May 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

bridging 2 call legs after a conference
by Tihomir Culjaga 27 May '13

27 May '13

hello folks, im controlling FS via ESL and what im trying to do is to make a 3-way conference and this works fine. What i have an issue with is when one conference member hangs up, i want to return back to a normal bridge between the 2 remaining members ... so what im doing is: On Incoming call from 1013 to FS answer the call: ["sendmsg 00349e91-117e-4dcc-b2db-45d3a54c5fee","call-command: execute","execute-app-name: answer"]) play an announcement to A leg: CMD uuid_broadcast "bgapi uuid_broadcast 00349e91-117e-4dcc-b2db-45d3a54c5fee 'greeting1.wav'" originate a new call and parks it: CMD originate "bgapi originate {origination_uuid=fc3e44c5-63af-45bb-8a8e-8a35bac52234,ignore_early_media=false,originate_timeout=15}user/1002 &park" bridge these 2 legs into a call: CMD uuid_bridge "bgapi uuid_bridge fc3e44c5-63af-45bb-8a8e-8a35bac52234 00349e91-117e-4dcc-b2db-45d3a54c5fee" originate a 2nd new call into a conference: CMD originate "bgapi originate {origination_uuid=3f516dc9-61a8-4c36-97f0-ea7ac18c6b9f,ignore_early_media=false,originate_timeout=0}user/1014 &conference(supervizor36@silent+flags{mute}) &park" join the 1st new call (user/1002) CMD_MULTI conference ["sendmsg fc3e44c5-63af-45bb-8a8e-8a35bac52234","call-command: execute","execute-app-name: conference","execute-app-arg: supervizor36@silent"] join incoming call into the same conference CMD_MULTI conference ["sendmsg 00349e91-117e-4dcc-b2db-45d3a54c5fee","call-command: execute","execute-app-name: conference","execute-app-arg: supervizor36@silent"] unmute uuid=fc3e44c5-63af-45bb-8a8e-8a35bac52234 CMD api "api conference supervizor36 unmute 20" Incoming call 00349e91-117e-4dcc-b2db-45d3a54c5fee must not hear uuid=fc3e44c5-63af-45bb-8a8e-8a35bac52234 CMD api "api conference supervizor36 relate 21 20 nohear" so far its great! now, call uuid=3f516dc9-61a8-4c36-97f0-ea7ac18c6b9f (the one originated into the conference) hangs up, 2 remaining calls (incoming call and 1st originated calls) can still hear each others... but i want to get rid of a conference since there are just 2 members hence no sense keeping the conference up... so i do: bridge the two remaining conference members together CMD uuid_bridge "bgapi uuid_bridge fc3e44c5-63af-45bb-8a8e-8a35bac52234 00349e91-117e-4dcc-b2db-45d3a54c5fee" and i get no audio!! looks like FS setting both channels on park ?? any idea ? 1. 2013-05-27 15:34:13.724037 [NOTICE] switch_core_session.c:1367 Session 30 (sofia/internal/sip:1014@192.168.254.112:5060) Ended 2. 2013-05-27 15:34:13.724037 [NOTICE] switch_core_session.c:1369 Close Channel sofia/internal/sip:1014@192.168.254.112:5060 [CS_DESTROY] 3. 2013-05-27 15:34:13.724037 [DEBUG] switch_core_state_machine.c:491 ( sofia/internal/sip:1014@192.168.254.112:5060) Callstate Change HANGUP -> DOWN 4. 2013-05-27 15:34:13.724037 [DEBUG] switch_core_state_machine.c:494 ( sofia/internal/sip:1014@192.168.254.112:5060) Running State Change CS_DESTROY 5. 2013-05-27 15:34:13.724037 [DEBUG] switch_core_state_machine.c:504 ( sofia/internal/sip:1014@192.168.254.112:5060) State DESTROY 6. 2013-05-27 15:34:13.724037 [DEBUG] mod_sofia.c:363 sofia/internal/sip: 1014@192.168.254.112:5060 SOFIA DESTROY 7. 2013-05-27 15:34:13.724037 [DEBUG] switch_core_state_machine.c:86 sofia/internal/sip:1014@192.168.254.112:5060 Standard DESTROY 8. 2013-05-27 15:34:13.724037 [DEBUG] switch_core_state_machine.c:504 ( sofia/internal/sip:1014@192.168.254.112:5060) State DESTROY going to sleep 9. 2013-05-27 15:34:13.724037 [DEBUG] switch_ivr_bridge.c:1521 (sofia/internal/ sip:1002@192.168.254.116:5060) State Change CS_SOFT_EXECUTE -> CS_HIBERNATE 10. 2013-05-27 15:34:13.724037 [DEBUG] switch_core_session.c:1175 Send signal sofia/internal/sip:1002@192.168.254.116:5060 [BREAK] 11. 