April 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Random Freeswitch Crash
by Jai Rangi 24 Apr '15

24 Apr '15

FS Server has been running fine for last 5 months, suddenly since yesterday it started crashing every few hours. Checked the log files and did not see any error message. I made few changes recently: Enabled Shared line in sip_profile/internal <param name="presence-probe-on-register" value="true"/> <param name="manage-shared-appearance" value="true"/> Using a LUA script for group call, change the originate to enterprise originate. From bridge(user/101@domainname,user/102@domainname) To bridge(<all_threads=true>user/101@domainname:_:user/102@domainname) The reason I did this, was that, if shared line is registered on more than one device, we want all the devices to ring and other people should be able to take call by just pressing the line numbers. Changes the debug to level to 10 and still no information in the log file. No core dump was generated for me to share or look at. FS is running a server with decent specs. Dual CPU, Quad Core, 48 GB Ram. SSD Hard drive. Wondering if anyone else had the smiler issue and if I can get a hint where I can look for more hints on why FS dies all of a sudden. Thank you, -Jai

5 4

Suspend RTP Transmission Until RTP Has Been Received From Remote Party
by JP 24 Apr '15

24 Apr '15

I've observed that FreeSWITCH will initially transmit RTP to the remote party's private IP address until it has received at least 10 packets, at which time it will start sending to the remote party's NATed address. Is there some way to suspend the transmission of RTP altogether until at least one packet has been received from the remote party? It would be ideal if this could be controlled on a per call and/or per SIP profile basis. Thanks,

2 1

Music on hold for multi-tenant environment
by Paul Atreides 24 Apr '15

24 Apr '15

Hi, I am using a Grandstream GXP2140. When I am having an active call and another person is calling me then I can switch between the two calls. The person set to hold should hear the hold music but it only plays the beginning and then just comes silence. I don't get any error messages. How can I set it up that it keeps playing the music? Log: 2015-04-23 21:19:00.154957 [DEBUG] switch_channel.c:1961 (sofia/internal/ sip:13@10.0.200.13:5060) Callstate Change HELD -> UNHELD 2015-04-23 21:19:00.154957 [DEBUG] switch_core_io.c:1026 (sofia/internal/ sip:13@10.0.200.13:5060) Callstate Change UNHELD -> ACTIVE Dialplan: <action application="bridge_export" data="hold_music=$${sounds_dir}/music/song.mp3"/> Thank you

1 1

FreeSWITCH Friday FreeForAll Reminder!
by Ken Rice 24 Apr '15

24 Apr '15

FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH Friday FreeFor All Visit http://ift.tt/1n3h0Pf and Click Call 888 with your WebRTC enabled Browser and headset, Call sip:888@conference.freeswitch.org or see http://ift.tt/1prwIZL for access info! -- Ken FreeSWITCH.org ClueCon.com OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon

1 0

[Warning][switch_core_media.c:1945 Changing current codec to PCMU (payload type 0)]
by Bilal Abbasi 24 Apr '15

24 Apr '15

Hi FS Users, I am currently getting the following Warning message on my FS server. 2015-04-24 03:00:37.433196 [WARNING] switch_core_media.c:1945 Changing current codec to PCMU (payload type 0). 2015-04-24 03:00:37.433196 [DEBUG] sofia.c:6413 Channel sofia/x.x.x.x entering state [ready][200] 2015-04-24 03:00:37.453197 [DEBUG] switch_core_media.c:2260 Changing Codec from G722@20ms@8000hz to PCMU@20ms@8000hz 2015-04-24 03:00:37.493187 [DEBUG] switch_core_media.c:2352 Set Codec sofia/sipinterface_1/simple(a)x.x.x.x PCMU/8000 20 ms 160 samples 64000 bits 1 channels 2015-04-24 03:00:37.493187 [DEBUG] switch_core_codec.c:123 sofia/sipinterface_1/simple(a)x.x.x.x Original read codec replaced with PCMU:0 2015-04-24 03:00:37.493187 [NOTICE] switch_core_io.c:1257 Activating write resampler Please help me, Regards Bilal Abbasi

