK ?? Record-Route: ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ;tag=qCSaB2YJOAsAr2dJ-RWNMhPoDdn1RFvs ?? CSeq: 28614398 INVITE ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, REGISTER ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 354 bytes to udp/[78.138.90.58]:5060 at 23:09:16.419543: ?? ------------------------------------------------------------------------ ?? ACK sip:1002 at 78.138.90.58:5060 SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKapDgFeegKyHXK ?? Max-Forwards: 67 ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ;tag=qCSaB2YJOAsAr2dJ-RWNMhPoDdn1RFvs ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? CSeq: 28614398 ACK ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 564 bytes from udp/[41.34.123.243]:58881 at 23:13:13.612418: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjHQIpv.ACHGbXD4IHrMr72LwZ8kYdlu-J ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=wOP.FVT4Crk8O5E.bB7BWUhNO6QG91IG ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60926 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 744 bytes to udp/[41.34.123.243]:58881 at 23:13:13.613152: ?? ------------------------------------------------------------------------ ?? SIP/2.0 401 Unauthorized ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjHQIpv.ACHGbXD4IHrMr72LwZ8kYdlu-J;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=wOP.FVT4Crk8O5E.bB7BWUhNO6QG91IG ?? To: "Sherif 1003" ;tag=USeHUmjpmrFUB ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60926 REGISTER ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? WWW-Authenticate: Digest realm="sip.pcfone.com", nonce="09674764-a5f6-11e1-a83c-7b1f73a7ffcf", algorithm=MD5, qop="auth" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 830 bytes from udp/[41.34.123.243]:58881 at 23:13:13.847746: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjFr7rIAdAJPRQd2GmrMrVib3KT3Rup0q4 ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=wOP.FVT4Crk8O5E.bB7BWUhNO6QG91IG ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60927 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Authorization: Digest username="1002", realm="sip.pcfone.com", nonce="09674764-a5f6-11e1-a83c-7b1f73a7ffcf", uri="sip:sip.pcfone.com:6090", response="3bc0bb82a803ded3641704c4eb78d58f", algorithm=MD5, cnonce="sc3AJfhuev18Bs-L-6tORmI.RXlxnPOz", qop=auth, nc=00000001 ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 706 bytes to udp/[41.34.123.243]:58881 at 23:13:13.850197: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjFr7rIAdAJPRQd2GmrMrVib3KT3Rup0q4;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=wOP.FVT4Crk8O5E.bB7BWUhNO6QG91IG ?? To: "Sherif 1003" ;tag=v279vF3SH15DQ ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60927 REGISTER ?? Contact: ;expires=300 ?? Date: Thu, 24 May 2012 23:13:13 GMT ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ send 989 bytes to udp/[41.34.123.243]:58881 at 23:13:13.888717: ?? ------------------------------------------------------------------------ ?? NOTIFY sip:1002 at 192.168.178.26:58881;ob SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKBZ68g9yKg77FF ?? Route: ;ob ?? Max-Forwards: 70 ?? From: ;tag=XB12yamXeav0j ?? To: ?? Call-ID: e0efa252-2098-1230-8985-00163e6bb553 ?? CSeq: 28614532 NOTIFY ?? Contact: ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? 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=noresource ?? Content-Type: application/simple-message-summary ?? Content-Length: 91 ?? ?? Messages-Waiting: yes ?? Message-Account: sip:1002 at 78.138.90.58 ?? Voice-Message: 1/0 (0/0) ?? ?? ------------------------------------------------------------------------ recv 308 bytes from udp/[41.34.123.243]:58881 at 23:13:14.298850: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;received=78.138.90.58;branch=z9hG4bKBZ68g9yKg77FF ?? Call-ID: e0efa252-2098-1230-8985-00163e6bb553 ?? From: ;tag=XB12yamXeav0j ?? To: ;tag=z9hG4bKBZ68g9yKg77FF ?? CSeq: 28614532 NOTIFY ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ recv 564 bytes from udp/[41.34.123.243]:58881 at 23:18:09.067232: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjABTslmZIhiHYRtLqQn2erwmX19MVYHFC ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=JBSXL5WZLWbLzpvRFoTSP5x8qQslI1ST ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60928 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 744 bytes to udp/[41.34.123.243]:58881 at 23:18:09.068121: ?? ------------------------------------------------------------------------ ?? SIP/2.0 401 Unauthorized ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjABTslmZIhiHYRtLqQn2erwmX19MVYHFC;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=JBSXL5WZLWbLzpvRFoTSP5x8qQslI1ST ?? To: "Sherif 1003" ;tag=ymtU0540BKjKe ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60928 REGISTER ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? WWW-Authenticate: Digest realm="sip.pcfone.com", nonce="b9821df4-a5f6-11e1-a83d-7b1f73a7ffcf", algorithm=MD5, qop="auth" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 830 bytes from udp/[41.34.123.243]:58881 at 23:18:09.302944: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjRgtztcy63vMT97Jp5pH9n311li5EFeY- ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=JBSXL5WZLWbLzpvRFoTSP5x8qQslI1ST ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60929 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Authorization: Digest username="1002", realm="sip.pcfone.com", nonce="b9821df4-a5f6-11e1-a83d-7b1f73a7ffcf", uri="sip:sip.pcfone.com:6090", response="519c6301816b51f3bbb98d97a347e92b", algorithm=MD5, cnonce="sc3AJfhuev18Bs-L-6tORmI.RXlxnPOz", qop=auth, nc=00000001 ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 706 bytes to udp/[41.34.123.243]:58881 at 23:18:09.305217: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjRgtztcy63vMT97Jp5pH9n311li5EFeY-;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=JBSXL5WZLWbLzpvRFoTSP5x8qQslI1ST ?? To: "Sherif 1003" ;tag=ZXKm20N48U85S ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60929 REGISTER ?? Contact: ;expires=300 ?? Date: Thu, 24 May 2012 23:18:09 GMT ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ send 989 bytes to udp/[41.34.123.243]:58881 at 23:18:09.379987: ?? ------------------------------------------------------------------------ ?? NOTIFY sip:1002 at 192.168.178.26:58881;ob SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKc8Z1j4FQDgy2a ?? Route: ;ob ?? Max-Forwards: 70 ?? From: ;tag=06cD4U6754yrN ?? To: ?? Call-ID: 91100602-2099-1230-8985-00163e6bb553 ?? CSeq: 28614680 NOTIFY ?? Contact: ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? 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=noresource ?? Content-Type: application/simple-message-summary ?? Content-Length: 91 ?? ?? Messages-Waiting: yes ?? Message-Account: sip:1002 at 78.138.90.58 ?? Voice-Message: 1/0 (0/0) ?? ?? ------------------------------------------------------------------------ recv 308 bytes from udp/[41.34.123.243]:58881 at 23:18:09.773629: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;received=78.138.90.58;branch=z9hG4bKc8Z1j4FQDgy2a ?? Call-ID: 91100602-2099-1230-8985-00163e6bb553 ?? From: ;tag=06cD4U6754yrN ?? To: ;tag=z9hG4bKc8Z1j4FQDgy2a ?? CSeq: 28614680 NOTIFY ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ recv 564 bytes from udp/[41.34.123.243]:58881 at 23:23:04.522455: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjaCuKyaBi6ZqcsHyp5HhH3WthJStom0SK ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=qfUt1fQWuE-Ro3zWld6X0r6WidtD6B1q ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60930 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 744 bytes to udp/[41.34.123.243]:58881 at 23:23:04.523158: ?? ------------------------------------------------------------------------ ?? SIP/2.0 401 Unauthorized ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjaCuKyaBi6ZqcsHyp5HhH3WthJStom0SK;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=qfUt1fQWuE-Ro3zWld6X0r6WidtD6B1q ?? To: "Sherif 1003" ;tag=1F655pQB3DNBH ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60930 REGISTER ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? WWW-Authenticate: Digest realm="sip.pcfone.com", nonce="699cfe2a-a5f7-11e1-a83e-7b1f73a7ffcf", algorithm=MD5, qop="auth" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 830 bytes from udp/[41.34.123.243]:58881 at 23:23:04.757798: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjs8SAufeoyzCLrePaAbYhZuOqQF47XIc- ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=qfUt1fQWuE-Ro3zWld6X0r6WidtD6B1q ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60931 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Authorization: Digest username="1002", realm="sip.pcfone.com", nonce="699cfe2a-a5f7-11e1-a83e-7b1f73a7ffcf", uri="sip:sip.pcfone.com:6090", response="23a886934aaf049f08432b6bb226bd37", algorithm=MD5, cnonce="sc3AJfhuev18Bs-L-6tORmI.RXlxnPOz", qop=auth, nc=00000001 ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 706 bytes to udp/[41.34.123.243]:58881 at 23:23:04.759924: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjs8SAufeoyzCLrePaAbYhZuOqQF47XIc-;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=qfUt1fQWuE-Ro3zWld6X0r6WidtD6B1q ?? To: "Sherif 1003" ;tag=2rZy7H8e0pByc ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60931 REGISTER ?? Contact: ;expires=300 ?? Date: Thu, 24 May 2012 23:23:04 GMT ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ send 989 bytes to udp/[41.34.123.243]:58881 at 23:23:04.872728: ?? ------------------------------------------------------------------------ ?? NOTIFY sip:1002 at 192.168.178.26:58881;ob SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKDHStmZ0taSmNp ?? Route: ;ob ?? Max-Forwards: 70 ?? From: ;tag=31rQ9cSjXZ1gr ?? To: ?? Call-ID: 4130a1ce-209a-1230-8985-00163e6bb553 ?? CSeq: 28614828 NOTIFY ?? Contact: ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? 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=noresource ?? Content-Type: application/simple-message-summary ?? Content-Length: 91 ?? ?? Messages-Waiting: yes ?? Message-Account: sip:1002 at 78.138.90.58 ?? Voice-Message: 1/0 (0/0) ?? ?? ------------------------------------------------------------------------ recv 308 bytes from udp/[41.34.123.243]:58881 at 23:23:05.269063: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;received=78.138.90.58;branch=z9hG4bKDHStmZ0taSmNp ?? Call-ID: 4130a1ce-209a-1230-8985-00163e6bb553 ?? From: ;tag=31rQ9cSjXZ1gr ?? To: ;tag=z9hG4bKDHStmZ0taSmNp ?? CSeq: 28614828 NOTIFY ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 916 bytes to udp/[217.74.179.29]:5060 at 23:25:45.873423: ?? ------------------------------------------------------------------------ ?? REGISTER sip:bluesip.net;transport=udp SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKetjKptHy71a8H ?? Max-Forwards: 70 ?? From: ;tag=6r0vBQZS650Fg ?? To: ?? Call-ID: 2a3e4696-a5b9-11e1-a7ed-7b1f73a7ffcf ?? CSeq: 28601493 REGISTER ?? Contact: ?? Expires: 3600 ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? Authorization: Digest username="bluesip/sherifomran", realm="bluesip.net", nonce="4fbeba4ebfb7414d5c701bd6f994402d551f3acf", algorithm=MD5, uri="sip:bluesip.net;transport=udp", response="c6cdcafe0418e519bc9ee0d8fa3d4d74" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 722 bytes from udp/[217.74.179.29]:5060 at 23:25:45.904176: ?? ------------------------------------------------------------------------ ?? SIP/2.0 401 Unauthorized ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKetjKptHy71a8H ?? From: ;tag=6r0vBQZS650Fg ?? To: ;tag=0354a2e1b960c9cc2279eca4e5f84e20.1b16 ?? Call-ID: 2a3e4696-a5b9-11e1-a7ed-7b1f73a7ffcf ?? CSeq: 28601493 REGISTER ?? WWW-Authenticate: Digest realm="bluesip.net", nonce="4fbec4a4ba82d1f1feaa32ccec01b85be054fcf7", stale=true ?? Server: Sip EXpress router (0.9.7 (i386/linux)) ?? Content-Length: 0 ?? Warning: 392 217.74.179.29:5060 "Noisy feedback tells:? pid=5455 req_src_ip=78.138.90.58 req_src_port=6090 in_uri=sip:bluesip.net;transport=udp out_uri=sip:bluesip.net;transport=udp via_cnt==1" ?? ?? ------------------------------------------------------------------------ send 916 bytes to udp/[217.74.179.29]:5060 at 23:25:45.904416: ?? ------------------------------------------------------------------------ ?? REGISTER sip:bluesip.net;transport=udp SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKF3BcrN214a1tD ?? Max-Forwards: 70 ?? From: ;tag=6r0vBQZS650Fg ?? To: ?? Call-ID: 2a3e4696-a5b9-11e1-a7ed-7b1f73a7ffcf ?? CSeq: 28601494 REGISTER ?? Contact: ?? Expires: 3600 ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? Authorization: Digest username="bluesip/sherifomran", realm="bluesip.net", nonce="4fbec4a4ba82d1f1feaa32ccec01b85be054fcf7", algorithm=MD5, uri="sip:bluesip.net;transport=udp", response="4c09dbe4b9accb52d4104b40dfe20040" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 699 bytes from udp/[217.74.179.29]:5060 at 23:25:45.935046: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKF3BcrN214a1tD ?? From: ;tag=6r0vBQZS650Fg ?? To: ;tag=0354a2e1b960c9cc2279eca4e5f84e20.2017 ?? Call-ID: 2a3e4696-a5b9-11e1-a7ed-7b1f73a7ffcf ?? CSeq: 28601494 REGISTER ?? Contact: ;q=0.5;expires=3600 ?? Server: Sip EXpress router (0.9.7 (i386/linux)) ?? Content-Length: 0 ?? Warning: 392 217.74.179.29:5060 "Noisy feedback tells:? pid=5462 req_src_ip=78.138.90.58 req_src_port=6090 in_uri=sip:bluesip.net;transport=udp out_uri=sip:bluesip.net;transport=udp via_cnt==1" ?? ?? ------------------------------------------------------------------------ recv 564 bytes from udp/[41.34.123.243]:58881 at 23:27:59.977057: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjEOP.a4NoXqoFqC7a.k2tyeX-mYlUyH4H ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60932 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 744 bytes to udp/[41.34.123.243]:58881 at 23:27:59.977960: ?? ------------------------------------------------------------------------ ?? SIP/2.0 401 Unauthorized ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjEOP.a4NoXqoFqC7a.k2tyeX-mYlUyH4H;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K ?? To: "Sherif 1003" ;tag=4ajgB89Nt8Q3K ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60932 REGISTER ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? WWW-Authenticate: Digest realm="sip.pcfone.com", nonce="19b7cc40-a5f8-11e1-a83f-7b1f73a7ffcf", algorithm=MD5, qop="auth" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 564 bytes from udp/[41.34.123.243]:58881 at 23:28:00.468487: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjEOP.a4NoXqoFqC7a.k2tyeX-mYlUyH4H ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60932 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 744 bytes to udp/[41.34.123.243]:58881 at 23:28:00.468679: ?? ------------------------------------------------------------------------ ?? SIP/2.0 401 Unauthorized ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjEOP.a4NoXqoFqC7a.k2tyeX-mYlUyH4H;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K ?? To: "Sherif 1003" ;tag=4ajgB89Nt8Q3K ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60932 REGISTER ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? WWW-Authenticate: Digest realm="sip.pcfone.com", nonce="19b7cc40-a5f8-11e1-a83f-7b1f73a7ffcf", algorithm=MD5, qop="auth" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 830 bytes from udp/[41.34.123.243]:58881 at 23:28:00.717446: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPj0iHd3-XUjnvcBs9DiJUCWzrmYTmWxFLt ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60933 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Authorization: Digest username="1002", realm="sip.pcfone.com", nonce="19b7cc40-a5f8-11e1-a83f-7b1f73a7ffcf", uri="sip:sip.pcfone.com:6090", response="e21c903e6d98b5adcdd2eefa75217157", algorithm=MD5, cnonce="sc3AJfhuev18Bs-L-6tORmI.RXlxnPOz", qop=auth, nc=00000001 ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 706 bytes to udp/[41.34.123.243]:58881 at 23:28:00.719947: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPj0iHd3-XUjnvcBs9DiJUCWzrmYTmWxFLt;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K ?? To: "Sherif 1003" ;tag=5KB9c3tSQHepF ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60933 REGISTER ?? Contact: ;expires=300 ?? Date: Thu, 24 May 2012 23:28:00 GMT ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ send 989 bytes to udp/[41.34.123.243]:58881 at 23:28:00.868368: ?? ------------------------------------------------------------------------ ?? NOTIFY sip:1002 at 192.168.178.26:58881;ob SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKgc54SgK51KQDS ?? Route: ;ob ?? Max-Forwards: 70 ?? From: ;tag=6v41eyBXmt48a ?? To: ?? Call-ID: f19dfbce-209a-1230-8985-00163e6bb553 ?? CSeq: 28614976 NOTIFY ?? Contact: ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? 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=noresource ?? Content-Type: application/simple-message-summary ?? Content-Length: 91 ?? ?? Messages-Waiting: yes ?? Message-Account: sip:1002 at 78.138.90.58 ?? Voice-Message: 1/0 (0/0) ?? ?? ------------------------------------------------------------------------ recv 308 bytes from udp/[41.34.123.243]:58881 at 23:28:01.128279: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;received=78.138.90.58;branch=z9hG4bKgc54SgK51KQDS ?? Call-ID: f19dfbce-209a-1230-8985-00163e6bb553 ?? From: ;tag=6v41eyBXmt48a ?? To: ;tag=z9hG4bKgc54SgK51KQDS ?? CSeq: 28614976 NOTIFY ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 913 bytes to udp/[217.74.179.29]:5060 at 23:28:11.885213: ?? ------------------------------------------------------------------------ ?? REGISTER sip:bluesip.net;transport=udp SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKHNyXUB48yvD0m ?? Max-Forwards: 70 ?? From: ;tag=5F739Uep9vaXm ?? To: ?? Call-ID: 2a3e49c0-a5b9-11e1-a7ee-7b1f73a7ffcf ?? CSeq: 28601493 REGISTER ?? Contact: ?? Expires: 3600 ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? Authorization: Digest username="bluesip/salahomran", realm="bluesip.net", nonce="4fbebeca79a5cff7c417463814cb9054d313bcac", algorithm=MD5, uri="sip:bluesip.net;transport=udp", response="bb1babb90f4ea1dc8fbb9aa45d6038c7" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 720 bytes from udp/[217.74.179.29]:5060 at 23:28:11.915764: ?? ------------------------------------------------------------------------ ?? SIP/2.0 401 Unauthorized ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKHNyXUB48yvD0m ?? From: ;tag=5F739Uep9vaXm ?? To: ;tag=0354a2e1b960c9cc2279eca4e5f84e20.3454 ?? Call-ID: 2a3e49c0-a5b9-11e1-a7ee-7b1f73a7ffcf ?? CSeq: 28601493 REGISTER ?? WWW-Authenticate: Digest realm="bluesip.net", nonce="4fbec5365af3a174f1cc0b8cda770c727d25bf71", stale=true ?? Server: Sip EXpress router (0.9.7 (i386/linux)) ?? Content-Length: 0 ?? Warning: 392 217.74.179.29:5060 "Noisy feedback tells:? pid=5462 req_src_ip=78.138.90.58 req_src_port=6090 in_uri=sip:bluesip.net;transport=udp out_uri=sip:bluesip.net;transport=udp via_cnt==1" ?? ?? ------------------------------------------------------------------------ send 913 bytes to udp/[217.74.179.29]:5060 at 23:28:11.915914: ?? ------------------------------------------------------------------------ ?? REGISTER sip:bluesip.net;transport=udp SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKjyQpX6mcv53jg ?? Max-Forwards: 70 ?? From: ;tag=5F739Uep9vaXm ?? To: ?? Call-ID: 2a3e49c0-a5b9-11e1-a7ee-7b1f73a7ffcf ?? CSeq: 28601494 REGISTER ?? Contact: ?? Expires: 3600 ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? Authorization: Digest username="bluesip/salahomran", realm="bluesip.net", nonce="4fbec5365af3a174f1cc0b8cda770c727d25bf71", algorithm=MD5, uri="sip:bluesip.net;transport=udp", response="e260f6dc9b01ce8bea8c5ffb36bd4ddd" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 697 bytes from udp/[217.74.179.29]:5060 at 23:28:11.946325: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKjyQpX6mcv53jg ?? From: ;tag=5F739Uep9vaXm ?? To: ;tag=0354a2e1b960c9cc2279eca4e5f84e20.4b5c ?? Call-ID: 2a3e49c0-a5b9-11e1-a7ee-7b1f73a7ffcf ?? CSeq: 28601494 REGISTER ?? Contact: ;q=0.5;expires=3600 ?? Server: Sip EXpress router (0.9.7 (i386/linux)) ?? Content-Length: 0 ?? Warning: 392 217.74.179.29:5060 "Noisy feedback tells:? pid=5455 req_src_ip=78.138.90.58 req_src_port=6090 in_uri=sip:bluesip.net;transport=udp out_uri=sip:bluesip.net;transport=udp via_cnt==1" ?? ?? ------------------------------------------------------------------------ recv 564 bytes from udp/[41.34.123.243]:58881 at 23:32:55.930149: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjuk8DZGiaXfN7BmtslNHVjtSg5UXsbHXc ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=Pz.t-0tGXj3saM.ehBiB1EAMmmitze1D ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60934 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 744 bytes to udp/[41.34.123.243]:58881 at 23:32:55.931324: ?? ------------------------------------------------------------------------ ?? SIP/2.0 401 Unauthorized ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjuk8DZGiaXfN7BmtslNHVjtSg5UXsbHXc;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=Pz.t-0tGXj3saM.ehBiB1EAMmmitze1D ?? To: "Sherif 1003" ;tag=75XtgSv0H3tUp ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60934 REGISTER ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? WWW-Authenticate: Digest realm="sip.pcfone.com", nonce="ca1eabee-a5f8-11e1-a840-7b1f73a7ffcf", algorithm=MD5, qop="auth" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 830 bytes from udp/[41.34.123.243]:58881 at 23:32:56.165624: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjKEQ6xXmyzZxA6vCM88vHNYAOPAjUqFsD ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=Pz.t-0tGXj3saM.ehBiB1EAMmmitze1D ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60935 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Authorization: Digest username="1002", realm="sip.pcfone.com", nonce="ca1eabee-a5f8-11e1-a840-7b1f73a7ffcf", uri="sip:sip.pcfone.com:6090", response="ff6d12f4ec7abd4b37187222b3af2358", algorithm=MD5, cnonce="sc3AJfhuev18Bs-L-6tORmI.RXlxnPOz", qop=auth, nc=00000001 ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 706 bytes to udp/[41.34.123.243]:58881 at 23:32:56.168033: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjKEQ6xXmyzZxA6vCM88vHNYAOPAjUqFsD;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=Pz.t-0tGXj3saM.ehBiB1EAMmmitze1D ?? To: "Sherif 1003" ;tag=8eQKjmD4ecHej ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60935 REGISTER ?? Contact: ;expires=300 ?? Date: Thu, 24 May 2012 23:32:56 GMT ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ send 989 bytes to udp/[41.34.123.243]:58881 at 23:32:56.360238: ?? ------------------------------------------------------------------------ ?? NOTIFY sip:1002 at 192.168.178.26:58881;ob SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKK7gFZ15FSet5B ?? Route: ;ob ?? Max-Forwards: 70 ?? From: ;tag=9QgcmFy7BN70D ?? To: ?? Call-ID: a1be7708-209b-1230-8985-00163e6bb553 ?? CSeq: 28615124 NOTIFY ?? Contact: ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? 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=noresource ?? Content-Type: application/simple-message-summary ?? Content-Length: 91 ?? ?? Messages-Waiting: yes ?? Message-Account: sip:1002 at 78.138.90.58 ?? Voice-Message: 1/0 (0/0) ?? ?? ------------------------------------------------------------------------ recv 308 bytes from udp/[41.34.123.243]:58881 at 23:32:56.623554: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;received=78.138.90.58;branch=z9hG4bKK7gFZ15FSet5B ?? Call-ID: a1be7708-209b-1230-8985-00163e6bb553 ?? From: ;tag=9QgcmFy7BN70D ?? To: ;tag=z9hG4bKK7gFZ15FSet5B ?? CSeq: 28615124 NOTIFY ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ freeswitch at internal> --- On Thu, 5/24/12, Kristian Kielhofner wrote: From: Kristian Kielhofner Subject: Re: [Freeswitch-users] Freeswitch (SBC + Media) does not know that gateway phone is picked up To: "FreeSWITCH Users Help" Date: Thursday, May 24, 2012, 7:51 PM Siptrace and logs please. On Thu, May 24, 2012 at 11:53 AM, Sherif Omran wrote: > > Hi all, > > My topology is as follows: > > Kamailio? -> FS (SBS+Media server) > > I came across an issue with my system as follows.? I have a Hardphone registered. When I do local call inside kamailio, it gets to FS and returns back well and FS understands when I lift the handset. However, I added a gateway (german landline server), when I call my self from another phone, the call gets to FS and then transmits to Kamailio, it rings my extension but when I lift the handset FS does not notice it and keeps ringing. > > Any body has an Idea? Here is my gateway trunk. > > > ??????? > ??????? > ??????? > ??????? > ??????? > ????? > > > thanks in advance > Sherif Omran > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Kristian Kielhofner _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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 at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -----Inline Attachment Follows----- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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 at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -----Inline Attachment Follows----- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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 at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -----Inline Attachment Follows----- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120526/89453bb9/attachment-0001.html From neilp at cs.stanford.edu Sat May 26 20:39:45 2012 From: neilp at cs.stanford.edu (Neil Patel) Date: Sat, 26 May 2012 09:39:45 -0700 Subject: [Freeswitch-users] Unable to build Git tree on Mac OSX Lion In-Reply-To: References: Message-ID: I was able to get around this by taking a hint from this comment . I modified this part of configure.in before running bootstrap: #set SOLINK variable based on compiler and host if test "x${ax_cv_c_compiler_vendor}" = "xsun" ; then SOLINK="-Bdynamic -dy -G" elif test "x${ax_cv_c_compiler_vendor}" = "xgnu" ; then case "$host" in *darwin10.*) SOLINK="-dynamic -force-flat-namespace" ;; *darwin*) SOLINK="-dynamic -bundle -force-flat-namespace" ;; *-solaris2*) SOLINK="-shared -Xlinker" ;; *) SOLINK="-shared -Xlinker -x" ;; esac else AC_ERROR([Please update configure.in with SOLINK values for your compiler]) fi OSX Lion uses darwin11, so the first case was not hit. I removed the -bundle flag from the highlighted case and got it to avoid the error. This is probably something the build system should patch up. -Neil On Fri, May 25, 2012 at 6:44 PM, Ken Rice wrote: > Can you do a git bisect on this and figure out where it broke? This > should build just fine... > > K > > > > On 5/25/12 6:50 PM, "Mario G" wrote: > > This is not much help but I just updated with the current git and the make > current went fine. However, I am on 10.6.8. Just out of curiosity I did a > FIND on -bundle and found it was only used once with mod_lua, -dynamiclib > was found only once but it looked to be referenced by several modules. I am > not a linux developer so have little idea what the messages mean exactly. I > found a few hits on the web, all involved source fixes, I know, not much > help... > Mario G > > On May 25, 2012, at 4:03 PM, Neil Patel wrote: > > Hi Mario, > > This looks like a deeper problem. I tried commenting out mod_amr but then > it broke on other codecs. > > The error says -bundle not allowed with -dyanmiclib. Seems like these > flags are mutually exclusive. Is there a problem in how the FS is > configuring the build? > > Thanks, > Neil > > On Fri, May 25, 2012 at 3:53 PM, Mario G wrote: > > The impression I get from the wiki is that mod_amr is optional. I would > try to comment it out in modules.conf in the FS source dir and see what > happens. I wish I had time to futz with FS on Lion but I am trying to > resolve 2 other FS issues for over a month with no luck. This is why I put > off updating the wiki, I won't until everything is perfect. > > On May 25, 2012, at 2:53 PM, Neil Patel wrote: > > Hi All, I am getting this error when trying to build latest git on Mac OSX > Lion, after fresh checkout and running ./boostraph.sh and ./configure: > > making all mod_amr > Creating mod_amr.so... > i686-apple-darwin11-llvm-gcc-4.2: -bundle not allowed with -dynamiclib > gcc -I/usr/local/src/freeswitch/libs/curl/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/libs/libteletone/src -Werror > -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb > -DMACOSX -DHAVE_OPENSSL -g -O2 -Wall -std=c99 -pedantic > -Wdeclaration-after-statement -D_GNU_SOURCE -DAMR_PASSTHROUGH -shared -o > .libs/mod_amr.so -dynamic -bundle -force-flat-namespace .libs/mod_amr.o > /usr/local/src/freeswitch/.libs/libfreeswitch.dylib -lodbc -lresolv > -L/usr/local/src/freeswitch/libs/apr-util/xml/expat/lib -lpq > /usr/local/src/freeswitch/libs/apr-util/xml/expat/lib/.libs/libexpat.a > /usr/local/src/freeswitch/libs/apr/.libs/libapr-1.a -lpthread -lm > -L/usr/local/src/freeswitch/libs/srtp -ldl -lssl -lcrypto -lz -lncurses > /usr/local/lib/libjpeg.dylib /usr/local/lib/libodbc.dylib -liconv > make[5]: *** [mod_amr.so] Error 1 > make[4]: *** [all] Error 1 > make[3]: *** [mod_amr-all] Error 1 > make[2]: *** [all-recursive] Error 1 > make[1]: *** [all-recursive] Error 1 > make: *** [all] Error 2 > > Heeding Mario's advice, I tried making with open-ssl flag, but that didn't > help. If I go about commenting out offending modules, the same "bundle not > allowed with -dynamiclib" appears for any number of other mods. I have > installed all the prerequisites for Mac OSX (to my knowledge), including > Apple and Unix dev tools. I am 100% macports/fink/brew free, AFAIK. Jira > filed, but no response for a while: > > http://jira.freeswitch.org/browse/FS-4240 > > My sense this is a missing tool or config issue with my setup (pretty much > fresh Lion install). Thanks in advance for your help! > Neil > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120526/283f7db3/attachment.html From neilp at cs.stanford.edu Sat May 26 20:56:03 2012 From: neilp at cs.stanford.edu (Neil Patel) Date: Sat, 26 May 2012 09:56:03 -0700 Subject: [Freeswitch-users] libtool on Mac OSX Lion Message-ID: Hi All, another OSX Lion build issue: On bootstrap I'm getting this output: anitya:freeswitch neil$ ./bootstrap.sh bootstrap: checking installation... bootstrap: autoconf version 2.59 (ok) bootstrap: automake version 1.7.1 (ok) bootstrap: aclocal version 1.7.1 (ok) bootstrap: libtool version 1.5.24 (ok) Bootstrapping using: autoconf : /usr/local/bin/autoconf automake : /usr/local/bin/automake aclocal : /usr/local/bin/aclocal libtool : /usr/bin/glibtool (1.5.24.) libtoolize: /usr/bin/glibtoolize make : /usr/bin/make (GNU Make 3.81) awk : () Entering directory /usr/local/src/freeswitch/libs/apr Copying libtool helper files ... Using `AC_PROG_RANLIB' is rendered obsolete by `AC_PROG_LIBTOOL' ... Remember to add `AC_PROG_LIBTOOL' to `configure.ac'. You should add the contents of `/usr/share/aclocal/libtool.m4' to `aclocal.m4'. Creating configure bootstrap: autoconf configure:6475: error: possibly undefined macro: AC_PROG_LIBTOOL If this token and others are legitimate, please use m4_pattern_allow. See the Autoconf documentation. ... I believe this is behind a build error I'm currently up against on mod_enum: making all mod_enum checking for i386-apple-darwin11.3.0-gcc... gcc checking for C compiler default output file name... a.out checking whether the C compiler works... yes checking whether we are cross compiling... no checking for suffix of executables... checking for suffix of object files... o checking whether we are using the GNU C compiler... yes checking whether gcc accepts -g... yes checking for gcc option to accept ANSI C... none needed checking how to run the C preprocessor... gcc -E checking for egrep... grep -E checking for AIX... no rm: conftest.dSYM: is a directory checking for i386-apple-darwin11.3.0-gcc... (cached) gcc checking whether we are using the GNU C compiler... (cached) yes checking whether gcc accepts -g... (cached) yes checking for gcc option to accept ANSI C... (cached) none needed checking whether make sets $(MAKE)... yes checking whether gcc supports -std=c99... yes checking whether gcc supports -xc99... no checking for an ANSI C-conforming const... yes checking whether gcc supports -g... yes checking whether gcc supports -O2... yes checking whether gcc supports -Wall... yes checking whether gcc supports -W... yes checking whether gcc supports -Wwrite-strings... yes checking for ANSI C header files... rm: conftest.dSYM: is a directory rm: conftest.dSYM: is a directory yes checking for sys/types.h... yes checking for sys/stat.h... yes checking for stdlib.h... yes checking for string.h... yes checking for memory.h... yes checking for strings.h... yes checking for inttypes.h... yes checking for stdint.h... yes checking for unistd.h... yes checking for getopt.h... yes checking for time.h... yes checking for winsock2.h... no checking for ws2tcpip.h... no checking whether gcc supports -Werror... yes checking whether gcc supports -Wall... (cached) yes checking whether gcc supports -std=c99... (cached) yes checking whether gcc supports -xc99... (cached) no checking for getopt.h... (cached) yes checking for time.h... (cached) yes checking whether we need -std=c99 -D__EXTENSIONS__ -D_BSD_SOURCE -D_POSIX_C_SOURCE=200112 -D_XOPEN_SOURCE=600 -D_XOPEN_SOURCE_EXTENDED=1 -D_ALL_SOURCE as a flag for gcc... no checking whether we need -std=c99 -D__EXTENSIONS__ -D_BSD_SOURCE -D_POSIX_C_SOURCE=200112 -D_XOPEN_SOURCE=600 -D_ALL_SOURCE as a flag for gcc... no checking whether we need -std=c99 as a flag for gcc... no checking whether we need -D_BSD_SOURCE as a flag for gcc... no checking whether we need -D_GNU_SOURCE as a flag for gcc... no checking whether we need -D_GNU_SOURCE -D_FRSRESGID as a flag for gcc... failed checking whether we need -D_POSIX_C_SOURCE=200112 as a flag for gcc... no checking whether we need -D__EXTENSIONS__ as a flag for gcc... no checking for inline... inline checking for int8_t... yes checking for int16_t... yes checking for int32_t... yes checking for int64_t... yes checking for uint8_t... yes checking for uint16_t... yes checking for uint32_t... yes checking for uint64_t... yes checking for doxygen... no checking for library containing socket... none required checking for library containing inet_pton... none required checking build system type... i386-apple-darwin11.3.0 checking host system type... i386-apple-darwin11.3.0 checking for i386-apple-darwin11.3.0-ar... no checking for ar... /usr/bin/ar /usr/local/src/freeswitch/libs/ldns/configure: line 6475: AC_PROG_LIBTOOL: command not found checking for openssl/ssl.h... yes checking for openssl/err.h... yes checking for openssl/rand.h... yes checking for EVP_sha256... no checking whether byte ordering is bigendian... no checking for ANSI C header files... (cached) yes checking for getopt.h... (cached) yes checking for stdarg.h... yes checking for stdbool.h... yes checking for openssl/ssl.h... (cached) yes checking for netinet/in.h... yes checking for time.h... (cached) yes checking for arpa/inet.h... yes checking for netdb.h... yes checking for sys/param.h... yes checking for sys/mount.h... yes checking for sys/socket.h... yes checking for inttypes.h... (cached) yes checking for sys/types.h... (cached) yes checking for unistd.h... (cached) yes checking for socklen_t... yes checking for ssize_t... yes checking for in_addr_t... yes checking for in_port_t... yes checking for struct sockaddr_storage.ss_family... yes checking for stdlib.h... (cached) yes checking for GNU libc compatible malloc... yes checking for stdlib.h... (cached) yes checking for GNU libc compatible realloc... yes checking for b64_pton... no checking for b64_ntop... no checking for b32_pton... no checking for b32_ntop... no checking for timegm... yes checking for gmtime_r... yes checking for ctime_r... yes checking for isblank... yes checking for isascii... yes checking for inet_aton... yes checking for inet_pton... yes checking for inet_ntop... yes checking for snprintf... yes checking for strlcpy... yes checking for memmove... yes checking for endprotoent... yes checking for endservent... yes checking for sleep... yes checking for random... yes checking for fcntl... yes checking for strtoul... yes checking for getaddrinfo... yes checking for ioctlsocket... no checking whether the C compiler (gcc) accepts the "format" attribute... yes checking whether the C compiler (gcc) accepts the "unused" attribute... yes configure: creating ./config.status config.status: creating Makefile config.status: creating ldns/net.h config.status: creating ldns/util.h config.status: creating packaging/libldns.pc config.status: creating packaging/ldns-config config.status: creating ldns/config.h copying header files make[6]: ./libtool: No such file or directory make[6]: *** [rdata.o] Error 1 make[5]: *** [/usr/local/src/freeswitch/libs/ldns/libldns.la] Error 2 make[4]: *** [all] Error 1 make[3]: *** [mod_enum-all] Error 1 make[2]: *** [all-recursive] Error 1 make[1]: *** [all-recursive] Error 1 make: *** [all] Error 2 I don't remember exactly how I installed libtool, but it was a script I ran from within FS that automatically wget'd, compiled, and installed it (suggested by wiki or mailing list). It definitely wasn't through macports/fink/brew. Any suggestions on what the problem may be? Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120526/d120135c/attachment-0001.html From krice at freeswitch.org Sat May 26 21:52:38 2012 From: krice at freeswitch.org (Ken Rice) Date: Sat, 26 May 2012 12:52:38 -0500 Subject: [Freeswitch-users] libtool on Mac OSX Lion In-Reply-To: Message-ID: Are you using the Apple Developers Kit or are you using something else? On 5/26/12 11:56 AM, "Neil Patel" wrote: > Hi All, another OSX Lion build issue: > > On bootstrap I'm getting this output: > > anitya:freeswitch neil$ ./bootstrap.sh? > bootstrap: checking installation... > bootstrap: autoconf version 2.59 (ok) > bootstrap: automake version 1.7.1 (ok) > bootstrap: aclocal version 1.7.1 (ok) > bootstrap: libtool version 1.5.24 (ok) > Bootstrapping using: > ? autoconf ?: /usr/local/bin/autoconf > ? automake ?: /usr/local/bin/automake > ? aclocal ? : /usr/local/bin/aclocal > ??libtool ? : /usr/bin/glibtool (1.5.24.) > ? libtoolize: /usr/bin/glibtoolize > ? make ? ? ?: /usr/bin/make (GNU Make 3.81) > ? awk ? ? ? : ?() > > Entering directory /usr/local/src/freeswitch/libs/apr > Copying libtool helper files ... > Using `AC_PROG_RANLIB' is rendered obsolete by `AC_PROG_LIBTOOL' > ... > Remember to add `AC_PROG_LIBTOOL' to `configure.ac '. > You should add the contents of `/usr/share/aclocal/libtool.m4' to > `aclocal.m4'. > Creating configure > bootstrap: autoconf > configure:6475: error: possibly undefined macro: AC_PROG_LIBTOOL > ? ? ? If this token and others are legitimate, please use m4_pattern_allow. > ? ? ? See the Autoconf documentation. > ... > > I believe this is behind a build error I'm currently up against on mod_enum: > making all mod_enum > checking for i386-apple-darwin11.3.0-gcc... gcc > checking for C compiler default output file name... a.out > checking whether the C compiler works... yes > checking whether we are cross compiling... no > checking for suffix of executables...? > checking for suffix of object files... o > checking whether we are using the GNU C compiler... yes > checking whether gcc accepts -g... yes > checking for gcc option to accept ANSI C... none needed > checking how to run the C preprocessor... gcc -E > checking for egrep... grep -E > checking for AIX... no > rm: conftest.dSYM: is a directory > checking for i386-apple-darwin11.3.0-gcc... (cached) gcc > checking whether we are using the GNU C compiler... (cached) yes > checking whether gcc accepts -g... (cached) yes > checking for gcc option to accept ANSI C... (cached) none needed > checking whether make sets $(MAKE)... yes > checking whether gcc supports -std=c99... yes > checking whether gcc supports -xc99... no > checking for an ANSI C-conforming const... yes > checking whether gcc supports -g... yes > checking whether gcc supports -O2... yes > checking whether gcc supports -Wall... yes > checking whether gcc supports -W... yes > checking whether gcc supports -Wwrite-strings... yes > checking for ANSI C header files... rm: conftest.dSYM: is a directory > rm: conftest.dSYM: is a directory > yes > checking for sys/types.h... yes > checking for sys/stat.h... yes > checking for stdlib.h... yes > checking for string.h... yes > checking for memory.h... yes > checking for strings.h... yes > checking for inttypes.h... yes > checking for stdint.h... yes > checking for unistd.h... yes > checking for getopt.h... yes > checking for time.h... yes > checking for winsock2.h... no > checking for ws2tcpip.h... no > checking whether gcc supports -Werror... yes > checking whether gcc supports -Wall... (cached) yes > checking whether gcc supports -std=c99... (cached) yes > checking whether gcc supports -xc99... (cached) no > checking for getopt.h... (cached) yes > checking for time.h... (cached) yes > checking whether we need -std=c99 -D__EXTENSIONS__ -D_BSD_SOURCE > -D_POSIX_C_SOURCE=200112 -D_XOPEN_SOURCE=600 -D_XOPEN_SOURCE_EXTENDED=1 > -D_ALL_SOURCE as a flag for gcc... no > checking whether we need -std=c99 -D__EXTENSIONS__ -D_BSD_SOURCE > -D_POSIX_C_SOURCE=200112 -D_XOPEN_SOURCE=600 -D_ALL_SOURCE as a flag for > gcc... no > checking whether we need -std=c99 as a flag for gcc... no > checking whether we need -D_BSD_SOURCE as a flag for gcc... no > checking whether we need -D_GNU_SOURCE as a flag for gcc... no > checking whether we need -D_GNU_SOURCE -D_FRSRESGID as a flag for gcc... > failed > checking whether we need -D_POSIX_C_SOURCE=200112 as a flag for gcc... no > checking whether we need -D__EXTENSIONS__ as a flag for gcc... no > checking for inline... inline > checking for int8_t... yes > checking for int16_t... yes > checking for int32_t... yes > checking for int64_t... yes > checking for uint8_t... yes > checking for uint16_t... yes > checking for uint32_t... yes > checking for uint64_t... yes > checking for doxygen... no > checking for library containing socket... none required > checking for library containing inet_pton... none required > checking build system type... i386-apple-darwin11.3.0 > checking host system type... i386-apple-darwin11.3.0 > checking for i386-apple-darwin11.3.0-ar... no > checking for ar... /usr/bin/ar > /usr/local/src/freeswitch/libs/ldns/configure: line 6475: AC_PROG_LIBTOOL: > command not found > checking for openssl/ssl.h... yes > checking for openssl/err.h... yes > checking for openssl/rand.h... yes > checking for EVP_sha256... no > checking whether byte ordering is bigendian... no > checking for ANSI C header files... (cached) yes > checking for getopt.h... (cached) yes > checking for stdarg.h... yes > checking for stdbool.h... yes > checking for openssl/ssl.h... (cached) yes > checking for netinet/in.h... yes > checking for time.h... (cached) yes > checking for arpa/inet.h... yes > checking for netdb.h... yes > checking for sys/param.h... yes > checking for sys/mount.h... yes > checking for sys/socket.h... yes > checking for inttypes.h... (cached) yes > checking for sys/types.h... (cached) yes > checking for unistd.h... (cached) yes > checking for socklen_t... yes > checking for ssize_t... yes > checking for in_addr_t... yes > checking for in_port_t... yes > checking for struct sockaddr_storage.ss_family... yes > checking for stdlib.h... (cached) yes > checking for GNU libc compatible malloc... yes > checking for stdlib.h... (cached) yes > checking for GNU libc compatible realloc... yes > checking for b64_pton... no > checking for b64_ntop... no > checking for b32_pton... no > checking for b32_ntop... no > checking for timegm... yes > checking for gmtime_r... yes > checking for ctime_r... yes > checking for isblank... yes > checking for isascii... yes > checking for inet_aton... yes > checking for inet_pton... yes > checking for inet_ntop... yes > checking for snprintf... yes > checking for strlcpy... yes > checking for memmove... yes > checking for endprotoent... yes > checking for endservent... yes > checking for sleep... yes > checking for random... yes > checking for fcntl... yes > checking for strtoul... yes > checking for getaddrinfo... yes > checking for ioctlsocket... no > checking whether the C compiler (gcc) accepts the "format" attribute... yes > checking whether the C compiler (gcc) accepts the "unused" attribute... yes > configure: creating ./config.status > config.status: creating Makefile > config.status: creating ldns/net.h > config.status: creating ldns/util.h > config.status: creating packaging/libldns.pc > config.status: creating packaging/ldns-config > config.status: creating ldns/config.h > copying header files > make[6]: ./libtool: No such file or directory > make[6]: *** [rdata.o] Error 1 > make[5]: *** [/usr/local/src/freeswitch/libs/ldns/libldns.la > ] Error 2 > make[4]: *** [all] Error 1 > make[3]: *** [mod_enum-all] Error 1 > make[2]: *** [all-recursive] Error 1 > make[1]: *** [all-recursive] Error 1 > make: *** [all] Error 2 > > I don't remember exactly how I installed libtool, but it was a script I ran > from within FS that automatically wget'd, compiled, and installed it > (suggested by wiki or mailing list). It definitely wasn't through > macports/fink/brew. > > Any suggestions on what the problem may be? > > Thanks! > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120526/60e91c8b/attachment.html From neilp at cs.stanford.edu Sat May 26 21:56:11 2012 From: neilp at cs.stanford.edu (Neil Patel) Date: Sat, 26 May 2012 10:56:11 -0700 Subject: [Freeswitch-users] libtool on Mac OSX Lion In-Reply-To: References: Message-ID: Developer's kit. I have nothing installed using ports/brew/fink On Sat, May 26, 2012 at 10:52 AM, Ken Rice wrote: > Are you using the Apple Developers Kit or are you using something else? > > > > On 5/26/12 11:56 AM, "Neil Patel" wrote: > > Hi All, another OSX Lion build issue: > > On bootstrap I'm getting this output: > > anitya:freeswitch neil$ ./bootstrap.sh > bootstrap: checking installation... > bootstrap: autoconf version 2.59 (ok) > bootstrap: automake version 1.7.1 (ok) > bootstrap: aclocal version 1.7.1 (ok) > bootstrap: libtool version 1.5.24 (ok) > Bootstrapping using: > autoconf : /usr/local/bin/autoconf > automake : /usr/local/bin/automake > aclocal : /usr/local/bin/aclocal > libtool : /usr/bin/glibtool (1.5.24.) > libtoolize: /usr/bin/glibtoolize > make : /usr/bin/make (GNU Make 3.81) > awk : () > > Entering directory /usr/local/src/freeswitch/libs/apr > Copying libtool helper files ... > Using `AC_PROG_RANLIB' is rendered obsolete by `AC_PROG_LIBTOOL' > ... > Remember to add `AC_PROG_LIBTOOL' to `configure.ac > '. > > You should add the contents of `/usr/share/aclocal/libtool.m4' to > `aclocal.m4'. > Creating configure > bootstrap: autoconf > configure:6475: error: possibly undefined macro: AC_PROG_LIBTOOL > If this token and others are legitimate, please use m4_pattern_allow. > See the Autoconf documentation. > ... > > I believe this is behind a build error I'm currently up against on > mod_enum: > making all mod_enum > checking for i386-apple-darwin11.3.0-gcc... gcc > checking for C compiler default output file name... a.out > checking whether the C compiler works... yes > checking whether we are cross compiling... no > checking for suffix of executables... > checking for suffix of object files... o > checking whether we are using the GNU C compiler... yes > checking whether gcc accepts -g... yes > checking for gcc option to accept ANSI C... none needed > checking how to run the C preprocessor... gcc -E > checking for egrep... grep -E > checking for AIX... no > rm: conftest.dSYM: is a directory > checking for i386-apple-darwin11.3.0-gcc... (cached) gcc > checking whether we are using the GNU C compiler... (cached) yes > checking whether gcc accepts -g... (cached) yes > checking for gcc option to accept ANSI C... (cached) none needed > checking whether make sets $(MAKE)... yes > checking whether gcc supports -std=c99... yes > checking whether gcc supports -xc99... no > checking for an ANSI C-conforming const... yes > checking whether gcc supports -g... yes > checking whether gcc supports -O2... yes > checking whether gcc supports -Wall... yes > checking whether gcc supports -W... yes > checking whether gcc supports -Wwrite-strings... yes > checking for ANSI C header files... rm: conftest.dSYM: is a directory > rm: conftest.dSYM: is a directory > yes > checking for sys/types.h... yes > checking for sys/stat.h... yes > checking for stdlib.h... yes > checking for string.h... yes > checking for memory.h... yes > checking for strings.h... yes > checking for inttypes.h... yes > checking for stdint.h... yes > checking for unistd.h... yes > checking for getopt.h... yes > checking for time.h... yes > checking for winsock2.h... no > checking for ws2tcpip.h... no > checking whether gcc supports -Werror... yes > checking whether gcc supports -Wall... (cached) yes > checking whether gcc supports -std=c99... (cached) yes > checking whether gcc supports -xc99... (cached) no > checking for getopt.h... (cached) yes > checking for time.h... (cached) yes > checking whether we need -std=c99 -D__EXTENSIONS__ -D_BSD_SOURCE > -D_POSIX_C_SOURCE=200112 -D_XOPEN_SOURCE=600 -D_XOPEN_SOURCE_EXTENDED=1 > -D_ALL_SOURCE as a flag for gcc... no > checking whether we need -std=c99 -D__EXTENSIONS__ -D_BSD_SOURCE > -D_POSIX_C_SOURCE=200112 -D_XOPEN_SOURCE=600 -D_ALL_SOURCE as a flag for > gcc... no > checking whether we need -std=c99 as a flag for gcc... no > checking whether we need -D_BSD_SOURCE as a flag for gcc... no > checking whether we need -D_GNU_SOURCE as a flag for gcc... no > checking whether we need -D_GNU_SOURCE -D_FRSRESGID as a flag for gcc... > failed > checking whether we need -D_POSIX_C_SOURCE=200112 as a flag for gcc... no > checking whether we need -D__EXTENSIONS__ as a flag for gcc... no > checking for inline... inline > checking for int8_t... yes > checking for int16_t... yes > checking for int32_t... yes > checking for int64_t... yes > checking for uint8_t... yes > checking for uint16_t... yes > checking for uint32_t... yes > checking for uint64_t... yes > checking for doxygen... no > checking for library containing socket... none required > checking for library containing inet_pton... none required > checking build system type... i386-apple-darwin11.3.0 > checking host system type... i386-apple-darwin11.3.0 > checking for i386-apple-darwin11.3.0-ar... no > checking for ar... /usr/bin/ar > /usr/local/src/freeswitch/libs/ldns/configure: line 6475: AC_PROG_LIBTOOL: > command not found > checking for openssl/ssl.h... yes > checking for openssl/err.h... yes > checking for openssl/rand.h... yes > checking for EVP_sha256... no > checking whether byte ordering is bigendian... no > checking for ANSI C header files... (cached) yes > checking for getopt.h... (cached) yes > checking for stdarg.h... yes > checking for stdbool.h... yes > checking for openssl/ssl.h... (cached) yes > checking for netinet/in.h... yes > checking for time.h... (cached) yes > checking for arpa/inet.h... yes > checking for netdb.h... yes > checking for sys/param.h... yes > checking for sys/mount.h... yes > checking for sys/socket.h... yes > checking for inttypes.h... (cached) yes > checking for sys/types.h... (cached) yes > checking for unistd.h... (cached) yes > checking for socklen_t... yes > checking for ssize_t... yes > checking for in_addr_t... yes > checking for in_port_t... yes > checking for struct sockaddr_storage.ss_family... yes > checking for stdlib.h... (cached) yes > checking for GNU libc compatible malloc... yes > checking for stdlib.h... (cached) yes > checking for GNU libc compatible realloc... yes > checking for b64_pton... no > checking for b64_ntop... no > checking for b32_pton... no > checking for b32_ntop... no > checking for timegm... yes > checking for gmtime_r... yes > checking for ctime_r... yes > checking for isblank... yes > checking for isascii... yes > checking for inet_aton... yes > checking for inet_pton... yes > checking for inet_ntop... yes > checking for snprintf... yes > checking for strlcpy... yes > checking for memmove... yes > checking for endprotoent... yes > checking for endservent... yes > checking for sleep... yes > checking for random... yes > checking for fcntl... yes > checking for strtoul... yes > checking for getaddrinfo... yes > checking for ioctlsocket... no > checking whether the C compiler (gcc) accepts the "format" attribute... yes > checking whether the C compiler (gcc) accepts the "unused" attribute... yes > configure: creating ./config.status > config.status: creating Makefile > config.status: creating ldns/net.h > config.status: creating ldns/util.h > config.status: creating packaging/libldns.pc > config.status: creating packaging/ldns-config > config.status: creating ldns/config.h > copying header files > make[6]: ./libtool: No such file or directory > make[6]: *** [rdata.o] Error 1 > make[5]: *** [/usr/local/src/freeswitch/libs/ldns/libldns.la < > http://libldns.la/> ] Error 2 > > make[4]: *** [all] Error 1 > make[3]: *** [mod_enum-all] Error 1 > make[2]: *** [all-recursive] Error 1 > make[1]: *** [all-recursive] Error 1 > make: *** [all] Error 2 > > I don't remember exactly how I installed libtool, but it was a script I > ran from within FS that automatically wget'd, compiled, and installed it > (suggested by wiki or mailing list). It definitely wasn't through > macports/fink/brew. > > Any suggestions on what the problem may be? > > Thanks! > > ------------------------------ > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120526/1bdfbfae/attachment-0001.html From msc at freeswitch.org Sun May 27 01:29:00 2012 From: msc at freeswitch.org (Michael Collins) Date: Sat, 26 May 2012 14:29:00 -0700 Subject: [Freeswitch-users] sofia current calls for a user In-Reply-To: References: Message-ID: On Tue, May 22, 2012 at 8:30 AM, Brian Foster wrote: > http://wiki.freeswitch.org/wiki/Limit > On May 22, 2012 11:25 AM, "Brian Foster" wrote: > >> As stated by MC, mod_limit fulfills that role as a side effect. >> >> -BDF >> > FYI, I updated the limit wiki page to line up with what's in the source code, namely that if you set max to zero or a negative value that the limit is not a "limit" but basically just a counter. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120526/8d08ad8c/attachment.html From all.eforums at gmail.com Sun May 27 02:13:18 2012 From: all.eforums at gmail.com (A E [Gmail]) Date: Sat, 26 May 2012 18:13:18 -0400 Subject: [Freeswitch-users] Common/Reasonable Assumption on DID/Channel over-subscription Message-ID: Hello All, just throwing this out there. What are people generally using these days when designing their services, esp. those that require a user to call a DID to access their system, similar to calling card services. There was a time when this used to be 50 to 1 for DIDs, and about 10 to 1 for number of channels bought in SMB with IP-PBX. I believe this would have changed today and assuming a service is pretty popular, the ALOCs are longer due to cheaper rates and convenience of calling. Does anyone have any real world numbers they can share? Is 10 to 1 a good ratio to ensure a user practically never gets a "circuits are busy"? Thanks in advance -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120526/29ee7ce2/attachment.html From nathandownes at hotmail.com Sun May 27 02:28:48 2012 From: nathandownes at hotmail.com (Mr Nathan Downes) Date: Sun, 27 May 2012 08:28:48 +1000 Subject: [Freeswitch-users] RTP media issue In-Reply-To: <014401cd3b43$d4004ed0$7c00ec70$@visionutveckling.se> References: <02a201cd3a1a$2a658770$7f309650$@gmail.com> <007301cd3a96$d106ed90$7314c8b0$@gmail.com>, <014401cd3b43$d4004ed0$7c00ec70$@visionutveckling.se> Message-ID: Hi Peter, Is it possible using pcapsipdump is causing it to look like that? I will be onsite again tomorrow and I will do a few tshark captures on both sides, I just have to start and stop it after each call, as they get huge if I leave it running. I can also get the capture from the provider, and ones from the near end and far end of the fibre network. I did find where to put cng_plc in public.xml as an absolute condition so it runs on each inbound call, the same with that jitterbuffer statement. Should I take them out just for this tracing? Thank you for your help. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Peter Olsson Sent: Saturday, 26 May 2012 10:42 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] RTP media issue I think the first step is to check the actual signalling. Just as Anthony mentions, the call setup looks weird. Between FS and the ATA (if I understood the different IP's/peers correctly) it seems FS sends early media (183), and it takes the ATA 10 seconds to ACK this. During the time the call is answered on the other end, which dosn't seem to be passed to the ATA (probably because it's so late with the ACK - I'm not sure about that though). So, first check this - it seems to me that FS might still be in "early media" state during the entire call. About the actual RTP, I see one problem here, it's packet 8146 in outbound.pcap. This is a packet that is probably generated by FS (since by that exact time, you're missing one packet from the provider), the problem here is that the timestamp if way off (I'm not sure where FS gets this ts from). It's because of this faulty timestamp that a few packets after this is dropped, since FS won't send packets with a lower timestamp than before. This is also why packet 8152 seems to have a strange timestamp, but it's actually just passed from the other leg, and is the first timestamp that is greater then packet 8146. One possible solution would be to enable rewrite of timestamps (rtp-rewrite-timestamps in the sofia config), also mentioned here: http://wiki.freeswitch.org/wiki/RTP_Issues#Dropped_Audio However, I think you also need to check the strangness in the SIP signalling. /Peter ________________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] för Mr Nathan Downes [nathandownes at hotmail.com] Skickat: den 26 maj 2012 00:03 Till: 'FreeSWITCH Users Help' ?mne: Re: [Freeswitch-users] RTP media issue Hi Anthony, FS version = FreeSWITCH Version 1.1.beta1 (git-f1b5044 2012-04-26 11-28-47 -0500) I don't have a debug log, but I could probably get it with another trace of both sides of the call, but it would be hard to capture as there is constant calls to this, unless there is a way to do it on a per call basis? I can also only do testing onsite as we don't have the same fibre equipment. I already have jitterbuffer set in both profiles in an attempt to try and stop it using , is there a way to set the cng_plc in the profile itself rather than diaplpan as there are 70 or so numbers in it. In the outbound dialplans I also added because I kept seeing PAUSE JITTERBUFFER in the FS logs when calls were made outbound so I wasn't sure it was doing something and read somewhere it pauses it when it bridges the call The inbound dialplan for all of those people consists of It doesn't affect SIP phones or normal ATA devices we have connected and only affects these FTTH GPON ATA's, but with almost 100 residents in this retirement village and them constantly complaining we have been given til Wednesday to come up with a solution or risk losing our position as the internet/phone provider for that retirement village. It appeared to me that what happens in that trace isn't normal behaviour, I did try rewriting timestamps last week, but as you suggested that appeared to mask the issue but not stop it from happening. That was when I was losing a packet from them each second or so, which by the time it arrived to end user sounded horrible. It has settled down a lot now and maybe 1 or 2 packets per call, but if what is in this trace is the cause each time, that would explain the poor end users experience. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Saturday, 26 May 2012 2:06 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] RTP media issue What version of FS are you running? Do you have the debug logs of those calls? you could try using the jitterbuffer. in the inbound DP to FS *before* you answer. Also it looks a little odd to me in this trace if this is the same call, it seems like you answered the call before placing the call to the phone and that phone never answers.... On Thu, May 24, 2012 at 9:51 PM, Nathan Downes wrote: > Hi, > > enable-soa > > > > Set the value to "false" to diable SIP SOA from sofia to tell sofia > not to touch the exchange of SDP > > I don't think this is related to the exchange of an SDP message.. Can > you elaborate more before I try it? I can't make things worse or > change things I don't understand. > > ________________________________ > From: djbinter at gmail.com > To: freeswitch-users at lists.freeswitch.org > CC: nathan at nortec.com.au > Subject: Re: [Freeswitch-users] RTP media issue > Date: Fri, 25 May 2012 11:19:46 +1000 > > > > > > Sent from my iPad > > On May 24, 2012, at 5:01 PM, Nathan Downes wrote: > > Hi, > > I had previous reported an issue with poor voice quality, appearing to > stem from occasion wrong timestamps coming from provider, but the end > user's experience was much worse than what I could see/hear in the trace. > > I have finally captured an event inbound and outbound. The thing I > don't understand is I thought even though FS proxied the media it > didn't touch it or change anything, but it appears it is. > > The 2 traces are http://www.nortec.com.au/inbound.pcap.gz and > http://www.nortec.com.au/outbound.pcap.gz > > Inbound is from my trunk provider to FS box and outbound is FS box to > ATA in FTTH GPON. > > The event I am talking about, if both traces are open, is in the > inbound one inbetween packet 8114 and 8117 the provider drops a packet > or I don't receive it. In the corresponding outbound trace, between > packet 8144 and 8152, it appears FS misses a whole heap of packets > (.1 seconds) between > 8146 and 8152 then it increases the timestamp only by 40 rather than > 160 on packet 8152. This seems to not affect SIP phones themselves > but causes issues with the FTTH GPON ATA. > > This causes a gap in the audio for the end user, and when they miss a > high number of packets even though it sounds good on the inbound trace > the end users experience is horrible. This trace is actually a good > one, but the wrong timestamp can occur once per second, causing end > user to lose 10%+ of incoming audio only. The issue only affects the > audio coming from provider to FS to end user. > > I am chasing it up with the voice provider to try and eliminate the > occasional packet loss, but if I could stop/fix FS from doing its > adjustment/gap/something the end user wouldn't even notice it. > > > > ______________________________________________________________________ > ___ > > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org > > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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 at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4fbfffef32764666710318! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From anthony.minessale at gmail.com Sun May 27 04:29:01 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 26 May 2012 19:29:01 -0500 Subject: [Freeswitch-users] RTP media issue In-Reply-To: <1FFF97C269757C458224B7C895F35F150D9857@cantor.std.visionutv.se> References: <02a201cd3a1a$2a658770$7f309650$@gmail.com> <007301cd3a96$d106ed90$7314c8b0$@gmail.com> <1FFF97C269757C458224B7C895F35F150D9857@cantor.std.visionutv.se> Message-ID: Also update, you are running git head so you have to update frequently, you rev is a month old. On May 26, 2012 7:43 AM, "Peter Olsson" wrote: > I think the first step is to check the actual signalling. > > Just as Anthony mentions, the call setup looks weird. Between FS and the > ATA (if I understood the different IP's/peers correctly) it seems FS sends > early media (183), and it takes the ATA 10 seconds to ACK this. During the > time the call is answered on the other end, which dosn't seem to be passed > to the ATA (probably because it's so late with the ACK - I'm not sure about > that though). So, first check this - it seems to me that FS might still be > in "early media" state during the entire call. > > About the actual RTP, I see one problem here, it's packet 8146 in > outbound.pcap. This is a packet that is probably generated by FS (since by > that exact time, you're missing one packet from the provider), the problem > here is that the timestamp if way off (I'm not sure where FS gets this ts > from). It's because of this faulty timestamp that a few packets after this > is dropped, since FS won't send packets with a lower timestamp than before. > This is also why packet 8152 seems to have a strange timestamp, but it's > actually just passed from the other leg, and is the first timestamp that is > greater then packet 8146. > > One possible solution would be to enable rewrite of timestamps > (rtp-rewrite-timestamps in the sofia config), also mentioned here: > http://wiki.freeswitch.org/wiki/RTP_Issues#Dropped_Audio > > However, I think you also need to check the strangness in the SIP > signalling. > > /Peter > > ________________________________________ > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [ > freeswitch-users-bounces at lists.freeswitch.org] för Mr Nathan Downes [ > nathandownes at hotmail.com] > Skickat: den 26 maj 2012 00:03 > Till: 'FreeSWITCH Users Help' > ?mne: Re: [Freeswitch-users] RTP media issue > > Hi Anthony, > > FS version = FreeSWITCH Version 1.1.beta1 (git-f1b5044 2012-04-26 11-28-47 > -0500) > > I don't have a debug log, but I could probably get it with another trace of > both sides of the call, but it would be hard to capture as there is > constant > calls to this, unless there is a way to do it on a per call basis? I can > also only do testing onsite as we don't have the same fibre equipment. I > already have jitterbuffer set in both profiles in an attempt to try and > stop > it using , is there a way > to set the cng_plc in the profile itself rather than diaplpan as there are > 70 or so numbers in it. In the outbound dialplans I also added application="set" data="sip_jitter_buffer_during_bridge=true" /> because I > kept seeing PAUSE JITTERBUFFER in the FS logs when calls were made outbound > so I wasn't sure it was doing something and read somewhere it pauses it > when > it bridges the call > > The inbound dialplan for all of those people consists of > > > > > > > > > > > > > It doesn't affect SIP phones or normal ATA devices we have connected and > only affects these FTTH GPON ATA's, but with almost 100 residents in this > retirement village and them constantly complaining we have been given til > Wednesday to come up with a solution or risk losing our position as the > internet/phone provider for that retirement village. > > It appeared to me that what happens in that trace isn't normal behaviour, I > did try rewriting timestamps last week, but as you suggested that appeared > to mask the issue but not stop it from happening. That was when I was > losing a packet from them each second or so, which by the time it arrived > to > end user sounded horrible. It has settled down a lot now and maybe 1 or 2 > packets per call, but if what is in this trace is the cause each time, that > would explain the poor end users experience. > > > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Anthony > Minessale > Sent: Saturday, 26 May 2012 2:06 AM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] RTP media issue > > What version of FS are you running? > Do you have the debug logs of those calls? > > you could try using the jitterbuffer. > > > > in the inbound DP to FS *before* you answer. > > Also it looks a little odd to me in this trace if this is the same call, it > seems like you answered the call before placing the call to the phone and > that phone never answers.... > > > > > > > > > On Thu, May 24, 2012 at 9:51 PM, Nathan Downes > wrote: > > Hi, > > > > enable-soa > > > > > > > > Set the value to "false" to diable SIP SOA from sofia to tell sofia > > not to touch the exchange of SDP > > > > I don't think this is related to the exchange of an SDP message.. Can > > you elaborate more before I try it? I can't make things worse or > > change things I don't understand. > > > > ________________________________ > > From: djbinter at gmail.com > > To: freeswitch-users at lists.freeswitch.org > > CC: nathan at nortec.com.au > > Subject: Re: [Freeswitch-users] RTP media issue > > Date: Fri, 25 May 2012 11:19:46 +1000 > > > > > > > > > > > > Sent from my iPad > > > > On May 24, 2012, at 5:01 PM, Nathan Downes > wrote: > > > > Hi, > > > > I had previous reported an issue with poor voice quality, appearing to > > stem from occasion wrong timestamps coming from provider, but the end > > user's experience was much worse than what I could see/hear in the trace. > > > > I have finally captured an event inbound and outbound. The thing I > > don't understand is I thought even though FS proxied the media it > > didn't touch it or change anything, but it appears it is. > > > > The 2 traces are http://www.nortec.com.au/inbound.pcap.gz and > > http://www.nortec.com.au/outbound.pcap.gz > > > > Inbound is from my trunk provider to FS box and outbound is FS box to > > ATA in FTTH GPON. > > > > The event I am talking about, if both traces are open, is in the > > inbound one inbetween packet 8114 and 8117 the provider drops a packet > > or I don't receive it. In the corresponding outbound trace, between > > packet 8144 and 8152, it appears FS misses a whole heap of packets > > (.1 seconds) between > > 8146 and 8152 then it increases the timestamp only by 40 rather than > > 160 on packet 8152. This seems to not affect SIP phones themselves > > but causes issues with the FTTH GPON ATA. > > > > This causes a gap in the audio for the end user, and when they miss a > > high number of packets even though it sounds good on the inbound trace > > the end users experience is horrible. This trace is actually a good > > one, but the wrong timestamp can occur once per second, causing end > > user to lose 10%+ of incoming audio only. The issue only affects the > > audio coming from provider to FS to end user. > > > > I am chasing it up with the voice provider to try and eliminate the > > occasional packet loss, but if I could stop/fix FS from doing its > > adjustment/gap/something the end user wouldn't even notice it. > > > > > > > > ______________________________________________________________________ > > ___ > > > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > > rs > > http://www.freeswitch.org > > > > > > ______________________________________________________________________ > > ___ Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > > rs > > http://www.freeswitch.org > > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > !DSPAM:4fbfffef32764666710318! > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120526/cb53ac03/attachment-0001.html From govoiper at gmail.com Sun May 27 09:36:41 2012 From: govoiper at gmail.com (SamyGo) Date: Sun, 27 May 2012 10:36:41 +0500 Subject: [Freeswitch-users] Freeswitch (SBC + Media) does not know that gateway phone is picked up In-Reply-To: <1338044784.74842.YahooMailClassic@web110809.mail.gq1.yahoo.com> References: <1338044784.74842.YahooMailClassic@web110809.mail.gq1.yahoo.com> Message-ID: OK Sheriff, I've gone through this. This says that destination on kamailio end is ringing and keeps on ringing state. What Im more interested in are the sip traces on port 5060 the story thats happening in between the destination phone and the kamailo..if u know what I mean !! On Sat, May 26, 2012 at 8:06 PM, Sherif Omran wrote: > Hi Sanny, > > Please see the following links > > http://pastebin.freeswitch.org/19175 > http://pastebin.freeswitch.org/19174 > > > regards, > Sherif > > > --- On *Sat, 5/26/12, SamyGo * wrote: > > > From: SamyGo > Subject: Re: [Freeswitch-users] Freeswitch (SBC + Media) does not know > that gateway phone is picked up > To: "FreeSWITCH Users Help" > Date: Saturday, May 26, 2012, 5:52 PM > > Hey Sherrif, > You might wanna resend the pcap file as I couldn;t find any single INVITE > or any call in that capture. Please review. > > BR > SaGo. > > On Sat, May 26, 2012 at 1:05 AM, Sherif Omran > > wrote: > > Hello Sammy > > FS uses port 6090 and registers the GW. When a call comes, it rings the > extension but then gives a busy signal and FS keeps ringing till I cancel > the call from the GW. > > > > 1. recv 477 bytes from udp/[217.74.179.29]:5060 at 19:52:05.662344: > 2. > ------------------------------------------------------------------------ > 3. CANCEL sip:gw+trunk_1000 at 78.138.90.58:6090;transport=udp;gw=trunk_1000 > SIP/2.0 > 4. Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bKe567.13b0b9e.0 > 5. Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK16d1a257;rport= > 5060 > 6. From: "+41793940965" >;tag=as1dccab06 > 7. To: > 8. Call-ID: 5cd37edb776a6b3a35e9713a453a3425 at bluesip.net > 9. CSeq: 102 CANCEL > 10. User-Agent: blueSIP PSTN GW > 11. Max-Forwards: 69 > 12. Content-Length: 0 > 13. P-hint: USRLOC > > > > > --- On *Fri, 5/25/12, SamyGo > >* wrote: > > > From: SamyGo > > > Subject: Re: [Freeswitch-users] Freeswitch (SBC + Media) does not know > that gateway phone is picked up > To: "FreeSWITCH Users Help" > > > Date: Friday, May 25, 2012, 10:46 AM > > Hi again, > If you want kamailio register to the provider then use UAC > module. > Kamailio will use the username/password and register with the provider. > > Regards, > Sammy > > > On Fri, May 25, 2012 at 12:20 PM, Sherif Omran > > wrote: > > Dear Sammy, > > Thank you for your question ... Yes, the GW is registered through FS > because I did not know how to register it to kamailio. But it seems better > to register it to kamailio. > One more information, calls from 1001 to 1002 go to kamailio then to FS > then return back to kamailio smoothly. Thus I would suggest that I change > registering the gateway from FS to kamailio. but How to? > > > > --- On *Fri, 5/25/12, SamyGo > >* wrote: > > > From: SamyGo > > > Subject: Re: [Freeswitch-users] Freeswitch (SBC + Media) does not know > that gateway phone is picked up > To: "FreeSWITCH Users Help" > > > Date: Friday, May 25, 2012, 8:08 AM > > Hi, > These are SIP traces on FreeSWITCH console, whereas you are saying and it > seems that kamailio is not detecting the answering (200 OK)of the call from > extension 1002. Please, can you take a sip trace..!! I see you've both > kamailio and FS on same server! Please take a pcap from the linux console > using the following command. > > #tcpdump -i any -s 0 -w new-call-trace.pcap -vvv > > Please be quick on turning the sip trace on and off as quickly as possible > to avoid extra packets. Once done open the file in wireshark > ; apply filter "sip || rtp" and then save the > resulting capture in separate file. Send us the new file to analyse. > > One more silly question probably, I see REGISTERs coming to your FS as > well and the calls to gateways are made from FS too !!, umm...just thinking > what are you using kamailio for!!? > > Thanks > BR, > Sammy > > > On Fri, May 25, 2012 at 4:40 AM, Sherif Omran > > wrote: > > Hi all, > > here is the siptrace: To figure it out > 1- gateway called bluesip.net. It send invide using caller > number at bluesip.net > 2- This call should go to extension kb-1002. kb means go from freeswitch > port 6090 to kamailio port 5060 > 3- It should go to call extension 1002 in Kamailio > 4- Extension 1002 rings but when I reply, it does not notice I replied > > > ./fs_cli > _____ ____ ____ _ ___ > | ___/ ___| / ___| | |_ _| > | |_ \___ \ | | | | | | > | _| ___) | | |___| |___ | | > |_| |____/ \____|_____|___| > > ******************************************************* > * Anthony Minessale II, Ken Rice, * > * Michael Jerris, Travis Cross * > * FreeSWITCH (http://www.freeswitch.org) * > * Paypal Donations Appreciated: paypal at freeswitch.org* > * Brought to you by ClueCon http://www.cluecon.com/ * > ******************************************************* > > Type /help to see a list of commands > > > +OK log level [7] > freeswitch at internal> tracelevel > -ERR tracelevel Command not found! > > freeswitch at internal> sofia global siptrace on > +OK Global siptrace on > recv 1104 bytes from udp/[217.74.179.29]:5060 at 23:08:45.822962: > ------------------------------------------------------------------------ > INVITE sip:gw+trunk_1000 at 78.138.90.58:6090;transport=udp;gw=trunk_1000 > SIP/2.0 > Record-Route: > Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 > Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 > From: "+41793940965" ;tag=as00589402 > To: > Contact: > Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net > CSeq: 102 INVITE > User-Agent: blueSIP PSTN GW > Max-Forwards: 69 > Date: Thu, 24 May 2012 23:08:44 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Content-Type: application/sdp > Content-Length: 367 > P-hint: USRLOC > > v=0 > o=root 20076 20076 IN IP4 217.74.179.28 > s=session > c=IN IP4 217.74.179.28 > t=0 0 > m=audio 25626 RTP/AVP 8 0 18 111 3 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:111 G726-32/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > a=sendrecv > ------------------------------------------------------------------------ > send 494 bytes to udp/[217.74.179.29]:5060 at 23:08:45.844486: > ------------------------------------------------------------------------ > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 > Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 > Record-Route: > From: "+41793940965" ;tag=as00589402 > To: > Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net > CSeq: 102 INVITE > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Content-Length: 0 > > ------------------------------------------------------------------------ > 2012-05-24 23:08:45.840749 [NOTICE] switch_channel.c:926 New Channel > sofia/internal/+41793940965 at bluesip.net[69cd22e6-a5f5-11e1-a833-7b1f73a7ffcf] > 2012-05-24 23:08:45.840749 [DEBUG] sofia.c:5532 Channel sofia/internal/ > +41793940965 at bluesip.net entering state [received][100] > 2012-05-24 23:08:45.840749 [DEBUG] sofia.c:5543 Remote SDP: > v=0 > o=root 20076 20076 IN IP4 217.74.179.28 > s=session > c=IN IP4 217.74.179.28 > t=0 0 > m=audio 25626 RTP/AVP 8 0 18 111 3 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:111 G726-32/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [PCMA:8:8000:20:64000]/[iLBC:97:8000:30:13330] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [PCMU:0:8000:20:64000]/[iLBC:97:8000:30:13330] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [G729:18:8000:20:8000]/[iLBC:97:8000:30:13330] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [G726-32:111:8000:20:0]/[iLBC:97:8000:30:13330] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [GSM:3:8000:20:13200]/[iLBC:97:8000:30:13330] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [telephone-event:101:8000:20:0]/[iLBC:97:8000:30:13330] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf > send/recv payload to 101 > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [PCMA:8:8000:20:64000]/[G7221:115:32000:20:48000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [PCMU:0:8000:20:64000]/[G7221:115:32000:20:48000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [G729:18:8000:20:8000]/[G7221:115:32000:20:48000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [G726-32:111:8000:20:0]/[G7221:115:32000:20:48000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [GSM:3:8000:20:13200]/[G7221:115:32000:20:48000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [telephone-event:101:8000:20:0]/[G7221:115:32000:20:48000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf > send/recv payload to 101 > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [PCMA:8:8000:20:64000]/[G7221:107:16000:20:32000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [PCMU:0:8000:20:64000]/[G7221:107:16000:20:32000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [G729:18:8000:20:8000]/[G7221:107:16000:20:32000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [G726-32:111:8000:20:0]/[G7221:107:16000:20:32000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [GSM:3:8000:20:13200]/[G7221:107:16000:20:32000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [telephone-event:101:8000:20:0]/[G7221:107:16000:20:32000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf > send/recv payload to 101 > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [PCMA:8:8000:20:64000]/[G722:9:8000:20:64000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [G729:18:8000:20:8000]/[G722:9:8000:20:64000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [G726-32:111:8000:20:0]/[G722:9:8000:20:64000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [GSM:3:8000:20:13200]/[G722:9:8000:20:64000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [telephone-event:101:8000:20:0]/[G722:9:8000:20:64000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf > send/recv payload to 101 > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [PCMA:8:8000:20:64000]/[PCMU:0:8000:20:64000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:2996 Set Codec > sofia/internal/+41793940965 at bluesip.net PCMU/8000 20 ms 160 samples 64000 > bits > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_codec.c:111 sofia/internal/ > +41793940965 at bluesip.net Original read codec set to PCMU:0 > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf > send/recv payload to 101 > 2012-05-24 23:08:45.840749 [DEBUG] sofia.c:5757 (sofia/internal/ > +41793940965 at bluesip.net) State Change CS_NEW -> CS_INIT > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_INIT > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:401 > (sofia/internal/+41793940965 at bluesip.net) State INIT > 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:85 sofia/internal/ > +41793940965 at bluesip.net SOFIA INIT > 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:125 (sofia/internal/ > +41793940965 at bluesip.net) State Change CS_INIT -> CS_ROUTING > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:401 > (sofia/internal/+41793940965 at bluesip.net) State INIT going to sleep > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_ROUTING > 2012-05-24 23:08:45.840749 [DEBUG] switch_channel.c:1886 (sofia/internal/ > +41793940965 at bluesip.net) Callstate Change DOWN -> RINGING > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 > (sofia/internal/+41793940965 at bluesip.net) State ROUTING > 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:148 sofia/internal/ > +41793940965 at bluesip.net SOFIA ROUTING > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:104 > sofia/internal/+41793940965 at bluesip.net Standard ROUTING > 2012-05-24 23:08:45.840749 [INFO] mod_dialplan_xml.c:485 Processing > +41793940965 <+41793940965>->kb-1002 in context public > Dialplan: sofia/internal/+41793940965 at bluesip.net parsing > [public->from_kamailio] continue=false > Dialplan: sofia/internal/+41793940965 at bluesip.net Regex (PASS) > [from_kamailio] destination_number(kb-1002) =~ /^(.+)$/ break=on-false > Dialplan: sofia/internal/+41793940965 at bluesip.net Action transfer(kb-1002 > XML default) > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:154 > (sofia/internal/+41793940965 at bluesip.net) State Change CS_ROUTING -> > CS_EXECUTE > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 > (sofia/internal/+41793940965 at bluesip.net) State ROUTING going to sleep > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_EXECUTE > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:417 > (sofia/internal/+41793940965 at bluesip.net) State EXECUTE > 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:241 sofia/internal/ > +41793940965 at bluesip.net SOFIA EXECUTE > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:192 > sofia/internal/+41793940965 at bluesip.net Standard EXECUTE > EXECUTE sofia/internal/+41793940965 at bluesip.net transfer(kb-1002 XML > default) > 2012-05-24 23:08:45.840749 [DEBUG] switch_ivr.c:1711 (sofia/internal/ > +41793940965 at bluesip.net) State Change CS_EXECUTE -> CS_ROUTING > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:731 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:08:45.840749 [NOTICE] switch_ivr.c:1717 Transfer > sofia/internal/+41793940965 at bluesip.net to XML[kb-1002 at default] > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:417 > (sofia/internal/+41793940965 at bluesip.net) State EXECUTE going to sleep > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_ROUTING > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 > (sofia/internal/+41793940965 at bluesip.net) State ROUTING > 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:148 sofia/internal/ > +41793940965 at bluesip.net SOFIA ROUTING > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:104 > sofia/internal/+41793940965 at bluesip.net Standard ROUTING > 2012-05-24 23:08:45.840749 [INFO] mod_dialplan_xml.c:485 Processing > +41793940965 <+41793940965>->kb-1002 in context default > Dialplan: sofia/internal/+41793940965 at bluesip.net parsing [default->vbox] > continue=false > Dialplan: sofia/internal/+41793940965 at bluesip.net Regex (FAIL) [vbox] > destination_number(kb-1002) =~ /^vb-([0-9][0-9][0-9])$/ break=on-false > Dialplan: sofia/internal/+41793940965 at bluesip.net parsing > [default->vmenu] continue=false > Dialplan: sofia/internal/+41793940965 at bluesip.net Regex (FAIL) [vmenu] > destination_number(kb-1002) =~ /^vm-([0-9][0-9][0-9][0-9])$/ break=on-false > Dialplan: sofia/internal/+41793940965 at bluesip.net parsing > [default->kbridge] continue=false > Dialplan: sofia/internal/+41793940965 at bluesip.net Regex (PASS) [kbridge] > destination_number(kb-1002) =~ /^kb-(.