2013-05-27 15:34:13.724037 [DEBUG] switch_ivr_bridge.c:1523 ( sofia/external/1013(a)192.168.254.93) State Change CS_EXCHANGE_MEDIA -> CS_HIBERNATE 12. 2013-05-27 15:34:13.724037 [DEBUG] switch_core_session.c:1175 Send signal sofia/external/1013(a)192.168.254.93 [BREAK] 13. 2013-05-27 15:34:13.724037 [DEBUG] switch_core_session.c:786 Send signal sofia/external/1013(a)192.168.254.93 [BREAK] 14. 2013-05-27 15:34:13.724037 [DEBUG] switch_core_session.c:786 Send signal sofia/internal/sip:1002@192.168.254.116:5060 [BREAK] 15. 2013-05-27 15:34:13.744036 [DEBUG] mod_conference.c:2899 Channel leaving conference, cause: NONE 16. 2013-05-27 15:34:13.744036 [DEBUG] mod_conference.c:2899 Channel leaving conference, cause: NONE 17. 2013-05-27 15:34:13.744036 [DEBUG] switch_core_session.c:724 Send signal sofia/internal/sip:1002@192.168.254.116:5060 [BREAK] 18. 2013-05-27 15:34:13.744036 [DEBUG] switch_core_codec.c:141 sofia/internal/sip:1002@192.168.254.116:5060 Restore previous codec PCMA:8. 19. 2013-05-27 15:34:13.744036 [DEBUG] switch_core_session.c:724 Send signal sofia/internal/sip:1002@192.168.254.116:5060 [BREAK] 20. 2013-05-27 15:34:13.744036 [DEBUG] switch_ivr_bridge.c:329 Send signal sofia/external/1013(a)192.168.254.93 [BREAK] 21. 2013-05-27 15:34:13.744036 [DEBUG] switch_core_session.c:724 Send signal sofia/external/1013(a)192.168.254.93 [BREAK] 22. 2013-05-27 15:34:13.744036 [DEBUG] switch_core_codec.c:141 sofia/external/1013(a)192.168.254.93 Restore previous codec PCMA:8. 23. 2013-05-27 15:34:13.744036 [DEBUG] switch_core_session.c:724 Send signal sofia/external/1013(a)192.168.254.93 [BREAK] 24. 2013-05-27 15:34:13.744036 [DEBUG] switch_ivr_bridge.c:329 Send signal sofia/internal/sip:1002@192.168.254.116:5060 [BREAK] 25. 2013-05-27 15:34:13.764036 [DEBUG] mod_conference.c:1602 Write Lock ON 26. 2013-05-27 15:34:13.764036 [DEBUG] switch_ivr_bridge.c:586 BRIDGE THREAD DONE [sofia/internal/sip:1002@192.168.254.116:5060] 27. 2013-05-27 15:34:13.764036 [DEBUG] mod_conference.c:1605 Write Lock OFF 28. 2013-05-27 15:34:13.764036 [DEBUG] switch_ivr_bridge.c:606 Send signal sofia/external/1013(a)192.168.254.93 [BREAK] 29. 2013-05-27 15:34:13.764036 [DEBUG] switch_core_session.c:724 Send signal sofia/external/1013(a)192.168.254.93 [BREAK] 30. 2013-05-27 15:34:13.764036 [DEBUG] switch_core_session.c:724 Send signal sofia/internal/sip:1002@192.168.254.116:5060 [BREAK] 31. 2013-05-27 15:34:13.764036 [DEBUG] switch_ivr_bridge.c:1366 ( sofia/internal/sip:1002@192.168.254.116:5060) State Change CS_HIBERNATE -> CS_RESET 32. 2013-05-27 15:34:13.764036 [DEBUG] switch_core_session.c:1175 Send signal sofia/internal/sip:1002@192.168.254.116:5060 [BREAK] 33. 2013-05-27 15:34:13.764036 [DEBUG] switch_core_state_machine.c:423 (sofia/internal/ sip:1002@192.168.254.116:5060) State SOFT_EXECUTE going to sleep 34. 2013-05-27 15:34:13.764036 [DEBUG] switch_core_state_machine.c:362 ( sofia/internal/sip:1002@192.168.254.116:5060) Running State Change CS_RESET 35. 2013-05-27 15:34:13.764036 [DEBUG] switch_core_state_machine.c:413 ( sofia/internal/sip:1002@192.168.254.116:5060) State RESET 36. 2013-05-27 15:34:13.764036 [DEBUG] mod_sofia.c:166 sofia/internal/sip:1002@192.168.254.116:5060 SOFIA RESET 37. 2013-05-27 15:34:13.764036 [DEBUG] switch_ivr_bridge.c:721 sofia/internal/sip:1002@192.168.254.116:5060 CUSTOM RESET 38. 2013-05-27 15:34:13.764036 [DEBUG] switch_ivr_bridge.c:728 ( sofia/internal/sip:1002@192.168.254.116:5060) State Change CS_RESET -> CS_SOFT_EXECUTE 39. 2013-05-27 15:34:13.764036 [DEBUG] switch_core_session.c:1175 Send signal sofia/internal/sip:1002@192.168.254.116:5060 [BREAK] 40. 2013-05-27 15:34:13.764036 [DEBUG] switch_core_state_machine.c:413 ( sofia/internal/sip:1002@192.168.254.116:5060) State RESET going to sleep 41. 