2 1

freeswitch sometimes crashes with core dump
by Кирилл Преслер 24 Apr '15

24 Apr '15

Please help me to analyze and understand what's going on. We have FS with python script dial plan. Python script asks SOAP API to get dial plan for call, parses result and returns Dialplan XML via fetch_xml function. I can't distinguish main source of error and crash, is problem in FS itself, mod_python, python or in my script. Please help me, below some system data and link to dump log CentOS release 6.6 (Final) linux 2.6.32-504.8.1.el6.x86_64 FreeSWITCH Version 1.2.24+git~20141120T223008Z~5dfc7cf803~64bit (git 5dfc7cf 2014-11-20 22:30:08Z 64bit) Python 2.6.6 (r266:84292, Jan 22 2014, 09:42:36) (SUDS,lxml...) http://pastebin.com/BkSmzq73

3 2

Duplicate DTMF Issue
by Jeremy 24 Apr '15

24 Apr '15

Hey Giovanni, Awesome! I will compile it tonight and let you know how it goes. Thank you again for your help. Jeremy Hello Jeremy, ok, understood now. Gsmopen uses internally spandsp to detect DTMFs. Because of possible false multiple detections, there is an hardcoded minimum duration of 350 ms for each DTMF. In your specific case, edit mod_gsmopen.cpp in /usr/src/freeswitch/src/mod/endpoints/mod_gsmopen, and modify the value at line 3302: if ((new_dtmf_timestamp - tech_pvt->old_dtmf_timestamp) > 350000) { //FIXME: make it configurable change that value to 50000 or little less (is not configurable, as you can see :) ) and recompile (eg: from /usr/src/freeswitch type make mod_gsmopen-install) That should do the trick. Let us know if you encounter problems. -giovanni On Thu, Apr 23, 2015 at 2:39 PM, Jeremy <jhorsburgh at gmail.com <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>> wrote: >* Hi Giovanni, *>>* Thank you for your reply, sorry I didn't explain the problem very well. *>>* I am trying to detect a string of DTMF's with a short pulse duration (about 50ms) and the normal DTMF detection is not able to reliably do this (it seems to miss almost all of the tones). *>>* Spandsp appears to be able to reliably do this but when I enable spandsp, I get duplicate tones detected by both spandsp and gsmopen (I think its gsmopen, example below). *>>* You are correct. Is it possible for gsmopen not to recognize DTMF's and use only spandsp to analyze the audio? *>>* Thank you for your help in advance. *>>* Jeremy. *>>>* P.S. Greatly appreciate the work you have put into GSMopen, brilliant work! *>>>* Hello Jeremy, *>>* I've not understood the problem, and the solution you are looking for. *>>* You want gsmopen not to recognize DTMFs, so you can analyze the audio with *>* spandsp? *>>* Can you give more info ? *>>* -giovanni *>>* On Wed, Apr 22, 2015 at 2:58 PM, jhorsburgh <jhorsburgh at gmail.com <http://gmail.com> <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>>> wrote: *>>* >* Hello All, *>* *>* I am having some issues detecting DTMF tones and was hoping *>* *>* someone could shed some light on the