- FreeSWITCH-users - FreeSWITCH Mailing Lists

dbh:query - insert id
by Lloyd Aloysius 28 May '13

28 May '13

Hello All How to get the id value after insert a record a record using dbh:query *table_a - columns*. id - auto increment field1 field2 dbh:query("insert into table_a ( field1,field2) values ('11','Test')") After insert how to get the table_a - id value for the inserted record? Thanks Lloyd

11 18

Splitting CDRs on call forward
by Jon Schøpzinsky 28 May '13

28 May '13

Hi List, I am implementing call forwarding on a multi tenant system, and therefore need to split CDR's when the call forward happens, so that if the receiving user also has his account call forwarded, he pays for his part of the call. A calls B B forwards to C C forwards to an external mobile phone. B has a free call from B to C, but C needs to pay for the forwarding to the mobile phone. Therefore i need a separate CDR for the C to Mobile phone call. Another example would be this A works in Company A, and B works in Company B They are both users on our system, and therefore is on the same freeswitch. A calls B B Forwards to an external mobile phone. Here A needs to pay for the call from A to B, and B needs to pay for the call being forwarded to his mobile phone. Do anybody have an idea as to how to implement this in freeswitch. Back in my Asterisk days, this would be done by the ForkCDR command. Venlig hilsen/kind regards Jon Leren Schøpzinsky

5 9

Centralized SIP directory
by mehroz 28 May '13

28 May '13

Hi all, I have a DB server, storing SIP users and auth information. 2 freeswitch servers configured with mod_xml_curl , and an attempt of registration on both servers looks for SIP authentication from DB server. At the moment, i am defining 2 domains on DB server (FS-1 IP and FS-2 IP), and creating unique user , 2 times to address both domains. I want to get rid of creating 2 users for both domains. What is the best approach. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Centralized-SIP-directory-tp7… Sent from the freeswitch-users mailing list archive at Nabble.com.

4 7

FS bypass ZRTP with TURN relay
by adahary 28 May '13

28 May '13

I've setup the FS with bypass_media mode to let endpoints to pass media through a TURN server. The dialplan is set dynamically to bypass_mode=true. The endpoints can now register SIP with FS and then pass media between thenself via the TURN relay (FS not in between) with no one-way-audio issue. The problem is when I setup FS and the endpoints with ZRTP. Even though all RTPs are passing directly between endpoints and FS log indications show that ZRTP is activated on both endpoints (including a=zrtp-hash: in both SDPs) and still the ZRTP is not poping up in the endpoints display. I assume that once media is being relayed between endpoints then ZRTP should pass between like it happens when both endpoints on the same LAN network. What could cause the ZRTP not to work between endpoints when media is bypassing the FS? regards Assaf -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FS-bypass-ZRTP-with-TURN-rela… Sent from the freeswitch-users mailing list archive at Nabble.com.

2 4

SSL/TLS customized encryption.
by mehroz 28 May '13

28 May '13

Hi all, I am trying to create certificates and keys on a customized encryption scheme. AFAIK gentls_cert generates certificates and keys with : Public Key Info: Public Key Algorithm: rsaEncryption Public-Key: (2048 bit) I want the encryption to be according the SUITE-B standard ,i .e 1. (AES) with key sizes of 128 and 256 bits 2 . Elliptic Curve Digital Signature Algorithm 3 . Elliptic Curve Diffie–Hellman (ECDH) — key agreement 4 .Secure Hash Algorithm 2 (SHA-256 and SHA-384) — message digest I can see FS script (gentls_cert) is using openssl at the back. And as i have very little idea of encryption and all using openssl , i need to know if this is possible to mend this script to our requirements, if so ... how and where should i look at first! Thanks -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/SSL-TLS-customized-encryption… Sent from the freeswitch-users mailing list archive at Nabble.com.

2 6

Getting CHANNEL_PROGRESS event only sometimes when originating call
by mbo 27 May '13

27 May '13

We are using Freeswitch version 1.2.9. connected with an Sangoma A104 card to the PSTN. If I originate a call though esl socket connection, I'm not always getting a CHANNEL_PROGRESS event. In the test scenario, I'm repeating the originate a couple of times with the same phone number and same parameters, sometimes I get CHANNEL_PROGRESS sometimes not. In the Q.931 trace I can see that I always get a ALERT message, which if I see it correctly, should be converted into a CHANNEL_PROGRESS event. We also have a demo system here with 2 Sangoma A101 cards connected with an crossed E1 cable (also Freeswitch 1.2.9). If I perform the same test here, I always get the CHANNEL_PROGRESS event. Are there any known issues that CHANNEL_PROGRESS event isn't fired reliable when originating calls via freetdm? Thanks Markus

1 0

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