2013-05-27 15:34:13.764036 [DEBUG] switch_core_state_machine.c:362 ( sofia/internal/sip:1002@192.168.254.116:5060) Running State Change CS_SOFT_EXECUTE 42. 2013-05-27 15:34:13.764036 [DEBUG] switch_core_state_machine.c:423 ( sofia/internal/sip:1002@192.168.254.116:5060) State SOFT_EXECUTE 43. 2013-05-27 15:34:13.764036 [DEBUG] mod_sofia.c:572 SOFIA SOFT_EXECUTE 44. 2013-05-27 15:34:13.764036 [DEBUG] switch_ivr_bridge.c:746 sofia/internal/sip:1002@192.168.254.116:5060 CUSTOM SOFT_EXECUTE 45. 2013-05-27 15:34:13.764036 [DEBUG] switch_ivr_bridge.c:586 BRIDGE THREAD DONE [sofia/external/1013(a)192.168.254.93] 46. 2013-05-27 15:34:13.764036 [DEBUG] switch_ivr_bridge.c:606 Send signal sofia/internal/sip:1002@192.168.254.116:5060 [BREAK] 47. 2013-05-27 15:34:13.764036 [DEBUG] switch_ivr.c:2422 (sofia/external/ 1013(a)192.168.254.93) State Change CS_HIBERNATE -> CS_PARK 48. 2013-05-27 15:34:13.764036 [DEBUG] switch_core_session.c:1175 Send signal sofia/external/1013(a)192.168.254.93 [BREAK] 49. 2013-05-27 15:34:13.764036 [DEBUG] switch_core_state_machine.c:420 ( sofia/external/1013(a)192.168.254.93) State EXCHANGE_MEDIA going to sleep 50. 2013-05-27 15:34:13.764036 [DEBUG] switch_core_state_machine.c:362 ( sofia/external/1013(a)192.168.254.93) Running State Change CS_PARK 51. 2013-05-27 15:34:13.764036 [DEBUG] switch_core_state_machine.c:426 ( sofia/external/1013(a)192.168.254.93) State PARK 52. 2013-05-27 15:34:13.764036 [DEBUG] switch_core_state_machine.c:247 sofia/external/1013(a)192.168.254.93 Standard PARK 53. 54. 55. 56. freeswitch@internal> 2013-05-27 15:34:18.457165 [WARNING] sofia_reg.c:1400 SIP auth challenge (REGISTER) on sofia profile 'internal' for [1014(a)192.168.254.93]from ip 192.168.254.112 57. 58. freeswitch@internal> 59. freeswitch@internal> 60. freeswitch@internal> show calls 61. uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,hostname,sent_callee_name,sent_callee_num,b_uuid,b_direction,b_created,b_created_epoch,b_name,b_state,b_cid_name,b_cid_num,b_ip_addr,b_dest,b_presence_id,b_presence_data,b_callstate,b_callee_name,b_callee_num,b_callee_direction,b_sent_callee_name,b_sent_callee_num,call_created_epoch 62. 00349e91-117e-4dcc-b2db-45d3a54c5fee,inbound,2013-05-2715:32:49, 1369661569,sofia/external/1013(a)192.168.254.93,CS_PARK,1013,1013,192.168 .254.112,38515494471,,,ACTIVE,,,SEND,00349e91-117e-4dcc-b2db-45d3a54c5fee,cc01,Outbound Call,1002,,,,,,,,,,,,,,,,,,, 63. fc3e44c5-63af-45bb-8a8e-8a35bac52234,outbound,2013-05-27 15:33:05, 1369661585,sofia/internal/sip:1002@192.168.254.116:5060,CS_SOFT_EXECUTE,Outbound Call,1002,,1002,1002(a)192.168.254.93 ,,ACTIVE,,,SEND,fc3e44c5-63af-45bb-8a8e-8a35bac52234,cc01,1013,1013 ,,,,,,,,,,,,,,,,,,, 64. 65. 2 total. 66. 67. freeswitch@internal> show channels 68. uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,read_bit_rate,write_codec,write_rate,write_bit_rate,secure,hostname,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,sent_callee_name,sent_callee_num 69. 00349e91-117e-4dcc-b2db-45d3a54c5fee,inbound,2013-05-2715:32:49, 1369661569,sofia/external/1013(a)192.168.254.93,CS_PARK,1013,1013,192.168 .254.112,38515494471,conference,supervizor36@silent,XML,default,PCMA,8000 ,64000,PCMA,8000,64000,,cc01,,,ACTIVE,,,SEND,00349e91-117e-4dcc-b2db-45d3a54c5fee,Outbound Call,1002 70. fc3e44c5-63af-45bb-8a8e-8a35bac52234,outbound,2013-05-27 15:33:05, 1369661585,sofia/internal/sip:1002@192.168.254.116:5060,CS_SOFT_EXECUTE,Outbound Call,1002,,1002,conference,supervizor36@silent,,default,PCMA,8000,64000 ,PCMA,8000,64000,,cc01,1002(a)192.168.254.93 ,,ACTIVE,,,SEND,fc3e44c5-63af-45bb-8a8e-8a35bac52234,1013,1013 71. 72. 2 total. 73. 74. freeswitch@internal> 75. freeswitch@internal> 76. freeswitch@internal> 77. freeswitch@internal> 2013-05-27 15:34:57.732290 [DEBUG] switch_core_session.c:870 Send signal sofia/internal/sip:1002@192.168 .254.116:5060 [BREAK] Tihomir.