problem. *>* *>>* *Background* *>* *>* I am using Freeswitch (1.4.14~32bit) with GSMopen to receive calls and *>* *>* process high-speed DTMF signal for use with Alarm Systems. Unfortunately *>* *>* the *>* *>* standard inband DTMF detection is not quite up to the task but spandsp *>* *>* works *>* *>* perfectly. *>* *>>* *Problem* *>* *>* The problem I am having is that the standard inband DTMF detection remains *>* *>* enabled when spandsp is enabled producing duplicates of each detected tone. *>* *>>* Is there a way of disabling DTMF detection with the exception of spandsp? *>* *>>* Any help would be greatly appreciated. *>* *>>* Kind regards, *>* *>* Jeremy *>* *>>>>* 2015-04-22 12:18:22.129468 [DEBUG] mod_gsmopen.cpp:919 rev 1.4.14~32bit *>* *>* [(nil)|37 ][DEBUG_GSMOPEN 919 ][interface1][-1, 5, 5] *>* *>* gsmopen/interface1 CHANNEL got *>* *>* SWITCH_MESSAGE_INDICATE_TRANSCODING_NECESSARY *>* *>* 2015-04-22 12:18:22.569494 [DEBUG] switch_channel.c:488 RECV DTMF 5:2000 *>* *>* 2015-04-22 12:18:22.569494 [DEBUG] mod_gsmopen.cpp:745 rev 1.4.14~32bit *>* *>* [(nil)|37 ][DEBUG_GSMOPEN 745 ][interface1][-1, 5, 5] DTMF DETECTED: *>* *>* [5] new_dtmf_timestamp: 1652802529, delta_t: 515839955 *>* *>* 2015-04-22 12:18:22.569494 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN *>* *>* DETECTED: [5] *>* *>* 2015-04-22 12:18:22.569494 [DEBUG] switch_channel.c:488 RECV DTMF 5:2000 *>* *>* 2015-04-22 12:18:22.749457 [DEBUG] mod_spandsp_dsp.c:385 DTMF END DETECTED: *>* *>* [5], duration = 178 ms *>* *>* 2015-04-22 12:18:23.049446 [DEBUG] switch_channel.c:488 RECV DTMF 3:2000 *>* *>* 2015-04-22 12:18:23.049446 [DEBUG] mod_gsmopen.cpp:745 rev 1.4.14~32bit *>* *>* [(nil)|37 ][DEBUG_GSMOPEN 745 ][interface1][-1, 5, 5] DTMF DETECTED: *>* *>* [3] new_dtmf_timestamp: 1653282480, delta_t: 479951 *>* *>* 2015-04-22 12:18:23.049446 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN *>* *>* DETECTED: [3] *>* *>* 2015-04-22 12:18:23.049446 [DEBUG] switch_channel.c:488 RECV DTMF 3:2000 *>* *>* 2015-04-22 12:18:23.229447 [DEBUG] mod_spandsp_dsp.c:385 DTMF END DETECTED: *>* *>* [3], duration = 165 ms *>* *>* 2015-04-22 12:18:23.529449 [DEBUG] switch_channel.c:488 RECV DTMF 5:2000 *>* *>* 2015-04-22 12:18:23.529449 [DEBUG] mod_gsmopen.cpp:745 rev 1.4.14~32bit *>* *>* [(nil)|37 ][DEBUG_GSMOPEN 745 ][interface1][-1, 5, 5] DTMF DETECTED: *>* *>* [5] new_dtmf_timestamp: 1653762488, delta_t: 480008 *>* *>* 2015-04-22 12:18:23.549464 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN *>* *>* DETECTED: [5] *>* *>* 2015-04-22 12:18:23.549464 [DEBUG] switch_channel.c:488 RECV DTMF 5:2000 *>* *>* 2015-04-22 12:18:23.689495 [DEBUG] mod_spandsp_dsp.c:385 DTMF END DETECTED: *>* *>* [5], duration = 140 ms *>* *>* 2015-04-22 12:18:24.029440 [DEBUG] switch_channel.c:488 RECV DTMF 2:2000 *>* *>* 2015-04-22 12:18:24.029440 [DEBUG] mod_gsmopen.cpp:745 rev 1.4.14~32bit *>* *>* [(nil)|37 ][DEBUG_GSMOPEN 745 ][interface1][-1, 5, 5] DTMF