+)$/ break=on-false > Dialplan: sofia/internal/+41793940965 at bluesip.net Action > set(proxy_media=true) > Dialplan: sofia/internal/+41793940965 at bluesip.net Action > set(call_timeout=50) > Dialplan: sofia/internal/+41793940965 at bluesip.net Action > set(continue_on_fail=true) > Dialplan: sofia/internal/+41793940965 at bluesip.net Action > set(hangup_after_bridge=true) > Dialplan: sofia/internal/+41793940965 at bluesip.net Action > set(sip_invite_domain=78.138.90.58) > Dialplan: sofia/internal/+41793940965 at bluesip.net Action > export(sip_contact_user=ufs) > Dialplan: sofia/internal/+41793940965 at bluesip.net Action bridge(sofia/ > 78.138.90.58/1002 at 78.138.90.58:5060) > Dialplan: sofia/internal/+41793940965 at bluesip.net Action answer() > Dialplan: sofia/internal/+41793940965 at bluesip.net Action > voicemail(default ${domain_name} 1002) > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:154 > (sofia/internal/+41793940965 at bluesip.net) State Change CS_ROUTING -> > CS_EXECUTE > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 > (sofia/internal/+41793940965 at bluesip.net) State ROUTING going to sleep > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_EXECUTE > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:417 > (sofia/internal/+41793940965 at bluesip.net) State EXECUTE > 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:241 sofia/internal/ > +41793940965 at bluesip.net SOFIA EXECUTE > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:192 > sofia/internal/+41793940965 at bluesip.net Standard EXECUTE > EXECUTE sofia/internal/+41793940965 at bluesip.net set(proxy_media=true) > 2012-05-24 23:08:45.840749 [DEBUG] mod_dptools.c:1281 sofia/internal/ > +41793940965 at bluesip.net SET [proxy_media]=[true] > EXECUTE sofia/internal/+41793940965 at bluesip.net set(call_timeout=50) > 2012-05-24 23:08:45.840749 [DEBUG] mod_dptools.c:1281 sofia/internal/ > +41793940965 at bluesip.net SET [call_timeout]=[50] > EXECUTE sofia/internal/+41793940965 at bluesip.net set(continue_on_fail=true) > 2012-05-24 23:08:45.840749 [DEBUG] mod_dptools.c:1281 sofia/internal/ > +41793940965 at bluesip.net SET [continue_on_fail]=[true] > EXECUTE sofia/internal/+41793940965 at bluesip.netset(hangup_after_bridge=true) > 2012-05-24 23:08:45.840749 [DEBUG] mod_dptools.c:1281 sofia/internal/ > +41793940965 at bluesip.net SET [hangup_after_bridge]=[true] > EXECUTE sofia/internal/+41793940965 at bluesip.netset(sip_invite_domain=78.138.90.58) > 2012-05-24 23:08:45.840749 [DEBUG] mod_dptools.c:1281 sofia/internal/ > +41793940965 at bluesip.net SET [sip_invite_domain]=[78.138.90.58] > EXECUTE sofia/internal/+41793940965 at bluesip.netexport(sip_contact_user=ufs) > 2012-05-24 23:08:45.840749 [DEBUG] switch_channel.c:1093 EXPORT > (export_vars) [sip_contact_user]=[ufs] > EXECUTE sofia/internal/+41793940965 at bluesip.net bridge(sofia/ > 78.138.90.58/1002 at 78.138.90.58:5060) > 2012-05-24 23:08:45.840749 [DEBUG] switch_channel.c:1047 sofia/internal/ > +41793940965 at bluesip.net EXPORTING[export_vars] [sip_contact_user]=[ufs] > to event > 2012-05-24 23:08:45.840749 [DEBUG] switch_ivr_originate.c:1884 Parsing > global variables > 2012-05-24 23:08:45.840749 [NOTICE] switch_channel.c:926 New Channel > sofia/internal/1002 at 78.138.90.58:5060[69ce1f0c-a5f5-11e1-a838-7b1f73a7ffcf] > 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:4691 (sofia/internal/ > 1002 at 78.138.90.58:5060) State Change CS_NEW -> CS_INIT > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/1002 at 78.138.90.58:5060) Running State Change CS_INIT > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:401 > (sofia/internal/1002 at 78.138.90.58:5060) State INIT > 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:85 sofia/internal/ > 1002 at 78.138.90.58:5060 SOFIA INIT > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:1871 sofia/internal/ > 1002 at 78.138.90.58:5060 Patched SDP > --- > v=0 > o=root 20076 20076 IN IP4 217.74.179.28 > s=session > c=IN IP4 217.74.179.28 > t=0 0 > m=audio 25626 RTP/AVP 8 0 18 111 3 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:111 G726-32/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > > +++ > v=0 > o=FreeSWITCH 1629510322 1629510323 IN IP4 78.138.90.58 > s=FreeSWITCH > c=IN IP4 78.138.90.58 > t=0 0 > m=audio 31178 RTP/AVP 8 0 18 111 3 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:111 G726-32/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > > send 1340 bytes to udp/[78.138.90.58]:5060 at 23:08:45.852280: > ------------------------------------------------------------------------ > INVITE sip:1002 at 78.138.90.58:5060 SIP/2.0 > Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKapDgFeegKyHXK > Max-Forwards: 67 > From: "+41793940965" ;tag=tgNrSS1jQFS8F > To: > Call-ID: 412c8312-2098-1230-8985-00163e6bb553 > CSeq: 28614398 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > 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: 372 > P-hint: USRLOC > X-FS-Support: update_display,send_info > Remote-Party-ID: "+41793940965" >;party=calling;screen=yes;privacy=off > > v=0 > o=FreeSWITCH 1629510322 1629510323 IN IP4 78.138.90.58 > s=FreeSWITCH > c=IN IP4 78.138.90.58 > t=0 0 > m=audio 31178 RTP/AVP 8 0 18 111 3 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:111 G726-32/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > ------------------------------------------------------------------------ > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:125 (sofia/internal/ > 1002 at 78.138.90.58:5060) State Change CS_INIT -> CS_ROUTING > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:401 > (sofia/internal/1002 at 78.138.90.58:5060) State INIT going to sleep > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/1002 at 78.138.90.58:5060) Running State Change CS_ROUTING > 2012-05-24 23:08:45.840749 [DEBUG] switch_channel.c:1886 (sofia/internal/ > 1002 at 78.138.90.58:5060) Callstate Change DOWN -> RINGING > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 > (sofia/internal/1002 at 78.138.90.58:5060) State ROUTING > 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:148 sofia/internal/ > 1002 at 78.138.90.58:5060 SOFIA ROUTING > 2012-05-24 23:08:45.840749 [DEBUG] switch_ivr_originate.c:66 > (sofia/internal/1002 at 78.138.90.58:5060) State Change CS_ROUTING -> > CS_CONSUME_MEDIA > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 > (sofia/internal/1002 at 78.138.90.58:5060) State ROUTING going to sleep > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/1002 at 78.138.90.58:5060) Running State Change > CS_CONSUME_MEDIA > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:429 > (sofia/internal/1002 at 78.138.90.58:5060) State CONSUME_MEDIA > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:429 > (sofia/internal/1002 at 78.138.90.58:5060) State CONSUME_MEDIA going to sleep > 2012-05-24 23:08:45.840749 [DEBUG] sofia.c:5532 Channel sofia/internal/ > 1002 at 78.138.90.58:5060 entering state [calling][0] > recv 365 bytes from udp/[78.138.90.58]:5060 at 23:08:45.964309: > ------------------------------------------------------------------------ > SIP/2.0 100 trying -- your call is important to us > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;branch=z9hG4bKapDgFeegKyHXK > From: "+41793940965" ;tag=tgNrSS1jQFS8F > To: > Call-ID: 412c8312-2098-1230-8985-00163e6bb553 > CSeq: 28614398 INVITE > Server: kamailio (3.1.5 (x86_64/linux)) > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 496 bytes from udp/[78.138.90.58]:5060 at 23:08:47.414766: > ------------------------------------------------------------------------ > SIP/2.0 180 Ringing > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;branch=z9hG4bKapDgFeegKyHXK > Record-Route: > Call-ID: 412c8312-2098-1230-8985-00163e6bb553 > From: "+41793940965" ;tag=tgNrSS1jQFS8F > To: > >;tag=qCSaB2YJOAsAr2dJ-RWNMhPoDdn1RFvs > CSeq: 28614398 INVITE > Contact: "Mama" > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, REGISTER > Content-Length: 0 > > ------------------------------------------------------------------------ > 2012-05-24 23:08:47.400741 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:47.400741 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:47.400741 [DEBUG] sofia.c:5532 Channel sofia/internal/ > 1002 at 78.138.90.58:5060 entering state [proceeding][180] > 2012-05-24 23:08:47.400741 [NOTICE] sofia.c:5624 Ring-Ready sofia/internal/ > 1002 at 78.138.90.58:5060! > send 995 bytes to udp/[217.74.179.29]:5060 at 23:08:47.416538: > ------------------------------------------------------------------------ > SIP/2.0 180 Ringing > Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 > Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 > Record-Route: > From: "+41793940965" ;tag=as00589402 > To: ;tag=S7UZQygFt62Nm > Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net > CSeq: 102 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Accept: application/sdp > 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-Length: 0 > Remote-Party-ID: "Outbound Call" > >;party=calling;privacy=off;screen=no > > ------------------------------------------------------------------------ > 2012-05-24 23:08:47.400741 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:08:47.400741 [NOTICE] mod_sofia.c:2514 Ring-Ready > sofia/internal/+41793940965 at bluesip.net! > 2012-05-24 23:08:47.400741 [DEBUG] sofia.c:5532 Channel sofia/internal/ > +41793940965 at bluesip.net entering state [early][180] > 2012-05-24 23:08:47.400741 [DEBUG] switch_core_session.c:731 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:08:47.400741 [NOTICE] switch_ivr_originate.c:483 Ring Ready > sofia/internal/+41793940965 at bluesip.net! > recv 496 bytes from udp/[78.138.90.58]:5060 at 23:08:47.804967: > ------------------------------------------------------------------------ > SIP/2.0 180 Ringing > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;branch=z9hG4bKapDgFeegKyHXK > Record-Route: > From: "+41793940965" ;tag=tgNrSS1jQFS8F > To: ;tag=549D2DD03BBA7C67 > Call-ID: 412c8312-2098-1230-8985-00163e6bb553 > CSeq: 28614398 INVITE > Contact: > ;uniq=6AC0DF4D2E498C8ACA82CB22226B9> > User-Agent: AVM FRITZ!Box Fon WLAN 7170 29.04.80 (Jan 27 2010) > Content-Length: 0 > > ------------------------------------------------------------------------ > 2012-05-24 23:08:47.800746 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:47.800746 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:47.800746 [DEBUG] sofia.c:5532 Channel sofia/internal/ > 1002 at 78.138.90.58:5060 entering state [proceeding][180] > recv 496 bytes from udp/[78.138.90.58]:5060 at 23:08:48.182098: > ------------------------------------------------------------------------ > SIP/2.0 180 Ringing > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;branch=z9hG4bKapDgFeegKyHXK > Record-Route: > Call-ID: 412c8312-2098-1230-8985-00163e6bb553 > From: "+41793940965" ;tag=tgNrSS1jQFS8F > To: > >;tag=qCSaB2YJOAsAr2dJ-RWNMhPoDdn1RFvs > CSeq: 28614398 INVITE > Contact: "Mama" > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, REGISTER > Content-Length: 0 > > ------------------------------------------------------------------------ > 2012-05-24 23:08:48.180741 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:48.180741 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:48.180741 [DEBUG] sofia.c:5532 Channel sofia/internal/ > 1002 at 78.138.90.58:5060 entering state [proceeding][180] > recv 496 bytes from udp/[78.138.90.58]:5060 at 23:08:48.417403: > ------------------------------------------------------------------------ > SIP/2.0 180 Ringing > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;branch=z9hG4bKapDgFeegKyHXK > Record-Route: > From: "+41793940965" ;tag=tgNrSS1jQFS8F > To: ;tag=549D2DD03BBA7C67 > Call-ID: 412c8312-2098-1230-8985-00163e6bb553 > CSeq: 28614398 INVITE > Contact: > ;uniq=6AC0DF4D2E498C8ACA82CB22226B9> > User-Agent: AVM FRITZ!Box Fon WLAN 7170 29.04.80 (Jan 27 2010) > Content-Length: 0 > > ------------------------------------------------------------------------ > 2012-05-24 23:08:48.400744 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:48.400744 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:48.400744 [DEBUG] sofia.c:5532 Channel sofia/internal/ > 1002 at 78.138.90.58:5060 entering state [proceeding][180] > recv 496 bytes from udp/[78.138.90.58]:5060 at 23:08:49.292599: > ------------------------------------------------------------------------ > SIP/2.0 180 Ringing > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;branch=z9hG4bKapDgFeegKyHXK > Record-Route: > From: "+41793940965" ;tag=tgNrSS1jQFS8F > To: ;tag=549D2DD03BBA7C67 > Call-ID: 412c8312-2098-1230-8985-00163e6bb553 > CSeq: 28614398 INVITE > Contact: > ;uniq=6AC0DF4D2E498C8ACA82CB22226B9> > User-Agent: AVM FRITZ!Box Fon WLAN 7170 29.04.80 (Jan 27 2010) > Content-Length: 0 > > ------------------------------------------------------------------------ > 2012-05-24 23:08:49.280742 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:49.280742 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:49.280742 [DEBUG] sofia.c:5532 Channel sofia/internal/ > 1002 at 78.138.90.58:5060 entering state [proceeding][180] > recv 478 bytes from udp/[217.74.179.29]:5060 at 23:09:15.066012: > ------------------------------------------------------------------------ > CANCEL sip:gw+trunk_1000 at 78.138.90.58:6090;transport=udp;gw=trunk_1000 > SIP/2.0 > Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 > Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 > From: "+41793940965" ;tag=as00589402 > To: > Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net > CSeq: 102 CANCEL > User-Agent: blueSIP PSTN GW > Max-Forwards: 69 > Content-Length: 0 > P-hint: USRLOC > > ------------------------------------------------------------------------ > send 369 bytes to udp/[217.74.179.29]:5060 at 23:09:15.066176: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 > Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 > From: "+41793940965" ;tag=as00589402 > To: ;tag=S7UZQygFt62Nm > Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net > CSeq: 102 CANCEL > Content-Length: 0 > > ------------------------------------------------------------------------ > send 766 bytes to udp/[217.74.179.29]:5060 at 23:09:15.066289: > ------------------------------------------------------------------------ > SIP/2.0 487 Request Terminated > Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 > Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 > From: "+41793940965" ;tag=as00589402 > To: ;tag=S7UZQygFt62Nm > Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net > CSeq: 102 INVITE > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > 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-Length: 0 > > ------------------------------------------------------------------------ > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:09:15.060742 [DEBUG] sofia.c:5532 Channel sofia/internal/ > +41793940965 at bluesip.net entering state [terminated][487] > 2012-05-24 23:09:15.060742 [DEBUG] switch_channel.c:2848 (sofia/internal/ > +41793940965 at bluesip.net) Callstate Change RINGING -> HANGUP > 2012-05-24 23:09:15.060742 [NOTICE] sofia.c:6301 Hangup sofia/internal/ > +41793940965 at bluesip.net [CS_EXECUTE] [ORIGINATOR_CANCEL] > 2012-05-24 23:09:15.060742 [DEBUG] switch_channel.c:2871 Send signal > sofia/internal/+41793940965 at bluesip.net [KILL] > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:09:15.060742 [DEBUG] switch_channel.c:2848 (sofia/internal/ > 1002 at 78.138.90.58:5060) Callstate Change RINGING -> HANGUP > 2012-05-24 23:09:15.060742 [NOTICE] switch_ivr_originate.c:3182 Hangup > sofia/internal/1002 at 78.138.90.58:5060 [CS_CONSUME_MEDIA] > [ORIGINATOR_CANCEL] > 2012-05-24 23:09:15.060742 [DEBUG] switch_channel.c:2871 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [KILL] > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/1002 at 78.138.90.58:5060) Running State Change CS_HANGUP > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:602 > (sofia/internal/1002 at 78.138.90.58:5060) State HANGUP > 2012-05-24 23:09:15.060742 [DEBUG] mod_sofia.c:469 Channel sofia/internal/ > 1002 at 78.138.90.58:5060 hanging up, cause: ORIGINATOR_CANCEL > 2012-05-24 23:09:15.060742 [DEBUG] mod_sofia.c:523 Sending CANCEL to > sofia/internal/1002 at 78.138.90.58:5060 > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:47 > sofia/internal/1002 at 78.138.90.58:5060 Standard HANGUP, cause: > ORIGINATOR_CANCEL > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:602 > (sofia/internal/1002 at 78.138.90.58:5060) State HANGUP going to sleep > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:393 > (sofia/internal/1002 at 78.138.90.58:5060) State Change CS_HANGUP -> > CS_REPORTING > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/1002 at 78.138.90.58:5060) Running State Change CS_REPORTING > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:662 > (sofia/internal/1002 at 78.138.90.58:5060) State REPORTING > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:79 > sofia/internal/1002 at 78.138.90.58:5060 Standard REPORTING, cause: > ORIGINATOR_CANCEL > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:662 > (sofia/internal/1002 at 78.138.90.58:5060) State REPORTING going to sleep > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:387 > (sofia/internal/1002 at 78.138.90.58:5060) State Change CS_REPORTING -> > CS_DESTROY > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1382 Session 9 > (sofia/internal/1002 at 78.138.90.58:5060) Locked, Waiting on external > entities > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:09:15.060742 [DEBUG] switch_ivr_originate.c:3358 Originate > Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL] > 2012-05-24 23:09:15.060742 [NOTICE] switch_core_session.c:1400 Session 9 > (sofia/internal/1002 at 78.138.90.58:5060) Ended > 2012-05-24 23:09:15.060742 [NOTICE] switch_core_session.c:1402 Close > Channel sofia/internal/1002 at 78.138.90.58:5060 [CS_DESTROY] > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:491 > (sofia/internal/1002 at 78.138.90.58:5060) Callstate Change HANGUP -> DOWN > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:494 > (sofia/internal/1002 at 78.138.90.58:5060) Running State Change CS_DESTROY > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:504 > (sofia/internal/1002 at 78.138.90.58:5060) State DESTROY > 2012-05-24 23:09:15.060742 [DEBUG] mod_sofia.c:374 sofia/internal/ > 1002 at 78.138.90.58:5060 SOFIA DESTROY > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:86 > sofia/internal/1002 at 78.138.90.58:5060 Standard DESTROY > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:504 > (sofia/internal/1002 at 78.138.90.58:5060) State DESTROY going to sleep > 2012-05-24 23:09:15.060742 [INFO] mod_dptools.c:2922 Originate Failed. > Cause: ORIGINATOR_CANCEL > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:2287 > sofia/internal/+41793940965 at bluesip.net skip receive message > [APPLICATION_EXEC_COMPLETE] (channel is hungup already) > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:417 > (sofia/internal/+41793940965 at bluesip.net) State EXECUTE going to sleep > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_HANGUP > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:602 > (sofia/internal/+41793940965 at bluesip.net) State HANGUP > 2012-05-24 23:09:15.060742 [DEBUG] mod_sofia.c:469 Channel sofia/internal/ > +41793940965 at bluesip.net hanging up, cause: ORIGINATOR_CANCEL > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:47 > sofia/internal/+41793940965 at bluesip.net Standard HANGUP, cause: > ORIGINATOR_CANCEL > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:602 > (sofia/internal/+41793940965 at bluesip.net) State HANGUP going to sleep > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:393 > (sofia/internal/+41793940965 at bluesip.net) State Change CS_HANGUP -> > CS_REPORTING > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/+41793940965 at bluesip.net) Running State Change > CS_REPORTING > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:662 > (sofia/internal/+41793940965 at bluesip.net) State REPORTING > send 383 bytes to udp/[78.138.90.58]:5060 at 23:09:15.087695: > ------------------------------------------------------------------------ > CANCEL sip:1002 at 78.138.90.58:5060 SIP/2.0 > Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKapDgFeegKyHXK > Max-Forwards: 67 > From: "+41793940965" ;tag=tgNrSS1jQFS8F > To: > Call-ID: 412c8312-2098-1230-8985-00163e6bb553 > CSeq: 28614398 CANCEL > Reason: FreeSWITCH;cause=487;text="ORIGINATOR_CANCEL" > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 408 bytes from udp/[217.74.179.29]:5060 at 23:09:15.094670: > ------------------------------------------------------------------------ > ACK sip:gw+trunk_1000 at 78.138.90.58:6090;transport=udp;gw=trunk_1000 > SIP/2.0 > Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 > From: "+41793940965" ;tag=as00589402 > Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net > To: ;tag=S7UZQygFt62Nm > CSeq: 102 ACK > User-Agent: Sip EXpress router(0.9.7 (i386/linux)) > Content-Length: 0 > > ------------------------------------------------------------------------ > 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:79 > sofia/internal/+41793940965 at bluesip.net Standard REPORTING, cause: > ORIGINATOR_CANCEL > 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:662 > (sofia/internal/+41793940965 at bluesip.net) State REPORTING going to sleep > 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:387 > (sofia/internal/+41793940965 at bluesip.net) State Change CS_REPORTING -> > CS_DESTROY > 2012-05-24 23:09:15.120807 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:09:15.120807 [DEBUG] switch_core_session.c:1382 Session 8 > (sofia/internal/+41793940965 at bluesip.net) Locked, Waiting on external > entities > 2012-05-24 23:09:15.120807 [NOTICE] switch_core_session.c:1400 Session 8 > (sofia/internal/+41793940965 at bluesip.net) Ended > 2012-05-24 23:09:15.120807 [NOTICE] switch_core_session.c:1402 Close > Channel sofia/internal/+41793940965 at bluesip.net [CS_DESTROY] > 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:491 > (sofia/internal/+41793940965 at bluesip.net) Callstate Change HANGUP -> DOWN > 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:494 > (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_DESTROY > 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:504 > (sofia/internal/+41793940965 at bluesip.net) State DESTROY > 2012-05-24 23:09:15.120807 [DEBUG] mod_sofia.c:374 sofia/internal/ > +41793940965 at bluesip.net SOFIA DESTROY > 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:86 > sofia/internal/+41793940965 at bluesip.net Standard DESTROY > 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:504 > (sofia/internal/+41793940965 at bluesip.net) State DESTROY going to sleep > recv 378 bytes from udp/[78.138.90.58]:5060 at 23:09:15.135235: > ------------------------------------------------------------------------ > SIP/2.0 200 canceling > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;branch=z9hG4bKapDgFeegKyHXK > From: "+41793940965" ;tag=tgNrSS1jQFS8F > To: >;tag=a6a1c5f60faecf035a1ae5b6e96e979a-8c5b > Call-ID: 412c8312-2098-1230-8985-00163e6bb553 > CSeq: 28614398 CANCEL > Server: kamailio (3.1.5 (x86_64/linux)) > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 446 bytes from udp/[78.138.90.58]:5060 at 23:09:16.419308: > ------------------------------------------------------------------------ > SIP/2.0 487 Request Terminated > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;branch=z9hG4bKapDgFeegKyHXK > Record-Route: > Call-ID: 412c8312-2098-1230-8985-00163e6bb553 > From: "+41793940965" ;tag=tgNrSS1jQFS8F > To: > >;tag=qCSaB2YJOAsAr2dJ-RWNMhPoDdn1RFvs > CSeq: 28614398 INVITE > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, REGISTER > Content-Length: 0 > > ------------------------------------------------------------------------ > send 354 bytes to udp/[78.138.90.58]:5060 at 23:09:16.419543: > ------------------------------------------------------------------------ > ACK sip:1002 at 78.138.90.58:5060 SIP/2.0 > Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKapDgFeegKyHXK > Max-Forwards: 67 > From: "+41793940965" ;tag=tgNrSS1jQFS8F > To: > >;tag=qCSaB2YJOAsAr2dJ-RWNMhPoDdn1RFvs > Call-ID: 412c8312-2098-1230-8985-00163e6bb553 > CSeq: 28614398 ACK > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 564 bytes from udp/[41.34.123.243]:58881 at 23:13:13.612418: > ------------------------------------------------------------------------ > REGISTER sip:sip.pcfone.com:6090 SIP/2.0 > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport;branch=z9hG4bKPjHQIpv.ACHGbXD4IHrMr72LwZ8kYdlu-J > Max-Forwards: 70 > From: "Sherif 1003" > >;tag=wOP.FVT4Crk8O5E.bB7BWUhNO6QG91IG > To: "Sherif 1003" > > > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60926 REGISTER > User-Agent: Telephone 1.0.2 > Contact: "Sherif 1003" > Expires: 300 > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, > REFER, MESSAGE, OPTIONS > Content-Length: 0 > > ------------------------------------------------------------------------ > send 744 bytes to udp/[41.34.123.243]:58881 at 23:13:13.613152: > ------------------------------------------------------------------------ > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport=58881;branch=z9hG4bKPjHQIpv.ACHGbXD4IHrMr72LwZ8kYdlu-J;received=41.34.123.243 > From: "Sherif 1003" > >;tag=wOP.FVT4Crk8O5E.bB7BWUhNO6QG91IG > To: "Sherif 1003" > >;tag=USeHUmjpmrFUB > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60926 REGISTER > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > WWW-Authenticate: Digest realm="sip.pcfone.com", > nonce="09674764-a5f6-11e1-a83c-7b1f73a7ffcf", algorithm=MD5, qop="auth" > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 830 bytes from udp/[41.34.123.243]:58881 at 23:13:13.847746: > ------------------------------------------------------------------------ > REGISTER sip:sip.pcfone.com:6090 SIP/2.0 > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport;branch=z9hG4bKPjFr7rIAdAJPRQd2GmrMrVib3KT3Rup0q4 > Max-Forwards: 70 > From: "Sherif 1003" > >;tag=wOP.FVT4Crk8O5E.bB7BWUhNO6QG91IG > To: "Sherif 1003" > > > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60927 REGISTER > User-Agent: Telephone 1.0.2 > Contact: "Sherif 1003" > Expires: 300 > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, > REFER, MESSAGE, OPTIONS > Authorization: Digest username="1002", realm="sip.pcfone.com", > nonce="09674764-a5f6-11e1-a83c-7b1f73a7ffcf", uri="sip:sip.pcfone.com:6090", > response="3bc0bb82a803ded3641704c4eb78d58f", algorithm=MD5, > cnonce="sc3AJfhuev18Bs-L-6tORmI.RXlxnPOz", qop=auth, nc=00000001 > Content-Length: 0 > > ------------------------------------------------------------------------ > send 706 bytes to udp/[41.34.123.243]:58881 at 23:13:13.850197: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport=58881;branch=z9hG4bKPjFr7rIAdAJPRQd2GmrMrVib3KT3Rup0q4;received=41.34.123.243 > From: "Sherif 1003" > >;tag=wOP.FVT4Crk8O5E.bB7BWUhNO6QG91IG > To: "Sherif 1003" > >;tag=v279vF3SH15DQ > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60927 REGISTER > Contact: ;expires=300 > Date: Thu, 24 May 2012 23:13:13 GMT > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Content-Length: 0 > > ------------------------------------------------------------------------ > send 989 bytes to udp/[41.34.123.243]:58881 at 23:13:13.888717: > ------------------------------------------------------------------------ > NOTIFY sip:1002 at 192.168.178.26:58881;ob SIP/2.0 > Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKBZ68g9yKg77FF > Route: ;ob > Max-Forwards: 70 > From: > >;tag=XB12yamXeav0j > To: > > > Call-ID: e0efa252-2098-1230-8985-00163e6bb553 > CSeq: 28614532 NOTIFY > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > 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=noresource > Content-Type: application/simple-message-summary > Content-Length: 91 > > Messages-Waiting: yes > Message-Account: sip:1002 at 78.138.90.58 > Voice-Message: 1/0 (0/0) > > ------------------------------------------------------------------------ > recv 308 bytes from udp/[41.34.123.243]:58881 at 23:13:14.298850: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;received=78.138.90.58;branch=z9hG4bKBZ68g9yKg77FF > Call-ID: e0efa252-2098-1230-8985-00163e6bb553 > From: > >;tag=XB12yamXeav0j > To: > >;tag=z9hG4bKBZ68g9yKg77FF > CSeq: 28614532 NOTIFY > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 564 bytes from udp/[41.34.123.243]:58881 at 23:18:09.067232: > ------------------------------------------------------------------------ > REGISTER sip:sip.pcfone.com:6090 SIP/2.0 > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport;branch=z9hG4bKPjABTslmZIhiHYRtLqQn2erwmX19MVYHFC > Max-Forwards: 70 > From: "Sherif 1003" > >;tag=JBSXL5WZLWbLzpvRFoTSP5x8qQslI1ST > To: "Sherif 1003" > > > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60928 REGISTER > User-Agent: Telephone 1.0.2 > Contact: "Sherif 1003" > Expires: 300 > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, > REFER, MESSAGE, OPTIONS > Content-Length: 0 > > ------------------------------------------------------------------------ > send 744 bytes to udp/[41.34.123.243]:58881 at 23:18:09.068121: > ------------------------------------------------------------------------ > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport=58881;branch=z9hG4bKPjABTslmZIhiHYRtLqQn2erwmX19MVYHFC;received=41.34.123.243 > From: "Sherif 1003" > >;tag=JBSXL5WZLWbLzpvRFoTSP5x8qQslI1ST > To: "Sherif 1003" > >;tag=ymtU0540BKjKe > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60928 REGISTER > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > WWW-Authenticate: Digest realm="sip.pcfone.com", > nonce="b9821df4-a5f6-11e1-a83d-7b1f73a7ffcf", algorithm=MD5, qop="auth" > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 830 bytes from udp/[41.34.123.243]:58881 at 23:18:09.302944: > ------------------------------------------------------------------------ > REGISTER sip:sip.pcfone.com:6090 SIP/2.0 > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport;branch=z9hG4bKPjRgtztcy63vMT97Jp5pH9n311li5EFeY- > Max-Forwards: 70 > From: "Sherif 1003" > >;tag=JBSXL5WZLWbLzpvRFoTSP5x8qQslI1ST > To: "Sherif 1003" > > > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60929 REGISTER > User-Agent: Telephone 1.0.2 > Contact: "Sherif 1003" > Expires: 300 > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, > REFER, MESSAGE, OPTIONS > Authorization: Digest username="1002", realm="sip.pcfone.com", > nonce="b9821df4-a5f6-11e1-a83d-7b1f73a7ffcf", uri="sip:sip.pcfone.com:6090", > response="519c6301816b51f3bbb98d97a347e92b", algorithm=MD5, > cnonce="sc3AJfhuev18Bs-L-6tORmI.RXlxnPOz", qop=auth, nc=00000001 > Content-Length: 0 > > ------------------------------------------------------------------------ > send 706 bytes to udp/[41.34.123.243]:58881 at 23:18:09.305217: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport=58881;branch=z9hG4bKPjRgtztcy63vMT97Jp5pH9n311li5EFeY-;received=41.34.123.243 > From: "Sherif 1003" > >;tag=JBSXL5WZLWbLzpvRFoTSP5x8qQslI1ST > To: "Sherif 1003" > >;tag=ZXKm20N48U85S > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60929 REGISTER > Contact: ;expires=300 > Date: Thu, 24 May 2012 23:18:09 GMT > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Content-Length: 0 > > ------------------------------------------------------------------------ > send 989 bytes to udp/[41.34.123.243]:58881 at 23:18:09.379987: > ------------------------------------------------------------------------ > NOTIFY sip:1002 at 192.168.178.26:58881;ob SIP/2.0 > Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKc8Z1j4FQDgy2a > Route: ;ob > Max-Forwards: 70 > From: > >;tag=06cD4U6754yrN > To: > > > Call-ID: 91100602-2099-1230-8985-00163e6bb553 > CSeq: 28614680 NOTIFY > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > 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=noresource > Content-Type: application/simple-message-summary > Content-Length: 91 > > Messages-Waiting: yes > Message-Account: sip:1002 at 78.138.90.58 > Voice-Message: 1/0 (0/0) > > ------------------------------------------------------------------------ > recv 308 bytes from udp/[41.34.123.243]:58881 at 23:18:09.773629: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;received=78.138.90.58;branch=z9hG4bKc8Z1j4FQDgy2a > Call-ID: 91100602-2099-1230-8985-00163e6bb553 > From: > >;tag=06cD4U6754yrN > To: > >;tag=z9hG4bKc8Z1j4FQDgy2a > CSeq: 28614680 NOTIFY > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 564 bytes from udp/[41.34.123.243]:58881 at 23:23:04.522455: > ------------------------------------------------------------------------ > REGISTER sip:sip.pcfone.com:6090 SIP/2.0 > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport;branch=z9hG4bKPjaCuKyaBi6ZqcsHyp5HhH3WthJStom0SK > Max-Forwards: 70 > From: "Sherif 1003" > >;tag=qfUt1fQWuE-Ro3zWld6X0r6WidtD6B1q > To: "Sherif 1003" > > > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60930 REGISTER > User-Agent: Telephone 1.0.2 > Contact: "Sherif 1003" > Expires: 300 > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, > REFER, MESSAGE, OPTIONS > Content-Length: 0 > > ------------------------------------------------------------------------ > send 744 bytes to udp/[41.34.123.243]:58881 at 23:23:04.523158: > ------------------------------------------------------------------------ > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport=58881;branch=z9hG4bKPjaCuKyaBi6ZqcsHyp5HhH3WthJStom0SK;received=41.34.123.243 > From: "Sherif 1003" > >;tag=qfUt1fQWuE-Ro3zWld6X0r6WidtD6B1q > To: "Sherif 1003" > >;tag=1F655pQB3DNBH > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60930 REGISTER > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > WWW-Authenticate: Digest realm="sip.pcfone.com", > nonce="699cfe2a-a5f7-11e1-a83e-7b1f73a7ffcf", algorithm=MD5, qop="auth" > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 830 bytes from udp/[41.34.123.243]:58881 at 23:23:04.757798: > ------------------------------------------------------------------------ > REGISTER sip:sip.pcfone.com:6090 SIP/2.0 > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport;branch=z9hG4bKPjs8SAufeoyzCLrePaAbYhZuOqQF47XIc- > Max-Forwards: 70 > From: "Sherif 1003" > >;tag=qfUt1fQWuE-Ro3zWld6X0r6WidtD6B1q > To: "Sherif 1003" > > > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60931 REGISTER > User-Agent: Telephone 1.0.2 > Contact: "Sherif 1003" > Expires: 300 > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, > REFER, MESSAGE, OPTIONS > Authorization: Digest username="1002", realm="sip.pcfone.com", > nonce="699cfe2a-a5f7-11e1-a83e-7b1f73a7ffcf", uri="sip:sip.pcfone.com:6090", > response="23a886934aaf049f08432b6bb226bd37", algorithm=MD5, > cnonce="sc3AJfhuev18Bs-L-6tORmI.RXlxnPOz", qop=auth, nc=00000001 > Content-Length: 0 > > ------------------------------------------------------------------------ > send 706 bytes to udp/[41.34.123.243]:58881 at 23:23:04.759924: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport=58881;branch=z9hG4bKPjs8SAufeoyzCLrePaAbYhZuOqQF47XIc-;received=41.34.123.243 > From: "Sherif 1003" > >;tag=qfUt1fQWuE-Ro3zWld6X0r6WidtD6B1q > To: "Sherif 1003" > >;tag=2rZy7H8e0pByc > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60931 REGISTER > Contact: ;expires=300 > Date: Thu, 24 May 2012 23:23:04 GMT > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Content-Length: 0 > > ------------------------------------------------------------------------ > send 989 bytes to udp/[41.34.123.243]:58881 at 23:23:04.872728: > ------------------------------------------------------------------------ > NOTIFY sip:1002 at 192.168.178.26:58881;ob SIP/2.0 > Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKDHStmZ0taSmNp > Route: ;ob > Max-Forwards: 70 > From: > >;tag=31rQ9cSjXZ1gr > To: > > > Call-ID: 4130a1ce-209a-1230-8985-00163e6bb553 > CSeq: 28614828 NOTIFY > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > 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=noresource > Content-Type: application/simple-message-summary > Content-Length: 91 > > Messages-Waiting: yes > Message-Account: sip:1002 at 78.138.90.58 > Voice-Message: 1/0 (0/0) > > ------------------------------------------------------------------------ > recv 308 bytes from udp/[41.34.123.243]:58881 at 23:23:05.269063: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;received=78.138.90.58;branch=z9hG4bKDHStmZ0taSmNp > Call-ID: 4130a1ce-209a-1230-8985-00163e6bb553 > From: > >;tag=31rQ9cSjXZ1gr > To: > >;tag=z9hG4bKDHStmZ0taSmNp > CSeq: 28614828 NOTIFY > Content-Length: 0 > > ------------------------------------------------------------------------ > send 916 bytes to udp/[217.74.179.29]:5060 at 23:25:45.873423: > ------------------------------------------------------------------------ > REGISTER sip:bluesip.net;transport=udp SIP/2.0 > Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKetjKptHy71a8H > Max-Forwards: 70 > From: > ;transport=udp>;tag=6r0vBQZS650Fg > To: > ;transport=udp> > Call-ID: 2a3e4696-a5b9-11e1-a7ed-7b1f73a7ffcf > CSeq: 28601493 REGISTER > Contact: ;transport=udp;gw=trunk_1000> > Expires: 3600 > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Authorization: Digest username="bluesip/sherifomran", realm=" > bluesip.net", nonce="4fbeba4ebfb7414d5c701bd6f994402d551f3acf", > algorithm=MD5, uri="sip:bluesip.net;transport=udp", > response="c6cdcafe0418e519bc9ee0d8fa3d4d74" > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 722 bytes from udp/[217.74.179.29]:5060 at 23:25:45.904176: > ------------------------------------------------------------------------ > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;branch=z9hG4bKetjKptHy71a8H > From: > ;transport=udp>;tag=6r0vBQZS650Fg > To: > ;transport=udp>;tag=0354a2e1b960c9cc2279eca4e5f84e20.1b16 > Call-ID: 2a3e4696-a5b9-11e1-a7ed-7b1f73a7ffcf > CSeq: 28601493 REGISTER > WWW-Authenticate: Digest realm="bluesip.net", > nonce="4fbec4a4ba82d1f1feaa32ccec01b85be054fcf7", stale=true > Server: Sip EXpress router (0.9.7 (i386/linux)) > Content-Length: 0 > Warning: 392 217.74.179.29:5060 "Noisy feedback tells: pid=5455 > req_src_ip=78.138.90.58 req_src_port=6090 in_uri=sip:bluesip.net;transport=udp > out_uri=sip:bluesip.net;transport=udp via_cnt==1" > > ------------------------------------------------------------------------ > send 916 bytes to udp/[217.74.179.29]:5060 at 23:25:45.904416: > ------------------------------------------------------------------------ > REGISTER sip:bluesip.net;transport=udp SIP/2.0 > Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKF3BcrN214a1tD > Max-Forwards: 70 > From: > ;transport=udp>;tag=6r0vBQZS650Fg > To: > ;transport=udp> > Call-ID: 2a3e4696-a5b9-11e1-a7ed-7b1f73a7ffcf > CSeq: 28601494 REGISTER > Contact: ;transport=udp;gw=trunk_1000> > Expires: 3600 > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Authorization: Digest username="bluesip/sherifomran", realm=" > bluesip.net", nonce="4fbec4a4ba82d1f1feaa32ccec01b85be054fcf7", > algorithm=MD5, uri="sip:bluesip.net;transport=udp", > response="4c09dbe4b9accb52d4104b40dfe20040" > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 699 bytes from udp/[217.74.179.29]:5060 at 23:25:45.935046: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;branch=z9hG4bKF3BcrN214a1tD > From: > ;transport=udp>;tag=6r0vBQZS650Fg > To: > ;transport=udp>;tag=0354a2e1b960c9cc2279eca4e5f84e20.2017 > Call-ID: 2a3e4696-a5b9-11e1-a7ed-7b1f73a7ffcf > CSeq: 28601494 REGISTER > Contact: ;transport=udp;gw=trunk_1000>;q=0.5;expires=3600 > Server: Sip EXpress router (0.9.7 (i386/linux)) > Content-Length: 0 > Warning: 392 217.74.179.29:5060 "Noisy feedback tells: pid=5462 > req_src_ip=78.138.90.58 req_src_port=6090 in_uri=sip:bluesip.net;transport=udp > out_uri=sip:bluesip.net;transport=udp via_cnt==1" > > ------------------------------------------------------------------------ > recv 564 bytes from udp/[41.34.123.243]:58881 at 23:27:59.977057: > ------------------------------------------------------------------------ > REGISTER sip:sip.pcfone.com:6090 SIP/2.0 > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport;branch=z9hG4bKPjEOP.a4NoXqoFqC7a.k2tyeX-mYlUyH4H > Max-Forwards: 70 > From: "Sherif 1003" > >;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K > To: "Sherif 1003" > > > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60932 REGISTER > User-Agent: Telephone 1.0.2 > Contact: "Sherif 1003" > Expires: 300 > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, > REFER, MESSAGE, OPTIONS > Content-Length: 0 > > ------------------------------------------------------------------------ > send 744 bytes to udp/[41.34.123.243]:58881 at 23:27:59.977960: > ------------------------------------------------------------------------ > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport=58881;branch=z9hG4bKPjEOP.a4NoXqoFqC7a.k2tyeX-mYlUyH4H;received=41.34.123.243 > From: "Sherif 1003" > >;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K > To: "Sherif 1003" > >;tag=4ajgB89Nt8Q3K > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60932 REGISTER > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > WWW-Authenticate: Digest realm="sip.pcfone.com", > nonce="19b7cc40-a5f8-11e1-a83f-7b1f73a7ffcf", algorithm=MD5, qop="auth" > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 564 bytes from udp/[41.34.123.243]:58881 at 23:28:00.468487: > ------------------------------------------------------------------------ > REGISTER sip:sip.pcfone.com:6090 SIP/2.0 > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport;branch=z9hG4bKPjEOP.a4NoXqoFqC7a.k2tyeX-mYlUyH4H > Max-Forwards: 70 > From: "Sherif 1003" > >;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K > To: "Sherif 1003" > > > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60932 REGISTER > User-Agent: Telephone 1.0.2 > Contact: "Sherif 1003" > Expires: 300 > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, > REFER, MESSAGE, OPTIONS > Content-Length: 0 > > ------------------------------------------------------------------------ > send 744 bytes to udp/[41.34.123.243]:58881 at 23:28:00.468679: > ------------------------------------------------------------------------ > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport=58881;branch=z9hG4bKPjEOP.a4NoXqoFqC7a.k2tyeX-mYlUyH4H;received=41.34.123.243 > From: "Sherif 1003" > >;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K > To: "Sherif 1003" > >;tag=4ajgB89Nt8Q3K > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60932 REGISTER > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > WWW-Authenticate: Digest realm="sip.pcfone.com", > nonce="19b7cc40-a5f8-11e1-a83f-7b1f73a7ffcf", algorithm=MD5, qop="auth" > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 830 bytes from udp/[41.34.123.243]:58881 at 23:28:00.717446: > ------------------------------------------------------------------------ > REGISTER sip:sip.pcfone.com:6090 SIP/2.0 > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport;branch=z9hG4bKPj0iHd3-XUjnvcBs9DiJUCWzrmYTmWxFLt > Max-Forwards: 70 > From: "Sherif 1003" > >;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K > To: "Sherif 1003" > > > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60933 REGISTER > User-Agent: Telephone 1.0.2 > Contact: "Sherif 1003" > Expires: 300 > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, > REFER, MESSAGE, OPTIONS > Authorization: Digest username="1002", realm="sip.pcfone.com", > nonce="19b7cc40-a5f8-11e1-a83f-7b1f73a7ffcf", uri="sip:sip.pcfone.com:6090", > response="e21c903e6d98b5adcdd2eefa75217157", algorithm=MD5, > cnonce="sc3AJfhuev18Bs-L-6tORmI.RXlxnPOz", qop=auth, nc=00000001 > Content-Length: 0 > > ------------------------------------------------------------------------ > send 706 bytes to udp/[41.34.123.243]:58881 at 23:28:00.719947: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport=58881;branch=z9hG4bKPj0iHd3-XUjnvcBs9DiJUCWzrmYTmWxFLt;received=41.34.123.243 > From: "Sherif 1003" > >;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K > To: "Sherif 1003" > >;tag=5KB9c3tSQHepF > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60933 REGISTER > Contact: ;expires=300 > Date: Thu, 24 May 2012 23:28:00 GMT > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Content-Length: 0 > > ------------------------------------------------------------------------ > send 989 bytes to udp/[41.34.123.243]:58881 at 23:28:00.868368: > ------------------------------------------------------------------------ > NOTIFY sip:1002 at 192.168.178.26:58881;ob SIP/2.0 > Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKgc54SgK51KQDS > Route: ;ob > Max-Forwards: 70 > From: > >;tag=6v41eyBXmt48a > To: > > > Call-ID: f19dfbce-209a-1230-8985-00163e6bb553 > CSeq: 28614976 NOTIFY > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > 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=noresource > Content-Type: application/simple-message-summary > Content-Length: 91 > > Messages-Waiting: yes > Message-Account: sip:1002 at 78.138.90.58 > Voice-Message: 1/0 (0/0) > > ------------------------------------------------------------------------ > recv 308 bytes from udp/[41.34.123.243]:58881 at 23:28:01.128279: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;received=78.138.90.58;branch=z9hG4bKgc54SgK51KQDS > Call-ID: f19dfbce-209a-1230-8985-00163e6bb553 > From: > >;tag=6v41eyBXmt48a > To: > >;tag=z9hG4bKgc54SgK51KQDS > CSeq: 28614976 NOTIFY > Content-Length: 0 > > ------------------------------------------------------------------------ > send 913 bytes to udp/[217.74.179.29]:5060 at 23:28:11.885213: > ------------------------------------------------------------------------ > REGISTER sip:bluesip.net;transport=udp SIP/2.0 > Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKHNyXUB48yvD0m > Max-Forwards: 70 > From: > ;transport=udp>;tag=5F739Uep9vaXm > To: > ;transport=udp> > Call-ID: 2a3e49c0-a5b9-11e1-a7ee-7b1f73a7ffcf > CSeq: 28601493 REGISTER > Contact: ;transport=udp;gw=trunk_1002> > Expires: 3600 > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Authorization: Digest username="bluesip/salahomran", realm="bluesip.net", > nonce="4fbebeca79a5cff7c417463814cb9054d313bcac", algorithm=MD5, uri="sip: > bluesip.net;transport=udp", response="bb1babb90f4ea1dc8fbb9aa45d6038c7" > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 720 bytes from udp/[217.74.179.29]:5060 at 23:28:11.915764: > ------------------------------------------------------------------------ > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;branch=z9hG4bKHNyXUB48yvD0m > From: > ;transport=udp>;tag=5F739Uep9vaXm > To: > ;transport=udp>;tag=0354a2e1b960c9cc2279eca4e5f84e20.3454 > Call-ID: 2a3e49c0-a5b9-11e1-a7ee-7b1f73a7ffcf > CSeq: 28601493 REGISTER > WWW-Authenticate: Digest realm="bluesip.net", > nonce="4fbec5365af3a174f1cc0b8cda770c727d25bf71", stale=true > Server: Sip EXpress router (0.9.7 (i386/linux)) > Content-Length: 0 > Warning: 392 217.74.179.29:5060 "Noisy feedback tells: pid=5462 > req_src_ip=78.138.90.58 req_src_port=6090 in_uri=sip:bluesip.net;transport=udp > out_uri=sip:bluesip.net;transport=udp via_cnt==1" > > ------------------------------------------------------------------------ > send 913 bytes to udp/[217.74.179.29]:5060 at 23:28:11.915914: > ------------------------------------------------------------------------ > REGISTER sip:bluesip.net;transport=udp SIP/2.0 > Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKjyQpX6mcv53jg > Max-Forwards: 70 > From: > ;transport=udp>;tag=5F739Uep9vaXm > To: > ;transport=udp> > Call-ID: 2a3e49c0-a5b9-11e1-a7ee-7b1f73a7ffcf > CSeq: 28601494 REGISTER > Contact: ;transport=udp;gw=trunk_1002> > Expires: 3600 > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Authorization: Digest username="bluesip/salahomran", realm="bluesip.net", > nonce="4fbec5365af3a174f1cc0b8cda770c727d25bf71", algorithm=MD5, uri="sip: > bluesip.net;transport=udp", response="e260f6dc9b01ce8bea8c5ffb36bd4ddd" > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 697 bytes from udp/[217.74.179.29]:5060 at 23:28:11.946325: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;branch=z9hG4bKjyQpX6mcv53jg > From: > ;transport=udp>;tag=5F739Uep9vaXm > To: > ;transport=udp>;tag=0354a2e1b960c9cc2279eca4e5f84e20.4b5c > Call-ID: 2a3e49c0-a5b9-11e1-a7ee-7b1f73a7ffcf > CSeq: 28601494 REGISTER > Contact: ;transport=udp;gw=trunk_1002>;q=0.5;expires=3600 > Server: Sip EXpress router (0.9.7 (i386/linux)) > Content-Length: 0 > Warning: 392 217.74.179.29:5060 "Noisy feedback tells: pid=5455 > req_src_ip=78.138.90.58 req_src_port=6090 in_uri=sip:bluesip.net;transport=udp > out_uri=sip:bluesip.net;transport=udp via_cnt==1" > > ------------------------------------------------------------------------ > recv 564 bytes from udp/[41.34.123.243]:58881 at 23:32:55.930149: > ------------------------------------------------------------------------ > REGISTER sip:sip.pcfone.com:6090 SIP/2.0 > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport;branch=z9hG4bKPjuk8DZGiaXfN7BmtslNHVjtSg5UXsbHXc > Max-Forwards: 70 > From: "Sherif 1003" > >;tag=Pz.t-0tGXj3saM.ehBiB1EAMmmitze1D > To: "Sherif 1003" > > > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60934 REGISTER > User-Agent: Telephone 1.0.2 > Contact: "Sherif 1003" > Expires: 300 > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, > REFER, MESSAGE, OPTIONS > Content-Length: 0 > > ------------------------------------------------------------------------ > send 744 bytes to udp/[41.34.123.243]:58881 at 23:32:55.931324: > ------------------------------------------------------------------------ > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport=58881;branch=z9hG4bKPjuk8DZGiaXfN7BmtslNHVjtSg5UXsbHXc;received=41.34.123.243 > From: "Sherif 1003" > >;tag=Pz.t-0tGXj3saM.ehBiB1EAMmmitze1D > To: "Sherif 1003" > >;tag=75XtgSv0H3tUp > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60934 REGISTER > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > WWW-Authenticate: Digest realm="sip.pcfone.com", > nonce="ca1eabee-a5f8-11e1-a840-7b1f73a7ffcf", algorithm=MD5, qop="auth" > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 830 bytes from udp/[41.34.123.243]:58881 at 23:32:56.165624: > ------------------------------------------------------------------------ > REGISTER sip:sip.pcfone.com:6090 SIP/2.0 > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport;branch=z9hG4bKPjKEQ6xXmyzZxA6vCM88vHNYAOPAjUqFsD > Max-Forwards: 70 > From: "Sherif 1003" > >;tag=Pz.t-0tGXj3saM.ehBiB1EAMmmitze1D > To: "Sherif 1003" > > > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60935 REGISTER > User-Agent: Telephone 1.0.2 > Contact: "Sherif 1003" > Expires: 300 > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, > REFER, MESSAGE, OPTIONS > Authorization: Digest username="1002", realm="sip.pcfone.com", > nonce="ca1eabee-a5f8-11e1-a840-7b1f73a7ffcf", uri="sip:sip.pcfone.com:6090", > response="ff6d12f4ec7abd4b37187222b3af2358", algorithm=MD5, > cnonce="sc3AJfhuev18Bs-L-6tORmI.RXlxnPOz", qop=auth, nc=00000001 > Content-Length: 0 > > ------------------------------------------------------------------------ > send 706 bytes to udp/[41.34.123.243]:58881 at 23:32:56.168033: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport=58881;branch=z9hG4bKPjKEQ6xXmyzZxA6vCM88vHNYAOPAjUqFsD;received=41.34.123.243 > From: "Sherif 1003" > >;tag=Pz.t-0tGXj3saM.ehBiB1EAMmmitze1D > To: "Sherif 1003" > >;tag=8eQKjmD4ecHej > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60935 REGISTER > Contact: ;expires=300 > Date: Thu, 24 May 2012 23:32:56 GMT > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Content-Length: 0 > > ------------------------------------------------------------------------ > send 989 bytes to udp/[41.34.123.243]:58881 at 23:32:56.360238: > ------------------------------------------------------------------------ > NOTIFY sip:1002 at 192.168.178.26:58881;ob SIP/2.0 > Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKK7gFZ15FSet5B > Route: ;ob > Max-Forwards: 70 > From: > >;tag=9QgcmFy7BN70D > To: > > > Call-ID: a1be7708-209b-1230-8985-00163e6bb553 > CSeq: 28615124 NOTIFY > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > 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=noresource > Content-Type: application/simple-message-summary > Content-Length: 91 > > Messages-Waiting: yes > Message-Account: sip:1002 at 78.138.90.58 > Voice-Message: 1/0 (0/0) > > ------------------------------------------------------------------------ > recv 308 bytes from udp/[41.34.123.243]:58881 at 23:32:56.623554: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;received=78.138.90.58;branch=z9hG4bKK7gFZ15FSet5B > Call-ID: a1be7708-209b-1230-8985-00163e6bb553 > From: > >;tag=9QgcmFy7BN70D > To: > >;tag=z9hG4bKK7gFZ15FSet5B > CSeq: 28615124 NOTIFY > Content-Length: 0 > > ------------------------------------------------------------------------ > freeswitch at internal> > > > --- On *Thu, 5/24/12, Kristian Kielhofner > >* wrote: > > > From: Kristian Kielhofner > > > Subject: Re: [Freeswitch-users] Freeswitch (SBC + Media) does not know > that gateway phone is picked up > To: "FreeSWITCH Users Help" > > > Date: Thursday, May 24, 2012, 7:51 PM > > > Siptrace and logs please. > > On Thu, May 24, 2012 at 11:53 AM, Sherif Omran > > > wrote: > > > > Hi all, > > > > My topology is as follows: > > > > Kamailio -> FS (SBS+Media server) > > > > I came across an issue with my system as follows. I have a Hardphone > registered. When I do local call inside kamailio, it gets to FS and returns > back well and FS understands when I lift the handset. However, I added a > gateway (german landline server), when I call my self from another phone, > the call gets to FS and then transmits to Kamailio, it rings my extension > but when I lift the handset FS does not notice it and keeps ringing. > > > > Any body has an Idea? Here is my gateway trunk. > > > > > > > > > > > > > > > > > > > > > > thanks in advance > > Sherif Omran > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > Kristian Kielhofner > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -----Inline Attachment Follows----- > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -----Inline Attachment Follows----- > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -----Inline Attachment Follows----- > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120527/2b393e10/attachment-0001.html From saami_mh at ymail.com Sun May 27 13:18:02 2012 From: saami_mh at ymail.com (Samira Mh) Date: Sun, 27 May 2012 02:18:02 -0700 (PDT) Subject: [Freeswitch-users] How to connect dialplan on freeswitch to Mysql or ODBC? Message-ID: <1338110282.29296.YahooMailNeo@web120101.mail.ne1.yahoo.com> hi guys, i have created mysql database name "x" using ODBC in freeswitch, now i have the table named "y" in ?the "x" databases; now,how can i connect the sipuser in my direcroty to the "x" database so that use the result of quert for taht user? and how can i insert query into the dialplan to connect myuser to mysql table how can i do that ? thanks alot -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120527/f1d49ada/attachment.html From sherifomran2000 at yahoo.com Sun May 27 13:30:01 2012 From: sherifomran2000 at yahoo.com (Sherif Omran) Date: Sun, 27 May 2012 02:30:01 -0700 (PDT) Subject: [Freeswitch-users] Freeswitch (SBC + Media) does not know that gateway phone is picked up In-Reply-To: Message-ID: <1338111001.12711.YahooMailClassic@web110816.mail.gq1.yahoo.com> Hi Sammy I found the solution. Made a new extension and set proxy media to false. Thats why the call was looping between Kamailio and FS. It now works fine ? regards, Sherif --- On Sun, 5/27/12, SamyGo wrote: From: SamyGo Subject: Re: [Freeswitch-users] Freeswitch (SBC + Media) does not know that gateway phone is picked up To: "FreeSWITCH Users Help" Date: Sunday, May 27, 2012, 8:36 AM OK Sheriff, I've gone through this.?This says that destination on kamailio end is ringing and keeps on ringing state. What Im more interested in are the sip traces on port 5060 the story thats happening in between the destination phone and the kamailo..if u know what I mean !! On Sat, May 26, 2012 at 8:06 PM, Sherif Omran wrote: Hi Sanny, Please see the following links http://pastebin.freeswitch.org/19175 http://pastebin.freeswitch.org/19174 regards, Sherif --- On Sat, 5/26/12, SamyGo wrote: From: SamyGo Subject: Re: [Freeswitch-users] Freeswitch (SBC + Media) does not know that gateway phone is picked up To: "FreeSWITCH Users Help" Date: Saturday, May 26, 2012, 5:52 PM Hey Sherrif, You might wanna resend the pcap file as I couldn;t find any single INVITE or any call in that capture. Please review. BRSaGo. On Sat, May 26, 2012 at 1:05 AM, Sherif Omran wrote: Hello Sammy FS uses port 6090 and registers the GW. When a call comes, it rings the extension but then gives a busy signal and FS keeps ringing till I cancel the call from the GW. recv 477 bytes from udp/[217.74.179.29]:5060 at 19:52:05.662344: ? ?------------------------------------------------------------------------? ?CANCEL sip:gw+trunk_1000 at 78.138.90.58:6090;transport=udp;gw=trunk_1000 SIP/2.0 ? ?Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bKe567.13b0b9e.0? ?Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK16d1a257;rport=5060 ? ?From: "+41793940965" ;tag=as1dccab06 ? ?To: ? ?Call-ID: 5cd37edb776a6b3a35e9713a453a3425 at bluesip.net? ?CSeq: 102 CANCEL ? ?User-Agent: blueSIP PSTN GW ? ?Max-Forwards: 69? ?Content-Length: 0? ?P-hint: USRLOC --- On Fri, 5/25/12, SamyGo wrote: From: SamyGo Subject: Re: [Freeswitch-users] Freeswitch (SBC + Media) does not know that gateway phone is picked up To: "FreeSWITCH Users Help" Date: Friday, May 25, 2012, 10:46 AM Hi again,If you want kamailio register to the provider then use UAC module. Kamailio will use the username/password and register with the provider. Regards,Sammy On Fri, May 25, 2012 at 12:20 PM, Sherif Omran wrote: Dear Sammy, Thank you for your question ... Yes, the GW is registered through FS because I did not know how to register it to kamailio. But it seems better to register it to kamailio. One more information, calls from 1001 to 1002 go to kamailio then to FS then return back to kamailio smoothly. Thus I would suggest that I change registering the gateway from FS to kamailio. but How to? --- On Fri, 5/25/12, SamyGo wrote: From: SamyGo Subject: Re: [Freeswitch-users] Freeswitch (SBC + Media) does not know that gateway phone is picked up To: "FreeSWITCH Users Help" Date: Friday, May 25, 2012, 8:08 AM Hi,These are SIP traces on FreeSWITCH console, whereas you are saying and it seems that kamailio is not detecting the answering (200 OK)of the call from extension 1002. Please, can you take a sip trace..!! I see you've both kamailio and FS on same server! Please take a pcap from the linux console using the following command. #tcpdump -i any -s 0 -w new-call-trace.pcap -vvv Please be quick on turning the sip trace on and off as quickly as possible to avoid extra packets. Once done open the file in wireshark ;?apply filter "sip || rtp" and then save the resulting capture in separate file. Send us the new file to analyse. One more silly question probably, I see REGISTERs coming to your FS as well and the calls to gateways are made from FS too !!, umm...just thinking what are you using kamailio for!!? ThanksBR,Sammy On Fri, May 25, 2012 at 4:40 AM, Sherif Omran wrote: Hi all, here is the siptrace: To figure it out 1- gateway called bluesip.net. It send invide using caller number at bluesip.net 2- This call should go to extension kb-1002. kb means go from freeswitch port 6090 to kamailio port 5060 3- It should go to call extension 1002 in Kamailio 4- Extension 1002 rings but when I reply, it does not notice I replied ./fs_cli ??????????? _____ ____???? ____ _???? ___????????????? ?????????? |? ___/ ___|?? / ___| |?? |_ _|???????????? ?????????? | |_? \___ \? | |?? | |??? | |??????????? ?????????? |? _|? ___) | | |___| |___ | |????????????? ?????????? |_|?? |____/?? \____|_____|___|??????????? ******************************************************* * Anthony Minessale II, Ken Rice,???????????????????? * * Michael Jerris, Travis Cross??????????????????????? * * FreeSWITCH (http://www.freeswitch.org)????????????? * * Paypal Donations Appreciated: paypal at freeswitch.org * * Brought to you by ClueCon http://www.cluecon.com/?? * ******************************************************* Type /help to see a list of commands +OK log level? [7] freeswitch at internal> tracelevel -ERR tracelevel Command not found! freeswitch at internal> sofia global siptrace on +OK Global siptrace on recv 1104 bytes from udp/[217.74.179.29]:5060 at 23:08:45.822962: ?? ------------------------------------------------------------------------ ?? INVITE sip:gw+trunk_1000 at 78.138.90.58:6090;transport=udp;gw=trunk_1000 SIP/2.0 ?? Record-Route: ?? Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 ?? Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 ?? From: "+41793940965" ;tag=as00589402 ?? To: ?? Contact: ?? Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net ?? CSeq: 102 INVITE ?? User-Agent: blueSIP PSTN GW ?? Max-Forwards: 69 ?? Date: Thu, 24 May 2012 23:08:44 GMT ?? Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY ?? Supported: replaces ?? Content-Type: application/sdp ?? Content-Length: 367 ?? P-hint: USRLOC ?? ?? v=0 ?? o=root 20076 20076 IN IP4 217.74.179.28 ?? s=session ?? c=IN IP4 217.74.179.28 ?? t=0 0 ?? m=audio 25626 RTP/AVP 8 0 18 111 3 101 ?? a=rtpmap:8 PCMA/8000 ?? a=rtpmap:0 PCMU/8000 ?? a=rtpmap:18 G729/8000 ?? a=fmtp:18 annexb=no ?? a=rtpmap:111 G726-32/8000 ?? a=rtpmap:3 GSM/8000 ?? a=rtpmap:101 telephone-event/8000 ?? a=fmtp:101 0-16 ?? a=silenceSupp:off - - - - ?? a=ptime:20 ?? a=sendrecv ?? ------------------------------------------------------------------------ send 494 bytes to udp/[217.74.179.29]:5060 at 23:08:45.844486: ?? ------------------------------------------------------------------------ ?? SIP/2.0 100 Trying ?? Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 ?? Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 ?? Record-Route: ?? From: "+41793940965" ;tag=as00589402 ?? To: ?? Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net ?? CSeq: 102 INVITE ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ 2012-05-24 23:08:45.840749 [NOTICE] switch_channel.c:926 New Channel sofia/internal/+41793940965 at bluesip.net [69cd22e6-a5f5-11e1-a833-7b1f73a7ffcf] 2012-05-24 23:08:45.840749 [DEBUG] sofia.c:5532 Channel sofia/internal/+41793940965 at bluesip.net entering state [received][100] 2012-05-24 23:08:45.840749 [DEBUG] sofia.c:5543 Remote SDP: v=0 o=root 20076 20076 IN IP4 217.74.179.28 s=session c=IN IP4 217.74.179.28 t=0 0 m=audio 25626 RTP/AVP 8 0 18 111 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:111 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMA:8:8000:20:64000]/[iLBC:97:8000:30:13330] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMU:0:8000:20:64000]/[iLBC:97:8000:30:13330] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G729:18:8000:20:8000]/[iLBC:97:8000:30:13330] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G726-32:111:8000:20:0]/[iLBC:97:8000:30:13330] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [GSM:3:8000:20:13200]/[iLBC:97:8000:30:13330] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [telephone-event:101:8000:20:0]/[iLBC:97:8000:30:13330] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf send/recv payload to 101 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMA:8:8000:20:64000]/[G7221:115:32000:20:48000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMU:0:8000:20:64000]/[G7221:115:32000:20:48000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G729:18:8000:20:8000]/[G7221:115:32000:20:48000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G726-32:111:8000:20:0]/[G7221:115:32000:20:48000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [GSM:3:8000:20:13200]/[G7221:115:32000:20:48000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [telephone-event:101:8000:20:0]/[G7221:115:32000:20:48000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf send/recv payload to 101 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMA:8:8000:20:64000]/[G7221:107:16000:20:32000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMU:0:8000:20:64000]/[G7221:107:16000:20:32000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G729:18:8000:20:8000]/[G7221:107:16000:20:32000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G726-32:111:8000:20:0]/[G7221:107:16000:20:32000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [GSM:3:8000:20:13200]/[G7221:107:16000:20:32000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [telephone-event:101:8000:20:0]/[G7221:107:16000:20:32000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf send/recv payload to 101 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMA:8:8000:20:64000]/[G722:9:8000:20:64000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G729:18:8000:20:8000]/[G722:9:8000:20:64000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G726-32:111:8000:20:0]/[G722:9:8000:20:64000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [GSM:3:8000:20:13200]/[G722:9:8000:20:64000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [telephone-event:101:8000:20:0]/[G722:9:8000:20:64000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf send/recv payload to 101 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMA:8:8000:20:64000]/[PCMU:0:8000:20:64000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:2996 Set Codec sofia/internal/+41793940965 at bluesip.net PCMU/8000 20 ms 160 samples 64000 bits 2012-05-24 23:08:45.840749 [DEBUG] switch_core_codec.c:111 sofia/internal/+41793940965 at bluesip.net Original read codec set to PCMU:0 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf send/recv payload to 101 2012-05-24 23:08:45.840749 [DEBUG] sofia.c:5757 (sofia/internal/+41793940965 at bluesip.net) State Change CS_NEW -> CS_INIT 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_INIT 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/+41793940965 at bluesip.net) State INIT 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:85 sofia/internal/+41793940965 at bluesip.net SOFIA INIT 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:125 (sofia/internal/+41793940965 at bluesip.net) State Change CS_INIT -> CS_ROUTING 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/+41793940965 at bluesip.net) State INIT going to sleep 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_ROUTING 2012-05-24 23:08:45.840749 [DEBUG] switch_channel.c:1886 (sofia/internal/+41793940965 at bluesip.net) Callstate Change DOWN -> RINGING 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/+41793940965 at bluesip.net) State ROUTING 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:148 sofia/internal/+41793940965 at bluesip.net SOFIA ROUTING 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:104 sofia/internal/+41793940965 at bluesip.net Standard ROUTING 2012-05-24 23:08:45.840749 [INFO] mod_dialplan_xml.c:485 Processing +41793940965 <+41793940965>->kb-1002 in context public Dialplan: sofia/internal/+41793940965 at bluesip.net parsing [public->from_kamailio] continue=false Dialplan: sofia/internal/+41793940965 at bluesip.net Regex (PASS) [from_kamailio] destination_number(kb-1002) =~ /^(.+)$/ break=on-false Dialplan: sofia/internal/+41793940965 at bluesip.net Action transfer(kb-1002 XML default) 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:154 (sofia/internal/+41793940965 at bluesip.net) State Change CS_ROUTING -> CS_EXECUTE 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/+41793940965 at bluesip.net) State ROUTING going to sleep 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_EXECUTE 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/+41793940965 at bluesip.net) State EXECUTE 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:241 sofia/internal/+41793940965 at bluesip.net SOFIA EXECUTE 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:192 sofia/internal/+41793940965 at bluesip.net Standard EXECUTE EXECUTE sofia/internal/+41793940965 at bluesip.net transfer(kb-1002 XML default) 2012-05-24 23:08:45.840749 [DEBUG] switch_ivr.c:1711 (sofia/internal/+41793940965 at bluesip.net) State Change CS_EXECUTE -> CS_ROUTING 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:731 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:08:45.840749 [NOTICE] switch_ivr.c:1717 Transfer sofia/internal/+41793940965 at bluesip.net to XML[kb-1002 at default] 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/+41793940965 at bluesip.net) State EXECUTE going to sleep 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_ROUTING 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/+41793940965 at bluesip.net) State ROUTING 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:148 sofia/internal/+41793940965 at bluesip.net SOFIA ROUTING 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:104 sofia/internal/+41793940965 at bluesip.net Standard ROUTING 2012-05-24 23:08:45.840749 [INFO] mod_dialplan_xml.c:485 Processing +41793940965 <+41793940965>->kb-1002 in context default Dialplan: sofia/internal/+41793940965 at bluesip.net parsing [default->vbox] continue=false Dialplan: sofia/internal/+41793940965 at bluesip.net Regex (FAIL) [vbox] destination_number(kb-1002) =~ /^vb-([0-9][0-9][0-9])$/ break=on-false Dialplan: sofia/internal/+41793940965 at bluesip.net parsing [default->vmenu] continue=false Dialplan: sofia/internal/+41793940965 at bluesip.net Regex (FAIL) [vmenu] destination_number(kb-1002) =~ /^vm-([0-9][0-9][0-9][0-9])$/ break=on-false Dialplan: sofia/internal/+41793940965 at bluesip.net parsing [default->kbridge] continue=false Dialplan: sofia/internal/+41793940965 at bluesip.net Regex (PASS) [kbridge] destination_number(kb-1002) =~ /^kb-(.+)$/ break=on-false Dialplan: sofia/internal/+41793940965 at bluesip.net Action set(proxy_media=true) Dialplan: sofia/internal/+41793940965 at bluesip.net Action set(call_timeout=50) Dialplan: sofia/internal/+41793940965 at bluesip.net Action set(continue_on_fail=true) Dialplan: sofia/internal/+41793940965 at bluesip.net Action set(hangup_after_bridge=true) Dialplan: sofia/internal/+41793940965 at bluesip.net Action set(sip_invite_domain=78.138.90.58) Dialplan: sofia/internal/+41793940965 at bluesip.net Action export(sip_contact_user=ufs) Dialplan: sofia/internal/+41793940965 at bluesip.net Action bridge(sofia/78.138.90.58/1002 at 78.138.90.58:5060) Dialplan: sofia/internal/+41793940965 at bluesip.net Action answer() Dialplan: sofia/internal/+41793940965 at bluesip.net Action voicemail(default ${domain_name} 1002) 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:154 (sofia/internal/+41793940965 at bluesip.net) State Change CS_ROUTING -> CS_EXECUTE 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/+41793940965 at bluesip.net) State ROUTING going to sleep 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_EXECUTE 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/+41793940965 at bluesip.net) State EXECUTE 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:241 sofia/internal/+41793940965 at bluesip.net SOFIA EXECUTE 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:192 sofia/internal/+41793940965 at bluesip.net Standard EXECUTE EXECUTE sofia/internal/+41793940965 at bluesip.net set(proxy_media=true) 2012-05-24 23:08:45.840749 [DEBUG] mod_dptools.c:1281 sofia/internal/+41793940965 at bluesip.net SET [proxy_media]=[true] EXECUTE sofia/internal/+41793940965 at bluesip.net set(call_timeout=50) 2012-05-24 23:08:45.840749 [DEBUG] mod_dptools.c:1281 sofia/internal/+41793940965 at bluesip.net SET [call_timeout]=[50] EXECUTE sofia/internal/+41793940965 at bluesip.net set(continue_on_fail=true) 2012-05-24 23:08:45.840749 [DEBUG] mod_dptools.c:1281 sofia/internal/+41793940965 at bluesip.net SET [continue_on_fail]=[true] EXECUTE sofia/internal/+41793940965 at bluesip.net set(hangup_after_bridge=true) 2012-05-24 23:08:45.840749 [DEBUG] mod_dptools.c:1281 sofia/internal/+41793940965 at bluesip.net SET [hangup_after_bridge]=[true] EXECUTE sofia/internal/+41793940965 at bluesip.net set(sip_invite_domain=78.138.90.58) 2012-05-24 23:08:45.840749 [DEBUG] mod_dptools.c:1281 sofia/internal/+41793940965 at bluesip.net SET [sip_invite_domain]=[78.138.90.58] EXECUTE sofia/internal/+41793940965 at bluesip.net export(sip_contact_user=ufs) 2012-05-24 23:08:45.840749 [DEBUG] switch_channel.c:1093 EXPORT (export_vars) [sip_contact_user]=[ufs] EXECUTE sofia/internal/+41793940965 at bluesip.net bridge(sofia/78.138.90.58/1002 at 78.138.90.58:5060) 2012-05-24 23:08:45.840749 [DEBUG] switch_channel.c:1047 sofia/internal/+41793940965 at bluesip.net EXPORTING[export_vars] [sip_contact_user]=[ufs] to event 2012-05-24 23:08:45.840749 [DEBUG] switch_ivr_originate.c:1884 Parsing global variables 2012-05-24 23:08:45.840749 [NOTICE] switch_channel.c:926 New Channel sofia/internal/1002 at 78.138.90.58:5060 [69ce1f0c-a5f5-11e1-a838-7b1f73a7ffcf] 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:4691 (sofia/internal/1002 at 78.138.90.58:5060) State Change CS_NEW -> CS_INIT 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1002 at 78.138.90.58:5060) Running State Change CS_INIT 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/1002 at 78.138.90.58:5060) State INIT 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:85 sofia/internal/1002 at 78.138.90.58:5060 SOFIA INIT 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:1871 sofia/internal/1002 at 78.138.90.58:5060 Patched SDP --- v=0 o=root 20076 20076 IN IP4 217.74.179.28 s=session c=IN IP4 217.74.179.28 t=0 0 m=audio 25626 RTP/AVP 8 0 18 111 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:111 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 +++ v=0 o=FreeSWITCH 1629510322 1629510323 IN IP4 78.138.90.58 s=FreeSWITCH c=IN IP4 78.138.90.58 t=0 0 m=audio 31178 RTP/AVP 8 0 18 111 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:111 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 send 1340 bytes to udp/[78.138.90.58]:5060 at 23:08:45.852280: ?? ------------------------------------------------------------------------ ?? INVITE sip:1002 at 78.138.90.58:5060 SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKapDgFeegKyHXK ?? Max-Forwards: 67 ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? CSeq: 28614398 INVITE ?? Contact: ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? 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: 372 ?? P-hint: USRLOC ?? X-FS-Support: update_display,send_info ?? Remote-Party-ID: "+41793940965" ;party=calling;screen=yes;privacy=off ?? ?? v=0 ?? o=FreeSWITCH 1629510322 1629510323 IN IP4 78.138.90.58 ?? s=FreeSWITCH ?? c=IN IP4 78.138.90.58 ?? t=0 0 ?? m=audio 31178 RTP/AVP 8 0 18 111 3 101 ?? a=rtpmap:8 PCMA/8000 ?? a=rtpmap:0 PCMU/8000 ?? a=rtpmap:18 G729/8000 ?? a=fmtp:18 annexb=no ?? a=rtpmap:111 G726-32/8000 ?? a=rtpmap:3 GSM/8000 ?? a=rtpmap:101 telephone-event/8000 ?? a=fmtp:101 0-16 ?? a=silenceSupp:off - - - - ?? a=ptime:20 ?? ------------------------------------------------------------------------ 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:125 (sofia/internal/1002 at 78.138.90.58:5060) State Change CS_INIT -> CS_ROUTING 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/1002 at 78.138.90.58:5060) State INIT going to sleep 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1002 at 78.138.90.58:5060) Running State Change CS_ROUTING 2012-05-24 23:08:45.840749 [DEBUG] switch_channel.c:1886 (sofia/internal/1002 at 78.138.90.58:5060) Callstate Change DOWN -> RINGING 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/1002 at 78.138.90.58:5060) State ROUTING 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:148 sofia/internal/1002 at 78.138.90.58:5060 SOFIA ROUTING 2012-05-24 23:08:45.840749 [DEBUG] switch_ivr_originate.c:66 (sofia/internal/1002 at 78.138.90.58:5060) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/1002 at 78.138.90.58:5060) State ROUTING going to sleep 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1002 at 78.138.90.58:5060) Running State Change CS_CONSUME_MEDIA 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:429 (sofia/internal/1002 at 78.138.90.58:5060) State CONSUME_MEDIA 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:429 (sofia/internal/1002 at 78.138.90.58:5060) State CONSUME_MEDIA going to sleep 2012-05-24 23:08:45.840749 [DEBUG] sofia.c:5532 Channel sofia/internal/1002 at 78.138.90.58:5060 entering state [calling][0] recv 365 bytes from udp/[78.138.90.58]:5060 at 23:08:45.964309: ?? ------------------------------------------------------------------------ ?? SIP/2.0 100 trying -- your call is important to us ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKapDgFeegKyHXK ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? CSeq: 28614398 INVITE ?? Server: kamailio (3.1.5 (x86_64/linux)) ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 496 bytes from udp/[78.138.90.58]:5060 at 23:08:47.414766: ?? ------------------------------------------------------------------------ ?? SIP/2.0 180 Ringing ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKapDgFeegKyHXK ?? Record-Route: ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ;tag=qCSaB2YJOAsAr2dJ-RWNMhPoDdn1RFvs ?? CSeq: 28614398 INVITE ?? Contact: "Mama" ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, REGISTER ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ 2012-05-24 23:08:47.400741 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:47.400741 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:47.400741 [DEBUG] sofia.c:5532 Channel sofia/internal/1002 at 78.138.90.58:5060 entering state [proceeding][180] 2012-05-24 23:08:47.400741 [NOTICE] sofia.c:5624 Ring-Ready sofia/internal/1002 at 78.138.90.58:5060! send 995 bytes to udp/[217.74.179.29]:5060 at 23:08:47.416538: ?? ------------------------------------------------------------------------ ?? SIP/2.0 180 Ringing ?? Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 ?? Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 ?? Record-Route: ?? From: "+41793940965" ;tag=as00589402 ?? To: ;tag=S7UZQygFt62Nm ?? Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net ?? CSeq: 102 INVITE ?? Contact: ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Accept: application/sdp ?? 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-Length: 0 ?? Remote-Party-ID: "Outbound Call" ;party=calling;privacy=off;screen=no ?? ?? ------------------------------------------------------------------------ 2012-05-24 23:08:47.400741 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:08:47.400741 [NOTICE] mod_sofia.c:2514 Ring-Ready sofia/internal/+41793940965 at bluesip.net! 2012-05-24 23:08:47.400741 [DEBUG] sofia.c:5532 Channel sofia/internal/+41793940965 at bluesip.net entering state [early][180] 2012-05-24 23:08:47.400741 [DEBUG] switch_core_session.c:731 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:08:47.400741 [NOTICE] switch_ivr_originate.c:483 Ring Ready sofia/internal/+41793940965 at bluesip.net! recv 496 bytes from udp/[78.138.90.58]:5060 at 23:08:47.804967: ?? ------------------------------------------------------------------------ ?? SIP/2.0 180 Ringing ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKapDgFeegKyHXK ?? Record-Route: ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ;tag=549D2DD03BBA7C67 ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? CSeq: 28614398 INVITE ?? Contact: ?? User-Agent: AVM FRITZ!Box Fon WLAN 7170 29.04.80 (Jan 27 2010) ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ 2012-05-24 23:08:47.800746 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:47.800746 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:47.800746 [DEBUG] sofia.c:5532 Channel sofia/internal/1002 at 78.138.90.58:5060 entering state [proceeding][180] recv 496 bytes from udp/[78.138.90.58]:5060 at 23:08:48.182098: ?? ------------------------------------------------------------------------ ?? SIP/2.0 180 Ringing ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKapDgFeegKyHXK ?? Record-Route: ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ;tag=qCSaB2YJOAsAr2dJ-RWNMhPoDdn1RFvs ?? CSeq: 28614398 INVITE ?? Contact: "Mama" ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, REGISTER ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ 2012-05-24 23:08:48.180741 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:48.180741 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:48.180741 [DEBUG] sofia.c:5532 Channel sofia/internal/1002 at 78.138.90.58:5060 entering state [proceeding][180] recv 496 bytes from udp/[78.138.90.58]:5060 at 23:08:48.417403: ?? ------------------------------------------------------------------------ ?? SIP/2.0 180 Ringing ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKapDgFeegKyHXK ?? Record-Route: ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ;tag=549D2DD03BBA7C67 ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? CSeq: 28614398 INVITE ?? Contact: ?? User-Agent: AVM FRITZ!Box Fon WLAN 7170 29.04.80 (Jan 27 2010) ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ 2012-05-24 23:08:48.400744 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:48.400744 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:48.400744 [DEBUG] sofia.c:5532 Channel sofia/internal/1002 at 78.138.90.58:5060 entering state [proceeding][180] recv 496 bytes from udp/[78.138.90.58]:5060 at 23:08:49.292599: ?? ------------------------------------------------------------------------ ?? SIP/2.0 180 Ringing ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKapDgFeegKyHXK ?? Record-Route: ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ;tag=549D2DD03BBA7C67 ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? CSeq: 28614398 INVITE ?? Contact: ?? User-Agent: AVM FRITZ!Box Fon WLAN 7170 29.04.80 (Jan 27 2010) ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ 2012-05-24 23:08:49.280742 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:49.280742 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:49.280742 [DEBUG] sofia.c:5532 Channel sofia/internal/1002 at 78.138.90.58:5060 entering state [proceeding][180] recv 478 bytes from udp/[217.74.179.29]:5060 at 23:09:15.066012: ?? ------------------------------------------------------------------------ ?? CANCEL sip:gw+trunk_1000 at 78.138.90.58:6090;transport=udp;gw=trunk_1000 SIP/2.0 ?? Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 ?? Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 ?? From: "+41793940965" ;tag=as00589402 ?? To: ?? Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net ?? CSeq: 102 CANCEL ?? User-Agent: blueSIP PSTN GW ?? Max-Forwards: 69 ?? Content-Length: 0 ?? P-hint: USRLOC ?? ?? ------------------------------------------------------------------------ send 369 bytes to udp/[217.74.179.29]:5060 at 23:09:15.066176: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 ?? Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 ?? From: "+41793940965" ;tag=as00589402 ?? To: ;tag=S7UZQygFt62Nm ?? Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net ?? CSeq: 102 CANCEL ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ send 766 bytes to udp/[217.74.179.29]:5060 at 23:09:15.066289: ?? ------------------------------------------------------------------------ ?? SIP/2.0 487 Request Terminated ?? Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 ?? Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 ?? From: "+41793940965" ;tag=as00589402 ?? To: ;tag=S7UZQygFt62Nm ?? Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net ?? CSeq: 102 INVITE ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? 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-Length: 0 ?? ?? ------------------------------------------------------------------------ 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:09:15.060742 [DEBUG] sofia.c:5532 Channel sofia/internal/+41793940965 at bluesip.net entering state [terminated][487] 2012-05-24 23:09:15.060742 [DEBUG] switch_channel.c:2848 (sofia/internal/+41793940965 at bluesip.net) Callstate Change RINGING -> HANGUP 2012-05-24 23:09:15.060742 [NOTICE] sofia.c:6301 Hangup sofia/internal/+41793940965 at bluesip.net [CS_EXECUTE] [ORIGINATOR_CANCEL] 2012-05-24 23:09:15.060742 [DEBUG] switch_channel.c:2871 Send signal sofia/internal/+41793940965 at bluesip.net [KILL] 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:09:15.060742 [DEBUG] switch_channel.c:2848 (sofia/internal/1002 at 78.138.90.58:5060) Callstate Change RINGING -> HANGUP 2012-05-24 23:09:15.060742 [NOTICE] switch_ivr_originate.c:3182 Hangup sofia/internal/1002 at 78.138.90.58:5060 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL] 2012-05-24 23:09:15.060742 [DEBUG] switch_channel.c:2871 Send signal sofia/internal/1002 at 78.138.90.58:5060 [KILL] 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1002 at 78.138.90.58:5060) Running State Change CS_HANGUP 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/1002 at 78.138.90.58:5060) State HANGUP 2012-05-24 23:09:15.060742 [DEBUG] mod_sofia.c:469 Channel sofia/internal/1002 at 78.138.90.58:5060 hanging up, cause: ORIGINATOR_CANCEL 2012-05-24 23:09:15.060742 [DEBUG] mod_sofia.c:523 Sending CANCEL to sofia/internal/1002 at 78.138.90.58:5060 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:47 sofia/internal/1002 at 78.138.90.58:5060 Standard HANGUP, cause: ORIGINATOR_CANCEL 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/1002 at 78.138.90.58:5060) State HANGUP going to sleep 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:393 (sofia/internal/1002 at 78.138.90.58:5060) State Change CS_HANGUP -> CS_REPORTING 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1002 at 78.138.90.58:5060) Running State Change CS_REPORTING 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/1002 at 78.138.90.58:5060) State REPORTING 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:79 sofia/internal/1002 at 78.138.90.58:5060 Standard REPORTING, cause: ORIGINATOR_CANCEL 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/1002 at 78.138.90.58:5060) State REPORTING going to sleep 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:387 (sofia/internal/1002 at 78.138.90.58:5060) State Change CS_REPORTING -> CS_DESTROY 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1382 Session 9 (sofia/internal/1002 at 78.138.90.58:5060) Locked, Waiting on external entities 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:09:15.060742 [DEBUG] switch_ivr_originate.c:3358 Originate Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL] 2012-05-24 23:09:15.060742 [NOTICE] switch_core_session.c:1400 Session 9 (sofia/internal/1002 at 78.138.90.58:5060) Ended 2012-05-24 23:09:15.060742 [NOTICE] switch_core_session.c:1402 Close Channel sofia/internal/1002 at 78.138.90.58:5060 [CS_DESTROY] 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:491 (sofia/internal/1002 at 78.138.90.58:5060) Callstate Change HANGUP -> DOWN 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:494 (sofia/internal/1002 at 78.138.90.58:5060) Running State Change CS_DESTROY 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/1002 at 78.138.90.58:5060) State DESTROY 2012-05-24 23:09:15.060742 [DEBUG] mod_sofia.c:374 sofia/internal/1002 at 78.138.90.58:5060 SOFIA DESTROY 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:86 sofia/internal/1002 at 78.138.90.58:5060 Standard DESTROY 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/1002 at 78.138.90.58:5060) State DESTROY going to sleep 2012-05-24 23:09:15.060742 [INFO] mod_dptools.c:2922 Originate Failed.? Cause: ORIGINATOR_CANCEL 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:2287 sofia/internal/+41793940965 at bluesip.net skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/+41793940965 at bluesip.net) State EXECUTE going to sleep 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_HANGUP 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/+41793940965 at bluesip.net) State HANGUP 2012-05-24 23:09:15.060742 [DEBUG] mod_sofia.c:469 Channel sofia/internal/+41793940965 at bluesip.net hanging up, cause: ORIGINATOR_CANCEL 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:47 sofia/internal/+41793940965 at bluesip.net Standard HANGUP, cause: ORIGINATOR_CANCEL 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/+41793940965 at bluesip.net) State HANGUP going to sleep 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:393 (sofia/internal/+41793940965 at bluesip.net) State Change CS_HANGUP -> CS_REPORTING 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_REPORTING 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/+41793940965 at bluesip.net) State REPORTING send 383 bytes to udp/[78.138.90.58]:5060 at 23:09:15.087695: ?? ------------------------------------------------------------------------ ?? CANCEL sip:1002 at 78.138.90.58:5060 SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKapDgFeegKyHXK ?? Max-Forwards: 67 ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? CSeq: 28614398 CANCEL ?? Reason: FreeSWITCH;cause=487;text="ORIGINATOR_CANCEL" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 408 bytes from udp/[217.74.179.29]:5060 at 23:09:15.094670: ?? ------------------------------------------------------------------------ ?? ACK sip:gw+trunk_1000 at 78.138.90.58:6090;transport=udp;gw=trunk_1000 SIP/2.0 ?? Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 ?? From: "+41793940965" ;tag=as00589402 ?? Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net ?? To: ;tag=S7UZQygFt62Nm ?? CSeq: 102 ACK ?? User-Agent: Sip EXpress router(0.9.7 (i386/linux)) ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:79 sofia/internal/+41793940965 at bluesip.net Standard REPORTING, cause: ORIGINATOR_CANCEL 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/+41793940965 at bluesip.net) State REPORTING going to sleep 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:387 (sofia/internal/+41793940965 at bluesip.net) State Change CS_REPORTING -> CS_DESTROY 2012-05-24 23:09:15.120807 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:09:15.120807 [DEBUG] switch_core_session.c:1382 Session 8 (sofia/internal/+41793940965 at bluesip.net) Locked, Waiting on external entities 2012-05-24 23:09:15.120807 [NOTICE] switch_core_session.c:1400 Session 8 (sofia/internal/+41793940965 at bluesip.net) Ended 2012-05-24 23:09:15.120807 [NOTICE] switch_core_session.c:1402 Close Channel sofia/internal/+41793940965 at bluesip.net [CS_DESTROY] 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:491 (sofia/internal/+41793940965 at bluesip.net) Callstate Change HANGUP -> DOWN 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:494 (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_DESTROY 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/+41793940965 at bluesip.net) State DESTROY 2012-05-24 23:09:15.120807 [DEBUG] mod_sofia.c:374 sofia/internal/+41793940965 at bluesip.net SOFIA DESTROY 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:86 sofia/internal/+41793940965 at bluesip.net Standard DESTROY 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/+41793940965 at bluesip.net) State DESTROY going to sleep recv 378 bytes from udp/[78.138.90.58]:5060 at 23:09:15.135235: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 canceling ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKapDgFeegKyHXK ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ;tag=a6a1c5f60faecf035a1ae5b6e96e979a-8c5b ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? CSeq: 28614398 CANCEL ?? Server: kamailio (3.1.5 (x86_64/linux)) ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 446 bytes from udp/[78.138.90.58]:5060 at 23:09:16.419308: ?? ------------------------------------------------------------------------ ?? SIP/2.0 487 Request Terminated ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKapDgFeegKyHXK ?? Record-Route: ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ;tag=qCSaB2YJOAsAr2dJ-RWNMhPoDdn1RFvs ?? CSeq: 28614398 INVITE ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, REGISTER ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 354 bytes to udp/[78.138.90.58]:5060 at 23:09:16.419543: ?? ------------------------------------------------------------------------ ?? ACK sip:1002 at 78.138.90.58:5060 SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKapDgFeegKyHXK ?? Max-Forwards: 67 ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ;tag=qCSaB2YJOAsAr2dJ-RWNMhPoDdn1RFvs ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? CSeq: 28614398 ACK ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 564 bytes from udp/[41.34.123.243]:58881 at 23:13:13.612418: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjHQIpv.ACHGbXD4IHrMr72LwZ8kYdlu-J ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=wOP.FVT4Crk8O5E.bB7BWUhNO6QG91IG ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60926 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 744 bytes to udp/[41.34.123.243]:58881 at 23:13:13.613152: ?? ------------------------------------------------------------------------ ?? SIP/2.0 401 Unauthorized ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjHQIpv.ACHGbXD4IHrMr72LwZ8kYdlu-J;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=wOP.FVT4Crk8O5E.bB7BWUhNO6QG91IG ?? To: "Sherif 1003" ;tag=USeHUmjpmrFUB ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60926 REGISTER ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? WWW-Authenticate: Digest realm="sip.pcfone.com", nonce="09674764-a5f6-11e1-a83c-7b1f73a7ffcf", algorithm=MD5, qop="auth" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 830 bytes from udp/[41.34.123.243]:58881 at 23:13:13.847746: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjFr7rIAdAJPRQd2GmrMrVib3KT3Rup0q4 ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=wOP.FVT4Crk8O5E.bB7BWUhNO6QG91IG ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60927 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Authorization: Digest username="1002", realm="sip.pcfone.com", nonce="09674764-a5f6-11e1-a83c-7b1f73a7ffcf", uri="sip:sip.pcfone.com:6090", response="3bc0bb82a803ded3641704c4eb78d58f", algorithm=MD5, cnonce="sc3AJfhuev18Bs-L-6tORmI.RXlxnPOz", qop=auth, nc=00000001 ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 706 bytes to udp/[41.34.123.243]:58881 at 23:13:13.850197: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjFr7rIAdAJPRQd2GmrMrVib3KT3Rup0q4;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=wOP.FVT4Crk8O5E.bB7BWUhNO6QG91IG ?? To: "Sherif 1003" ;tag=v279vF3SH15DQ ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60927 REGISTER ?? Contact: ;expires=300 ?? Date: Thu, 24 May 2012 23:13:13 GMT ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ send 989 bytes to udp/[41.34.123.243]:58881 at 23:13:13.888717: ?? ------------------------------------------------------------------------ ?? NOTIFY sip:1002 at 192.168.178.26:58881;ob SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKBZ68g9yKg77FF ?? Route: ;ob ?? Max-Forwards: 70 ?? From: ;tag=XB12yamXeav0j ?? To: ?? Call-ID: e0efa252-2098-1230-8985-00163e6bb553 ?? CSeq: 28614532 NOTIFY ?? Contact: ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? 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=noresource ?? Content-Type: application/simple-message-summary ?? Content-Length: 91 ?? ?? Messages-Waiting: yes ?? Message-Account: sip:1002 at 78.138.90.58 ?? Voice-Message: 1/0 (0/0) ?? ?? ------------------------------------------------------------------------ recv 308 bytes from udp/[41.34.123.243]:58881 at 23:13:14.298850: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;received=78.138.90.58;branch=z9hG4bKBZ68g9yKg77FF ?? Call-ID: e0efa252-2098-1230-8985-00163e6bb553 ?? From: ;tag=XB12yamXeav0j ?? To: ;tag=z9hG4bKBZ68g9yKg77FF ?? CSeq: 28614532 NOTIFY ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ recv 564 bytes from udp/[41.34.123.243]:58881 at 23:18:09.067232: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjABTslmZIhiHYRtLqQn2erwmX19MVYHFC ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=JBSXL5WZLWbLzpvRFoTSP5x8qQslI1ST ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60928 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 744 bytes to udp/[41.34.123.243]:58881 at 23:18:09.068121: ?? ------------------------------------------------------------------------ ?? SIP/2.0 401 Unauthorized ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjABTslmZIhiHYRtLqQn2erwmX19MVYHFC;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=JBSXL5WZLWbLzpvRFoTSP5x8qQslI1ST ?? To: "Sherif 1003" ;tag=ymtU0540BKjKe ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60928 REGISTER ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? WWW-Authenticate: Digest realm="sip.pcfone.com", nonce="b9821df4-a5f6-11e1-a83d-7b1f73a7ffcf", algorithm=MD5, qop="auth" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 830 bytes from udp/[41.34.123.243]:58881 at 23:18:09.302944: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjRgtztcy63vMT97Jp5pH9n311li5EFeY- ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=JBSXL5WZLWbLzpvRFoTSP5x8qQslI1ST ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60929 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Authorization: Digest username="1002", realm="sip.pcfone.com", nonce="b9821df4-a5f6-11e1-a83d-7b1f73a7ffcf", uri="sip:sip.pcfone.com:6090", response="519c6301816b51f3bbb98d97a347e92b", algorithm=MD5, cnonce="sc3AJfhuev18Bs-L-6tORmI.RXlxnPOz", qop=auth, nc=00000001 ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 706 bytes to udp/[41.34.123.243]:58881 at 23:18:09.305217: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjRgtztcy63vMT97Jp5pH9n311li5EFeY-;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=JBSXL5WZLWbLzpvRFoTSP5x8qQslI1ST ?? To: "Sherif 1003" ;tag=ZXKm20N48U85S ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60929 REGISTER ?? Contact: ;expires=300 ?? Date: Thu, 24 May 2012 23:18:09 GMT ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ send 989 bytes to udp/[41.34.123.243]:58881 at 23:18:09.379987: ?? ------------------------------------------------------------------------ ?? NOTIFY sip:1002 at 192.168.178.26:58881;ob SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKc8Z1j4FQDgy2a ?? Route: ;ob ?? Max-Forwards: 70 ?? From: ;tag=06cD4U6754yrN ?? To: ?? Call-ID: 91100602-2099-1230-8985-00163e6bb553 ?? CSeq: 28614680 NOTIFY ?? Contact: ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? 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=noresource ?? Content-Type: application/simple-message-summary ?? Content-Length: 91 ?? ?? Messages-Waiting: yes ?? Message-Account: sip:1002 at 78.138.90.58 ?? Voice-Message: 1/0 (0/0) ?? ?? ------------------------------------------------------------------------ recv 308 bytes from udp/[41.34.123.243]:58881 at 23:18:09.773629: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;received=78.138.90.58;branch=z9hG4bKc8Z1j4FQDgy2a ?? Call-ID: 91100602-2099-1230-8985-00163e6bb553 ?? From: ;tag=06cD4U6754yrN ?? To: ;tag=z9hG4bKc8Z1j4FQDgy2a ?? CSeq: 28614680 NOTIFY ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ recv 564 bytes from udp/[41.34.123.243]:58881 at 23:23:04.522455: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjaCuKyaBi6ZqcsHyp5HhH3WthJStom0SK ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=qfUt1fQWuE-Ro3zWld6X0r6WidtD6B1q ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60930 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 744 bytes to udp/[41.34.123.243]:58881 at 23:23:04.523158: ?? ------------------------------------------------------------------------ ?? SIP/2.0 401 Unauthorized ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjaCuKyaBi6ZqcsHyp5HhH3WthJStom0SK;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=qfUt1fQWuE-Ro3zWld6X0r6WidtD6B1q ?? To: "Sherif 1003" ;tag=1F655pQB3DNBH ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60930 REGISTER ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? WWW-Authenticate: Digest realm="sip.pcfone.com", nonce="699cfe2a-a5f7-11e1-a83e-7b1f73a7ffcf", algorithm=MD5, qop="auth" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 830 bytes from udp/[41.34.123.243]:58881 at 23:23:04.757798: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjs8SAufeoyzCLrePaAbYhZuOqQF47XIc- ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=qfUt1fQWuE-Ro3zWld6X0r6WidtD6B1q ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60931 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Authorization: Digest username="1002", realm="sip.pcfone.com", nonce="699cfe2a-a5f7-11e1-a83e-7b1f73a7ffcf", uri="sip:sip.pcfone.com:6090", response="23a886934aaf049f08432b6bb226bd37", algorithm=MD5, cnonce="sc3AJfhuev18Bs-L-6tORmI.RXlxnPOz", qop=auth, nc=00000001 ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 706 bytes to udp/[41.34.123.243]:58881 at 23:23:04.759924: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjs8SAufeoyzCLrePaAbYhZuOqQF47XIc-;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=qfUt1fQWuE-Ro3zWld6X0r6WidtD6B1q ?? To: "Sherif 1003" ;tag=2rZy7H8e0pByc ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60931 REGISTER ?? Contact: ;expires=300 ?? Date: Thu, 24 May 2012 23:23:04 GMT ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ send 989 bytes to udp/[41.34.123.243]:58881 at 23:23:04.872728: ?? ------------------------------------------------------------------------ ?? NOTIFY sip:1002 at 192.168.178.26:58881;ob SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKDHStmZ0taSmNp ?? Route: ;ob ?? Max-Forwards: 70 ?? From: ;tag=31rQ9cSjXZ1gr ?? To: ?? Call-ID: 4130a1ce-209a-1230-8985-00163e6bb553 ?? CSeq: 28614828 NOTIFY ?? Contact: ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? 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=noresource ?? Content-Type: application/simple-message-summary ?? Content-Length: 91 ?? ?? Messages-Waiting: yes ?? Message-Account: sip:1002 at 78.138.90.58 ?? Voice-Message: 1/0 (0/0) ?? ?? ------------------------------------------------------------------------ recv 308 bytes from udp/[41.34.123.243]:58881 at 23:23:05.269063: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;received=78.138.90.58;branch=z9hG4bKDHStmZ0taSmNp ?? Call-ID: 4130a1ce-209a-1230-8985-00163e6bb553 ?? From: ;tag=31rQ9cSjXZ1gr ?? To: ;tag=z9hG4bKDHStmZ0taSmNp ?? CSeq: 28614828 NOTIFY ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 916 bytes to udp/[217.74.179.29]:5060 at 23:25:45.873423: ?? ------------------------------------------------------------------------ ?? REGISTER sip:bluesip.net;transport=udp SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKetjKptHy71a8H ?? Max-Forwards: 70 ?? From: ;tag=6r0vBQZS650Fg ?? To: ?? Call-ID: 2a3e4696-a5b9-11e1-a7ed-7b1f73a7ffcf ?? CSeq: 28601493 REGISTER ?? Contact: ?? Expires: 3600 ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? Authorization: Digest username="bluesip/sherifomran", realm="bluesip.net", nonce="4fbeba4ebfb7414d5c701bd6f994402d551f3acf", algorithm=MD5, uri="sip:bluesip.net;transport=udp", response="c6cdcafe0418e519bc9ee0d8fa3d4d74" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 722 bytes from udp/[217.74.179.29]:5060 at 23:25:45.904176: ?? ------------------------------------------------------------------------ ?? SIP/2.0 401 Unauthorized ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKetjKptHy71a8H ?? From: ;tag=6r0vBQZS650Fg ?? To: ;tag=0354a2e1b960c9cc2279eca4e5f84e20.1b16 ?? Call-ID: 2a3e4696-a5b9-11e1-a7ed-7b1f73a7ffcf ?? CSeq: 28601493 REGISTER ?? WWW-Authenticate: Digest realm="bluesip.net", nonce="4fbec4a4ba82d1f1feaa32ccec01b85be054fcf7", stale=true ?? Server: Sip EXpress router (0.9.7 (i386/linux)) ?? Content-Length: 0 ?? Warning: 392 217.74.179.29:5060 "Noisy feedback tells:? pid=5455 req_src_ip=78.138.90.58 req_src_port=6090 in_uri=sip:bluesip.net;transport=udp out_uri=sip:bluesip.net;transport=udp via_cnt==1" ?? ?? ------------------------------------------------------------------------ send 916 bytes to udp/[217.74.179.29]:5060 at 23:25:45.904416: ?? ------------------------------------------------------------------------ ?? REGISTER sip:bluesip.net;transport=udp SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKF3BcrN214a1tD ?? Max-Forwards: 70 ?? From: ;tag=6r0vBQZS650Fg ?? To: ?? Call-ID: 2a3e4696-a5b9-11e1-a7ed-7b1f73a7ffcf ?? CSeq: 28601494 REGISTER ?? Contact: ?? Expires: 3600 ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? Authorization: Digest username="bluesip/sherifomran", realm="bluesip.net", nonce="4fbec4a4ba82d1f1feaa32ccec01b85be054fcf7", algorithm=MD5, uri="sip:bluesip.net;transport=udp", response="4c09dbe4b9accb52d4104b40dfe20040" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 699 bytes from udp/[217.74.179.29]:5060 at 23:25:45.935046: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKF3BcrN214a1tD ?? From: ;tag=6r0vBQZS650Fg ?? To: ;tag=0354a2e1b960c9cc2279eca4e5f84e20.2017 ?? Call-ID: 2a3e4696-a5b9-11e1-a7ed-7b1f73a7ffcf ?? CSeq: 28601494 REGISTER ?? Contact: ;q=0.5;expires=3600 ?? Server: Sip EXpress router (0.9.7 (i386/linux)) ?? Content-Length: 0 ?? Warning: 392 217.74.179.29:5060 "Noisy feedback tells:? pid=5462 req_src_ip=78.138.90.58 req_src_port=6090 in_uri=sip:bluesip.net;transport=udp out_uri=sip:bluesip.net;transport=udp via_cnt==1" ?? ?? ------------------------------------------------------------------------ recv 564 bytes from udp/[41.34.123.243]:58881 at 23:27:59.977057: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjEOP.a4NoXqoFqC7a.k2tyeX-mYlUyH4H ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60932 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 744 bytes to udp/[41.34.123.243]:58881 at 23:27:59.977960: ?? ------------------------------------------------------------------------ ?? SIP/2.0 401 Unauthorized ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjEOP.a4NoXqoFqC7a.k2tyeX-mYlUyH4H;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K ?? To: "Sherif 1003" ;tag=4ajgB89Nt8Q3K ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60932 REGISTER ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? WWW-Authenticate: Digest realm="sip.pcfone.com", nonce="19b7cc40-a5f8-11e1-a83f-7b1f73a7ffcf", algorithm=MD5, qop="auth" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 564 bytes from udp/[41.34.123.243]:58881 at 23:28:00.468487: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjEOP.a4NoXqoFqC7a.k2tyeX-mYlUyH4H ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60932 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 744 bytes to udp/[41.34.123.243]:58881 at 23:28:00.468679: ?? ------------------------------------------------------------------------ ?? SIP/2.0 401 Unauthorized ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjEOP.a4NoXqoFqC7a.k2tyeX-mYlUyH4H;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K ?? To: "Sherif 1003" ;tag=4ajgB89Nt8Q3K ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60932 REGISTER ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? WWW-Authenticate: Digest realm="sip.pcfone.com", nonce="19b7cc40-a5f8-11e1-a83f-7b1f73a7ffcf", algorithm=MD5, qop="auth" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 830 bytes from udp/[41.34.123.243]:58881 at 23:28:00.717446: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPj0iHd3-XUjnvcBs9DiJUCWzrmYTmWxFLt ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60933 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Authorization: Digest username="1002", realm="sip.pcfone.com", nonce="19b7cc40-a5f8-11e1-a83f-7b1f73a7ffcf", uri="sip:sip.pcfone.com:6090", response="e21c903e6d98b5adcdd2eefa75217157", algorithm=MD5, cnonce="sc3AJfhuev18Bs-L-6tORmI.RXlxnPOz", qop=auth, nc=00000001 ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 706 bytes to udp/[41.34.123.243]:58881 at 23:28:00.719947: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPj0iHd3-XUjnvcBs9DiJUCWzrmYTmWxFLt;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K ?? To: "Sherif 1003" ;tag=5KB9c3tSQHepF ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60933 REGISTER ?? Contact: ;expires=300 ?? Date: Thu, 24 May 2012 23:28:00 GMT ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ send 989 bytes to udp/[41.34.123.243]:58881 at 23:28:00.868368: ?? ------------------------------------------------------------------------ ?? NOTIFY sip:1002 at 192.168.178.26:58881;ob SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKgc54SgK51KQDS ?? Route: ;ob ?? Max-Forwards: 70 ?? From: ;tag=6v41eyBXmt48a ?? To: ?? Call-ID: f19dfbce-209a-1230-8985-00163e6bb553 ?? CSeq: 28614976 NOTIFY ?? Contact: ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? 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=noresource ?? Content-Type: application/simple-message-summary ?? Content-Length: 91 ?? ?? Messages-Waiting: yes ?? Message-Account: sip:1002 at 78.138.90.58 ?? Voice-Message: 1/0 (0/0) ?? ?? ------------------------------------------------------------------------ recv 308 bytes from udp/[41.34.123.243]:58881 at 23:28:01.128279: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;received=78.138.90.58;branch=z9hG4bKgc54SgK51KQDS ?? Call-ID: f19dfbce-209a-1230-8985-00163e6bb553 ?? From: ;tag=6v41eyBXmt48a ?? To: ;tag=z9hG4bKgc54SgK51KQDS ?? CSeq: 28614976 NOTIFY ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 913 bytes to udp/[217.74.179.29]:5060 at 23:28:11.885213: ?? ------------------------------------------------------------------------ ?? REGISTER sip:bluesip.net;transport=udp SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKHNyXUB48yvD0m ?? Max-Forwards: 70 ?? From: ;tag=5F739Uep9vaXm ?? To: ?? Call-ID: 2a3e49c0-a5b9-11e1-a7ee-7b1f73a7ffcf ?? CSeq: 28601493 REGISTER ?? Contact: ?? Expires: 3600 ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? Authorization: Digest username="bluesip/salahomran", realm="bluesip.net", nonce="4fbebeca79a5cff7c417463814cb9054d313bcac", algorithm=MD5, uri="sip:bluesip.net;transport=udp", response="bb1babb90f4ea1dc8fbb9aa45d6038c7" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 720 bytes from udp/[217.74.179.29]:5060 at 23:28:11.915764: ?? ------------------------------------------------------------------------ ?? SIP/2.0 401 Unauthorized ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKHNyXUB48yvD0m ?? From: ;tag=5F739Uep9vaXm ?? To: ;tag=0354a2e1b960c9cc2279eca4e5f84e20.3454 ?? Call-ID: 2a3e49c0-a5b9-11e1-a7ee-7b1f73a7ffcf ?? CSeq: 28601493 REGISTER ?? WWW-Authenticate: Digest realm="bluesip.net", nonce="4fbec5365af3a174f1cc0b8cda770c727d25bf71", stale=true ?? Server: Sip EXpress router (0.9.7 (i386/linux)) ?? Content-Length: 0 ?? Warning: 392 217.74.179.29:5060 "Noisy feedback tells:? pid=5462 req_src_ip=78.138.90.58 req_src_port=6090 in_uri=sip:bluesip.net;transport=udp out_uri=sip:bluesip.net;transport=udp via_cnt==1" ?? ?? ------------------------------------------------------------------------ send 913 bytes to udp/[217.74.179.29]:5060 at 23:28:11.915914: ?? ------------------------------------------------------------------------ ?? REGISTER sip:bluesip.net;transport=udp SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKjyQpX6mcv53jg ?? Max-Forwards: 70 ?? From: ;tag=5F739Uep9vaXm ?? To: ?? Call-ID: 2a3e49c0-a5b9-11e1-a7ee-7b1f73a7ffcf ?? CSeq: 28601494 REGISTER ?? Contact: ?? Expires: 3600 ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? Authorization: Digest username="bluesip/salahomran", realm="bluesip.net", nonce="4fbec5365af3a174f1cc0b8cda770c727d25bf71", algorithm=MD5, uri="sip:bluesip.net;transport=udp", response="e260f6dc9b01ce8bea8c5ffb36bd4ddd" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 697 bytes from udp/[217.74.179.29]:5060 at 23:28:11.946325: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKjyQpX6mcv53jg ?? From: ;tag=5F739Uep9vaXm ?? To: ;tag=0354a2e1b960c9cc2279eca4e5f84e20.4b5c ?? Call-ID: 2a3e49c0-a5b9-11e1-a7ee-7b1f73a7ffcf ?? CSeq: 28601494 REGISTER ?? Contact: ;q=0.5;expires=3600 ?? Server: Sip EXpress router (0.9.7 (i386/linux)) ?? Content-Length: 0 ?? Warning: 392 217.74.179.29:5060 "Noisy feedback tells:? pid=5455 req_src_ip=78.138.90.58 req_src_port=6090 in_uri=sip:bluesip.net;transport=udp out_uri=sip:bluesip.net;transport=udp via_cnt==1" ?? ?? ------------------------------------------------------------------------ recv 564 bytes from udp/[41.34.123.243]:58881 at 23:32:55.930149: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjuk8DZGiaXfN7BmtslNHVjtSg5UXsbHXc ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=Pz.t-0tGXj3saM.ehBiB1EAMmmitze1D ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60934 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 744 bytes to udp/[41.34.123.243]:58881 at 23:32:55.931324: ?? ------------------------------------------------------------------------ ?? SIP/2.0 401 Unauthorized ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjuk8DZGiaXfN7BmtslNHVjtSg5UXsbHXc;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=Pz.t-0tGXj3saM.ehBiB1EAMmmitze1D ?? To: "Sherif 1003" ;tag=75XtgSv0H3tUp ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60934 REGISTER ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? WWW-Authenticate: Digest realm="sip.pcfone.com", nonce="ca1eabee-a5f8-11e1-a840-7b1f73a7ffcf", algorithm=MD5, qop="auth" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 830 bytes from udp/[41.34.123.243]:58881 at 23:32:56.165624: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjKEQ6xXmyzZxA6vCM88vHNYAOPAjUqFsD ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=Pz.t-0tGXj3saM.ehBiB1EAMmmitze1D ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60935 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Authorization: Digest username="1002", realm="sip.pcfone.com", nonce="ca1eabee-a5f8-11e1-a840-7b1f73a7ffcf", uri="sip:sip.pcfone.com:6090", response="ff6d12f4ec7abd4b37187222b3af2358", algorithm=MD5, cnonce="sc3AJfhuev18Bs-L-6tORmI.RXlxnPOz", qop=auth, nc=00000001 ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 706 bytes to udp/[41.34.123.243]:58881 at 23:32:56.168033: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjKEQ6xXmyzZxA6vCM88vHNYAOPAjUqFsD;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=Pz.t-0tGXj3saM.ehBiB1EAMmmitze1D ?? To: "Sherif 1003" ;tag=8eQKjmD4ecHej ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60935 REGISTER ?? Contact: ;expires=300 ?? Date: Thu, 24 May 2012 23:32:56 GMT ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ send 989 bytes to udp/[41.34.123.243]:58881 at 23:32:56.360238: ?? ------------------------------------------------------------------------ ?? NOTIFY sip:1002 at 192.168.178.26:58881;ob SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKK7gFZ15FSet5B ?? Route: ;ob ?? Max-Forwards: 70 ?? From: ;tag=9QgcmFy7BN70D ?? To: ?? Call-ID: a1be7708-209b-1230-8985-00163e6bb553 ?? CSeq: 28615124 NOTIFY ?? Contact: ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? 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=noresource ?? Content-Type: application/simple-message-summary ?? Content-Length: 91 ?? ?? Messages-Waiting: yes ?? Message-Account: sip:1002 at 78.138.90.58 ?? Voice-Message: 1/0 (0/0) ?? ?? ------------------------------------------------------------------------ recv 308 bytes from udp/[41.34.123.243]:58881 at 23:32:56.623554: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;received=78.138.90.58;branch=z9hG4bKK7gFZ15FSet5B ?? Call-ID: a1be7708-209b-1230-8985-00163e6bb553 ?? From: ;tag=9QgcmFy7BN70D ?? To: ;tag=z9hG4bKK7gFZ15FSet5B ?? CSeq: 28615124 NOTIFY ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ freeswitch at internal> --- On Thu, 5/24/12, Kristian Kielhofner wrote: From: Kristian Kielhofner Subject: Re: [Freeswitch-users] Freeswitch (SBC + Media) does not know that gateway phone is picked up To: "FreeSWITCH Users Help" Date: Thursday, May 24, 2012, 7:51 PM Siptrace and logs please. On Thu, May 24, 2012 at 11:53 AM, Sherif Omran wrote: > > Hi all, > > My topology is as follows: > > Kamailio? -> FS (SBS+Media server) > > I came across an issue with my system as follows.? I have a Hardphone registered. When I do local call inside kamailio, it gets to FS and returns back well and FS understands when I lift the handset. However, I added a gateway (german landline server), when I call my self from another phone, the call gets to FS and then transmits to Kamailio, it rings my extension but when I lift the handset FS does not notice it and keeps ringing. > > Any body has an Idea? Here is my gateway trunk. > > > ??????? > ??????? > ??????? > ??????? > ??????? > ????? > > > thanks in advance > Sherif Omran > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Kristian Kielhofner _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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 at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -----Inline Attachment Follows----- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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 at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -----Inline Attachment Follows----- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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 at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -----Inline Attachment Follows----- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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 at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -----Inline Attachment Follows----- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120527/0ae138a6/attachment-0001.html From sherifomran2000 at yahoo.com Sun May 27 13:32:05 2012 From: sherifomran2000 at yahoo.com (Sherif Omran) Date: Sun, 27 May 2012 02:32:05 -0700 (PDT) Subject: [Freeswitch-users] how to alert existing voice mail Message-ID: <1338111125.35732.YahooMailClassic@web110812.mail.gq1.yahoo.com> Hello guys, I need to alert existing voice mail by either one of the following ways 1- send email 2- FS server should call every 1 hr to deliver the message any ideas on how to commit it thanks Sherif -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120527/bc74a68e/attachment.html From albert_nguyen16 at hotmail.com Sun May 27 15:47:19 2012 From: albert_nguyen16 at hotmail.com (Albert Nguyen) Date: Sun, 27 May 2012 11:47:19 +0000 Subject: [Freeswitch-users] What is the different between FS linux and MS windows version Message-ID: Hi I have installed a FS windows version on a DELL R210 server running windows 2003 server. Is there any different between the FS windows and linux version in term of capability and performance? Thanks in advance, Al -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120527/ee091d73/attachment.html From saami_mh at ymail.com Sun May 27 16:59:10 2012 From: saami_mh at ymail.com (Samira Mh) Date: Sun, 27 May 2012 05:59:10 -0700 (PDT) Subject: [Freeswitch-users] query on freeswitch Message-ID: <1338123550.34364.YahooMailNeo@web120104.mail.ne1.yahoo.com> how to implement mod query odbc on dialplan of freeswitch? the wiki is colud't help me? plz help? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120527/3e25028c/attachment.html From peter.olsson at visionutveckling.se Sun May 27 17:02:40 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Sun, 27 May 2012 13:02:40 +0000 Subject: [Freeswitch-users] What is the different between FS linux and MS windows version In-Reply-To: References: Message-ID: <1FFF97C269757C458224B7C895F35F150D9F47@cantor.std.visionutv.se> I would say it's more or less the same (FS build natively on both platforms). I'm using it on both platforms, and it's been working perfect on both. There is a bigger user base on Linux I guess (if that's important), and if you want to run HA-setups, you would need Linux to setup Kamailio etc. If you're running Windows, I recommend ________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f?r Albert Nguyen [albert_nguyen16 at hotmail.com] Skickat: den 27 maj 2012 13:47 Till: freeswitch-users at lists.freeswitch.org ?mne: [Freeswitch-users] What is the different between FS linux and MS windows version Hi I have installed a FS windows version on a DELL R210 server running windows 2003 server. Is there any different between the FS windows and linux version in term of capability and performance? Thanks in advance, Al !DSPAM:4fc214f832761559443750! From peter.olsson at visionutveckling.se Sun May 27 17:05:03 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Sun, 27 May 2012 13:05:03 +0000 Subject: [Freeswitch-users] What is the different between FS linux and MS windows version In-Reply-To: <1FFF97C269757C458224B7C895F35F150D9F47@cantor.std.visionutv.se> References: , <1FFF97C269757C458224B7C895F35F150D9F47@cantor.std.visionutv.se> Message-ID: <1FFF97C269757C458224B7C895F35F150D9F55@cantor.std.visionutv.se> Hit "send" to soon... :) What I meant to say last was that I recommend using Windows 2008 R2, or at least Windows 2008. Microsoft has improved their timing in the kernel, starting with these kernels (and also Vista / Windows 7). /Peter ________________________________ Fr?n: Peter Olsson Skickat: den 27 maj 2012 15:02 Till: FreeSWITCH Users Help ?mne: RE: [Freeswitch-users] What is the different between FS linux and MS windows version I would say it's more or less the same (FS build natively on both platforms). I'm using it on both platforms, and it's been working perfect on both. There is a bigger user base on Linux I guess (if that's important), and if you want to run HA-setups, you would need Linux to setup Kamailio etc. If you're running Windows, I recommend ________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f?r Albert Nguyen [albert_nguyen16 at hotmail.com] Skickat: den 27 maj 2012 13:47 Till: freeswitch-users at lists.freeswitch.org ?mne: [Freeswitch-users] What is the different between FS linux and MS windows version Hi I have installed a FS windows version on a DELL R210 server running windows 2003 server. Is there any different between the FS windows and linux version in term of capability and performance? Thanks in advance, Al !DSPAM:4fc214f832761559443750! From albert_nguyen16 at hotmail.com Sun May 27 17:15:35 2012 From: albert_nguyen16 at hotmail.com (Albert Nguyen) Date: Sun, 27 May 2012 13:15:35 +0000 Subject: [Freeswitch-users] What is the different between FS linux and MS windows version In-Reply-To: <1FFF97C269757C458224B7C895F35F150D9F55@cantor.std.visionutv.se> References: , , <1FFF97C269757C458224B7C895F35F150D9F47@cantor.std.visionutv.se>, <1FFF97C269757C458224B7C895F35F150D9F55@cantor.std.visionutv.se> Message-ID: Hi Peter, Thanks very much for your info. I'll put it on win 2008 R2. Regards, Al > From: peter.olsson at visionutveckling.se > To: freeswitch-users at lists.freeswitch.org > Date: Sun, 27 May 2012 13:05:03 +0000 > Subject: Re: [Freeswitch-users] What is the different between FS linux and MS windows version > > Hit "send" to soon... :) > > What I meant to say last was that I recommend using Windows 2008 R2, or at least Windows 2008. Microsoft has improved their timing in the kernel, starting with these kernels (and also Vista / Windows 7). > > /Peter > ________________________________ > Fr?n: Peter Olsson > Skickat: den 27 maj 2012 15:02 > Till: FreeSWITCH Users Help > ?mne: RE: [Freeswitch-users] What is the different between FS linux and MS windows version > > I would say it's more or less the same (FS build natively on both platforms). > > I'm using it on both platforms, and it's been working perfect on both. There is a bigger user base on Linux I guess (if that's important), and if you want to run HA-setups, you would need Linux to setup Kamailio etc. > > If you're running Windows, I recommend > > > > ________________________________ > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f?r Albert Nguyen [albert_nguyen16 at hotmail.com] > Skickat: den 27 maj 2012 13:47 > Till: freeswitch-users at lists.freeswitch.org > ?mne: [Freeswitch-users] What is the different between FS linux and MS windows version > > Hi > > I have installed a FS windows version on a DELL R210 server running windows 2003 server. Is there any different between the FS windows and linux version in term of capability and performance? > > Thanks in advance, > > Al > !DSPAM:4fc214f832761559443750! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120527/65e093d5/attachment.html From govoiper at gmail.com Sun May 27 17:26:11 2012 From: govoiper at gmail.com (SamyGo) Date: Sun, 27 May 2012 18:26:11 +0500 Subject: [Freeswitch-users] Freeswitch (SBC + Media) does not know that gateway phone is picked up In-Reply-To: <1338111001.12711.YahooMailClassic@web110816.mail.gq1.yahoo.com> References: <1338111001.12711.YahooMailClassic@web110816.mail.gq1.yahoo.com> Message-ID: Its good to read that its working for you, I'm still not sure how disabling your proxying media relate to the correction of the SIP signalling flow for a call !! I would still like to investigate your final solution to this. Caller====>Provider=====> FS ====>Kamailio====>Callee Kamailio keeps on returning ringing FS keeps on relaying ringing 200 OK from Callee never reached FS, (definitely getting lost in kamailio, maybe due to bad/missing record-route or something) Solution you say; disable proxy-media and define new extension !! Can you compare the SIP traces of successful calls now (If you like to )!? Regards, Sammy G. On Sun, May 27, 2012 at 2:30 PM, Sherif Omran wrote: > Hi Sammy > > I found the solution. Made a new extension and set proxy media to false. > Thats why the call was looping between Kamailio and FS. It now works fine > > > > > regards, > Sherif > > > > --- On *Sun, 5/27/12, SamyGo * wrote: > > > From: SamyGo > Subject: Re: [Freeswitch-users] Freeswitch (SBC + Media) does not know > that gateway phone is picked up > To: "FreeSWITCH Users Help" > Date: Sunday, May 27, 2012, 8:36 AM > > OK Sheriff, I've gone through this. This says that destination on kamailio > end is ringing and keeps on ringing state. What Im more interested in are > the sip traces on port 5060 the story thats happening in between the > destination phone and the kamailo..if u know what I mean !! > > > On Sat, May 26, 2012 at 8:06 PM, Sherif Omran > > wrote: > > Hi Sanny, > > Please see the following links > > http://pastebin.freeswitch.org/19175 > http://pastebin.freeswitch.org/19174 > > > regards, > Sherif > > > --- On *Sat, 5/26/12, SamyGo > >* wrote: > > > From: SamyGo > > > Subject: Re: [Freeswitch-users] Freeswitch (SBC + Media) does not know > that gateway phone is picked up > To: "FreeSWITCH Users Help" > > > Date: Saturday, May 26, 2012, 5:52 PM > > Hey Sherrif, > You might wanna resend the pcap file as I couldn;t find any single INVITE > or any call in that capture. Please review. > > BR > SaGo. > > On Sat, May 26, 2012 at 1:05 AM, Sherif Omran > > wrote: > > Hello Sammy > > FS uses port 6090 and registers the GW. When a call comes, it rings the > extension but then gives a busy signal and FS keeps ringing till I cancel > the call from the GW. > > > > 1. recv 477 bytes from udp/[217.74.179.29]:5060 at 19:52:05.662344: > 2. > ------------------------------------------------------------------------ > 3. CANCEL sip:gw+trunk_1000 at 78.138.90.58:6090;transport=udp;gw=trunk_1000 > SIP/2.0 > 4. Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bKe567.13b0b9e.0 > 5. Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK16d1a257;rport= > 5060 > 6. From: "+41793940965" >;tag=as1dccab06 > 7. To: > 8. Call-ID: 5cd37edb776a6b3a35e9713a453a3425 at bluesip.net > 9. CSeq: 102 CANCEL > 10. User-Agent: blueSIP PSTN GW > 11. Max-Forwards: 69 > 12. Content-Length: 0 > 13. P-hint: USRLOC > > > > > --- On *Fri, 5/25/12, SamyGo > >* wrote: > > > From: SamyGo > > > Subject: Re: [Freeswitch-users] Freeswitch (SBC + Media) does not know > that gateway phone is picked up > To: "FreeSWITCH Users Help" > > > Date: Friday, May 25, 2012, 10:46 AM > > Hi again, > If you want kamailio register to the provider then use UAC > module. > Kamailio will use the username/password and register with the provider. > > Regards, > Sammy > > > On Fri, May 25, 2012 at 12:20 PM, Sherif Omran > > wrote: > > Dear Sammy, > > Thank you for your question ... Yes, the GW is registered through FS > because I did not know how to register it to kamailio. But it seems better > to register it to kamailio. > One more information, calls from 1001 to 1002 go to kamailio then to FS > then return back to kamailio smoothly. Thus I would suggest that I change > registering the gateway from FS to kamailio. but How to? > > > > --- On *Fri, 5/25/12, SamyGo > >* wrote: > > > From: SamyGo > > > Subject: Re: [Freeswitch-users] Freeswitch (SBC + Media) does not know > that gateway phone is picked up > To: "FreeSWITCH Users Help" > > > Date: Friday, May 25, 2012, 8:08 AM > > Hi, > These are SIP traces on FreeSWITCH console, whereas you are saying and it > seems that kamailio is not detecting the answering (200 OK)of the call from > extension 1002. Please, can you take a sip trace..!! I see you've both > kamailio and FS on same server! Please take a pcap from the linux console > using the following command. > > #tcpdump -i any -s 0 -w new-call-trace.pcap -vvv > > Please be quick on turning the sip trace on and off as quickly as possible > to avoid extra packets. Once done open the file in wireshark > ; apply filter "sip || rtp" and then save the > resulting capture in separate file. Send us the new file to analyse. > > One more silly question probably, I see REGISTERs coming to your FS as > well and the calls to gateways are made from FS too !!, umm...just thinking > what are you using kamailio for!!? > > Thanks > BR, > Sammy > > > On Fri, May 25, 2012 at 4:40 AM, Sherif Omran > > wrote: > > Hi all, > > here is the siptrace: To figure it out > 1- gateway called bluesip.net. It send invide using caller > number at bluesip.net > 2- This call should go to extension kb-1002. kb means go from freeswitch > port 6090 to kamailio port 5060 > 3- It should go to call extension 1002 in Kamailio > 4- Extension 1002 rings but when I reply, it does not notice I replied > > > ./fs_cli > _____ ____ ____ _ ___ > | ___/ ___| / ___| | |_ _| > | |_ \___ \ | | | | | | > | _| ___) | | |___| |___ | | > |_| |____/ \____|_____|___| > > ******************************************************* > * Anthony Minessale II, Ken Rice, * > * Michael Jerris, Travis Cross * > * FreeSWITCH (http://www.freeswitch.org) * > * Paypal Donations Appreciated: paypal at freeswitch.org* > * Brought to you by ClueCon http://www.cluecon.com/ * > ******************************************************* > > Type /help to see a list of commands > > > +OK log level [7] > freeswitch at internal> tracelevel > -ERR tracelevel Command not found! > > freeswitch at internal> sofia global siptrace on > +OK Global siptrace on > recv 1104 bytes from udp/[217.74.179.29]:5060 at 23:08:45.822962: > ------------------------------------------------------------------------ > INVITE sip:gw+trunk_1000 at 78.138.90.58:6090;transport=udp;gw=trunk_1000 > SIP/2.0 > Record-Route: > Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 > Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 > From: "+41793940965" ;tag=as00589402 > To: > Contact: > Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net > CSeq: 102 INVITE > User-Agent: blueSIP PSTN GW > Max-Forwards: 69 > Date: Thu, 24 May 2012 23:08:44 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces > Content-Type: application/sdp > Content-Length: 367 > P-hint: USRLOC > > v=0 > o=root 20076 20076 IN IP4 217.74.179.28 > s=session > c=IN IP4 217.74.179.28 > t=0 0 > m=audio 25626 RTP/AVP 8 0 18 111 3 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:111 G726-32/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > a=sendrecv > ------------------------------------------------------------------------ > send 494 bytes to udp/[217.74.179.29]:5060 at 23:08:45.844486: > ------------------------------------------------------------------------ > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 > Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 > Record-Route: > From: "+41793940965" ;tag=as00589402 > To: > Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net > CSeq: 102 INVITE > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Content-Length: 0 > > ------------------------------------------------------------------------ > 2012-05-24 23:08:45.840749 [NOTICE] switch_channel.c:926 New Channel > sofia/internal/+41793940965 at bluesip.net[69cd22e6-a5f5-11e1-a833-7b1f73a7ffcf] > 2012-05-24 23:08:45.840749 [DEBUG] sofia.c:5532 Channel sofia/internal/ > +41793940965 at bluesip.net entering state [received][100] > 2012-05-24 23:08:45.840749 [DEBUG] sofia.c:5543 Remote SDP: > v=0 > o=root 20076 20076 IN IP4 217.74.179.28 > s=session > c=IN IP4 217.74.179.28 > t=0 0 > m=audio 25626 RTP/AVP 8 0 18 111 3 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:111 G726-32/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [PCMA:8:8000:20:64000]/[iLBC:97:8000:30:13330] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [PCMU:0:8000:20:64000]/[iLBC:97:8000:30:13330] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [G729:18:8000:20:8000]/[iLBC:97:8000:30:13330] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [G726-32:111:8000:20:0]/[iLBC:97:8000:30:13330] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [GSM:3:8000:20:13200]/[iLBC:97:8000:30:13330] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [telephone-event:101:8000:20:0]/[iLBC:97:8000:30:13330] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf > send/recv payload to 101 > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [PCMA:8:8000:20:64000]/[G7221:115:32000:20:48000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [PCMU:0:8000:20:64000]/[G7221:115:32000:20:48000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [G729:18:8000:20:8000]/[G7221:115:32000:20:48000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [G726-32:111:8000:20:0]/[G7221:115:32000:20:48000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [GSM:3:8000:20:13200]/[G7221:115:32000:20:48000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [telephone-event:101:8000:20:0]/[G7221:115:32000:20:48000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf > send/recv payload to 101 > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [PCMA:8:8000:20:64000]/[G7221:107:16000:20:32000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [PCMU:0:8000:20:64000]/[G7221:107:16000:20:32000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [G729:18:8000:20:8000]/[G7221:107:16000:20:32000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [G726-32:111:8000:20:0]/[G7221:107:16000:20:32000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [GSM:3:8000:20:13200]/[G7221:107:16000:20:32000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [telephone-event:101:8000:20:0]/[G7221:107:16000:20:32000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf > send/recv payload to 101 > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [PCMA:8:8000:20:64000]/[G722:9:8000:20:64000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [G729:18:8000:20:8000]/[G722:9:8000:20:64000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [G726-32:111:8000:20:0]/[G722:9:8000:20:64000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [GSM:3:8000:20:13200]/[G722:9:8000:20:64000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [telephone-event:101:8000:20:0]/[G722:9:8000:20:64000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf > send/recv payload to 101 > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [PCMA:8:8000:20:64000]/[PCMU:0:8000:20:64000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare > [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:2996 Set Codec > sofia/internal/+41793940965 at bluesip.net PCMU/8000 20 ms 160 samples 64000 > bits > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_codec.c:111 sofia/internal/ > +41793940965 at bluesip.net Original read codec set to PCMU:0 > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf > send/recv payload to 101 > 2012-05-24 23:08:45.840749 [DEBUG] sofia.c:5757 (sofia/internal/ > +41793940965 at bluesip.net) State Change CS_NEW -> CS_INIT > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_INIT > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:401 > (sofia/internal/+41793940965 at bluesip.net) State INIT > 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:85 sofia/internal/ > +41793940965 at bluesip.net SOFIA INIT > 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:125 (sofia/internal/ > +41793940965 at bluesip.net) State Change CS_INIT -> CS_ROUTING > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:401 > (sofia/internal/+41793940965 at bluesip.net) State INIT going to sleep > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_ROUTING > 2012-05-24 23:08:45.840749 [DEBUG] switch_channel.c:1886 (sofia/internal/ > +41793940965 at bluesip.net) Callstate Change DOWN -> RINGING > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 > (sofia/internal/+41793940965 at bluesip.net) State ROUTING > 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:148 sofia/internal/ > +41793940965 at bluesip.net SOFIA ROUTING > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:104 > sofia/internal/+41793940965 at bluesip.net Standard ROUTING > 2012-05-24 23:08:45.840749 [INFO] mod_dialplan_xml.c:485 Processing > +41793940965 <+41793940965>->kb-1002 in context public > Dialplan: sofia/internal/+41793940965 at bluesip.net parsing > [public->from_kamailio] continue=false > Dialplan: sofia/internal/+41793940965 at bluesip.net Regex (PASS) > [from_kamailio] destination_number(kb-1002) =~ /^(.+)$/ break=on-false > Dialplan: sofia/internal/+41793940965 at bluesip.net Action transfer(kb-1002 > XML default) > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:154 > (sofia/internal/+41793940965 at bluesip.net) State Change CS_ROUTING -> > CS_EXECUTE > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 > (sofia/internal/+41793940965 at bluesip.net) State ROUTING going to sleep > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_EXECUTE > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:417 > (sofia/internal/+41793940965 at bluesip.net) State EXECUTE > 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:241 sofia/internal/ > +41793940965 at bluesip.net SOFIA EXECUTE > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:192 > sofia/internal/+41793940965 at bluesip.net Standard EXECUTE > EXECUTE sofia/internal/+41793940965 at bluesip.net transfer(kb-1002 XML > default) > 2012-05-24 23:08:45.840749 [DEBUG] switch_ivr.c:1711 (sofia/internal/ > +41793940965 at bluesip.net) State Change CS_EXECUTE -> CS_ROUTING > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:731 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:08:45.840749 [NOTICE] switch_ivr.c:1717 Transfer > sofia/internal/+41793940965 at bluesip.net to XML[kb-1002 at default] > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:417 > (sofia/internal/+41793940965 at bluesip.net) State EXECUTE going to sleep > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_ROUTING > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 > (sofia/internal/+41793940965 at bluesip.net) State ROUTING > 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:148 sofia/internal/ > +41793940965 at bluesip.net SOFIA ROUTING > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:104 > sofia/internal/+41793940965 at bluesip.net Standard ROUTING > 2012-05-24 23:08:45.840749 [INFO] mod_dialplan_xml.c:485 Processing > +41793940965 <+41793940965>->kb-1002 in context default > Dialplan: sofia/internal/+41793940965 at bluesip.net parsing [default->vbox] > continue=false > Dialplan: sofia/internal/+41793940965 at bluesip.net Regex (FAIL) [vbox] > destination_number(kb-1002) =~ /^vb-([0-9][0-9][0-9])$/ break=on-false > Dialplan: sofia/internal/+41793940965 at bluesip.net parsing > [default->vmenu] continue=false > Dialplan: sofia/internal/+41793940965 at bluesip.net Regex (FAIL) [vmenu] > destination_number(kb-1002) =~ /^vm-([0-9][0-9][0-9][0-9])$/ break=on-false > Dialplan: sofia/internal/+41793940965 at bluesip.net parsing > [default->kbridge] continue=false > Dialplan: sofia/internal/+41793940965 at bluesip.net Regex (PASS) [kbridge] > destination_number(kb-1002) =~ /^kb-(.+)$/ break=on-false > Dialplan: sofia/internal/+41793940965 at bluesip.net Action > set(proxy_media=true) > Dialplan: sofia/internal/+41793940965 at bluesip.net Action > set(call_timeout=50) > Dialplan: sofia/internal/+41793940965 at bluesip.net Action > set(continue_on_fail=true) > Dialplan: sofia/internal/+41793940965 at bluesip.net Action > set(hangup_after_bridge=true) > Dialplan: sofia/internal/+41793940965 at bluesip.net Action > set(sip_invite_domain=78.138.90.58) > Dialplan: sofia/internal/+41793940965 at bluesip.net Action > export(sip_contact_user=ufs) > Dialplan: sofia/internal/+41793940965 at bluesip.net Action bridge(sofia/ > 78.138.90.58/1002 at 78.138.90.58:5060) > Dialplan: sofia/internal/+41793940965 at bluesip.net Action answer() > Dialplan: sofia/internal/+41793940965 at bluesip.net Action > voicemail(default ${domain_name} 1002) > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:154 > (sofia/internal/+41793940965 at bluesip.net) State Change CS_ROUTING -> > CS_EXECUTE > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 > (sofia/internal/+41793940965 at bluesip.net) State ROUTING going to sleep > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_EXECUTE > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:417 > (sofia/internal/+41793940965 at bluesip.net) State EXECUTE > 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:241 sofia/internal/ > +41793940965 at bluesip.net SOFIA EXECUTE > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:192 > sofia/internal/+41793940965 at bluesip.net Standard EXECUTE > EXECUTE sofia/internal/+41793940965 at bluesip.net set(proxy_media=true) > 2012-05-24 23:08:45.840749 [DEBUG] mod_dptools.c:1281 sofia/internal/ > +41793940965 at bluesip.net SET [proxy_media]=[true] > EXECUTE sofia/internal/+41793940965 at bluesip.net set(call_timeout=50) > 2012-05-24 23:08:45.840749 [DEBUG] mod_dptools.c:1281 sofia/internal/ > +41793940965 at bluesip.net SET [call_timeout]=[50] > EXECUTE sofia/internal/+41793940965 at bluesip.net set(continue_on_fail=true) > 2012-05-24 23:08:45.840749 [DEBUG] mod_dptools.c:1281 sofia/internal/ > +41793940965 at bluesip.net SET [continue_on_fail]=[true] > EXECUTE sofia/internal/+41793940965 at bluesip.netset(hangup_after_bridge=true) > 2012-05-24 23:08:45.840749 [DEBUG] mod_dptools.c:1281 sofia/internal/ > +41793940965 at bluesip.net SET [hangup_after_bridge]=[true] > EXECUTE sofia/internal/+41793940965 at bluesip.netset(sip_invite_domain=78.138.90.58) > 2012-05-24 23:08:45.840749 [DEBUG] mod_dptools.c:1281 sofia/internal/ > +41793940965 at bluesip.net SET [sip_invite_domain]=[78.138.90.58] > EXECUTE sofia/internal/+41793940965 at bluesip.netexport(sip_contact_user=ufs) > 2012-05-24 23:08:45.840749 [DEBUG] switch_channel.c:1093 EXPORT > (export_vars) [sip_contact_user]=[ufs] > EXECUTE sofia/internal/+41793940965 at bluesip.net bridge(sofia/ > 78.138.90.58/1002 at 78.138.90.58:5060) > 2012-05-24 23:08:45.840749 [DEBUG] switch_channel.c:1047 sofia/internal/ > +41793940965 at bluesip.net EXPORTING[export_vars] [sip_contact_user]=[ufs] > to event > 2012-05-24 23:08:45.840749 [DEBUG] switch_ivr_originate.c:1884 Parsing > global variables > 2012-05-24 23:08:45.840749 [NOTICE] switch_channel.c:926 New Channel > sofia/internal/1002 at 78.138.90.58:5060[69ce1f0c-a5f5-11e1-a838-7b1f73a7ffcf] > 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:4691 (sofia/internal/ > 1002 at 78.138.90.58:5060) State Change CS_NEW -> CS_INIT > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/1002 at 78.138.90.58:5060) Running State Change CS_INIT > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:401 > (sofia/internal/1002 at 78.138.90.58:5060) State INIT > 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:85 sofia/internal/ > 1002 at 78.138.90.58:5060 SOFIA INIT > 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:1871 sofia/internal/ > 1002 at 78.138.90.58:5060 Patched SDP > --- > v=0 > o=root 20076 20076 IN IP4 217.74.179.28 > s=session > c=IN IP4 217.74.179.28 > t=0 0 > m=audio 25626 RTP/AVP 8 0 18 111 3 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:111 G726-32/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > > +++ > v=0 > o=FreeSWITCH 1629510322 1629510323 IN IP4 78.138.90.58 > s=FreeSWITCH > c=IN IP4 78.138.90.58 > t=0 0 > m=audio 31178 RTP/AVP 8 0 18 111 3 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:111 G726-32/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > > send 1340 bytes to udp/[78.138.90.58]:5060 at 23:08:45.852280: > ------------------------------------------------------------------------ > INVITE sip:1002 at 78.138.90.58:5060 SIP/2.0 > Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKapDgFeegKyHXK > Max-Forwards: 67 > From: "+41793940965" ;tag=tgNrSS1jQFS8F > To: > Call-ID: 412c8312-2098-1230-8985-00163e6bb553 > CSeq: 28614398 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > 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: 372 > P-hint: USRLOC > X-FS-Support: update_display,send_info > Remote-Party-ID: "+41793940965" >;party=calling;screen=yes;privacy=off > > v=0 > o=FreeSWITCH 1629510322 1629510323 IN IP4 78.138.90.58 > s=FreeSWITCH > c=IN IP4 78.138.90.58 > t=0 0 > m=audio 31178 RTP/AVP 8 0 18 111 3 101 > a=rtpmap:8 PCMA/8000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:111 G726-32/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=silenceSupp:off - - - - > a=ptime:20 > ------------------------------------------------------------------------ > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:125 (sofia/internal/ > 1002 at 78.138.90.58:5060) State Change CS_INIT -> CS_ROUTING > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:401 > (sofia/internal/1002 at 78.138.90.58:5060) State INIT going to sleep > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/1002 at 78.138.90.58:5060) Running State Change CS_ROUTING > 2012-05-24 23:08:45.840749 [DEBUG] switch_channel.c:1886 (sofia/internal/ > 1002 at 78.138.90.58:5060) Callstate Change DOWN -> RINGING > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 > (sofia/internal/1002 at 78.138.90.58:5060) State ROUTING > 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:148 sofia/internal/ > 1002 at 78.138.90.58:5060 SOFIA ROUTING > 2012-05-24 23:08:45.840749 [DEBUG] switch_ivr_originate.c:66 > (sofia/internal/1002 at 78.138.90.58:5060) State Change CS_ROUTING -> > CS_CONSUME_MEDIA > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 > (sofia/internal/1002 at 78.138.90.58:5060) State ROUTING going to sleep > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/1002 at 78.138.90.58:5060) Running State Change > CS_CONSUME_MEDIA > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:429 > (sofia/internal/1002 at 78.138.90.58:5060) State CONSUME_MEDIA > 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:429 > (sofia/internal/1002 at 78.138.90.58:5060) State CONSUME_MEDIA going to sleep > 2012-05-24 23:08:45.840749 [DEBUG] sofia.c:5532 Channel sofia/internal/ > 1002 at 78.138.90.58:5060 entering state [calling][0] > recv 365 bytes from udp/[78.138.90.58]:5060 at 23:08:45.964309: > ------------------------------------------------------------------------ > SIP/2.0 100 trying -- your call is important to us > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;branch=z9hG4bKapDgFeegKyHXK > From: "+41793940965" ;tag=tgNrSS1jQFS8F > To: > Call-ID: 412c8312-2098-1230-8985-00163e6bb553 > CSeq: 28614398 INVITE > Server: kamailio (3.1.5 (x86_64/linux)) > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 496 bytes from udp/[78.138.90.58]:5060 at 23:08:47.414766: > ------------------------------------------------------------------------ > SIP/2.0 180 Ringing > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;branch=z9hG4bKapDgFeegKyHXK > Record-Route: > Call-ID: 412c8312-2098-1230-8985-00163e6bb553 > From: "+41793940965" ;tag=tgNrSS1jQFS8F > To: > >;tag=qCSaB2YJOAsAr2dJ-RWNMhPoDdn1RFvs > CSeq: 28614398 INVITE > Contact: "Mama" > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, REGISTER > Content-Length: 0 > > ------------------------------------------------------------------------ > 2012-05-24 23:08:47.400741 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:47.400741 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:47.400741 [DEBUG] sofia.c:5532 Channel sofia/internal/ > 1002 at 78.138.90.58:5060 entering state [proceeding][180] > 2012-05-24 23:08:47.400741 [NOTICE] sofia.c:5624 Ring-Ready sofia/internal/ > 1002 at 78.138.90.58:5060! > send 995 bytes to udp/[217.74.179.29]:5060 at 23:08:47.416538: > ------------------------------------------------------------------------ > SIP/2.0 180 Ringing > Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 > Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 > Record-Route: > From: "+41793940965" ;tag=as00589402 > To: ;tag=S7UZQygFt62Nm > Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net > CSeq: 102 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Accept: application/sdp > 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-Length: 0 > Remote-Party-ID: "Outbound Call" > >;party=calling;privacy=off;screen=no > > ------------------------------------------------------------------------ > 2012-05-24 23:08:47.400741 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:08:47.400741 [NOTICE] mod_sofia.c:2514 Ring-Ready > sofia/internal/+41793940965 at bluesip.net! > 2012-05-24 23:08:47.400741 [DEBUG] sofia.c:5532 Channel sofia/internal/ > +41793940965 at bluesip.net entering state [early][180] > 2012-05-24 23:08:47.400741 [DEBUG] switch_core_session.c:731 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:08:47.400741 [NOTICE] switch_ivr_originate.c:483 Ring Ready > sofia/internal/+41793940965 at bluesip.net! > recv 496 bytes from udp/[78.138.90.58]:5060 at 23:08:47.804967: > ------------------------------------------------------------------------ > SIP/2.0 180 Ringing > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;branch=z9hG4bKapDgFeegKyHXK > Record-Route: > From: "+41793940965" ;tag=tgNrSS1jQFS8F > To: ;tag=549D2DD03BBA7C67 > Call-ID: 412c8312-2098-1230-8985-00163e6bb553 > CSeq: 28614398 INVITE > Contact: > ;uniq=6AC0DF4D2E498C8ACA82CB22226B9> > User-Agent: AVM FRITZ!Box Fon WLAN 7170 29.04.80 (Jan 27 2010) > Content-Length: 0 > > ------------------------------------------------------------------------ > 2012-05-24 23:08:47.800746 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:47.800746 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:47.800746 [DEBUG] sofia.c:5532 Channel sofia/internal/ > 1002 at 78.138.90.58:5060 entering state [proceeding][180] > recv 496 bytes from udp/[78.138.90.58]:5060 at 23:08:48.182098: > ------------------------------------------------------------------------ > SIP/2.0 180 Ringing > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;branch=z9hG4bKapDgFeegKyHXK > Record-Route: > Call-ID: 412c8312-2098-1230-8985-00163e6bb553 > From: "+41793940965" ;tag=tgNrSS1jQFS8F > To: > >;tag=qCSaB2YJOAsAr2dJ-RWNMhPoDdn1RFvs > CSeq: 28614398 INVITE > Contact: "Mama" > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, REGISTER > Content-Length: 0 > > ------------------------------------------------------------------------ > 2012-05-24 23:08:48.180741 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:48.180741 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:48.180741 [DEBUG] sofia.c:5532 Channel sofia/internal/ > 1002 at 78.138.90.58:5060 entering state [proceeding][180] > recv 496 bytes from udp/[78.138.90.58]:5060 at 23:08:48.417403: > ------------------------------------------------------------------------ > SIP/2.0 180 Ringing > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;branch=z9hG4bKapDgFeegKyHXK > Record-Route: > From: "+41793940965" ;tag=tgNrSS1jQFS8F > To: ;tag=549D2DD03BBA7C67 > Call-ID: 412c8312-2098-1230-8985-00163e6bb553 > CSeq: 28614398 INVITE > Contact: > ;uniq=6AC0DF4D2E498C8ACA82CB22226B9> > User-Agent: AVM FRITZ!Box Fon WLAN 7170 29.04.80 (Jan 27 2010) > Content-Length: 0 > > ------------------------------------------------------------------------ > 2012-05-24 23:08:48.400744 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:48.400744 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:48.400744 [DEBUG] sofia.c:5532 Channel sofia/internal/ > 1002 at 78.138.90.58:5060 entering state [proceeding][180] > recv 496 bytes from udp/[78.138.90.58]:5060 at 23:08:49.292599: > ------------------------------------------------------------------------ > SIP/2.0 180 Ringing > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;branch=z9hG4bKapDgFeegKyHXK > Record-Route: > From: "+41793940965" ;tag=tgNrSS1jQFS8F > To: ;tag=549D2DD03BBA7C67 > Call-ID: 412c8312-2098-1230-8985-00163e6bb553 > CSeq: 28614398 INVITE > Contact: > ;uniq=6AC0DF4D2E498C8ACA82CB22226B9> > User-Agent: AVM FRITZ!Box Fon WLAN 7170 29.04.80 (Jan 27 2010) > Content-Length: 0 > > ------------------------------------------------------------------------ > 2012-05-24 23:08:49.280742 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:49.280742 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:08:49.280742 [DEBUG] sofia.c:5532 Channel sofia/internal/ > 1002 at 78.138.90.58:5060 entering state [proceeding][180] > recv 478 bytes from udp/[217.74.179.29]:5060 at 23:09:15.066012: > ------------------------------------------------------------------------ > CANCEL sip:gw+trunk_1000 at 78.138.90.58:6090;transport=udp;gw=trunk_1000 > SIP/2.0 > Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 > Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 > From: "+41793940965" ;tag=as00589402 > To: > Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net > CSeq: 102 CANCEL > User-Agent: blueSIP PSTN GW > Max-Forwards: 69 > Content-Length: 0 > P-hint: USRLOC > > ------------------------------------------------------------------------ > send 369 bytes to udp/[217.74.179.29]:5060 at 23:09:15.066176: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 > Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 > From: "+41793940965" ;tag=as00589402 > To: ;tag=S7UZQygFt62Nm > Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net > CSeq: 102 CANCEL > Content-Length: 0 > > ------------------------------------------------------------------------ > send 766 bytes to udp/[217.74.179.29]:5060 at 23:09:15.066289: > ------------------------------------------------------------------------ > SIP/2.0 487 Request Terminated > Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 > Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 > From: "+41793940965" ;tag=as00589402 > To: ;tag=S7UZQygFt62Nm > Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net > CSeq: 102 INVITE > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > 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-Length: 0 > > ------------------------------------------------------------------------ > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:877 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:09:15.060742 [DEBUG] sofia.c:5532 Channel sofia/internal/ > +41793940965 at bluesip.net entering state [terminated][487] > 2012-05-24 23:09:15.060742 [DEBUG] switch_channel.c:2848 (sofia/internal/ > +41793940965 at bluesip.net) Callstate Change RINGING -> HANGUP > 2012-05-24 23:09:15.060742 [NOTICE] sofia.c:6301 Hangup sofia/internal/ > +41793940965 at bluesip.net [CS_EXECUTE] [ORIGINATOR_CANCEL] > 2012-05-24 23:09:15.060742 [DEBUG] switch_channel.c:2871 Send signal > sofia/internal/+41793940965 at bluesip.net [KILL] > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:09:15.060742 [DEBUG] switch_channel.c:2848 (sofia/internal/ > 1002 at 78.138.90.58:5060) Callstate Change RINGING -> HANGUP > 2012-05-24 23:09:15.060742 [NOTICE] switch_ivr_originate.c:3182 Hangup > sofia/internal/1002 at 78.138.90.58:5060 [CS_CONSUME_MEDIA] > [ORIGINATOR_CANCEL] > 2012-05-24 23:09:15.060742 [DEBUG] switch_channel.c:2871 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [KILL] > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/1002 at 78.138.90.58:5060) Running State Change CS_HANGUP > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:602 > (sofia/internal/1002 at 78.138.90.58:5060) State HANGUP > 2012-05-24 23:09:15.060742 [DEBUG] mod_sofia.c:469 Channel sofia/internal/ > 1002 at 78.138.90.58:5060 hanging up, cause: ORIGINATOR_CANCEL > 2012-05-24 23:09:15.060742 [DEBUG] mod_sofia.c:523 Sending CANCEL to > sofia/internal/1002 at 78.138.90.58:5060 > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:47 > sofia/internal/1002 at 78.138.90.58:5060 Standard HANGUP, cause: > ORIGINATOR_CANCEL > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:602 > (sofia/internal/1002 at 78.138.90.58:5060) State HANGUP going to sleep > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:393 > (sofia/internal/1002 at 78.138.90.58:5060) State Change CS_HANGUP -> > CS_REPORTING > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/1002 at 78.138.90.58:5060) Running State Change CS_REPORTING > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:662 > (sofia/internal/1002 at 78.138.90.58:5060) State REPORTING > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:79 > sofia/internal/1002 at 78.138.90.58:5060 Standard REPORTING, cause: > ORIGINATOR_CANCEL > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:662 > (sofia/internal/1002 at 78.138.90.58:5060) State REPORTING going to sleep > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:387 > (sofia/internal/1002 at 78.138.90.58:5060) State Change CS_REPORTING -> > CS_DESTROY > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1382 Session 9 > (sofia/internal/1002 at 78.138.90.58:5060) Locked, Waiting on external > entities > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/1002 at 78.138.90.58:5060 [BREAK] > 2012-05-24 23:09:15.060742 [DEBUG] switch_ivr_originate.c:3358 Originate > Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL] > 2012-05-24 23:09:15.060742 [NOTICE] switch_core_session.c:1400 Session 9 > (sofia/internal/1002 at 78.138.90.58:5060) Ended > 2012-05-24 23:09:15.060742 [NOTICE] switch_core_session.c:1402 Close > Channel sofia/internal/1002 at 78.138.90.58:5060 [CS_DESTROY] > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:491 > (sofia/internal/1002 at 78.138.90.58:5060) Callstate Change HANGUP -> DOWN > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:494 > (sofia/internal/1002 at 78.138.90.58:5060) Running State Change CS_DESTROY > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:504 > (sofia/internal/1002 at 78.138.90.58:5060) State DESTROY > 2012-05-24 23:09:15.060742 [DEBUG] mod_sofia.c:374 sofia/internal/ > 1002 at 78.138.90.58:5060 SOFIA DESTROY > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:86 > sofia/internal/1002 at 78.138.90.58:5060 Standard DESTROY > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:504 > (sofia/internal/1002 at 78.138.90.58:5060) State DESTROY going to sleep > 2012-05-24 23:09:15.060742 [INFO] mod_dptools.c:2922 Originate Failed. > Cause: ORIGINATOR_CANCEL > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:2287 > sofia/internal/+41793940965 at bluesip.net skip receive message > [APPLICATION_EXEC_COMPLETE] (channel is hungup already) > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:417 > (sofia/internal/+41793940965 at bluesip.net) State EXECUTE going to sleep > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_HANGUP > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:602 > (sofia/internal/+41793940965 at bluesip.net) State HANGUP > 2012-05-24 23:09:15.060742 [DEBUG] mod_sofia.c:469 Channel sofia/internal/ > +41793940965 at bluesip.net hanging up, cause: ORIGINATOR_CANCEL > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:47 > sofia/internal/+41793940965 at bluesip.net Standard HANGUP, cause: > ORIGINATOR_CANCEL > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:602 > (sofia/internal/+41793940965 at bluesip.net) State HANGUP going to sleep > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:393 > (sofia/internal/+41793940965 at bluesip.net) State Change CS_HANGUP -> > CS_REPORTING > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:362 > (sofia/internal/+41793940965 at bluesip.net) Running State Change > CS_REPORTING > 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:662 > (sofia/internal/+41793940965 at bluesip.net) State REPORTING > send 383 bytes to udp/[78.138.90.58]:5060 at 23:09:15.087695: > ------------------------------------------------------------------------ > CANCEL sip:1002 at 78.138.90.58:5060 SIP/2.0 > Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKapDgFeegKyHXK > Max-Forwards: 67 > From: "+41793940965" ;tag=tgNrSS1jQFS8F > To: > Call-ID: 412c8312-2098-1230-8985-00163e6bb553 > CSeq: 28614398 CANCEL > Reason: FreeSWITCH;cause=487;text="ORIGINATOR_CANCEL" > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 408 bytes from udp/[217.74.179.29]:5060 at 23:09:15.094670: > ------------------------------------------------------------------------ > ACK sip:gw+trunk_1000 at 78.138.90.58:6090;transport=udp;gw=trunk_1000 > SIP/2.0 > Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 > From: "+41793940965" ;tag=as00589402 > Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net > To: ;tag=S7UZQygFt62Nm > CSeq: 102 ACK > User-Agent: Sip EXpress router(0.9.7 (i386/linux)) > Content-Length: 0 > > ------------------------------------------------------------------------ > 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:79 > sofia/internal/+41793940965 at bluesip.net Standard REPORTING, cause: > ORIGINATOR_CANCEL > 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:662 > (sofia/internal/+41793940965 at bluesip.net) State REPORTING going to sleep > 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:387 > (sofia/internal/+41793940965 at bluesip.net) State Change CS_REPORTING -> > CS_DESTROY > 2012-05-24 23:09:15.120807 [DEBUG] switch_core_session.c:1182 Send signal > sofia/internal/+41793940965 at bluesip.net [BREAK] > 2012-05-24 23:09:15.120807 [DEBUG] switch_core_session.c:1382 Session 8 > (sofia/internal/+41793940965 at bluesip.net) Locked, Waiting on external > entities > 2012-05-24 23:09:15.120807 [NOTICE] switch_core_session.c:1400 Session 8 > (sofia/internal/+41793940965 at bluesip.net) Ended > 2012-05-24 23:09:15.120807 [NOTICE] switch_core_session.c:1402 Close > Channel sofia/internal/+41793940965 at bluesip.net [CS_DESTROY] > 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:491 > (sofia/internal/+41793940965 at bluesip.net) Callstate Change HANGUP -> DOWN > 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:494 > (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_DESTROY > 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:504 > (sofia/internal/+41793940965 at bluesip.net) State DESTROY > 2012-05-24 23:09:15.120807 [DEBUG] mod_sofia.c:374 sofia/internal/ > +41793940965 at bluesip.net SOFIA DESTROY > 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:86 > sofia/internal/+41793940965 at bluesip.net Standard DESTROY > 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:504 > (sofia/internal/+41793940965 at bluesip.net) State DESTROY going to sleep > recv 378 bytes from udp/[78.138.90.58]:5060 at 23:09:15.135235: > ------------------------------------------------------------------------ > SIP/2.0 200 canceling > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;branch=z9hG4bKapDgFeegKyHXK > From: "+41793940965" ;tag=tgNrSS1jQFS8F > To: >;tag=a6a1c5f60faecf035a1ae5b6e96e979a-8c5b > Call-ID: 412c8312-2098-1230-8985-00163e6bb553 > CSeq: 28614398 CANCEL > Server: kamailio (3.1.5 (x86_64/linux)) > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 446 bytes from udp/[78.138.90.58]:5060 at 23:09:16.419308: > ------------------------------------------------------------------------ > SIP/2.0 487 Request Terminated > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;branch=z9hG4bKapDgFeegKyHXK > Record-Route: > Call-ID: 412c8312-2098-1230-8985-00163e6bb553 > From: "+41793940965" ;tag=tgNrSS1jQFS8F > To: > >;tag=qCSaB2YJOAsAr2dJ-RWNMhPoDdn1RFvs > CSeq: 28614398 INVITE > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, REGISTER > Content-Length: 0 > > ------------------------------------------------------------------------ > send 354 bytes to udp/[78.138.90.58]:5060 at 23:09:16.419543: > ------------------------------------------------------------------------ > ACK sip:1002 at 78.138.90.58:5060 SIP/2.0 > Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKapDgFeegKyHXK > Max-Forwards: 67 > From: "+41793940965" ;tag=tgNrSS1jQFS8F > To: > >;tag=qCSaB2YJOAsAr2dJ-RWNMhPoDdn1RFvs > Call-ID: 412c8312-2098-1230-8985-00163e6bb553 > CSeq: 28614398 ACK > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 564 bytes from udp/[41.34.123.243]:58881 at 23:13:13.612418: > ------------------------------------------------------------------------ > REGISTER sip:sip.pcfone.com:6090 SIP/2.0 > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport;branch=z9hG4bKPjHQIpv.ACHGbXD4IHrMr72LwZ8kYdlu-J > Max-Forwards: 70 > From: "Sherif 1003" > >;tag=wOP.FVT4Crk8O5E.bB7BWUhNO6QG91IG > To: "Sherif 1003" > > > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60926 REGISTER > User-Agent: Telephone 1.0.2 > Contact: "Sherif 1003" > Expires: 300 > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, > REFER, MESSAGE, OPTIONS > Content-Length: 0 > > ------------------------------------------------------------------------ > send 744 bytes to udp/[41.34.123.243]:58881 at 23:13:13.613152: > ------------------------------------------------------------------------ > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport=58881;branch=z9hG4bKPjHQIpv.ACHGbXD4IHrMr72LwZ8kYdlu-J;received=41.34.123.243 > From: "Sherif 1003" > >;tag=wOP.FVT4Crk8O5E.bB7BWUhNO6QG91IG > To: "Sherif 1003" > >;tag=USeHUmjpmrFUB > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60926 REGISTER > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > WWW-Authenticate: Digest realm="sip.pcfone.com", > nonce="09674764-a5f6-11e1-a83c-7b1f73a7ffcf", algorithm=MD5, qop="auth" > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 830 bytes from udp/[41.34.123.243]:58881 at 23:13:13.847746: > ------------------------------------------------------------------------ > REGISTER sip:sip.pcfone.com:6090 SIP/2.0 > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport;branch=z9hG4bKPjFr7rIAdAJPRQd2GmrMrVib3KT3Rup0q4 > Max-Forwards: 70 > From: "Sherif 1003" > >;tag=wOP.FVT4Crk8O5E.bB7BWUhNO6QG91IG > To: "Sherif 1003" > > > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60927 REGISTER > User-Agent: Telephone 1.0.2 > Contact: "Sherif 1003" > Expires: 300 > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, > REFER, MESSAGE, OPTIONS > Authorization: Digest username="1002", realm="sip.pcfone.com", > nonce="09674764-a5f6-11e1-a83c-7b1f73a7ffcf", uri="sip:sip.pcfone.com:6090", > response="3bc0bb82a803ded3641704c4eb78d58f", algorithm=MD5, > cnonce="sc3AJfhuev18Bs-L-6tORmI.RXlxnPOz", qop=auth, nc=00000001 > Content-Length: 0 > > ------------------------------------------------------------------------ > send 706 bytes to udp/[41.34.123.243]:58881 at 23:13:13.850197: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport=58881;branch=z9hG4bKPjFr7rIAdAJPRQd2GmrMrVib3KT3Rup0q4;received=41.34.123.243 > From: "Sherif 1003" > >;tag=wOP.FVT4Crk8O5E.bB7BWUhNO6QG91IG > To: "Sherif 1003" > >;tag=v279vF3SH15DQ > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60927 REGISTER > Contact: ;expires=300 > Date: Thu, 24 May 2012 23:13:13 GMT > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Content-Length: 0 > > ------------------------------------------------------------------------ > send 989 bytes to udp/[41.34.123.243]:58881 at 23:13:13.888717: > ------------------------------------------------------------------------ > NOTIFY sip:1002 at 192.168.178.26:58881;ob SIP/2.0 > Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKBZ68g9yKg77FF > Route: ;ob > Max-Forwards: 70 > From: > >;tag=XB12yamXeav0j > To: > > > Call-ID: e0efa252-2098-1230-8985-00163e6bb553 > CSeq: 28614532 NOTIFY > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > 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=noresource > Content-Type: application/simple-message-summary > Content-Length: 91 > > Messages-Waiting: yes > Message-Account: sip:1002 at 78.138.90.58 > Voice-Message: 1/0 (0/0) > > ------------------------------------------------------------------------ > recv 308 bytes from udp/[41.34.123.243]:58881 at 23:13:14.298850: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;received=78.138.90.58;branch=z9hG4bKBZ68g9yKg77FF > Call-ID: e0efa252-2098-1230-8985-00163e6bb553 > From: > >;tag=XB12yamXeav0j > To: > >;tag=z9hG4bKBZ68g9yKg77FF > CSeq: 28614532 NOTIFY > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 564 bytes from udp/[41.34.123.243]:58881 at 23:18:09.067232: > ------------------------------------------------------------------------ > REGISTER sip:sip.pcfone.com:6090 SIP/2.0 > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport;branch=z9hG4bKPjABTslmZIhiHYRtLqQn2erwmX19MVYHFC > Max-Forwards: 70 > From: "Sherif 1003" > >;tag=JBSXL5WZLWbLzpvRFoTSP5x8qQslI1ST > To: "Sherif 1003" > > > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60928 REGISTER > User-Agent: Telephone 1.0.2 > Contact: "Sherif 1003" > Expires: 300 > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, > REFER, MESSAGE, OPTIONS > Content-Length: 0 > > ------------------------------------------------------------------------ > send 744 bytes to udp/[41.34.123.243]:58881 at 23:18:09.068121: > ------------------------------------------------------------------------ > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport=58881;branch=z9hG4bKPjABTslmZIhiHYRtLqQn2erwmX19MVYHFC;received=41.34.123.243 > From: "Sherif 1003" > >;tag=JBSXL5WZLWbLzpvRFoTSP5x8qQslI1ST > To: "Sherif 1003" > >;tag=ymtU0540BKjKe > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60928 REGISTER > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > WWW-Authenticate: Digest realm="sip.pcfone.com", > nonce="b9821df4-a5f6-11e1-a83d-7b1f73a7ffcf", algorithm=MD5, qop="auth" > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 830 bytes from udp/[41.34.123.243]:58881 at 23:18:09.302944: > ------------------------------------------------------------------------ > REGISTER sip:sip.pcfone.com:6090 SIP/2.0 > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport;branch=z9hG4bKPjRgtztcy63vMT97Jp5pH9n311li5EFeY- > Max-Forwards: 70 > From: "Sherif 1003" > >;tag=JBSXL5WZLWbLzpvRFoTSP5x8qQslI1ST > To: "Sherif 1003" > > > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60929 REGISTER > User-Agent: Telephone 1.0.2 > Contact: "Sherif 1003" > Expires: 300 > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, > REFER, MESSAGE, OPTIONS > Authorization: Digest username="1002", realm="sip.pcfone.com", > nonce="b9821df4-a5f6-11e1-a83d-7b1f73a7ffcf", uri="sip:sip.pcfone.com:6090", > response="519c6301816b51f3bbb98d97a347e92b", algorithm=MD5, > cnonce="sc3AJfhuev18Bs-L-6tORmI.RXlxnPOz", qop=auth, nc=00000001 > Content-Length: 0 > > ------------------------------------------------------------------------ > send 706 bytes to udp/[41.34.123.243]:58881 at 23:18:09.305217: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport=58881;branch=z9hG4bKPjRgtztcy63vMT97Jp5pH9n311li5EFeY-;received=41.34.123.243 > From: "Sherif 1003" > >;tag=JBSXL5WZLWbLzpvRFoTSP5x8qQslI1ST > To: "Sherif 1003" > >;tag=ZXKm20N48U85S > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60929 REGISTER > Contact: ;expires=300 > Date: Thu, 24 May 2012 23:18:09 GMT > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Content-Length: 0 > > ------------------------------------------------------------------------ > send 989 bytes to udp/[41.34.123.243]:58881 at 23:18:09.379987: > ------------------------------------------------------------------------ > NOTIFY sip:1002 at 192.168.178.26:58881;ob SIP/2.0 > Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKc8Z1j4FQDgy2a > Route: ;ob > Max-Forwards: 70 > From: > >;tag=06cD4U6754yrN > To: > > > Call-ID: 91100602-2099-1230-8985-00163e6bb553 > CSeq: 28614680 NOTIFY > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > 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=noresource > Content-Type: application/simple-message-summary > Content-Length: 91 > > Messages-Waiting: yes > Message-Account: sip:1002 at 78.138.90.58 > Voice-Message: 1/0 (0/0) > > ------------------------------------------------------------------------ > recv 308 bytes from udp/[41.34.123.243]:58881 at 23:18:09.773629: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;received=78.138.90.58;branch=z9hG4bKc8Z1j4FQDgy2a > Call-ID: 91100602-2099-1230-8985-00163e6bb553 > From: > >;tag=06cD4U6754yrN > To: > >;tag=z9hG4bKc8Z1j4FQDgy2a > CSeq: 28614680 NOTIFY > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 564 bytes from udp/[41.34.123.243]:58881 at 23:23:04.522455: > ------------------------------------------------------------------------ > REGISTER sip:sip.pcfone.com:6090 SIP/2.0 > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport;branch=z9hG4bKPjaCuKyaBi6ZqcsHyp5HhH3WthJStom0SK > Max-Forwards: 70 > From: "Sherif 1003" > >;tag=qfUt1fQWuE-Ro3zWld6X0r6WidtD6B1q > To: "Sherif 1003" > > > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60930 REGISTER > User-Agent: Telephone 1.0.2 > Contact: "Sherif 1003" > Expires: 300 > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, > REFER, MESSAGE, OPTIONS > Content-Length: 0 > > ------------------------------------------------------------------------ > send 744 bytes to udp/[41.34.123.243]:58881 at 23:23:04.523158: > ------------------------------------------------------------------------ > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport=58881;branch=z9hG4bKPjaCuKyaBi6ZqcsHyp5HhH3WthJStom0SK;received=41.34.123.243 > From: "Sherif 1003" > >;tag=qfUt1fQWuE-Ro3zWld6X0r6WidtD6B1q > To: "Sherif 1003" > >;tag=1F655pQB3DNBH > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60930 REGISTER > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > WWW-Authenticate: Digest realm="sip.pcfone.com", > nonce="699cfe2a-a5f7-11e1-a83e-7b1f73a7ffcf", algorithm=MD5, qop="auth" > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 830 bytes from udp/[41.34.123.243]:58881 at 23:23:04.757798: > ------------------------------------------------------------------------ > REGISTER sip:sip.pcfone.com:6090 SIP/2.0 > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport;branch=z9hG4bKPjs8SAufeoyzCLrePaAbYhZuOqQF47XIc- > Max-Forwards: 70 > From: "Sherif 1003" > >;tag=qfUt1fQWuE-Ro3zWld6X0r6WidtD6B1q > To: "Sherif 1003" > > > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60931 REGISTER > User-Agent: Telephone 1.0.2 > Contact: "Sherif 1003" > Expires: 300 > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, > REFER, MESSAGE, OPTIONS > Authorization: Digest username="1002", realm="sip.pcfone.com", > nonce="699cfe2a-a5f7-11e1-a83e-7b1f73a7ffcf", uri="sip:sip.pcfone.com:6090", > response="23a886934aaf049f08432b6bb226bd37", algorithm=MD5, > cnonce="sc3AJfhuev18Bs-L-6tORmI.RXlxnPOz", qop=auth, nc=00000001 > Content-Length: 0 > > ------------------------------------------------------------------------ > send 706 bytes to udp/[41.34.123.243]:58881 at 23:23:04.759924: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport=58881;branch=z9hG4bKPjs8SAufeoyzCLrePaAbYhZuOqQF47XIc-;received=41.34.123.243 > From: "Sherif 1003" > >;tag=qfUt1fQWuE-Ro3zWld6X0r6WidtD6B1q > To: "Sherif 1003" > >;tag=2rZy7H8e0pByc > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60931 REGISTER > Contact: ;expires=300 > Date: Thu, 24 May 2012 23:23:04 GMT > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Content-Length: 0 > > ------------------------------------------------------------------------ > send 989 bytes to udp/[41.34.123.243]:58881 at 23:23:04.872728: > ------------------------------------------------------------------------ > NOTIFY sip:1002 at 192.168.178.26:58881;ob SIP/2.0 > Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKDHStmZ0taSmNp > Route: ;ob > Max-Forwards: 70 > From: > >;tag=31rQ9cSjXZ1gr > To: > > > Call-ID: 4130a1ce-209a-1230-8985-00163e6bb553 > CSeq: 28614828 NOTIFY > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > 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=noresource > Content-Type: application/simple-message-summary > Content-Length: 91 > > Messages-Waiting: yes > Message-Account: sip:1002 at 78.138.90.58 > Voice-Message: 1/0 (0/0) > > ------------------------------------------------------------------------ > recv 308 bytes from udp/[41.34.123.243]:58881 at 23:23:05.269063: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;received=78.138.90.58;branch=z9hG4bKDHStmZ0taSmNp > Call-ID: 4130a1ce-209a-1230-8985-00163e6bb553 > From: > >;tag=31rQ9cSjXZ1gr > To: > >;tag=z9hG4bKDHStmZ0taSmNp > CSeq: 28614828 NOTIFY > Content-Length: 0 > > ------------------------------------------------------------------------ > send 916 bytes to udp/[217.74.179.29]:5060 at 23:25:45.873423: > ------------------------------------------------------------------------ > REGISTER sip:bluesip.net;transport=udp SIP/2.0 > Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKetjKptHy71a8H > Max-Forwards: 70 > From: > ;transport=udp>;tag=6r0vBQZS650Fg > To: > ;transport=udp> > Call-ID: 2a3e4696-a5b9-11e1-a7ed-7b1f73a7ffcf > CSeq: 28601493 REGISTER > Contact: ;transport=udp;gw=trunk_1000> > Expires: 3600 > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Authorization: Digest username="bluesip/sherifomran", realm=" > bluesip.net", nonce="4fbeba4ebfb7414d5c701bd6f994402d551f3acf", > algorithm=MD5, uri="sip:bluesip.net;transport=udp", > response="c6cdcafe0418e519bc9ee0d8fa3d4d74" > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 722 bytes from udp/[217.74.179.29]:5060 at 23:25:45.904176: > ------------------------------------------------------------------------ > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;branch=z9hG4bKetjKptHy71a8H > From: > ;transport=udp>;tag=6r0vBQZS650Fg > To: > ;transport=udp>;tag=0354a2e1b960c9cc2279eca4e5f84e20.1b16 > Call-ID: 2a3e4696-a5b9-11e1-a7ed-7b1f73a7ffcf > CSeq: 28601493 REGISTER > WWW-Authenticate: Digest realm="bluesip.net", > nonce="4fbec4a4ba82d1f1feaa32ccec01b85be054fcf7", stale=true > Server: Sip EXpress router (0.9.7 (i386/linux)) > Content-Length: 0 > Warning: 392 217.74.179.29:5060 "Noisy feedback tells: pid=5455 > req_src_ip=78.138.90.58 req_src_port=6090 in_uri=sip:bluesip.net;transport=udp > out_uri=sip:bluesip.net;transport=udp via_cnt==1" > > ------------------------------------------------------------------------ > send 916 bytes to udp/[217.74.179.29]:5060 at 23:25:45.904416: > ------------------------------------------------------------------------ > REGISTER sip:bluesip.net;transport=udp SIP/2.0 > Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKF3BcrN214a1tD > Max-Forwards: 70 > From: > ;transport=udp>;tag=6r0vBQZS650Fg > To: > ;transport=udp> > Call-ID: 2a3e4696-a5b9-11e1-a7ed-7b1f73a7ffcf > CSeq: 28601494 REGISTER > Contact: ;transport=udp;gw=trunk_1000> > Expires: 3600 > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Authorization: Digest username="bluesip/sherifomran", realm=" > bluesip.net", nonce="4fbec4a4ba82d1f1feaa32ccec01b85be054fcf7", > algorithm=MD5, uri="sip:bluesip.net;transport=udp", > response="4c09dbe4b9accb52d4104b40dfe20040" > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 699 bytes from udp/[217.74.179.29]:5060 at 23:25:45.935046: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;branch=z9hG4bKF3BcrN214a1tD > From: > ;transport=udp>;tag=6r0vBQZS650Fg > To: > ;transport=udp>;tag=0354a2e1b960c9cc2279eca4e5f84e20.2017 > Call-ID: 2a3e4696-a5b9-11e1-a7ed-7b1f73a7ffcf > CSeq: 28601494 REGISTER > Contact: ;transport=udp;gw=trunk_1000>;q=0.5;expires=3600 > Server: Sip EXpress router (0.9.7 (i386/linux)) > Content-Length: 0 > Warning: 392 217.74.179.29:5060 "Noisy feedback tells: pid=5462 > req_src_ip=78.138.90.58 req_src_port=6090 in_uri=sip:bluesip.net;transport=udp > out_uri=sip:bluesip.net;transport=udp via_cnt==1" > > ------------------------------------------------------------------------ > recv 564 bytes from udp/[41.34.123.243]:58881 at 23:27:59.977057: > ------------------------------------------------------------------------ > REGISTER sip:sip.pcfone.com:6090 SIP/2.0 > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport;branch=z9hG4bKPjEOP.a4NoXqoFqC7a.k2tyeX-mYlUyH4H > Max-Forwards: 70 > From: "Sherif 1003" > >;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K > To: "Sherif 1003" > > > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60932 REGISTER > User-Agent: Telephone 1.0.2 > Contact: "Sherif 1003" > Expires: 300 > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, > REFER, MESSAGE, OPTIONS > Content-Length: 0 > > ------------------------------------------------------------------------ > send 744 bytes to udp/[41.34.123.243]:58881 at 23:27:59.977960: > ------------------------------------------------------------------------ > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport=58881;branch=z9hG4bKPjEOP.a4NoXqoFqC7a.k2tyeX-mYlUyH4H;received=41.34.123.243 > From: "Sherif 1003" > >;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K > To: "Sherif 1003" > >;tag=4ajgB89Nt8Q3K > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60932 REGISTER > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > WWW-Authenticate: Digest realm="sip.pcfone.com", > nonce="19b7cc40-a5f8-11e1-a83f-7b1f73a7ffcf", algorithm=MD5, qop="auth" > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 564 bytes from udp/[41.34.123.243]:58881 at 23:28:00.468487: > ------------------------------------------------------------------------ > REGISTER sip:sip.pcfone.com:6090 SIP/2.0 > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport;branch=z9hG4bKPjEOP.a4NoXqoFqC7a.k2tyeX-mYlUyH4H > Max-Forwards: 70 > From: "Sherif 1003" > >;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K > To: "Sherif 1003" > > > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60932 REGISTER > User-Agent: Telephone 1.0.2 > Contact: "Sherif 1003" > Expires: 300 > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, > REFER, MESSAGE, OPTIONS > Content-Length: 0 > > ------------------------------------------------------------------------ > send 744 bytes to udp/[41.34.123.243]:58881 at 23:28:00.468679: > ------------------------------------------------------------------------ > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport=58881;branch=z9hG4bKPjEOP.a4NoXqoFqC7a.k2tyeX-mYlUyH4H;received=41.34.123.243 > From: "Sherif 1003" > >;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K > To: "Sherif 1003" > >;tag=4ajgB89Nt8Q3K > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60932 REGISTER > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > WWW-Authenticate: Digest realm="sip.pcfone.com", > nonce="19b7cc40-a5f8-11e1-a83f-7b1f73a7ffcf", algorithm=MD5, qop="auth" > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 830 bytes from udp/[41.34.123.243]:58881 at 23:28:00.717446: > ------------------------------------------------------------------------ > REGISTER sip:sip.pcfone.com:6090 SIP/2.0 > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport;branch=z9hG4bKPj0iHd3-XUjnvcBs9DiJUCWzrmYTmWxFLt > Max-Forwards: 70 > From: "Sherif 1003" > >;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K > To: "Sherif 1003" > > > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60933 REGISTER > User-Agent: Telephone 1.0.2 > Contact: "Sherif 1003" > Expires: 300 > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, > REFER, MESSAGE, OPTIONS > Authorization: Digest username="1002", realm="sip.pcfone.com", > nonce="19b7cc40-a5f8-11e1-a83f-7b1f73a7ffcf", uri="sip:sip.pcfone.com:6090", > response="e21c903e6d98b5adcdd2eefa75217157", algorithm=MD5, > cnonce="sc3AJfhuev18Bs-L-6tORmI.RXlxnPOz", qop=auth, nc=00000001 > Content-Length: 0 > > ------------------------------------------------------------------------ > send 706 bytes to udp/[41.34.123.243]:58881 at 23:28:00.719947: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport=58881;branch=z9hG4bKPj0iHd3-XUjnvcBs9DiJUCWzrmYTmWxFLt;received=41.34.123.243 > From: "Sherif 1003" > >;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K > To: "Sherif 1003" > >;tag=5KB9c3tSQHepF > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60933 REGISTER > Contact: ;expires=300 > Date: Thu, 24 May 2012 23:28:00 GMT > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Content-Length: 0 > > ------------------------------------------------------------------------ > send 989 bytes to udp/[41.34.123.243]:58881 at 23:28:00.868368: > ------------------------------------------------------------------------ > NOTIFY sip:1002 at 192.168.178.26:58881;ob SIP/2.0 > Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKgc54SgK51KQDS > Route: ;ob > Max-Forwards: 70 > From: > >;tag=6v41eyBXmt48a > To: > > > Call-ID: f19dfbce-209a-1230-8985-00163e6bb553 > CSeq: 28614976 NOTIFY > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > 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=noresource > Content-Type: application/simple-message-summary > Content-Length: 91 > > Messages-Waiting: yes > Message-Account: sip:1002 at 78.138.90.58 > Voice-Message: 1/0 (0/0) > > ------------------------------------------------------------------------ > recv 308 bytes from udp/[41.34.123.243]:58881 at 23:28:01.128279: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;received=78.138.90.58;branch=z9hG4bKgc54SgK51KQDS > Call-ID: f19dfbce-209a-1230-8985-00163e6bb553 > From: > >;tag=6v41eyBXmt48a > To: > >;tag=z9hG4bKgc54SgK51KQDS > CSeq: 28614976 NOTIFY > Content-Length: 0 > > ------------------------------------------------------------------------ > send 913 bytes to udp/[217.74.179.29]:5060 at 23:28:11.885213: > ------------------------------------------------------------------------ > REGISTER sip:bluesip.net;transport=udp SIP/2.0 > Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKHNyXUB48yvD0m > Max-Forwards: 70 > From: > ;transport=udp>;tag=5F739Uep9vaXm > To: > ;transport=udp> > Call-ID: 2a3e49c0-a5b9-11e1-a7ee-7b1f73a7ffcf > CSeq: 28601493 REGISTER > Contact: ;transport=udp;gw=trunk_1002> > Expires: 3600 > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Authorization: Digest username="bluesip/salahomran", realm="bluesip.net", > nonce="4fbebeca79a5cff7c417463814cb9054d313bcac", algorithm=MD5, uri="sip: > bluesip.net;transport=udp", response="bb1babb90f4ea1dc8fbb9aa45d6038c7" > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 720 bytes from udp/[217.74.179.29]:5060 at 23:28:11.915764: > ------------------------------------------------------------------------ > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;branch=z9hG4bKHNyXUB48yvD0m > From: > ;transport=udp>;tag=5F739Uep9vaXm > To: > ;transport=udp>;tag=0354a2e1b960c9cc2279eca4e5f84e20.3454 > Call-ID: 2a3e49c0-a5b9-11e1-a7ee-7b1f73a7ffcf > CSeq: 28601493 REGISTER > WWW-Authenticate: Digest realm="bluesip.net", > nonce="4fbec5365af3a174f1cc0b8cda770c727d25bf71", stale=true > Server: Sip EXpress router (0.9.7 (i386/linux)) > Content-Length: 0 > Warning: 392 217.74.179.29:5060 "Noisy feedback tells: pid=5462 > req_src_ip=78.138.90.58 req_src_port=6090 in_uri=sip:bluesip.net;transport=udp > out_uri=sip:bluesip.net;transport=udp via_cnt==1" > > ------------------------------------------------------------------------ > send 913 bytes to udp/[217.74.179.29]:5060 at 23:28:11.915914: > ------------------------------------------------------------------------ > REGISTER sip:bluesip.net;transport=udp SIP/2.0 > Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKjyQpX6mcv53jg > Max-Forwards: 70 > From: > ;transport=udp>;tag=5F739Uep9vaXm > To: > ;transport=udp> > Call-ID: 2a3e49c0-a5b9-11e1-a7ee-7b1f73a7ffcf > CSeq: 28601494 REGISTER > Contact: ;transport=udp;gw=trunk_1002> > Expires: 3600 > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Authorization: Digest username="bluesip/salahomran", realm="bluesip.net", > nonce="4fbec5365af3a174f1cc0b8cda770c727d25bf71", algorithm=MD5, uri="sip: > bluesip.net;transport=udp", response="e260f6dc9b01ce8bea8c5ffb36bd4ddd" > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 697 bytes from udp/[217.74.179.29]:5060 at 23:28:11.946325: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;branch=z9hG4bKjyQpX6mcv53jg > From: > ;transport=udp>;tag=5F739Uep9vaXm > To: > ;transport=udp>;tag=0354a2e1b960c9cc2279eca4e5f84e20.4b5c > Call-ID: 2a3e49c0-a5b9-11e1-a7ee-7b1f73a7ffcf > CSeq: 28601494 REGISTER > Contact: ;transport=udp;gw=trunk_1002>;q=0.5;expires=3600 > Server: Sip EXpress router (0.9.7 (i386/linux)) > Content-Length: 0 > Warning: 392 217.74.179.29:5060 "Noisy feedback tells: pid=5455 > req_src_ip=78.138.90.58 req_src_port=6090 in_uri=sip:bluesip.net;transport=udp > out_uri=sip:bluesip.net;transport=udp via_cnt==1" > > ------------------------------------------------------------------------ > recv 564 bytes from udp/[41.34.123.243]:58881 at 23:32:55.930149: > ------------------------------------------------------------------------ > REGISTER sip:sip.pcfone.com:6090 SIP/2.0 > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport;branch=z9hG4bKPjuk8DZGiaXfN7BmtslNHVjtSg5UXsbHXc > Max-Forwards: 70 > From: "Sherif 1003" > >;tag=Pz.t-0tGXj3saM.ehBiB1EAMmmitze1D > To: "Sherif 1003" > > > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60934 REGISTER > User-Agent: Telephone 1.0.2 > Contact: "Sherif 1003" > Expires: 300 > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, > REFER, MESSAGE, OPTIONS > Content-Length: 0 > > ------------------------------------------------------------------------ > send 744 bytes to udp/[41.34.123.243]:58881 at 23:32:55.931324: > ------------------------------------------------------------------------ > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport=58881;branch=z9hG4bKPjuk8DZGiaXfN7BmtslNHVjtSg5UXsbHXc;received=41.34.123.243 > From: "Sherif 1003" > >;tag=Pz.t-0tGXj3saM.ehBiB1EAMmmitze1D > To: "Sherif 1003" > >;tag=75XtgSv0H3tUp > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60934 REGISTER > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > WWW-Authenticate: Digest realm="sip.pcfone.com", > nonce="ca1eabee-a5f8-11e1-a840-7b1f73a7ffcf", algorithm=MD5, qop="auth" > Content-Length: 0 > > ------------------------------------------------------------------------ > recv 830 bytes from udp/[41.34.123.243]:58881 at 23:32:56.165624: > ------------------------------------------------------------------------ > REGISTER sip:sip.pcfone.com:6090 SIP/2.0 > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport;branch=z9hG4bKPjKEQ6xXmyzZxA6vCM88vHNYAOPAjUqFsD > Max-Forwards: 70 > From: "Sherif 1003" > >;tag=Pz.t-0tGXj3saM.ehBiB1EAMmmitze1D > To: "Sherif 1003" > > > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60935 REGISTER > User-Agent: Telephone 1.0.2 > Contact: "Sherif 1003" > Expires: 300 > Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, > REFER, MESSAGE, OPTIONS > Authorization: Digest username="1002", realm="sip.pcfone.com", > nonce="ca1eabee-a5f8-11e1-a840-7b1f73a7ffcf", uri="sip:sip.pcfone.com:6090", > response="ff6d12f4ec7abd4b37187222b3af2358", algorithm=MD5, > cnonce="sc3AJfhuev18Bs-L-6tORmI.RXlxnPOz", qop=auth, nc=00000001 > Content-Length: 0 > > ------------------------------------------------------------------------ > send 706 bytes to udp/[41.34.123.243]:58881 at 23:32:56.168033: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.178.26:58881 > ;rport=58881;branch=z9hG4bKPjKEQ6xXmyzZxA6vCM88vHNYAOPAjUqFsD;received=41.34.123.243 > From: "Sherif 1003" > >;tag=Pz.t-0tGXj3saM.ehBiB1EAMmmitze1D > To: "Sherif 1003" > >;tag=8eQKjmD4ecHej > Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e > CSeq: 60935 REGISTER > Contact: ;expires=300 > Date: Thu, 24 May 2012 23:32:56 GMT > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Content-Length: 0 > > ------------------------------------------------------------------------ > send 989 bytes to udp/[41.34.123.243]:58881 at 23:32:56.360238: > ------------------------------------------------------------------------ > NOTIFY sip:1002 at 192.168.178.26:58881;ob SIP/2.0 > Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKK7gFZ15FSet5B > Route: ;ob > Max-Forwards: 70 > From: > >;tag=9QgcmFy7BN70D > To: > > > Call-ID: a1be7708-209b-1230-8985-00163e6bb553 > CSeq: 28615124 NOTIFY > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 > 05-16-47 +0000 > 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=noresource > Content-Type: application/simple-message-summary > Content-Length: 91 > > Messages-Waiting: yes > Message-Account: sip:1002 at 78.138.90.58 > Voice-Message: 1/0 (0/0) > > ------------------------------------------------------------------------ > recv 308 bytes from udp/[41.34.123.243]:58881 at 23:32:56.623554: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 78.138.90.58:6090 > ;rport=6090;received=78.138.90.58;branch=z9hG4bKK7gFZ15FSet5B > Call-ID: a1be7708-209b-1230-8985-00163e6bb553 > From: > >;tag=9QgcmFy7BN70D > To: > >;tag=z9hG4bKK7gFZ15FSet5B > CSeq: 28615124 NOTIFY > Content-Length: 0 > > ------------------------------------------------------------------------ > freeswitch at internal> > > > --- On *Thu, 5/24/12, Kristian Kielhofner > >* wrote: > > > From: Kristian Kielhofner > > > Subject: Re: [Freeswitch-users] Freeswitch (SBC + Media) does not know > that gateway phone is picked up > To: "FreeSWITCH Users Help" > > > Date: Thursday, May 24, 2012, 7:51 PM > > > Siptrace and logs please. > > On Thu, May 24, 2012 at 11:53 AM, Sherif Omran > > > wrote: > > > > Hi all, > > > > My topology is as follows: > > > > Kamailio -> FS (SBS+Media server) > > > > I came across an issue with my system as follows. I have a Hardphone > registered. When I do local call inside kamailio, it gets to FS and returns > back well and FS understands when I lift the handset. However, I added a > gateway (german landline server), when I call my self from another phone, > the call gets to FS and then transmits to Kamailio, it rings my extension > but when I lift the handset FS does not notice it and keeps ringing. > > > > Any body has an Idea? Here is my gateway trunk. > > > > > > > > > > > > > > > > > > > > > > thanks in advance > > Sherif Omran > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > Kristian Kielhofner > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -----Inline Attachment Follows----- > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -----Inline Attachment Follows----- > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -----Inline Attachment Follows----- > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -----Inline Attachment Follows----- > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120527/aae6f6d4/attachment-0001.html From sherifomran2000 at yahoo.com Sun May 27 17:46:23 2012 From: sherifomran2000 at yahoo.com (Sherif Omran) Date: Sun, 27 May 2012 06:46:23 -0700 (PDT) Subject: [Freeswitch-users] Freeswitch (SBC + Media) does not know that gateway phone is picked up In-Reply-To: Message-ID: <1338126383.39153.YahooMailClassic@web110801.mail.gq1.yahoo.com> hi Sammy, Local calls work well: ?Client A - Kamailio (change in the header) - FS (Prepare for header change by setting the media proxy to enable so that it flows back to kamailio) - Kamailio -? B client Wrong Setting for incoming gateways ?Client A - FS? (Prepare for header change by setting the media proxy to enable so that it flows back to kamailio)- Kamailio? - B Client This was the wrong setting. I tested it by using an extension connected to FS directly before kamailio and it worked well. Thus the problem was when it gets to Kamailio and the header was prepared for a media proxy which was not true Correct Setting for incoming gateways Client A - FS - Kamailo (no need to disable media proxy because it did not go through Kamailio before) - B client Pingo .... It worked well in this setting. I hope this helps. regards, Sherif --- On Sun, 5/27/12, SamyGo wrote: From: SamyGo Subject: Re: [Freeswitch-users] Freeswitch (SBC + Media) does not know that gateway phone is picked up To: "FreeSWITCH Users Help" Date: Sunday, May 27, 2012, 4:26 PM Its good to read that its working for you, I'm still not sure how disabling your proxying media relate to the correction of the SIP signalling flow for a call !! I would?still?like to investigate your final solution to this. Caller====>Provider=====> FS ====>Kamailio====>CalleeKamailio keeps on returning ringingFS keeps on relaying ringing200 OK from Callee never reached FS, (definitely getting lost in kamailio, maybe due to bad/missing record-route or something) Solution you say;disable proxy-media and define new extension !!?? Can you compare the SIP traces of successful calls now (If you like to )!? Regards, Sammy G. On Sun, May 27, 2012 at 2:30 PM, Sherif Omran wrote: Hi Sammy I found the solution. Made a new extension and set proxy media to false. Thats why the call was looping between Kamailio and FS. It now works fine ? regards, Sherif --- On Sun, 5/27/12, SamyGo wrote: From: SamyGo Subject: Re: [Freeswitch-users] Freeswitch (SBC + Media) does not know that gateway phone is picked up To: "FreeSWITCH Users Help" Date: Sunday, May 27, 2012, 8:36 AM OK Sheriff, I've gone through this.?This says that destination on kamailio end is ringing and keeps on ringing state. What Im more interested in are the sip traces on port 5060 the story thats happening in between the destination phone and the kamailo..if u know what I mean !! On Sat, May 26, 2012 at 8:06 PM, Sherif Omran wrote: Hi Sanny, Please see the following links http://pastebin.freeswitch.org/19175 http://pastebin.freeswitch.org/19174 regards, Sherif --- On Sat, 5/26/12, SamyGo wrote: From: SamyGo Subject: Re: [Freeswitch-users] Freeswitch (SBC + Media) does not know that gateway phone is picked up To: "FreeSWITCH Users Help" Date: Saturday, May 26, 2012, 5:52 PM Hey Sherrif, You might wanna resend the pcap file as I couldn;t find any single INVITE or any call in that capture. Please review. BRSaGo. On Sat, May 26, 2012 at 1:05 AM, Sherif Omran wrote: Hello Sammy FS uses port 6090 and registers the GW. When a call comes, it rings the extension but then gives a busy signal and FS keeps ringing till I cancel the call from the GW. recv 477 bytes from udp/[217.74.179.29]:5060 at 19:52:05.662344: ? ?------------------------------------------------------------------------? ?CANCEL sip:gw+trunk_1000 at 78.138.90.58:6090;transport=udp;gw=trunk_1000 SIP/2.0 ? ?Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bKe567.13b0b9e.0? ?Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK16d1a257;rport=5060 ? ?From: "+41793940965" ;tag=as1dccab06 ? ?To: ? ?Call-ID: 5cd37edb776a6b3a35e9713a453a3425 at bluesip.net? ?CSeq: 102 CANCEL ? ?User-Agent: blueSIP PSTN GW ? ?Max-Forwards: 69? ?Content-Length: 0? ?P-hint: USRLOC --- On Fri, 5/25/12, SamyGo wrote: From: SamyGo Subject: Re: [Freeswitch-users] Freeswitch (SBC + Media) does not know that gateway phone is picked up To: "FreeSWITCH Users Help" Date: Friday, May 25, 2012, 10:46 AM Hi again,If you want kamailio register to the provider then use UAC module. Kamailio will use the username/password and register with the provider. Regards,Sammy On Fri, May 25, 2012 at 12:20 PM, Sherif Omran wrote: Dear Sammy, Thank you for your question ... Yes, the GW is registered through FS because I did not know how to register it to kamailio. But it seems better to register it to kamailio. One more information, calls from 1001 to 1002 go to kamailio then to FS then return back to kamailio smoothly. Thus I would suggest that I change registering the gateway from FS to kamailio. but How to? --- On Fri, 5/25/12, SamyGo wrote: From: SamyGo Subject: Re: [Freeswitch-users] Freeswitch (SBC + Media) does not know that gateway phone is picked up To: "FreeSWITCH Users Help" Date: Friday, May 25, 2012, 8:08 AM Hi,These are SIP traces on FreeSWITCH console, whereas you are saying and it seems that kamailio is not detecting the answering (200 OK)of the call from extension 1002. Please, can you take a sip trace..!! I see you've both kamailio and FS on same server! Please take a pcap from the linux console using the following command. #tcpdump -i any -s 0 -w new-call-trace.pcap -vvv Please be quick on turning the sip trace on and off as quickly as possible to avoid extra packets. Once done open the file in wireshark ;?apply filter "sip || rtp" and then save the resulting capture in separate file. Send us the new file to analyse. One more silly question probably, I see REGISTERs coming to your FS as well and the calls to gateways are made from FS too !!, umm...just thinking what are you using kamailio for!!? ThanksBR,Sammy On Fri, May 25, 2012 at 4:40 AM, Sherif Omran wrote: Hi all, here is the siptrace: To figure it out 1- gateway called bluesip.net. It send invide using caller number at bluesip.net 2- This call should go to extension kb-1002. kb means go from freeswitch port 6090 to kamailio port 5060 3- It should go to call extension 1002 in Kamailio 4- Extension 1002 rings but when I reply, it does not notice I replied ./fs_cli ??????????? _____ ____???? ____ _???? ___????????????? ?????????? |? ___/ ___|?? / ___| |?? |_ _|???????????? ?????????? | |_? \___ \? | |?? | |??? | |??????????? ?????????? |? _|? ___) | | |___| |___ | |????????????? ?????????? |_|?? |____/?? \____|_____|___|??????????? ******************************************************* * Anthony Minessale II, Ken Rice,???????????????????? * * Michael Jerris, Travis Cross??????????????????????? * * FreeSWITCH (http://www.freeswitch.org)????????????? * * Paypal Donations Appreciated: paypal at freeswitch.org * * Brought to you by ClueCon http://www.cluecon.com/?? * ******************************************************* Type /help to see a list of commands +OK log level? [7] freeswitch at internal> tracelevel -ERR tracelevel Command not found! freeswitch at internal> sofia global siptrace on +OK Global siptrace on recv 1104 bytes from udp/[217.74.179.29]:5060 at 23:08:45.822962: ?? ------------------------------------------------------------------------ ?? INVITE sip:gw+trunk_1000 at 78.138.90.58:6090;transport=udp;gw=trunk_1000 SIP/2.0 ?? Record-Route: ?? Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 ?? Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 ?? From: "+41793940965" ;tag=as00589402 ?? To: ?? Contact: ?? Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net ?? CSeq: 102 INVITE ?? User-Agent: blueSIP PSTN GW ?? Max-Forwards: 69 ?? Date: Thu, 24 May 2012 23:08:44 GMT ?? Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY ?? Supported: replaces ?? Content-Type: application/sdp ?? Content-Length: 367 ?? P-hint: USRLOC ?? ?? v=0 ?? o=root 20076 20076 IN IP4 217.74.179.28 ?? s=session ?? c=IN IP4 217.74.179.28 ?? t=0 0 ?? m=audio 25626 RTP/AVP 8 0 18 111 3 101 ?? a=rtpmap:8 PCMA/8000 ?? a=rtpmap:0 PCMU/8000 ?? a=rtpmap:18 G729/8000 ?? a=fmtp:18 annexb=no ?? a=rtpmap:111 G726-32/8000 ?? a=rtpmap:3 GSM/8000 ?? a=rtpmap:101 telephone-event/8000 ?? a=fmtp:101 0-16 ?? a=silenceSupp:off - - - - ?? a=ptime:20 ?? a=sendrecv ?? ------------------------------------------------------------------------ send 494 bytes to udp/[217.74.179.29]:5060 at 23:08:45.844486: ?? ------------------------------------------------------------------------ ?? SIP/2.0 100 Trying ?? Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 ?? Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 ?? Record-Route: ?? From: "+41793940965" ;tag=as00589402 ?? To: ?? Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net ?? CSeq: 102 INVITE ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ 2012-05-24 23:08:45.840749 [NOTICE] switch_channel.c:926 New Channel sofia/internal/+41793940965 at bluesip.net [69cd22e6-a5f5-11e1-a833-7b1f73a7ffcf] 2012-05-24 23:08:45.840749 [DEBUG] sofia.c:5532 Channel sofia/internal/+41793940965 at bluesip.net entering state [received][100] 2012-05-24 23:08:45.840749 [DEBUG] sofia.c:5543 Remote SDP: v=0 o=root 20076 20076 IN IP4 217.74.179.28 s=session c=IN IP4 217.74.179.28 t=0 0 m=audio 25626 RTP/AVP 8 0 18 111 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:111 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMA:8:8000:20:64000]/[iLBC:97:8000:30:13330] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMU:0:8000:20:64000]/[iLBC:97:8000:30:13330] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G729:18:8000:20:8000]/[iLBC:97:8000:30:13330] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G726-32:111:8000:20:0]/[iLBC:97:8000:30:13330] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [GSM:3:8000:20:13200]/[iLBC:97:8000:30:13330] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [telephone-event:101:8000:20:0]/[iLBC:97:8000:30:13330] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf send/recv payload to 101 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMA:8:8000:20:64000]/[G7221:115:32000:20:48000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMU:0:8000:20:64000]/[G7221:115:32000:20:48000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G729:18:8000:20:8000]/[G7221:115:32000:20:48000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G726-32:111:8000:20:0]/[G7221:115:32000:20:48000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [GSM:3:8000:20:13200]/[G7221:115:32000:20:48000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [telephone-event:101:8000:20:0]/[G7221:115:32000:20:48000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf send/recv payload to 101 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMA:8:8000:20:64000]/[G7221:107:16000:20:32000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMU:0:8000:20:64000]/[G7221:107:16000:20:32000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G729:18:8000:20:8000]/[G7221:107:16000:20:32000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G726-32:111:8000:20:0]/[G7221:107:16000:20:32000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [GSM:3:8000:20:13200]/[G7221:107:16000:20:32000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [telephone-event:101:8000:20:0]/[G7221:107:16000:20:32000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf send/recv payload to 101 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMA:8:8000:20:64000]/[G722:9:8000:20:64000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G729:18:8000:20:8000]/[G722:9:8000:20:64000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [G726-32:111:8000:20:0]/[G722:9:8000:20:64000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [GSM:3:8000:20:13200]/[G722:9:8000:20:64000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [telephone-event:101:8000:20:0]/[G722:9:8000:20:64000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf send/recv payload to 101 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMA:8:8000:20:64000]/[PCMU:0:8000:20:64000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:4879 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:2996 Set Codec sofia/internal/+41793940965 at bluesip.net PCMU/8000 20 ms 160 samples 64000 bits 2012-05-24 23:08:45.840749 [DEBUG] switch_core_codec.c:111 sofia/internal/+41793940965 at bluesip.net Original read codec set to PCMU:0 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:5000 Set 2833 dtmf send/recv payload to 101 2012-05-24 23:08:45.840749 [DEBUG] sofia.c:5757 (sofia/internal/+41793940965 at bluesip.net) State Change CS_NEW -> CS_INIT 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_INIT 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/+41793940965 at bluesip.net) State INIT 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:85 sofia/internal/+41793940965 at bluesip.net SOFIA INIT 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:125 (sofia/internal/+41793940965 at bluesip.net) State Change CS_INIT -> CS_ROUTING 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/+41793940965 at bluesip.net) State INIT going to sleep 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_ROUTING 2012-05-24 23:08:45.840749 [DEBUG] switch_channel.c:1886 (sofia/internal/+41793940965 at bluesip.net) Callstate Change DOWN -> RINGING 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/+41793940965 at bluesip.net) State ROUTING 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:148 sofia/internal/+41793940965 at bluesip.net SOFIA ROUTING 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:104 sofia/internal/+41793940965 at bluesip.net Standard ROUTING 2012-05-24 23:08:45.840749 [INFO] mod_dialplan_xml.c:485 Processing +41793940965 <+41793940965>->kb-1002 in context public Dialplan: sofia/internal/+41793940965 at bluesip.net parsing [public->from_kamailio] continue=false Dialplan: sofia/internal/+41793940965 at bluesip.net Regex (PASS) [from_kamailio] destination_number(kb-1002) =~ /^(.+)$/ break=on-false Dialplan: sofia/internal/+41793940965 at bluesip.net Action transfer(kb-1002 XML default) 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:154 (sofia/internal/+41793940965 at bluesip.net) State Change CS_ROUTING -> CS_EXECUTE 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/+41793940965 at bluesip.net) State ROUTING going to sleep 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_EXECUTE 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/+41793940965 at bluesip.net) State EXECUTE 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:241 sofia/internal/+41793940965 at bluesip.net SOFIA EXECUTE 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:192 sofia/internal/+41793940965 at bluesip.net Standard EXECUTE EXECUTE sofia/internal/+41793940965 at bluesip.net transfer(kb-1002 XML default) 2012-05-24 23:08:45.840749 [DEBUG] switch_ivr.c:1711 (sofia/internal/+41793940965 at bluesip.net) State Change CS_EXECUTE -> CS_ROUTING 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:731 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:08:45.840749 [NOTICE] switch_ivr.c:1717 Transfer sofia/internal/+41793940965 at bluesip.net to XML[kb-1002 at default] 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/+41793940965 at bluesip.net) State EXECUTE going to sleep 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_ROUTING 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/+41793940965 at bluesip.net) State ROUTING 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:148 sofia/internal/+41793940965 at bluesip.net SOFIA ROUTING 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:104 sofia/internal/+41793940965 at bluesip.net Standard ROUTING 2012-05-24 23:08:45.840749 [INFO] mod_dialplan_xml.c:485 Processing +41793940965 <+41793940965>->kb-1002 in context default Dialplan: sofia/internal/+41793940965 at bluesip.net parsing [default->vbox] continue=false Dialplan: sofia/internal/+41793940965 at bluesip.net Regex (FAIL) [vbox] destination_number(kb-1002) =~ /^vb-([0-9][0-9][0-9])$/ break=on-false Dialplan: sofia/internal/+41793940965 at bluesip.net parsing [default->vmenu] continue=false Dialplan: sofia/internal/+41793940965 at bluesip.net Regex (FAIL) [vmenu] destination_number(kb-1002) =~ /^vm-([0-9][0-9][0-9][0-9])$/ break=on-false Dialplan: sofia/internal/+41793940965 at bluesip.net parsing [default->kbridge] continue=false Dialplan: sofia/internal/+41793940965 at bluesip.net Regex (PASS) [kbridge] destination_number(kb-1002) =~ /^kb-(.+)$/ break=on-false Dialplan: sofia/internal/+41793940965 at bluesip.net Action set(proxy_media=true) Dialplan: sofia/internal/+41793940965 at bluesip.net Action set(call_timeout=50) Dialplan: sofia/internal/+41793940965 at bluesip.net Action set(continue_on_fail=true) Dialplan: sofia/internal/+41793940965 at bluesip.net Action set(hangup_after_bridge=true) Dialplan: sofia/internal/+41793940965 at bluesip.net Action set(sip_invite_domain=78.138.90.58) Dialplan: sofia/internal/+41793940965 at bluesip.net Action export(sip_contact_user=ufs) Dialplan: sofia/internal/+41793940965 at bluesip.net Action bridge(sofia/78.138.90.58/1002 at 78.138.90.58:5060) Dialplan: sofia/internal/+41793940965 at bluesip.net Action answer() Dialplan: sofia/internal/+41793940965 at bluesip.net Action voicemail(default ${domain_name} 1002) 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:154 (sofia/internal/+41793940965 at bluesip.net) State Change CS_ROUTING -> CS_EXECUTE 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/+41793940965 at bluesip.net) State ROUTING going to sleep 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_EXECUTE 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/+41793940965 at bluesip.net) State EXECUTE 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:241 sofia/internal/+41793940965 at bluesip.net SOFIA EXECUTE 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:192 sofia/internal/+41793940965 at bluesip.net Standard EXECUTE EXECUTE sofia/internal/+41793940965 at bluesip.net set(proxy_media=true) 2012-05-24 23:08:45.840749 [DEBUG] mod_dptools.c:1281 sofia/internal/+41793940965 at bluesip.net SET [proxy_media]=[true] EXECUTE sofia/internal/+41793940965 at bluesip.net set(call_timeout=50) 2012-05-24 23:08:45.840749 [DEBUG] mod_dptools.c:1281 sofia/internal/+41793940965 at bluesip.net SET [call_timeout]=[50] EXECUTE sofia/internal/+41793940965 at bluesip.net set(continue_on_fail=true) 2012-05-24 23:08:45.840749 [DEBUG] mod_dptools.c:1281 sofia/internal/+41793940965 at bluesip.net SET [continue_on_fail]=[true] EXECUTE sofia/internal/+41793940965 at bluesip.net set(hangup_after_bridge=true) 2012-05-24 23:08:45.840749 [DEBUG] mod_dptools.c:1281 sofia/internal/+41793940965 at bluesip.net SET [hangup_after_bridge]=[true] EXECUTE sofia/internal/+41793940965 at bluesip.net set(sip_invite_domain=78.138.90.58) 2012-05-24 23:08:45.840749 [DEBUG] mod_dptools.c:1281 sofia/internal/+41793940965 at bluesip.net SET [sip_invite_domain]=[78.138.90.58] EXECUTE sofia/internal/+41793940965 at bluesip.net export(sip_contact_user=ufs) 2012-05-24 23:08:45.840749 [DEBUG] switch_channel.c:1093 EXPORT (export_vars) [sip_contact_user]=[ufs] EXECUTE sofia/internal/+41793940965 at bluesip.net bridge(sofia/78.138.90.58/1002 at 78.138.90.58:5060) 2012-05-24 23:08:45.840749 [DEBUG] switch_channel.c:1047 sofia/internal/+41793940965 at bluesip.net EXPORTING[export_vars] [sip_contact_user]=[ufs] to event 2012-05-24 23:08:45.840749 [DEBUG] switch_ivr_originate.c:1884 Parsing global variables 2012-05-24 23:08:45.840749 [NOTICE] switch_channel.c:926 New Channel sofia/internal/1002 at 78.138.90.58:5060 [69ce1f0c-a5f5-11e1-a838-7b1f73a7ffcf] 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:4691 (sofia/internal/1002 at 78.138.90.58:5060) State Change CS_NEW -> CS_INIT 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1002 at 78.138.90.58:5060) Running State Change CS_INIT 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/1002 at 78.138.90.58:5060) State INIT 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:85 sofia/internal/1002 at 78.138.90.58:5060 SOFIA INIT 2012-05-24 23:08:45.840749 [DEBUG] sofia_glue.c:1871 sofia/internal/1002 at 78.138.90.58:5060 Patched SDP --- v=0 o=root 20076 20076 IN IP4 217.74.179.28 s=session c=IN IP4 217.74.179.28 t=0 0 m=audio 25626 RTP/AVP 8 0 18 111 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:111 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 +++ v=0 o=FreeSWITCH 1629510322 1629510323 IN IP4 78.138.90.58 s=FreeSWITCH c=IN IP4 78.138.90.58 t=0 0 m=audio 31178 RTP/AVP 8 0 18 111 3 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:111 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 send 1340 bytes to udp/[78.138.90.58]:5060 at 23:08:45.852280: ?? ------------------------------------------------------------------------ ?? INVITE sip:1002 at 78.138.90.58:5060 SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKapDgFeegKyHXK ?? Max-Forwards: 67 ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? CSeq: 28614398 INVITE ?? Contact: ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? 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: 372 ?? P-hint: USRLOC ?? X-FS-Support: update_display,send_info ?? Remote-Party-ID: "+41793940965" ;party=calling;screen=yes;privacy=off ?? ?? v=0 ?? o=FreeSWITCH 1629510322 1629510323 IN IP4 78.138.90.58 ?? s=FreeSWITCH ?? c=IN IP4 78.138.90.58 ?? t=0 0 ?? m=audio 31178 RTP/AVP 8 0 18 111 3 101 ?? a=rtpmap:8 PCMA/8000 ?? a=rtpmap:0 PCMU/8000 ?? a=rtpmap:18 G729/8000 ?? a=fmtp:18 annexb=no ?? a=rtpmap:111 G726-32/8000 ?? a=rtpmap:3 GSM/8000 ?? a=rtpmap:101 telephone-event/8000 ?? a=fmtp:101 0-16 ?? a=silenceSupp:off - - - - ?? a=ptime:20 ?? ------------------------------------------------------------------------ 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:125 (sofia/internal/1002 at 78.138.90.58:5060) State Change CS_INIT -> CS_ROUTING 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/1002 at 78.138.90.58:5060) State INIT going to sleep 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1002 at 78.138.90.58:5060) Running State Change CS_ROUTING 2012-05-24 23:08:45.840749 [DEBUG] switch_channel.c:1886 (sofia/internal/1002 at 78.138.90.58:5060) Callstate Change DOWN -> RINGING 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/1002 at 78.138.90.58:5060) State ROUTING 2012-05-24 23:08:45.840749 [DEBUG] mod_sofia.c:148 sofia/internal/1002 at 78.138.90.58:5060 SOFIA ROUTING 2012-05-24 23:08:45.840749 [DEBUG] switch_ivr_originate.c:66 (sofia/internal/1002 at 78.138.90.58:5060) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2012-05-24 23:08:45.840749 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/1002 at 78.138.90.58:5060) State ROUTING going to sleep 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1002 at 78.138.90.58:5060) Running State Change CS_CONSUME_MEDIA 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:429 (sofia/internal/1002 at 78.138.90.58:5060) State CONSUME_MEDIA 2012-05-24 23:08:45.840749 [DEBUG] switch_core_state_machine.c:429 (sofia/internal/1002 at 78.138.90.58:5060) State CONSUME_MEDIA going to sleep 2012-05-24 23:08:45.840749 [DEBUG] sofia.c:5532 Channel sofia/internal/1002 at 78.138.90.58:5060 entering state [calling][0] recv 365 bytes from udp/[78.138.90.58]:5060 at 23:08:45.964309: ?? ------------------------------------------------------------------------ ?? SIP/2.0 100 trying -- your call is important to us ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKapDgFeegKyHXK ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? CSeq: 28614398 INVITE ?? Server: kamailio (3.1.5 (x86_64/linux)) ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 496 bytes from udp/[78.138.90.58]:5060 at 23:08:47.414766: ?? ------------------------------------------------------------------------ ?? SIP/2.0 180 Ringing ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKapDgFeegKyHXK ?? Record-Route: ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ;tag=qCSaB2YJOAsAr2dJ-RWNMhPoDdn1RFvs ?? CSeq: 28614398 INVITE ?? Contact: "Mama" ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, REGISTER ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ 2012-05-24 23:08:47.400741 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:47.400741 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:47.400741 [DEBUG] sofia.c:5532 Channel sofia/internal/1002 at 78.138.90.58:5060 entering state [proceeding][180] 2012-05-24 23:08:47.400741 [NOTICE] sofia.c:5624 Ring-Ready sofia/internal/1002 at 78.138.90.58:5060! send 995 bytes to udp/[217.74.179.29]:5060 at 23:08:47.416538: ?? ------------------------------------------------------------------------ ?? SIP/2.0 180 Ringing ?? Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 ?? Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 ?? Record-Route: ?? From: "+41793940965" ;tag=as00589402 ?? To: ;tag=S7UZQygFt62Nm ?? Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net ?? CSeq: 102 INVITE ?? Contact: ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Accept: application/sdp ?? 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-Length: 0 ?? Remote-Party-ID: "Outbound Call" ;party=calling;privacy=off;screen=no ?? ?? ------------------------------------------------------------------------ 2012-05-24 23:08:47.400741 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:08:47.400741 [NOTICE] mod_sofia.c:2514 Ring-Ready sofia/internal/+41793940965 at bluesip.net! 2012-05-24 23:08:47.400741 [DEBUG] sofia.c:5532 Channel sofia/internal/+41793940965 at bluesip.net entering state [early][180] 2012-05-24 23:08:47.400741 [DEBUG] switch_core_session.c:731 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:08:47.400741 [NOTICE] switch_ivr_originate.c:483 Ring Ready sofia/internal/+41793940965 at bluesip.net! recv 496 bytes from udp/[78.138.90.58]:5060 at 23:08:47.804967: ?? ------------------------------------------------------------------------ ?? SIP/2.0 180 Ringing ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKapDgFeegKyHXK ?? Record-Route: ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ;tag=549D2DD03BBA7C67 ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? CSeq: 28614398 INVITE ?? Contact: ?? User-Agent: AVM FRITZ!Box Fon WLAN 7170 29.04.80 (Jan 27 2010) ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ 2012-05-24 23:08:47.800746 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:47.800746 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:47.800746 [DEBUG] sofia.c:5532 Channel sofia/internal/1002 at 78.138.90.58:5060 entering state [proceeding][180] recv 496 bytes from udp/[78.138.90.58]:5060 at 23:08:48.182098: ?? ------------------------------------------------------------------------ ?? SIP/2.0 180 Ringing ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKapDgFeegKyHXK ?? Record-Route: ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ;tag=qCSaB2YJOAsAr2dJ-RWNMhPoDdn1RFvs ?? CSeq: 28614398 INVITE ?? Contact: "Mama" ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, REGISTER ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ 2012-05-24 23:08:48.180741 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:48.180741 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:48.180741 [DEBUG] sofia.c:5532 Channel sofia/internal/1002 at 78.138.90.58:5060 entering state [proceeding][180] recv 496 bytes from udp/[78.138.90.58]:5060 at 23:08:48.417403: ?? ------------------------------------------------------------------------ ?? SIP/2.0 180 Ringing ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKapDgFeegKyHXK ?? Record-Route: ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ;tag=549D2DD03BBA7C67 ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? CSeq: 28614398 INVITE ?? Contact: ?? User-Agent: AVM FRITZ!Box Fon WLAN 7170 29.04.80 (Jan 27 2010) ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ 2012-05-24 23:08:48.400744 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:48.400744 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:48.400744 [DEBUG] sofia.c:5532 Channel sofia/internal/1002 at 78.138.90.58:5060 entering state [proceeding][180] recv 496 bytes from udp/[78.138.90.58]:5060 at 23:08:49.292599: ?? ------------------------------------------------------------------------ ?? SIP/2.0 180 Ringing ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKapDgFeegKyHXK ?? Record-Route: ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ;tag=549D2DD03BBA7C67 ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? CSeq: 28614398 INVITE ?? Contact: ?? User-Agent: AVM FRITZ!Box Fon WLAN 7170 29.04.80 (Jan 27 2010) ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ 2012-05-24 23:08:49.280742 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:49.280742 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:08:49.280742 [DEBUG] sofia.c:5532 Channel sofia/internal/1002 at 78.138.90.58:5060 entering state [proceeding][180] recv 478 bytes from udp/[217.74.179.29]:5060 at 23:09:15.066012: ?? ------------------------------------------------------------------------ ?? CANCEL sip:gw+trunk_1000 at 78.138.90.58:6090;transport=udp;gw=trunk_1000 SIP/2.0 ?? Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 ?? Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 ?? From: "+41793940965" ;tag=as00589402 ?? To: ?? Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net ?? CSeq: 102 CANCEL ?? User-Agent: blueSIP PSTN GW ?? Max-Forwards: 69 ?? Content-Length: 0 ?? P-hint: USRLOC ?? ?? ------------------------------------------------------------------------ send 369 bytes to udp/[217.74.179.29]:5060 at 23:09:15.066176: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 ?? Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 ?? From: "+41793940965" ;tag=as00589402 ?? To: ;tag=S7UZQygFt62Nm ?? Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net ?? CSeq: 102 CANCEL ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ send 766 bytes to udp/[217.74.179.29]:5060 at 23:09:15.066289: ?? ------------------------------------------------------------------------ ?? SIP/2.0 487 Request Terminated ?? Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 ?? Via: SIP/2.0/UDP 217.74.179.28:5060;branch=z9hG4bK086622bb;rport=5060 ?? From: "+41793940965" ;tag=as00589402 ?? To: ;tag=S7UZQygFt62Nm ?? Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net ?? CSeq: 102 INVITE ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? 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-Length: 0 ?? ?? ------------------------------------------------------------------------ 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:877 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:09:15.060742 [DEBUG] sofia.c:5532 Channel sofia/internal/+41793940965 at bluesip.net entering state [terminated][487] 2012-05-24 23:09:15.060742 [DEBUG] switch_channel.c:2848 (sofia/internal/+41793940965 at bluesip.net) Callstate Change RINGING -> HANGUP 2012-05-24 23:09:15.060742 [NOTICE] sofia.c:6301 Hangup sofia/internal/+41793940965 at bluesip.net [CS_EXECUTE] [ORIGINATOR_CANCEL] 2012-05-24 23:09:15.060742 [DEBUG] switch_channel.c:2871 Send signal sofia/internal/+41793940965 at bluesip.net [KILL] 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:09:15.060742 [DEBUG] switch_channel.c:2848 (sofia/internal/1002 at 78.138.90.58:5060) Callstate Change RINGING -> HANGUP 2012-05-24 23:09:15.060742 [NOTICE] switch_ivr_originate.c:3182 Hangup sofia/internal/1002 at 78.138.90.58:5060 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL] 2012-05-24 23:09:15.060742 [DEBUG] switch_channel.c:2871 Send signal sofia/internal/1002 at 78.138.90.58:5060 [KILL] 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1002 at 78.138.90.58:5060) Running State Change CS_HANGUP 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/1002 at 78.138.90.58:5060) State HANGUP 2012-05-24 23:09:15.060742 [DEBUG] mod_sofia.c:469 Channel sofia/internal/1002 at 78.138.90.58:5060 hanging up, cause: ORIGINATOR_CANCEL 2012-05-24 23:09:15.060742 [DEBUG] mod_sofia.c:523 Sending CANCEL to sofia/internal/1002 at 78.138.90.58:5060 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:47 sofia/internal/1002 at 78.138.90.58:5060 Standard HANGUP, cause: ORIGINATOR_CANCEL 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/1002 at 78.138.90.58:5060) State HANGUP going to sleep 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:393 (sofia/internal/1002 at 78.138.90.58:5060) State Change CS_HANGUP -> CS_REPORTING 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1002 at 78.138.90.58:5060) Running State Change CS_REPORTING 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/1002 at 78.138.90.58:5060) State REPORTING 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:79 sofia/internal/1002 at 78.138.90.58:5060 Standard REPORTING, cause: ORIGINATOR_CANCEL 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/1002 at 78.138.90.58:5060) State REPORTING going to sleep 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:387 (sofia/internal/1002 at 78.138.90.58:5060) State Change CS_REPORTING -> CS_DESTROY 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1382 Session 9 (sofia/internal/1002 at 78.138.90.58:5060) Locked, Waiting on external entities 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/1002 at 78.138.90.58:5060 [BREAK] 2012-05-24 23:09:15.060742 [DEBUG] switch_ivr_originate.c:3358 Originate Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL] 2012-05-24 23:09:15.060742 [NOTICE] switch_core_session.c:1400 Session 9 (sofia/internal/1002 at 78.138.90.58:5060) Ended 2012-05-24 23:09:15.060742 [NOTICE] switch_core_session.c:1402 Close Channel sofia/internal/1002 at 78.138.90.58:5060 [CS_DESTROY] 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:491 (sofia/internal/1002 at 78.138.90.58:5060) Callstate Change HANGUP -> DOWN 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:494 (sofia/internal/1002 at 78.138.90.58:5060) Running State Change CS_DESTROY 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/1002 at 78.138.90.58:5060) State DESTROY 2012-05-24 23:09:15.060742 [DEBUG] mod_sofia.c:374 sofia/internal/1002 at 78.138.90.58:5060 SOFIA DESTROY 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:86 sofia/internal/1002 at 78.138.90.58:5060 Standard DESTROY 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/1002 at 78.138.90.58:5060) State DESTROY going to sleep 2012-05-24 23:09:15.060742 [INFO] mod_dptools.c:2922 Originate Failed.? Cause: ORIGINATOR_CANCEL 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:2287 sofia/internal/+41793940965 at bluesip.net skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/+41793940965 at bluesip.net) State EXECUTE going to sleep 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_HANGUP 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/+41793940965 at bluesip.net) State HANGUP 2012-05-24 23:09:15.060742 [DEBUG] mod_sofia.c:469 Channel sofia/internal/+41793940965 at bluesip.net hanging up, cause: ORIGINATOR_CANCEL 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:47 sofia/internal/+41793940965 at bluesip.net Standard HANGUP, cause: ORIGINATOR_CANCEL 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/+41793940965 at bluesip.net) State HANGUP going to sleep 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:393 (sofia/internal/+41793940965 at bluesip.net) State Change CS_HANGUP -> CS_REPORTING 2012-05-24 23:09:15.060742 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_REPORTING 2012-05-24 23:09:15.060742 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/+41793940965 at bluesip.net) State REPORTING send 383 bytes to udp/[78.138.90.58]:5060 at 23:09:15.087695: ?? ------------------------------------------------------------------------ ?? CANCEL sip:1002 at 78.138.90.58:5060 SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKapDgFeegKyHXK ?? Max-Forwards: 67 ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? CSeq: 28614398 CANCEL ?? Reason: FreeSWITCH;cause=487;text="ORIGINATOR_CANCEL" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 408 bytes from udp/[217.74.179.29]:5060 at 23:09:15.094670: ?? ------------------------------------------------------------------------ ?? ACK sip:gw+trunk_1000 at 78.138.90.58:6090;transport=udp;gw=trunk_1000 SIP/2.0 ?? Via: SIP/2.0/UDP 217.74.179.29;branch=z9hG4bK0dc9.284b9a03.0 ?? From: "+41793940965" ;tag=as00589402 ?? Call-ID: 4253e4bd2d624caf7f18faf85390b68f at bluesip.net ?? To: ;tag=S7UZQygFt62Nm ?? CSeq: 102 ACK ?? User-Agent: Sip EXpress router(0.9.7 (i386/linux)) ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:79 sofia/internal/+41793940965 at bluesip.net Standard REPORTING, cause: ORIGINATOR_CANCEL 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/+41793940965 at bluesip.net) State REPORTING going to sleep 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:387 (sofia/internal/+41793940965 at bluesip.net) State Change CS_REPORTING -> CS_DESTROY 2012-05-24 23:09:15.120807 [DEBUG] switch_core_session.c:1182 Send signal sofia/internal/+41793940965 at bluesip.net [BREAK] 2012-05-24 23:09:15.120807 [DEBUG] switch_core_session.c:1382 Session 8 (sofia/internal/+41793940965 at bluesip.net) Locked, Waiting on external entities 2012-05-24 23:09:15.120807 [NOTICE] switch_core_session.c:1400 Session 8 (sofia/internal/+41793940965 at bluesip.net) Ended 2012-05-24 23:09:15.120807 [NOTICE] switch_core_session.c:1402 Close Channel sofia/internal/+41793940965 at bluesip.net [CS_DESTROY] 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:491 (sofia/internal/+41793940965 at bluesip.net) Callstate Change HANGUP -> DOWN 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:494 (sofia/internal/+41793940965 at bluesip.net) Running State Change CS_DESTROY 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/+41793940965 at bluesip.net) State DESTROY 2012-05-24 23:09:15.120807 [DEBUG] mod_sofia.c:374 sofia/internal/+41793940965 at bluesip.net SOFIA DESTROY 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:86 sofia/internal/+41793940965 at bluesip.net Standard DESTROY 2012-05-24 23:09:15.120807 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/+41793940965 at bluesip.net) State DESTROY going to sleep recv 378 bytes from udp/[78.138.90.58]:5060 at 23:09:15.135235: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 canceling ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKapDgFeegKyHXK ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ;tag=a6a1c5f60faecf035a1ae5b6e96e979a-8c5b ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? CSeq: 28614398 CANCEL ?? Server: kamailio (3.1.5 (x86_64/linux)) ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 446 bytes from udp/[78.138.90.58]:5060 at 23:09:16.419308: ?? ------------------------------------------------------------------------ ?? SIP/2.0 487 Request Terminated ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKapDgFeegKyHXK ?? Record-Route: ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ;tag=qCSaB2YJOAsAr2dJ-RWNMhPoDdn1RFvs ?? CSeq: 28614398 INVITE ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, REGISTER ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 354 bytes to udp/[78.138.90.58]:5060 at 23:09:16.419543: ?? ------------------------------------------------------------------------ ?? ACK sip:1002 at 78.138.90.58:5060 SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKapDgFeegKyHXK ?? Max-Forwards: 67 ?? From: "+41793940965" ;tag=tgNrSS1jQFS8F ?? To: ;tag=qCSaB2YJOAsAr2dJ-RWNMhPoDdn1RFvs ?? Call-ID: 412c8312-2098-1230-8985-00163e6bb553 ?? CSeq: 28614398 ACK ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 564 bytes from udp/[41.34.123.243]:58881 at 23:13:13.612418: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjHQIpv.ACHGbXD4IHrMr72LwZ8kYdlu-J ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=wOP.FVT4Crk8O5E.bB7BWUhNO6QG91IG ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60926 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 744 bytes to udp/[41.34.123.243]:58881 at 23:13:13.613152: ?? ------------------------------------------------------------------------ ?? SIP/2.0 401 Unauthorized ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjHQIpv.ACHGbXD4IHrMr72LwZ8kYdlu-J;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=wOP.FVT4Crk8O5E.bB7BWUhNO6QG91IG ?? To: "Sherif 1003" ;tag=USeHUmjpmrFUB ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60926 REGISTER ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? WWW-Authenticate: Digest realm="sip.pcfone.com", nonce="09674764-a5f6-11e1-a83c-7b1f73a7ffcf", algorithm=MD5, qop="auth" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 830 bytes from udp/[41.34.123.243]:58881 at 23:13:13.847746: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjFr7rIAdAJPRQd2GmrMrVib3KT3Rup0q4 ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=wOP.FVT4Crk8O5E.bB7BWUhNO6QG91IG ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60927 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Authorization: Digest username="1002", realm="sip.pcfone.com", nonce="09674764-a5f6-11e1-a83c-7b1f73a7ffcf", uri="sip:sip.pcfone.com:6090", response="3bc0bb82a803ded3641704c4eb78d58f", algorithm=MD5, cnonce="sc3AJfhuev18Bs-L-6tORmI.RXlxnPOz", qop=auth, nc=00000001 ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 706 bytes to udp/[41.34.123.243]:58881 at 23:13:13.850197: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjFr7rIAdAJPRQd2GmrMrVib3KT3Rup0q4;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=wOP.FVT4Crk8O5E.bB7BWUhNO6QG91IG ?? To: "Sherif 1003" ;tag=v279vF3SH15DQ ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60927 REGISTER ?? Contact: ;expires=300 ?? Date: Thu, 24 May 2012 23:13:13 GMT ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ send 989 bytes to udp/[41.34.123.243]:58881 at 23:13:13.888717: ?? ------------------------------------------------------------------------ ?? NOTIFY sip:1002 at 192.168.178.26:58881;ob SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKBZ68g9yKg77FF ?? Route: ;ob ?? Max-Forwards: 70 ?? From: ;tag=XB12yamXeav0j ?? To: ?? Call-ID: e0efa252-2098-1230-8985-00163e6bb553 ?? CSeq: 28614532 NOTIFY ?? Contact: ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? 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=noresource ?? Content-Type: application/simple-message-summary ?? Content-Length: 91 ?? ?? Messages-Waiting: yes ?? Message-Account: sip:1002 at 78.138.90.58 ?? Voice-Message: 1/0 (0/0) ?? ?? ------------------------------------------------------------------------ recv 308 bytes from udp/[41.34.123.243]:58881 at 23:13:14.298850: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;received=78.138.90.58;branch=z9hG4bKBZ68g9yKg77FF ?? Call-ID: e0efa252-2098-1230-8985-00163e6bb553 ?? From: ;tag=XB12yamXeav0j ?? To: ;tag=z9hG4bKBZ68g9yKg77FF ?? CSeq: 28614532 NOTIFY ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ recv 564 bytes from udp/[41.34.123.243]:58881 at 23:18:09.067232: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjABTslmZIhiHYRtLqQn2erwmX19MVYHFC ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=JBSXL5WZLWbLzpvRFoTSP5x8qQslI1ST ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60928 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 744 bytes to udp/[41.34.123.243]:58881 at 23:18:09.068121: ?? ------------------------------------------------------------------------ ?? SIP/2.0 401 Unauthorized ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjABTslmZIhiHYRtLqQn2erwmX19MVYHFC;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=JBSXL5WZLWbLzpvRFoTSP5x8qQslI1ST ?? To: "Sherif 1003" ;tag=ymtU0540BKjKe ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60928 REGISTER ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? WWW-Authenticate: Digest realm="sip.pcfone.com", nonce="b9821df4-a5f6-11e1-a83d-7b1f73a7ffcf", algorithm=MD5, qop="auth" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 830 bytes from udp/[41.34.123.243]:58881 at 23:18:09.302944: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjRgtztcy63vMT97Jp5pH9n311li5EFeY- ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=JBSXL5WZLWbLzpvRFoTSP5x8qQslI1ST ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60929 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Authorization: Digest username="1002", realm="sip.pcfone.com", nonce="b9821df4-a5f6-11e1-a83d-7b1f73a7ffcf", uri="sip:sip.pcfone.com:6090", response="519c6301816b51f3bbb98d97a347e92b", algorithm=MD5, cnonce="sc3AJfhuev18Bs-L-6tORmI.RXlxnPOz", qop=auth, nc=00000001 ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 706 bytes to udp/[41.34.123.243]:58881 at 23:18:09.305217: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjRgtztcy63vMT97Jp5pH9n311li5EFeY-;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=JBSXL5WZLWbLzpvRFoTSP5x8qQslI1ST ?? To: "Sherif 1003" ;tag=ZXKm20N48U85S ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60929 REGISTER ?? Contact: ;expires=300 ?? Date: Thu, 24 May 2012 23:18:09 GMT ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ send 989 bytes to udp/[41.34.123.243]:58881 at 23:18:09.379987: ?? ------------------------------------------------------------------------ ?? NOTIFY sip:1002 at 192.168.178.26:58881;ob SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKc8Z1j4FQDgy2a ?? Route: ;ob ?? Max-Forwards: 70 ?? From: ;tag=06cD4U6754yrN ?? To: ?? Call-ID: 91100602-2099-1230-8985-00163e6bb553 ?? CSeq: 28614680 NOTIFY ?? Contact: ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? 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=noresource ?? Content-Type: application/simple-message-summary ?? Content-Length: 91 ?? ?? Messages-Waiting: yes ?? Message-Account: sip:1002 at 78.138.90.58 ?? Voice-Message: 1/0 (0/0) ?? ?? ------------------------------------------------------------------------ recv 308 bytes from udp/[41.34.123.243]:58881 at 23:18:09.773629: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;received=78.138.90.58;branch=z9hG4bKc8Z1j4FQDgy2a ?? Call-ID: 91100602-2099-1230-8985-00163e6bb553 ?? From: ;tag=06cD4U6754yrN ?? To: ;tag=z9hG4bKc8Z1j4FQDgy2a ?? CSeq: 28614680 NOTIFY ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ recv 564 bytes from udp/[41.34.123.243]:58881 at 23:23:04.522455: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjaCuKyaBi6ZqcsHyp5HhH3WthJStom0SK ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=qfUt1fQWuE-Ro3zWld6X0r6WidtD6B1q ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60930 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 744 bytes to udp/[41.34.123.243]:58881 at 23:23:04.523158: ?? ------------------------------------------------------------------------ ?? SIP/2.0 401 Unauthorized ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjaCuKyaBi6ZqcsHyp5HhH3WthJStom0SK;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=qfUt1fQWuE-Ro3zWld6X0r6WidtD6B1q ?? To: "Sherif 1003" ;tag=1F655pQB3DNBH ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60930 REGISTER ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? WWW-Authenticate: Digest realm="sip.pcfone.com", nonce="699cfe2a-a5f7-11e1-a83e-7b1f73a7ffcf", algorithm=MD5, qop="auth" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 830 bytes from udp/[41.34.123.243]:58881 at 23:23:04.757798: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjs8SAufeoyzCLrePaAbYhZuOqQF47XIc- ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=qfUt1fQWuE-Ro3zWld6X0r6WidtD6B1q ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60931 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Authorization: Digest username="1002", realm="sip.pcfone.com", nonce="699cfe2a-a5f7-11e1-a83e-7b1f73a7ffcf", uri="sip:sip.pcfone.com:6090", response="23a886934aaf049f08432b6bb226bd37", algorithm=MD5, cnonce="sc3AJfhuev18Bs-L-6tORmI.RXlxnPOz", qop=auth, nc=00000001 ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 706 bytes to udp/[41.34.123.243]:58881 at 23:23:04.759924: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjs8SAufeoyzCLrePaAbYhZuOqQF47XIc-;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=qfUt1fQWuE-Ro3zWld6X0r6WidtD6B1q ?? To: "Sherif 1003" ;tag=2rZy7H8e0pByc ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60931 REGISTER ?? Contact: ;expires=300 ?? Date: Thu, 24 May 2012 23:23:04 GMT ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ send 989 bytes to udp/[41.34.123.243]:58881 at 23:23:04.872728: ?? ------------------------------------------------------------------------ ?? NOTIFY sip:1002 at 192.168.178.26:58881;ob SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKDHStmZ0taSmNp ?? Route: ;ob ?? Max-Forwards: 70 ?? From: ;tag=31rQ9cSjXZ1gr ?? To: ?? Call-ID: 4130a1ce-209a-1230-8985-00163e6bb553 ?? CSeq: 28614828 NOTIFY ?? Contact: ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? 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=noresource ?? Content-Type: application/simple-message-summary ?? Content-Length: 91 ?? ?? Messages-Waiting: yes ?? Message-Account: sip:1002 at 78.138.90.58 ?? Voice-Message: 1/0 (0/0) ?? ?? ------------------------------------------------------------------------ recv 308 bytes from udp/[41.34.123.243]:58881 at 23:23:05.269063: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;received=78.138.90.58;branch=z9hG4bKDHStmZ0taSmNp ?? Call-ID: 4130a1ce-209a-1230-8985-00163e6bb553 ?? From: ;tag=31rQ9cSjXZ1gr ?? To: ;tag=z9hG4bKDHStmZ0taSmNp ?? CSeq: 28614828 NOTIFY ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 916 bytes to udp/[217.74.179.29]:5060 at 23:25:45.873423: ?? ------------------------------------------------------------------------ ?? REGISTER sip:bluesip.net;transport=udp SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKetjKptHy71a8H ?? Max-Forwards: 70 ?? From: ;tag=6r0vBQZS650Fg ?? To: ?? Call-ID: 2a3e4696-a5b9-11e1-a7ed-7b1f73a7ffcf ?? CSeq: 28601493 REGISTER ?? Contact: ?? Expires: 3600 ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? Authorization: Digest username="bluesip/sherifomran", realm="bluesip.net", nonce="4fbeba4ebfb7414d5c701bd6f994402d551f3acf", algorithm=MD5, uri="sip:bluesip.net;transport=udp", response="c6cdcafe0418e519bc9ee0d8fa3d4d74" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 722 bytes from udp/[217.74.179.29]:5060 at 23:25:45.904176: ?? ------------------------------------------------------------------------ ?? SIP/2.0 401 Unauthorized ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKetjKptHy71a8H ?? From: ;tag=6r0vBQZS650Fg ?? To: ;tag=0354a2e1b960c9cc2279eca4e5f84e20.1b16 ?? Call-ID: 2a3e4696-a5b9-11e1-a7ed-7b1f73a7ffcf ?? CSeq: 28601493 REGISTER ?? WWW-Authenticate: Digest realm="bluesip.net", nonce="4fbec4a4ba82d1f1feaa32ccec01b85be054fcf7", stale=true ?? Server: Sip EXpress router (0.9.7 (i386/linux)) ?? Content-Length: 0 ?? Warning: 392 217.74.179.29:5060 "Noisy feedback tells:? pid=5455 req_src_ip=78.138.90.58 req_src_port=6090 in_uri=sip:bluesip.net;transport=udp out_uri=sip:bluesip.net;transport=udp via_cnt==1" ?? ?? ------------------------------------------------------------------------ send 916 bytes to udp/[217.74.179.29]:5060 at 23:25:45.904416: ?? ------------------------------------------------------------------------ ?? REGISTER sip:bluesip.net;transport=udp SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKF3BcrN214a1tD ?? Max-Forwards: 70 ?? From: ;tag=6r0vBQZS650Fg ?? To: ?? Call-ID: 2a3e4696-a5b9-11e1-a7ed-7b1f73a7ffcf ?? CSeq: 28601494 REGISTER ?? Contact: ?? Expires: 3600 ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? Authorization: Digest username="bluesip/sherifomran", realm="bluesip.net", nonce="4fbec4a4ba82d1f1feaa32ccec01b85be054fcf7", algorithm=MD5, uri="sip:bluesip.net;transport=udp", response="4c09dbe4b9accb52d4104b40dfe20040" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 699 bytes from udp/[217.74.179.29]:5060 at 23:25:45.935046: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKF3BcrN214a1tD ?? From: ;tag=6r0vBQZS650Fg ?? To: ;tag=0354a2e1b960c9cc2279eca4e5f84e20.2017 ?? Call-ID: 2a3e4696-a5b9-11e1-a7ed-7b1f73a7ffcf ?? CSeq: 28601494 REGISTER ?? Contact: ;q=0.5;expires=3600 ?? Server: Sip EXpress router (0.9.7 (i386/linux)) ?? Content-Length: 0 ?? Warning: 392 217.74.179.29:5060 "Noisy feedback tells:? pid=5462 req_src_ip=78.138.90.58 req_src_port=6090 in_uri=sip:bluesip.net;transport=udp out_uri=sip:bluesip.net;transport=udp via_cnt==1" ?? ?? ------------------------------------------------------------------------ recv 564 bytes from udp/[41.34.123.243]:58881 at 23:27:59.977057: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjEOP.a4NoXqoFqC7a.k2tyeX-mYlUyH4H ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60932 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 744 bytes to udp/[41.34.123.243]:58881 at 23:27:59.977960: ?? ------------------------------------------------------------------------ ?? SIP/2.0 401 Unauthorized ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjEOP.a4NoXqoFqC7a.k2tyeX-mYlUyH4H;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K ?? To: "Sherif 1003" ;tag=4ajgB89Nt8Q3K ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60932 REGISTER ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? WWW-Authenticate: Digest realm="sip.pcfone.com", nonce="19b7cc40-a5f8-11e1-a83f-7b1f73a7ffcf", algorithm=MD5, qop="auth" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 564 bytes from udp/[41.34.123.243]:58881 at 23:28:00.468487: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjEOP.a4NoXqoFqC7a.k2tyeX-mYlUyH4H ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60932 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 744 bytes to udp/[41.34.123.243]:58881 at 23:28:00.468679: ?? ------------------------------------------------------------------------ ?? SIP/2.0 401 Unauthorized ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjEOP.a4NoXqoFqC7a.k2tyeX-mYlUyH4H;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K ?? To: "Sherif 1003" ;tag=4ajgB89Nt8Q3K ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60932 REGISTER ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? WWW-Authenticate: Digest realm="sip.pcfone.com", nonce="19b7cc40-a5f8-11e1-a83f-7b1f73a7ffcf", algorithm=MD5, qop="auth" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 830 bytes from udp/[41.34.123.243]:58881 at 23:28:00.717446: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPj0iHd3-XUjnvcBs9DiJUCWzrmYTmWxFLt ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60933 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Authorization: Digest username="1002", realm="sip.pcfone.com", nonce="19b7cc40-a5f8-11e1-a83f-7b1f73a7ffcf", uri="sip:sip.pcfone.com:6090", response="e21c903e6d98b5adcdd2eefa75217157", algorithm=MD5, cnonce="sc3AJfhuev18Bs-L-6tORmI.RXlxnPOz", qop=auth, nc=00000001 ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 706 bytes to udp/[41.34.123.243]:58881 at 23:28:00.719947: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPj0iHd3-XUjnvcBs9DiJUCWzrmYTmWxFLt;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=FYtvWBFYne8l.-.5Q6JgnCajTtrXQY-K ?? To: "Sherif 1003" ;tag=5KB9c3tSQHepF ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60933 REGISTER ?? Contact: ;expires=300 ?? Date: Thu, 24 May 2012 23:28:00 GMT ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ send 989 bytes to udp/[41.34.123.243]:58881 at 23:28:00.868368: ?? ------------------------------------------------------------------------ ?? NOTIFY sip:1002 at 192.168.178.26:58881;ob SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKgc54SgK51KQDS ?? Route: ;ob ?? Max-Forwards: 70 ?? From: ;tag=6v41eyBXmt48a ?? To: ?? Call-ID: f19dfbce-209a-1230-8985-00163e6bb553 ?? CSeq: 28614976 NOTIFY ?? Contact: ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? 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=noresource ?? Content-Type: application/simple-message-summary ?? Content-Length: 91 ?? ?? Messages-Waiting: yes ?? Message-Account: sip:1002 at 78.138.90.58 ?? Voice-Message: 1/0 (0/0) ?? ?? ------------------------------------------------------------------------ recv 308 bytes from udp/[41.34.123.243]:58881 at 23:28:01.128279: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;received=78.138.90.58;branch=z9hG4bKgc54SgK51KQDS ?? Call-ID: f19dfbce-209a-1230-8985-00163e6bb553 ?? From: ;tag=6v41eyBXmt48a ?? To: ;tag=z9hG4bKgc54SgK51KQDS ?? CSeq: 28614976 NOTIFY ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 913 bytes to udp/[217.74.179.29]:5060 at 23:28:11.885213: ?? ------------------------------------------------------------------------ ?? REGISTER sip:bluesip.net;transport=udp SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKHNyXUB48yvD0m ?? Max-Forwards: 70 ?? From: ;tag=5F739Uep9vaXm ?? To: ?? Call-ID: 2a3e49c0-a5b9-11e1-a7ee-7b1f73a7ffcf ?? CSeq: 28601493 REGISTER ?? Contact: ?? Expires: 3600 ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? Authorization: Digest username="bluesip/salahomran", realm="bluesip.net", nonce="4fbebeca79a5cff7c417463814cb9054d313bcac", algorithm=MD5, uri="sip:bluesip.net;transport=udp", response="bb1babb90f4ea1dc8fbb9aa45d6038c7" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 720 bytes from udp/[217.74.179.29]:5060 at 23:28:11.915764: ?? ------------------------------------------------------------------------ ?? SIP/2.0 401 Unauthorized ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKHNyXUB48yvD0m ?? From: ;tag=5F739Uep9vaXm ?? To: ;tag=0354a2e1b960c9cc2279eca4e5f84e20.3454 ?? Call-ID: 2a3e49c0-a5b9-11e1-a7ee-7b1f73a7ffcf ?? CSeq: 28601493 REGISTER ?? WWW-Authenticate: Digest realm="bluesip.net", nonce="4fbec5365af3a174f1cc0b8cda770c727d25bf71", stale=true ?? Server: Sip EXpress router (0.9.7 (i386/linux)) ?? Content-Length: 0 ?? Warning: 392 217.74.179.29:5060 "Noisy feedback tells:? pid=5462 req_src_ip=78.138.90.58 req_src_port=6090 in_uri=sip:bluesip.net;transport=udp out_uri=sip:bluesip.net;transport=udp via_cnt==1" ?? ?? ------------------------------------------------------------------------ send 913 bytes to udp/[217.74.179.29]:5060 at 23:28:11.915914: ?? ------------------------------------------------------------------------ ?? REGISTER sip:bluesip.net;transport=udp SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKjyQpX6mcv53jg ?? Max-Forwards: 70 ?? From: ;tag=5F739Uep9vaXm ?? To: ?? Call-ID: 2a3e49c0-a5b9-11e1-a7ee-7b1f73a7ffcf ?? CSeq: 28601494 REGISTER ?? Contact: ?? Expires: 3600 ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? Authorization: Digest username="bluesip/salahomran", realm="bluesip.net", nonce="4fbec5365af3a174f1cc0b8cda770c727d25bf71", algorithm=MD5, uri="sip:bluesip.net;transport=udp", response="e260f6dc9b01ce8bea8c5ffb36bd4ddd" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 697 bytes from udp/[217.74.179.29]:5060 at 23:28:11.946325: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;branch=z9hG4bKjyQpX6mcv53jg ?? From: ;tag=5F739Uep9vaXm ?? To: ;tag=0354a2e1b960c9cc2279eca4e5f84e20.4b5c ?? Call-ID: 2a3e49c0-a5b9-11e1-a7ee-7b1f73a7ffcf ?? CSeq: 28601494 REGISTER ?? Contact: ;q=0.5;expires=3600 ?? Server: Sip EXpress router (0.9.7 (i386/linux)) ?? Content-Length: 0 ?? Warning: 392 217.74.179.29:5060 "Noisy feedback tells:? pid=5455 req_src_ip=78.138.90.58 req_src_port=6090 in_uri=sip:bluesip.net;transport=udp out_uri=sip:bluesip.net;transport=udp via_cnt==1" ?? ?? ------------------------------------------------------------------------ recv 564 bytes from udp/[41.34.123.243]:58881 at 23:32:55.930149: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjuk8DZGiaXfN7BmtslNHVjtSg5UXsbHXc ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=Pz.t-0tGXj3saM.ehBiB1EAMmmitze1D ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60934 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 744 bytes to udp/[41.34.123.243]:58881 at 23:32:55.931324: ?? ------------------------------------------------------------------------ ?? SIP/2.0 401 Unauthorized ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjuk8DZGiaXfN7BmtslNHVjtSg5UXsbHXc;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=Pz.t-0tGXj3saM.ehBiB1EAMmmitze1D ?? To: "Sherif 1003" ;tag=75XtgSv0H3tUp ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60934 REGISTER ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? WWW-Authenticate: Digest realm="sip.pcfone.com", nonce="ca1eabee-a5f8-11e1-a840-7b1f73a7ffcf", algorithm=MD5, qop="auth" ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ recv 830 bytes from udp/[41.34.123.243]:58881 at 23:32:56.165624: ?? ------------------------------------------------------------------------ ?? REGISTER sip:sip.pcfone.com:6090 SIP/2.0 ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport;branch=z9hG4bKPjKEQ6xXmyzZxA6vCM88vHNYAOPAjUqFsD ?? Max-Forwards: 70 ?? From: "Sherif 1003" ;tag=Pz.t-0tGXj3saM.ehBiB1EAMmmitze1D ?? To: "Sherif 1003" ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60935 REGISTER ?? User-Agent: Telephone 1.0.2 ?? Contact: "Sherif 1003" ?? Expires: 300 ?? Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS ?? Authorization: Digest username="1002", realm="sip.pcfone.com", nonce="ca1eabee-a5f8-11e1-a840-7b1f73a7ffcf", uri="sip:sip.pcfone.com:6090", response="ff6d12f4ec7abd4b37187222b3af2358", algorithm=MD5, cnonce="sc3AJfhuev18Bs-L-6tORmI.RXlxnPOz", qop=auth, nc=00000001 ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ send 706 bytes to udp/[41.34.123.243]:58881 at 23:32:56.168033: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 192.168.178.26:58881;rport=58881;branch=z9hG4bKPjKEQ6xXmyzZxA6vCM88vHNYAOPAjUqFsD;received=41.34.123.243 ?? From: "Sherif 1003" ;tag=Pz.t-0tGXj3saM.ehBiB1EAMmmitze1D ?? To: "Sherif 1003" ;tag=8eQKjmD4ecHej ?? Call-ID: ohCjnPIlf85DIrNvetdMy.1RQM-rLj4e ?? CSeq: 60935 REGISTER ?? Contact: ;expires=300 ?? Date: Thu, 24 May 2012 23:32:56 GMT ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ?? Supported: timer, precondition, path, replaces ?? Content-Length: 0 ?? ?? ------------------------------------------------------------------------ send 989 bytes to udp/[41.34.123.243]:58881 at 23:32:56.360238: ?? ------------------------------------------------------------------------ ?? NOTIFY sip:1002 at 192.168.178.26:58881;ob SIP/2.0 ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport;branch=z9hG4bKK7gFZ15FSet5B ?? Route: ;ob ?? Max-Forwards: 70 ?? From: ;tag=9QgcmFy7BN70D ?? To: ?? Call-ID: a1be7708-209b-1230-8985-00163e6bb553 ?? CSeq: 28615124 NOTIFY ?? Contact: ?? User-Agent: FreeSWITCH-mod_sofia/1.1.beta1-git-8edfd01 2012-03-24 05-16-47 +0000 ?? 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=noresource ?? Content-Type: application/simple-message-summary ?? Content-Length: 91 ?? ?? Messages-Waiting: yes ?? Message-Account: sip:1002 at 78.138.90.58 ?? Voice-Message: 1/0 (0/0) ?? ?? ------------------------------------------------------------------------ recv 308 bytes from udp/[41.34.123.243]:58881 at 23:32:56.623554: ?? ------------------------------------------------------------------------ ?? SIP/2.0 200 OK ?? Via: SIP/2.0/UDP 78.138.90.58:6090;rport=6090;received=78.138.90.58;branch=z9hG4bKK7gFZ15FSet5B ?? Call-ID: a1be7708-209b-1230-8985-00163e6bb553 ?? From: ;tag=9QgcmFy7BN70D ?? To: ;tag=z9hG4bKK7gFZ15FSet5B ?? CSeq: 28615124 NOTIFY ?? Content-Length:? 0 ?? ?? ------------------------------------------------------------------------ freeswitch at internal> --- On Thu, 5/24/12, Kristian Kielhofner wrote: From: Kristian Kielhofner Subject: Re: [Freeswitch-users] Freeswitch (SBC + Media) does not know that gateway phone is picked up To: "FreeSWITCH Users Help" Date: Thursday, May 24, 2012, 7:51 PM Siptrace and logs please. On Thu, May 24, 2012 at 11:53 AM, Sherif Omran wrote: > > Hi all, > > My topology is as follows: > > Kamailio? -> FS (SBS+Media server) > > I came across an issue with my system as follows.? I have a Hardphone registered. When I do local call inside kamailio, it gets to FS and returns back well and FS understands when I lift the handset. However, I added a gateway (german landline server), when I call my self from another phone, the call gets to FS and then transmits to Kamailio, it rings my extension but when I lift the handset FS does not notice it and keeps ringing. > > Any body has an Idea? Here is my gateway trunk. > > > ??????? > ??????? > ??????? > ??????? > ??????? > ????? > > > thanks in advance > Sherif Omran > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Kristian Kielhofner _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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 at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -----Inline Attachment Follows----- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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 at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -----Inline Attachment Follows----- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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 at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -----Inline Attachment Follows----- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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 at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -----Inline Attachment Follows----- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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 at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -----Inline Attachment Follows----- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120527/da3ab790/attachment-0001.html From Nabble_01394 at slickdeals.endjunk.com Sun May 27 18:11:01 2012 From: Nabble_01394 at slickdeals.endjunk.com (mazilo) Date: Sun, 27 May 2012 07:11:01 -0700 (PDT) Subject: [Freeswitch-users] Facebook... In-Reply-To: References: <4B95889A.1090907@todandlorna.com> <5D0F46F9-5977-4095-AAEB-49CE5A0D0BD6@freeswitch.org> Message-ID: <1338127861000-7578266.post@n2.nabble.com> Any update on this topic? ----- FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3 Watts of electricity. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Facebook-tp4699084p7578266.html Sent from the freeswitch-users mailing list archive at Nabble.com. From bdfoster at endigotech.com Sun May 27 18:36:45 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Sun, 27 May 2012 10:36:45 -0400 Subject: [Freeswitch-users] Facebook... In-Reply-To: <1338127861000-7578266.post@n2.nabble.com> References: <4B95889A.1090907@todandlorna.com> <5D0F46F9-5977-4095-AAEB-49CE5A0D0BD6@freeswitch.org> <1338127861000-7578266.post@n2.nabble.com> Message-ID: Since it's been two years... I'm guessing NO. -BDF On May 27, 2012 10:11 AM, "mazilo" wrote: > Any update on this topic? > > ----- > FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3 > Watts of electricity. > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Facebook-tp4699084p7578266.