2 3

OpenVZ tuning tips
by Tamas Jalsovszky 27 May '13

27 May '13

Hello, Do you have any recommendations regarding how to set up correctly (for production) CentOS5 openvz and FS 1.2.stable? Is there any trick to tuneup the system to be rock solid? Right now we use centos5 openvz and ubuntu 10.04 LTS in container with FS 1.2.8 and RTP deltas are varying from 15 to around 40ms. We guess that something is not well configured around timers, however mod_posix_timer did not help anything (running FS with -rp). We use our own bare metal and can reproduce those delatas eirher when only one VE is on the HW. Maybe time to check out centos6 with openvz? Any idea, recommendation, experience can be very helpful. Regards, Jalsot

8 16

Freeswitch just won't transcode the call.
by Daniel Ivanov 27 May '13

27 May '13

I'm having media problems with a FS here. UAs are sending mixed capabilites all the time and i wanna gracefully degradate on them. This is what i have in vars.xml: <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,GSM,CODEC2"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMA,G729"/> This is in external.xml: <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="outbound-codec-prefs" value="G729"/> <param name="inbound-late-negotiation" value="false"/> <param name="disable-transcoding" value="false"/> <param name="inbound-proxy-media" value="true"/> I need to make sure that when a UA dials in, the A-leg is negotiated between g729 and gsm(and codec2) and the call is then transcoded to the B-leg if it's not G729 instead of just resending the SDP directly to the gateways. I've tried disabling and enabling proxy-media, late-negotiation and disable-transcoding along with passing the strict codec_string and absolute_codec_string. Point me on what i'm failing to apprehend here.