DETECTED: *>* *>* [2] new_dtmf_timestamp: 1654262471, delta_t: 499983 *>* *>* 2015-04-22 12:18:24.029440 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN *>* *>* DETECTED: [2] *>* *>* 2015-04-22 12:18:24.029440 [DEBUG] switch_channel.c:488 RECV DTMF 2:2000 *>* *>* 2015-04-22 12:18:24.169442 [DEBUG] mod_spandsp_dsp.c:385 DTMF END DETECTED: *>* *>* [2], duration = 153 ms *>* *>* 2015-04-22 12:18:24.489445 [DEBUG] switch_channel.c:488 RECV DTMF 5:2000 *>* *>* 2015-04-22 12:18:24.489445 [DEBUG] mod_gsmopen.cpp:745 rev 1.4.14~32bit *>* *>* [(nil)|37 ][DEBUG_GSMOPEN 745 ][interface1][-1, 5, 5] DTMF DETECTED: *>* *>* [5] new_dtmf_timestamp: 1654722477, delta_t: 460006 *>* *>* 2015-04-22 12:18:24.489445 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN *>* *>* DETECTED: [5] *>* *>* 2015-04-22 12:18:24.489445 [DEBUG] switch_channel.c:488 RECV DTMF 5:2000 *>* *>* 2015-04-22 12:18:24.649447 [DEBUG] mod_spandsp_dsp.c:385 DTMF END DETECTED: *>* *>* [5], duration = 153 ms *>* *>* 2015-04-22 12:18:25.609450 [INFO] switch_cpp.cpp:1328 Failed to validate *>* *>* Checksum - '5533552255' *>* *>>>>>>>* -- *>* *>* View this message in context: *>* *>* http://freeswitch-users.2379917.n2.nabble.com/Duplicate-DTMF-Issue-tp759615… <http://freeswitch-users.2379917.n2.nabble.com/Duplicate-DTMF-Issue-tp759615…> <http://freeswitch-users.2379917.n2.nabble.com/Duplicate-DTMF-Issue-tp759615… <http://freeswitch-users.2379917.n2.nabble.com/Duplicate-DTMF-Issue-tp759615…>> *>* *>* Sent from the freeswitch-users mailing list archive at Nabble.com. *>* *>>* _________________________________________________________________________ *>* *>* Professional FreeSWITCH Consulting Services: *>* *>* consulting at freeswitch.org <http://freeswitch.org> <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>> *>* *>* http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/> <http://www.freeswitchsolutions.com/ <http://www.freeswitchsolutions.com/>> *>* *>>* Official FreeSWITCH Sites *>* *>* http://www.freeswitch.org <http://www.freeswitch.org/> <http://www.freeswitch.org/ <http://www.freeswitch.org/>> *>* *>* http://confluence.freeswitch.org <http://confluence.freeswitch.org/> <http://confluence.freeswitch.org/ <http://confluence.freeswitch.org/>> *>* *>* http://www.cluecon.com <http://www.cluecon.com/> <http://www.cluecon.com/ <http://www.cluecon.com/>> *>* *>>* FreeSWITCH-users mailing list *>* *>* FreeSWITCH-users at lists.freeswitch.org <http://lists.freeswitch.org> <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>> *>* *>* http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>> *>* *>* UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users> <http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users>> *>* *>* http://www.freeswitch.org <http://www.freeswitch.org/> <http://www.freeswitch.org/ <http://www.freeswitch.org/>> *>* *> *>>>* -- *>* Sincerely, *>>* Giovanni Maruzzelli *>* Cell : +39-347-2665618 *>*