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120527/21e7089e/attachment.html From rmorin at blie-ent.com Sun May 27 19:24:08 2012 From: rmorin at blie-ent.com (Rob Morin) Date: Sun, 27 May 2012 11:24:08 -0400 Subject: [Freeswitch-users] how to alert existing voice mail In-Reply-To: <1338111125.35732.YahooMailClassic@web110812.mail.gq1.yahoo.com> References: <1338111125.35732.YahooMailClassic@web110812.mail.gq1.yahoo.com> Message-ID: <025501cd3c1c$c3131150$493933f0$@blie-ent.com> To send an email notification, put In the extension configuration. Of course, replace email at gmail.com with the appropriate email address J. Best of luck, Rob From: Sherif Omran [mailto:sherifomran2000 at yahoo.com] Sent: Sunday, May 27, 2012 5:32 AM To: FreeSWITCH Users Help Subject: [Freeswitch-users] how to alert existing voice mail Hello guys, I need to alert existing voice mail by either one of the following ways 1- send email 2- FS server should call every 1 hr to deliver the message any ideas on how to commit it thanks Sherif -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120527/18ff3407/attachment.html From sherifomran2000 at yahoo.com Sun May 27 20:42:29 2012 From: sherifomran2000 at yahoo.com (Sherif Omran) Date: Sun, 27 May 2012 09:42:29 -0700 (PDT) Subject: [Freeswitch-users] how to alert existing voice mail In-Reply-To: <025501cd3c1c$c3131150$493933f0$@blie-ent.com> Message-ID: <1338136949.21616.YahooMailClassic@web110804.mail.gq1.yahoo.com> It does not send emails out in my case. Is there something else to pay attention to? S. --- On Sun, 5/27/12, Rob Morin wrote: From: Rob Morin Subject: Re: [Freeswitch-users] how to alert existing voice mail To: "'FreeSWITCH Users Help'" Date: Sunday, May 27, 2012, 6:24 PM To send an email notification, put ??????In the extension configuration. ?Of course, replace email at gmail.com with the appropriate email address J. ?Best of luck, Rob ?From: Sherif Omran [mailto:sherifomran2000 at yahoo.com] Sent: Sunday, May 27, 2012 5:32 AM To: FreeSWITCH Users Help Subject: [Freeswitch-users] how to alert existing voice mail ?Hello guys, I need to alert existing voice mail by either one of the following ways 1- send email 2- FS server should call every 1 hr to deliver the message any ideas on how to commit it thanks Sherif ? -----Inline Attachment Follows----- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120527/aec44fe9/attachment.html From mario_fs at mgtech.com Sun May 27 22:21:08 2012 From: mario_fs at mgtech.com (Mario G) Date: Sun, 27 May 2012 11:21:08 -0700 Subject: [Freeswitch-users] how to alert existing voice mail In-Reply-To: <1338136949.21616.YahooMailClassic@web110804.mail.gq1.yahoo.com> References: <1338136949.21616.YahooMailClassic@web110804.mail.gq1.yahoo.com> Message-ID: <1A0501C8-8EFE-4B8B-AE5E-740CA35B4D26@mgtech.com> See this wiki I wrote, although it's OSX it should help you get it all working. It has the linux setup followed by what you need in FreeSwitch: http://wiki.freeswitch.org/wiki/Installation_and_Tips_for_Mac_OS_X#Email_Voicemail_to_an_iPhone Mario G On May 27, 2012, at 9:42 AM, Sherif Omran wrote: > It does not send emails out in my case. Is there something else to pay attention to? > S. > > > --- On Sun, 5/27/12, Rob Morin wrote: > > From: Rob Morin > Subject: Re: [Freeswitch-users] how to alert existing voice mail > To: "'FreeSWITCH Users Help'" > Date: Sunday, May 27, 2012, 6:24 PM > > To send an email notification, put > > > > In the extension configuration. > > > Of course, replace email at gmail.com with the appropriate email address J. > > > Best of luck, > Rob > > > From: Sherif Omran [mailto:sherifomran2000 at yahoo.com] > Sent: Sunday, May 27, 2012 5:32 AM > To: FreeSWITCH Users Help > Subject: [Freeswitch-users] how to alert existing voice mail > > > Hello guys, > > I need to alert existing voice mail by either one of the following ways > > 1- send email > 2- FS server should call every 1 hr to deliver the message > > any ideas on how to commit it > > thanks > Sherif > > > > -----Inline Attachment Follows----- > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120527/40cb30a1/attachment-0001.html From drk at drkngs.net Sun May 27 22:11:43 2012 From: drk at drkngs.net (Dave R. Kompel) Date: Sun, 27 May 2012 11:11:43 -0700 Subject: [Freeswitch-users] What is the different between FS linux and MS windows version In-Reply-To: Message-ID: <20120527181143.62d2a620@mail.tritonwest.net> Just to chime in a little. 2008 R2 it will kick ass, or of you are using an earlier version, of windows, at least use an x64 one. You can also use Win7 X64, and should get about the same results. FreeSWITCH will also run fine on windows as a Hyper-V virtual machine, with no timing or other issues. Also the HA stuff works the same, at least on the FS side, and there is no reason that you couldn't write a simple module using mod_managed and C# to integrate to windows clustering. That will give you the IP address fallover from one system to the other builtin. You would just have to subscribe to the event for the cluster node becoming active, and issue the API command to restore the active calls. I haven't tested that yet, but it's on my todo list. --Dave _____ From: Albert Nguyen [mailto:albert_nguyen16 at hotmail.com] To: freeswitch-users at lists.freeswitch.org Sent: Sun, 27 May 2012 06:15:35 -0700 Subject: Re: [Freeswitch-users] What is the different between FS linux and MS windows version Hi Peter, Thanks very much for your info. I'll put it on win 2008 R2. Regards, Al > From: peter.olsson at visionutveckling.se > To: freeswitch-users at lists.freeswitch.org > Date: Sun, 27 May 2012 13:05:03 +0000 > Subject: Re: [Freeswitch-users] What is the different between FS linux and MS windows version > > Hit "send" to soon... :) > > What I meant to say last was that I recommend using Windows 2008 R2, or at least Windows 2008. Microsoft has improved their timing in the kernel, starting with these kernels (and also Vista / Windows 7). > > /Peter > ________________________________ > Fr?n: Peter Olsson > Skickat: den 27 maj 2012 15:02 > Till: FreeSWITCH Users Help > ?mne: RE: [Freeswitch-users] What is the different between FS linux and MS windows version > > I would say it's more or less the same (FS build natively on both platforms). > > I'm using it on both platforms, and it's been working perfect on both. There is a bigger user base on Linux I guess (if that's important), and if you want to run HA-setups, you would need Linux to setup Kamailio etc. > > If you're running Windows, I recommend > > > > ________________________________ > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f?r Albert Nguyen [albert_nguyen16 at hotmail.com] > Skickat: den 27 maj 2012 13:47 > Till: freeswitch-users at lists.freeswitch.org > ?mne: [Freeswitch-users] What is the different between FS linux and MS windows version > > Hi > > I have installed a FS windows version on a DELL R210 server running windows 2003 server. Is there any different between the FS windows and linux version in term of capability and performance? > > Thanks in advance, > > Al > !DSPAM:4fc214f832761559443750! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120527/cf33c2fc/attachment.html From sherifomran2000 at yahoo.com Mon May 28 00:02:48 2012 From: sherifomran2000 at yahoo.com (Sherif Omran) Date: Sun, 27 May 2012 13:02:48 -0700 (PDT) Subject: [Freeswitch-users] how to alert existing voice mail In-Reply-To: <1A0501C8-8EFE-4B8B-AE5E-740CA35B4D26@mgtech.com> Message-ID: <1338148968.15942.YahooMailClassic@web110801.mail.gq1.yahoo.com> Hello Mario Thank you for the wiki. I follwed it and installed postfix. It sends test emails out. I adjusted the profile and the parameters accordingly. However it still does not send emails out. Could you help me trace this issue? Thank you S. --- On Sun, 5/27/12, Mario G wrote: From: Mario G Subject: Re: [Freeswitch-users] how to alert existing voice mail To: "FreeSWITCH Users Help" Date: Sunday, May 27, 2012, 9:21 PM See this wiki I wrote, although it's OSX it should help you get it all working. It has the linux setup followed by what you need in FreeSwitch:http://wiki.freeswitch.org/wiki/Installation_and_Tips_for_Mac_OS_X#Email_Voicemail_to_an_iPhoneMario G On May 27, 2012, at 9:42 AM, Sherif Omran wrote: It does not send emails out in my case. Is there something else to pay attention to? S. --- On Sun, 5/27/12, Rob Morin wrote: From: Rob Morin Subject: Re: [Freeswitch-users] how to alert existing voice mail To: "'FreeSWITCH Users Help'" Date: Sunday, May 27, 2012, 6:24 PM To send an email notification, put ??????In the extension configuration. ? Of course, replace email at gmail.com with the appropriate email address J. ? Best of luck, Rob ? From: Sherif Omran [mailto:sherifomran2000 at yahoo.com] Sent: Sunday, May 27, 2012 5:32 AM To: FreeSWITCH Users Help Subject: [Freeswitch-users] how to alert existing voice mail ? Hello guys, I need to alert existing voice mail by either one of the following ways 1- send email 2- FS server should call every 1 hr to deliver the message any ideas on how to commit it thanks Sherif ? -----Inline Attachment Follows----- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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 at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -----Inline Attachment Follows----- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120527/4c652fb0/attachment.html From sherifomran2000 at yahoo.com Mon May 28 00:15:04 2012 From: sherifomran2000 at yahoo.com (Sherif Omran) Date: Sun, 27 May 2012 13:15:04 -0700 (PDT) Subject: [Freeswitch-users] how to alert existing voice mail In-Reply-To: <1338148968.15942.YahooMailClassic@web110801.mail.gq1.yahoo.com> Message-ID: <1338149704.52346.YahooMailClassic@web110805.mail.gq1.yahoo.com> Mario should i use this parameters in switch.xml as well? thanks Sherif --- On Sun, 5/27/12, Sherif Omran wrote: From: Sherif Omran Subject: Re: [Freeswitch-users] how to alert existing voice mail To: "FreeSWITCH Users Help" Date: Sunday, May 27, 2012, 11:02 PM Hello Mario Thank you for the wiki. I follwed it and installed postfix. It sends test emails out. I adjusted the profile and the parameters accordingly. However it still does not send emails out. Could you help me trace this issue? Thank you S. --- On Sun, 5/27/12, Mario G wrote: From: Mario G Subject: Re: [Freeswitch-users] how to alert existing voice mail To: "FreeSWITCH Users Help" Date: Sunday, May 27, 2012, 9:21 PM See this wiki I wrote, although it's OSX it should help you get it all working. It has the linux setup followed by what you need in FreeSwitch:http://wiki.freeswitch.org/wiki/Installation_and_Tips_for_Mac_OS_X#Email_Voicemail_to_an_iPhoneMario G On May 27, 2012, at 9:42 AM, Sherif Omran wrote: It does not send emails out in my case. Is there something else to pay attention to? S. --- On Sun, 5/27/12, Rob Morin wrote: From: Rob Morin Subject: Re: [Freeswitch-users] how to alert existing voice mail To: "'FreeSWITCH Users Help'" Date: Sunday, May 27, 2012, 6:24 PM To send an email notification, put ??????In the extension configuration. ? Of course, replace email at gmail.com with the appropriate email address J. ? Best of luck, Rob ? From: Sherif Omran [mailto:sherifomran2000 at yahoo.com] Sent: Sunday, May 27, 2012 5:32 AM To: FreeSWITCH Users Help Subject: [Freeswitch-users] how to alert existing voice mail ? Hello guys, I need to alert existing voice mail by either one of the following ways 1- send email 2- FS server should call every 1 hr to deliver the message any ideas on how to commit it thanks Sherif ? -----Inline Attachment Follows----- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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 at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -----Inline Attachment Follows----- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -----Inline Attachment Follows----- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120527/d0b5d3f9/attachment-0001.html From sherifomran2000 at yahoo.com Mon May 28 00:28:33 2012 From: sherifomran2000 at yahoo.com (Sherif Omran) Date: Sun, 27 May 2012 13:28:33 -0700 (PDT) Subject: [Freeswitch-users] how to alert existing voice mail (solved) In-Reply-To: <1338149704.52346.YahooMailClassic@web110805.mail.gq1.yahoo.com> Message-ID: <1338150513.20267.YahooMailClassic@web110808.mail.gq1.yahoo.com> It is now working thank you. It would be nice if? you add this note ??? ??? --- On Sun, 5/27/12, Sherif Omran wrote: From: Sherif Omran Subject: Re: [Freeswitch-users] how to alert existing voice mail To: "FreeSWITCH Users Help" Date: Sunday, May 27, 2012, 11:15 PM Mario should i use this parameters in switch.xml as well? thanks Sherif --- On Sun, 5/27/12, Sherif Omran wrote: From: Sherif Omran Subject: Re: [Freeswitch-users] how to alert existing voice mail To: "FreeSWITCH Users Help" Date: Sunday, May 27, 2012, 11:02 PM Hello Mario Thank you for the wiki. I follwed it and installed postfix. It sends test emails out. I adjusted the profile and the parameters accordingly. However it still does not send emails out. Could you help me trace this issue? Thank you S. --- On Sun, 5/27/12, Mario G wrote: From: Mario G Subject: Re: [Freeswitch-users] how to alert existing voice mail To: "FreeSWITCH Users Help" Date: Sunday, May 27, 2012, 9:21 PM See this wiki I wrote, although it's OSX it should help you get it all working. It has the linux setup followed by what you need in FreeSwitch:http://wiki.freeswitch.org/wiki/Installation_and_Tips_for_Mac_OS_X#Email_Voicemail_to_an_iPhoneMario G On May 27, 2012, at 9:42 AM, Sherif Omran wrote: It does not send emails out in my case. Is there something else to pay attention to? S. --- On Sun, 5/27/12, Rob Morin wrote: From: Rob Morin Subject: Re: [Freeswitch-users] how to alert existing voice mail To: "'FreeSWITCH Users Help'" Date: Sunday, May 27, 2012, 6:24 PM To send an email notification, put ??????In the extension configuration. ? Of course, replace email at gmail.com with the appropriate email address J. ? Best of luck, Rob ? From: Sherif Omran [mailto:sherifomran2000 at yahoo.com] Sent: Sunday, May 27, 2012 5:32 AM To: FreeSWITCH Users Help Subject: [Freeswitch-users] how to alert existing voice mail ? Hello guys, I need to alert existing voice mail by either one of the following ways 1- send email 2- FS server should call every 1 hr to deliver the message any ideas on how to commit it thanks Sherif ? -----Inline Attachment Follows----- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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 at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -----Inline Attachment Follows----- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -----Inline Attachment Follows----- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -----Inline Attachment Follows----- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120527/d993722c/attachment.html From covici at ccs.covici.com Mon May 28 00:50:04 2012 From: covici at ccs.covici.com (covici at ccs.covici.com) Date: Sun, 27 May 2012 16:50:04 -0400 Subject: [Freeswitch-users] how to alert existing voice mail In-Reply-To: <1338136949.21616.YahooMailClassic@web110804.mail.gq1.yahoo.com> References: <1338136949.21616.YahooMailClassic@web110804.mail.gq1.yahoo.com> Message-ID: <28914.1338151804@ccs.covici.com> You put the variable in the directory entry for the user, not the extension. Sherif Omran wrote: > It does not send emails out in my case. Is there something else to pay attention to? > S. > > > --- On Sun, 5/27/12, Rob Morin wrote: > > From: Rob Morin > Subject: Re: [Freeswitch-users] how to alert existing voice mail > To: "'FreeSWITCH Users Help'" > Date: Sunday, May 27, 2012, 6:24 PM > > To send an email notification, put ??????In the extension configuration. ?Of course, replace email at gmail.com with the appropriate email address J. ?Best of luck, > Rob ?From: Sherif Omran [mailto:sherifomran2000 at yahoo.com] > Sent: Sunday, May 27, 2012 5:32 AM > To: FreeSWITCH Users Help > Subject: [Freeswitch-users] how to alert existing voice mail ?Hello guys, > > I need to alert existing voice mail by either one of the following ways > > 1- send email > 2- FS server should call every 1 hr to deliver the message > > any ideas on how to commit it > > thanks > Sherif ? > -----Inline Attachment Follows----- > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > ---------------------------------------------------- > Alternatives: > > ---------------------------------------------------- > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici at ccs.covici.com From rmorin at blie-ent.com Mon May 28 01:08:52 2012 From: rmorin at blie-ent.com (Rob Morin) Date: Sun, 27 May 2012 17:08:52 -0400 Subject: [Freeswitch-users] how to alert existing voice mail In-Reply-To: <1338149704.52346.YahooMailClassic@web110805.mail.gq1.yahoo.com> References: <1338148968.15942.YahooMailClassic@web110801.mail.gq1.yahoo.com> <1338149704.52346.YahooMailClassic@web110805.mail.gq1.yahoo.com> Message-ID: <02c501cd3c4c$ebc91f60$c35b5e20$@blie-ent.com> What's your OS? There's a problem with some linux OS's (CentOS being one) and sendmail. They'll work from the command line, but not through FS. From: Sherif Omran [mailto:sherifomran2000 at yahoo.com] Sent: Sunday, May 27, 2012 4:15 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] how to alert existing voice mail Mario should i use this parameters in switch.xml as well? thanks Sherif --- On Sun, 5/27/12, Sherif Omran wrote: From: Sherif Omran Subject: Re: [Freeswitch-users] how to alert existing voice mail To: "FreeSWITCH Users Help" Date: Sunday, May 27, 2012, 11:02 PM Hello Mario Thank you for the wiki. I follwed it and installed postfix. It sends test emails out. I adjusted the profile and the parameters accordingly. However it still does not send emails out. Could you help me trace this issue? Thank you S. --- On Sun, 5/27/12, Mario G wrote: From: Mario G Subject: Re: [Freeswitch-users] how to alert existing voice mail To: "FreeSWITCH Users Help" Date: Sunday, May 27, 2012, 9:21 PM See this wiki I wrote, although it's OSX it should help you get it all working. It has the linux setup followed by what you need in FreeSwitch: http://wiki.freeswitch.org/wiki/Installation_and_Tips_for_Mac_OS_X#Email_Voi cemail_to_an_iPhone Mario G On May 27, 2012, at 9:42 AM, Sherif Omran wrote: It does not send emails out in my case. Is there something else to pay attention to? S. --- On Sun, 5/27/12, Rob Morin wrote: From: Rob Morin Subject: Re: [Freeswitch-users] how to alert existing voice mail To: "'FreeSWITCH Users Help'" Date: Sunday, May 27, 2012, 6:24 PM To send an email notification, put In the extension configuration. Of course, replace email at gmail.com with the appropriate email address J. Best of luck, Rob From: Sherif Omran [mailto:sherifomran2000 at yahoo.com] Sent: Sunday, May 27, 2012 5:32 AM To: FreeSWITCH Users Help Subject: [Freeswitch-users] how to alert existing voice mail Hello guys, I need to alert existing voice mail by either one of the following ways 1- send email 2- FS server should call every 1 hr to deliver the message any ideas on how to commit it thanks Sherif -----Inline Attachment Follows----- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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 at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -----Inline Attachment Follows----- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -----Inline Attachment Follows----- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120527/427511f8/attachment-0001.html From sherifomran2000 at yahoo.com Mon May 28 02:08:24 2012 From: sherifomran2000 at yahoo.com (Sherif Omran) Date: Sun, 27 May 2012 15:08:24 -0700 (PDT) Subject: [Freeswitch-users] how to alert existing voice mail In-Reply-To: <02c501cd3c4c$ebc91f60$c35b5e20$@blie-ent.com> Message-ID: <1338156504.19221.YahooMailClassic@web110811.mail.gq1.yahoo.com> Hello guys, It worked as I stated previously but for 1 time only and then it did not work again. My OS is debian linux any idea? regards, Sherif. --- On Mon, 5/28/12, Rob Morin wrote: From: Rob Morin Subject: Re: [Freeswitch-users] how to alert existing voice mail To: "'FreeSWITCH Users Help'" Date: Monday, May 28, 2012, 12:08 AM What?s your OS? There?s a problem with some linux OS?s (CentOS being one) and sendmail. They?ll work from the command line, but not through FS. ?From: Sherif Omran [mailto:sherifomran2000 at yahoo.com] Sent: Sunday, May 27, 2012 4:15 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] how to alert existing voice mail ?Mario should i use this parameters in switch.xml as well???? ??? thanks Sherif --- On Sun, 5/27/12, Sherif Omran wrote: From: Sherif Omran Subject: Re: [Freeswitch-users] how to alert existing voice mail To: "FreeSWITCH Users Help" Date: Sunday, May 27, 2012, 11:02 PMHello Mario Thank you for the wiki. I follwed it and installed postfix. It sends test emails out. I adjusted the profile and the parameters accordingly. However it still does not send emails out. Could you help me trace this issue? Thank you S. --- On Sun, 5/27/12, Mario G wrote: From: Mario G Subject: Re: [Freeswitch-users] how to alert existing voice mail To: "FreeSWITCH Users Help" Date: Sunday, May 27, 2012, 9:21 PMSee this wiki I wrote, although it's OSX it should help you get it all working. It has the linux setup followed by what you need in FreeSwitch:http://wiki.freeswitch.org/wiki/Installation_and_Tips_for_Mac_OS_X#Email_Voicemail_to_an_iPhoneMario G ?On May 27, 2012, at 9:42 AM, Sherif Omran wrote: It does not send emails out in my case. Is there something else to pay attention to? S. --- On Sun, 5/27/12, Rob Morin wrote: From: Rob Morin Subject: Re: [Freeswitch-users] how to alert existing voice mail To: "'FreeSWITCH Users Help'" Date: Sunday, May 27, 2012, 6:24 PMTo send an email notification, put ??????In the extension configuration. ?Of course, replace email at gmail.com with the appropriate email address J.?Best of luck, Rob?From: Sherif Omran [mailto:sherifomran2000 at yahoo.com] Sent: Sunday, May 27, 2012 5:32 AM To: FreeSWITCH Users Help Subject: [Freeswitch-users] how to alert existing voice mail?Hello guys, I need to alert existing voice mail by either one of the following ways 1- send email 2- FS server should call every 1 hr to deliver the message any ideas on how to commit it thanks Sherif? -----Inline Attachment Follows-----_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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 at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ? -----Inline Attachment Follows-----_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -----Inline Attachment Follows-----_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ? -----Inline Attachment Follows----- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120527/e00156c5/attachment.html From avi at avimarcus.net Mon May 28 02:31:56 2012 From: avi at avimarcus.net (Avi Marcus) Date: Mon, 28 May 2012 01:31:56 +0300 Subject: [Freeswitch-users] query on freeswitch In-Reply-To: <1338123550.34364.YahooMailNeo@web120104.mail.ne1.yahoo.com> References: <1338123550.34364.YahooMailNeo@web120104.mail.ne1.yahoo.com> Message-ID: http://wiki.freeswitch.org/wiki/Mod_odbc_query is pretty descriptive. -Avi On Sun, May 27, 2012 at 3:59 PM, Samira Mh wrote: > how to implement mod query odbc on dialplan of freeswitch? > the wiki is colud't help me > plz help? > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120528/0caf6a47/attachment-0001.html From lazyvirus at gmx.com Mon May 28 02:48:38 2012 From: lazyvirus at gmx.com (Bzzz) Date: Mon, 28 May 2012 00:48:38 +0200 Subject: [Freeswitch-users] how to alert existing voice mail In-Reply-To: <02c501cd3c4c$ebc91f60$c35b5e20$@blie-ent.com> References: <1338148968.15942.YahooMailClassic@web110801.mail.gq1.yahoo.com> <1338149704.52346.YahooMailClassic@web110805.mail.gq1.yahoo.com> <02c501cd3c4c$ebc91f60$c35b5e20$@blie-ent.com> Message-ID: <20120528004838.4619c9be@anubis.defcon1> On Sun, 27 May 2012 17:08:52 -0400 "Rob Morin" wrote: > There's a problem with some linux OS's (CentOS > being one) and sendmail. They'll work from the command line, but > not through FS. Hi Rob, Do you know what is the string(d?) format which is sent to the perl mail script? JY -- No matter who you are, some scholar can show you the great idea you had was had by someone before you. From sherifomran2000 at yahoo.com Mon May 28 03:01:01 2012 From: sherifomran2000 at yahoo.com (Sherif Omran) Date: Sun, 27 May 2012 16:01:01 -0700 (PDT) Subject: [Freeswitch-users] how to alert existing voice mail In-Reply-To: <20120528004838.4619c9be@anubis.defcon1> Message-ID: <1338159661.29137.YahooMailClassic@web110810.mail.gq1.yahoo.com> Hello guys, I get the following in the debug but the mail will not be sent 2012-05-27 22:52:37.470744 [DEBUG] mod_voicemail.c:2654 Deliver VM to 1000 at xxx.xxx.xxx.xxx 2012-05-27 22:52:37.540746 [DEBUG] switch_utils.c:767 Emailed file [/tmp/mail.13381591576764] to [email at server.com] 2012-05-27 22:52:37.540746 [DEBUG] mod_voicemail.c:2844 Sending message to email at server.com Any idea? /tmp/ does not have any files inside thanks S. --- On Mon, 5/28/12, Bzzz wrote: From: Bzzz Subject: Re: [Freeswitch-users] how to alert existing voice mail To: freeswitch-users at lists.freeswitch.org Date: Monday, May 28, 2012, 1:48 AM On Sun, 27 May 2012 17:08:52 -0400 "Rob Morin" wrote: >? There's a problem with some linux OS's (CentOS > being one) and sendmail. They'll work from the command line, but > not through FS. Hi Rob, Do you know what is the string(d?) format which is sent to the perl mail script? JY -- No matter who you are, some scholar can show you the great idea you had was had by someone before you. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120527/7e33772f/attachment.html From sherifomran2000 at yahoo.com Mon May 28 03:14:05 2012 From: sherifomran2000 at yahoo.com (Sherif Omran) Date: Sun, 27 May 2012 16:14:05 -0700 (PDT) Subject: [Freeswitch-users] how to alert existing voice mail (solved) In-Reply-To: <1338159661.29137.YahooMailClassic@web110810.mail.gq1.yahoo.com> Message-ID: <1338160445.1063.YahooMailClassic@web110801.mail.gq1.yahoo.com> I used the -t parameter instead of the -d and it worked well ??? ??? For error debugging I checked /var/log/mail.log and I saw the following errors May 27 22:52:37 sip postfix/sendmail[18911]: fatal: Recipient addresses must be specified on the command line or via the -t option May 27 22:52:37 sip postfix/postdrop[18912]: warning: stdin: unexpected EOF in data, record type 78 length 72 May 27 22:52:37 sip postfix/postdrop[18912]: fatal: uid=1000: malformed input Thanks S. --- On Mon, 5/28/12, Sherif Omran wrote: From: Sherif Omran Subject: Re: [Freeswitch-users] how to alert existing voice mail To: "FreeSWITCH Users Help" Date: Monday, May 28, 2012, 2:01 AM Hello guys, I get the following in the debug but the mail will not be sent 2012-05-27 22:52:37.470744 [DEBUG] mod_voicemail.c:2654 Deliver VM to 1000 at xxx.xxx.xxx.xxx 2012-05-27 22:52:37.540746 [DEBUG] switch_utils.c:767 Emailed file [/tmp/mail.13381591576764] to [email at server.com] 2012-05-27 22:52:37.540746 [DEBUG] mod_voicemail.c:2844 Sending message to email at server.com Any idea? /tmp/ does not have any files inside thanks S. --- On Mon, 5/28/12, Bzzz wrote: From: Bzzz Subject: Re: [Freeswitch-users] how to alert existing voice mail To: freeswitch-users at lists.freeswitch.org Date: Monday, May 28, 2012, 1:48 AM On Sun, 27 May 2012 17:08:52 -0400 "Rob Morin" wrote: >? There's a problem with some linux OS's (CentOS > being one) and sendmail. They'll work from the command line, but > not through FS. Hi Rob, Do you know what is the string(d?) format which is sent to the perl mail script? JY -- No matter who you are, some scholar can show you the great idea you had was had by someone before you. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -----Inline Attachment Follows----- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120527/dd49c0f7/attachment.html From lazyvirus at gmx.com Mon May 28 03:26:22 2012 From: lazyvirus at gmx.com (Bzzz) Date: Mon, 28 May 2012 01:26:22 +0200 Subject: [Freeswitch-users] how to alert existing voice mail In-Reply-To: <1338159661.29137.YahooMailClassic@web110810.mail.gq1.yahoo.com> References: <20120528004838.4619c9be@anubis.defcon1> <1338159661.29137.YahooMailClassic@web110810.mail.gq1.yahoo.com> Message-ID: <20120528012622.5e21ee62@anubis.defcon1> On Sun, 27 May 2012 16:01:01 -0700 (PDT) Sherif Omran wrote: > > Any idea? /tmp/ does not have any files inside As its name suggest, content of /tmp is often fugitive. If you want to get the message, use a script such as: #!/bin/sh for i in `seq 1 50000`; do echo "----------------------------------------------------------------------------------------------------------------" echo "----------------------------------------------------------------------------------------------------------------" cp /tmp/mail.* /whatever/ echo "num = $i " echo "----------------------------------------------------------------------------------------------------------------" echo "----------------------------------------------------------------------------------------------------------------" done Echos are only there to speed down the loop. JY -- I'm an evolutionist; I refuse to believe that I could have been created by man. From mario_fs at mgtech.com Mon May 28 04:34:48 2012 From: mario_fs at mgtech.com (Mario G) Date: Sun, 27 May 2012 17:34:48 -0700 Subject: [Freeswitch-users] how to alert existing voice mail (solved) In-Reply-To: <1338150513.20267.YahooMailClassic@web110808.mail.gq1.yahoo.com> References: <1338150513.20267.YahooMailClassic@web110808.mail.gq1.yahoo.com> Message-ID: <7E30DC20-FDC3-433F-8BEA-BF8DD8778B22@mgtech.com> That wiki is for Mac OSX so the info you mentioned below does not apply to that wiki. You are native Linux (I think). Mario G On May 27, 2012, at 1:28 PM, Sherif Omran wrote: > It is now working thank you. It would be nice if you add this note > > > > > > --- On Sun, 5/27/12, Sherif Omran wrote: > > From: Sherif Omran > Subject: Re: [Freeswitch-users] how to alert existing voice mail > To: "FreeSWITCH Users Help" > Date: Sunday, May 27, 2012, 11:15 PM > > Mario > > should i use this parameters in switch.xml as well? > > > > > > thanks > Sherif > > > > --- On Sun, 5/27/12, Sherif Omran wrote: > > From: Sherif Omran > Subject: Re: [Freeswitch-users] how to alert existing voice mail > To: "FreeSWITCH Users Help" > Date: Sunday, May 27, 2012, 11:02 PM > > Hello Mario > > Thank you for the wiki. I follwed it and installed postfix. It sends test emails out. > I adjusted the profile and the parameters accordingly. However it still does not send emails out. > > Could you help me trace this issue? > > Thank you > > S. > > --- On Sun, 5/27/12, Mario G wrote: > > From: Mario G > Subject: Re: [Freeswitch-users] how to alert existing voice mail > To: "FreeSWITCH Users Help" > Date: Sunday, May 27, 2012, 9:21 PM > > See this wiki I wrote, although it's OSX it should help you get it all working. It has the linux setup followed by what you need in FreeSwitch: > http://wiki.freeswitch.org/wiki/Installation_and_Tips_for_Mac_OS_X#Email_Voicemail_to_an_iPhone > Mario G > > On May 27, 2012, at 9:42 AM, Sherif Omran wrote: > >> It does not send emails out in my case. Is there something else to pay attention to? >> S. >> >> >> --- On Sun, 5/27/12, Rob Morin wrote: >> >> From: Rob Morin >> Subject: Re: [Freeswitch-users] how to alert existing voice mail >> To: "'FreeSWITCH Users Help'" >> Date: Sunday, May 27, 2012, 6:24 PM >> >> To send an email notification, put >> >> In the extension configuration. >> >> Of course, replace email at gmail.com with the appropriate email address J. >> >> Best of luck, >> Rob >> >> From: Sherif Omran [mailto:sherifomran2000 at yahoo.com] >> Sent: Sunday, May 27, 2012 5:32 AM >> To: FreeSWITCH Users Help >> Subject: [Freeswitch-users] how to alert existing voice mail >> >> Hello guys, >> >> I need to alert existing voice mail by either one of the following ways >> >> 1- send email >> 2- FS server should call every 1 hr to deliver the message >> >> any ideas on how to commit it >> >> thanks >> Sherif >> >> >> >> -----Inline Attachment Follows----- >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > -----Inline Attachment Follows----- > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -----Inline Attachment Follows----- > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -----Inline Attachment Follows----- > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120527/1edfc978/attachment-0001.html From mario_fs at mgtech.com Mon May 28 05:09:12 2012 From: mario_fs at mgtech.com (Mario G) Date: Sun, 27 May 2012 18:09:12 -0700 Subject: [Freeswitch-users] how to alert existing voice mail (solved) In-Reply-To: <1338160445.1063.YahooMailClassic@web110801.mail.gq1.yahoo.com> References: <1338160445.1063.YahooMailClassic@web110801.mail.gq1.yahoo.com> Message-ID: Another thing, "sendmail" and "-t" are the defaults in switch.conf so normally not needed to be changed. If you had /usr/sbin in the linux path then you should not have needed to change the lines below. Mario G On May 27, 2012, at 4:14 PM, Sherif Omran wrote: > I used the -t parameter instead of the -d and it worked well > > > > > For error debugging I checked /var/log/mail.log and I saw the following errors > > May 27 22:52:37 sip postfix/sendmail[18911]: fatal: Recipient addresses must be specified on the command line or via the -t option > May 27 22:52:37 sip postfix/postdrop[18912]: warning: stdin: unexpected EOF in data, record type 78 length 72 > May 27 22:52:37 sip postfix/postdrop[18912]: fatal: uid=1000: malformed input > > > Thanks > S. > > > --- On Mon, 5/28/12, Sherif Omran wrote: > > From: Sherif Omran > Subject: Re: [Freeswitch-users] how to alert existing voice mail > To: "FreeSWITCH Users Help" > Date: Monday, May 28, 2012, 2:01 AM > > Hello guys, > > I get the following in the debug but the mail will not be sent > > > 2012-05-27 22:52:37.470744 [DEBUG] mod_voicemail.c:2654 Deliver VM to 1000 at xxx.xxx.xxx.xxx > 2012-05-27 22:52:37.540746 [DEBUG] switch_utils.c:767 Emailed file [/tmp/mail.13381591576764] to [email at server.com] > 2012-05-27 22:52:37.540746 [DEBUG] mod_voicemail.c:2844 Sending message to email at server.com > > > Any idea? /tmp/ does not have any files inside > > thanks > S. > > --- On Mon, 5/28/12, Bzzz wrote: > > From: Bzzz > Subject: Re: [Freeswitch-users] how to alert existing voice mail > To: freeswitch-users at lists.freeswitch.org > Date: Monday, May 28, 2012, 1:48 AM > > On Sun, 27 May 2012 17:08:52 -0400 > "Rob Morin" wrote: > > > There's a problem with some linux OS's (CentOS > > being one) and sendmail. They'll work from the command line, but > > not through FS. > > Hi Rob, > > Do you know what is the string(d?) format which is sent to the perl > mail script? > > JY > -- > No matter who you are, some scholar can show you the great idea you > had was had by someone before you. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -----Inline Attachment Follows----- > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120527/839e668d/attachment.html From curriegrad2004 at gmail.com Mon May 28 05:40:12 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Sun, 27 May 2012 18:40:12 -0700 Subject: [Freeswitch-users] freeswitch - sangoma CAS T1 - a101 configuration In-Reply-To: References: Message-ID: Sangoma's Wiki has all of the necessary information you're looking for. If you're doing CAS for voice, it's going to be under RBS. On Fri, May 25, 2012 at 10:29 PM, copycall wrote: > hi, > > i'm having trouble getting instructions to configure freeswitch with a > sangoma a101 T1 card for CAS, B8ZS, ESF. > > the first application i seek to solve is using this setup as a gateway > (replacing a dialogic dmg 2000) for sip from my itsp to the tdm endpoint: a > legacy centigram - mitel voicemail server. > > any help will be greatly appreciated. > > thank you, > dave > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From albert_nguyen16 at hotmail.com Mon May 28 08:08:55 2012 From: albert_nguyen16 at hotmail.com (Albert Nguyen) Date: Mon, 28 May 2012 04:08:55 +0000 Subject: [Freeswitch-users] What is the different between FS linux and MS windows version In-Reply-To: <20120527181143.62d2a620@mail.tritonwest.net> References: , <20120527181143.62d2a620@mail.tritonwest.net> Message-ID: Thanks Dave, I'll do a load test on both to check it out. Al From: drk at drkngs.net To: freeswitch-users at lists.freeswitch.org Date: Sun, 27 May 2012 11:11:43 -0700 Subject: Re: [Freeswitch-users] What is the different between FS linux and MS windows version Just to chime in a little. 2008 R2 it will kick ass, or of you are using an earlier version, of windows, at least use an x64 one. You can also use Win7 X64, and should get about the same results. FreeSWITCH will also run fine on windows as a Hyper-V virtual machine, with no timing or other issues. Also the HA stuff works the same, at least on the FS side, and there is no reason that you couldn't write a simple module using mod_managed and C# to integrate to windows clustering. That will give you the IP address fallover from one system to the other builtin. You would just have to subscribe to the event for the cluster node becoming active, and issue the API command to restore the active calls. I haven't tested that yet, but it's on my todo list. --Dave From: Albert Nguyen [mailto:albert_nguyen16 at hotmail.com] To: freeswitch-users at lists.freeswitch.org Sent: Sun, 27 May 2012 06:15:35 -0700 Subject: Re: [Freeswitch-users] What is the different between FS linux and MS windows version Hi Peter, Thanks very much for your info. I'll put it on win 2008 R2. Regards, Al > From: peter.olsson at visionutveckling.se > To: freeswitch-users at lists.freeswitch.org > Date: Sun, 27 May 2012 13:05:03 +0000 > Subject: Re: [Freeswitch-users] What is the different between FS linux and MS windows version > > Hit "send" to soon... :) > > What I meant to say last was that I recommend using Windows 2008 R2, or at least Windows 2008. Microsoft has improved their timing in the kernel, starting with these kernels (and also Vista / Windows 7). > > /Peter > ________________________________ > Fr?n: Peter Olsson > Skickat: den 27 maj 2012 15:02 > Till: FreeSWITCH Users Help > ?mne: RE: [Freeswitch-users] What is the different between FS linux and MS windows version > > I would say it's more or less the same (FS build natively on both platforms). > > I'm using it on both platforms, and it's been working perfect on both. There is a bigger user base on Linux I guess (if that's important), and if you want to run HA-setups, you would need Linux to setup Kamailio etc. > > If you're running Windows, I recommend > > > > ________________________________ > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f?r Albert Nguyen [albert_nguyen16 at hotmail.com] > Skickat: den 27 maj 2012 13:47 > Till: freeswitch-users at lists.freeswitch.org > ?mne: [Freeswitch-users] What is the different between FS linux and MS windows version > > Hi > > I have installed a FS windows version on a DELL R210 server running windows 2003 server. Is there any different between the FS windows and linux version in term of capability and performance? > > Thanks in advance, > > Al > !DSPAM:4fc214f832761559443750! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120528/c0ef0758/attachment-0001.html From chris at gonumina.com Mon May 28 09:45:31 2012 From: chris at gonumina.com (Chris Ferreira) Date: Sun, 27 May 2012 23:45:31 -0600 Subject: [Freeswitch-users] What is the different between FS linux and MS windows version In-Reply-To: <20120527181143.62d2a620@mail.tritonwest.net> References: <20120527181143.62d2a620@mail.tritonwest.net> Message-ID: <3820475734403626844@unknownmsgid> Great info Dave, much appreciated. As far as timing goes with Hyper-V, have you messed at all with T.38 or any faxing stuff? Thanks, -Chris ___________________ Mobile Reply On May 27, 2012, at 12:33 PM, "Dave R. Kompel" wrote: Just to chime in a little. 2008 R2 it will kick ass, or of you are using an earlier version, of windows, at least use an x64 one. You can also use Win7 X64, and should get about the same results. FreeSWITCH will also run fine on windows as a Hyper-V virtual machine, with no timing or other issues. Also the HA stuff works the same, at least on the FS side, and there is no reason that you couldn't write a simple module using mod_managed and C# to integrate to windows clustering. That will give you the IP address fallover from one system to the other builtin. You would just have to subscribe to the event for the cluster node becoming active, and issue the API command to restore the active calls. I haven't tested that yet, but it's on my todo list. --Dave ------------------------------ *From:* Albert Nguyen [mailto:albert_nguyen16 at hotmail.com] *To:* freeswitch-users at lists.freeswitch.org *Sent:* Sun, 27 May 2012 06:15:35 -0700 *Subject:* Re: [Freeswitch-users] What is the different between FS linux and MS windows version Hi Peter, Thanks very much for your info. I'll put it on win 2008 R2. Regards, Al > From: peter.olsson at visionutveckling.se > To: freeswitch-users at lists.freeswitch.org > Date: Sun, 27 May 2012 13:05:03 +0000 > Subject: Re: [Freeswitch-users] What is the different between FS linux and MS windows version > > Hit "send" to soon... :) > > What I meant to say last was that I recommend using Windows 2008 R2, or at least Windows 2008. Microsoft has improved their timing in the kernel, starting with these kernels (and also Vista / Windows 7). > > /Peter > ________________________________ > Fr?n: Peter Olsson > Skickat: den 27 maj 2012 15:02 > Till: FreeSWITCH Users Help > ?mne: RE: [Freeswitch-users] What is the different between FS linux and MS windows version > > I would say it's more or less the same (FS build natively on both platforms). > > I'm using it on both platforms, and it's been working perfect on both. There is a bigger user base on Linux I guess (if that's important), and if you want to run HA-setups, you would need Linux to setup Kamailio etc. > > If you're running Windows, I recommend > > > > ________________________________ > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [ freeswitch-users-bounces at lists.freeswitch.org] f?r Albert Nguyen [ albert_nguyen16 at hotmail.com] > Skickat: den 27 maj 2012 13:47 > Till: freeswitch-users at lists.freeswitch.org > ?mne: [Freeswitch-users] What is the different between FS linux and MS windows version > > Hi > > I have installed a FS windows version on a DELL R210 server running windows 2003 server. Is there any different between the FS windows and linux version in term of capability and performance? > > Thanks in advance, > > Al > !DSPAM:4fc214f832761559443750! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120527/ceb2af5a/attachment.html From dave at copycall.com Mon May 28 08:49:56 2012 From: dave at copycall.com (copycall) Date: Sun, 27 May 2012 21:49:56 -0700 Subject: [Freeswitch-users] freeswitch - sangoma CAS T1 - a101 configuration In-Reply-To: References: Message-ID: curriegrad2004, thanks for responding. unfortunately, the sangoma tech support has not been as crack as usual. i can go into the specifics, but it probably isn't necessary in a public forum. i'm not looking to piss anyone off. if you have you actually configured a a101 sangoma card for a legacy CAS T1 card, i would greatly appreciate your assistance. thank you, dave cook On Sun, May 27, 2012 at 6:40 PM, curriegrad2004 wrote: > Sangoma's Wiki has all of the necessary information you're looking > for. If you're doing CAS for voice, it's going to be under RBS. > > On Fri, May 25, 2012 at 10:29 PM, copycall wrote: > > hi, > > > > i'm having trouble getting instructions to configure freeswitch with a > > sangoma a101 T1 card for CAS, B8ZS, ESF. > > > > the first application i seek to solve is using this setup as a gateway > > (replacing a dialogic dmg 2000) for sip from my itsp to the tdm > endpoint: a > > legacy centigram - mitel voicemail server. > > > > any help will be greatly appreciated. > > > > thank you, > > dave > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120527/62015d85/attachment.html From sherifomran2000 at yahoo.com Mon May 28 10:37:11 2012 From: sherifomran2000 at yahoo.com (Sherif Omran) Date: Sun, 27 May 2012 23:37:11 -0700 (PDT) Subject: [Freeswitch-users] how to alert existing voice mail (solved) In-Reply-To: <7E30DC20-FDC3-433F-8BEA-BF8DD8778B22@mgtech.com> Message-ID: <1338187031.89546.YahooMailClassic@web110814.mail.gq1.yahoo.com> Thank you Guys for your help. It is now working as a charm. The only point that should be fixed is the from header From: "${voicemail_caller_id_name}" -> Voicemail_caller_id_name is the name used in sip registration. However, I need to use the name existing in the database. Any Idea how to fetch it using an external script from inside the tpl file? thanks Sherif --- On Mon, 5/28/12, Mario G wrote: From: Mario G Subject: Re: [Freeswitch-users] how to alert existing voice mail (solved) To: "FreeSWITCH Users Help" Date: Monday, May 28, 2012, 3:34 AM That wiki is for Mac OSX so the info you mentioned below does not apply to that wiki. You are native Linux (I think).Mario G On May 27, 2012, at 1:28 PM, Sherif Omran wrote: It is now working thank you. It would be nice if? you add this note ??? ??? --- On Sun, 5/27/12, Sherif Omran wrote: From: Sherif Omran Subject: Re: [Freeswitch-users] how to alert existing voice mail To: "FreeSWITCH Users Help" Date: Sunday, May 27, 2012, 11:15 PM Mario should i use this parameters in switch.xml as well? thanks Sherif --- On Sun, 5/27/12, Sherif Omran wrote: From: Sherif Omran Subject: Re: [Freeswitch-users] how to alert existing voice mail To: "FreeSWITCH Users Help" Date: Sunday, May 27, 2012, 11:02 PM Hello Mario Thank you for the wiki. I follwed it and installed postfix. It sends test emails out. I adjusted the profile and the parameters accordingly. However it still does not send emails out. Could you help me trace this issue? Thank you S. --- On Sun, 5/27/12, Mario G wrote: From: Mario G Subject: Re: [Freeswitch-users] how to alert existing voice mail To: "FreeSWITCH Users Help" Date: Sunday, May 27, 2012, 9:21 PM See this wiki I wrote, although it's OSX it should help you get it all working. It has the linux setup followed by what you need in FreeSwitch:http://wiki.freeswitch.org/wiki/Installation_and_Tips_for_Mac_OS_X#Email_Voicemail_to_an_iPhoneMario G On May 27, 2012, at 9:42 AM, Sherif Omran wrote: It does not send emails out in my case. Is there something else to pay attention to? S. --- On Sun, 5/27/12, Rob Morin wrote: From: Rob Morin Subject: Re: [Freeswitch-users] how to alert existing voice mail To: "'FreeSWITCH Users Help'" Date: Sunday, May 27, 2012, 6:24 PM To send an email notification, put ??????In the extension configuration. ? Of course, replace email at gmail.com with the appropriate email address J. ? Best of luck, Rob ? From: Sherif Omran [mailto:sherifomran2000 at yahoo.com] Sent: Sunday, May 27, 2012 5:32 AM To: FreeSWITCH Users Help Subject: [Freeswitch-users] how to alert existing voice mail ? Hello guys, I need to alert existing voice mail by either one of the following ways 1- send email 2- FS server should call every 1 hr to deliver the message any ideas on how to commit it thanks Sherif ? -----Inline Attachment Follows----- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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 at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -----Inline Attachment Follows----- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -----Inline Attachment Follows----- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -----Inline Attachment Follows----- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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 at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -----Inline Attachment Follows----- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120527/67d6a1c7/attachment-0001.html From andy at fabulous4.co.uk Mon May 28 13:30:53 2012 From: andy at fabulous4.co.uk (Andy Ayers) Date: Mon, 28 May 2012 10:30:53 +0100 Subject: [Freeswitch-users] Best database setup for high volume In-Reply-To: References: <005401cd38be$2a9434b0$7fbc9e10$@fabulous4.co.uk> <01ef01cd390a$884b60f0$98e222d0$@fabulous4.co.uk> <008e01cd3990$0cefac00$26cf0400$@fabulous4.co.uk> Message-ID: <00c901cd3cb4$94775200$bd65f600$@fabulous4.co.uk> Disk at the moment. Is it easy to move it to memory? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Wesley Akio Sent: 24 May 2012 12:58 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Best database setup for high volume Out of curiosity, do you run sqlite from memory or disk? Sent from mobile, sorry for the typos.... Em 24/05/2012 06:33, "Andy Ayers" escreveu: Hi Avi, Yes sadly I've been through all those options and am still getting the error. Tried OPTION and OPTIONS but it doesn't seem to make any difference. There seem to be a few other folks on the users list that have had the same problem and not managed to find a solution. Cheers Andy From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi Marcus Sent: 23 May 2012 19:10 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Best database setup for high volume On Wed, May 23, 2012 at 8:36 PM, Andy Ayers wrote: Many thanks for your reply Avi, that's very helpful. I've disabled the sql scoreboard and presence and all seems to be ok so those should help. My traffic consists of relatively high volumes of incoming and outgoing calls but all one sided. i.e. It's the switch taking the caller through an ivr 'form' so there are no 2-way calls, no bridging, forwarding or directing of calls at all. The only time multiple callers are involved is when we use it for conferencing which is only small scale at the moment. It's the database corruption issue I'm really interested in solving so I'd like to get the odbc connection working if possible. Any info you can provide on how you got that to work would be greatly appreciated. Like I say I hit 2 problems: Firstly on load freeswitch complains that it can't run multiple statements. I've tried everything that's recommended in the MyOdbc docs including setting the options in odbc.ini but still get the error. Some posts talk about needing to use to _r version of the driver but I don't have that on my system. I'm running Debian if that's significant. Did you set: OPTIONS = 67108864 in the odbc.ini, for FLAG_MULTI_STATEMENTS? (Supposedly it's "OPTION" on centos) http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core -Avi The second issue was a message popping up in the logs every few seconds saying: Error in my_thread_global_end() nn threads didn't exit. Did you encounter either of these problems or find ways round them? Once again many thanks for any help. Cheers Andy From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi Marcus Sent: 23 May 2012 09:52 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Best database setup for high volume Is your issue the CDRs & Voicemail or session count, current calls, recovery data.. You can make sure track-calls is off... -nosql -- disable internal sql scoreboard I'm not sure if that kills presence or not. Disable presence if you don't need it - it's a real usage hog. I have odbc to mysql but I made calls,channels,sip_dialogs, sip_subscriptions, etc into memory tables a few months ago. (I left sip_registrations as non-memory for persistence of a sort) -Avi On Wed, May 23, 2012 at 11:29 AM, Andy Ayers wrote: Hi, Can anyone tell me what the best database setup is for dealing with high call volumes? The background: I've been running with the standard SQLite system for about 3 years without issue but recently am getting a lot of database corruption errors ('database disk image is malformed'). Easily solved by deleting the db and restarting but on occasion it brings my switch down. I've tried upgrading to odbc and mysql but hit 2 problems: FLAG_MULTIPLE_STATEMENTS Error in my_thread_global_end() Both of which are mentioned in the user groups but not with any solutions that worked. I've just upgraded to version 1.2 so this may solve the corruption problems but would really like to get my system set up to handle as much traffic as possible. Any advice or suggestions much appreciated. Kind regards Andy _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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 at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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 at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120528/bd001a33/attachment-0001.html From sherifomran2000 at yahoo.com Mon May 28 13:51:48 2012 From: sherifomran2000 at yahoo.com (Sherif Omran) Date: Mon, 28 May 2012 02:51:48 -0700 (PDT) Subject: [Freeswitch-users] how to schedule calls In-Reply-To: <1338187031.89546.YahooMailClassic@web110814.mail.gq1.yahoo.com> Message-ID: <1338198708.69195.YahooMailClassic@web110807.mail.gq1.yahoo.com> Hi guys, Any body knows how to schedule calls depending on some internal variables. Such as at 8pm, users with voice mail not yet delivered receive a call. Thanks S. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120528/a99a0b50/attachment.html From nbhatti at gmail.com Mon May 28 14:09:48 2012 From: nbhatti at gmail.com (Muhammad Naseer Bhatti) Date: Mon, 28 May 2012 13:09:48 +0300 Subject: [Freeswitch-users] Gateway failover with LUA Message-ID: Hi, I am trying to implement a gateway failover functionality with lua. Right now I am using limit and using hash backend to limit number of channels to each gateway. Since I am forwarding all calls with a default dialplan which is sending all calls to application lua and the bridge is done outside lua setting all the necessary vars. The idea is if all channels of first gateway are occupied, send the call to second gateway in the list and so on. Any idea how to implement this with lua? Can not seem to find the logic. If I query with limit_status it is going to be a query on every call. In a high call volume scenario, I don't think this would be really a good choice. Any ideas, thoughts? Thanks. From wesleyakio at tuntscorp.com Mon May 28 16:41:58 2012 From: wesleyakio at tuntscorp.com (Wesley Akio) Date: Mon, 28 May 2012 09:41:58 -0300 Subject: [Freeswitch-users] Best database setup for high volume In-Reply-To: <00c901cd3cb4$94775200$bd65f600$@fabulous4.co.uk> References: <005401cd38be$2a9434b0$7fbc9e10$@fabulous4.co.uk> <01ef01cd390a$884b60f0$98e222d0$@fabulous4.co.uk> <008e01cd3990$0cefac00$26cf0400$@fabulous4.co.uk> <00c901cd3cb4$94775200$bd65f600$@fabulous4.co.uk> Message-ID: Pretty easy, just format a ramdisk and mount it... Will send you my init script as soon as I get to the office... Sent from mobile, sorry for the typos.... Em 28/05/2012 06:32, "Andy Ayers" escreveu: > Disk at the moment. Is it easy to move it to memory?**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Wesley Akio > *Sent:* 24 May 2012 12:58 > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Best database setup for high volume**** > > ** ** > > Out of curiosity, do you run sqlite from memory or disk?**** > > Sent from mobile, sorry for the typos....**** > > Em 24/05/2012 06:33, "Andy Ayers" escreveu:**** > > Hi Avi,**** > > **** > > Yes sadly I?ve been through all those options and am still getting the > error. Tried OPTION and OPTIONS but it doesn?t seem to make any difference. > There seem to be a few other folks on the users list that have had the same > problem and not managed to find a solution.**** > > **** > > Cheers**** > > Andy**** > > **** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Avi Marcus > *Sent:* 23 May 2012 19:10 > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Best database setup for high volume**** > > **** > > On Wed, May 23, 2012 at 8:36 PM, Andy Ayers wrote:* > *** > > Many thanks for your reply Avi, that?s very helpful.**** > > **** > > I?ve disabled the sql scoreboard and presence and all seems to be ok so > those should help.**** > > **** > > My traffic consists of relatively high volumes of incoming and outgoing > calls but all one sided. i.e. It?s the switch taking the caller through an > ivr ?form? so there are no 2-way calls, no bridging, forwarding or > directing of calls at all. The only time multiple callers are involved is > when we use it for conferencing which is only small scale at the moment.** > ** > > **** > > It?s the database corruption issue I?m really interested in solving so I?d > like to get the odbc connection working if possible. Any info you can > provide on how you got that to work would be greatly appreciated. Like I > say I hit 2 problems:**** > > **** > > Firstly on load freeswitch complains that it can?t run multiple > statements. I?ve tried everything that?s recommended in the MyOdbc docs > including setting the options in odbc.ini but still get the error. Some > posts talk about needing to use to _r version of the driver but I don?t > have that on my system. I?m running Debian if that?s significant.**** > > Did you set:**** > > OPTIONS = 67108864**** > > in the odbc.ini, for FLAG_MULTI_STATEMENTS? (Supposedly it's "OPTION" on > centos)**** > > http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core**** > > **** > > -Avi**** > > **** > > The second issue was a message popping up in the logs every few seconds > saying: Error in my_thread_global_end() nn threads didn?t exit.**** > > Did you encounter either of these problems or find ways round them?**** > > **** > > Once again many thanks for any help.**** > > **** > > Cheers**** > > Andy**** > > **** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Avi Marcus > *Sent:* 23 May 2012 09:52 > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Best database setup for high volume**** > > **** > > Is your issue the CDRs & Voicemail or session count, current calls, > recovery data..**** > > You can make sure track-calls is off... **** > > **** > > **** > > **** > > **** > > -nosql -- disable internal sql scoreboard**** > > I'm not sure if that kills presence or not.**** > > **** > > Disable presence if you don't need it - it's a real usage hog.**** > > **** > > I have odbc to mysql but I made calls,channels,sip_dialogs, > sip_subscriptions, etc into memory tables a few months ago.**** > > **** > > (I left sip_registrations as non-memory for persistence of a sort)**** > > **** > > -Avi**** > > **** > > On Wed, May 23, 2012 at 11:29 AM, Andy Ayers wrote: > **** > > Hi,**** > > **** > > Can anyone tell me what the best database setup is for dealing with high > call volumes?**** > > **** > > The background:**** > > **** > > I?ve been running with the standard SQLite system for about 3 years > without issue but recently am getting a lot of database corruption errors > (?database disk image is malformed?). Easily solved by deleting the db and > restarting but on occasion it brings my switch down.**** > > **** > > I?ve tried upgrading to odbc and mysql but hit 2 problems:**** > > **** > > FLAG_MULTIPLE_STATEMENTS**** > > Error in my_thread_global_end()**** > > **** > > Both of which are mentioned in the user groups but not with any solutions > that worked.**** > > **** > > I?ve just upgraded to version 1.2 so this may solve the corruption > problems but would really like to get my system set up to handle as much > traffic as possible.**** > > **** > > Any advice or suggestions much appreciated.**** > > **** > > Kind regards**** > > Andy**** > > **** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120528/25d7cc66/attachment-0001.html From wesleyakio at tuntscorp.com Mon May 28 16:44:43 2012 From: wesleyakio at tuntscorp.com (Wesley Akio) Date: Mon, 28 May 2012 09:44:43 -0300 Subject: [Freeswitch-users] Gateway failover with LUA In-Reply-To: References: Message-ID: As far as you use a decent limit backend I see no problem with querying in every call... Sent from mobile, sorry for the typos.... Em 28/05/2012 07:11, "Muhammad Naseer Bhatti" escreveu: > Hi, I am trying to implement a gateway failover functionality with > lua. Right now I am using limit and using hash backend to limit number > of channels to each gateway. Since I am forwarding all calls with a > default dialplan which is sending all calls to application lua and the > bridge is done outside lua setting all the necessary vars. > The idea is if all channels of first gateway are occupied, send the > call to second gateway in the list and so on. Any idea how to > implement this with lua? Can not seem to find the logic. If I query > with limit_status it is going to be a query on every call. In a high > call volume scenario, I don't think this would be really a good > choice. Any ideas, thoughts? > > Thanks. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120528/4c95b991/attachment.html From nathandownes at hotmail.com Mon May 28 17:50:56 2012 From: nathandownes at hotmail.com (Mr Nathan Downes) Date: Mon, 28 May 2012 23:50:56 +1000 Subject: [Freeswitch-users] RTP media issue In-Reply-To: <01cd01cd3ba4$e7b13c80$b713b580$@gmail.com> References: <02a201cd3a1a$2a658770$7f309650$@gmail.com> <007301cd3a96$d106ed90$7314c8b0$@gmail.com> <1FFF97C269757C458224B7C895F35F150D9857@cantor.std.visionutv.se> <01cd01cd3ba4$e7b13c80$b713b580$@gmail.com> Message-ID: Hi, Made current as requested but nothing works afterwards, freeswitch.log has this in it 2012-05-28 23:39:41.704128 [CRIT] switch_loadable_module.c:1300 Error Loading module /usr/local/freeswitch/mod/mod_sofia.so **/usr/local/freeswitch/mod/mod_sofia.so: undefined symbol: tls_version** Had to copy older version back, lost a few changes from last month L From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Sunday, 27 May 2012 10:29 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] RTP media issue Also update, you are running git head so you have to update frequently, you rev is a month old. On May 26, 2012 7:43 AM, "Peter Olsson" wrote: I think the first step is to check the actual signalling. Just as Anthony mentions, the call setup looks weird. Between FS and the ATA (if I understood the different IP's/peers correctly) it seems FS sends early media (183), and it takes the ATA 10 seconds to ACK this. During the time the call is answered on the other end, which dosn't seem to be passed to the ATA (probably because it's so late with the ACK - I'm not sure about that though). So, first check this - it seems to me that FS might still be in "early media" state during the entire call. About the actual RTP, I see one problem here, it's packet 8146 in outbound.pcap. This is a packet that is probably generated by FS (since by that exact time, you're missing one packet from the provider), the problem here is that the timestamp if way off (I'm not sure where FS gets this ts from). It's because of this faulty timestamp that a few packets after this is dropped, since FS won't send packets with a lower timestamp than before. This is also why packet 8152 seems to have a strange timestamp, but it's actually just passed from the other leg, and is the first timestamp that is greater then packet 8146. One possible solution would be to enable rewrite of timestamps (rtp-rewrite-timestamps in the sofia config), also mentioned here: http://wiki.freeswitch.org/wiki/RTP_Issues#Dropped_Audio However, I think you also need to check the strangness in the SIP signalling. /Peter ________________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] för Mr Nathan Downes [nathandownes at hotmail.com] Skickat: den 26 maj 2012 00:03 Till: 'FreeSWITCH Users Help' ?mne: Re: [Freeswitch-users] RTP media issue Hi Anthony, FS version = FreeSWITCH Version 1.1.beta1 (git-f1b5044 2012-04-26 11-28-47 -0500) I don't have a debug log, but I could probably get it with another trace of both sides of the call, but it would be hard to capture as there is constant calls to this, unless there is a way to do it on a per call basis? I can also only do testing onsite as we don't have the same fibre equipment. I already have jitterbuffer set in both profiles in an attempt to try and stop it using , is there a way to set the cng_plc in the profile itself rather than diaplpan as there are 70 or so numbers in it. In the outbound dialplans I also added because I kept seeing PAUSE JITTERBUFFER in the FS logs when calls were made outbound so I wasn't sure it was doing something and read somewhere it pauses it when it bridges the call The inbound dialplan for all of those people consists of It doesn't affect SIP phones or normal ATA devices we have connected and only affects these FTTH GPON ATA's, but with almost 100 residents in this retirement village and them constantly complaining we have been given til Wednesday to come up with a solution or risk losing our position as the internet/phone provider for that retirement village. It appeared to me that what happens in that trace isn't normal behaviour, I did try rewriting timestamps last week, but as you suggested that appeared to mask the issue but not stop it from happening. That was when I was losing a packet from them each second or so, which by the time it arrived to end user sounded horrible. It has settled down a lot now and maybe 1 or 2 packets per call, but if what is in this trace is the cause each time, that would explain the poor end users experience. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Saturday, 26 May 2012 2:06 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] RTP media issue What version of FS are you running? Do you have the debug logs of those calls? you could try using the jitterbuffer. in the inbound DP to FS *before* you answer. Also it looks a little odd to me in this trace if this is the same call, it seems like you answered the call before placing the call to the phone and that phone never answers.... On Thu, May 24, 2012 at 9:51 PM, Nathan Downes wrote: > Hi, > > enable-soa > > > > Set the value to "false" to diable SIP SOA from sofia to tell sofia > not to touch the exchange of SDP > > I don't think this is related to the exchange of an SDP message.. Can > you elaborate more before I try it? I can't make things worse or > change things I don't understand. > > ________________________________ > From: djbinter at gmail.com > To: freeswitch-users at lists.freeswitch.org > CC: nathan at nortec.com.au > Subject: Re: [Freeswitch-users] RTP media issue > Date: Fri, 25 May 2012 11:19:46 +1000 > > > > > > Sent from my iPad > > On May 24, 2012, at 5:01 PM, Nathan Downes wrote: > > Hi, > > I had previous reported an issue with poor voice quality, appearing to > stem from occasion wrong timestamps coming from provider, but the end > user's experience was much worse than what I could see/hear in the trace. > > I have finally captured an event inbound and outbound. The thing I > don't understand is I thought even though FS proxied the media it > didn't touch it or change anything, but it appears it is. > > The 2 traces are http://www.nortec.com.au/inbound.pcap.gz and > http://www.nortec.com.au/outbound.pcap.gz > > Inbound is from my trunk provider to FS box and outbound is FS box to > ATA in FTTH GPON. > > The event I am talking about, if both traces are open, is in the > inbound one inbetween packet 8114 and 8117 the provider drops a packet > or I don't receive it. In the corresponding outbound trace, between > packet 8144 and 8152, it appears FS misses a whole heap of packets > (.1 seconds) between > 8146 and 8152 then it increases the timestamp only by 40 rather than > 160 on packet 8152. This seems to not affect SIP phones themselves > but causes issues with the FTTH GPON ATA. > > This causes a gap in the audio for the end user, and when they miss a > high number of packets even though it sounds good on the inbound trace > the end users experience is horrible. This trace is actually a good > one, but the wrong timestamp can occur once per second, causing end > user to lose 10%+ of incoming audio only. The issue only affects the > audio coming from provider to FS to end user. > > I am chasing it up with the voice provider to try and eliminate the > occasional packet loss, but if I could stop/fix FS from doing its > adjustment/gap/something the end user wouldn't even notice it. > > > > ______________________________________________________________________ > ___ > > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org > > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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 at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4fbfffef32764666710318! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120528/f30fddb3/attachment-0001.html From avi at avimarcus.net Mon May 28 20:19:20 2012 From: avi at avimarcus.net (Avi Marcus) Date: Mon, 28 May 2012 19:19:20 +0300 Subject: [Freeswitch-users] how to schedule calls In-Reply-To: <1338198708.69195.YahooMailClassic@web110807.mail.gq1.yahoo.com> References: <1338187031.89546.YahooMailClassic@web110814.mail.gq1.yahoo.com> <1338198708.69195.YahooMailClassic@web110807.mail.gq1.yahoo.com> Message-ID: With ESL , originates -- driven by your own custom scripting -- you can do all sorts of things. e.g. query the voicemail DB for.. unread? messages then originate a call to them with a specific message then directly to their VM. With your own code and ESL, you can instruct FS to do nearly anything. -Avi On Mon, May 28, 2012 at 12:51 PM, Sherif Omran wrote: > Hi guys, > > Any body knows how to schedule calls depending on some internal variables. > Such as at 8pm, users with voice mail not yet delivered receive a call. > > Thanks > S. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120528/f37dc6be/attachment.html From avi at avimarcus.net Mon May 28 20:20:32 2012 From: avi at avimarcus.net (Avi Marcus) Date: Mon, 28 May 2012 19:20:32 +0300 Subject: [Freeswitch-users] Gateway failover with LUA In-Reply-To: References: Message-ID: How about a bridge2, bridge3, etc variable? Then your static dialplan can see if those are set, and if so, use them. -Avi On Mon, May 28, 2012 at 1:09 PM, Muhammad Naseer Bhatti wrote: > Hi, I am trying to implement a gateway failover functionality with > lua. Right now I am using limit and using hash backend to limit number > of channels to each gateway. Since I am forwarding all calls with a > default dialplan which is sending all calls to application lua and the > bridge is done outside lua setting all the necessary vars. > The idea is if all channels of first gateway are occupied, send the > call to second gateway in the list and so on. Any idea how to > implement this with lua? Can not seem to find the logic. If I query > with limit_status it is going to be a query on every call. In a high > call volume scenario, I don't think this would be really a good > choice. Any ideas, thoughts? > > Thanks. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120528/8c55389f/attachment.html From jayesh.voip at gmail.com Mon May 28 21:41:03 2012 From: jayesh.voip at gmail.com (Jayesh Nambiar) Date: Mon, 28 May 2012 23:11:03 +0530 Subject: [Freeswitch-users] problem with fs_path in mod_callcenter agents Message-ID: Hello All, I am using mod_callcenter and have the agents definitions as follows: I include the fs_path parameter since my agents are not directly registered on freeswitch. The calls need to be sent to a different proxy server where the agents are registered. proxy_ip variable is defined in the vars.xml. Now the problem is as follows: When I start freeswitch, the calls routed to callcenter queues are not sent to the fs_path parameter defined. Instead freeswitch tries to resolve the domain "test.com" and send the call to the resolved IP address. But after I do a "reload -f mod_callcenter", the calls are properly routed to the IP Address mentioned in the fs_path parameter. I am running the 1.2 RC2 version of freeswitch. Can anyone explain why this might happen and why do I exclusively need to reload the callcenter module to make this work. Thanks for any suggestions or pointers to check for this problem. --- Jayesh -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120528/70cbb6ad/attachment.html From engelster at gmail.com Mon May 28 20:58:30 2012 From: engelster at gmail.com (Der Engel) Date: Mon, 28 May 2012 11:58:30 -0500 Subject: [Freeswitch-users] how to schedule calls In-Reply-To: References: <1338187031.89546.YahooMailClassic@web110814.mail.gq1.yahoo.com> <1338198708.69195.YahooMailClassic@web110807.mail.gq1.yahoo.com> Message-ID: Can any programming language be used with ESL? how is it differente from httapi ? On Mon, May 28, 2012 at 11:19 AM, Avi Marcus wrote: > With ESL , originates -- driven by > your own custom scripting -- you can do all sorts of things. e.g. query the > voicemail DB for.. unread? messages then originate a call to them with a > specific message then directly to their VM. > With your own code and ESL, you can instruct FS to do nearly anything. > > -Avi > > > > On Mon, May 28, 2012 at 12:51 PM, Sherif Omran > wrote: > >> Hi guys, >> >> Any body knows how to schedule calls depending on some internal >> variables. Such as at 8pm, users with voice mail not yet delivered receive >> a call. >> >> Thanks >> S. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120528/9c0fd3cb/attachment.html From saami_mh at ymail.com Mon May 28 22:35:28 2012 From: saami_mh at ymail.com (Samira Mh) Date: Mon, 28 May 2012 11:35:28 -0700 (PDT) Subject: [Freeswitch-users] "bash" programming on freeswitch? Message-ID: <1338230128.40509.YahooMailNeo@web120104.mail.ne1.yahoo.com> how to run bash script programming in freeswitch using AGI (like freeswitch)? i think freeswitch support JAVA/PERL/PYTHON/RUBY programming -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120528/29920607/attachment-0001.html From saami_mh at ymail.com Mon May 28 22:38:35 2012 From: saami_mh at ymail.com (Samira Mh) Date: Mon, 28 May 2012 11:38:35 -0700 (PDT) Subject: [Freeswitch-users] query on freeswitch In-Reply-To: References: <1338123550.34364.YahooMailNeo@web120104.mail.ne1.yahoo.com> Message-ID: <1338230315.40508.YahooMailNeo@web120102.mail.ne1.yahoo.com> thanks ?Avi for your reply,my problem solved; ________________________________ From: Avi Marcus To: FreeSWITCH Users Help Sent: Monday, May 28, 2012 3:01 AM Subject: Re: [Freeswitch-users] query on freeswitch http://wiki.freeswitch.org/wiki/Mod_odbc_query?is pretty descriptive. -Avi On Sun, May 27, 2012 at 3:59 PM, Samira Mh wrote: how to implement mod query odbc on dialplan of freeswitch? >the wiki is colud't help me? >plz help? > >? >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >Join Us At ClueCon - Aug 7-9, 2012 > >FreeSWITCH-users mailing list >FreeSWITCH-users at 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 at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120528/fde6f320/attachment.html From bdfoster at endigotech.com Mon May 28 22:48:11 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Mon, 28 May 2012 14:48:11 -0400 Subject: [Freeswitch-users] "bash" programming on freeswitch? In-Reply-To: <1338230128.40509.YahooMailNeo@web120104.mail.ne1.yahoo.com> References: <1338230128.40509.YahooMailNeo@web120104.mail.ne1.yahoo.com> Message-ID: system script.sh arg1 arg2 argN No FS library though so you can't actually control calls. -BDF Brian Foster Endigo Computer LLC Sent from a mobile device. On May 28, 2012 2:36 PM, "Samira Mh" wrote: > how to run bash script programming in freeswitch using AGI (like > freeswitch)? > i think freeswitch support JAVA/PERL/PYTHON/RUBY programming > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120528/a4544b77/attachment.html From peter.olsson at visionutveckling.se Mon May 28 22:55:58 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Mon, 28 May 2012 18:55:58 +0000 Subject: [Freeswitch-users] RTP media issue In-Reply-To: References: <02a201cd3a1a$2a658770$7f309650$@gmail.com> <007301cd3a96$d106ed90$7314c8b0$@gmail.com> <1FFF97C269757C458224B7C895F35F150D9857@cantor.std.visionutv.se> <01cd01cd3ba4$e7b13c80$b713b580$@gmail.com>, Message-ID: Did you really do "make current"? Seems there is a linking problem somewhere, but I know for sure current git builds and loads fine. Anyway, about the original issue, did you try using rtp-rewrite-timestamps, to see if that helped? /Peter 28 maj 2012 kl. 16:02 skrev "Mr Nathan Downes" >: Hi, Made current as requested but nothing works afterwards, freeswitch.log has this in it 2012-05-28 23:39:41.704128 [CRIT] switch_loadable_module.c:1300 Error Loading module /usr/local/freeswitch/mod/mod_sofia.so **/usr/local/freeswitch/mod/mod_sofia.so: undefined symbol: tls_version** Had to copy older version back, lost a few changes from last month? :( From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Sunday, 27 May 2012 10:29 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] RTP media issue Also update, you are running git head so you have to update frequently, you rev is a month old. On May 26, 2012 7:43 AM, "Peter Olsson" > wrote: I think the first step is to check the actual signalling. Just as Anthony mentions, the call setup looks weird. Between FS and the ATA (if I understood the different IP's/peers correctly) it seems FS sends early media (183), and it takes the ATA 10 seconds to ACK this. During the time the call is answered on the other end, which dosn't seem to be passed to the ATA (probably because it's so late with the ACK - I'm not sure about that though). So, first check this - it seems to me that FS might still be in "early media" state during the entire call. About the actual RTP, I see one problem here, it's packet 8146 in outbound.pcap. This is a packet that is probably generated by FS (since by that exact time, you're missing one packet from the provider), the problem here is that the timestamp if way off (I'm not sure where FS gets this ts from). It's because of this faulty timestamp that a few packets after this is dropped, since FS won't send packets with a lower timestamp than before. This is also why packet 8152 seems to have a strange timestamp, but it's actually just passed from the other leg, and is the first timestamp that is greater then packet 8146. One possible solution would be to enable rewrite of timestamps (rtp-rewrite-timestamps in the sofia config), also mentioned here: http://wiki.freeswitch.org/wiki/RTP_Issues#Dropped_Audio However, I think you also need to check the strangness in the SIP signalling. /Peter ________________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] för Mr Nathan Downes [nathandownes at hotmail.com] Skickat: den 26 maj 2012 00:03 Till: 'FreeSWITCH Users Help' ?mne: Re: [Freeswitch-users] RTP media issue Hi Anthony, FS version = FreeSWITCH Version 1.1.beta1 (git-f1b5044 2012-04-26 11-28-47 -0500) I don't have a debug log, but I could probably get it with another trace of both sides of the call, but it would be hard to capture as there is constant calls to this, unless there is a way to do it on a per call basis? I can also only do testing onsite as we don't have the same fibre equipment. I already have jitterbuffer set in both profiles in an attempt to try and stop it using , is there a way to set the cng_plc in the profile itself rather than diaplpan as there are 70 or so numbers in it. In the outbound dialplans I also added because I kept seeing PAUSE JITTERBUFFER in the FS logs when calls were made outbound so I wasn't sure it was doing something and read somewhere it pauses it when it bridges the call The inbound dialplan for all of those people consists of It doesn't affect SIP phones or normal ATA devices we have connected and only affects these FTTH GPON ATA's, but with almost 100 residents in this retirement village and them constantly complaining we have been given til Wednesday to come up with a solution or risk losing our position as the internet/phone provider for that retirement village. It appeared to me that what happens in that trace isn't normal behaviour, I did try rewriting timestamps last week, but as you suggested that appeared to mask the issue but not stop it from happening. That was when I was losing a packet from them each second or so, which by the time it arrived to end user sounded horrible. It has settled down a lot now and maybe 1 or 2 packets per call, but if what is in this trace is the cause each time, that would explain the poor end users experience. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Saturday, 26 May 2012 2:06 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] RTP media issue What version of FS are you running? Do you have the debug logs of those calls? you could try using the jitterbuffer. in the inbound DP to FS *before* you answer. Also it looks a little odd to me in this trace if this is the same call, it seems like you answered the call before placing the call to the phone and that phone never answers.... On Thu, May 24, 2012 at 9:51 PM, Nathan Downes > wrote: > Hi, > > enable-soa > > > > Set the value to "false" to diable SIP SOA from sofia to tell sofia > not to touch the exchange of SDP > > I don't think this is related to the exchange of an SDP message.. Can > you elaborate more before I try it? I can't make things worse or > change things I don't understand. > > ________________________________ > From: djbinter at gmail.com > To: freeswitch-users at lists.freeswitch.org > CC: nathan at nortec.com.au > Subject: Re: [Freeswitch-users] RTP media issue > Date: Fri, 25 May 2012 11:19:46 +1000 > > > > > > Sent from my iPad > > On May 24, 2012, at 5:01 PM, Nathan Downes > wrote: > > Hi, > > I had previous reported an issue with poor voice quality, appearing to > stem from occasion wrong timestamps coming from provider, but the end > user's experience was much worse than what I could see/hear in the trace. > > I have finally captured an event inbound and outbound. The thing I > don't understand is I thought even though FS proxied the media it > didn't touch it or change anything, but it appears it is. > > The 2 traces are http://www.nortec.com.au/inbound.pcap.gz and > http://www.nortec.com.au/outbound.pcap.gz > > Inbound is from my trunk provider to FS box and outbound is FS box to > ATA in FTTH GPON. > > The event I am talking about, if both traces are open, is in the > inbound one inbetween packet 8114 and 8117 the provider drops a packet > or I don't receive it. In the corresponding outbound trace, between > packet 8144 and 8152, it appears FS misses a whole heap of packets > (.1 seconds) between > 8146 and 8152 then it increases the timestamp only by 40 rather than > 160 on packet 8152. This seems to not affect SIP phones themselves > but causes issues with the FTTH GPON ATA. > > This causes a gap in the audio for the end user, and when they miss a > high number of packets even though it sounds good on the inbound trace > the end users experience is horrible. This trace is actually a good > one, but the wrong timestamp can occur once per second, causing end > user to lose 10%+ of incoming audio only. The issue only affects the > audio coming from provider to FS to end user. > > I am chasing it up with the voice provider to try and eliminate the > occasional packet loss, but if I could stop/fix FS from doing its > adjustment/gap/something the end user wouldn't even notice it. > > > > ______________________________________________________________________ > ___ > > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org > > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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 at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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 at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4fc3827732761296161717! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4fc3827732761296161717! From anthony.minessale at gmail.com Mon May 28 23:06:43 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 28 May 2012 14:06:43 -0500 Subject: [Freeswitch-users] RTP media issue In-Reply-To: References: <02a201cd3a1a$2a658770$7f309650$@gmail.com> <007301cd3a96$d106ed90$7314c8b0$@gmail.com> <1FFF97C269757C458224B7C895F35F150D9857@cantor.std.visionutv.se> <01cd01cd3ba4$e7b13c80$b713b580$@gmail.com> Message-ID: Lost changes in what? This is the reason I don't like dealing with issues on the list instead of Jira. You need to be able to run git head to reproduce an issue its our standard policy. You should really examine your environment and find out why you can't it may be a clue, you should download and build a fresh git checkout. On May 28, 2012 8:53 AM, "Mr Nathan Downes" wrote: > Hi,**** > > ** ** > > Made current as requested but nothing works afterwards, freeswitch.log has > this in it**** > > ** ** > > 2012-05-28 23:39:41.704128 [CRIT] switch_loadable_module.c:1300 Error > Loading module /usr/local/freeswitch/mod/mod_sofia.so**** > > **/usr/local/freeswitch/mod/mod_sofia.so: undefined symbol: tls_version*** > *** > > ** ** > > Had to copy older version back, lost a few changes from last month? L**** > > ** ** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony > Minessale > *Sent:* Sunday, 27 May 2012 10:29 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] RTP media issue**** > > ** ** > > Also update, you are running git head so you have to update frequently, > you rev is a month old.