3 3

WebRTC
by mbo 27 May '13

27 May '13

I've seen there have been a couple of discussions about WebRTC support earlier this year. Are there any news on that? Thanks Markus

4 3

Anyone used Instant messaging with FS
by Farrukh Ali 27 May '13

27 May '13

Dear all, I am trying to configure and use instant messaging with FS, can anyone give me a hint about configuration in FS and the soft phones? Regards, Muhammad Farrukh

2 1

Freetdm PRI Span Failover
by Ashish gautam 27 May '13

27 May '13

Hi, I have been using single span PRI card with FreeTDM but now I plan to use 8 Span card. I want to know if the channels on a particular span are all busy(or unavailable) at a time and a call needs to be originated, then how to automatically pass it through the other free span (failover kind of a thing) on which the channels are available? Please help! -- Ashish Gautam

1 0

missing wav files from yum install?
by Adam Cooley 27 May '13

27 May '13

Got freeswitch installed, and debugs on fs_cli show were missing audio files with a fresh install of 1.2.9 from yum on centos. 2013-05-25 19:48:27.071848 [ERR] mod_sndfile.c:202 Error Opening File [/usr/share/freeswitch/sounds/en/us/callie/ivr/ivr-to_call_the_freeswitch_conference.wav] [System error : No such file or directory.] Where do we get the default system wav files from? -a

1 0

SRTP media handling survey behaviour.
by Daniel Ivanov 26 May '13

26 May '13

I wanted to implement a mixed mode of srtp and zrtp crypto capabilities across our network and i was facing several issues. But succeeded nonetheless. What kicked me in the balls hardest was rfc3713(taken from memory) compliance of the UAs. Whenever i wanted to set SRTP to optional on the devices sent both avp and savp lines along with the according keys in the a=crypto lines in the sdp and the zrtp-hash too. If for a reason i wanted to force ZRTP on the channel, i was supposed to set sip_secure_media to false and that would disable srtp on the b-leg and zrtp handshake would take place in p2p mode. I was using csipsimple(pjsip) though and fs was just stripping the savp from the sdp and still passing the a=crypto lines. It turned out greedy and still started talking srtp instead of zrtp. I patched it myself to strip the a=crypto too if sip_secure_media is false and it worked like a german. I wanted to know if anyone has observed how other UAs handle the srtp negotiation and if it's worth proposing my patch as an improvement on the JIRA.

1 0

Strange issue with late negotiation
by Flavio Goncalves 25 May '13

25 May '13

Hello, I'm having a really strange issue on billing related to late negotiation. The call end up hanged with no BYE. There is a dialer sending thousands of calls through and OpenSIPS to a termination based on FreeSwitch. I have a SIP call flow as below. Some times during the day, when the volume is high, the UAC drops some 200Ok and FS send a reinvite in the middle of the Initial transaction. ----INVITE --------------- Proxy --INVITE ---------------> FS <----200 OK -------------- Proxy <---200 OK- ------------- FS <--REINVITE--------------- Proxy <-REINVITE--------------- FS ---481 leg does not exit-> Proxy ---481------------------> FS <-- ACK (REINVITE)-------- Proxy <-ACK(REINVITE)---------- FS ----CANCEL ---------------> Proxy ---CANCEL --------------> FS <---200 Ok ---------------- Proxy <----200 Ok ------------- FS FS is sending a reinvite before ACK comes from the client. I have two questions: 1) Is it valid to send a reinvite in the middle of an existing transaction? According to the RFC3261 Section 14. Note that a UAC MUST NOT initiate a new INVITE transaction within a dialog while another INVITE transaction is in progress in either direction. 1. If there is an ongoing INVITE client transaction, the TU MUST wait until the transaction reaches the completed or terminated state before initiating the new INVITE. 2. If there is an ongoing INVITE server transaction, the TU MUST wait until the transaction reaches the confirmed or terminated state before initiating the new INVITE. But it is confusing because just below it says the opposite. However, a UA MAY initiate a regular transaction while an INVITE transaction is in progress. A UA MAY also initiate an INVITE transaction while a regular transaction is in progress. 2. Shouldn't FS send a BYE after sending a 200Ok and not receiving the ACK? If a UAS generates a 2xx response and never receives an ACK, it SHOULD generate a BYE to terminate the dialog. Best regards, Flavio E. Goncalves

4 5

How fs handle the sip-T message?
by xiaofengcanyuexp＠163.com 25 May '13

25 May '13

Hi, support team I know FS can pass out SS7 info via properity header. While, how it handles an incoming SIP-T message with SS7 info encapsulated in "content-type". Will it parse it or discard it? Thanks Windy ------------------- 2013-05-25

3 3