1 0

Peer to Peer as compared to sipxecs
by Naveen Khanna 24 Apr '15

24 Apr '15

Hi, Recently I am being asked about peer to peer communication capabilities of freeswtich as compared to sipxecs. As I understand that NAT is a blocker for having peer to peer communication, but It appears that spixecs has some work arounds. So I thought I will bring my understanding to this forum and get some clarity on the Subject. There are three scenarios : #1 Freeswitch, and 2 SIP phones are in the same NAT. #2 Freeswitch and 2 SIP phones are in 2 separate NATs. i.e. Freeswitch is in one NAT and 2 SIP Phones are behind 1 common ADSL connection #3 Freeswitch and 2 SIP phones are in 3 separate NATs. i.e. Freeswitch is in on NAT and 2 SIP Phones are behind 2 ADSL connection provide by different ISP. As per my understanding #1 is fairly possible in freeswitch, however I need to how does freeswitch #2 and #3. Regards, Naveen Khanna

1 0

Stale channels with mod_callcenter in uuid-standby mode
by Saumar Hajjar 23 Apr '15

23 Apr '15

(Reposting: the 1st "version" didn't reach the list for some reason) After a mod_callcenter member hangs up, its channel is still present in the system, and its uuid is unkillable. I believe FS team calls it a "stale channel". The uuid-standby agent won't receive any more calls, until he/she hangs up - what would make uuid-standby a "agent hangup and call again" mode. What I'm trying to do, is nothing fancy. Basically I'm following the mod_callcenter documentation without much tweaking. Here's the test case setup: - Freeswitch with default config, mod_callcenter enabled also with default config Tested with the following versions: 1.4.18+git~20150312T185523Z~4eed221b69~64bit (git 4eed221 2015-03-12 18:55:23Z 64bit) 1.5.15b+git~20150117T062211Z~46cf8a4dce~64bit (git 46cf8a4 2015-01-17 06:22:11Z 64bit) 1.5.15b+git~20150421T235828Z~a4d877c189~64bit (git a4d877c 2015-04-21 23:58:28Z 64bit) - Dialplan mod_callcenter entries:  <extension> <condition field="destination_number" expression="^\*91$"> <action application="callcenter" data="support@default"/> </condition> </extension>  <extension> <condition field="destination_number" expression="^(4099)$"> <action application="set" data="transfer_after_bridge=4099"/>  <action application="sleep" data="300"/>  <action application="set" data="res=${callcenter_config(agent set uuid ${caller_id_number}@default '${uuid}')}" /> <action application="set" data="res=${callcenter_config(agent set type ${caller_id_number}@default 'uuid-standby')}" /> <action application="set" data="res=${callcenter_config(agent set status ${caller_id_number}@default 'Available (On Demand)')}" /> <action application="set" data="res=${callcenter_config(agent set state ${caller_id_number}@default 'Waiting')}" /> <action application="set" data="cc_warning_tone=tone_stream://%(200,0,500,600,700)"/> <action application="answer" /> <action application="playback" data="$${hold_music}"/> </condition> </extension> - Start FS and add a test agent: callcenter_config agent add 1002@default uuid-standby - Agent (extension 1002) dials 4099: The agent starts hearing music on hold. show channels 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,initial_cid_name,initial_cid_num,initial_ip_addr,initial_dest,initial_dialplan,initial_context a1c9b1a4-7926-4da9-9041-100df3d1bbbb,inbound,2015-04-22 