**** > > On May 26, 2012 7:43 AM, "Peter Olsson" > wrote:**** > > I think the first step is to check the actual signalling. > > Just as Anthony mentions, the call setup looks weird. Between FS and the > ATA (if I understood the different IP's/peers correctly) it seems FS sends > early media (183), and it takes the ATA 10 seconds to ACK this. During the > time the call is answered on the other end, which dosn't seem to be passed > to the ATA (probably because it's so late with the ACK - I'm not sure about > that though). So, first check this - it seems to me that FS might still be > in "early media" state during the entire call. > > About the actual RTP, I see one problem here, it's packet 8146 in > outbound.pcap. This is a packet that is probably generated by FS (since by > that exact time, you're missing one packet from the provider), the problem > here is that the timestamp if way off (I'm not sure where FS gets this ts > from). It's because of this faulty timestamp that a few packets after this > is dropped, since FS won't send packets with a lower timestamp than before. > This is also why packet 8152 seems to have a strange timestamp, but it's > actually just passed from the other leg, and is the first timestamp that is > greater then packet 8146. > > One possible solution would be to enable rewrite of timestamps > (rtp-rewrite-timestamps in the sofia config), also mentioned here: > http://wiki.freeswitch.org/wiki/RTP_Issues#Dropped_Audio > > However, I think you also need to check the strangness in the SIP > signalling. > > /Peter > > ________________________________________ > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [ > freeswitch-users-bounces at lists.freeswitch.org] för Mr Nathan Downes [ > nathandownes at hotmail.com] > Skickat: den 26 maj 2012 00:03 > Till: 'FreeSWITCH Users Help' > ?mne: Re: [Freeswitch-users] RTP media issue > > Hi Anthony, > > FS version = FreeSWITCH Version 1.1.beta1 (git-f1b5044 2012-04-26 11-28-47 > -0500) > > I don't have a debug log, but I could probably get it with another trace of > both sides of the call, but it would be hard to capture as there is > constant > calls to this, unless there is a way to do it on a per call basis? I can > also only do testing onsite as we don't have the same fibre equipment. I > already have jitterbuffer set in both profiles in an attempt to try and > stop > it using , is there a way > to set the cng_plc in the profile itself rather than diaplpan as there are > 70 or so numbers in it. In the outbound dialplans I also added application="set" data="sip_jitter_buffer_during_bridge=true" /> because I > kept seeing PAUSE JITTERBUFFER in the FS logs when calls were made outbound > so I wasn't sure it was doing something and read somewhere it pauses it > when > it bridges the call > > The inbound dialplan for all of those people consists of > > > > > > > > > > > > > It doesn't affect SIP phones or normal ATA devices we have connected and > only affects these FTTH GPON ATA's, but with almost 100 residents in this > retirement village and them constantly complaining we have been given til > Wednesday to come up with a solution or risk losing our position as the > internet/phone provider for that retirement village. > > It appeared to me that what happens in that trace isn't normal behaviour, I > did try rewriting timestamps last week, but as you suggested that appeared > to mask the issue but not stop it from happening. That was when I was > losing a packet from them each second or so, which by the time it arrived > to > end user sounded horrible. It has settled down a lot now and maybe 1 or 2 > packets per call, but if what is in this trace is the cause each time, that > would explain the poor end users experience. > > > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Anthony > Minessale > Sent: Saturday, 26 May 2012 2:06 AM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] RTP media issue > > What version of FS are you running? > Do you have the debug logs of those calls? > > you could try using the jitterbuffer. > > > > in the inbound DP to FS *before* you answer. > > Also it looks a little odd to me in this trace if this is the same call, it > seems like you answered the call before placing the call to the phone and > that phone never answers.... > > > > > > > > > On Thu, May 24, 2012 at 9:51 PM, Nathan Downes > wrote: > > Hi, > > > > enable-soa > > > > > > > > Set the value to "false" to diable SIP SOA from sofia to tell sofia > > not to touch the exchange of SDP > > > > I don't think this is related to the exchange of an SDP message.. Can > > you elaborate more before I try it? I can't make things worse or > > change things I don't understand. > > > > ________________________________ > > From: djbinter at gmail.com > > To: freeswitch-users at lists.freeswitch.org > > CC: nathan at nortec.com.au > > Subject: Re: [Freeswitch-users] RTP media issue > > Date: Fri, 25 May 2012 11:19:46 +1000 > > > > > > > > > > > > Sent from my iPad > > > > On May 24, 2012, at 5:01 PM, Nathan Downes > wrote: > > > > Hi, > > > > I had previous reported an issue with poor voice quality, appearing to > > stem from occasion wrong timestamps coming from provider, but the end > > user's experience was much worse than what I could see/hear in the trace. > > > > I have finally captured an event inbound and outbound. The thing I > > don't understand is I thought even though FS proxied the media it > > didn't touch it or change anything, but it appears it is. > > > > The 2 traces are http://www.nortec.com.au/inbound.pcap.gz and > > http://www.nortec.com.au/outbound.pcap.gz > > > > Inbound is from my trunk provider to FS box and outbound is FS box to > > ATA in FTTH GPON. > > > > The event I am talking about, if both traces are open, is in the > > inbound one inbetween packet 8114 and 8117 the provider drops a packet > > or I don't receive it. In the corresponding outbound trace, between > > packet 8144 and 8152, it appears FS misses a whole heap of packets > > (.1 seconds) between > > 8146 and 8152 then it increases the timestamp only by 40 rather than > > 160 on packet 8152. This seems to not affect SIP phones themselves > > but causes issues with the FTTH GPON ATA. > > > > This causes a gap in the audio for the end user, and when they miss a > > high number of packets even though it sounds good on the inbound trace > > the end users experience is horrible. This trace is actually a good > > one, but the wrong timestamp can occur once per second, causing end > > user to lose 10%+ of incoming audio only. The issue only affects the > > audio coming from provider to FS to end user. > > > > I am chasing it up with the voice provider to try and eliminate the > > occasional packet loss, but if I could stop/fix FS from doing its > > adjustment/gap/something the end user wouldn't even notice it. > > > > > > > > ______________________________________________________________________ > > ___ > > > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > > rs > > http://www.freeswitch.org > > > > > > ______________________________________________________________________ > > ___ Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > > rs > > http://www.freeswitch.org > > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > !DSPAM:4fbfffef32764666710318! > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120528/50b1dd36/attachment-0001.html From saami_mh at ymail.com Mon May 28 23:20:57 2012 From: saami_mh at ymail.com (Samira Mh) Date: Mon, 28 May 2012 12:20:57 -0700 (PDT) Subject: [Freeswitch-users] "bash" programming on freeswitch? In-Reply-To: References: <1338230128.40509.YahooMailNeo@web120104.mail.ne1.yahoo.com> Message-ID: <1338232857.21351.YahooMailNeo@web120102.mail.ne1.yahoo.com> Hi Brian, thanks for your reply, but i want to do more things on freeswitch using Bash script on AGI, (like asterisk) i want to run for/while loop , if-else condition usin bash and use of the result of running AGI on freeswitch dialplan ?On asterisk the dial plan looks as follows exten => 1500,1,AGI(agi://192.168.0.131/hello.agi). I need something similar to this. How do i do this from freeswitch? thanks alot ________________________________ From: Brian Foster To: FreeSWITCH Users Help Sent: Monday, May 28, 2012 11:18 PM Subject: Re: [Freeswitch-users] "bash" programming on freeswitch? system script.sh arg1 arg2 argN No FS library though so you can't actually control calls. -BDF Brian Foster Endigo Computer LLC Sent from a mobile device. On May 28, 2012 2:36 PM, "Samira Mh" wrote: how to run bash script programming in freeswitch using AGI (like freeswitch)? >i think freeswitch support JAVA/PERL/PYTHON/RUBY programming >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >Join Us At ClueCon - Aug 7-9, 2012 > >FreeSWITCH-users mailing list >FreeSWITCH-users at 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 at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120528/1201a565/attachment.html From moises.silva at gmail.com Mon May 28 23:24:06 2012 From: moises.silva at gmail.com (Moises Silva) Date: Mon, 28 May 2012 15:24:06 -0400 Subject: [Freeswitch-users] freeswitch - sangoma CAS T1 - a101 configuration In-Reply-To: References: Message-ID: On Mon, May 28, 2012 at 12:49 AM, copycall wrote: > curriegrad2004, > > thanks for responding. > > unfortunately, the sangoma tech support has not been as crack as usual. > > i can go into the specifics, but it probably isn't necessary in a public > forum. i'm not looking to piss anyone off. > > if you have you actually configured a a101 sangoma card for a legacy CAS > T1 card, i would greatly appreciate your assistance. > > It's been a while since I configured for RBS, but I may be able to give you some pointers. Where are you at? did you have the drivers compiled? http://wiki.sangoma.com/wanpipe-freeswitch-ftdm As that web page states, RBS with FreeSWITCH is only supported using Wanpipe + DAHDI, not in native Wanpipe mode. http://wiki.sangoma.com/FreeSWITCH-Dahdi-Mode The module you want to use is ftmod_analog *Moises Silva **Manager, Software Engineering*** msilva at sangoma.com Sangoma Technologies 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada t. +1 800 388 2475 (N. America) t. +1 905 474 1990 x128 f. +1 905 474 9223 ** Products | Solutions | Events | Contact | Wiki | Facebook | Twitter`| | YouTube -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120528/7225331c/attachment.html From avi at avimarcus.net Mon May 28 23:35:50 2012 From: avi at avimarcus.net (Avi Marcus) Date: Mon, 28 May 2012 22:35:50 +0300 Subject: [Freeswitch-users] "bash" programming on freeswitch? In-Reply-To: <1338232857.21351.YahooMailNeo@web120102.mail.ne1.yahoo.com> References: <1338230128.40509.YahooMailNeo@web120104.mail.ne1.yahoo.com> <1338232857.21351.YahooMailNeo@web120102.mail.ne1.yahoo.com> Message-ID: Lua is good for this. Or any language over ESL or event socket outbound, which might be the most comparable to running an external script on AGI. -Avi On Mon, May 28, 2012 at 10:20 PM, Samira Mh wrote: > Hi Brian, > thanks for your reply, > but i want to do more things on freeswitch using Bash script on AGI, (like > asterisk) > i want to run for/while loop , if-else condition usin bash and use of the > result of running AGI on freeswitch dialplan > On asterisk the dial plan looks as follows > > exten => 1500,1,AGI(agi://192.168.0.131/hello.agi). > > I need something similar to this. How do i do this from freeswitch? > thanks alot > > ------------------------------ > *From:* Brian Foster > *To:* FreeSWITCH Users Help > *Sent:* Monday, May 28, 2012 11:18 PM > *Subject:* Re: [Freeswitch-users] "bash" programming on freeswitch? > > system script.sh arg1 arg2 argN > No FS library though so you can't actually control calls. > -BDF > Brian Foster > Endigo Computer LLC > Sent from a mobile device. > On May 28, 2012 2:36 PM, "Samira Mh" wrote: > > how to run bash script programming in freeswitch using AGI (like > freeswitch)? > i think freeswitch support JAVA/PERL/PYTHON/RUBY programming > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120528/13c1092d/attachment-0001.html From bdfoster at endigotech.com Mon May 28 23:44:26 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Mon, 28 May 2012 15:44:26 -0400 Subject: [Freeswitch-users] "bash" programming on freeswitch? In-Reply-To: References: <1338230128.40509.YahooMailNeo@web120104.mail.ne1.yahoo.com> <1338232857.21351.YahooMailNeo@web120102.mail.ne1.yahoo.com> Message-ID: Adding to Avi's comments, LUA is also more efficient and highly embeddable when used with FS. Brian Foster Endigo Computer LLC Sent from a mobile device. On May 28, 2012 3:36 PM, "Avi Marcus" wrote: > Lua is good for this. Or any language over ESL or event socket outbound, > which might be the most comparable to running an external script on AGI. > -Avi > > > On Mon, May 28, 2012 at 10:20 PM, Samira Mh wrote: > >> Hi Brian, >> thanks for your reply, >> but i want to do more things on freeswitch using Bash script on AGI, >> (like asterisk) >> i want to run for/while loop , if-else condition usin bash and use of the >> result of running AGI on freeswitch dialplan >> On asterisk the dial plan looks as follows >> >> exten => 1500,1,AGI(agi://192.168.0.131/hello.agi). >> >> I need something similar to this. How do i do this from freeswitch? >> thanks alot >> >> ------------------------------ >> *From:* Brian Foster >> *To:* FreeSWITCH Users Help >> *Sent:* Monday, May 28, 2012 11:18 PM >> *Subject:* Re: [Freeswitch-users] "bash" programming on freeswitch? >> >> system script.sh arg1 arg2 argN >> No FS library though so you can't actually control calls. >> -BDF >> Brian Foster >> Endigo Computer LLC >> Sent from a mobile device. >> On May 28, 2012 2:36 PM, "Samira Mh" wrote: >> >> how to run bash script programming in freeswitch using AGI (like >> freeswitch)? >> i think freeswitch support JAVA/PERL/PYTHON/RUBY programming >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120528/10b99bc1/attachment.html From nathandownes at hotmail.com Tue May 29 01:25:19 2012 From: nathandownes at hotmail.com (Mr Nathan Downes) Date: Tue, 29 May 2012 07:25:19 +1000 Subject: [Freeswitch-users] RTP media issue In-Reply-To: <042b01cd3d0b$b3f22d60$1bd68820$@gmail.com> References: <02a201cd3a1a$2a658770$7f309650$@gmail.com> <007301cd3a96$d106ed90$7314c8b0$@gmail.com> <1FFF97C269757C458224B7C895F35F150D9857@cantor.std.visionutv.se> <01cd01cd3ba4$e7b13c80$b713b580$@gmail.com> <042b01cd3d0b$b3f22d60$1bd68820$@gmail.com> Message-ID: I copy /usr/local/freeswitch before making current in case something goes wrong, so I have quite a few backups, I made current and it didn?t work so I deleted that folder then copied a backup back. Then though I had an idea why it didn?t run (didn?t check log straight away) made current again, but forgot a backup. Then when it didn?t work and I found that I copied another backup back.. it was from February. Completely my fault and I shouldn?t be doing it at midnight when tired J As for the 3 possible things to try to fix it, at 9:23am on Friday the people we trunk with found an issue on their IP network and corrected it, so I have 0% packet loss from the upstream provider again as we both are in the same data centre. I?ll keep tracing until I find a call with a missing packet so I can see the effect. I have some captures at work using tshark I?ll send across tomorrow so you can check it?s not staying in early media, I see an ACK after the 100, but not sure that is all good. Pcapsipdump appears to only capture calls in one direction on that side of things as well. I?ll sort out the build and get it current, and try and locate a way to simulate that one lost packet from their side, and we will see how those fixes have gone, the call quality has been fine at the site since they resolved that packet loss. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Tuesday, 29 May 2012 5:07 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] RTP media issue Lost changes in what? This is the reason I don't like dealing with issues on the list instead of Jira. You need to be able to run git head to reproduce an issue its our standard policy. You should really examine your environment and find out why you can't it may be a clue, you should download and build a fresh git checkout. On May 28, 2012 8:53 AM, "Mr Nathan Downes" wrote: Hi, Made current as requested but nothing works afterwards, freeswitch.log has this in it 2012-05-28 23:39:41.704128 [CRIT] switch_loadable_module.c:1300 Error Loading module /usr/local/freeswitch/mod/mod_sofia.so **/usr/local/freeswitch/mod/mod_sofia.so: undefined symbol: tls_version** Had to copy older version back, lost a few changes from last month L From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Sunday, 27 May 2012 10:29 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] RTP media issue Also update, you are running git head so you have to update frequently, you rev is a month old. On May 26, 2012 7:43 AM, "Peter Olsson" wrote: I think the first step is to check the actual signalling. Just as Anthony mentions, the call setup looks weird. Between FS and the ATA (if I understood the different IP's/peers correctly) it seems FS sends early media (183), and it takes the ATA 10 seconds to ACK this. During the time the call is answered on the other end, which dosn't seem to be passed to the ATA (probably because it's so late with the ACK - I'm not sure about that though). So, first check this - it seems to me that FS might still be in "early media" state during the entire call. About the actual RTP, I see one problem here, it's packet 8146 in outbound.pcap. This is a packet that is probably generated by FS (since by that exact time, you're missing one packet from the provider), the problem here is that the timestamp if way off (I'm not sure where FS gets this ts from). It's because of this faulty timestamp that a few packets after this is dropped, since FS won't send packets with a lower timestamp than before. This is also why packet 8152 seems to have a strange timestamp, but it's actually just passed from the other leg, and is the first timestamp that is greater then packet 8146. One possible solution would be to enable rewrite of timestamps (rtp-rewrite-timestamps in the sofia config), also mentioned here: http://wiki.freeswitch.org/wiki/RTP_Issues#Dropped_Audio However, I think you also need to check the strangness in the SIP signalling. /Peter ________________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] för Mr Nathan Downes [nathandownes at hotmail.com] Skickat: den 26 maj 2012 00:03 Till: 'FreeSWITCH Users Help' ?mne: Re: [Freeswitch-users] RTP media issue Hi Anthony, FS version = FreeSWITCH Version 1.1.beta1 (git-f1b5044 2012-04-26 11-28-47 -0500) I don't have a debug log, but I could probably get it with another trace of both sides of the call, but it would be hard to capture as there is constant calls to this, unless there is a way to do it on a per call basis? I can also only do testing onsite as we don't have the same fibre equipment. I already have jitterbuffer set in both profiles in an attempt to try and stop it using , is there a way to set the cng_plc in the profile itself rather than diaplpan as there are 70 or so numbers in it. In the outbound dialplans I also added because I kept seeing PAUSE JITTERBUFFER in the FS logs when calls were made outbound so I wasn't sure it was doing something and read somewhere it pauses it when it bridges the call The inbound dialplan for all of those people consists of It doesn't affect SIP phones or normal ATA devices we have connected and only affects these FTTH GPON ATA's, but with almost 100 residents in this retirement village and them constantly complaining we have been given til Wednesday to come up with a solution or risk losing our position as the internet/phone provider for that retirement village. It appeared to me that what happens in that trace isn't normal behaviour, I did try rewriting timestamps last week, but as you suggested that appeared to mask the issue but not stop it from happening. That was when I was losing a packet from them each second or so, which by the time it arrived to end user sounded horrible. It has settled down a lot now and maybe 1 or 2 packets per call, but if what is in this trace is the cause each time, that would explain the poor end users experience. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Saturday, 26 May 2012 2:06 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] RTP media issue What version of FS are you running? Do you have the debug logs of those calls? you could try using the jitterbuffer. in the inbound DP to FS *before* you answer. Also it looks a little odd to me in this trace if this is the same call, it seems like you answered the call before placing the call to the phone and that phone never answers.... On Thu, May 24, 2012 at 9:51 PM, Nathan Downes wrote: > Hi, > > enable-soa > > > > Set the value to "false" to diable SIP SOA from sofia to tell sofia > not to touch the exchange of SDP > > I don't think this is related to the exchange of an SDP message.. Can > you elaborate more before I try it? I can't make things worse or > change things I don't understand. > > ________________________________ > From: djbinter at gmail.com > To: freeswitch-users at lists.freeswitch.org > CC: nathan at nortec.com.au > Subject: Re: [Freeswitch-users] RTP media issue > Date: Fri, 25 May 2012 11:19:46 +1000 > > > > > > Sent from my iPad > > On May 24, 2012, at 5:01 PM, Nathan Downes wrote: > > Hi, > > I had previous reported an issue with poor voice quality, appearing to > stem from occasion wrong timestamps coming from provider, but the end > user's experience was much worse than what I could see/hear in the trace. > > I have finally captured an event inbound and outbound. The thing I > don't understand is I thought even though FS proxied the media it > didn't touch it or change anything, but it appears it is. > > The 2 traces are http://www.nortec.com.au/inbound.pcap.gz and > http://www.nortec.com.au/outbound.pcap.gz > > Inbound is from my trunk provider to FS box and outbound is FS box to > ATA in FTTH GPON. > > The event I am talking about, if both traces are open, is in the > inbound one inbetween packet 8114 and 8117 the provider drops a packet > or I don't receive it. In the corresponding outbound trace, between > packet 8144 and 8152, it appears FS misses a whole heap of packets > (.1 seconds) between > 8146 and 8152 then it increases the timestamp only by 40 rather than > 160 on packet 8152. This seems to not affect SIP phones themselves > but causes issues with the FTTH GPON ATA. > > This causes a gap in the audio for the end user, and when they miss a > high number of packets even though it sounds good on the inbound trace > the end users experience is horrible. This trace is actually a good > one, but the wrong timestamp can occur once per second, causing end > user to lose 10%+ of incoming audio only. The issue only affects the > audio coming from provider to FS to end user. > > I am chasing it up with the voice provider to try and eliminate the > occasional packet loss, but if I could stop/fix FS from doing its > adjustment/gap/something the end user wouldn't even notice it. > > > > ______________________________________________________________________ > ___ > > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org > > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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 at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4fbfffef32764666710318! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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 at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120529/0de5144f/attachment-0001.html From Nabble_01394 at slickdeals.endjunk.com Tue May 29 03:13:49 2012 From: Nabble_01394 at slickdeals.endjunk.com (mazilo) Date: Mon, 28 May 2012 16:13:49 -0700 (PDT) Subject: [Freeswitch-users] Bobsled Message-ID: <1338246829179-7579077.post@n2.nabble.com> Is is possible to hack http://bobsled.com BobSled so that it will work with FS (like Google Voice)? ----- FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3 Watts of electricity. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Bobsled-tp7579077.html Sent from the freeswitch-users mailing list archive at Nabble.com. From bdfoster at endigotech.com Tue May 29 04:30:53 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Mon, 28 May 2012 20:30:53 -0400 Subject: [Freeswitch-users] Bobsled In-Reply-To: <1338246829179-7579077.post@n2.nabble.com> References: <1338246829179-7579077.post@n2.nabble.com> Message-ID: You might be able to but many are going to be hesitant about answering that question. Brian Foster Endigo Computer LLC Sent from a mobile device. On May 28, 2012 7:14 PM, "mazilo" wrote: > Is is possible to hack http://bobsled.com BobSled so that it will work > with > FS (like Google Voice)? > > ----- > FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3 > Watts of electricity. > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Bobsled-tp7579077.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120528/f5f9007e/attachment.html From saami_mh at ymail.com Tue May 29 10:19:13 2012 From: saami_mh at ymail.com (Samira Mh) Date: Mon, 28 May 2012 23:19:13 -0700 (PDT) Subject: [Freeswitch-users] "bash" programming on freeswitch? In-Reply-To: References: <1338230128.40509.YahooMailNeo@web120104.mail.ne1.yahoo.com> <1338232857.21351.YahooMailNeo@web120102.mail.ne1.yahoo.com> Message-ID: <1338272353.11657.YahooMailNeo@web120104.mail.ne1.yahoo.com> Hello guys, thanks (Avi,Brian)?alot for your help, ________________________________ From: Avi Marcus To: FreeSWITCH Users Help Sent: Tuesday, May 29, 2012 12:05 AM Subject: Re: [Freeswitch-users] "bash" programming on freeswitch? Lua is good for this. Or any language over ESL or event socket outbound, which might be the most comparable to running an external script on AGI. -Avi On Mon, May 28, 2012 at 10:20 PM, Samira Mh wrote: Hi Brian, >thanks for your reply, >but i want to do more things on freeswitch using Bash script on AGI, (like asterisk) >i want to run for/while loop , if-else condition usin bash and use of the result of running AGI on freeswitch dialplan >?On asterisk the dial plan looks as follows > >exten => 1500,1,AGI(agi://192.168.0.131/hello.agi). > >I need something similar to this. How do i do this from freeswitch? > >thanks alot > > > >________________________________ > From: Brian Foster >To: FreeSWITCH Users Help >Sent: Monday, May 28, 2012 11:18 PM >Subject: Re: [Freeswitch-users] "bash" programming on freeswitch? > > > >system script.sh arg1 arg2 argN >No FS library though so you can't actually control calls. >-BDF > >Brian Foster >Endigo Computer LLC >Sent from a mobile device. >On May 28, 2012 2:36 PM, "Samira Mh" wrote: > >how to run bash script programming in freeswitch using AGI (like freeswitch)? >>i think freeswitch support JAVA/PERL/PYTHON/RUBY programming >>_________________________________________________________________________ >>Professional FreeSWITCH Consulting Services: >>consulting at freeswitch.org >>http://www.freeswitchsolutions.com >> >> >> >> >>Official FreeSWITCH Sites >>http://www.freeswitch.org >>http://wiki.freeswitch.org >>http://www.cluecon.com >> >>Join Us At ClueCon - Aug 7-9, 2012 >> >>FreeSWITCH-users mailing list >>FreeSWITCH-users at 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 at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >Join Us At ClueCon - Aug 7-9, 2012 > >FreeSWITCH-users mailing list >FreeSWITCH-users at 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 at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >Join Us At ClueCon - Aug 7-9, 2012 > >FreeSWITCH-users mailing list >FreeSWITCH-users at 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 at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120528/7e8acc7a/attachment.html From ocset at the800group.com Tue May 29 10:33:47 2012 From: ocset at the800group.com (ocset) Date: Tue, 29 May 2012 14:33:47 +0800 Subject: [Freeswitch-users] Bridge incoming call to external number In-Reply-To: <20120525185002.0bc97db6@mail.tritonwest.net> References: <20120525185002.0bc97db6@mail.tritonwest.net> Message-ID: <4FC46DCB.3000305@the800group.com> Dave Thanks for the reply. Isolation testing - If I make or receive a call using either of the two lines (one line at a time), then I don't have any issues with call disconnection. It is only when both lines are used together that this occurs. Also, if I use a SIP gateway(PennyTel) in place of one of the lines for either leg of the bridge, then everything works fine. I have also tested using a separate FXO device(SPA3102) just to see if the problem may be related to the GXW4104 but that had the same end result. Is there a way to set up a "session-timeout" or "rtp-timeout-sec" for only this connection/bridge? Would that work? Thanks in advance Regards O On 26/05/12 02:50, Dave R. Kompel wrote: > This is not a FreeSWITCH issue. A lot of FXO terminal adaptors do not > detect CPC right. In some cases they don't even do it on outgoing > calls. Make sure your device is set to detect CPC (Calling Party > Control) signalls form the phone line. Make sure the device you are > using cal do that too. > --Dave > > ------------------------------------------------------------------------ > *From:* ocset [mailto:ocset at the800group.com] > *To:* FreeSWITCH Users Help > [mailto:freeswitch-users at lists.freeswitch.org] > *Sent:* Fri, 25 May 2012 00:41:54 -0700 > *Subject:* [Freeswitch-users] Bridge incoming call to external number > > Hi > > Setup > Windows 7 > Freeswitch > GXW4104 FXO > 2 x POTS lines > > I have setup Freeswitch to bridge an incoming call to an external > mobile > phone using a GXW4104 FXO device. The bridge works great but I am > having > a real headache with the connection not being disconnected when the > users hang up. > > There is no disconnection issue when a call comes in on either > line and > is connected to an internal extension (like ext 1001). It is when one > line is bride to the second that the issue occurs. > > > > > > > data="sofia/internal/0766666666 at 192.168.1.160 > :5060 > > > > > > > 1. Would this be a limitation of the GXW4104 > 2. Is it possible to write a "lua" script that could manage the > disconnect some how? > 3. If I enable "Polarity Reversal" under "FXO Termination" then the > disconnect works but I have issues with sound and the system not > registering when someone picks up. > > Please help me fix/diagnose this problem. > > Thanks > O > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120529/2e187266/attachment-0001.html From dave at copycall.com Tue May 29 01:53:15 2012 From: dave at copycall.com (copycall) Date: Mon, 28 May 2012 14:53:15 -0700 Subject: [Freeswitch-users] freeswitch - sangoma CAS T1 - a101 configuration In-Reply-To: References: Message-ID: moises, thank you for your response. the T1 linecard at the end of this call is CAS T1, with B8ZS, and ESF, extended super frame, with E&M trunk signaling. as i understand it, this is not robbed bit signaling, but a later variation of it. i will paste the sangoma linecard config, below: ######################################################################## # Sangoma Wanpipe # # Dahdi/Zaptel/SMG/TDMAPI/BOOT Configuration Script # # v2.39 # # Sangoma Technologies Inc. # # Copyright(c) 2009. # ######################################################################## Would you like to change FreeSWITCH Configuration Directory? Default: /usr/local/freeswitch/conf 1. NO 2. YES [1-2, ENTER='NO']:NO Error: Invalid option, input an integer [1-2, ENTER='NO']:1 --------------------------------------------- Configuring T1/E1 cards [A101/A102/A104/A108] --------------------------------------------- A101 detected on slot:4 bus:3 ----------------------------------------------------------- Configuring port 1 on A101 slot:4 bus:3. ----------------------------------------------------------- Select media type for AFT-A101 on port 1 [slot:4 bus:3 span:1] 1. T1 2. E1 3. Unused 4. Exit [1-4]:1 Configuring port 1 on AFT-A101 as: T1, coding:B8ZS, framing:ESF. 1. YES - Keep these settings 2. NO - Configure line coding and framing [1-2, ENTER='YES']:YES Error: Invalid option, input an integer [1-2, ENTER='YES']:1 Select clock for AFT-A101 on port 1 [slot:4 bus:3 span:1] 1. NORMAL 2. MASTER [1-2]:1 Select Switchtype for AFT-A101 on port 1 [slot:4 bus:3 span:1] 1. National 2. Nortel DMS100 3. Lucent 5ESS 4. Lucent 4ESS [1-4]:1 Select signalling type for AFT-A101 on port 1 [slot:4 bus:3 span:1] 1. PRI CPE 2. PRI NET [1-2]:1 Select dialplan context for AFT-A101 on port 1 1. default 2. public 3. Custom [1-3]:1 Input the dialing group for this port : 1 Would you like to enable hardware DTMF detection? 1. YES 2. NO [1-2, ENTER='YES']:1 Would you like to enable hardware fax detection? 1. YES 2. NO [1-2, ENTER='NO']:1 Port 1 on AFT-A101 configuration complete... Press any key to continue: T1/E1 card configuration complete. Press any key to continue: ------------------------------------ Configuring ISDN BRI cards [A500/B700] ------------------------------------ No Sangoma ISDN BRI cards detected Press any key to continue: ------------------------------------ Configuring GSM cards [W400] ------------------------------------ No Sangoma GSM cards detected Press any key to continue: ------------------------------------ Configuring analog cards [A200/A400/B600/B610/B700/B800] ------------------------------------ ------------------------------------ Configuring USB devices [U100] ------------------------------------ ################################################################### # SUMMARY # ################################################################### 1 T1/E1 port(s) detected, 1 configured 0 ISDN BRI port(s) detected, 0 configured 0 analog card(s) detected, 0 configured 0 GSM card(s) detected, 0 configured 0 usb device(s) detected, 0 configured Configurator will create the following files: 1. Wanpipe config files in /etc/wanpipe 2. freetdm config file /usr/local/freeswitch/conf/freetdm.conf 3. freetdm_xml config file /usr/local/freeswitch/conf/freetdm.conf.xml Your configuration has been saved in /etc/wanpipe/debug-2012-05-18.tgz. When requesting support, email this file to techdesk at sangoma.com ################################################################### Configuration Complete! Please select following: 1. YES - Continue 2. NO - Exit [1-2]:1 Wanpipe configuration complete: choose action 1. Save cfg: Stop Wanpipe now 2. Do not save cfg: Exit [1-2]:1 sh: asterisk: not found sh: asterisk: not found Stopping Wanpipe... Removing old configuration files... Copying new Wanpipe configuration files... Copying new freetdm configuration files (/usr/local/freeswitch/conf/freetdm.conf)... Copying new freetdm configuration files (/usr/local/freeswitch/conf/autoload_configs/freetdm.conf.xml)... Wanrouter start complete... Current boot level is 2 Wanrouter boot scripts configuration... Removing existing wanrouter boot scripts...OK Would you like wanrouter to start on system boot? 1. YES 2. NO [1-2]:1 Verifying Zaptel boot scripts... Enabling wanrouter init scripts ...(start:8, stop:91) Sangoma cards configuration complete, exiting... root at copycall:~# thanks, dave cook On Mon, May 28, 2012 at 12:24 PM, Moises Silva wrote: > On Mon, May 28, 2012 at 12:49 AM, copycall wrote: > >> curriegrad2004, >> >> thanks for responding. >> >> unfortunately, the sangoma tech support has not been as crack as usual. >> >> i can go into the specifics, but it probably isn't necessary in a public >> forum. i'm not looking to piss anyone off. >> >> if you have you actually configured a a101 sangoma card for a legacy CAS >> T1 card, i would greatly appreciate your assistance. >> >> > > It's been a while since I configured for RBS, but I may be able to give > you some pointers. Where are you at? did you have the drivers compiled? > > http://wiki.sangoma.com/wanpipe-freeswitch-ftdm > > As that web page states, RBS with FreeSWITCH is only supported using > Wanpipe + DAHDI, not in native Wanpipe mode. > http://wiki.sangoma.com/FreeSWITCH-Dahdi-Mode > > The module you want to use is ftmod_analog > > > *Moises Silva > **Manager, Software Engineering*** > > msilva at sangoma.com > > Sangoma Technologies > > 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada > > > t. +1 800 388 2475 (N. America) > > t. +1 905 474 1990 x128 > > f. +1 905 474 9223 > > > > ** > > Products > | Solutions > | Events > | Contact > | Wiki > | Facebook > | Twitter`| > | YouTube > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120528/4522f54a/attachment.html From saami_mh at ymail.com Tue May 29 13:38:35 2012 From: saami_mh at ymail.com (Samira Mh) Date: Tue, 29 May 2012 02:38:35 -0700 (PDT) Subject: [Freeswitch-users] how to run Lua commands in freeswitch? Message-ID: <1338284315.78221.YahooMailNeo@web120101.mail.ne1.yahoo.com> i have created the file named "helloworld.lua" like this: session:answer(); session:streamFile("blah.wav"); session:hangup(); and it working correctly, **now i want to use lua command -not freeswitch command -in lua? for example run: op='+' if op == "+" then r = a + b elseif op == "-" then r = a - b elseif op == "*" then r = a*b elseif op == "/" then r = a/b else error("invalid operation") end but how can do that? thanks alot -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120529/3ae47ef9/attachment-0001.html From govoiper at gmail.com Tue May 29 13:54:10 2012 From: govoiper at gmail.com (SamyGo) Date: Tue, 29 May 2012 14:54:10 +0500 Subject: [Freeswitch-users] how to run Lua commands in freeswitch? In-Reply-To: <1338284315.78221.YahooMailNeo@web120101.mail.ne1.yahoo.com> References: <1338284315.78221.YahooMailNeo@web120101.mail.ne1.yahoo.com> Message-ID: Thats a very good question, let me show you how you can do it. session:answer(); session:streamFile("blah.wav"); op='+' if op == "+" then r = a + b session:streamFile("finally-I-did-this.wav"); elseif op == "-" then r = a - b session:streamFile("It-was-vry-difficult.wav"); elseif op == "*" then r = a*b session:streamFile("just-kidding.wav"); elseif op == "/" then r = a/b session:streamFile("only-took-5-hrs-for-this.wav"); else error("invalid operation") end session:hangup(); But whatever you want to do if it requires you to include some header/library on top don't forget to add it. Regards, Sammy Go. On Tue, May 29, 2012 at 2:38 PM, Samira Mh wrote: > i have created the file named "helloworld.lua" like this: > session:answer(); > session:streamFile("blah.wav"); > session:hangup(); > and it working correctly, > **now i want to use lua command -not freeswitch command -in lua > for example run: > > op='+' > > if op == "+" then > r = a + b > elseif op == "-" then > r = a - b > elseif op == "*" then > r = a*b > elseif op == "/" then > r = a/b > else > error("invalid operation") > end > > but how can do that? > > thanks alot > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120529/73be099e/attachment.html From saami_mh at ymail.com Tue May 29 14:25:15 2012 From: saami_mh at ymail.com (Samira Mh) Date: Tue, 29 May 2012 03:25:15 -0700 (PDT) Subject: [Freeswitch-users] how to run Lua commands in freeswitch? In-Reply-To: References: <1338284315.78221.YahooMailNeo@web120101.mail.ne1.yahoo.com> Message-ID: <1338287115.30092.YahooMailNeo@web120102.mail.ne1.yahoo.com> hi SamyGo, thanks for your help, it is kind of you to answer my questions,but i have one question again. i am going to put the result of "select query" on the text file ?and insert the result in array ?on lua how can i do that ? is it possible to use lua commands directly? i know how to connect to database ,but i don't know how to put the select query result in array or text file and match the number i want on the array or text file? ________________________________ From: SamyGo To: FreeSWITCH Users Help Sent: Tuesday, May 29, 2012 2:24 PM Subject: Re: [Freeswitch-users] how to run Lua commands in freeswitch? Thats a very good question, let me show you how you can do it. session:answer(); session:streamFile("blah.wav"); op='+' if op == "+" then r = a + b session:streamFile("finally-I-did-this.wav");? elseif op == "-" then r = a - b session:streamFile("It-was-vry-difficult.wav"); ? elseif op == "*" then r = a*b session:streamFile("just-kidding.wav"); ? elseif op == "/" then r = a/b session:streamFile("only-took-5-hrs-for-this.wav"); ? else error("invalid operation") end session:hangup(); But whatever you want to do if it requires you to include some header/library on top don't forget to add it.? Regards, Sammy Go. On Tue, May 29, 2012 at 2:38 PM, Samira Mh wrote: i have created the file named "helloworld.lua" like this: >session:answer(); >session:streamFile("blah.wav"); >session:hangup(); >and it working correctly, >**now i want to use lua command -not freeswitch command -in lua? >for example run: > > >op='+' >if op == "+" then r = a + b elseif op == "-" then r = a - b elseif op == "*" then r = a*b elseif op == "/" then r = a/b else error("invalid operation") end >but how can do that? >thanks alot >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >Join Us At ClueCon - Aug 7-9, 2012 > >FreeSWITCH-users mailing list >FreeSWITCH-users at 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 at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120529/b0df76dd/attachment.html From andy at fabulous4.co.uk Tue May 29 15:00:23 2012 From: andy at fabulous4.co.uk (Andy Ayers) Date: Tue, 29 May 2012 12:00:23 +0100 Subject: [Freeswitch-users] Best database setup for high volume In-Reply-To: References: <005401cd38be$2a9434b0$7fbc9e10$@fabulous4.co.uk> <01ef01cd390a$884b60f0$98e222d0$@fabulous4.co.uk> <008e01cd3990$0cefac00$26cf0400$@fabulous4.co.uk> <00c901cd3cb4$94775200$bd65f600$@fabulous4.co.uk> Message-ID: <00c101cd3d8a$40d9a900$c28cfb00$@fabulous4.co.uk> Many thanks Wesley that would be extremely helpful. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Wesley Akio Sent: 28 May 2012 13:42 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Best database setup for high volume Pretty easy, just format a ramdisk and mount it... Will send you my init script as soon as I get to the office... Sent from mobile, sorry for the typos.... Em 28/05/2012 06:32, "Andy Ayers" escreveu: Disk at the moment. Is it easy to move it to memory? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Wesley Akio Sent: 24 May 2012 12:58 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Best database setup for high volume Out of curiosity, do you run sqlite from memory or disk? Sent from mobile, sorry for the typos.... Em 24/05/2012 06:33, "Andy Ayers" escreveu: Hi Avi, Yes sadly I've been through all those options and am still getting the error. Tried OPTION and OPTIONS but it doesn't seem to make any difference. There seem to be a few other folks on the users list that have had the same problem and not managed to find a solution. Cheers Andy From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi Marcus Sent: 23 May 2012 19:10 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Best database setup for high volume On Wed, May 23, 2012 at 8:36 PM, Andy Ayers wrote: Many thanks for your reply Avi, that's very helpful. I've disabled the sql scoreboard and presence and all seems to be ok so those should help. My traffic consists of relatively high volumes of incoming and outgoing calls but all one sided. i.e. It's the switch taking the caller through an ivr 'form' so there are no 2-way calls, no bridging, forwarding or directing of calls at all. The only time multiple callers are involved is when we use it for conferencing which is only small scale at the moment. It's the database corruption issue I'm really interested in solving so I'd like to get the odbc connection working if possible. Any info you can provide on how you got that to work would be greatly appreciated. Like I say I hit 2 problems: Firstly on load freeswitch complains that it can't run multiple statements. I've tried everything that's recommended in the MyOdbc docs including setting the options in odbc.ini but still get the error. Some posts talk about needing to use to _r version of the driver but I don't have that on my system. I'm running Debian if that's significant. Did you set: OPTIONS = 67108864 in the odbc.ini, for FLAG_MULTI_STATEMENTS? (Supposedly it's "OPTION" on centos) http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core -Avi The second issue was a message popping up in the logs every few seconds saying: Error in my_thread_global_end() nn threads didn't exit. Did you encounter either of these problems or find ways round them? Once again many thanks for any help. Cheers Andy From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi Marcus Sent: 23 May 2012 09:52 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Best database setup for high volume Is your issue the CDRs & Voicemail or session count, current calls, recovery data.. You can make sure track-calls is off... -nosql -- disable internal sql scoreboard I'm not sure if that kills presence or not. Disable presence if you don't need it - it's a real usage hog. I have odbc to mysql but I made calls,channels,sip_dialogs, sip_subscriptions, etc into memory tables a few months ago. (I left sip_registrations as non-memory for persistence of a sort) -Avi On Wed, May 23, 2012 at 11:29 AM, Andy Ayers wrote: Hi, Can anyone tell me what the best database setup is for dealing with high call volumes? The background: I've been running with the standard SQLite system for about 3 years without issue but recently am getting a lot of database corruption errors ('database disk image is malformed'). Easily solved by deleting the db and restarting but on occasion it brings my switch down. I've tried upgrading to odbc and mysql but hit 2 problems: FLAG_MULTIPLE_STATEMENTS Error in my_thread_global_end() Both of which are mentioned in the user groups but not with any solutions that worked. I've just upgraded to version 1.2 so this may solve the corruption problems but would really like to get my system set up to handle as much traffic as possible. Any advice or suggestions much appreciated. Kind regards Andy _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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 at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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 at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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 at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120529/21693120/attachment-0001.html From sherifomran2000 at yahoo.com Tue May 29 15:45:13 2012 From: sherifomran2000 at yahoo.com (Sherif Omran) Date: Tue, 29 May 2012 04:45:13 -0700 (PDT) Subject: [Freeswitch-users] add international prefix to gateway caller ID Message-ID: <1338291913.82803.YahooMailClassic@web110809.mail.gq1.yahoo.com> Hello guys, I have a german gateway among other gateways going to the same dialling plan and it does not send the interational caller ID prefix so I want to add 0049 and strip the first 0 in the caller ID sent. But Is it possible to add a condition inside the gateway setting as follows (action applicaiton set ... etc)? ??????? ??????? ??????? ??????? ??????? ??????? ??? ??????? ????? References: <005401cd38be$2a9434b0$7fbc9e10$@fabulous4.co.uk> <01ef01cd390a$884b60f0$98e222d0$@fabulous4.co.uk> <008e01cd3990$0cefac00$26cf0400$@fabulous4.co.uk> <00c901cd3cb4$94775200$bd65f600$@fabulous4.co.uk> <00c101cd3d8a$40d9a900$c28cfb00$@fabulous4.co.uk> Message-ID: Hi Andy, This is as simple as it gets, and performace of sqlite is inarguably superior... mkfs.ext2 /dev/ram0 &2> /dev/null mount /dev/ram0 /usr/local/freeswitch/db &2> /dev/null exec /usr/local/freeswitch/bin/freeswitch -nc -nonat Works for me and it has for quite some time... Wesley Akio TuntsCorp.com On Tue, May 29, 2012 at 8:00 AM, Andy Ayers wrote: > Many thanks Wesley that would be extremely helpful.**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Wesley Akio > *Sent:* 28 May 2012 13:42 > > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Best database setup for high volume**** > > ** ** > > Pretty easy, just format a ramdisk and mount it... Will send you my init > script as soon as I get to the office...**** > > Sent from mobile, sorry for the typos....**** > > Em 28/05/2012 06:32, "Andy Ayers" escreveu:**** > > Disk at the moment. Is it easy to move it to memory?**** > > **** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Wesley Akio > *Sent:* 24 May 2012 12:58 > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Best database setup for high volume**** > > **** > > Out of curiosity, do you run sqlite from memory or disk?**** > > Sent from mobile, sorry for the typos....**** > > Em 24/05/2012 06:33, "Andy Ayers" escreveu:**** > > Hi Avi,**** > > **** > > Yes sadly I?ve been through all those options and am still getting the > error. Tried OPTION and OPTIONS but it doesn?t seem to make any difference. > There seem to be a few other folks on the users list that have had the same > problem and not managed to find a solution.**** > > **** > > Cheers**** > > Andy**** > > **** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Avi Marcus > *Sent:* 23 May 2012 19:10 > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Best database setup for high volume**** > > **** > > On Wed, May 23, 2012 at 8:36 PM, Andy Ayers wrote:* > *** > > Many thanks for your reply Avi, that?s very helpful.**** > > **** > > I?ve disabled the sql scoreboard and presence and all seems to be ok so > those should help.**** > > **** > > My traffic consists of relatively high volumes of incoming and outgoing > calls but all one sided. i.e. It?s the switch taking the caller through an > ivr ?form? so there are no 2-way calls, no bridging, forwarding or > directing of calls at all. The only time multiple callers are involved is > when we use it for conferencing which is only small scale at the moment.** > ** > > **** > > It?s the database corruption issue I?m really interested in solving so I?d > like to get the odbc connection working if possible. Any info you can > provide on how you got that to work would be greatly appreciated. Like I > say I hit 2 problems:**** > > **** > > Firstly on load freeswitch complains that it can?t run multiple > statements. I?ve tried everything that?s recommended in the MyOdbc docs > including setting the options in odbc.ini but still get the error. Some > posts talk about needing to use to _r version of the driver but I don?t > have that on my system. I?m running Debian if that?s significant.**** > > Did you set:**** > > OPTIONS = 67108864**** > > in the odbc.ini, for FLAG_MULTI_STATEMENTS? (Supposedly it's "OPTION" on > centos)**** > > http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core**** > > **** > > -Avi**** > > **** > > The second issue was a message popping up in the logs every few seconds > saying: Error in my_thread_global_end() nn threads didn?t exit.**** > > Did you encounter either of these problems or find ways round them?**** > > **** > > Once again many thanks for any help.**** > > **** > > Cheers**** > > Andy**** > > **** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Avi Marcus > *Sent:* 23 May 2012 09:52 > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Best database setup for high volume**** > > **** > > Is your issue the CDRs & Voicemail or session count, current calls, > recovery data..**** > > You can make sure track-calls is off... **** > > **** > > **** > > **** > > **** > > -nosql -- disable internal sql scoreboard**** > > I'm not sure if that kills presence or not.**** > > **** > > Disable presence if you don't need it - it's a real usage hog.**** > > **** > > I have odbc to mysql but I made calls,channels,sip_dialogs, > sip_subscriptions, etc into memory tables a few months ago.**** > > **** > > (I left sip_registrations as non-memory for persistence of a sort)**** > > **** > > -Avi**** > > **** > > On Wed, May 23, 2012 at 11:29 AM, Andy Ayers wrote: > **** > > Hi,**** > > **** > > Can anyone tell me what the best database setup is for dealing with high > call volumes?**** > > **** > > The background:**** > > **** > > I?ve been running with the standard SQLite system for about 3 years > without issue but recently am getting a lot of database corruption errors > (?database disk image is malformed?). Easily solved by deleting the db and > restarting but on occasion it brings my switch down.**** > > **** > > I?ve tried upgrading to odbc and mysql but hit 2 problems:**** > > **** > > FLAG_MULTIPLE_STATEMENTS**** > > Error in my_thread_global_end()**** > > **** > > Both of which are mentioned in the user groups but not with any solutions > that worked.**** > > **** > > I?ve just upgraded to version 1.2 so this may solve the corruption > problems but would really like to get my system set up to handle as much > traffic as possible.**** > > **** > > Any advice or suggestions much appreciated.**** > > **** > > Kind regards**** > > Andy**** > > **** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120529/7a569306/attachment-0001.html From avi at avimarcus.net Tue May 29 16:53:58 2012 From: avi at avimarcus.net (Avi Marcus) Date: Tue, 29 May 2012 15:53:58 +0300 Subject: [Freeswitch-users] Best database setup for high volume In-Reply-To: References: <005401cd38be$2a9434b0$7fbc9e10$@fabulous4.co.uk> <01ef01cd390a$884b60f0$98e222d0$@fabulous4.co.uk> <008e01cd3990$0cefac00$26cf0400$@fabulous4.co.uk> <00c901cd3cb4$94775200$bd65f600$@fabulous4.co.uk> <00c101cd3d8a$40d9a900$c28cfb00$@fabulous4.co.uk> Message-ID: Do you do anything to keep the voicemail DB on the actual hard drive rather than also in ramdisk? I presume that in the case of a system crash, everything in the ramdisk is lost... -Avi On Tue, May 29, 2012 at 2:56 PM, Wesley Akio wrote: > Hi Andy, > > This is as simple as it gets, and performace of sqlite is inarguably > superior... > > mkfs.ext2 /dev/ram0 &2> /dev/null > mount /dev/ram0 /usr/local/freeswitch/db &2> /dev/null > exec /usr/local/freeswitch/bin/freeswitch -nc -nonat > > Works for me and it has for quite some time... > > Wesley Akio > TuntsCorp.com > > > > On Tue, May 29, 2012 at 8:00 AM, Andy Ayers wrote: > >> Many thanks Wesley that would be extremely helpful.**** >> >> ** ** >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Wesley Akio >> *Sent:* 28 May 2012 13:42 >> >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] Best database setup for high volume**** >> >> ** ** >> >> Pretty easy, just format a ramdisk and mount it... Will send you my init >> script as soon as I get to the office...**** >> >> Sent from mobile, sorry for the typos....**** >> >> Em 28/05/2012 06:32, "Andy Ayers" escreveu:**** >> >> Disk at the moment. Is it easy to move it to memory?**** >> >> **** >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Wesley Akio >> *Sent:* 24 May 2012 12:58 >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] Best database setup for high volume**** >> >> **** >> >> Out of curiosity, do you run sqlite from memory or disk?**** >> >> Sent from mobile, sorry for the typos....**** >> >> Em 24/05/2012 06:33, "Andy Ayers" escreveu:**** >> >> Hi Avi,**** >> >> **** >> >> Yes sadly I?ve been through all those options and am still getting the >> error. Tried OPTION and OPTIONS but it doesn?t seem to make any difference. >> There seem to be a few other folks on the users list that have had the same >> problem and not managed to find a solution.**** >> >> **** >> >> Cheers**** >> >> Andy**** >> >> **** >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Avi Marcus >> *Sent:* 23 May 2012 19:10 >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] Best database setup for high volume**** >> >> **** >> >> On Wed, May 23, 2012 at 8:36 PM, Andy Ayers wrote: >> **** >> >> Many thanks for your reply Avi, that?s very helpful.**** >> >> **** >> >> I?ve disabled the sql scoreboard and presence and all seems to be ok so >> those should help.**** >> >> **** >> >> My traffic consists of relatively high volumes of incoming and outgoing >> calls but all one sided. i.e. It?s the switch taking the caller through an >> ivr ?form? so there are no 2-way calls, no bridging, forwarding or >> directing of calls at all. The only time multiple callers are involved is >> when we use it for conferencing which is only small scale at the moment.* >> *** >> >> **** >> >> It?s the database corruption issue I?m really interested in solving so >> I?d like to get the odbc connection working if possible. Any info you can >> provide on how you got that to work would be greatly appreciated. Like I >> say I hit 2 problems:**** >> >> **** >> >> Firstly on load freeswitch complains that it can?t run multiple >> statements. I?ve tried everything that?s recommended in the MyOdbc docs >> including setting the options in odbc.ini but still get the error. Some >> posts talk about needing to use to _r version of the driver but I don?t >> have that on my system. I?m running Debian if that?s significant.**** >> >> Did you set:**** >> >> OPTIONS = 67108864**** >> >> in the odbc.ini, for FLAG_MULTI_STATEMENTS? (Supposedly it's "OPTION" >> on centos)**** >> >> http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core**** >> >> **** >> >> -Avi**** >> >> **** >> >> The second issue was a message popping up in the logs every few seconds >> saying: Error in my_thread_global_end() nn threads didn?t exit.**** >> >> Did you encounter either of these problems or find ways round them?**** >> >> **** >> >> Once again many thanks for any help.**** >> >> **** >> >> Cheers**** >> >> Andy**** >> >> **** >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Avi Marcus >> *Sent:* 23 May 2012 09:52 >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] Best database setup for high volume**** >> >> **** >> >> Is your issue the CDRs & Voicemail or session count, current calls, >> recovery data..**** >> >> You can make sure track-calls is off... **** >> >> **** >> >> **** >> >> **** >> >> **** >> >> -nosql -- disable internal sql scoreboard**** >> >> I'm not sure if that kills presence or not.**** >> >> **** >> >> Disable presence if you don't need it - it's a real usage hog.**** >> >> **** >> >> I have odbc to mysql but I made calls,channels,sip_dialogs, >> sip_subscriptions, etc into memory tables a few months ago.**** >> >> **** >> >> (I left sip_registrations as non-memory for persistence of a sort)**** >> >> **** >> >> -Avi**** >> >> **** >> >> On Wed, May 23, 2012 at 11:29 AM, Andy Ayers >> wrote:**** >> >> Hi,**** >> >> **** >> >> Can anyone tell me what the best database setup is for dealing with high >> call volumes?**** >> >> **** >> >> The background:**** >> >> **** >> >> I?ve been running with the standard SQLite system for about 3 years >> without issue but recently am getting a lot of database corruption errors >> (?database disk image is malformed?). Easily solved by deleting the db and >> restarting but on occasion it brings my switch down.**** >> >> **** >> >> I?ve tried upgrading to odbc and mysql but hit 2 problems:**** >> >> **** >> >> FLAG_MULTIPLE_STATEMENTS**** >> >> Error in my_thread_global_end()**** >> >> **** >> >> Both of which are mentioned in the user groups but not with any solutions >> that worked.**** >> >> **** >> >> I?ve just upgraded to version 1.2 so this may solve the corruption >> problems but would really like to get my system set up to handle as much >> traffic as possible.**** >> >> **** >> >> Any advice or suggestions much appreciated.**** >> >> **** >> >> Kind regards**** >> >> Andy**** >> >> **** >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120529/3a0e29cd/attachment-0001.html From avi at avimarcus.net Tue May 29 16:55:02 2012 From: avi at avimarcus.net (Avi Marcus) Date: Tue, 29 May 2012 15:55:02 +0300 Subject: [Freeswitch-users] add international prefix to gateway caller ID In-Reply-To: <1338291913.82803.YahooMailClassic@web110809.mail.gq1.yahoo.com> References: <1338291913.82803.YahooMailClassic@web110809.mail.gq1.yahoo.com> Message-ID: Best I can think of: mod_lcr has a regex you can apply to the different gateways. You need to set it per row, however, rather than per provider. -Avi On Tue, May 29, 2012 at 2:45 PM, Sherif Omran wrote: > Hello guys, > > I have a german gateway among other gateways going to the same dialling > plan and it does not send the interational caller ID prefix so I want to > add 0049 and strip the first 0 in the caller ID sent. > > But Is it possible to add a condition inside the gateway setting as > follows (action applicaiton set ... etc)? > > > > > > > data="effective_caller_id_number=0${caller_id_number}"/> > > > I don't want to add it to the dialing plan in order not to affect the > caller id from other gateways. > Thank you > > Sherif > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120529/c40245e2/attachment.html From hiryu23 at gmail.com Tue May 29 13:13:06 2012 From: hiryu23 at gmail.com (hiryu23) Date: Tue, 29 May 2012 02:13:06 -0700 (PDT) Subject: [Freeswitch-users] Subscribe for MWI In-Reply-To: <1338282624250-7579201.post@n2.nabble.com> References: <9D1F6AAA-D64F-45BC-A70D-C6E469D38C30@5ninesolutions.com> <8b33f052-3895-4252-8f47-6ce672e810c6@blur> <850DE19F-0F8B-4E4F-9B22-534311287478@5ninesolutions.com> <1338282624250-7579201.post@n2.nabble.com> Message-ID: <1338282786982-7579202.post@n2.nabble.com> Hi Spencer I am configuring something similar. May i know what is your configuration for Kamailio for routing packets to FreeSwitch? Sorry i am pretty new here. Thanks in advance -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Subscribe-for-MWI-tp7557448p7579202.html Sent from the freeswitch-users mailing list archive at Nabble.com. From kris at kriskinc.com Tue May 29 19:52:44 2012 From: kris at kriskinc.com (Kristian Kielhofner) Date: Tue, 29 May 2012 11:52:44 -0400 Subject: [Freeswitch-users] Best database setup for high volume In-Reply-To: References: <005401cd38be$2a9434b0$7fbc9e10$@fabulous4.co.uk> <01ef01cd390a$884b60f0$98e222d0$@fabulous4.co.uk> <008e01cd3990$0cefac00$26cf0400$@fabulous4.co.uk> <00c901cd3cb4$94775200$bd65f600$@fabulous4.co.uk> <00c101cd3d8a$40d9a900$c28cfb00$@fabulous4.co.uk> Message-ID: This structure of commands is very bizarre... I highly, highly suggest using tmpfs instead of this method. Besides, it's easier: mount -t tmpfs none /usr/local/freeswitch/db On Tue, May 29, 2012 at 7:56 AM, Wesley Akio wrote: > Hi Andy, > > This is as simple as it gets, and performace of sqlite is inarguably > superior... > > mkfs.ext2 /dev/ram0 &2> /dev/null > mount /dev/ram0 /usr/local/freeswitch/db &2> /dev/null > exec /usr/local/freeswitch/bin/freeswitch -nc -nonat > > Works for me and it has for quite some time... > > Wesley Akio > TuntsCorp.com > > > > On Tue, May 29, 2012 at 8:00 AM, Andy Ayers wrote: >> >> Many thanks Wesley that would be extremely helpful. >> >> >> >> From: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Wesley >> Akio >> Sent: 28 May 2012 13:42 >> >> >> To: FreeSWITCH Users Help >> Subject: Re: [Freeswitch-users] Best database setup for high volume >> >> >> >> Pretty easy, just format a ramdisk and mount it... Will send you my init >> script as soon as I get to the office... >> >> Sent from mobile, sorry for the typos.... >> >> Em 28/05/2012 06:32, "Andy Ayers" escreveu: >> >> Disk at the moment. Is it easy to move it to memory? >> >> >> >> From: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Wesley >> Akio >> Sent: 24 May 2012 12:58 >> To: FreeSWITCH Users Help >> Subject: Re: [Freeswitch-users] Best database setup for high volume >> >> >> >> Out of curiosity, do you run sqlite from memory or disk? >> >> Sent from mobile, sorry for the typos.... >> >> Em 24/05/2012 06:33, "Andy Ayers" escreveu: >> >> Hi Avi, >> >> >> >> Yes sadly I?ve been through all those options and am still getting the >> error. Tried OPTION and OPTIONS but it doesn?t seem to make any difference. >> There seem to be a few other folks on the users list that have had the same >> problem and not managed to find a solution. >> >> >> >> Cheers >> >> Andy >> >> >> >> From: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi >> Marcus >> Sent: 23 May 2012 19:10 >> To: FreeSWITCH Users Help >> Subject: Re: [Freeswitch-users] Best database setup for high volume >> >> >> >> On Wed, May 23, 2012 at 8:36 PM, Andy Ayers wrote: >> >> Many thanks for your reply Avi, that?s very helpful. >> >> >> >> I?ve disabled the sql scoreboard and presence and all seems to be ok so >> those should help. >> >> >> >> My traffic consists of relatively high volumes of incoming and outgoing >> calls but all one sided. i.e. It?s the switch taking the caller through an >> ivr ?form? so there are no 2-way calls, no bridging, forwarding or directing >> of calls at all. The only time multiple callers are involved is when we use >> it for conferencing which is only small scale at the moment. >> >> >> >> It?s the database corruption issue I?m really interested in solving so I?d >> like to get the odbc connection working if possible. Any info you can >> provide on how you got that to work would be greatly appreciated. Like I say >> I hit 2 problems: >> >> >> >> Firstly on load freeswitch complains that it can?t run multiple >> statements. I?ve tried everything that?s recommended in the MyOdbc docs >> including setting the options in odbc.ini but still get the error. Some >> posts talk about needing to use to _r version of the driver but I don?t have >> that on my system. I?m running Debian if that?s significant. >> >> Did you set: >> >> OPTIONS ?= 67108864 >> >> in the odbc.ini, for??FLAG_MULTI_STATEMENTS? (Supposedly it's "OPTION" on >> centos) >> >> http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core >> >> >> >> -Avi >> >> >> >> The second issue was a message popping up in the logs every few seconds >> saying: Error in my_thread_global_end() nn threads didn?t exit. >> >> Did you encounter either of these problems or find ways round them? >> >> >> >> Once again many thanks for any help. >> >> >> >> Cheers >> >> Andy >> >> >> >> From: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi >> Marcus >> Sent: 23 May 2012 09:52 >> To: FreeSWITCH Users Help >> Subject: Re: [Freeswitch-users] Best database setup for high volume >> >> >> >> Is your issue the CDRs & Voicemail or session count, current calls, >> recovery data.. >> >> You can make sure track-calls is off... >> >> >> >> >> >> >> >> >> >> -nosql???????????????? -- disable internal sql scoreboard >> >> I'm not sure if that kills presence or not. >> >> >> >> Disable presence if you don't need it - it's a real usage hog. >> >> >> >> I have odbc to mysql but I made?calls,channels,sip_dialogs, >> sip_subscriptions, etc into memory tables a few months ago. >> >> >> >> (I left?sip_registrations as non-memory for persistence of a sort) >> >> >> >> -Avi >> >> >> >> On Wed, May 23, 2012 at 11:29 AM, Andy Ayers wrote: >> >> Hi, >> >> >> >> Can anyone tell me what the best database setup is for dealing with high >> call volumes? >> >> >> >> The background: >> >> >> >> I?ve been running with the standard SQLite system for about 3 years >> without issue but recently am getting a lot of database corruption errors >> (?database disk image is malformed?). Easily solved by deleting the db and >> restarting but on occasion it brings my switch down. >> >> >> >> I?ve tried upgrading to odbc and mysql but hit 2 problems: >> >> >> >> FLAG_MULTIPLE_STATEMENTS >> >> Error in my_thread_global_end() >> >> >> >> Both of which are mentioned in the user groups but not with any solutions >> that worked. >> >> >> >> I?ve just upgraded to version 1.2 so this may solve the corruption >> problems but would really like to get my system set up to handle as much >> traffic as possible. >> >> >> >> Any advice or suggestions much appreciated. >> >> >> >> Kind regards >> >> Andy >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Kristian Kielhofner From marketing at cluecon.com Tue May 29 20:40:23 2012 From: marketing at cluecon.com (Michael Collins) Date: Tue, 29 May 2012 09:40:23 -0700 Subject: [Freeswitch-users] FreeSWITCH Weekly News and Notes Message-ID: Happy Tuesday to all. We hope you enjoyed your holiday weekend. We've had a steady stream of presentations on the FreeSWITCH conference call over the past month. Ken Rice has graciously done two presentations on how to create a dialplan module in C, including a fully functional example module. This code is extremely helpful not just in understanding how to write a dialplan module but in writing any module that exposes new dialplan applications or command line APIs. This week we will take a break from the formal presentations and have an open discussion on any topics of interest to the community. Our ClueCon plans are gearing up as well. We are pleased to announce that Plivo, Inc. is a brand new silver sponsor for this year's event. Plivois an open-source framework for developing Web-based solutions with FreeSWITCH. We are also happy to report that the Illinois Institute of Technology (IIT) is again with us a media sponsor. IIT holds the annual Real-Time Communications conference and expo where academia and industry meet to discuss various aspects of global telecommunications. We invite you to visit our sponsors' Web sites to learn more about what they have to offer. As a reminder, ClueCon still has openings for sponsors and speakers. Please contact us via email or at the phone number below if you have any questions. Visit the registrationpage to get signed up and be sure to book your room at the Wyndham . (And don't forget that the Wyndham will shortly be renamed to the Hyatt Chicago Miracle Mile.) We look forward to meeting everyone in person this August! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE cc12-0529 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120529/81cb263b/attachment-0001.html From sdevoy at bizfocused.com Tue May 29 21:11:02 2012 From: sdevoy at bizfocused.com (Sean Devoy) Date: Tue, 29 May 2012 13:11:02 -0400 Subject: [Freeswitch-users] SCA not working inbound - Multi Domain In-Reply-To: <051501cd3ac2$8a187ee0$9e497ca0$@bizfocused.com> References: <040301cd3aa3$e93705f0$bba511d0$@bizfocused.com> <051501cd3ac2$8a187ee0$9e497ca0$@bizfocused.com> Message-ID: <075e01cd3dbe$06db5350$1491f9f0$@bizfocused.com> BUMP! Anyone have any ideas for me? Any other information I can provide? Thanks. Sean -----Original Message----- From: Sean Devoy [mailto:sdevoy at bizfocused.com] Sent: Friday, May 25, 2012 6:06 PM To: 'FreeSWITCH Users Help' Subject: Re: [Freeswitch-users] SCA not working inbound - Multi Domain Here is the result of select * from sip_subscriptions;" sip|220|fs_lan.bizfocused.com|220|fs_lan.bizfocused.com||call-info|"user" |38e107ab-6cde6635 at 10.10.40.30|"220" ;tag=fd508933c5f5924d|SIP/2.0/UDP 10.10.40.30:5060;branch=z9hG4bK-f5598bfb|1337980638|Cisco/SPA504G-7.4.8a||ex ternal|FreeSwitch1.sumdomain.com|5060|10.10.40.30|-1||"220" ;tag=VrGrXQaOH22R sip|220|fs_lan.bizfocused.com|220|fs_lan.bizfocused.com||call-info|"user" |c08f0c6a-c46e90d2 at 10.10.40.20|"Sean" ;tag=f22a978ae8838032|SIP/2.0/UDP 10.10.40.20:5064;branch=z9hG4bK-905b6faa|1337980643|Cisco/SPA504G-7.4.9c||ex ternal|FreeSwitch1.sumdomain.com|5064|10.10.40.20|-1||"Sean" ;tag=y5VtigPlIghD But it was in: sofia_reg_external.db not internal. I have sorted out all the sip trace data into 2 txt files for the 2 phones involved. They are zipped up at: http://www.bizfocused.com/sip_trace.zip Thank you again for your help. I am way over my head now. -----Original Message----- From: Anthony Minessale [mailto:anthony.minessale at gmail.com] Sent: Friday, May 25, 2012 2:35 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SCA not working inbound - Multi Domain What are the phones putting in the subscribe ? sofia global siptrace on sofia global debug presence|sla then watch for SUBSCRIBE also when you are not using odbc you can get the sql with this app sqlute3 /usr/local/freeswitch/db/sofia_reg_internal.db also try "select * from sip_subscriptions" its all about using the right host name across the board, IP's count as hostnames, they do not magically resolve any dns with SIP On Fri, May 25, 2012 at 1:26 PM, Sean Devoy wrote: > Hi all, > > > > I have a muti-tennnant configuration that is working nicely except for > Shared Call Appearance.? The desktop devices are CISCO 504Gs and they > are configured as described in the FS Wiki as well as Cisco Documentation. > > > > The SCA works perfectly for outbound calls ? if either phone pickups > like 220, the other phones indicator light flashes red.? However, > inbound calls will go to only one of the phones (which one has changed > a few times) and the other phones line still just stays green and does > not ring. > > > > Here is the sip interfaces config: > > > > ??? > > ????? > > ????? > > ????? > > ????? > > ????? > > ????? > > ????? > > ????? > > ????? > > ????? > > ????? > > ????? > > ????? > > ????? > > ????? > > ????? > > ????? > > ????? > > ????? > > ??? > > ? > > > > The directory entry which both phones connect using: > > ??? > > ????? > > ??????? > > ??????? value="410420BLEEP"/> > > ??????? > > ??????? > > ??????? > > ??????? > > ??????? > > ??????? > > ??????? > > ????? > > ????? > > ??????? " > > ??????? > > ??????? value="{presence_id=220 at mydomainname.com}${sofia_contact(220 at mydomainn > ame.com)}"/> > > ??????? > > ????? > > ??? > > > > And the dial plan for ext 220: > > ? > > ??? > > ????? data="effective_caller_id_number=${internal_caller_id_number}"/> > > ????? data="effective_caller_id_name=${internal_caller_id_name}"/> > > ????? > > ????? > > ????? > > ????? data="{sip_invite_domain=mydomainname.com}user/220 at mydomainname.com" > /> > > ?????? > > ????? > > ????? > > ??? > > ? > > > > > > > > I did see this in the wiki > (http://wiki.freeswitch.org/wiki/Shared_Line_Appearance): > > If SLA works for outgoing calls and SLA does not work for inbound > calls to the SLA phones, you may have some presence problem related to > mixed IP and domain names. When using ODBC you may issue the following > SQL statement > > select > sip_to_host,sip_from_user,sip_from_host,hostname,presence_id,call_info > from sip_dialogs; > > But I don?t have ODBC on this server, so I am a little lost. > > > > I have the phones login to domain names, not addresses.? I never refer > to IP addresses in my xml (except gateways addresses).? I am not > trying SLA across domain, only within the same domain. > > > > I hope someone can spot something.? Thanks for your help. > > > > Sean > > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From msc at freeswitch.org Tue May 29 22:40:31 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 29 May 2012 11:40:31 -0700 Subject: [Freeswitch-users] Best database setup for high volume In-Reply-To: References: <005401cd38be$2a9434b0$7fbc9e10$@fabulous4.co.uk> <01ef01cd390a$884b60f0$98e222d0$@fabulous4.co.uk> <008e01cd3990$0cefac00$26cf0400$@fabulous4.co.uk> <00c901cd3cb4$94775200$bd65f600$@fabulous4.co.uk> <00c101cd3d8a$40d9a900$c28cfb00$@fabulous4.co.uk> Message-ID: On Tue, May 29, 2012 at 8:52 AM, Kristian Kielhofner wrote: > This structure of commands is very bizarre... I highly, highly > suggest using tmpfs instead of this method. Besides, it's easier: > > mount -t tmpfs none /usr/local/freeswitch/db > > For future reference, this wiki page mentions tmpfs using fstab: http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations#FreeSWITCH.27s_core.db_I.2FO_bottleneck Same idea, just a different tack to get there. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120529/810753af/attachment.html From anthony.minessale at gmail.com Tue May 29 23:17:48 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 29 May 2012 14:17:48 -0500 Subject: [Freeswitch-users] SCA not working inbound - Multi Domain In-Reply-To: <075e01cd3dbe$06db5350$1491f9f0$@bizfocused.com> References: <040301cd3aa3$e93705f0$bba511d0$@bizfocused.com> <051501cd3ac2$8a187ee0$9e497ca0$@bizfocused.com> <075e01cd3dbe$06db5350$1491f9f0$@bizfocused.com> Message-ID: enter "sofia_contact 220 at mydomainname.com" at your cli and see if you get a large dial string. if you are only getting calls to one phone make sure they all work, its possible you have many phones behind the same nat and only one of them can poke through at a time. On Tue, May 29, 2012 at 12:11 PM, Sean Devoy wrote: > BUMP! > > Anyone have any ideas for me? > > Any other information I can provide? > > Thanks. > Sean > > -----Original Message----- > From: Sean Devoy [mailto:sdevoy at bizfocused.com] > Sent: Friday, May 25, 2012 6:06 PM > To: 'FreeSWITCH Users Help' > Subject: Re: [Freeswitch-users] SCA not working inbound - Multi Domain > > Here is the result of select * from sip_subscriptions;" > sip|220|fs_lan.bizfocused.com|220|fs_lan.bizfocused.com||call-info|"user" > |38e107ab-6cde6635 at 10.10.40.30|"220" > ;tag=fd508933c5f5924d|SIP/2.0/UDP > 10.10.40.30:5060;branch=z9hG4bK-f5598bfb|1337980638|Cisco/SPA504G-7.4.8a||ex > ternal|FreeSwitch1.sumdomain.com|5060|10.10.40.30|-1||"220" > ;tag=VrGrXQaOH22R > sip|220|fs_lan.bizfocused.com|220|fs_lan.bizfocused.com||call-info|"user" > |c08f0c6a-c46e90d2 at 10.10.40.20|"Sean" > ;tag=f22a978ae8838032|SIP/2.0/UDP > 10.10.40.20:5064;branch=z9hG4bK-905b6faa|1337980643|Cisco/SPA504G-7.4.9c||ex > ternal|FreeSwitch1.sumdomain.com|5064|10.10.40.20|-1||"Sean" > ;tag=y5VtigPlIghD > > But it was in: > sofia_reg_external.db not internal. > > I have sorted out all the sip trace data into 2 txt files for the 2 phones > involved. ?They are zipped up at: > http://www.bizfocused.com/sip_trace.zip > > Thank you again for your help. ?I am way over my head now. > > > -----Original Message----- > From: Anthony Minessale [mailto:anthony.minessale at gmail.com] > Sent: Friday, May 25, 2012 2:35 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] SCA not working inbound - Multi Domain > > What are the phones putting in the subscribe ? > > sofia global siptrace on > sofia global debug presence|sla > > then watch for SUBSCRIBE > > also when you are not using odbc you can get the sql with this app > > sqlute3 /usr/local/freeswitch/db/sofia_reg_internal.db > > also try "select * from sip_subscriptions" > > its all about using the right host name across the board, IP's count as > hostnames, they do not magically resolve any dns with SIP > > > > > On Fri, May 25, 2012 at 1:26 PM, Sean Devoy wrote: >> Hi all, >> >> >> >> I have a muti-tennnant configuration that is working nicely except for >> Shared Call Appearance.? The desktop devices are CISCO 504Gs and they >> are configured as described in the FS Wiki as well as Cisco Documentation. >> >> >> >> The SCA works perfectly for outbound calls ? if either phone pickups >> like 220, the other phones indicator light flashes red.? However, >> inbound calls will go to only one of the phones (which one has changed >> a few times) and the other phones line still just stays green and does >> not > ring. >> >> >> >> Here is the sip interfaces config: >> >> >> >> ??? >> >> ????? >> >> ????? >> >> ????? >> >> ????? >> >> ????? >> >> ????? >> >> ????? >> >> ????? >> >> ????? >> >> ????? >> >> ????? >> >> ????? >> >> ????? >> >> ????? >> >> ????? >> >> ????? >> >> ????? >> >> ????? >> >> ????? >> >> ??? >> >> ? >> >> >> >> The directory entry which both phones connect using: >> >> ??? >> >> ????? >> >> ??????? >> >> ??????? > value="410420BLEEP"/> >> >> ??????? >> >> ??????? >> >> ??????? >> >> ??????? >> >> ??????? >> >> ??????? >> >> ??????? >> >> ????? >> >> ????? >> >> ??????? " >> >> ??????? >> >> ??????? > value="{presence_id=220 at mydomainname.com}${sofia_contact(220 at mydomainn >> ame.com)}"/> >> >> ??????? >> >> ????? >> >> ??? >> >> >> >> And the dial plan for ext 220: >> >> ? >> >> ??? >> >> ????? > data="effective_caller_id_number=${internal_caller_id_number}"/> >> >> ????? > data="effective_caller_id_name=${internal_caller_id_name}"/> >> >> ????? >> >> ????? >> >> ????? >> >> ????? > data="{sip_invite_domain=mydomainname.com}user/220 at mydomainname.com" >> /> >> >> ?????? >> >> ????? >> >> ????? >> >> ??? >> >> ? >> >> >> >> >> >> >> >> I did see this in the wiki >> (http://wiki.freeswitch.org/wiki/Shared_Line_Appearance): >> >> If SLA works for outgoing calls and SLA does not work for inbound >> calls to the SLA phones, you may have some presence problem related to >> mixed IP and domain names. When using ODBC you may issue the following >> SQL statement >> >> select >> sip_to_host,sip_from_user,sip_from_host,hostname,presence_id,call_info >> from sip_dialogs; >> >> But I don?t have ODBC on this server, so I am a little lost. >> >> >> >> I have the phones login to domain names, not addresses.? I never refer >> to IP addresses in my xml (except gateways addresses).? I am not >> trying SLA across domain, only within the same domain. >> >> >> >> I hope someone can spot something.? Thanks for your help. >> >> >> >> Sean >> >> >> ______________________________________________________________________ >> ___ Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use >> rs >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > > > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From msc at freeswitch.org Tue May 29 23:35:33 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 29 May 2012 12:35:33 -0700 Subject: [Freeswitch-users] how to schedule calls In-Reply-To: <1338198708.69195.YahooMailClassic@web110807.mail.gq1.yahoo.com> References: <1338187031.89546.YahooMailClassic@web110814.mail.gq1.yahoo.com> <1338198708.69195.YahooMailClassic@web110807.mail.gq1.yahoo.com> Message-ID: On Mon, May 28, 2012 at 2:51 AM, Sherif Omran wrote: > Hi guys, > > Any body knows how to schedule calls depending on some internal variables. > Such as at 8pm, users with voice mail not yet delivered receive a call. > > Thanks > S. > > You can also use the sched_api API command: http://wiki.freeswitch.org/wiki/Mod_commands#sched_api You'll need to convert to epoch seconds or use an offset from the current time. For example: sched_api +3600 foo originate user/1000 &park Will call user 1000 in 3600 seconds (1 hour) and park the call. (Obviously you'll send the call somewhere other than 'park') If you want an absolute time then you'll need to convert to epoch seconds. Use 'strepoch' to convert a date/time string into epoch seconds. The 'strftime' displays the current date/time in the proper format: freeswitch at internal> strftime 2012-05-29 12:40:22 By itself, 'strepoch' displays the current date/time in epoch seconds: freeswitch at internal> strepoch 1338320491 To get a specific date/time just feed it as an argument. 6PM tonight is this: freeswitch at internal> strepoch 2012-05-29 18:00:00 1338339600 Feed that into your sched_api: freeswitch at internal> sched_api 1338339600 foo originate user/1001 &park +OK Added: 10 2012-05-29 12:44:01.405063 [DEBUG] switch_scheduler.c:214 Added task 10 sched_api_function (foo) to run at 1338339600 And verify that task 10 is there: freeswitch at internal> show tasks task_id,task_desc,task_group,task_sql_manager,hostname 1,zrtp_cache_save,core,0,fs.debian 2,heartbeat,core,0,fs.debian 3,check_ip,core,0,fs.debian 4,limit_hash_cleanup,mod_hash,0,fs.debian *10,sched_api_function,foo,0,fs.debian* 5 total. freeswitch at internal> And use 'sched_del' to remove it if you no longer need it: freeswitch at internal> sched_del 10 +OK Deleted: 1 2012-05-29 12:46:22.845040 [DEBUG] switch_scheduler.c:138 Deleting task 10 sched_api_function (foo) Bonus feature: use 'expand' API to roll it all into a single command: freeswitch at internal> expand sched_api ${strepoch 2012-05-29 18:00:00} foo originate user/1001 &park +OK Added: 11 2012-05-29 12:48:11.785045 [DEBUG] switch_scheduler.c:214 Added task 11 sched_api_function (foo) to run at 1338339600 Please tinker away and let us know what you decide to do. You'll need something to talk to the FreeSWITCH event socket to send all these commands. Any ESL (event socket library) compatible language is a good choice. We have people who use Perl, Python, PHP, and Ruby all successfully, so any of those is a good choice. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120529/642ff640/attachment-0001.html From msc at freeswitch.org Wed May 30 00:04:20 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 29 May 2012 13:04:20 -0700 Subject: [Freeswitch-users] How do you split bridged calls ? In-Reply-To: <20120523163913.b07558d1@mail.tritonwest.net> References: <20120523163913.b07558d1@mail.tritonwest.net> Message-ID: Sorry for the late reply, but I just wanted to mention for the sake of posterity/SEO that you could also use 'uuid_dual_transfer' if you want to send each leg to a different destination. http://wiki.freeswitch.org/wiki/Mod_commands#uuid_dual_transfer -MC On Wed, May 23, 2012 at 9:39 AM, Dave R. Kompel wrote: > ** > Or you could use uuid_transfer, such as "uuid_transfer -both park > inline". If you want to use dialplan apps in the transfer just use the > "inline" dialplan, and not XML. > > --Dave > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120529/86bbcaeb/attachment.html From sdevoy at bizfocused.com Wed May 30 00:19:14 2012 From: sdevoy at bizfocused.com (Sean Devoy) Date: Tue, 29 May 2012 16:19:14 -0400 Subject: [Freeswitch-users] SCA not working inbound - Multi Domain In-Reply-To: References: <040301cd3aa3$e93705f0$bba511d0$@bizfocused.com> <051501cd3ac2$8a187ee0$9e497ca0$@bizfocused.com> <075e01cd3dbe$06db5350$1491f9f0$@bizfocused.com> Message-ID: <085901cd3dd8$5157c7f0$f40757d0$@bizfocused.com> Thanks Anthony. First I should have pointed out that the 2 test phones are on the local Lan with the switch, NO NAT. The final task is to move them out with NAT. Second, it is working!! After I dumped everything for you I noticed one of the phones was connecting on the wrong port (sofia external, not lan). I changed it last week but it did not seem to work. Now having waited (and registrations having renewed) it is working!! Thank you for your help. Now, out to the site and their crappy router and to see if this works on through NAT. Sean -----Original Message----- From: Anthony Minessale [mailto:anthony.minessale at gmail.com] Sent: Tuesday, May 29, 2012 3:18 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SCA not working inbound - Multi Domain enter "sofia_contact 220 at mydomainname.com" at your cli and see if you get a large dial string. if you are only getting calls to one phone make sure they all work, its possible you have many phones behind the same nat and only one of them can poke through at a time. On Tue, May 29, 2012 at 12:11 PM, Sean Devoy wrote: > BUMP! > > Anyone have any ideas for me? > > Any other information I can provide? > > Thanks. > Sean > > -----Original Message----- > From: Sean Devoy [mailto:sdevoy at bizfocused.com] > Sent: Friday, May 25, 2012 6:06 PM > To: 'FreeSWITCH Users Help' > Subject: Re: [Freeswitch-users] SCA not working inbound - Multi Domain > > Here is the result of select * from sip_subscriptions;" > sip|220|fs_lan.bizfocused.com|220|fs_lan.bizfocused.com||call-info|"user" > |38e107ab-6cde6635 at 10.10.40.30|"220" > ;tag=fd508933c5f5924d|SIP/2.0/UDP > 10.10.40.30:5060;branch=z9hG4bK-f5598bfb|1337980638|Cisco/SPA504G-7.4. > 8a||ex > ternal|FreeSwitch1.sumdomain.com|5060|10.10.40.30|-1||"220" > ;tag=VrGrXQaOH22R > sip|220|fs_lan.bizfocused.com|220|fs_lan.bizfocused.com||call-info|"user" > |c08f0c6a-c46e90d2 at 10.10.40.20|"Sean" > ;tag=f22a978ae8838032|SIP/2.0/UDP > 10.10.40.20:5064;branch=z9hG4bK-905b6faa|1337980643|Cisco/SPA504G-7.4. > 9c||ex > ternal|FreeSwitch1.sumdomain.com|5064|10.10.40.20|-1||"Sean" > ;tag=y5VtigPlIghD > > But it was in: > sofia_reg_external.db not internal. > > I have sorted out all the sip trace data into 2 txt files for the 2 > phones involved. ?They are zipped up at: > http://www.bizfocused.com/sip_trace.zip > > Thank you again for your help. ?I am way over my head now. > > > -----Original Message----- > From: Anthony Minessale [mailto:anthony.minessale at gmail.com] > Sent: Friday, May 25, 2012 2:35 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] SCA not working inbound - Multi Domain > > What are the phones putting in the subscribe ? > > sofia global siptrace on > sofia global debug presence|sla > > then watch for SUBSCRIBE > > also when you are not using odbc you can get the sql with this app > > sqlute3 /usr/local/freeswitch/db/sofia_reg_internal.db > > also try "select * from sip_subscriptions" > > its all about using the right host name across the board, IP's count > as hostnames, they do not magically resolve any dns with SIP > > > > > On Fri, May 25, 2012 at 1:26 PM, Sean Devoy wrote: >> Hi all, >> >> >> >> I have a muti-tennnant configuration that is working nicely except >> for Shared Call Appearance.? The desktop devices are CISCO 504Gs and >> they are configured as described in the FS Wiki as well as Cisco Documentation. >> >> >> >> The SCA works perfectly for outbound calls ? if either phone pickups >> like 220, the other phones indicator light flashes red.? However, >> inbound calls will go to only one of the phones (which one has >> changed a few times) and the other phones line still just stays green >> and does not > ring. >> >> >> >> Here is the sip interfaces config: >> >> >> >> ??? >> >> ????? >> >> ????? >> >> ????? >> >> ????? >> >> ????? >> >> ????? >> >> ????? >> >> ????? >> >> ????? > value="true"/> >> >> ????? >> >> ????? >> >> ????? >> >> ????? >> >> ????? >> >> ????? >> >> ????? >> >> ????? >> >> ????? >> >> ????? >> >> ??? >> >> ? >> >> >> >> The directory entry which both phones connect using: >> >> ??? >> >> ????? >> >> ??????? >> >> ??????? > value="410420BLEEP"/> >> >> ??????? >> >> ??????? >> >> ??????? >> >> ??????? >> >> ??????? >> >> ??????? >> >> ??????? >> >> ????? >> >> ????? >> >> ??????? " >> >> ??????? >> >> ??????? > value="{presence_id=220 at mydomainname.com}${sofia_contact(220 at mydomain >> n >> ame.com)}"/> >> >> ??????? >> >> ????? >> >> ??? >> >> >> >> And the dial plan for ext 220: >> >> ? >> >> ??? >> >> ????? > data="effective_caller_id_number=${internal_caller_id_number}"/> >> >> ????? > data="effective_caller_id_name=${internal_caller_id_name}"/> >> >> ????? >> >> ????? >> >> ????? >> >> ????? > data="{sip_invite_domain=mydomainname.com}user/220 at mydomainname.com" >> /> >> >> ?????? >> >> ????? >> >> ????? >> >> ??? >> >> ? >> >> >> >> >> >> >> >> I did see this in the wiki >> (http://wiki.freeswitch.org/wiki/Shared_Line_Appearance): >> >> If SLA works for outgoing calls and SLA does not work for inbound >> calls to the SLA phones, you may have some presence problem related >> to mixed IP and domain names. When using ODBC you may issue the >> following SQL statement >> >> select >> sip_to_host,sip_from_user,sip_from_host,hostname,presence_id,call_inf >> o >> from sip_dialogs; >> >> But I don?t have ODBC on this server, so I am a little lost. >> >> >> >> I have the phones login to domain names, not addresses.? I never >> refer to IP addresses in my xml (except gateways addresses).? I am >> not trying SLA across domain, only within the same domain. >> >> >> >> I hope someone can spot something.? Thanks for your help. >> >> >> >> Sean >> >> >> _____________________________________________________________________ >> _ ___ Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us >> e >> rs >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > > > > > > > > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From msc at freeswitch.org Wed May 30 00:24:51 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 29 May 2012 13:24:51 -0700 Subject: [Freeswitch-users] How to connect dialplan on freeswitch to Mysql or ODBC? In-Reply-To: <1338110282.29296.YahooMailNeo@web120101.mail.ne1.yahoo.com> References: <1338110282.29296.YahooMailNeo@web120101.mail.ne1.yahoo.com> Message-ID: I'm pretty sure the stuff you need for mod_sofia is found here: http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core However, I'm not sure what exactly you mean about the query in the dialplan. Could you clarify what you mean? -MC On Sun, May 27, 2012 at 2:18 AM, Samira Mh wrote: > hi guys, > i have created mysql database name "x" using ODBC in freeswitch, > now i have the table named "y" in the "x" databases; > now,how can i connect the sipuser in my direcroty to the "x" database so > that use the result of quert for taht user? > and how can i insert query into the dialplan to connect myuser to mysql > table > how can i do that ? > thanks alot > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120529/e2fdb819/attachment.html From sdevoy at bizfocused.com Wed May 30 00:28:18 2012 From: sdevoy at bizfocused.com (Sean Devoy) Date: Tue, 29 May 2012 16:28:18 -0400 Subject: [Freeswitch-users] SCA not working inbound - Multi Domain In-Reply-To: References: <040301cd3aa3$e93705f0$bba511d0$@bizfocused.com> <051501cd3ac2$8a187ee0$9e497ca0$@bizfocused.com> <075e01cd3dbe$06db5350$1491f9f0$@bizfocused.com> Message-ID: <087401cd3dd9$954cfa60$bfe6ef20$@bizfocused.com> One follow up question about SCA and Gateways ... I have one Gateway setup w/out NAT for phones on the local LAN and a second for NAT phones (on a different port). Is that necessary or can I support NAT and NO NAT phones on the same Gateway/port? If 2 ports/gateways is the best approach will SCA work across them? I don?t see how as one would be 220 at local_ip_name.com and other would be 220 at external_ip_name.com. Thanks again, Sean -----Original Message----- From: Anthony Minessale [mailto:anthony.minessale at gmail.com] Sent: Tuesday, May 29, 2012 3:18 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SCA not working inbound - Multi Domain enter "sofia_contact 220 at mydomainname.com" at your cli and see if you get a large dial string. if you are only getting calls to one phone make sure they all work, its possible you have many phones behind the same nat and only one of them can poke through at a time. On Tue, May 29, 2012 at 12:11 PM, Sean Devoy wrote: > BUMP! > > Anyone have any ideas for me? > > Any other information I can provide? > > Thanks. > Sean > > -----Original Message----- > From: Sean Devoy [mailto:sdevoy at bizfocused.com] > Sent: Friday, May 25, 2012 6:06 PM > To: 'FreeSWITCH Users Help' > Subject: Re: [Freeswitch-users] SCA not working inbound - Multi Domain > > Here is the result of select * from sip_subscriptions;" > sip|220|fs_lan.bizfocused.com|220|fs_lan.bizfocused.com||call-info|"user" > |38e107ab-6cde6635 at 10.10.40.30|"220" > ;tag=fd508933c5f5924d|SIP/2.0/UDP > 10.10.40.30:5060;branch=z9hG4bK-f5598bfb|1337980638|Cisco/SPA504G-7.4. > 8a||ex > ternal|FreeSwitch1.sumdomain.com|5060|10.10.40.30|-1||"220" > ;tag=VrGrXQaOH22R > sip|220|fs_lan.bizfocused.com|220|fs_lan.bizfocused.com||call-info|"user" > |c08f0c6a-c46e90d2 at 10.10.40.20|"Sean" > ;tag=f22a978ae8838032|SIP/2.0/UDP > 10.10.40.20:5064;branch=z9hG4bK-905b6faa|1337980643|Cisco/SPA504G-7.4. > 9c||ex > ternal|FreeSwitch1.sumdomain.com|5064|10.10.40.20|-1||"Sean" > ;tag=y5VtigPlIghD > > But it was in: > sofia_reg_external.db not internal. > > I have sorted out all the sip trace data into 2 txt files for the 2 > phones involved. ?They are zipped up at: > http://www.bizfocused.com/sip_trace.zip > > Thank you again for your help. ?I am way over my head now. > > > -----Original Message----- > From: Anthony Minessale [mailto:anthony.minessale at gmail.com] > Sent: Friday, May 25, 2012 2:35 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] SCA not working inbound - Multi Domain > > What are the phones putting in the subscribe ? > > sofia global siptrace on > sofia global debug presence|sla > > then watch for SUBSCRIBE > > also when you are not using odbc you can get the sql with this app > > sqlute3 /usr/local/freeswitch/db/sofia_reg_internal.db > > also try "select * from sip_subscriptions" > > its all about using the right host name across the board, IP's count > as hostnames, they do not magically resolve any dns with SIP > > > > > On Fri, May 25, 2012 at 1:26 PM, Sean Devoy wrote: >> Hi all, >> >> >> >> I have a muti-tennnant configuration that is working nicely except >> for Shared Call Appearance.? The desktop devices are CISCO 504Gs and >> they are configured as described in the FS Wiki as well as Cisco Documentation. >> >> >> >> The SCA works perfectly for outbound calls ? if either phone pickups >> like 220, the other phones indicator light flashes red.? However, >> inbound calls will go to only one of the phones (which one has >> changed a few times) and the other phones line still just stays green >> and does not > ring. >> >> >> >> Here is the sip interfaces config: >> >> >> >> ??? >> >> ????? >> >> ????? >> >> ????? >> >> ????? >> >> ????? >> >> ????? >> >> ????? >> >> ????? >> >> ????? > value="true"/> >> >> ????? >> >> ????? >> >> ????? >> >> ????? >> >> ????? >> >> ????? >> >> ????? >> >> ????? >> >> ????? >> >> ????? >> >> ??? >> >> ? >> >> >> >> The directory entry which both phones connect using: >> >> ??? >> >> ????? >> >> ??????? >> >> ??????? > value="410420BLEEP"/> >> >> ??????? >> >> ??????? >> >> ??????? >> >> ??????? >> >> ??????? >> >> ??????? >> >> ??????? >> >> ????? >> >> ????? >> >> ??????? " >> >> ??????? >> >> ??????? > value="{presence_id=220 at mydomainname.com}${sofia_contact(220 at mydomain >> n >> ame.com)}"/> >> >> ??????? >> >> ????? >> >> ??? >> >> >> >> And the dial plan for ext 220: >> >> ? >> >> ??? >> >> ????? > data="effective_caller_id_number=${internal_caller_id_number}"/> >> >> ????? > data="effective_caller_id_name=${internal_caller_id_name}"/> >> >> ????? >> >> ????? >> >> ????? >> >> ????? > data="{sip_invite_domain=mydomainname.com}user/220 at mydomainname.com" >> /> >> >> ?????? >> >> ????? >> >> ????? >> >> ??? >> >> ? >> >> >> >> >> >> >> >> I did see this in the wiki >> (http://wiki.freeswitch.org/wiki/Shared_Line_Appearance): >> >> If SLA works for outgoing calls and SLA does not work for inbound >> calls to the SLA phones, you may have some presence problem related >> to mixed IP and domain names. When using ODBC you may issue the >> following SQL statement >> >> select >> sip_to_host,sip_from_user,sip_from_host,hostname,presence_id,call_inf >> o >> from sip_dialogs; >> >> But I don?t have ODBC on this server, so I am a little lost. >> >> >> >> I have the phones login to domain names, not addresses.? I never >> refer to IP addresses in my xml (except gateways addresses).? I am >> not trying SLA across domain, only within the same domain. >> >> >> >> I hope someone can spot something.? Thanks for your help. >> >> >> >> Sean >> >> >> _____________________________________________________________________ >> _ ___ Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us >> e >> rs >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > > > > > > > > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From msc at freeswitch.org Wed May 30 00:29:32 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 29 May 2012 13:29:32 -0700 Subject: [Freeswitch-users] SCA not working inbound - Multi Domain In-Reply-To: <085901cd3dd8$5157c7f0$f40757d0$@bizfocused.com> References: <040301cd3aa3$e93705f0$bba511d0$@bizfocused.com> <051501cd3ac2$8a187ee0$9e497ca0$@bizfocused.com> <075e01cd3dbe$06db5350$1491f9f0$@bizfocused.com> <085901cd3dd8$5157c7f0$f40757d0$@bizfocused.com> Message-ID: Sean, Thanks for taking the time to follow up. If you get it working we'd love to have a snapshot of your configuration to put up on the wiki. Thanks, MC On Tue, May 29, 2012 at 1:19 PM, Sean Devoy wrote: > Thanks Anthony. > > First I should have pointed out that the 2 test phones are on the local Lan > with the switch, NO NAT. The final task is to move them out with NAT. > > Second, it is working!! > > After I dumped everything for you I noticed one of the phones was > connecting > on the wrong port (sofia external, not lan). I changed it last week but it > did not seem to work. Now having waited (and registrations having renewed) > it is working!! > > Thank you for your help. > > Now, out to the site and their crappy router and to see if this works on > through NAT. > > Sean > -----Original Message----- > From: Anthony Minessale [mailto:anthony.minessale at gmail.com] > Sent: Tuesday, May 29, 2012 3:18 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] SCA not working inbound - Multi Domain > > enter "sofia_contact 220 at mydomainname.com" at your cli and see if you get > a > large dial string. > if you are only getting calls to one phone make sure they all work, its > possible you have many phones behind the same nat and only one of them can > poke through at a time. > > > > On Tue, May 29, 2012 at 12:11 PM, Sean Devoy > wrote: > > BUMP! > > > > Anyone have any ideas for me? > > > > Any other information I can provide? > > > > Thanks. > > Sean > > > > -----Original Message----- > > From: Sean Devoy [mailto:sdevoy at bizfocused.com] > > Sent: Friday, May 25, 2012 6:06 PM > > To: 'FreeSWITCH Users Help' > > Subject: Re: [Freeswitch-users] SCA not working inbound - Multi Domain > > > > Here is the result of select * from sip_subscriptions;" > > sip|220|fs_lan.bizfocused.com|220|fs_lan.bizfocused.com > ||call-info|"user" > > |38e107ab-6cde6635 at 10.10.40.30|"220" > > ;tag=fd508933c5f5924d|SIP/2.0/UDP > > 10.10.40.30:5060;branch=z9hG4bK-f5598bfb|1337980638|Cisco/SPA504G-7.4. > > 8a||ex > > ternal|FreeSwitch1.sumdomain.com|5060|10.10.40.30|-1||"220" > > ;tag=VrGrXQaOH22R > > sip|220|fs_lan.bizfocused.com|220|fs_lan.bizfocused.com > ||call-info|"user" > > |c08f0c6a-c46e90d2 at 10.10.40.20|"Sean" > > ;tag=f22a978ae8838032|SIP/2.0/UDP > > 10.10.40.20:5064;branch=z9hG4bK-905b6faa|1337980643|Cisco/SPA504G-7.4. > > 9c||ex > > ternal|FreeSwitch1.sumdomain.com|5064|10.10.40.20|-1||"Sean" > > ;tag=y5VtigPlIghD > > > > But it was in: > > sofia_reg_external.db not internal. > > > > I have sorted out all the sip trace data into 2 txt files for the 2 > > phones involved. They are zipped up at: > > http://www.bizfocused.com/sip_trace.zip > > > > Thank you again for your help. I am way over my head now. > > > > > > -----Original Message----- > > From: Anthony Minessale [mailto:anthony.minessale at gmail.com] > > Sent: Friday, May 25, 2012 2:35 PM > > To: FreeSWITCH Users Help > > Subject: Re: [Freeswitch-users] SCA not working inbound - Multi Domain > > > > What are the phones putting in the subscribe ? > > > > sofia global siptrace on > > sofia global debug presence|sla > > > > then watch for SUBSCRIBE > > > > also when you are not using odbc you can get the sql with this app > > > > sqlute3 /usr/local/freeswitch/db/sofia_reg_internal.db > > > > also try "select * from sip_subscriptions" > > > > its all about using the right host name across the board, IP's count > > as hostnames, they do not magically resolve any dns with SIP > > > > > > > > > > On Fri, May 25, 2012 at 1:26 PM, Sean Devoy > wrote: > >> Hi all, > >> > >> > >> > >> I have a muti-tennnant configuration that is working nicely except > >> for Shared Call Appearance. The desktop devices are CISCO 504Gs and > >> they are configured as described in the FS Wiki as well as Cisco > Documentation. > >> > >> > >> > >> The SCA works perfectly for outbound calls ? if either phone pickups > >> like 220, the other phones indicator light flashes red. However, > >> inbound calls will go to only one of the phones (which one has > >> changed a few times) and the other phones line still just stays green > >> and does not > > ring. > >> > >> > >> > >> Here is the sip interfaces config: > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> >> value="true"/> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> The directory entry which both phones connect using: > >> > >> > >> > >> > >> > >> > >> > >> >> value="410420BLEEP"/> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> " > >> > >> > >> > >> >> value="{presence_id=220 at mydomainname.com}${sofia_contact(220 at mydomain > >> n > >> ame.com)}"/> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> And the dial plan for ext 220: > >> > >> > >> > >> > >> > >> >> data="effective_caller_id_number=${internal_caller_id_number}"/> > >> > >> >> data="effective_caller_id_name=${internal_caller_id_name}"/> > >> > >> > >> > >> > >> > >> > >> > >> >> data="{sip_invite_domain=mydomainname.com}user/220 at mydomainname.com" > >> /> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> I did see this in the wiki > >> (http://wiki.freeswitch.org/wiki/Shared_Line_Appearance): > >> > >> If SLA works for outgoing calls and SLA does not work for inbound > >> calls to the SLA phones, you may have some presence problem related > >> to mixed IP and domain names. When using ODBC you may issue the > >> following SQL statement > >> > >> select > >> sip_to_host,sip_from_user,sip_from_host,hostname,presence_id,call_inf > >> o > >> from sip_dialogs; > >> > >> But I don?t have ODBC on this server, so I am a little lost. > >> > >> > >> > >> I have the phones login to domain names, not addresses. I never > >> refer to IP addresses in my xml (except gateways addresses). I am > >> not trying SLA across domain, only within the same domain. > >> > >> > >> > >> I hope someone can spot something. Thanks for your help. > >> > >> > >> > >> Sean > >> > >> > >> _____________________________________________________________________ > >> _ ___ Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> Join Us At ClueCon - Aug 7-9, 2012 > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us > >> e > >> rs > >> http://www.freeswitch.org > >> > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > googletalk:conf+888 at conference.freeswitch.org > > pstn:+19193869900 > > > > > > > > > > > > > > > > > > > > ______________________________________________________________________ > > ___ Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > > rs > > http://www.freeswitch.org > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120529/2ac719f6/attachment-0001.html From anthony.minessale at gmail.com Wed May 30 00:31:22 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 29 May 2012 15:31:22 -0500 Subject: [Freeswitch-users] SCA not working inbound - Multi Domain In-Reply-To: <087401cd3dd9$954cfa60$bfe6ef20$@bizfocused.com> References: <040301cd3aa3$e93705f0$bba511d0$@bizfocused.com> <051501cd3ac2$8a187ee0$9e497ca0$@bizfocused.com> <075e01cd3dbe$06db5350$1491f9f0$@bizfocused.com> <087401cd3dd9$954cfa60$bfe6ef20$@bizfocused.com> Message-ID: They should work either way. On Tue, May 29, 2012 at 3:28 PM, Sean Devoy wrote: > One follow up question about SCA and Gateways ... > > I have one Gateway setup w/out NAT for phones on the local LAN and a second > for NAT phones (on a different port). Is that necessary or can I support NAT > and NO NAT phones on the same Gateway/port? ?If 2 ports/gateways is the best > approach will SCA work across them? ?I don?t see how as one would be > 220 at local_ip_name.com and other would be 220 at external_ip_name.com. > > Thanks again, > Sean > > -----Original Message----- > From: Anthony Minessale [mailto:anthony.minessale at gmail.com] > Sent: Tuesday, May 29, 2012 3:18 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] SCA not working inbound - Multi Domain > > enter "sofia_contact 220 at mydomainname.com" at your cli and see if you get a > large dial string. > if you are only getting calls to one phone make sure they all work, its > possible you have many phones behind the same nat and only one of them can > poke through at a time. > > > > On Tue, May 29, 2012 at 12:11 PM, Sean Devoy wrote: >> BUMP! >> >> Anyone have any ideas for me? >> >> Any other information I can provide? >> >> Thanks. >> Sean >> >> -----Original Message----- >> From: Sean Devoy [mailto:sdevoy at bizfocused.com] >> Sent: Friday, May 25, 2012 6:06 PM >> To: 'FreeSWITCH Users Help' >> Subject: Re: [Freeswitch-users] SCA not working inbound - Multi Domain >> >> Here is the result of select * from sip_subscriptions;" >> sip|220|fs_lan.bizfocused.com|220|fs_lan.bizfocused.com||call-info|"user" >> |38e107ab-6cde6635 at 10.10.40.30|"220" >> ;tag=fd508933c5f5924d|SIP/2.0/UDP >> 10.10.40.30:5060;branch=z9hG4bK-f5598bfb|1337980638|Cisco/SPA504G-7.4. >> 8a||ex >> ternal|FreeSwitch1.sumdomain.com|5060|10.10.40.30|-1||"220" >> ;tag=VrGrXQaOH22R >> sip|220|fs_lan.bizfocused.com|220|fs_lan.bizfocused.com||call-info|"user" >> |c08f0c6a-c46e90d2 at 10.10.40.20|"Sean" >> ;tag=f22a978ae8838032|SIP/2.0/UDP >> 10.10.40.20:5064;branch=z9hG4bK-905b6faa|1337980643|Cisco/SPA504G-7.4. >> 9c||ex >> ternal|FreeSwitch1.sumdomain.com|5064|10.10.40.20|-1||"Sean" >> ;tag=y5VtigPlIghD >> >> But it was in: >> sofia_reg_external.db not internal. >> >> I have sorted out all the sip trace data into 2 txt files for the 2 >> phones involved. ?They are zipped up at: >> http://www.bizfocused.com/sip_trace.zip >> >> Thank you again for your help. ?I am way over my head now. >> >> >> -----Original Message----- >> From: Anthony Minessale [mailto:anthony.minessale at gmail.com] >> Sent: Friday, May 25, 2012 2:35 PM >> To: FreeSWITCH Users Help >> Subject: Re: [Freeswitch-users] SCA not working inbound - Multi Domain >> >> What are the phones putting in the subscribe ? >> >> sofia global siptrace on >> sofia global debug presence|sla >> >> then watch for SUBSCRIBE >> >> also when you are not using odbc you can get the sql with this app >> >> sqlute3 /usr/local/freeswitch/db/sofia_reg_internal.db >> >> also try "select * from sip_subscriptions" >> >> its all about using the right host name across the board, IP's count >> as hostnames, they do not magically resolve any dns with SIP >> >> >> >> >> On Fri, May 25, 2012 at 1:26 PM, Sean Devoy wrote: >>> Hi all, >>> >>> >>> >>> I have a muti-tennnant configuration that is working nicely except >>> for Shared Call Appearance.? The desktop devices are CISCO 504Gs and >>> they are configured as described in the FS Wiki as well as Cisco > Documentation. >>> >>> >>> >>> The SCA works perfectly for outbound calls ? if either phone pickups >>> like 220, the other phones indicator light flashes red.? However, >>> inbound calls will go to only one of the phones (which one has >>> changed a few times) and the other phones line still just stays green >>> and does not >> ring. >>> >>> >>> >>> Here is the sip interfaces config: >>> >>> >>> >>> ??? >>> >>> ????? >>> >>> ????? >>> >>> ????? >>> >>> ????? >>> >>> ????? >>> >>> ????? >>> >>> ????? >>> >>> ????? >>> >>> ????? >> value="true"/> >>> >>> ????? >>> >>> ????? >>> >>> ????? >>> >>> ????? >>> >>> ????? >>> >>> ????? >>> >>> ????? >>> >>> ????? >>> >>> ????? >>> >>> ????? >>> >>> ??? >>> >>> ? >>> >>> >>> >>> The directory entry which both phones connect using: >>> >>> ??? >>> >>> ????? >>> >>> ??????? >>> >>> ??????? >> value="410420BLEEP"/> >>> >>> ??????? >>> >>> ??????? >>> >>> ??????? >>> >>> ??????? >>> >>> ??????? >>> >>> ??????? >>> >>> ??????? >>> >>> ????? >>> >>> ????? >>> >>> ??????? " >>> >>> ??????? >>> >>> ??????? >> value="{presence_id=220 at mydomainname.com}${sofia_contact(220 at mydomain >>> n >>> ame.com)}"/> >>> >>> ??????? >>> >>> ????? >>> >>> ??? >>> >>> >>> >>> And the dial plan for ext 220: >>> >>> ? >>> >>> ??? >>> >>> ????? >> data="effective_caller_id_number=${internal_caller_id_number}"/> >>> >>> ????? >> data="effective_caller_id_name=${internal_caller_id_name}"/> >>> >>> ????? >>> >>> ????? >>> >>> ????? >>> >>> ????? >> data="{sip_invite_domain=mydomainname.com}user/220 at mydomainname.com" >>> /> >>> >>> ?????? >>> >>> ????? >>> >>> ????? >>> >>> ??? >>> >>> ? >>> >>> >>> >>> >>> >>> >>> >>> I did see this in the wiki >>> (http://wiki.freeswitch.org/wiki/Shared_Line_Appearance): >>> >>> If SLA works for outgoing calls and SLA does not work for inbound >>> calls to the SLA phones, you may have some presence problem related >>> to mixed IP and domain names. When using ODBC you may issue the >>> following SQL statement >>> >>> select >>> sip_to_host,sip_from_user,sip_from_host,hostname,presence_id,call_inf >>> o >>> from sip_dialogs; >>> >>> But I don?t have ODBC on this server, so I am a little lost. >>> >>> >>> >>> I have the phones login to domain names, not addresses.? I never >>> refer to IP addresses in my xml (except gateways addresses).? I am >>> not trying SLA across domain, only within the same domain. >>> >>> >>> >>> I hope someone can spot something.? Thanks for your help. >>> >>> >>> >>> Sean >>> >>> >>> _____________________________________________________________________ >>> _ ___ Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> Join Us At ClueCon - Aug 7-9, 2012 >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us >>> e >>> rs >>> http://www.freeswitch.org >>> >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 >> >> >> >> >> >> >> >> >> >> ______________________________________________________________________ >> ___ Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use >> rs >> http://www.freeswitch.org > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From msc at freeswitch.org Wed May 30 00:40:53 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 29 May 2012 13:40:53 -0700 Subject: [Freeswitch-users] Why I could not detect DTMF when using Originate command In-Reply-To: References: Message-ID: You may need explicitly to listen for inband DTMFs.Try adding to extension 8888 and retest. -MC On Thu, May 24, 2012 at 8:27 PM, Chaiyawut Sookplang wrote: > I want to make an automatic IVR call using originate command. I issued > command > > "originate {ignore_early_media=true}sofia/gateway/trunk_1/0860216060 8888" > > and then I found that DTMF from the receiver side couldn't be > detected. On the other hand, if I call to extension 8888, IVR and DTMF > work fine. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120529/74d9f4aa/attachment.html From nbhatti at gmail.com Wed May 30 00:49:55 2012 From: nbhatti at gmail.com (Muhammad Naseer Bhatti) Date: Tue, 29 May 2012 23:49:55 +0300 Subject: [Freeswitch-users] Gateway failover with LUA In-Reply-To: References: Message-ID: Possible but here is the problem. When limit is set against two of the gateways, session:execute("limit", "hash gateway_channels v132_5000 10") session:execute("limit", "hash gateway_channels v132_5010 10") It sets limit of 10 channels for each gateway. But when the bridge is made, [gateway=v132_5010]sofia/gateway/v132_5010/9198|[gateway=v132_5000]sofia/gateway/v132_5000/9198 The limit is imposed on both gateways. 2012-05-27 22:13:48.371539 [INFO] switch_limit.c:126 incr called: gateway_channels_v132_5000 max:10, interval:0 2012-05-27 22:13:48.371539 [INFO] mod_hash.c:202 Usage for gateway_channels_v132_5000 is now 3/10 2012-05-27 22:13:48.371539 [INFO] switch_limit.c:126 incr called: gateway_channels_v132_5010 max:10, interval:0 2012-05-27 22:13:48.371539 [INFO] mod_hash.c:202 Usage for gateway_channels_v132_5010 is now 3/10 Is this is a bug or supposed to work like this? Though the call is connected to gateway name v132_5010 only. Thanks. On Mon, May 28, 2012 at 7:20 PM, Avi Marcus wrote: > How about a bridge2, bridge3, etc variable? Then your static dialplan can > see if those are set, and if so, use them. > > -Avi > > > On Mon, May 28, 2012 at 1:09 PM, Muhammad Naseer Bhatti > wrote: >> >> Hi, I am trying to implement a gateway failover functionality with >> lua. Right now I am using limit and using hash backend to limit number >> of channels to each gateway. Since I am forwarding all calls with a >> default dialplan which is sending all calls to application lua and the >> bridge is done outside lua setting all the necessary vars. >> The idea is if all channels of first gateway are occupied, send the >> call to second gateway in the list and so on. Any idea how to >> implement this with lua? Can not seem to find the logic. If I query >> with limit_status it is going to be a query on every call. In a high >> call volume scenario, I don't think this would be really a good >> choice. Any ideas, thoughts? >> >> Thanks. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From nbhatti at gmail.com Wed May 30 01:12:01 2012 From: nbhatti at gmail.com (Muhammad Naseer Bhatti) Date: Wed, 30 May 2012 00:12:01 +0300 Subject: [Freeswitch-users] Gateway failover with LUA In-Reply-To: References: Message-ID: They are two different gateways with two different names, but the same IP address yet the port is different. Possible that limit subsystem is imposing the limit on the IP address itself and not able to distinguish between the different ports. On Tue, May 29, 2012 at 11:49 PM, Muhammad Naseer Bhatti wrote: > Possible but here is the problem. When limit is set against two of the > gateways, > > session:execute("limit", "hash gateway_channels v132_5000 10") > session:execute("limit", "hash gateway_channels v132_5010 10") > > It sets limit of 10 channels for each gateway. But when the bridge is made, > > [gateway=v132_5010]sofia/gateway/v132_5010/9198|[gateway=v132_5000]sofia/gateway/v132_5000/9198 > > The limit is imposed on both gateways. > > 2012-05-27 22:13:48.371539 [INFO] switch_limit.c:126 incr called: > gateway_channels_v132_5000 max:10, interval:0 > 2012-05-27 22:13:48.371539 [INFO] mod_hash.c:202 Usage for > gateway_channels_v132_5000 is now 3/10 > 2012-05-27 22:13:48.371539 [INFO] switch_limit.c:126 incr called: > gateway_channels_v132_5010 max:10, interval:0 > 2012-05-27 22:13:48.371539 [INFO] mod_hash.c:202 Usage for > gateway_channels_v132_5010 is now 3/10 > > Is this is a bug or ?supposed to work like this? ?Though the call is > connected to gateway name v132_5010 only. > > > Thanks. > > On Mon, May 28, 2012 at 7:20 PM, Avi Marcus wrote: >> How about a bridge2, bridge3, etc variable? Then your static dialplan can >> see if those are set, and if so, use them. >> >> -Avi >> >> >> On Mon, May 28, 2012 at 1:09 PM, Muhammad Naseer Bhatti >> wrote: >>> >>> Hi, I am trying to implement a gateway failover functionality with >>> lua. Right now I am using limit and using hash backend to limit number >>> of channels to each gateway. Since I am forwarding all calls with a >>> default dialplan which is sending all calls to application lua and the >>> bridge is done outside lua setting all the necessary vars. >>> The idea is if all channels of first gateway are occupied, send the >>> call to second gateway in the list and so on. Any idea how to >>> implement this with lua? Can not seem to find the logic. If I query >>> with limit_status it is going to be a query on every call. In a high >>> call volume scenario, I don't think this would be really a good >>> choice. Any ideas, thoughts? >>> >>> Thanks. >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> Join Us At ClueCon - Aug 7-9, 2012 >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> From jerry.richards at teotech.com Wed May 30 03:17:11 2012 From: jerry.richards at teotech.com (Jerry Richards) Date: Tue, 29 May 2012 23:17:11 +0000 Subject: [Freeswitch-users] Why does both core.db and sofia_reg_internal.db have registrations tables? Message-ID: <1545146083A72C4DB7B66584B7E5D9840E78282C@BY2PRD0410MB377.namprd04.prod.outlook.com> Just curious why core.db and sofia_reg_internal.db have registrations tables? Is this redundant? Thanks, Jerry From: Jerry Richards Sent: Wednesday, May 23, 2012 10:06 AM To: 'FreeSWITCH Users Help' Subject: Channel Variable To Disable Processing of 2833 DTMF Messages? Is there a channel variable to disable/ignore 2833 DTMF messages that are going from the internal network toward the PRI? I have a scenario where I am getting both inband and out-of-band DTMF (i.e. double DTMF digits). This happens when a call comes in through the PRI and routes back out the PRI. Inband DTMF coming from the caller is converted to out-of-band 2833 internally, so when the audio is routed back out the PRI, it contains both the inband DTMF and the 2833 conversion. In some cases, we don't want to use disable_dtmf, since we may need to detect inband DTMF in FreeSwitch. Thanks, Jerry -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120529/d8e0c441/attachment-0001.html From drk at drkngs.net Wed May 30 03:05:00 2012 From: drk at drkngs.net (Dave R. Kompel) Date: Tue, 29 May 2012 16:05:00 -0700 Subject: [Freeswitch-users] Gateway failover with LUA In-Reply-To: Message-ID: <20120529230500.4689f47a@mail.tritonwest.net> One trick I've noticed that nobody is doing for outbound limit, is using "execute_on_originate" to do the limit there on each channel. You can specify it in the local variables per channel. The following would do it for each B-Leg, w/o messing associating the coun't to your a-leg session: bridge:[execute_on_originate=limit hash foo gwa 10]sofia/gateway/gwa/$1|[execute_on_originate=limit hash foo gwb 10]sofia/gateway/gwb/$1 ... and so on... I use this all the time in a more extended way. I use execute_on_originate to call my own dialplan app, but you could use it to invoke a lua script as well on the session of the B-Leg, before the call is originated. If you just return from your dialplan app, the b-leg will continue, but if you hangup the channel, then it will look like the b-leg failed with the hangup cause you specify, and will get passed back as if the far end returned that cause. --Dave _____ From: Muhammad Naseer Bhatti [mailto:nbhatti at gmail.com] To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org] Sent: Tue, 29 May 2012 14:12:01 -0700 Subject: Re: [Freeswitch-users] Gateway failover with LUA They are two different gateways with two different names, but the same IP address yet the port is different. Possible that limit subsystem is imposing the limit on the IP address itself and not able to distinguish between the different ports. On Tue, May 29, 2012 at 11:49 PM, Muhammad Naseer Bhatti wrote: > Possible but here is the problem. When limit is set against two of the > gateways, > > session:execute("limit", "hash gateway_channels v132_5000 10") > session:execute("limit", "hash gateway_channels v132_5010 10") > > It sets limit of 10 channels for each gateway. But when the bridge is made, > > [gateway=v132_5010]sofia/gateway/v132_5010/9198|[gateway=v132_5000]sofia/gateway/v132_5000/9198 > > The limit is imposed on both gateways. > > 2012-05-27 22:13:48.371539 [INFO] switch_limit.c:126 incr called: > gateway_channels_v132_5000 max:10, interval:0 > 2012-05-27 22:13:48.371539 [INFO] mod_hash.c:202 Usage for > gateway_channels_v132_5000 is now 3/10 > 2012-05-27 22:13:48.371539 [INFO] switch_limit.c:126 incr called: > gateway_channels_v132_5010 max:10, interval:0 > 2012-05-27 22:13:48.371539 [INFO] mod_hash.c:202 Usage for > gateway_channels_v132_5010 is now 3/10 > > Is this is a bug or supposed to work like this? Though the call is > connected to gateway name v132_5010 only. > > > Thanks. > > On Mon, May 28, 2012 at 7:20 PM, Avi Marcus wrote: >> How about a bridge2, bridge3, etc variable? Then your static dialplan can >> see if those are set, and if so, use them. >> >> -Avi >> >> >> On Mon, May 28, 2012 at 1:09 PM, Muhammad Naseer Bhatti >> wrote: >>> >>> Hi, I am trying to implement a gateway failover functionality with >>> lua. Right now I am using limit and using hash backend to limit number >>> of channels to each gateway. Since I am forwarding all calls with a >>> default dialplan which is sending all calls to application lua and the >>> bridge is done outside lua setting all the necessary vars. >>> The idea is if all channels of first gateway are occupied, send the >>> call to second gateway in the list and so on. Any idea how to >>> implement this with lua? Can not seem to find the logic. If I query >>> with limit_status it is going to be a query on every call. In a high >>> call volume scenario, I don't think this would be really a good >>> choice. Any ideas, thoughts? >>> >>> Thanks. >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> Join Us At ClueCon - Aug 7-9, 2012 >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120529/c7766d7d/attachment.html From msc at freeswitch.org Wed May 30 03:49:32 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 29 May 2012 16:49:32 -0700 Subject: [Freeswitch-users] Gateway failover with LUA In-Reply-To: <20120529230500.4689f47a@mail.tritonwest.net> References: <20120529230500.4689f47a@mail.tritonwest.net> Message-ID: Dave, That sounds like a clever way to handle it. I'd like to see more people give that a try and report back how it works in their various scenarios. -MC On Tue, May 29, 2012 at 4:05 PM, Dave R. Kompel wrote: > ** > One trick I've noticed that nobody is doing for outbound limit, is using > "execute_on_originate" to do the limit there on each channel. You can > specify it in the local variables per channel. > > The following would do it for each B-Leg, w/o messing associating the > coun't to your a-leg session: > > bridge:[execute_on_originate=limit hash foo gwa > 10]sofia/gateway/gwa/$1|[execute_on_originate=limit hash foo gwb > 10]sofia/gateway/gwb/$1 ... and so on... > > I use this all the time in a more extended way. I use execute_on_originate > to call my own dialplan app, but you could use it to invoke a lua script as > well on the session of the B-Leg, before the call is originated. If you > just return from your dialplan app, the b-leg will continue, but if you > hangup the channel, then it will look like the b-leg failed with the hangup > cause you specify, and will get passed back as if the far end returned that > cause. > > --Dave > > ------------------------------ > *From:* Muhammad Naseer Bhatti [mailto:nbhatti at gmail.com] > *To:* FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org] > *Sent:* Tue, 29 May 2012 14:12:01 -0700 > *Subject:* Re: [Freeswitch-users] Gateway failover with LUA > > > They are two different gateways with two different names, but the same > IP address yet the port is different. Possible that limit subsystem is > imposing the limit on the IP address itself and not able to > distinguish between the different ports. > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120529/b61e3c99/attachment.html From Tim.Meade at Millicorp.com Wed May 30 04:55:53 2012 From: Tim.Meade at Millicorp.com (Tim Meade) Date: Wed, 30 May 2012 00:55:53 +0000 Subject: [Freeswitch-users] mod_sms dynamic chatplan Message-ID: <804D48104511D4468F0D60DF9D3100350929327E@MAILBOX.millicorp.com> A couple of short questions on mod_sms: Has anyone done any work with using xml_curl to get the chatplan? If that isn't available, how about ESL OUTBOUND in the chatplan? Much thanks..... Tim -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120530/d1fb9c53/attachment-0001.html From mitch.capper at gmail.com Wed May 30 06:28:09 2012 From: mitch.capper at gmail.com (Mitch Capper) Date: Tue, 29 May 2012 19:28:09 -0700 Subject: [Freeswitch-users] FSClient a Windows FreeSWITCH softphone Update Released Message-ID: An update had been posted for FSClient is a windows .NET softphone that uses libfreeswitch at the core. Info page at: http://wiki.freeswitch.org/wiki/FSClient Direct binary download: http://files.freeswitch.org/windows/installer/x86/FSClient.zip And source is in contrib repo under MitchCapper/FSClient Some of the changes include: Built against FS 1.2 RC isac codec support for contacts you can now right click and choose what account to call from if sofia goes offline update the accounts status Jabra headset improvements to avoid some crash conditions background task improvements to avoid the phone breaking without you knowing Feedback always welcome. With ZRTP into core now the next version may have some ZRTP related features. ~Mitch From saami_mh at ymail.com Wed May 30 08:50:13 2012 From: saami_mh at ymail.com (Samira Mh) Date: Tue, 29 May 2012 21:50:13 -0700 (PDT) Subject: [Freeswitch-users] how to route Calls to VoipGateway for making calls to other country? Message-ID: <1338353413.71387.YahooMailNeo@web120101.mail.ne1.yahoo.com> Hi, i am going to Route Call to the voipgateway that is located on my office so that make call to outside of office or to the world or to the other country? thanks so much -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120529/b6e719af/attachment.html From engineerzuhairraza at gmail.com Wed May 30 09:04:53 2012 From: engineerzuhairraza at gmail.com (Zohair Raza) Date: Wed, 30 May 2012 09:04:53 +0400 Subject: [Freeswitch-users] how to route Calls to VoipGateway for making calls to other country? In-Reply-To: <1338353413.71387.YahooMailNeo@web120101.mail.ne1.yahoo.com> References: <1338353413.71387.YahooMailNeo@web120101.mail.ne1.yahoo.com> Message-ID: Hi, http://wiki.freeswitch.org/wiki/Clarification:gateways http://wiki.freeswitch.org/wiki/SIP_Provider_Examples Regards, Zohair Raza On Wed, May 30, 2012 at 8:50 AM, Samira Mh wrote: > Hi, > i am going to Route Call to the voipgateway that is located on my office so > that make call to outside of office or to the world or to the other country? > > thanks so much > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From saami_mh at ymail.com Wed May 30 09:25:27 2012 From: saami_mh at ymail.com (Samira Mh) Date: Tue, 29 May 2012 22:25:27 -0700 (PDT) Subject: [Freeswitch-users] how to route Calls to VoipGateway for making calls to other country? In-Reply-To: References: <1338353413.71387.YahooMailNeo@web120101.mail.ne1.yahoo.com> Message-ID: <1338355527.3988.YahooMailNeo@web120104.mail.ne1.yahoo.com> THNAKS SO MUCH FOR YOUR HELP; ________________________________ From: Zohair Raza To: FreeSWITCH Users Help Sent: Wednesday, May 30, 2012 9:34 AM Subject: Re: [Freeswitch-users] how to route Calls to VoipGateway for making calls to other country? Hi, http://wiki.freeswitch.org/wiki/Clarification:gateways http://wiki.freeswitch.org/wiki/SIP_Provider_Examples Regards, Zohair Raza On Wed, May 30, 2012 at 8:50 AM, Samira Mh wrote: > Hi, > i am going to Route Call to the voipgateway that is located on my office so > that make call to outside of office or to the world or to the other country? > > thanks so much > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120529/0047c0bb/attachment.html From avi at avimarcus.net Wed May 30 15:30:22 2012 From: avi at avimarcus.net (Avi Marcus) Date: Wed, 30 May 2012 14:30:22 +0300 Subject: [Freeswitch-users] Per-channel fsctl loglevel? Message-ID: Hi. I've been developing a script for someone to handle credit card transactions via IVR. Is there a way to just set the particular channel/call to fsctl loglevel 6, rather than the entire switch..? (dtmf is shown in /log 7) And.. similarly is there a way to blank out the var digits_dialed in the xml_cdr, from within FS, before the end of the call? Thanks! -Avi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120530/90b54aa8/attachment.html From mytemike72 at gmail.com Wed May 30 16:08:45 2012 From: mytemike72 at gmail.com (Michael Lutz) Date: Wed, 30 May 2012 14:08:45 +0200 Subject: [Freeswitch-users] Why I could not detect DTMF when using Originate command In-Reply-To: References: Message-ID: If your outbound provider does not support RFC2833 (DTMF via SIP Info messages) you need to explicitly set the outbound leg to 'listen' for DTMF tones using the start_dtmf dialplan command. As you originate, there is no real dialplan, you can set it using the "execute_on_answer"in inside the originate command like: originate {execute_on_answer=start_dtmf,ignore_early_media=true}sofia/gateway/trunk_1/0860216060 8888 Regards, Mike. 2012/5/29 Michael Collins : > You may need explicitly to listen for inband DTMFs.Try adding application="start_dtmf"> to extension 8888 and retest. > -MC > > > On Thu, May 24, 2012 at 8:27 PM, Chaiyawut Sookplang > wrote: >> >> I want to make an automatic IVR call using originate command. I issued >> command >> >> "originate {ignore_early_media=true}sofia/gateway/trunk_1/0860216060 8888" >> >> and then I found that DTMF from the receiver side couldn't be >> detected. On the other hand, if I call to extension 8888, IVR and DTMF >> work fine. > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From mytemike72 at gmail.com Wed May 30 16:10:25 2012 From: mytemike72 at gmail.com (Michael Lutz) Date: Wed, 30 May 2012 14:10:25 +0200 Subject: [Freeswitch-users] Channel Variable To Disable Processing of 2833 DTMF Messages? In-Reply-To: <1545146083A72C4DB7B66584B7E5D98402BD7266@BY2PRD0410MB377.namprd04.prod.outlook.com> References: <1545146083A72C4DB7B66584B7E5D98402BD7266@BY2PRD0410MB377.namprd04.prod.outlook.com> Message-ID: Hi Jerry, Did you ever get this to work?, I have a simular issue and looking for a resolvement. Thanks, Mike. 2012/5/23 Jerry Richards : > Is there a channel variable to disable/ignore 2833 DTMF messages that are > going from the internal network toward the PRI?? I have a scenario where I > am getting both inband and out-of-band DTMF (i.e. double DTMF digits). > > > > This happens when a call comes in through the PRI and routes back out the > PRI.? Inband DTMF coming from the caller is converted to out-of-band 2833 > internally, so when the audio is routed back out the PRI, it contains both > the inband DTMF and the 2833 conversion. ?In some cases, we don?t want to > use disable_dtmf, since we may need to detect inband DTMF in FreeSwitch. > > > > Thanks, > > Jerry > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From B.Tietz at pinguin.ag Wed May 30 16:10:29 2012 From: B.Tietz at pinguin.ag (B.Tietz at pinguin.ag) Date: Wed, 30 May 2012 14:10:29 +0200 Subject: [Freeswitch-users] request SDP in 200 OK Message-ID: <07BF4904977CC645B485E970424193AD110A2945C8@localhost> Hi list, I want to receive faxes with T.38 and spandsp (app spandsp). The INVITES for the calls come with or without SDP-headers. If the SDP-header is in the INVITE to freeswitch, Freeswitch can receive the fax without problem. But if FS gets an INVITE without SDP it can not receive the call and ends with error code 'MANDATORY_IE_MISSING', because FS doesn't send an SDP-offer in the 200 OK message. How can I achieve this? Is it late-negotiation? Benjamin From nazim.aghabayov at gmail.com Wed May 30 17:11:44 2012 From: nazim.aghabayov at gmail.com (Nazim Aghabayov) Date: Wed, 30 May 2012 18:11:44 +0500 Subject: [Freeswitch-users] FreeSWITCH Weekly News and Notes In-Reply-To: References: Message-ID: <4FC61C90.10107@gmail.com> Hello Michael I've tried to fetch an audio torrent of Kens presentation but it's not available "http://torrents.freeswitch.org/conf_call_2012-05-23.torrent" Is source code of the dialplan module available? With Best Regards, Nazim On 05/29/2012 09:40 PM, Michael Collins wrote: > Happy Tuesday to all. We hope you enjoyed your holiday weekend. > > We've had a steady stream of presentations on the FreeSWITCH > conference call over the past month. Ken Rice has graciously done two > presentations on how to create a dialplan module in C, including a > fully functional example module. This code is extremely helpful not > just in understanding how to write a dialplan module but in writing > any module that exposes new dialplan applications or command line > APIs. This week we will take a break from the formal presentations and > have an open discussion on any topics of interest to the community. > > Our ClueCon plans are gearing up as well. We are pleased to announce > that Plivo, Inc. is a brand new silver sponsor for this year's event. > Plivo is an open-source framework for > developing Web-based solutions with FreeSWITCH. We are also happy to > report that the Illinois Institute of Technology (IIT) is again with > us a media sponsor. IIT holds the annual Real-Time Communications > conference and expo where > academia and industry meet to discuss various aspects of global > telecommunications. We invite you to visit our sponsors' Web sites to > learn more about what they have to offer. > > As a reminder, ClueCon still has openings for sponsors and speakers. > Please contact us via email or at the phone number below if you have > any questions. Visit the registration > page to get signed up and > be sure to book your room at the Wyndham > . (And don't forget that the > Wyndham will shortly be renamed to the Hyatt Chicago Miracle Mile.) > > We look forward to meeting everyone in person this August! > > -- > Michael S Collins > ClueCon Team > http://www.cluecon.com > 877-7-4ACLUE > cc12-0529 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120530/6532acf2/attachment.html From krice at freeswitch.org Wed May 30 17:26:10 2012 From: krice at freeswitch.org (Ken Rice) Date: Wed, 30 May 2012 08:26:10 -0500 Subject: [Freeswitch-users] FreeSWITCH Weekly News and Notes In-Reply-To: <4FC61C90.10107@gmail.com> Message-ID: Theres a better example actually in the freeswitch src tree called mod_enum... This module is examples of Dialplan, API and APP interfaces... On 5/30/12 8:11 AM, "Nazim Aghabayov" wrote: > Hello Michael > > I've tried to fetch an audio torrent of Kens presentation but it's not > available "http://torrents.freeswitch.org/conf_call_2012-05-23.torrent" > > Is source code of the dialplan module available? > > With Best Regards, > Nazim > > On 05/29/2012 09:40 PM, Michael Collins wrote: >> Happy Tuesday to all. We hope you enjoyed your holiday weekend. >> >> We've had a steady stream of presentations on the FreeSWITCH conference call >> over the past month. Ken Rice has graciously done two presentations on how to >> create a dialplan module in C, including a fully functional example module. >> This code is extremely helpful not just in understanding how to write a >> dialplan module but in writing any module that exposes new dialplan >> applications or command line APIs. This week we will take a break from the >> formal presentations and have an open discussion on any topics of interest to >> the community. >> >> Our ClueCon plans are gearing up as well. We are pleased to announce that >> Plivo, Inc. is a brand new silver sponsor for this year's event. Plivo >> is an open-source framework for developing >> Web-based solutions with FreeSWITCH. We are also happy to report that the >> Illinois Institute of Technology (IIT) is again with us a media sponsor. IIT >> holds the annual Real-Time Communications conference and expo >> where academia and industry meet >> to discuss various aspects of global telecommunications. We invite you to >> visit our sponsors' Web sites to learn more about what they have to offer. >> >> As a reminder, ClueCon still has openings for sponsors and speakers. Please >> contact us via email or at the phone number below if you have any questions. >> Visit the registration page to >> get signed up and be sure to book your room at the Wyndham >> . (And don't forget that the >> Wyndham will shortly be renamed to the Hyatt Chicago Miracle Mile.) >> >> We look forward to meeting everyone in person this August! >> >> -- >> Michael S Collins >> ClueCon Team >> >> http://www.cluecon.com >> >> 877-7-4ACLUE >> >> cc12-0529 >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120530/21d947cc/attachment.html From nazim.aghabayov at gmail.com Wed May 30 17:30:46 2012 From: nazim.aghabayov at gmail.com (Nazim Aghabayov) Date: Wed, 30 May 2012 18:30:46 +0500 Subject: [Freeswitch-users] FreeSWITCH Weekly News and Notes In-Reply-To: References: Message-ID: <4FC62106.5090003@gmail.com> Thanks! That was what I'm looking for. On 05/30/2012 06:26 PM, Ken Rice wrote: > Theres a better example actually in the freeswitch src tree called > mod_enum... This module is examples of Dialplan, API and APP interfaces... > > > On 5/30/12 8:11 AM, "Nazim Aghabayov" wrote: > > Hello Michael > > I've tried to fetch an audio torrent of Kens presentation but > it's not available > "http://torrents.freeswitch.org/conf_call_2012-05-23.torrent" > > Is source code of the dialplan module available? > > With Best Regards, > Nazim > > On 05/29/2012 09:40 PM, Michael Collins wrote: > > Happy Tuesday to all. We hope you enjoyed your holiday weekend. > > We've had a steady stream of presentations on the FreeSWITCH > conference call over the past month. Ken Rice has graciously > done two presentations on how to create a dialplan module in > C, including a fully functional example module. This code is > extremely helpful not just in understanding how to write a > dialplan module but in writing any module that exposes new > dialplan applications or command line APIs. This week we will > take a break from the formal presentations and have an open > discussion on any topics of interest to the community. > > Our ClueCon plans are gearing up as well. We are pleased to > announce that Plivo, Inc. is a brand new silver sponsor for > this year's event. Plivo is > an open-source framework for developing Web-based solutions > with FreeSWITCH. We are also happy to report that the Illinois > Institute of Technology (IIT) is again with us a media > sponsor. IIT holds the annual Real-Time Communications > conference and expo > where academia and industry meet to discuss various aspects > of global telecommunications. We invite you to visit our > sponsors' Web sites to learn more about what they have to offer. > > As a reminder, ClueCon still has openings for sponsors and > speakers. Please contact us via email or at the phone number > below if you have any questions. Visit the registration > page to get > signed up and be sure to book your room at the Wyndham > . (And don't forget > that the Wyndham will shortly be renamed to the Hyatt Chicago > Miracle Mile.) > > We look forward to meeting everyone in person this August! > > -- > Michael S Collins > ClueCon Team > > http://www.cluecon.com > > 877-7-4ACLUE > > cc12-0529 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120530/d32c2a84/attachment-0001.html From Hector.Geraldino at ipsoft.com Wed May 30 18:28:35 2012 From: Hector.Geraldino at ipsoft.com (Hector Geraldino) Date: Wed, 30 May 2012 10:28:35 -0400 Subject: [Freeswitch-users] uuid_record and recording output format Message-ID: <6A6B4C284AD15042B429EB9D904544AD022EE80BB4@NY1-EXMB-01.ip-soft.net> Greetings, I'm using a 3rd party application (ndev dragonmobile) to get the transcription of some audio recorded by FreeSWITCH. Think about it as a voicemail transcription service. The problem I'm facing is that, when I record a session using uuid_record, the output file is encoded in PCM 16-bit @ 8khz. Correct me if I'm wrong, but my understanding is that if I want to capture audio from calls coming from the PSTN (analog/landlines), the best I can do is to record it in 8-bits (using G.711). I don't want to use sox (or any other tool) to resample the output file, and what I've tried so far is setting the sample_rate variable on the diaplan as recommended on the wiki: http://wiki.freeswitch.org/wiki/Variable_record_rate This doesn't have any effect on the generated wav file, which is still encoded in 16-bits. So my question is: does this variable affects the behavior of the uuid_record command? Or, do I really need to encode the audio output in 8-bits when the origin of the call comes from the PSTN? How is FreeSWITCH encoding the audio in 16-bits if, in theory, the best rate we can get from an analog line is 8-bits? Sorry if I'm misunderstanding something, but I'm not a telephony/voip guy, more like a java developer. Thanks for your help. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120530/a08d4af7/attachment.html From modesto at isimples.com.br Wed May 30 18:55:01 2012 From: modesto at isimples.com.br (Antonio Modesto) Date: Wed, 30 May 2012 11:55:01 -0300 Subject: [Freeswitch-users] tones.conf for Brazil Message-ID: <1338389701.3112.28.camel@modesto.localdomain.net> Hi, I am testing FreeSWITCH/FreeTDM with a digium TDM410P card, I noticed that the default tones.conf has configuration for only a few contries, does anybody know how can I get or configure by myself tone configuration for brazil? Thanks From anthony.minessale at gmail.com Wed May 30 19:14:55 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 30 May 2012 10:14:55 -0500 Subject: [Freeswitch-users] Why does both core.db and sofia_reg_internal.db have registrations tables? In-Reply-To: <1545146083A72C4DB7B66584B7E5D9840E78282C@BY2PRD0410MB377.namprd04.prod.outlook.com> References: <1545146083A72C4DB7B66584B7E5D9840E78282C@BY2PRD0410MB377.namprd04.prod.outlook.com> Message-ID: Its the groundwork for a central registration system so you can mingle the registrations of multiple protocols into one place. On Tue, May 29, 2012 at 6:17 PM, Jerry Richards wrote: > Just curious why core.db and sofia_reg_internal.db have registrations > tables?? Is this redundant? > > > > Thanks, > > Jerry > > > > > > From: Jerry Richards > Sent: Wednesday, May 23, 2012 10:06 AM > To: 'FreeSWITCH Users Help' > Subject: Channel Variable To Disable Processing of 2833 DTMF Messages? > > > > Is there a channel variable to disable/ignore 2833 DTMF messages that are > going from the internal network toward the PRI?? I have a scenario where I > am getting both inband and out-of-band DTMF (i.e. double DTMF digits). > > > > This happens when a call comes in through the PRI and routes back out the > PRI.? Inband DTMF coming from the caller is converted to out-of-band 2833 > internally, so when the audio is routed back out the PRI, it contains both > the inband DTMF and the 2833 conversion. ?In some cases, we don?t want to > use disable_dtmf, since we may need to detect inband DTMF in FreeSwitch. > > > > Thanks, > > Jerry > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From msc at freeswitch.org Wed May 30 19:48:34 2012 From: msc at freeswitch.org (Michael Collins) Date: Wed, 30 May 2012 08:48:34 -0700 Subject: [Freeswitch-users] Per-channel fsctl loglevel? In-Reply-To: References: Message-ID: On Wed, May 30, 2012 at 4:30 AM, Avi Marcus wrote: > Hi. I've been developing a script for someone to handle credit card > transactions via IVR. > Is there a way to just set the particular channel/call to fsctl loglevel > 6, rather than the entire switch..? (dtmf is shown in /log 7) > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_session_loglevel > > And.. similarly is there a way to blank out the var digits_dialed in the > xml_cdr, from within FS, before the end of the call? > Why do you need to clear it out? What information does it collect that you don't need? -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120530/fcd97b27/attachment.html From msc at freeswitch.org Wed May 30 19:50:44 2012 From: msc at freeswitch.org (Michael Collins) Date: Wed, 30 May 2012 08:50:44 -0700 Subject: [Freeswitch-users] Why I could not detect DTMF when using Originate command In-Reply-To: References: Message-ID: Make sure that the start_dtmf is on the correct leg. I suggested putting it on the b leg, but this example sets it on the a leg. Try it either way to make sure. -MC On Wed, May 30, 2012 at 5:08 AM, Michael Lutz wrote: > If your outbound provider does not support RFC2833 (DTMF via SIP Info > messages) you need to explicitly set the outbound leg to 'listen' for > DTMF tones using the start_dtmf dialplan command. > As you originate, there is no real dialplan, you can set it using the > "execute_on_answer"in inside the originate command like: > > originate > {execute_on_answer=start_dtmf,ignore_early_media=true}sofia/gateway/trunk_1/0860216060 > 8888 > > > Regards, > Mike. > > 2012/5/29 Michael Collins : > > You may need explicitly to listen for inband DTMFs.Try adding > application="start_dtmf"> to extension 8888 and retest. > > -MC > > > > > > On Thu, May 24, 2012 at 8:27 PM, Chaiyawut Sookplang > > wrote: > >> > >> I want to make an automatic IVR call using originate command. I issued > >> command > >> > >> "originate {ignore_early_media=true}sofia/gateway/trunk_1/0860216060 > 8888" > >> > >> and then I found that DTMF from the receiver side couldn't be > >> detected. On the other hand, if I call to extension 8888, IVR and DTMF > >> work fine. > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120530/fbadf847/attachment.html From krice at freeswitch.org Wed May 30 20:03:14 2012 From: krice at freeswitch.org (Ken Rice) Date: Wed, 30 May 2012 11:03:14 -0500 Subject: [Freeswitch-users] Request for a little documentation help Message-ID: Hey Guys, If you are currently maintaining a module in FreeSWITCH and you are the primary maintainer, can you please tag those in the header of the main C module below the Contributors Section and above the file description The only person exempted from this request is Tony. Everyone else please help us out... We want to make sure this is documented for a variety of reasons. See Example Below I snagged from mod_lcr.c /* * FreeSWITCH Modular Media Switching Software Library / Soft-Switch Application * Copyright (C) 2005-2011, Anthony Minessale II * * Version: MPL 1.1 * * The contents of this file are subject to the Mozilla Public License Version * 1.1 (the "License"); you may not use this file except in compliance with * the License. You may obtain a copy of the License at * http://www.mozilla.org/MPL/ * * Software distributed under the License is distributed on an "AS IS" basis, * WITHOUT WARRANTY OF ANY KIND, either express or implied. See the License * for the specific language governing rights and limitations under the * License. * * The Original Code is FreeSWITCH Modular Media Switching Software Library / Soft-Switch Application * * The Initial Developer of the Original Code is * Anthony Minessale II * Portions created by the Initial Developer are Copyright (C) * the Initial Developer. All Rights Reserved. * * Contributor(s): * * Raymond Chandler * Rupa Schomaker * * Maintainer: Raymond Chandler * * mod_lcr.c -- Least Cost Routing Module * */ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120530/f5c53bb1/attachment-0001.html From freeswitch-list at puzzled.xs4all.nl Wed May 30 20:08:52 2012 From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists) Date: Wed, 30 May 2012 18:08:52 +0200 Subject: [Freeswitch-users] Per-channel fsctl loglevel? In-Reply-To: References: Message-ID: <4FC64614.5040205@puzzled.xs4all.nl> On 30-05-12 17:48, Michael Collins wrote: > And.. similarly is there a way to blank out the var digits_dialed in > the xml_cdr, from within FS, before the end of the call? > > Why do you need to clear it out? What information does it collect that > you don't need? Since it's credit card data I can imagine Avi does not want it logged for security purposes. Regards, Patrick From msc at freeswitch.org Wed May 30 20:31:57 2012 From: msc at freeswitch.org (Michael Collins) Date: Wed, 30 May 2012 09:31:57 -0700 Subject: [Freeswitch-users] FreeSWITCH Conf Call: 30 minutes Message-ID: Hello all! The FS weekly conf call will start shortly. No formal presentation today, but several small things to talk about: http://wiki.freeswitch.org/wiki/FS_weekly_2012_05_30 Talk to you soon! -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120530/7ad7981e/attachment.html From mytemike72 at gmail.com Wed May 30 20:37:38 2012 From: mytemike72 at gmail.com (Michael Lutz) Date: Wed, 30 May 2012 18:37:38 +0200 Subject: [Freeswitch-users] Why I could not detect DTMF when using Originate command In-Reply-To: References: Message-ID: Hi Michael, Can you explain a bit more? If you would initiate an outbound call using an originate api call, there is only one leg? How would you specify to listen to the b-leg, other than this? It might explain some issues I have recognizing (valid) dtmf's... Regards, Mike. 2012/5/30 Michael Collins : > Make sure that the start_dtmf is on the correct leg. I suggested putting it > on the b leg, but this example sets it on the a leg. Try it either way to > make sure. > -MC > > > On Wed, May 30, 2012 at 5:08 AM, Michael Lutz wrote: >> >> If your outbound provider does not support RFC2833 (DTMF via SIP Info >> messages) you need to explicitly set the outbound leg to 'listen' for >> DTMF tones using the start_dtmf dialplan command. >> As you originate, there is no real dialplan, you can set it using the >> "execute_on_answer"in inside the originate command like: >> >> originate >> {execute_on_answer=start_dtmf,ignore_early_media=true}sofia/gateway/trunk_1/0860216060 >> 8888 >> >> >> Regards, >> Mike. >> >> 2012/5/29 Michael Collins : >> > You may need explicitly to listen for inband DTMFs.Try adding > > application="start_dtmf"> to extension 8888 and retest. >> > -MC >> > >> > >> > On Thu, May 24, 2012 at 8:27 PM, Chaiyawut Sookplang >> > wrote: >> >> >> >> I want to make an automatic IVR call using originate command. I issued >> >> command >> >> >> >> "originate {ignore_early_media=true}sofia/gateway/trunk_1/0860216060 >> >> 8888" >> >> >> >> and then I found that DTMF from the receiver side couldn't be >> >> detected. On the other hand, if I call to extension 8888, IVR and DTMF >> >> work fine. >> > >> > >> > >> > >> > _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > Join Us At ClueCon - Aug 7-9, 2012 >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at 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 at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From msc at freeswitch.org Wed May 30 21:11:25 2012 From: msc at freeswitch.org (Michael Collins) Date: Wed, 30 May 2012 10:11:25 -0700 Subject: [Freeswitch-users] Per-channel fsctl loglevel? In-Reply-To: <4FC64614.5040205@puzzled.xs4all.nl> References: <4FC64614.5040205@puzzled.xs4all.nl> Message-ID: If it's a compliance issue then I'd triple-check to make sure that no one unauthorized can get to any of your FS logs or CDR data. I suspect that logging vs. not logging dialed_digits is not a make-or-break proposition. If you're doing xml_cdrs then you've probably got that same data in other log lines. -MC On Wed, May 30, 2012 at 9:08 AM, Patrick Lists < freeswitch-list at puzzled.xs4all.nl> wrote: > On 30-05-12 17:48, Michael Collins wrote: > > And.. similarly is there a way to blank out the var digits_dialed in > > the xml_cdr, from within FS, before the end of the call? > > > > Why do you need to clear it out? What information does it collect that > > you don't need? > > Since it's credit card data I can imagine Avi does not want it logged > for security purposes. > > Regards, > Patrick > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120530/7789b6f7/attachment.html From msc at freeswitch.org Wed May 30 21:15:22 2012 From: msc at freeswitch.org (Michael Collins) Date: Wed, 30 May 2012 10:15:22 -0700 Subject: [Freeswitch-users] Why I could not detect DTMF when using Originate command In-Reply-To: References: Message-ID: The originate creates one leg and connects it to another leg. In this example: {execute_on_answer=start_dtmf,ignore_early_media=true}sofia/gateway/trunk_1/0860216060 8888 Everything up to 8888 creates the A leg. The "8888" is the B leg. -MC On Wed, May 30, 2012 at 9:37 AM, Michael Lutz wrote: > Hi Michael, > > Can you explain a bit more? If you would initiate an outbound call > using an originate api call, there is only one leg? How would you > specify to listen to the b-leg, other than this? > > > It might explain some issues I have recognizing (valid) dtmf's... > > Regards, > Mike. > > > 2012/5/30 Michael Collins : > > Make sure that the start_dtmf is on the correct leg. I suggested putting > it > > on the b leg, but this example sets it on the a leg. Try it either way to > > make sure. > > -MC > > > > > > On Wed, May 30, 2012 at 5:08 AM, Michael Lutz > wrote: > >> > >> If your outbound provider does not support RFC2833 (DTMF via SIP Info > >> messages) you need to explicitly set the outbound leg to 'listen' for > >> DTMF tones using the start_dtmf dialplan command. > >> As you originate, there is no real dialplan, you can set it using the > >> "execute_on_answer"in inside the originate command like: > >> > >> originate > >> > {execute_on_answer=start_dtmf,ignore_early_media=true}sofia/gateway/trunk_1/0860216060 > >> 8888 > >> > >> > >> Regards, > >> Mike. > >> > >> 2012/5/29 Michael Collins : > >> > You may need explicitly to listen for inband DTMFs.Try adding >> > application="start_dtmf"> to extension 8888 and retest. > >> > -MC > >> > > >> > > >> > On Thu, May 24, 2012 at 8:27 PM, Chaiyawut Sookplang > >> > wrote: > >> >> > >> >> I want to make an automatic IVR call using originate command. I > issued > >> >> command > >> >> > >> >> "originate {ignore_early_media=true}sofia/gateway/trunk_1/0860216060 > >> >> 8888" > >> >> > >> >> and then I found that DTMF from the receiver side couldn't be > >> >> detected. On the other hand, if I call to extension 8888, IVR and > DTMF > >> >> work fine. > >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120530/50a02800/attachment.html From mytemike72 at gmail.com Wed May 30 21:23:30 2012 From: mytemike72 at gmail.com (Michael Lutz) Date: Wed, 30 May 2012 19:23:30 +0200 Subject: [Freeswitch-users] Why I could not detect DTMF when using Originate command In-Reply-To: References: Message-ID: Ok, but with if I do an originate with a &(park) and bridge the calls later using the api bridge command using ESL. When I want to listen to the DTMF's generated by the originated (called) number, I shoul use {execute_on_answer=start_dtmf, ..} or do I need to specify the 'b-leg' and if so, how? Regards, Mike. 2012/5/30 Michael Collins : > The originate creates one leg and connects it to another leg. In this > example: > > {execute_on_answer=start_dtmf,ignore_early_media=true}sofia/gateway/trunk_1/0860216060 > 8888 > > Everything up to 8888 creates the A leg. The "8888" is the B leg. > > -MC > > > On Wed, May 30, 2012 at 9:37 AM, Michael Lutz wrote: >> >> Hi Michael, >> >> Can you explain a bit more? If you would initiate an outbound call >> using an originate api call, there is only one leg? How would you >> specify to listen to the b-leg, other than this? >> >> >> It might explain some issues I have recognizing (valid) dtmf's... >> >> Regards, >> Mike. >> >> >> 2012/5/30 Michael Collins : >> > Make sure that the start_dtmf is on the correct leg. I suggested putting >> > it >> > on the b leg, but this example sets it on the a leg. Try it either way >> > to >> > make sure. >> > -MC >> > >> > >> > On Wed, May 30, 2012 at 5:08 AM, Michael Lutz >> > wrote: >> >> >> >> If your outbound provider does not support RFC2833 (DTMF via SIP Info >> >> messages) you need to explicitly set the outbound leg to 'listen' for >> >> DTMF tones using the start_dtmf dialplan command. >> >> As you originate, there is no real dialplan, you can set it using the >> >> "execute_on_answer"in inside the originate command like: >> >> >> >> originate >> >> >> >> {execute_on_answer=start_dtmf,ignore_early_media=true}sofia/gateway/trunk_1/0860216060 >> >> 8888 >> >> >> >> >> >> Regards, >> >> Mike. >> >> >> >> 2012/5/29 Michael Collins : >> >> > You may need explicitly to listen for inband DTMFs.Try adding > >> > application="start_dtmf"> to extension 8888 and retest. >> >> > -MC >> >> > >> >> > >> >> > On Thu, May 24, 2012 at 8:27 PM, Chaiyawut Sookplang >> >> > wrote: >> >> >> >> >> >> I want to make an automatic IVR call using originate command. I >> >> >> issued >> >> >> command >> >> >> >> >> >> "originate {ignore_early_media=true}sofia/gateway/trunk_1/0860216060 >> >> >> 8888" >> >> >> >> >> >> and then I found that DTMF from the receiver side couldn't be >> >> >> detected. On the other hand, if I call to extension 8888, IVR and >> >> >> DTMF >> >> >> work fine. >> >> > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From avi at avimarcus.net Wed May 30 22:20:52 2012 From: avi at avimarcus.net (Avi Marcus) Date: Wed, 30 May 2012 21:20:52 +0300 Subject: [Freeswitch-users] Per-channel fsctl loglevel? In-Reply-To: References: <4FC64614.5040205@puzzled.xs4all.nl> Message-ID: The PCI-DSS (Payment Card Industry Data Security Standard) requires encryption, not merely permission restriction, for sensitive data. Hence I'm looking at the DTMF logging which can probably be easily re-patterned back into the digits, the curl POST which also shows everything in the log, the dialed_digits in a standard xml_cdr.. Otherwise, afaik, lua won't log things unless you explicitly tell it to. Any suggestions other than setting the entire switch to fsctl loglevel 6 and not storing the xml_cdrs in their raw form? -Avi On Wed, May 30, 2012 at 8:11 PM, Michael Collins wrote: > If it's a compliance issue then I'd triple-check to make sure that no one > unauthorized can get to any of your FS logs or CDR data. I suspect that > logging vs. not logging dialed_digits is not a make-or-break proposition. > If you're doing xml_cdrs then you've probably got that same data in other > log lines. > > -MC > > > On Wed, May 30, 2012 at 9:08 AM, Patrick Lists < > freeswitch-list at puzzled.xs4all.nl> wrote: > >> On 30-05-12 17:48, Michael Collins wrote: >> > And.. similarly is there a way to blank out the var digits_dialed in >> > the xml_cdr, from within FS, before the end of the call? >> > >> > Why do you need to clear it out? What information does it collect that >> > you don't need? >> >> Since it's credit card data I can imagine Avi does not want it logged >> for security purposes. >> >> Regards, >> Patrick >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120530/2da6bcfa/attachment-0001.html From msc at freeswitch.org Wed May 30 22:21:41 2012 From: msc at freeswitch.org (Michael Collins) Date: Wed, 30 May 2012 11:21:41 -0700 Subject: [Freeswitch-users] Why I could not detect DTMF when using Originate command In-Reply-To: References: Message-ID: If you need to do a start_dtmf on the B leg then just use uuid_broadcast: uuid_broadcast start_dtmf Keep in mind that uuid_broadcast will let you execute an arbitrary app on any leg. Check out the wiki entry: http://wiki.freeswitch.org/wiki/Mod_commands#uuid_broadcast -MC On Wed, May 30, 2012 at 10:23 AM, Michael Lutz wrote: > Ok, but with if I do an originate with a &(park) and bridge the calls > later using the api bridge command using ESL. > > When I want to listen to the DTMF's generated by the originated > (called) number, I shoul use {execute_on_answer=start_dtmf, ..} > or do I need to specify the 'b-leg' and if so, how? > > Regards, > Mike. > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120530/806139ec/attachment.html From modesto at isimples.com.br Wed May 30 22:39:28 2012 From: modesto at isimples.com.br (Antonio Modesto) Date: Wed, 30 May 2012 15:39:28 -0300 Subject: [Freeswitch-users] tones.conf for Brazil In-Reply-To: <1338389701.3112.28.camel@modesto.localdomain.net> References: <1338389701.3112.28.camel@modesto.localdomain.net> Message-ID: <1338403168.3112.32.camel@modesto.localdomain.net> On Wed, 2012-05-30 at 11:55 -0300, Antonio Modesto wrote: > Hi, > > I am testing FreeSWITCH/FreeTDM with a digium TDM410P card, I noticed > that the default tones.conf has configuration for only a few contries, > does anybody know how can I get or configure by myself tone > configuration for brazil? I found the frequencies in this website:http://pic.dhe.ibm.com/infocenter/wvraix/v6r1m0/index.jsp?topic= %2Fcom.ibm.wvraix.config.doc%2Fi572210.html and added the br section to tones.conf: [br] generate-dial => v=-7;%(1000,0,425) detect-dial => 425 generate-ring => v=-7;%(1000,4000,425) detect-ring => 425 generate-busy => v=-7;%(500,500,425) detect-busy => 425 generate-attn => v=0;%(100,100,1400,2060,2450,2600) detect-attn => 1400,2060,2450,2600 generate-callwaiting-sas => v=0;%(300,0,440) detect-callwaiting-sas => 440 generate-callwaiting-cas => v=0;%(80,0,2750,2130) detect-callwaiting-cas => 2750,2130 detect-fail1 => 913.8 detect-fail2 => 1370.6 detect-fail3 => 776.7 If somebody needs in the future... Regards. > > Thanks > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From modesto at isimples.com.br Wed May 30 22:49:59 2012 From: modesto at isimples.com.br (Antonio Modesto) Date: Wed, 30 May 2012 15:49:59 -0300 Subject: [Freeswitch-users] FreeTDM CallerID Detection with DTMF Message-ID: <1338403799.3112.43.camel@modesto.localdomain.net> Hi, I have a Digium TDM410P card, I am using it here in Brazil, where the signaling is not FSK, it's DTMF. It is working, though I am not receiving the callerid: Caller-Caller-ID-Number: [0000000000] Here is my freetdm.conf: [span zt FXO1] fxo-channel => 1 [span zt FXO2] fxo-channel => 2 [span zt FXO3] fxo-channel => 3 Here is one section of my autoload_configs/freetdm.conf.xml: Is it possible to enable the callerid detection in these conditions, or is it a hardware/driver limitation? Regards. From msc at freeswitch.org Thu May 31 00:18:46 2012 From: msc at freeswitch.org (Michael Collins) Date: Wed, 30 May 2012 13:18:46 -0700 Subject: [Freeswitch-users] Per-channel fsctl loglevel? In-Reply-To: References: <4FC64614.5040205@puzzled.xs4all.nl> Message-ID: How are you protecting everything else? If the XML CDR is sent over HTTP instead of HTTPS then everything about the call is plain text. And what about the FS logs? Are you encrypting those somehow? It seems to me that you need a more comprehensive solution than just scrubbing a single channel variable. However, if you need an interim solution I would suggest commenting out the line that sets digits_dialed: http://fisheye.freeswitch.org/browse/freeswitch.git/src/switch_channel.c?r=HEAD#to3912 A more permanent solution might be to create a channel variable that controls whether stuff like this gets logged. Something like "no_dtmf_logging=true" or whatever. That's a bit more involved because you have to decide if there are other places where DTMF info gets logged and if so, decide whether or not you want not to log them. What would be the ideal solution for your scenario? That answer might yield the best course of action. -MC On Wed, May 30, 2012 at 11:20 AM, Avi Marcus wrote: > The PCI-DSS (Payment Card Industry Data Security Standard) requires > encryption, not merely permission restriction, for sensitive data. Hence > I'm looking at the DTMF logging which can probably be easily re-patterned > back into the digits, the curl POST which also shows everything in the log, > the dialed_digits in a standard xml_cdr.. > Otherwise, afaik, lua won't log things unless you explicitly tell it to. > > Any suggestions other than setting the entire switch to fsctl loglevel 6 > and not storing the xml_cdrs in their raw form? > > -Avi > > On Wed, May 30, 2012 at 8:11 PM, Michael Collins wrote: > >> If it's a compliance issue then I'd triple-check to make sure that no one >> unauthorized can get to any of your FS logs or CDR data. I suspect that >> logging vs. not logging dialed_digits is not a make-or-break proposition. >> If you're doing xml_cdrs then you've probably got that same data in other >> log lines. >> >> -MC >> >> >> On Wed, May 30, 2012 at 9:08 AM, Patrick Lists < >> freeswitch-list at puzzled.xs4all.nl> wrote: >> >>> On 30-05-12 17:48, Michael Collins wrote: >>> > And.. similarly is there a way to blank out the var digits_dialed >>> in >>> > the xml_cdr, from within FS, before the end of the call? >>> > >>> > Why do you need to clear it out? What information does it collect that >>> > you don't need? >>> >>> Since it's credit card data I can imagine Avi does not want it logged >>> for security purposes. >>> >>> Regards, >>> Patrick >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120530/bced8273/attachment.html From avi at avimarcus.net Thu May 31 00:29:09 2012 From: avi at avimarcus.net (Avi Marcus) Date: Wed, 30 May 2012 23:29:09 +0300 Subject: [Freeswitch-users] Per-channel fsctl loglevel? In-Reply-To: References: <4FC64614.5040205@puzzled.xs4all.nl> Message-ID: On Wed, May 30, 2012 at 11:18 PM, Michael Collins wrote: > How are you protecting everything else? If the XML CDR is sent over HTTP > instead of HTTPS then everything about the call is plain text. As far as I know, the only thing sensitive in the xml_cdr is digits_dialed. > And what about the FS logs? Are you encrypting those somehow? It seems to > me that you need a more comprehensive solution than just scrubbing a single > channel variable. > No, I'm not encrypting them.. because t here wouldn't be anything sensitive. As far as I can tell, the only issue is the DTMF in DEBUG and the curl post message, again in DEBUG. Since this is a lua IVR it seems nearly nothing else makes it into the log. Only api:execute("curl",...) is in the log because it's not a native direct curl command (like session:playandgetdigits()) > However, if you need an interim solution I would suggest commenting out > the line that sets digits_dialed: > > http://fisheye.freeswitch.org/browse/freeswitch.git/src/switch_channel.c?r=HEAD#to3912 > > A more permanent solution might be to create a channel variable that > controls whether stuff like this gets logged. Something like > "no_dtmf_logging=true" or whatever. That's a bit more involved because you > have to decide if there are other places where DTMF info gets logged and if > so, decide whether or not you want not to log them. > That's an interesting idea... it might be more encompassing to have a loglevel=X channel variable instead that affects the logging for that channel. But this is probably overkill... > > What would be the ideal solution for your scenario? That answer might > yield the best course of action. > -MC > > > On Wed, May 30, 2012 at 11:20 AM, Avi Marcus wrote: > >> The PCI-DSS (Payment Card Industry Data Security Standard) requires >> encryption, not merely permission restriction, for sensitive data. Hence >> I'm looking at the DTMF logging which can probably be easily re-patterned >> back into the digits, the curl POST which also shows everything in the log, >> the dialed_digits in a standard xml_cdr.. >> Otherwise, afaik, lua won't log things unless you explicitly tell it to. >> >> Any suggestions other than setting the entire switch to fsctl loglevel 6 >> and not storing the xml_cdrs in their raw form? >> >> -Avi >> >> On Wed, May 30, 2012 at 8:11 PM, Michael Collins wrote: >> >>> If it's a compliance issue then I'd triple-check to make sure that no >>> one unauthorized can get to any of your FS logs or CDR data. I suspect that >>> logging vs. not logging dialed_digits is not a make-or-break proposition. >>> If you're doing xml_cdrs then you've probably got that same data in other >>> log lines. >>> >>> -MC >>> >>> >>> On Wed, May 30, 2012 at 9:08 AM, Patrick Lists < >>> freeswitch-list at puzzled.xs4all.nl> wrote: >>> >>>> On 30-05-12 17:48, Michael Collins wrote: >>>> > And.. similarly is there a way to blank out the var digits_dialed >>>> in >>>> > the xml_cdr, from within FS, before the end of the call? >>>> > >>>> > Why do you need to clear it out? What information does it collect that >>>> > you don't need? >>>> >>>> Since it's credit card data I can imagine Avi does not want it logged >>>> for security purposes. >>>> >>>> Regards, >>>> Patrick >>>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> Join Us At ClueCon - Aug 7-9, 2012 >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120530/8e960262/attachment-0001.html From msc at freeswitch.org Thu May 31 00:35:34 2012 From: msc at freeswitch.org (Michael Collins) Date: Wed, 30 May 2012 13:35:34 -0700 Subject: [Freeswitch-users] Per-channel fsctl loglevel? In-Reply-To: References: <4FC64614.5040205@puzzled.xs4all.nl> Message-ID: Avi, Can you think of any other places where the FS logging in general might contain sensitive data? Reason I ask is that maybe we could create something like "pcidss=true" and then use that as a flag to disable logging anything that might be considered sensitive. Just a thought. -MC On Wed, May 30, 2012 at 1:29 PM, Avi Marcus wrote: > On Wed, May 30, 2012 at 11:18 PM, Michael Collins wrote: > >> How are you protecting everything else? If the XML CDR is sent over HTTP >> instead of HTTPS then everything about the call is plain text. > > As far as I know, the only thing sensitive in the xml_cdr is digits_dialed. > > >> And what about the FS logs? Are you encrypting those somehow? It seems to >> me that you need a more comprehensive solution than just scrubbing a single >> channel variable. >> > No, I'm not encrypting them.. because t here wouldn't be anything > sensitive. As far as I can tell, the only issue is the DTMF in DEBUG and > the curl post message, again in DEBUG. > Since this is a lua IVR it seems nearly nothing else makes it into the > log. Only api:execute("curl",...) is in the log because it's not a native > direct curl command (like session:playandgetdigits()) > > >> However, if you need an interim solution I would suggest commenting out >> the line that sets digits_dialed: >> >> http://fisheye.freeswitch.org/browse/freeswitch.git/src/switch_channel.c?r=HEAD#to3912 >> >> A more permanent solution might be to create a channel variable that >> controls whether stuff like this gets logged. Something like >> "no_dtmf_logging=true" or whatever. That's a bit more involved because you >> have to decide if there are other places where DTMF info gets logged and if >> so, decide whether or not you want not to log them. >> > That's an interesting idea... it might be more encompassing to have a > loglevel=X channel variable instead that affects the logging for that > channel. But this is probably overkill... > >> >> What would be the ideal solution for your scenario? That answer might >> yield the best course of action. >> -MC >> >> >> On Wed, May 30, 2012 at 11:20 AM, Avi Marcus wrote: >> >>> The PCI-DSS (Payment Card Industry Data Security Standard) requires >>> encryption, not merely permission restriction, for sensitive data. Hence >>> I'm looking at the DTMF logging which can probably be easily re-patterned >>> back into the digits, the curl POST which also shows everything in the log, >>> the dialed_digits in a standard xml_cdr.. >>> Otherwise, afaik, lua won't log things unless you explicitly tell it to. >>> >>> Any suggestions other than setting the entire switch to fsctl loglevel 6 >>> and not storing the xml_cdrs in their raw form? >>> >>> -Avi >>> >>> On Wed, May 30, 2012 at 8:11 PM, Michael Collins wrote: >>> >>>> If it's a compliance issue then I'd triple-check to make sure that no >>>> one unauthorized can get to any of your FS logs or CDR data. I suspect that >>>> logging vs. not logging dialed_digits is not a make-or-break proposition. >>>> If you're doing xml_cdrs then you've probably got that same data in other >>>> log lines. >>>> >>>> -MC >>>> >>>> >>>> On Wed, May 30, 2012 at 9:08 AM, Patrick Lists < >>>> freeswitch-list at puzzled.xs4all.nl> wrote: >>>> >>>>> On 30-05-12 17:48, Michael Collins wrote: >>>>> > And.. similarly is there a way to blank out the var >>>>> digits_dialed in >>>>> > the xml_cdr, from within FS, before the end of the call? >>>>> > >>>>> > Why do you need to clear it out? What information does it collect >>>>> that >>>>> > you don't need? >>>>> >>>>> Since it's credit card data I can imagine Avi does not want it logged >>>>> for security purposes. >>>>> >>>>> Regards, >>>>> Patrick >>>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> Join Us At ClueCon - Aug 7-9, 2012 >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at 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 at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> Join Us At ClueCon - Aug 7-9, 2012 >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120530/826f3f38/attachment.html From avi at avimarcus.net Thu May 31 01:25:55 2012 From: avi at avimarcus.net (Avi Marcus) Date: Thu, 31 May 2012 00:25:55 +0300 Subject: [Freeswitch-users] Per-channel fsctl loglevel? In-Reply-To: References: <4FC64614.5040205@puzzled.xs4all.nl> Message-ID: Mostly credit cards.. and anything you do with it, e.g. submit it via https to authorize.net, stripe, etc where then you don't need the actual number anymore. So both the DTMF entry and the curl debug line. I can't think of anything else in particular. -Avi On Wed, May 30, 2012 at 11:35 PM, Michael Collins wrote: > Avi, > > Can you think of any other places where the FS logging in general might > contain sensitive data? Reason I ask is that maybe we could create > something like "pcidss=true" and then use that as a flag to disable logging > anything that might be considered sensitive. Just a thought. > > -MC > > > On Wed, May 30, 2012 at 1:29 PM, Avi Marcus wrote: > >> On Wed, May 30, 2012 at 11:18 PM, Michael Collins wrote: >> >>> How are you protecting everything else? If the XML CDR is sent over HTTP >>> instead of HTTPS then everything about the call is plain text. >> >> As far as I know, the only thing sensitive in the xml_cdr is >> digits_dialed. >> >> >>> And what about the FS logs? Are you encrypting those somehow? It seems >>> to me that you need a more comprehensive solution than just scrubbing a >>> single channel variable. >>> >> No, I'm not encrypting them.. because t here wouldn't be anything >> sensitive. As far as I can tell, the only issue is the DTMF in DEBUG and >> the curl post message, again in DEBUG. >> Since this is a lua IVR it seems nearly nothing else makes it into the >> log. Only api:execute("curl",...) is in the log because it's not a native >> direct curl command (like session:playandgetdigits()) >> >> >>> However, if you need an interim solution I would suggest commenting out >>> the line that sets digits_dialed: >>> >>> http://fisheye.freeswitch.org/browse/freeswitch.git/src/switch_channel.c?r=HEAD#to3912 >>> >>> A more permanent solution might be to create a channel variable that >>> controls whether stuff like this gets logged. Something like >>> "no_dtmf_logging=true" or whatever. That's a bit more involved because you >>> have to decide if there are other places where DTMF info gets logged and if >>> so, decide whether or not you want not to log them. >>> >> That's an interesting idea... it might be more encompassing to have a >> loglevel=X channel variable instead that affects the logging for that >> channel. But this is probably overkill... >> >>> >>> What would be the ideal solution for your scenario? That answer might >>> yield the best course of action. >>> -MC >>> >>> >>> On Wed, May 30, 2012 at 11:20 AM, Avi Marcus wrote: >>> >>>> The PCI-DSS (Payment Card Industry Data Security Standard) requires >>>> encryption, not merely permission restriction, for sensitive data. Hence >>>> I'm looking at the DTMF logging which can probably be easily re-patterned >>>> back into the digits, the curl POST which also shows everything in the log, >>>> the dialed_digits in a standard xml_cdr.. >>>> Otherwise, afaik, lua won't log things unless you explicitly tell it to. >>>> >>>> Any suggestions other than setting the entire switch to fsctl loglevel >>>> 6 and not storing the xml_cdrs in their raw form? >>>> >>>> -Avi >>>> >>>> On Wed, May 30, 2012 at 8:11 PM, Michael Collins wrote: >>>> >>>>> If it's a compliance issue then I'd triple-check to make sure that no >>>>> one unauthorized can get to any of your FS logs or CDR data. I suspect that >>>>> logging vs. not logging dialed_digits is not a make-or-break proposition. >>>>> If you're doing xml_cdrs then you've probably got that same data in other >>>>> log lines. >>>>> >>>>> -MC >>>>> >>>>> >>>>> On Wed, May 30, 2012 at 9:08 AM, Patrick Lists < >>>>> freeswitch-list at puzzled.xs4all.nl> wrote: >>>>> >>>>>> On 30-05-12 17:48, Michael Collins wrote: >>>>>> > And.. similarly is there a way to blank out the var >>>>>> digits_dialed in >>>>>> > the xml_cdr, from within FS, before the end of the call? >>>>>> > >>>>>> > Why do you need to clear it out? What information does it collect >>>>>> that >>>>>> > you don't need? >>>>>> >>>>>> Since it's credit card data I can imagine Avi does not want it logged >>>>>> for security purposes. >>>>>> >>>>>> Regards, >>>>>> Patrick >>>>>> >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> Join Us At ClueCon - Aug 7-9, 2012 >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at 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 at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> Join Us At ClueCon - Aug 7-9, 2012 >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at 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 at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> Join Us At ClueCon - Aug 7-9, 2012 >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120531/8243f16b/attachment-0001.html From ahe.sanath at gmail.com Thu May 31 06:29:53 2012 From: ahe.sanath at gmail.com (Sanath Prasanna) Date: Thu, 31 May 2012 07:59:53 +0530 Subject: [Freeswitch-users] Run LUA script in different server In-Reply-To: References: Message-ID: Hi, I transfer some calls from Server A Freeswitch box to Server B Freeswitch box. I need to get value of variable_sip_history_info parameter in Server B. It is coming to Server A. But not coming to Server B. Also I play few announcement from Server B. It is not hear to caller even though it is play from server B box. Pls help to sort out above 2 problems. Server A ip: 10.1.1.252 Server B ip: 10.1.1.253 Here is the dial plan of server A. Here is the acl.conf.xml server B Server A: Br, Sanath On Thu, May 24, 2012 at 11:30 AM, Sanath Prasanna wrote: > Hi all, > > Tx a lot for advice. Problem is port. Now I sorted it & call coming to B > server. In here diverted call coming to server A. So I used > variable_sip_history_info > parameter to extarct real B number.