18:01:46,1429736506,sofia/internal/1002@192.168.0.136,CS_EXECUTE,1002,1002,192.168.0.103,4099,playback,local_stream://moh,XML,default,PCMU,8000,64000,PCMU,8000,64000,,hajjar-notebook,1002@192.168.0.136,,ACTIVE,,,,,,,1002,1002,192.168.0.103,4099,XML,default callcenter_config agent list name|system|uuid|type|contact|status|state|max_no_answer|wrap_up_time|reject_delay_time|busy_delay_time|no_answer_delay_time|last_bridge_start|last_bridge_end|last_offered_call|last_status_change|no_answer_count|calls_answered|talk_time|ready_time 1002@default|single_box|a1c9b1a4-7926-4da9-9041-100df3d1bbbb|uuid-standby||Available (On Demand)|Waiting|1|5|0|5|5|1429734669|1429735319|1429734669|1429736506|0|0|0|0 - Member (extension 1000) dials *91: The member gets bridged to the uuid-stand by agent. Great. show channels 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,initial_cid_name,initial_cid_num,initial_ip_addr,initial_dest,initial_dialplan,initial_context a1c9b1a4-7926-4da9-9041-100df3d1bbbb,inbound,2015-04-22 18:01:46,1429736506,sofia/internal/1002@192.168.0.136,CS_EXCHANGE_MEDIA,1000,1000,192.168.0.103,4099,uuid_bridge,d298fa41-b855-4027-ba4a-ac112df641d5,XML,default,PCMU,8000,64000,PCMU,8000,64000,,hajjar-notebook,1002@192.168.0.136,,ACTIVE,1002,1002,SEND,d298fa41-b855-4027-ba4a-ac112df641d5,1000,1000,1002,1002,192.168.0.103,4099,XML,default d298fa41-b855-4027-ba4a-ac112df641d5,inbound,2015-04-22 18:04:47,1429736687,sofia/internal/1000@192.168.0.136,CS_SOFT_EXECUTE,1000,1000,192.168.0.120,*91,uuid_bridge,a1c9b1a4-7926-4da9-9041-100df3d1bbbb,XML,default,PCMU,8000,64000,PCMU,8000,64000,,hajjar-notebook,1000@192.168.0.136,,ACTIVE,1002,1002,SEND,d298fa41-b855-4027-ba4a-ac112df641d5,1002,1002,1000,1000,192.168.0.120,*91,XML,default 2 total. show calls 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 d298fa41-b855-4027-ba4a-ac112df641d5,inbound,2015-04-22 18:04:47,1429736687,sofia/internal/1000@192.168.0.136,CS_SOFT_EXECUTE,1000,1000,192.168.0.120,*91,1000@192.168.0.136,,ACTIVE,1002,1002,SEND,d298fa41-b855-4027-ba4a-ac112df641d5,hajjar-notebook,1002,1002,a1c9b1a4-7926-4da9-9041-100df3d1bbbb,inbound,2015-04-22 18:01:46,1429736506,sofia/internal/1002@192.168.0.136,CS_EXCHANGE_MEDIA,1000,1000,192.168.0.103,4099,1002@192.168.0.136,,ACTIVE,1002,1002,SEND,1000,1000,1429736687 1 total. callcenter_config agent list name|system|uuid|type|contact|status|state|max_no_answer|wrap_up_time|reject_delay_time|busy_delay_time|no_answer_delay_time|last_bridge_start|last_bridge_end|last_offered_call|last_status_change|no_answer_count|calls_answered|talk_time|ready_time 1002@default|single_box|a1c9b1a4-7926-4da9-9041-100df3d1bbbb|uuid-standby||Available (On Demand)|In a queue call|1|5|0|5|5|1429736687|1429735319|1429736687|1429736506|0|1|0|0 - Member hangs up show channels 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,initial_cid_name,initial_cid_num,initial_ip_addr,initial_dest,initial_dialplan,initial_context a1c9b1a4-7926-4da9-9041-100df3d1bbbb,inbound,2015-04-22 18:01:46,1429736506,sofia/internal/1002@192.168.0.136,CS_EXECUTE,1002,1002,192.168.0.103,4099,playback,local_stream://moh,XML,default,PCMU,8000,64000,PCMU,8000,64000,,hajjar-notebook,1002@192.168.0.136,,ACTIVE,1000,1000,RECV,d298fa41-b855-4027-ba4a-ac112df641d5,,,1002,1002,192.168.0.103,4099,XML,default d298fa41-b855-4027-ba4a-ac112df641d5,inbound,2015-04-22 18:04:47,1429736687,sofia/internal/1000@192.168.0.136,CS_SOFT_EXECUTE,1000,1000,192.168.0.120,*91,uuid_bridge,a1c9b1a4-7926-4da9-9041-100df3d1bbbb,XML,default,PCMU,8000,64000,PCMU,8000,64000,,hajjar-notebook,1000@192.168.0.136,,ACTIVE,1002,1002,SEND,d298fa41-b855-4027-ba4a-ac112df641d5,1002,1002,1000,1000,192.168.0.120,*91,XML,default 2 total. show calls 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 d298fa41-b855-4027-ba4a-ac112df641d5,inbound,2015-04-22 18:04:47,1429736687,sofia/internal/1000@192.168.0.136,CS_SOFT_EXECUTE,1000,1000,192.168.0.120,*91,1000@192.168.0.136,,ACTIVE,1002,1002,SEND,d298fa41-b855-4027-ba4a-ac112df641d5,hajjar-notebook,1002,1002,a1c9b1a4-7926-4da9-9041-100df3d1bbbb,inbound,2015-04-22 18:01:46,1429736506,sofia/internal/1002@192.168.0.136,CS_EXECUTE,1002,1002,192.168.0.103,4099,1002@192.168.0.136,,ACTIVE,1000,1000,RECV,,,1429736687 1 total. callcenter_config agent list name|system|uuid|type|contact|status|state|max_no_answer|wrap_up_time|reject_delay_time|busy_delay_time|no_answer_delay_time|last_bridge_start|last_bridge_end|last_offered_call|last_status_change|no_answer_count|calls_answered|talk_time|ready_time 1002@default|single_box|a1c9b1a4-7926-4da9-9041-100df3d1bbbb|uuid-standby||Available (On Demand)|Waiting|1|5|0|5|5|1429736687|1429735319|1429736687|1429736990|0|1|0|0 Here the agent gets transfered back to uuid-standby extension and restarts hearing moh. Its state is "Waiting" - but he/she won't answer new calls. But the state is "Waiting" only because it got overridden by the dialplan. In my tests, removing the callcenter_config(agent set state..) from dialplan (and controlling agent state via CLI), I've noticed that the agent state never gets back to "Idle" (if On Demand) or "Waiting". It gets stuck in "In a queue call". Trying to uuid_kill the member uuid, results in -ERR No such channel! - When agent finally hangs up: 2015-04-22 18:21:56.804183 [NOTICE] sofia.c:952 Hangup sofia/internal/1002(a)192.168.0.136 [CS_EXECUTE] [NORMAL_CLEARING] 2015-04-22 18:21:56.824219 [NOTICE] switch_core_session.c:1641 Session 6 (sofia/internal/1000(a)192.168.0.136) Ended 2015-04-22 18:21:56.824219 [NOTICE] switch_core_session.c:1645 Close Channel sofia/internal/1000(a)192.168.0.136 [CS_DESTROY] 2015-04-22 18:21:57.104198 [NOTICE] switch_core_session.c:1641 Session 4 (sofia/internal/1002(a)192.168.0.136) Ended 2015-04-22 18:21:57.104198 [NOTICE] switch_core_session.c:1645 Close Channel sofia/internal/1002(a)192.168.0.136 [CS_DESTROY] show channels, show calls return 0 If callcenter_config(agent set state...) is commented out in dialplan, callcenter_config agent list shows the agent as "Idle". Meaning mod_callcenter only perceives the call has been disconnected, after the agent hangs up... I'm not blaming mod_callcenter - maybe it's related to transfer_after_bridge. I already tried using exec_after_bridge_app/arg to a script that sleeps and transfer back to 4099. I also tried increasing the dialplan sleep. I'd appreciate any advice on how to make uuid-standby work. Thanks in advance, Saumar

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echo cancelation
by Sergey Safarov 23 Apr '15

23 Apr '15

Could you help me understand that prevents the use echo cancellation in FS On the one hand SpanDSP library compiled with FS and contains functions for echo canceler libs/spandsp/src/echo.c On the other hand FS does not contain applications of echo canceler. Also, I found the discussion in which states GPL is not compatible with the MPL1.1 and it prevent use Oslec (SpanDSP is uses Oslec). Sergey

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