(Real called party number) . But in > server B, that parameter is not coming. (variable_sip_history_info) Pls > help to solve that. > Br, > Sanath > > >> On Tue, May 22, 2012 at 12:53 AM, Michael Collins wrote: >> >>> Look at line #186 of your trace: >>> 2012-05-21 07:56:49.735660 [ERR] mod_sofia.c:3957 Invalid Profile >>> >>> You need to figure out why your internal profile isn't running. Try >>> "sofia profile internal restart" and see what happens. >>> >>> -MC >>> >>> >>> On Sun, May 20, 2012 at 10:45 PM, Sanath Prasanna wrote: >>> >>>> Hi MC, >>>> I did the change according to ure instruction. But error is coming. >>>> Here I attached freeswitch.log file >>>> >>>> I change the confs as follows in BOX A. (Operator connected Freeswitch >>>> box) >>>> BOX B ip is 10.22.29.253 >>>> >>>> vi /usr/local/freeswitch/conf/dialplan/default.xml >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> Also add following to ACL file in BOX B >>>> >>>> Pls advice to solve the problem here. >>>> Br, >>>> Sanath >>>> >>>> >>>> On Fri, May 18, 2012 at 9:57 AM, Michael Collins wrote: >>>> >>>>> If I understand your question correctly, yes you can do this. You can >>>>> send calls from one FreeSWITCH server to another. Start here: >>>>> http://wiki.freeswitch.org/wiki/Connect_Two_FreeSWITCH_Boxes >>>>> >>>>> Best way to learn is to get the FreeSWITCH books from Packt Publishing >>>>> and just start hacking code. >>>>> >>>>> -MC >>>>> >>>>> >>>>> On Thu, May 17, 2012 at 9:07 PM, Sanath Prasanna >>>> > wrote: >>>>> >>>>>> Tx for advice MC & Anita. Can I do work around like this . >>>>>> Another freeswitch instant will be start in other server & calls will >>>>>> be transfer from operator connected freeswitch instance to this new >>>>>> freeswitch instance & vise versa. Pls advice. >>>>>> >>>>>> >>>>>> On Thu, May 17, 2012 at 5:05 PM, Anita Hall >>>>> > wrote: >>>>>> >>>>>>> You could run a Lua ESL server on a different machine but this will >>>>>>> not be the same as running a Lua script. >>>>>>> http://wiki.freeswitch.org/wiki/Event_Socket_Library >>>>>>> >>>>>>> regards, >>>>>>> Anita >>>>>>> >>>>>>> >>>>>>> >>>>>>> On Thu, May 17, 2012 at 4:37 AM, Michael Collins >>>>>> > wrote: >>>>>>> >>>>>>>> I don't think you can directly do what you are describing. However, >>>>>>>> you might be able to use mod_httapi for this. There's some documentation on >>>>>>>> the wiki and in the module. Keep in mind that this is a relatively new >>>>>>>> module so we don't have lots of examples yet, so you'll probably be doing a >>>>>>>> fair amount of research and testing. >>>>>>>> >>>>>>>> -MC >>>>>>>> >>>>>>>> >>>>>>>> On Wed, May 16, 2012 at 5:59 AM, Sanath Prasanna < >>>>>>>> ahe.sanath at gmail.com> wrote: >>>>>>>> >>>>>>>>> Hi all, >>>>>>>>> I have 2 servers. One server has SIP GW connection From Operator & >>>>>>>>> IVR applications need to build in other server. How to call distributed LUA >>>>>>>>> applications with Mysql Databases from the SIP GW server ? Pls advice. >>>>>>>>> Main idea is, maintaining SIP connection in one server & all the IVR >>>>>>>>> applications in other server. >>>>>>>>> Br, >>>>>>>>> Sanath >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> Join Us At ClueCon - Aug 7-9, 2012 >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120531/4225206c/attachment.html From bdfoster at endigotech.com Thu May 31 06:33:57 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Wed, 30 May 2012 22:33:57 -0400 Subject: [Freeswitch-users] Run LUA script in different server In-Reply-To: References: Message-ID: Start another thread please. Brian Foster Endigo Computer LLC Sent from a mobile device. On May 30, 2012 10:30 PM, "Sanath Prasanna" wrote: > Hi, > I transfer some calls from Server A Freeswitch box to Server B Freeswitch > box. I need to get value of variable_sip_history_info parameter in Server > B. It is coming to Server A. But not coming to Server B. Also I play few > announcement from Server B. It is not hear to caller even though it is play > from server B box. > Pls help to sort out above 2 problems. > > Server A ip: 10.1.1.252 > Server B ip: 10.1.1.253 > > Here is the dial plan of server A. > > > > > > > > > Here is the acl.conf.xml > server B > > > > > Server A: > > > > > > Br, > Sanath > > On Thu, May 24, 2012 at 11:30 AM, Sanath Prasanna wrote: > >> Hi all, >> >> Tx a lot for advice. Problem is port. Now I sorted it & call coming to >> B server. In here diverted call coming to server A. So I used >> variable_sip_history_info >> parameter to extarct real B number.(Real called party number) . But in >> server B, that parameter is not coming. (variable_sip_history_info) Pls >> help to solve that. >> Br, >> Sanath >> >> >>> On Tue, May 22, 2012 at 12:53 AM, Michael Collins wrote: >>> >>>> Look at line #186 of your trace: >>>> 2012-05-21 07:56:49.735660 [ERR] mod_sofia.c:3957 Invalid Profile >>>> >>>> You need to figure out why your internal profile isn't running. Try >>>> "sofia profile internal restart" and see what happens. >>>> >>>> -MC >>>> >>>> >>>> On Sun, May 20, 2012 at 10:45 PM, Sanath Prasanna >>> > wrote: >>>> >>>>> Hi MC, >>>>> I did the change according to ure instruction. But error is coming. >>>>> Here I attached freeswitch.log file >>>>> >>>>> I change the confs as follows in BOX A. (Operator connected Freeswitch >>>>> box) >>>>> BOX B ip is 10.22.29.253 >>>>> >>>>> vi /usr/local/freeswitch/conf/dialplan/default.xml >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> Also add following to ACL file in BOX B >>>>> >>>>> Pls advice to solve the problem here. >>>>> Br, >>>>> Sanath >>>>> >>>>> >>>>> On Fri, May 18, 2012 at 9:57 AM, Michael Collins wrote: >>>>> >>>>>> If I understand your question correctly, yes you can do this. You can >>>>>> send calls from one FreeSWITCH server to another. Start here: >>>>>> http://wiki.freeswitch.org/wiki/Connect_Two_FreeSWITCH_Boxes >>>>>> >>>>>> Best way to learn is to get the FreeSWITCH books from Packt >>>>>> Publishing and just start hacking code. >>>>>> >>>>>> -MC >>>>>> >>>>>> >>>>>> On Thu, May 17, 2012 at 9:07 PM, Sanath Prasanna < >>>>>> ahe.sanath at gmail.com> wrote: >>>>>> >>>>>>> Tx for advice MC & Anita. Can I do work around like this . >>>>>>> Another freeswitch instant will be start in other server & calls >>>>>>> will be transfer from operator connected freeswitch instance to this new >>>>>>> freeswitch instance & vise versa. Pls advice. >>>>>>> >>>>>>> >>>>>>> On Thu, May 17, 2012 at 5:05 PM, Anita Hall < >>>>>>> anita.hall at simmortel.com> wrote: >>>>>>> >>>>>>>> You could run a Lua ESL server on a different machine but this will >>>>>>>> not be the same as running a Lua script. >>>>>>>> http://wiki.freeswitch.org/wiki/Event_Socket_Library >>>>>>>> >>>>>>>> regards, >>>>>>>> Anita >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> On Thu, May 17, 2012 at 4:37 AM, Michael Collins < >>>>>>>> msc at freeswitch.org> wrote: >>>>>>>> >>>>>>>>> I don't think you can directly do what you are describing. >>>>>>>>> However, you might be able to use mod_httapi for this. There's some >>>>>>>>> documentation on the wiki and in the module. Keep in mind that this is a >>>>>>>>> relatively new module so we don't have lots of examples yet, so you'll >>>>>>>>> probably be doing a fair amount of research and testing. >>>>>>>>> >>>>>>>>> -MC >>>>>>>>> >>>>>>>>> >>>>>>>>> On Wed, May 16, 2012 at 5:59 AM, Sanath Prasanna < >>>>>>>>> ahe.sanath at gmail.com> wrote: >>>>>>>>> >>>>>>>>>> Hi all, >>>>>>>>>> I have 2 servers. One server has SIP GW connection From Operator >>>>>>>>>> & IVR applications need to build in other server. How to call distributed >>>>>>>>>> LUA applications with Mysql Databases from the SIP GW server ? Pls advice. >>>>>>>>>> Main idea is, maintaining SIP connection in one server & all the IVR >>>>>>>>>> applications in other server. >>>>>>>>>> Br, >>>>>>>>>> Sanath >>>>>>>>>> >>>>>>>>>> >>>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>>>>> >>>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> Join Us At ClueCon - Aug 7-9, 2012 >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120530/1e76088f/attachment-0001.html From ahe.sanath at gmail.com Thu May 31 08:04:54 2012 From: ahe.sanath at gmail.com (Sanath Prasanna) Date: Thu, 31 May 2012 09:34:54 +0530 Subject: [Freeswitch-users] Transfer calls from Server A Freeswitch box to Server B Freeswitch box. Message-ID: Hi, I transfer some calls from Server A Freeswitch box to Server B Freeswitch box. I need to get value of variable_sip_history_info parameter in Server B. It is coming to Server A. But not coming to Server B. Also I play few announcement from Server B. It is not hear to caller even though it is play from server B box. Pls help to sort out above 2 problems. Server A ip: 10.1.1.252 Server B ip: 10.1.1.253 Here is the dial plan of server A. Here is the acl.conf.xml server B Server A: Br, Sanath -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120531/01e663fb/attachment.html From saami_mh at ymail.com Thu May 31 08:59:23 2012 From: saami_mh at ymail.com (Samira Mh) Date: Wed, 30 May 2012 21:59:23 -0700 (PDT) Subject: [Freeswitch-users] how to take the result of running lua script on dialplan of freeswitch? Message-ID: <1338440363.32524.YahooMailNeo@web120102.mail.ne1.yahoo.com> hi guys, ? ? ? ? ? ? i have created lua script like this: /usr/local/freeswitch/script/welcome.lua #!/usr/bin/lua local sf = string.format function binary_op(a, b, callback) ? ? ? ? return callback(a, b); end function ?plus(a, b) return a + b end function ?minus(a, b) return a - b end function ?times(a, b) return a * b end print(sf("Plus : %d", binary_op(5, 5, plus))); print(sf("Minus: %d", binary_op(5, 5, minus))); print(sf("Times: %d", binary_op(5, 5, times))); the result of running that is 3 number, but how can i get the 3 number in dialplan of ?freeswitch so that use the result of this number on it ? thanks so much -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120530/85ec3f8f/attachment.html From spencer at 5ninesolutions.com Thu May 31 09:59:41 2012 From: spencer at 5ninesolutions.com (Spencer Thomason) Date: Wed, 30 May 2012 22:59:41 -0700 Subject: [Freeswitch-users] Subscribe for MWI In-Reply-To: <1338282786982-7579202.post@n2.nabble.com> References: <9D1F6AAA-D64F-45BC-A70D-C6E469D38C30@5ninesolutions.com> <8b33f052-3895-4252-8f47-6ce672e810c6@blur> <850DE19F-0F8B-4E4F-9B22-534311287478@5ninesolutions.com> <1338282624250-7579201.post@n2.nabble.com> <1338282786982-7579202.post@n2.nabble.com> Message-ID: Hello, Our setup is as follows: Kamailio and FreeSWITCH run on the same machine on different ports. User accounts are managed with a custom Django application and stored in a Postgresql database. Kamailio handles registrations, auth, and NAT traversal using rtpproxy. Freeswitch is used for voicemail and CDR accounting. The cdrs are POSTed back to the local webserver using mod_json_cdr where the Django app performs rating and billing. We use 4 profiles for FreeSWITCH: Inbound(from PSTN), Outbound(to PSTN), Onnet(user to user), and Media(voicemail). Kamailio does all the necessary routing to the individual profiles and then we use mod_xml_curl to get the user info to FreeSWITCH for voicemail settings, etc. We then send the call back to Kamailio for routing either to another user, the PSTN or a failover destination. The reason for this complexity is so that when we need to scale, each piece can be scaled independently as needed(i.e. web server, database, cdr collection webserver, inbound proxy, outbound proxy, b2bua, and media server. To more concisely answer your original question, the basics are this: UAC sends a SUBSCRIBE to FreeSWITCH. Auth is performed by the proxy and the credentials are then stripped and the message is forwarded to the media profile. The proxy is authed by ACL. If the UAC is behind NAT, we replace the Contact header host with the IP of the Proxy so that the NOTIFYs get sent back to the proxy to handle NAT. FreeSWITCH will then send the NOTIFY to the proxy where we lookup the real contact address from the registration table and forward the packet to that URI. I hope this helps. Best, Spencer On May 29, 2012, at 2:13 AM, hiryu23 wrote: > Hi Spencer > > I am configuring something similar. May i know what is your configuration > for Kamailio for routing packets to FreeSwitch? > > Sorry i am pretty new here. > > Thanks in advance > > -- > View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Subscribe-for-MWI-tp7557448p7579202.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From spencer at 5ninesolutions.com Thu May 31 10:03:04 2012 From: spencer at 5ninesolutions.com (Spencer Thomason) Date: Wed, 30 May 2012 23:03:04 -0700 Subject: [Freeswitch-users] Subscribe for MWI In-Reply-To: References: <9D1F6AAA-D64F-45BC-A70D-C6E469D38C30@5ninesolutions.com> <8b33f052-3895-4252-8f47-6ce672e810c6@blur> <850DE19F-0F8B-4E4F-9B22-534311287478@5ninesolutions.com> <1338282624250-7579201.post@n2.nabble.com> <1338282786982-7579202.post@n2.nabble.com> Message-ID: Sorry its late here. This should read: > UAC sends a SUBSCRIBE to FreeSWITCH. UAC sends a SUBSCRIBE to Kamailio (or other proxy). On May 30, 2012, at 10:59 PM, Spencer Thomason wrote: > Hello, > Our setup is as follows: > Kamailio and FreeSWITCH run on the same machine on different ports. User accounts are managed with a custom Django application and stored in a Postgresql database. Kamailio handles registrations, auth, and NAT traversal using rtpproxy. Freeswitch is used for voicemail and CDR accounting. The cdrs are POSTed back to the local webserver using mod_json_cdr where the Django app performs rating and billing. We use 4 profiles for FreeSWITCH: Inbound(from PSTN), Outbound(to PSTN), Onnet(user to user), and Media(voicemail). Kamailio does all the necessary routing to the individual profiles and then we use mod_xml_curl to get the user info to FreeSWITCH for voicemail settings, etc. We then send the call back to Kamailio for routing either to another user, the PSTN or a failover destination. The reason for this complexity is so that when we need to scale, each piece can be scaled independently as needed(i.e. web server, database, cdr collection webserver, inbound proxy, outbound proxy, b2bua, and media server. To more concisely answer your original question, the basics are this: > > UAC sends a SUBSCRIBE to FreeSWITCH. > Auth is performed by the proxy and the credentials are then stripped and the message is forwarded to the media profile. > The proxy is authed by ACL. > If the UAC is behind NAT, we replace the Contact header host with the IP of the Proxy so that the NOTIFYs get sent back to the proxy to handle NAT. > FreeSWITCH will then send the NOTIFY to the proxy where we lookup the real contact address from the registration table and forward the packet to that URI. > > I hope this helps. > > Best, > Spencer > > On May 29, 2012, at 2:13 AM, hiryu23 wrote: > >> Hi Spencer >> >> I am configuring something similar. May i know what is your configuration >> for Kamailio for routing packets to FreeSwitch? >> >> Sorry i am pretty new here. >> >> Thanks in advance >> >> -- >> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Subscribe-for-MWI-tp7557448p7579202.html >> Sent from the freeswitch-users mailing list archive at Nabble.com. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From peter.olsson at visionutveckling.se Thu May 31 10:10:00 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Thu, 31 May 2012 06:10:00 +0000 Subject: [Freeswitch-users] uuid_record and recording output format In-Reply-To: <6A6B4C284AD15042B429EB9D904544AD022EE80BB4@NY1-EXMB-01.ip-soft.net> References: <6A6B4C284AD15042B429EB9D904544AD022EE80BB4@NY1-EXMB-01.ip-soft.net> Message-ID: The sample rate is the rate, so 8000 is the same as 8khz, it's got nothing to do with 8 or 16 bit. FreeSWITCH always converts to L16 (linear 16-bit), so that's why you get this result. You can force to record to a raw file, for instance, try recording to testfile.PCMA to record in G7.11 alaw format. /Peter 30 maj 2012 kl. 16:37 skrev "Hector Geraldino" >: Greetings, I?m using a 3rd party application (ndev dragonmobile) to get the transcription of some audio recorded by FreeSWITCH. Think about it as a voicemail transcription service. The problem I?m facing is that, when I record a session using uuid_record, the output file is encoded in PCM 16-bit @ 8khz. Correct me if I?m wrong, but my understanding is that if I want to capture audio from calls coming from the PSTN (analog/landlines), the best I can do is to record it in 8-bits (using G.711). I don?t want to use sox (or any other tool) to resample the output file, and what I?ve tried so far is setting the sample_rate variable on the diaplan as recommended on the wiki: http://wiki.freeswitch.org/wiki/Variable_record_rate This doesn?t have any effect on the generated wav file, which is still encoded in 16-bits. So my question is: does this variable affects the behavior of the uuid_record command? Or, do I really need to encode the audio output in 8-bits when the origin of the call comes from the PSTN? How is FreeSWITCH encoding the audio in 16-bits if, in theory, the best rate we can get from an analog line is 8-bits? Sorry if I?m misunderstanding something, but I?m not a telephony/voip guy, more like a java developer. Thanks for your help. !DSPAM:4fc62d7332761385138176! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4fc62d7332761385138176! From avi at avimarcus.net Thu May 31 10:15:07 2012 From: avi at avimarcus.net (Avi Marcus) Date: Thu, 31 May 2012 09:15:07 +0300 Subject: [Freeswitch-users] how to take the result of running lua script on dialplan of freeswitch? In-Reply-To: <1338440363.32524.YahooMailNeo@web120102.mail.ne1.yahoo.com> References: <1338440363.32524.YahooMailNeo@web120102.mail.ne1.yahoo.com> Message-ID: Set a channel variable in Lua: http://wiki.freeswitch.org/wiki/Lua#session:setVariable -Avi On Thu, May 31, 2012 at 7:59 AM, Samira Mh wrote: > hi guys, > > > > > > > > > i have created lua script like this: > /usr/local/freeswitch/script/welcome.lua > > #!/usr/bin/lua > > local sf = string.format > > function binary_op(a, b, callback) > return callback(a, b); > end > > function plus(a, b) return a + b end > function minus(a, b) return a - b end > function times(a, b) return a * b end > > print(sf("Plus : %d", binary_op(5, 5, plus))); > print(sf("Minus: %d", binary_op(5, 5, minus))); > print(sf("Times: %d", binary_op(5, 5, times))); > > the result of running that is 3 number, > but how can i get the 3 number in dialplan of freeswitch so that use the > result of this number on it ? > thanks so much > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120531/c0febbc9/attachment-0001.html From saami_mh at ymail.com Thu May 31 10:33:02 2012 From: saami_mh at ymail.com (Samira Mh) Date: Wed, 30 May 2012 23:33:02 -0700 (PDT) Subject: [Freeswitch-users] how to take the result of running lua script on dialplan of freeswitch?(solved) In-Reply-To: References: <1338440363.32524.YahooMailNeo@web120102.mail.ne1.yahoo.com> Message-ID: <1338445982.38790.YahooMailNeo@web120101.mail.ne1.yahoo.com> hi?Avi, thanks so much,solved problem... ________________________________ From: Avi Marcus To: FreeSWITCH Users Help Sent: Thursday, May 31, 2012 10:45 AM Subject: Re: [Freeswitch-users] how to take the result of running lua script on dialplan of freeswitch? Set a channel variable in Lua:? http://wiki.freeswitch.org/wiki/Lua#session:setVariable -Avi On Thu, May 31, 2012 at 7:59 AM, Samira Mh wrote: hi guys, > >? >? >? ? >? >? > > > >i have created lua script like this: >/usr/local/freeswitch/script/welcome.lua > > >#!/usr/bin/lua > > >local sf = string.format > > >function binary_op(a, b, callback) >? ? ? ? return callback(a, b); >end > > >function ?plus(a, b) return a + b end >function ?minus(a, b) return a - b end >function ?times(a, b) return a * b end > > >print(sf("Plus : %d", binary_op(5, 5, plus))); >print(sf("Minus: %d", binary_op(5, 5, minus))); >print(sf("Times: %d", binary_op(5, 5, times))); > > >the result of running that is 3 number, >but how can i get the 3 number in dialplan of ?freeswitch so that use the result of this number on it ? >thanks so much > > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >Join Us At ClueCon - Aug 7-9, 2012 > >FreeSWITCH-users mailing list >FreeSWITCH-users at 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 at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120530/89aeb4b5/attachment.html From gmaruzz at gmail.com Thu May 31 10:48:44 2012 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Thu, 31 May 2012 08:48:44 +0200 Subject: [Freeswitch-users] Request for a little documentation help In-Reply-To: References: Message-ID: You know, the Dude abides... commit 07bc7ba7639726985daa571ea20490c7d3dc75ce Author: Giovanni Maruzzelli Date: Thu May 31 07:40:59 2012 +0200 gsmopen and skypopen: updated License headers http://www.youtube.com/watch?v=TE_oIIIwWl0 On Wed, May 30, 2012 at 6:03 PM, Ken Rice wrote: > Hey Guys, > > If you are currently maintaining a module in FreeSWITCH and you are the > primary maintainer, can you please tag those in the header of the main C > module below the Contributors Section and above the file description > > The only person exempted from this request is Tony. Everyone else please > help us out... We want to make sure this is documented for a variety of > reasons. > > See Example Below I snagged from mod_lcr.c > > /* > ?* FreeSWITCH Modular Media Switching Software Library / Soft-Switch > Application > ?* Copyright (C) 2005-2011, Anthony Minessale II > ?* > ?* Version: MPL 1.1 > ?* > ?* The contents of this file are subject to the Mozilla Public License > Version > ?* 1.1 (the "License"); you may not use this file except in compliance with > ?* the License. You may obtain a copy of the License at > ?* http://www.mozilla.org/MPL/ > ?* > ?* Software distributed under the License is distributed on an "AS IS" > basis, > ?* WITHOUT WARRANTY OF ANY KIND, either express or implied. See the License > ?* for the specific language governing rights and limitations under the > ?* License. > ?* > ?* The Original Code is FreeSWITCH Modular Media Switching Software Library > / Soft-Switch Application > ?* > ?* The Initial Developer of the Original Code is > ?* Anthony Minessale II > ?* Portions created by the Initial Developer are Copyright (C) > ?* the Initial Developer. All Rights Reserved. > ?* > ?* Contributor(s): > ?* > ?* Raymond Chandler > ?* Rupa Schomaker > ?* > ?* Maintainer: Raymond Chandler > ?* > ?* mod_lcr.c -- Least Cost Routing Module > ?* > ?*/ > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From sameer2k3t at gmail.com Thu May 31 11:15:03 2012 From: sameer2k3t at gmail.com (Sameer Khan) Date: Thu, 31 May 2012 11:15:03 +0400 Subject: [Freeswitch-users] Chat Event in Dingaling Message-ID: Is this implemented? There is no event fired when a chat message is received -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120531/17275a6e/attachment.html From mytemike72 at gmail.com Thu May 31 13:12:20 2012 From: mytemike72 at gmail.com (Michael Lutz) Date: Thu, 31 May 2012 11:12:20 +0200 Subject: [Freeswitch-users] DTMF Passthrough on different legs Message-ID: Hi Guys, I am stuck with a bit of a problem I cannot figure out where it is going wrong.. Here's my situation, I have an inbound call, comming from an external provider, they don't support RFC2833 so I turn on inband detection using 'start_dtmf' in the dialplan. This works ok, for the incomming channel and IVR provided (via Lua) to the caller. (Though sometimes it seems to misinterpret and returns wrong keys..) (Session:A) After that, this call is bridged (via Lua to a a new inbound session on the same switch) (without specifying start_dtmf) so RC2388 is used by default (?) (Session:B) This new sessions is playing soem audio (Lua) and is heard by the original caller (the two sessions are bridged). Seperatly I have an inbound ESL connection where I do an origanate using a originate {execute_on_answer=start_dtmf, .... etc) &park() to setup an new external call via the same gateway (they don't support RFC2388 so I activate the inband detection on the b-leg to receive the digits. Let's call this leg Session:C. After the orignated call is answered I bridge the parked call with the previous session (Session:B), so at the end Session:A and Session:B can talk to each other. Al this works perfectly. [Incommming call] Session:A ----- INBAND ------> Bridge(FS.Lua) ---> Session:B ---- RFC2388 ----> FS.Bridge ---> Session:C --- INBAND ------> ESL destination number My symptoms are... 1. I have a customer who had a IVR on his own side, (Session:C) and requires input from Session:A. The weird thing is that it seems the input is wrong. Keys entered do not correspond to what his IVR is receiving. As if newly dtmfs are generated somewhere. When I switch to another inbound provider, making Session:A using rfc2388 and not inband, but do dialout via the provider using inband detection. It seems to work fine. 2. Not being able to get the dtmf's from the ESL Destination number (Session:C). I'm reading this by subscribing to events via ESL for that particulair uuid. When I have a connection (inbound) without inband detection it seems to get the DTMF's from Session:C without any problems. Which makes it weird for me. note: I know the setup is a bit odd, but Session A cannot exit it's Lua script, so I needed to do a workaround to be able to keep te lua alive and being able to control the call (audio) via ESL. I realy need some help on this one, becuase we have a lot of customers complaining about not responding to dtmf's or receving invalid input in the (FS) IVR. Thanks for you help, Regards, Mike. From mytemike72 at gmail.com Thu May 31 13:19:59 2012 From: mytemike72 at gmail.com (Michael Lutz) Date: Thu, 31 May 2012 11:19:59 +0200 Subject: [Freeswitch-users] Monitoring audio quality Message-ID: Hi Guys, I have a prety large system with a lot of traffic. Sometimes I get complains from customers about 'loud noises', 'not hearing audio', or just 'bad quality audio'. I can see a lot of info on the CDR's, no I have tried to search the wiki and groups for some explanaition about how to interpret them, but there is not much to find.. Is there someone who can tell me what these mean, and how to interpret them, could they help me in finding issues with audio? 573964 572932 3337 3331 47 0 0 0 6 0 485556 485556 2823 2823 0 0 0 0 0 Or are there other (better) methods of finding problems related to bad audio? Regards, Mike From ash at archerdrive.com Thu May 31 13:30:46 2012 From: ash at archerdrive.com (Ash) Date: Thu, 31 May 2012 19:30:46 +1000 Subject: [Freeswitch-users] Monitoring audio quality In-Reply-To: References: Message-ID: <2793676B-4E24-45F9-942B-FB5552A6EF06@archerdrive.com> Hi Michael, Not exactly an answer to your question, but I have quite a lot of call traffic running over three servers. I have found the best way to manage this is by using a monitoring tool. The one I use is http://www.voipmonitor.org/, the actual daemon is open source but you can buy a webUI which uses rrdtool to graph the call quality and display the calls history. Using a tool like this you can get an idea of call quality for each call. Cheers, Ash. On 31/05/2012, at 7:19 PM, Michael Lutz wrote: > Hi Guys, > > I have a prety large system with a lot of traffic. Sometimes I get > complains from customers about 'loud noises', 'not hearing audio', or > just 'bad quality audio'. > I can see a lot of info on the CDR's, no I have tried to search the > wiki and groups for some explanaition about how to interpret them, but > there is not much to find.. > > Is there someone who can tell me what these mean, and how to interpret > them, could they help me in finding issues with audio? > > 573964 > 572932 > 3337 > 3331 > 47 > 0 > 0 > 0 > 6 > 0 > 485556 > 485556 > 2823 > 2823 > 0 > 0 > 0 > 0 > 0 > > Or are there other (better) methods of finding problems related to bad audio? > > > Regards, > Mike > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From andy at fabulous4.co.uk Thu May 31 15:44:50 2012 From: andy at fabulous4.co.uk (Andy Ayers) Date: Thu, 31 May 2012 12:44:50 +0100 Subject: [Freeswitch-users] random DTMF after bridge attempt timeout in Javascript Message-ID: <011001cd3f22$ca8ba600$5fa2f200$@fabulous4.co.uk> Hi, I have the following script snippet (simplified for the purposes of this request): if (session.ready()) { // Play a welcome message, "please wait while we try to connect your call" session.sayPhrase('play', 'welcome.wav'); session.execute("set", "ringback=%(400,200,400,450);%(400,2000,400,450)"); session.execute("bridge", "{effective_caller_id_number=XXXX,origination_caller_id_number=XXXX,ignore_e arly_media=true,continue_on_fail=true,hangup_after_bridge=false,originate_ti meout=20}sofia/outbound/YYYY at sip.node4.co.uk"); if (session.getVariable("bridge_hangup_cause") != 'NORMAL_CLEARING') { // Play a sorry we could not connect your call message session.sayPhrase('play', 'sorry.wav'); // Ask if they would like a callback, "press 1 if you would like a callback" session.sayPhrase('play', 'callback.wav'); session.flushDigits(); var callback = session.getDigits(1, '', 5000); if (callback=='1') { //Do some more stuff } } } The problem I have is that if the bridge attempt times out, after playing the callback.wav request message and whilst waiting for input the session receives a random dtmf digit '6' without the caller pressing any keys on the keypad. Can anyone tell me where this digit comes from and how to prevent it from coming in on the original session? Or else tell me that I've got it all completely wrong and tell me how I can achieve the required call handling in a different way? Many thanks Andy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120531/be1fa241/attachment.html From adam.kelloway at newpace.ca Thu May 31 17:50:02 2012 From: adam.kelloway at newpace.ca (Adam Kelloway) Date: Thu, 31 May 2012 10:50:02 -0300 Subject: [Freeswitch-users] Running multiple SIP profiles Message-ID: <4FC7770A.8020807@newpace.ca> Hi there, My understanding is that a SIP profile is single-threaded, and that if you wish to increase the amount of possible concurrent call setups/sessions, you should run more SIP profiles. Am I correct so far? Assuming so, is there a threshold you can reach with the number of profiles you are running whereby it starts to degrade performance? Is there an optimal number of profiles you could run based on the number of processors you have available? Out of curiosity, how many profiles are other people running on your typical high-call load system? Thanks, Adam From hynek.cihlar at gmail.com Thu May 31 17:55:17 2012 From: hynek.cihlar at gmail.com (Hynek Cihlar) Date: Thu, 31 May 2012 15:55:17 +0200 Subject: [Freeswitch-users] Running multiple SIP profiles In-Reply-To: <4FC7770A.8020807@newpace.ca> References: <4FC7770A.8020807@newpace.ca> Message-ID: A subquestion here. What is the thread on the SIP profile responsible for? Is it only dispatching incoming/outgoing SIP requests and spanning new threads for new SIP conversations? Hynek On Thu, May 31, 2012 at 3:50 PM, Adam Kelloway wrote: > Hi there, > > My understanding is that a SIP profile is single-threaded, and that if > you wish to increase the amount of possible concurrent call > setups/sessions, you should run more SIP profiles. Am I correct so far? > Assuming so, is there a threshold you can reach with the number of > profiles you are running whereby it starts to degrade performance? Is > there an optimal number of profiles you could run based on the number of > processors you have available? Out of curiosity, how many profiles are > other people running on your typical high-call load system? > > Thanks, > > Adam > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120531/bbc10647/attachment.html From andrew at cassidywebservices.co.uk Thu May 31 18:09:29 2012 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Thu, 31 May 2012 15:09:29 +0100 Subject: [Freeswitch-users] mod_callcenter making no attempt to contact agents In-Reply-To: References: Message-ID: Further investigation shows this only occurs when if I set that to false then reload, it works fine. On 24 May 2012 16:51, Andrew Cassidy wrote: > Here's some pastebin links from earlier on IRC: > > Cli output and queue config: http://pastebin.freeswitch.org/19154 > Call log output: http://pastebin.freeswitch.org/19156 > > So, here's the thing, queue configured, agents and tiers are in the > database only (eventually going to be shared). Set up an extension, dial > it, get put in queue, hear MoH. > > However, there's no sign of any attempt to contact the agent, even though > it's registered and set as Available. No outgoing sip packets, nothing > logged. I've literally just trid a make current, makes no difference. > > Attached a pcap from server side, showing no invites to the agent. > > Any ideas folks? > > -- > *Andrew Cassidy BSc (Hons) MBCS SSCA* > Managing Director > > > *T *03300 100 960 *F > *03300 100 961 > *E *andrew at cassidywebservices.co.uk > *W *www.cassidywebservices.co.uk > > -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T *03300 100 960 *F *03300 100 961 *E *andrew at cassidywebservices.co.uk *W *www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120531/639c996d/attachment.html From steveayre at gmail.com Thu May 31 18:54:43 2012 From: steveayre at gmail.com (Steven Ayre) Date: Thu, 31 May 2012 15:54:43 +0100 Subject: [Freeswitch-users] Running multiple SIP profiles In-Reply-To: References: <4FC7770A.8020807@newpace.ca> Message-ID: Each sofia profile runs in a single thread (limitation of the sofia stack). This thread is only used for the processing of SIP packets and matching them to a call. Eventually there is a bottleneck if your call volume means the network I/O is faster than a single CPU core can keep up with. Adding more sofia profiles on other ip:ports lets you handle SIP packets on another core which is why it can help in extremely high call loads. At what point that occurs at will highly depend on your traffic and your hardware. Most people won't hit that threshold. Processing of calls (ie dialplan applications etc) and handling of RTP media occurs in their own threads. -Steve On 31 May 2012 14:55, Hynek Cihlar wrote: > A subquestion here. What is the thread on the SIP profile responsible for? > Is it only dispatching incoming/outgoing SIP requests and spanning new > threads for new SIP conversations? > > Hynek > > > > > On Thu, May 31, 2012 at 3:50 PM, Adam Kelloway > wrote: >> >> Hi there, >> >> My understanding is that a SIP profile is single-threaded, and that if >> you wish to increase the amount of possible concurrent call >> setups/sessions, you should run more SIP profiles. Am I correct so far? >> Assuming so, is there a threshold you can reach with the number of >> profiles you are running whereby it starts to degrade performance? Is >> there an optimal number of profiles you could run based on the number of >> processors you have available? Out of curiosity, how many profiles are >> other people running on your typical high-call load system? >> >> Thanks, >> >> Adam >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From a.villa at seletech.com Thu May 31 11:38:01 2012 From: a.villa at seletech.com (alberto Villa) Date: Thu, 31 May 2012 09:38:01 +0200 Subject: [Freeswitch-users] Custom Registration Message-ID: <4FC71FD9.5010004@seletech.com> Hello, I'm using FreeSWITCH in a very custom system and I need to setup this kind of registration: There are two SIP phones A and B: when each SIP phone have its own unique ID the registration to freeswitch goes as usual and no change is needed, but when A and B share the same ID I would like to reject the registration for the last SIP phone. So, if A and B have the same ID and A is registered first, I would like to reject the subsequent registration of B. I tried to change the values of some configuration parameters such as "sip-allow-multiple-registrations" and "max-registrations-per-extension" but it didn't work (by the way I'm not shure to have edited properly this parameters). How can I achive this kind of registration? Thank you Alberto -- Dr. Villa Alberto Sw Engineer SeleTech srl via Collodi, 8 20052 Monza (MI) tel: +39 039 5962000 fax: +39 039 9716905 email: a.villa at seletech.com web: www.seletech.com www.seletech.eu From krice at freeswitch.org Thu May 31 19:25:50 2012 From: krice at freeswitch.org (Ken Rice) Date: Thu, 31 May 2012 10:25:50 -0500 Subject: [Freeswitch-users] Running multiple SIP profiles In-Reply-To: Message-ID: This is not entirely true... While each profile is its own thread true, the SOA state engine is a single thread, and this is where most bottlenecks occur... Now that being said Tony has implemented some things in sofia to mitigate this issue... Tony demonstrated this update to the code at Cluecon (2011) with a machine with 1 sofia profile running 1000 calls/sec at 30second call duration leaving the box with 30K concurrent calls. Admittedly this was a pretty beefy box, but still, the added management overhead of the addition profiles (not just the freeswitch configs but other system admin required) may or may not be worth any perceived performance gains. Also if you are trying to do over 1000cps on a single machine, you need to re-evaluate what you are doing, as at that rate you should be able to afford additional hardware... That's just to much business grade traffic on 1 machine to risk a failure and loss of revenue K On 5/31/12 9:54 AM, "Steven Ayre" wrote: > Each sofia profile runs in a single thread (limitation of the sofia > stack). This thread is only used for the processing of SIP packets and > matching them to a call. Eventually there is a bottleneck if your call > volume means the network I/O is faster than a single CPU core can keep > up with. Adding more sofia profiles on other ip:ports lets you handle > SIP packets on another core which is why it can help in extremely high > call loads. At what point that occurs at will highly depend on your > traffic and your hardware. Most people won't hit that threshold. > > Processing of calls (ie dialplan applications etc) and handling of RTP > media occurs in their own threads. > > -Steve > > > > On 31 May 2012 14:55, Hynek Cihlar wrote: >> A subquestion here. What is the thread on the SIP profile responsible for? >> Is it only dispatching incoming/outgoing SIP requests and spanning new >> threads for new SIP conversations? >> >> Hynek >> >> >> >> >> On Thu, May 31, 2012 at 3:50 PM, Adam Kelloway >> wrote: >>> >>> Hi there, >>> >>> My understanding is that a SIP profile is single-threaded, and that if >>> you wish to increase the amount of possible concurrent call >>> setups/sessions, you should run more SIP profiles. Am I correct so far? >>> Assuming so, is there a threshold you can reach with the number of >>> profiles you are running whereby it starts to degrade performance? Is >>> there an optimal number of profiles you could run based on the number of >>> processors you have available? Out of curiosity, how many profiles are >>> other people running on your typical high-call load system? >>> >>> Thanks, >>> >>> Adam >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> Join Us At ClueCon - Aug 7-9, 2012 >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From jgallartm at gmail.com Thu May 31 19:51:03 2012 From: jgallartm at gmail.com (Javier Gallart) Date: Thu, 31 May 2012 17:51:03 +0200 Subject: [Freeswitch-users] Access to Cause filed in Reason header Message-ID: Hello list some of our providers include a Reason header in negative replies to an INVITE. For instance, they might include a cause=65 inside a Sip 503 reply. We need to react upon some of those ISDN causes: in this particular case with if Reason header were absent we would simply do nothing but let FS to send the response upstream; but if cause=65 in present we want to resend the call to the same provider with a different set of offered codecs. Is there any way to access to the content of that header?. Our applicatiion listens to the event socket in inbound mode but I haven't found a way to do this. Any help would be appreciated. Regards Javi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120531/e1ec3d40/attachment.html From johnrose at google.hm Thu May 31 20:00:10 2012 From: johnrose at google.hm (John Rose) Date: Thu, 31 May 2012 12:00:10 -0400 Subject: [Freeswitch-users] mod_sms MESSAGE request-uri Message-ID: <005a01cd3f46$75793230$606b9690$@google.hm> I'm trying to perform a SendEvent SMS::MESSAGE via ESL and have the MESSAGE go to a different URI than the "to" or "full_to" header that is set using AddHeader on the ESL event. Is it possible to set maybe a "sip_message_req_uri" akin to a "sip_invite_req_uri" variable or something similar for each MESSAGE? Looking at the FS source code it seems that "to" or "full_to" is always used. Thanks, John -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120531/3e0de537/attachment.html From andrew at cassidywebservices.co.uk Thu May 31 20:20:15 2012 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Thu, 31 May 2012 17:20:15 +0100 Subject: [Freeswitch-users] mod_callcenter making no attempt to contact agents In-Reply-To: References: Message-ID: Got it, just FYI all, If you have noone at tier 0, tier-rules-apply=true, tier-rule-no-agent-no-wait=false and a tier timeout, it'll wait for the tier timeout to pass before ringing up to tier 1. As the documentation only specifies calls at tier 1, I thought this was the first tier. I'll update the wiki accordingly. On 31 May 2012 15:09, Andrew Cassidy wrote: > Further investigation shows this only occurs when name="tier-rules-apply" value="true" /> > > if I set that to false then reload, it works fine. > > > On 24 May 2012 16:51, Andrew Cassidy wrote: > >> Here's some pastebin links from earlier on IRC: >> >> Cli output and queue config: http://pastebin.freeswitch.org/19154 >> Call log output: http://pastebin.freeswitch.org/19156 >> >> So, here's the thing, queue configured, agents and tiers are in the >> database only (eventually going to be shared). Set up an extension, dial >> it, get put in queue, hear MoH. >> >> However, there's no sign of any attempt to contact the agent, even though >> it's registered and set as Available. No outgoing sip packets, nothing >> logged. I've literally just trid a make current, makes no difference. >> >> Attached a pcap from server side, showing no invites to the agent. >> >> Any ideas folks? >> >> -- >> *Andrew Cassidy BSc (Hons) MBCS SSCA* >> Managing Director >> >> >> *T *03300 100 960 *F >> *03300 100 961 >> *E *andrew at cassidywebservices.co.uk >> *W *www.cassidywebservices.co.uk >> >> > > > -- > *Andrew Cassidy BSc (Hons) MBCS SSCA* > Managing Director > > > *T *03300 100 960 *F > *03300 100 961 > *E *andrew at cassidywebservices.co.uk > *W *www.cassidywebservices.co.uk > > -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T *03300 100 960 *F *03300 100 961 *E *andrew at cassidywebservices.co.uk *W *www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120531/fc103676/attachment-0001.html From msc at freeswitch.org Thu May 31 20:38:15 2012 From: msc at freeswitch.org (Michael Collins) Date: Thu, 31 May 2012 09:38:15 -0700 Subject: [Freeswitch-users] Transfer calls from Server A Freeswitch box to Server B Freeswitch box. In-Reply-To: References: Message-ID: Have you confirmed that the sip_history_info parameter is even defined at the time bridge the call? Add this line line right before the bridge and make a test call: Check the console for an orange colored line with the sip_history_info. On a different note: what information in the sip_history_info are you needing? In any case, you might be able to set an X header in server A that you can then collect in server B. I recommend checking this section of the mod_sofia page: http://wiki.freeswitch.org/wiki/Mod_sofia#Channel_Variables HTH, MC On Wed, May 30, 2012 at 9:04 PM, Sanath Prasanna wrote: > Hi, > I transfer some calls from Server A Freeswitch box to Server B Freeswitch > box. I need to get value of variable_sip_history_info parameter in Server > B. It is coming to Server A. But not coming to Server B. Also I play few > announcement from Server B. It is not hear to caller even though it is play > from server B box. > Pls help to sort out above 2 problems. > > Server A ip: 10.1.1.252 > Server B ip: 10.1.1.253 > > Here is the dial plan of server A. > > > > > > > > > Here is the acl.conf.xml > server B > > > > > Server A: > > > > > > Br, > Sanath > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120531/5e27b71d/attachment.html From modesto at isimples.com.br Thu May 31 21:10:50 2012 From: modesto at isimples.com.br (Antonio Modesto) Date: Thu, 31 May 2012 14:10:50 -0300 Subject: [Freeswitch-users] FreeTDM CallerID Detection with DTMF In-Reply-To: <1338403799.3112.43.camel@modesto.localdomain.net> References: <1338403799.3112.43.camel@modesto.localdomain.net> Message-ID: <1338484250.23766.17.camel@modesto.localdomain.net> Hi, Does anybody have any suggestions? I searched a lot about it but I was not successful, I found that here in brazil the callerid is sent via DTMF but without any warning or polarity reversal, it's a strange implementation. Any hints would be appreciated. Regards. On Wed, 2012-05-30 at 15:49 -0300, Antonio Modesto wrote: > Hi, > > I have a Digium TDM410P card, I am using it here in Brazil, where the > signaling is not FSK, it's DTMF. It is working, though I am not > receiving the callerid: > > Caller-Caller-ID-Number: [0000000000] > > Here is my freetdm.conf: > > [span zt FXO1] > fxo-channel => 1 > > [span zt FXO2] > fxo-channel => 2 > > [span zt FXO3] > fxo-channel => 3 > > Here is one section of my autoload_configs/freetdm.conf.xml: > > > > > > > > > > > > > > > > > > Is it possible to enable the callerid detection in these conditions, or > is it a hardware/driver limitation? > > > Regards. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From jrichey at itltd.net Thu May 31 21:16:09 2012 From: jrichey at itltd.net (JRichey) Date: Thu, 31 May 2012 10:16:09 -0700 Subject: [Freeswitch-users] DTMF Passthrough on different legs Message-ID: <6ECAF1527329364583AB525CF34ABF950B31A67C@ms.kallback.com> What CODECs are being used? In-band DTMF won't work correctly with compressed CODECs like G.729. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org]On Behalf Of Michael Lutz Sent: Thursday, May 31, 2012 2:12 AM To: FreeSWITCH Users Help Subject: [Freeswitch-users] DTMF Passthrough on different legs Hi Guys, I am stuck with a bit of a problem I cannot figure out where it is going wrong.. Here's my situation, I have an inbound call, comming from an external provider, they don't support RFC2833 so I turn on inband detection using 'start_dtmf' in the dialplan. This works ok, for the incomming channel and IVR provided (via Lua) to the caller. (Though sometimes it seems to misinterpret and returns wrong keys..) (Session:A) After that, this call is bridged (via Lua to a a new inbound session on the same switch) (without specifying start_dtmf) so RC2388 is used by default (?) (Session:B) This new sessions is playing soem audio (Lua) and is heard by the original caller (the two sessions are bridged). Seperatly I have an inbound ESL connection where I do an origanate using a originate {execute_on_answer=start_dtmf, .... etc) &park() to setup an new external call via the same gateway (they don't support RFC2388 so I activate the inband detection on the b-leg to receive the digits. Let's call this leg Session:C. After the orignated call is answered I bridge the parked call with the previous session (Session:B), so at the end Session:A and Session:B can talk to each other. Al this works perfectly. [Incommming call] Session:A ----- INBAND ------> Bridge(FS.Lua) ---> Session:B ---- RFC2388 ----> FS.Bridge ---> Session:C --- INBAND ------> ESL destination number My symptoms are... 1. I have a customer who had a IVR on his own side, (Session:C) and requires input from Session:A. The weird thing is that it seems the input is wrong. Keys entered do not correspond to what his IVR is receiving. As if newly dtmfs are generated somewhere. When I switch to another inbound provider, making Session:A using rfc2388 and not inband, but do dialout via the provider using inband detection. It seems to work fine. 2. Not being able to get the dtmf's from the ESL Destination number (Session:C). I'm reading this by subscribing to events via ESL for that particulair uuid. When I have a connection (inbound) without inband detection it seems to get the DTMF's from Session:C without any problems. Which makes it weird for me. note: I know the setup is a bit odd, but Session A cannot exit it's Lua script, so I needed to do a workaround to be able to keep te lua alive and being able to control the call (audio) via ESL. I realy need some help on this one, becuase we have a lot of customers complaining about not responding to dtmf's or receving invalid input in the (FS) IVR. Thanks for you help, Regards, Mike. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From sdevoy at bizfocused.com Thu May 31 21:50:06 2012 From: sdevoy at bizfocused.com (Sean Devoy) Date: Thu, 31 May 2012 13:50:06 -0400 Subject: [Freeswitch-users] SCA not working inbound - Multi Domain In-Reply-To: References: <040301cd3aa3$e93705f0$bba511d0$@bizfocused.com> <051501cd3ac2$8a187ee0$9e497ca0$@bizfocused.com> <075e01cd3dbe$06db5350$1491f9f0$@bizfocused.com> <085901cd3dd8$5157c7f0$f40757d0$@bizfocused.com> Message-ID: <032c01cd3f55$d0826930$71873b90$@bizfocused.com> I will Michael. Unfortunately, I cannot get it to work through their gateways/routers. I am opening a new thread for NAT issues and will followup with the working config when we have it. From: Michael Collins [mailto:msc at freeswitch.org] Sent: Tuesday, May 29, 2012 4:30 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SCA not working inbound - Multi Domain Sean, Thanks for taking the time to follow up. If you get it working we'd love to have a snapshot of your configuration to put up on the wiki. Thanks, MC On Tue, May 29, 2012 at 1:19 PM, Sean Devoy wrote: Thanks Anthony. First I should have pointed out that the 2 test phones are on the local Lan with the switch, NO NAT. The final task is to move them out with NAT. Second, it is working!! After I dumped everything for you I noticed one of the phones was connecting on the wrong port (sofia external, not lan). I changed it last week but it did not seem to work. Now having waited (and registrations having renewed) it is working!! Thank you for your help. Now, out to the site and their crappy router and to see if this works on through NAT. Sean -----Original Message----- From: Anthony Minessale [mailto:anthony.minessale at gmail.com] Sent: Tuesday, May 29, 2012 3:18 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SCA not working inbound - Multi Domain enter "sofia_contact 220 at mydomainname.com" at your cli and see if you get a large dial string. if you are only getting calls to one phone make sure they all work, its possible you have many phones behind the same nat and only one of them can poke through at a time. On Tue, May 29, 2012 at 12:11 PM, Sean Devoy wrote: > BUMP! > > Anyone have any ideas for me? > > Any other information I can provide? > > Thanks. > Sean > > -----Original Message----- > From: Sean Devoy [mailto:sdevoy at bizfocused.com] > Sent: Friday, May 25, 2012 6:06 PM > To: 'FreeSWITCH Users Help' > Subject: Re: [Freeswitch-users] SCA not working inbound - Multi Domain > > Here is the result of select * from sip_subscriptions;" > sip|220|fs_lan.bizfocused.com|220|fs_lan.bizfocused.com||call-info|"user" > |38e107ab-6cde6635 at 10.10.40.30|"220" > >;tag=fd508933c5f5924d|SIP/2.0/UDP > 10.10.40.30:5060;branch=z9hG4bK-f5598bfb|1337980638|Cisco/SPA504G-7.4. > 8a||ex > ternal|FreeSwitch1.sumdomain.com|5060|10.10.40.30|-1||"220" > >;tag=VrGrXQaOH22R > sip|220|fs_lan.bizfocused.com|220|fs_lan.bizfocused.com||call-info|"user" > |c08f0c6a-c46e90d2 at 10.10.40.20|"Sean" > >;tag=f22a978ae8838032|SIP/2.0/UDP > 10.10.40.20:5064;branch=z9hG4bK-905b6faa|1337980643|Cisco/SPA504G-7.4. > 9c||ex > ternal|FreeSwitch1.sumdomain.com|5064|10.10.40.20|-1||"Sean" > >;tag=y5VtigPlIghD > > But it was in: > sofia_reg_external.db not internal. > > I have sorted out all the sip trace data into 2 txt files for the 2 > phones involved. They are zipped up at: > http://www.bizfocused.com/sip_trace.zip > > Thank you again for your help. I am way over my head now. > > > -----Original Message----- > From: Anthony Minessale [mailto:anthony.minessale at gmail.com] > Sent: Friday, May 25, 2012 2:35 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] SCA not working inbound - Multi Domain > > What are the phones putting in the subscribe ? > > sofia global siptrace on > sofia global debug presence|sla > > then watch for SUBSCRIBE > > also when you are not using odbc you can get the sql with this app > > sqlute3 /usr/local/freeswitch/db/sofia_reg_internal.db > > also try "select * from sip_subscriptions" > > its all about using the right host name across the board, IP's count > as hostnames, they do not magically resolve any dns with SIP > > > > > On Fri, May 25, 2012 at 1:26 PM, Sean Devoy wrote: >> Hi all, >> >> >> >> I have a muti-tennnant configuration that is working nicely except >> for Shared Call Appearance. The desktop devices are CISCO 504Gs and >> they are configured as described in the FS Wiki as well as Cisco Documentation. >> >> >> >> The SCA works perfectly for outbound calls - if either phone pickups >> like 220, the other phones indicator light flashes red. However, >> inbound calls will go to only one of the phones (which one has >> changed a few times) and the other phones line still just stays green >> and does not > ring. >> >> >> >> Here is the sip interfaces config: >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> > value="true"/> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> The directory entry which both phones connect using: >> >> >> >> >> >> >> >> > value="410420BLEEP"/> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> " >> >> >> >> > value="{presence_id=220 at mydomainname.com}${sofia_contact(220 at mydomain >> n >> ame.com)}"/> >> >> >> >> >> >> >> >> >> >> And the dial plan for ext 220: >> >> >> >> >> >> > data="effective_caller_id_number=${internal_caller_id_number}"/> >> >> > data="effective_caller_id_name=${internal_caller_id_name}"/> >> >> >> >> >> >> >> >> > data="{sip_invite_domain=mydomainname.com}user/220 at mydomainname.com" >> /> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> I did see this in the wiki >> (http://wiki.freeswitch.org/wiki/Shared_Line_Appearance): >> >> If SLA works for outgoing calls and SLA does not work for inbound >> calls to the SLA phones, you may have some presence problem related >> to mixed IP and domain names. When using ODBC you may issue the >> following SQL statement >> >> select >> sip_to_host,sip_from_user,sip_from_host,hostname,presence_id,call_inf >> o >> from sip_dialogs; >> >> But I don't have ODBC on this server, so I am a little lost. >> >> >> >> I have the phones login to domain names, not addresses. I never >> refer to IP addresses in my xml (except gateways addresses). I am >> not trying SLA across domain, only within the same domain. >> >> >> >> I hope someone can spot something. Thanks for your help. >> >> >> >> Sean >> >> >> _____________________________________________________________________ >> _ ___ Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us >> e >> rs >> http://www.freeswitch.org >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > > > > > > > > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120531/b45ff9f0/attachment-0001.html From msc at freeswitch.org Thu May 31 21:53:21 2012 From: msc at freeswitch.org (Michael Collins) Date: Thu, 31 May 2012 10:53:21 -0700 Subject: [Freeswitch-users] SCA not working inbound - Multi Domain In-Reply-To: <032c01cd3f55$d0826930$71873b90$@bizfocused.com> References: <040301cd3aa3$e93705f0$bba511d0$@bizfocused.com> <051501cd3ac2$8a187ee0$9e497ca0$@bizfocused.com> <075e01cd3dbe$06db5350$1491f9f0$@bizfocused.com> <085901cd3dd8$5157c7f0$f40757d0$@bizfocused.com> <032c01cd3f55$d0826930$71873b90$@bizfocused.com> Message-ID: Sadface... Hopefully you get it ironed out. -MC On Thu, May 31, 2012 at 10:50 AM, Sean Devoy wrote: > I will Michael. Unfortunately, I cannot get it to work through their > gateways/routers.**** > > ** ** > > I am opening a new thread for NAT issues and will followup with the > working config when we have it.**** > > ** ** > > *From:* Michael Collins [mailto:msc at freeswitch.org] > *Sent:* Tuesday, May 29, 2012 4:30 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] SCA not working inbound - Multi Domain** > ** > > ** ** > > Sean, > > Thanks for taking the time to follow up. If you get it working we'd love > to have a snapshot of your configuration to put up on the wiki. > > Thanks, > MC**** > > On Tue, May 29, 2012 at 1:19 PM, Sean Devoy wrote: > **** > > Thanks Anthony. > > First I should have pointed out that the 2 test phones are on the local Lan > with the switch, NO NAT. The final task is to move them out with NAT. > > Second, it is working!! > > After I dumped everything for you I noticed one of the phones was > connecting > on the wrong port (sofia external, not lan). I changed it last week but it > did not seem to work. Now having waited (and registrations having renewed) > it is working!! > > Thank you for your help. > > Now, out to the site and their crappy router and to see if this works on > through NAT. > > Sean > -----Original Message----- > From: Anthony Minessale [mailto:anthony.minessale at gmail.com] > Sent: Tuesday, May 29, 2012 3:18 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] SCA not working inbound - Multi Domain > > enter "sofia_contact 220 at mydomainname.com" at your cli and see if you get > a > large dial string. > if you are only getting calls to one phone make sure they all work, its > possible you have many phones behind the same nat and only one of them can > poke through at a time. > > > > On Tue, May 29, 2012 at 12:11 PM, Sean Devoy > wrote: > > BUMP! > > > > Anyone have any ideas for me? > > > > Any other information I can provide? > > > > Thanks. > > Sean > > > > -----Original Message----- > > From: Sean Devoy [mailto:sdevoy at bizfocused.com] > > Sent: Friday, May 25, 2012 6:06 PM > > To: 'FreeSWITCH Users Help' > > Subject: Re: [Freeswitch-users] SCA not working inbound - Multi Domain > > > > Here is the result of select * from sip_subscriptions;" > > sip|220|fs_lan.bizfocused.com|220|fs_lan.bizfocused.com > ||call-info|"user" > > |38e107ab-6cde6635 at 10.10.40.30|"220" > > ;tag=fd508933c5f5924d|SIP/2.0/UDP > > 10.10.40.30:5060;branch=z9hG4bK-f5598bfb|1337980638|Cisco/SPA504G-7.4. > > 8a||ex > > ternal|FreeSwitch1.sumdomain.com|5060|10.10.40.30|-1||"220" > > ;tag=VrGrXQaOH22R > > sip|220|fs_lan.bizfocused.com|220|fs_lan.bizfocused.com > ||call-info|"user" > > |c08f0c6a-c46e90d2 at 10.10.40.20|"Sean" > > ;tag=f22a978ae8838032|SIP/2.0/UDP > > 10.10.40.20:5064;branch=z9hG4bK-905b6faa|1337980643|Cisco/SPA504G-7.4. > > 9c||ex > > ternal|FreeSwitch1.sumdomain.com|5064|10.10.40.20|-1||"Sean" > > ;tag=y5VtigPlIghD > > > > But it was in: > > sofia_reg_external.db not internal. > > > > I have sorted out all the sip trace data into 2 txt files for the 2 > > phones involved. They are zipped up at: > > http://www.bizfocused.com/sip_trace.zip > > > > Thank you again for your help. I am way over my head now. > > > > > > -----Original Message----- > > From: Anthony Minessale [mailto:anthony.minessale at gmail.com] > > Sent: Friday, May 25, 2012 2:35 PM > > To: FreeSWITCH Users Help > > Subject: Re: [Freeswitch-users] SCA not working inbound - Multi Domain > > > > What are the phones putting in the subscribe ? > > > > sofia global siptrace on > > sofia global debug presence|sla > > > > then watch for SUBSCRIBE > > > > also when you are not using odbc you can get the sql with this app > > > > sqlute3 /usr/local/freeswitch/db/sofia_reg_internal.db > > > > also try "select * from sip_subscriptions" > > > > its all about using the right host name across the board, IP's count > > as hostnames, they do not magically resolve any dns with SIP > > > > > > > > > > On Fri, May 25, 2012 at 1:26 PM, Sean Devoy > wrote: > >> Hi all, > >> > >> > >> > >> I have a muti-tennnant configuration that is working nicely except > >> for Shared Call Appearance. The desktop devices are CISCO 504Gs and > >> they are configured as described in the FS Wiki as well as Cisco > Documentation. > >> > >> > >> > >> The SCA works perfectly for outbound calls ? if either phone pickups > >> like 220, the other phones indicator light flashes red. However, > >> inbound calls will go to only one of the phones (which one has > >> changed a few times) and the other phones line still just stays green > >> and does not > > ring. > >> > >> > >> > >> Here is the sip interfaces config: > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> >> value="true"/> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> The directory entry which both phones connect using: > >> > >> > >> > >> > >> > >> > >> > >> >> value="410420BLEEP"/> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> " > >> > >> > >> > >> >> value="{presence_id=220 at mydomainname.com}${sofia_contact(220 at mydomain > >> n > >> ame.com)}"/> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> And the dial plan for ext 220: > >> > >> > >> > >> > >> > >> >> data="effective_caller_id_number=${internal_caller_id_number}"/> > >> > >> >> data="effective_caller_id_name=${internal_caller_id_name}"/> > >> > >> > >> > >> > >> > >> > >> > >> >> data="{sip_invite_domain=mydomainname.com}user/220 at mydomainname.com" > >> /> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> I did see this in the wiki > >> (http://wiki.freeswitch.org/wiki/Shared_Line_Appearance): > >> > >> If SLA works for outgoing calls and SLA does not work for inbound > >> calls to the SLA phones, you may have some presence problem related > >> to mixed IP and domain names. When using ODBC you may issue the > >> following SQL statement > >> > >> select > >> sip_to_host,sip_from_user,sip_from_host,hostname,presence_id,call_inf > >> o > >> from sip_dialogs; > >> > >> But I don?t have ODBC on this server, so I am a little lost. > >> > >> > >> > >> I have the phones login to domain names, not addresses. I never > >> refer to IP addresses in my xml (except gateways addresses). I am > >> not trying SLA across domain, only within the same domain. > >> > >> > >> > >> I hope someone can spot something. Thanks for your help. > >> > >> > >> > >> Sean > >> > >> > >> _____________________________________________________________________ > >> _ ___ Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> Join Us At ClueCon - Aug 7-9, 2012 > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us > >> e > >> rs > >> http://www.freeswitch.org > >> > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > googletalk:conf+888 at conference.freeswitch.org > > pstn:+19193869900 > > > > > > > > > > > > > > > > > > > > ______________________________________________________________________ > > ___ Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > > rs > > http://www.freeswitch.org > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120531/9f14c339/attachment-0001.html From msc at freeswitch.org Thu May 31 21:54:12 2012 From: msc at freeswitch.org (Michael Collins) Date: Thu, 31 May 2012 10:54:12 -0700 Subject: [Freeswitch-users] Chat Event in Dingaling In-Reply-To: References: Message-ID: I don't believe events are fired on chat messages - I think they just get routed to the destination. Can anyone verify this? -MC On Thu, May 31, 2012 at 12:15 AM, Sameer Khan wrote: > Is this implemented? > > There is no event fired when a chat message is received > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120531/dff531de/attachment.html From Hector.Geraldino at ipsoft.com Thu May 31 22:00:36 2012 From: Hector.Geraldino at ipsoft.com (Hector Geraldino) Date: Thu, 31 May 2012 14:00:36 -0400 Subject: [Freeswitch-users] uuid_record and recording output format In-Reply-To: References: <6A6B4C284AD15042B429EB9D904544AD022EE80BB4@NY1-EXMB-01.ip-soft.net> Message-ID: <6A6B4C284AD15042B429EB9D904544AD022EE80CAB@NY1-EXMB-01.ip-soft.net> Thanks Peter, it's much more clear now. By default, when recording a session to a wav file, I can see in the audio file properties that: the sample rate is 8000 Hz (8kz as mentioned), and the codec says "Uncompressed 16-bit PCM audio". By default, is it ulaw or alaw? Sorry for the confusion, I'm trying to catch up with all this information about codec negotiation, codecs and audio formats but it's a little bit overwhelming for a novice like me. Thanks again -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Peter Olsson Sent: Thursday, May 31, 2012 2:10 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] uuid_record and recording output format The sample rate is the rate, so 8000 is the same as 8khz, it's got nothing to do with 8 or 16 bit. FreeSWITCH always converts to L16 (linear 16-bit), so that's why you get this result. You can force to record to a raw file, for instance, try recording to testfile.PCMA to record in G7.11 alaw format. /Peter 30 maj 2012 kl. 16:37 skrev "Hector Geraldino" >: Greetings, I'm using a 3rd party application (ndev dragonmobile) to get the transcription of some audio recorded by FreeSWITCH. Think about it as a voicemail transcription service. The problem I'm facing is that, when I record a session using uuid_record, the output file is encoded in PCM 16-bit @ 8khz. Correct me if I'm wrong, but my understanding is that if I want to capture audio from calls coming from the PSTN (analog/landlines), the best I can do is to record it in 8-bits (using G.711). I don't want to use sox (or any other tool) to resample the output file, and what I've tried so far is setting the sample_rate variable on the diaplan as recommended on the wiki: http://wiki.freeswitch.org/wiki/Variable_record_rate This doesn't have any effect on the generated wav file, which is still encoded in 16-bits. So my question is: does this variable affects the behavior of the uuid_record command? Or, do I really need to encode the audio output in 8-bits when the origin of the call comes from the PSTN? How is FreeSWITCH encoding the audio in 16-bits if, in theory, the best rate we can get from an analog line is 8-bits? Sorry if I'm misunderstanding something, but I'm not a telephony/voip guy, more like a java developer. Thanks for your help. !DSPAM:4fc62d7332761385138176! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4fc62d7332761385138176! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From sdevoy at bizfocused.com Thu May 31 22:10:59 2012 From: sdevoy at bizfocused.com (Sean Devoy) Date: Thu, 31 May 2012 14:10:59 -0400 Subject: [Freeswitch-users] NAT issues and "best practices" Message-ID: <036901cd3f58$bb857510$32905f30$@bizfocused.com> HI All, I have a customer location that has just been a nightmare to implement. I am just learning that they "may" have multiple NAT routers in sequence at their location. I think I fully understand what NAT is trying to accomplish. There seem to be different levels and approaches. The most basic NAT setup (to me) is a HOME LAN with multiple PCs where NAT allows multiple devices to share a single routable IP address on the WAN side from multiple local devices the LAN side. Note I said OUTBOUND initiated connections. Even FTP can have trouble with this level. Almost all inbound traffic is blocked for security. Clearly for FS we need the switch to be able to punch through from the WAN to specific local IPs on the LAN to reach specific phones. This is INBOUND NAT and brings up many security issues for people. Even on devices where you get this "working" you may only be able to support one line per phone or a single inbound connection at a time. I understand NAT has PMP and UPnP protocols and FS "supports" both. What I can't find is where someone says "Here is a great setup that works with cheap, available "commodity" hardware from Cisco/Linksys that supports all the NAT you need for FS." I don't care if it is PMP or UPnP and I might not even care why you pick one over the other, although it is probably a "good read". Can someone just stand up say "FS works GREAT with the XYZ router in ABC mode from MY COMPANY using NAT to Cisco phones"? I have seen some articles about Freeware/Shareware firmware in this devices, but as a novice I want to limit the unknowns until I get more up to speed. My specific issue now is that I cannot get SCA to work at the NAT location. I issued: sofia_contact 220 at mydomain.com sofia/external/sip:200@:44234,sofia/external/sip:200@:1024 Only one phone rings on inbound and the line indicator light does not change when either is picked up. Same configuration is working on our LAN with the switch. I am absolutely ready to by a router to fix these issues, I don't want to lose this customer. Thanks for your thoughts, Sean -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120531/ccc242b0/attachment.html From msc at freeswitch.org Thu May 31 22:16:46 2012 From: msc at freeswitch.org (Michael Collins) Date: Thu, 31 May 2012 11:16:46 -0700 Subject: [Freeswitch-users] Access to Cause filed in Reason header In-Reply-To: References: Message-ID: I believe that information is available in other channel variables. I'd do an info dump right after the bridge and see what shakes out. -MC On Thu, May 31, 2012 at 8:51 AM, Javier Gallart wrote: > Hello list > > some of our providers include a Reason header in negative replies to an > INVITE. For instance, they might include a cause=65 inside a Sip 503 reply. > We need to react upon some of those ISDN causes: in this particular case > with if Reason header were absent we would simply do nothing but let FS to > send the response upstream; but if cause=65 in present we want to resend > the call to the same provider with a different set of offered codecs. Is > there any way to access to the content of that header?. Our applicatiion > listens to the event socket in inbound mode but I haven't found a way to do > this. Any help would be appreciated. > > Regards > > Javi > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120531/887481fe/attachment.html From msc at freeswitch.org Thu May 31 22:19:52 2012 From: msc at freeswitch.org (Michael Collins) Date: Thu, 31 May 2012 11:19:52 -0700 Subject: [Freeswitch-users] uuid_record and recording output format In-Reply-To: <6A6B4C284AD15042B429EB9D904544AD022EE80CAB@NY1-EXMB-01.ip-soft.net> References: <6A6B4C284AD15042B429EB9D904544AD022EE80BB4@NY1-EXMB-01.ip-soft.net> <6A6B4C284AD15042B429EB9D904544AD022EE80CAB@NY1-EXMB-01.ip-soft.net> Message-ID: 16-bit PCM audio is just "raw" audio, sometimes you'll see "SLIN" or "signed linear" or some other designations. Whenever you see that just know that FS is dealing with raw, unencoded audio. Therefore it is neither alaw nor ulaw. You *can* record to a file with specific encoding, but that's usually not needed. 99% of the time just a simple wav file with PCM audio is perfectly sufficient. -MC On Thu, May 31, 2012 at 11:00 AM, Hector Geraldino < Hector.Geraldino at ipsoft.com> wrote: > Thanks Peter, it's much more clear now. > > By default, when recording a session to a wav file, I can see in the audio > file properties that: the sample rate is 8000 Hz (8kz as mentioned), and > the codec says "Uncompressed 16-bit PCM audio". By default, is it ulaw or > alaw? > > Sorry for the confusion, I'm trying to catch up with all this information > about codec negotiation, codecs and audio formats but it's a little bit > overwhelming for a novice like me. > > Thanks again > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Peter Olsson > Sent: Thursday, May 31, 2012 2:10 AM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] uuid_record and recording output format > > The sample rate is the rate, so 8000 is the same as 8khz, it's got nothing > to do with 8 or 16 bit. > > FreeSWITCH always converts to L16 (linear 16-bit), so that's why you get > this result. You can force to record to a raw file, for instance, try > recording to testfile.PCMA to record in G7.11 alaw format. > > /Peter > > 30 maj 2012 kl. 16:37 skrev "Hector Geraldino" < > Hector.Geraldino at ipsoft.com>: > > Greetings, > > I'm using a 3rd party application (ndev dragonmobile) to get the > transcription of some audio recorded by FreeSWITCH. Think about it as a > voicemail transcription service. The problem I'm facing is that, when I > record a session using uuid_record, the output file is encoded in PCM > 16-bit @ 8khz. Correct me if I'm wrong, but my understanding is that if I > want to capture audio from calls coming from the PSTN (analog/landlines), > the best I can do is to record it in 8-bits (using G.711). > > I don't want to use sox (or any other tool) to resample the output file, > and what I've tried so far is setting the sample_rate variable on the > diaplan as recommended on the wiki: > http://wiki.freeswitch.org/wiki/Variable_record_rate > > > > This doesn't have any effect on the generated wav file, which is still > encoded in 16-bits. So my question is: does this variable affects the > behavior of the uuid_record command? Or, do I really need to encode the > audio output in 8-bits when the origin of the call comes from the PSTN? How > is FreeSWITCH encoding the audio in 16-bits if, in theory, the best rate we > can get from an analog line is 8-bits? > > Sorry if I'm misunderstanding something, but I'm not a telephony/voip guy, > more like a java developer. > > Thanks for your help. > > > !DSPAM:4fc62d7332761385138176! > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120531/6356b183/attachment-0001.html From errotan at elder.hu Thu May 31 22:42:08 2012 From: errotan at elder.hu (=?ISO-8859-1?Q?Pusk=E1s_Zsolt?=) Date: Thu, 31 May 2012 20:42:08 +0200 Subject: [Freeswitch-users] Custom Registration In-Reply-To: <4FC71FD9.5010004@seletech.com> References: <4FC71FD9.5010004@seletech.com> Message-ID: <4FC7BB80.9050600@elder.hu> Hi. Even if you wrote an ESL script to watch registration events you can't refuse or miss a registration so you have to modify the sofia source code to achive such a function as far as I know. Are you using IP address based authentication for phones and afraid the user might change the username or why you want such a function ? 2012-05-31 09:38 keltez?ssel, alberto Villa ?rta: > Hello, > > I'm using FreeSWITCH in a very custom system and I need to setup this > kind of registration: > > There are two SIP phones A and B: when each SIP phone have its own > unique ID the registration to freeswitch goes as usual and no change is > needed, but when A and B share the same ID I would like to reject the > registration for the last SIP phone. So, if A and B have the same ID and > A is registered first, I would like to reject the subsequent > registration of B. > > I tried to change the values of some configuration parameters such as > "sip-allow-multiple-registrations" and "max-registrations-per-extension" > but it didn't work (by the way I'm not shure to have edited properly > this parameters). > > How can I achive this kind of registration? Thank you > > Alberto > From sameer2k3t at gmail.com Thu May 31 23:13:44 2012 From: sameer2k3t at gmail.com (Sameer Khan) Date: Thu, 31 May 2012 23:13:44 +0400 Subject: [Freeswitch-users] Chat Event in Dingaling In-Reply-To: References: Message-ID: I verified making a connection to socket and listening to all events using "event plain all" There was no indication of such messages, but yes they were delivered to destination. However, Asterisk's gtalk module does fire an event when a chat message is received but I prefer to use freeswitch. On Thu, May 31, 2012 at 9:54 PM, Michael Collins wrote: > I don't believe events are fired on chat messages - I think they just get > routed to the destination. > > Can anyone verify this? > > -MC > > > On Thu, May 31, 2012 at 12:15 AM, Sameer Khan wrote: > >> Is this implemented? >> >> There is no event fired when a chat message is received >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120531/180bf5ff/attachment.html From anthony.minessale at gmail.com Thu May 31 23:20:26 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 31 May 2012 14:20:26 -0500 Subject: [Freeswitch-users] FreeTDM CallerID Detection with DTMF In-Reply-To: <1338484250.23766.17.camel@modesto.localdomain.net> References: <1338403799.3112.43.camel@modesto.localdomain.net> <1338484250.23766.17.camel@modesto.localdomain.net> Message-ID: I don't think it's implemented. You would have to ask Sangoma to implement that feature. On Thu, May 31, 2012 at 12:10 PM, Antonio Modesto wrote: > Hi, > > ? ? ? ?Does anybody have any suggestions? I searched a lot about it but I was > not successful, I found that here in brazil the callerid is sent via > DTMF but without any warning or polarity reversal, it's a strange > implementation. Any hints would be appreciated. > > Regards. > > > On Wed, 2012-05-30 at 15:49 -0300, Antonio Modesto wrote: >> Hi, >> >> ? ? ? I have a Digium TDM410P card, I am using it here in Brazil, where the >> signaling is not FSK, it's DTMF. It is working, though I am not >> receiving the callerid: >> >> Caller-Caller-ID-Number: [0000000000] >> >> Here is my freetdm.conf: >> >> [span zt FXO1] >> fxo-channel => 1 >> >> [span zt FXO2] >> fxo-channel => 2 >> >> [span zt FXO3] >> fxo-channel => 3 >> >> Here is one section of my autoload_configs/freetdm.conf.xml: >> >> ? ? >> ? ? ? >> ? ? ? >> ? ? ? >> ? ? ? >> ? ? ? >> ? ? ? >> ? ? ? >> ? ? ? >> ? ? ? >> ? ? ? >> ? ? ? >> ? ? ? >> ? ? ? >> ? ? >> >> Is it possible to enable the callerid detection in these conditions, or >> is it a hardware/driver limitation? >> >> >> Regards. >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From mytemike72 at gmail.com Thu May 31 23:24:01 2012 From: mytemike72 at gmail.com (Michael Lutz) Date: Thu, 31 May 2012 21:24:01 +0200 Subject: [Freeswitch-users] DTMF Passthrough on different legs In-Reply-To: <6ECAF1527329364583AB525CF34ABF950B31A67C@ms.kallback.com> References: <6ECAF1527329364583AB525CF34ABF950B31A67C@ms.kallback.com> Message-ID: According to CDRS's: Session:A -> Inbound, PCMU-8000 Session B -> FS to FS, PCMU-8000 Session C -> Outbound, PMCA-8000 According to vars.xml: Regards, Mike. 2012/5/31 JRichey : > What CODECs are being used? > > In-band DTMF won't work correctly with compressed CODECs like G.729. > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org]On Behalf Of > Michael Lutz > Sent: Thursday, May 31, 2012 2:12 AM > To: FreeSWITCH Users Help > Subject: [Freeswitch-users] DTMF Passthrough on different legs > > > Hi Guys, > > I am stuck with a bit of a problem I cannot figure out where it is going > wrong.. > > Here's my situation, > > I have an inbound call, comming from an external provider, they don't > support RFC2833 so I turn on inband detection using 'start_dtmf' in > the dialplan. > This works ok, for the incomming channel and IVR provided (via Lua) to > the caller. (Though sometimes it seems to misinterpret and returns > wrong keys..) (Session:A) > > After that, this call is bridged (via Lua to a a new inbound session > on the same switch) (without specifying start_dtmf) so RC2388 is used > by default (?) (Session:B) > > This new sessions is playing soem audio (Lua) and is heard by the > original caller (the two sessions are bridged). > > > Seperatly I have an inbound ESL connection where I do an origanate > using a originate {execute_on_answer=start_dtmf, .... etc) &park() to > setup an new external call via the same gateway (they don't support > RFC2388 so I activate the inband detection on the b-leg to receive the > digits. > Let's call this leg Session:C. > > After the orignated call is answered I bridge the parked call with the > previous session (Session:B), so at the end Session:A and Session:B > can talk to each other. > > Al this works perfectly. > > [Incommming call] Session:A ----- INBAND ------> Bridge(FS.Lua) ---> > Session:B ---- RFC2388 ----> ? FS.Bridge ---> Session:C --- INBAND > ------> ESL destination number > > > My symptoms are... > > 1. I have a customer who had a IVR on his own side, (Session:C) and > requires input from Session:A. The weird thing is that it seems the > input is wrong. > Keys entered do not correspond to what his IVR is receiving. As if > newly dtmfs are generated somewhere. > When I switch to another inbound provider, making Session:A using > rfc2388 and not inband, but do dialout via the provider using inband > detection. It seems to work fine. > > > 2. Not being able to get the dtmf's from the ESL Destination number > (Session:C). I'm reading this by subscribing to events via ESL for > that particulair uuid. > When I have a connection (inbound) without inband detection it seems > to get the DTMF's from Session:C without any problems. Which makes it > weird for me. > > note: I know the setup is a bit odd, but Session A cannot exit it's > Lua script, so I needed to do a workaround to be able to keep te lua > alive and being able to control the call (audio) via ESL. > > I realy need some help on this one, becuase we have a lot of customers > complaining about not responding to dtmf's or receving invalid input > in the (FS) IVR. > > > Thanks for you help, > > Regards, > Mike. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From jmesquita at freeswitch.org Thu May 31 23:51:26 2012 From: jmesquita at freeswitch.org (=?utf-8?Q?Jo=C3=A3o_Mesquita?=) Date: Thu, 31 May 2012 16:51:26 -0300 Subject: [Freeswitch-users] FreeTDM CallerID Detection with DTMF In-Reply-To: References: <1338403799.3112.43.camel@modesto.localdomain.net> <1338484250.23766.17.camel@modesto.localdomain.net> Message-ID: <6AFD632870AB4C3ABB086CAC635090A3@freeswitch.org> Just as an alternative, Khomp supports DTMF caller id detection since they are a brazilian manufacturer. It surely complies with the Brazilian standard. Regards, -- Jo?o Mesquita Sent with Sparrow (http://www.sparrowmailapp.com/?sig) On Thursday, May 31, 2012 at 4:20 PM, Anthony Minessale wrote: > I don't think it's implemented. > You would have to ask Sangoma to implement that feature. > > > > On Thu, May 31, 2012 at 12:10 PM, Antonio Modesto > wrote: > > Hi, > > > > Does anybody have any suggestions? I searched a lot about it but I was > > not successful, I found that here in brazil the callerid is sent via > > DTMF but without any warning or polarity reversal, it's a strange > > implementation. Any hints would be appreciated. > > > > Regards. > > > > > > On Wed, 2012-05-30 at 15:49 -0300, Antonio Modesto wrote: > > > Hi, > > > > > > I have a Digium TDM410P card, I am using it here in Brazil, where the > > > signaling is not FSK, it's DTMF. It is working, though I am not > > > receiving the callerid: > > > > > > Caller-Caller-ID-Number: [0000000000] > > > > > > Here is my freetdm.conf: > > > > > > [span zt FXO1] > > > fxo-channel => 1 > > > > > > [span zt FXO2] > > > fxo-channel => 2 > > > > > > [span zt FXO3] > > > fxo-channel => 3 > > > > > > Here is one section of my autoload_configs/freetdm.conf.xml: > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > Is it possible to enable the callerid detection in these conditions, or > > > is it a hardware/driver limitation? > > > > > > > > > Regards. > > > > > > > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting Services: > > > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://wiki.freeswitch.org > > > http://www.cluecon.com > > > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at 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 at freeswitch.org (mailto:consulting at freeswitch.org) > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com (mailto:anthony_minessale at hotmail.com) > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com (mailto:anthony.minessale at gmail.com) > IRC: irc.freenode.net (http://irc.freenode.net) #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org (mailto:888 at conference.freeswitch.org) > googletalk:conf+888 at conference.freeswitch.org (mailto:conf+888 at conference.freeswitch.org) > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... 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