September 2016 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Getting, sofia_glue.c:1031 Error creating HANDLE!
by devang nathwani 02 Sep '16

02 Sep '16

Hello, While routing the call from opensips server to freeswitch server i am getting this error in fs_cli log, 2016-08-26 14:05:13.611542 [CRIT] sofia_glue.c:1031 Error creating HANDLE! url_str=[sip:357xxxxxxxx@178.xxx.xxx.xxx] call_id=[N/A] to_str=[<sip:357xxxxxxxx@178.xxx.xxx.xxx>] from_str=["<null>" <sip:<null>@172.16.3.2>] invite_contact=[<sip:gw+IPComms@172.16.3.2:5060;transport=udp;gw=gtw-name>] I have tried configuring 1.6.8 and 1.6.9 versions of freeswitch but getting the same error. What am i missing?

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error: tport_logging: capture socket error: No such file or directory
by Ned Ludd 02 Sep '16

02 Sep '16

Trying to set up Homer and I get this in the "fs_cli" output. I saw something from 2014 in the IRC logs about it, there was debate whether it was a bug, but there didn't seem to be a resolution. I do have the appropriate entries in "sofia.conf.xml": <param name="capture-server" value="udp:10.0.0.84:5060"/> And my sip profile: <param name="sip-capture" value="yes"/> And I do see traffic on my capture server. Using 1.6.9. Any idea? ----- Ned Ludd

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One-way audio with Verto + Chrome (with video enabled)
by Joel Serrano 02 Sep '16

02 Sep '16

Hi, I'm getting started with WebRTC, Verto, etc. I'm facing an issue with chrome+video, I get one-way-audio. Scenario: I login to Verto and I call a conference that has other users in it (those users enter the conference by SIP). 1- Chrome: If video is enabled, I can send audio and other users will hear me but I can't hear them. 2- Chrome: If video is disabled, all works correctly. 3- Firefox: with video enabled, all works correctly. This definitely looks like a config issue but I have no idea where to start looking. I found this JIRA https://freeswitch.org/jira/browse/FS-7679 but in my case, I think the correct candidate is being selected. Should I attach the full logs from all 3 test calls? Please let me know also if I should open a new JIRA for this, I just want to make sure this isn't a config issue first. Thanks! Joel.

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Problems with group calls
by hartmut.schebl 02 Sep '16

02 Sep '16

I have some Problems with group calls. I'm using FreeSWITCH Version 1.6.6-13-d2d0b32~64bit (-13-d2d0b32 64bit) and mod_xml_curl with a java backend to control the FreeSWITCH. Now my Problem: I have groups, for example: <document type="freeswitch/xml"> <section name="directory"> <domain name="my.domain.de"> <groups> <group name="GROUP1"> <users> <user id="group.user1" type="pointer"/> <user id="group.user2" type="pointer"/> ... </users> </group> </groups> </domain> </section> </document> Some users inside the groups have multiple devices, for example: <document type="freeswitch/xml"> <section name="directory"> <domain name="my.domain.de"> <groups> <group name="default"> <users> <user id="group.user1" number-alias="163"> <params> </params> <variables/> </user> </users> </group> </groups> <variables> <variable name="user_context" value="my.domain.de"/> </variables> </domain> </section> </document> If I call this group with the +A option like this: <action application="bridge" data="${group_call(GROUP1(a)my.domain.de+A)}"/> only the first device of the user rings. The channel to the second device is ended immediately. The Log says: 2016-09-02 10:00:11.724379 [NOTICE] switch_ivr_originate.c:1415 Hangup sofia/internal/82040(a)172.16.0.21 [CS_INIT] [LOSE_RACE] If I call the group using the +E option like this: <action application="bridge" data="${group_call(GROUP1(a)my.domain.de+E)}"/> All devices are ringing and that problem is solved, but now I run into an other problem that doesn't exist with +A option. I use groups for fail over. If some user doesn't answer in a defined time a group should start ringing together with the user. The line in dialplan for this is: <action application="bridge" data="<ignore_early_media=true>user/some.user@my.domain.de:_:{origination_caller_id_name='Failover',is_failover='true'}[originate_delay_start=20000]${group_call(GROUP1(a)my.domain.de+E)}"/> In this case only the first user (group.user1 from example above) respects the delay all following users of the group start ringing immediately together with 'some.user'. I also tried an other syntax: <action application="bridge" data="<ignore_early_media=true>user/soma.user@my.domain.de:_:{originate_delay_start=20000,origination_caller_id_name='Failover',is_failover='true'}${group_call(GROUP1(a)my.domain.de+E)}"/> But I get the same problem and additionally the delay of 'group.user1' is doubled. I have no idea how to get out of this vicious circle. Either I can use only one device per user or I can't start the group delayed. Please help me to solve this problem. Best Regards, Hartmut -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Problems-with-group-calls-tp7… Sent from the freeswitch-users mailing list archive at Nabble.com.

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CHANNEL_PARK
by Tihomir Culjaga 02 Sep '16

02 Sep '16

hello, please can anyone help here. is there any way to distinguish a CHANNEL_PARK generated by att_xfer application from a "normal" CHANNEL_PARK generated by park application? this is a CHANNEL_PARK i got by using att_xfer application: Event-Name: CHANNEL_PARK Core-UUID: f63b2100-6eb0-11e6-b09e-e558b70924ad FreeSWITCH-Hostname: centralix.bbm.local FreeSWITCH-Switchname: centralix.bbm.local FreeSWITCH-IPv4: 192.168.141.60 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2016-09-01%2015%3A04%3A21 Event-Date-GMT: Thu,%2001%20Sep%202016%2013%3A04%3A21%20GMT Event-Date-Timestamp: 1472735061810863 Event-Calling-File: switch_ivr.c Event-Calling-Function: switch_ivr_park Event-Calling-Line-Number: 934 Event-Sequence: 78869 Channel-State: CS_PARK Channel-Call-State: ACTIVE Channel-State-Number: 6 Channel-Name: sofia/internal/888%4010.10.254.236%3A5060 Unique-ID: 934ee67c-7044-11e6-b404-e558b70924ad Call-Direction: outbound Presence-Call-Direction: outbound Channel-HIT-Dialplan: false Channel-Presence-ID: 888%40192.168.141.60 Channel-Call-UUID: 8cb36612-7044-11e6-b3d8-e558b70924ad Answer-State: answered Channel-Read-Codec-Name: PCMA Channel-Read-Codec-Rate: 8000 Channel-Read-Codec-Bit-Rate: 64000 Channel-Write-Codec-Name: PCMA Channel-Write-Codec-Rate: 8000 Channel-Write-Codec-Bit-Rate: 64000 Caller-Direction: outbound Caller-Logical-Direction: outbound Caller-Username: 844 Caller-Dialplan: XML Caller-Caller-ID-Name: 844 Caller-Caller-ID-Number: 844 Caller-Orig-Caller-ID-Name: 844 Caller-Orig-Caller-ID-Number: 844 Caller-Callee-ID-Name: Outbound%20Call Caller-Callee-ID-Number: 888 Caller-Network-Addr: 10.10.254.236 Caller-ANI: 844 Caller-Destination-Number: 888 Caller-Unique-ID: 934ee67c-7044-11e6-b404-e558b70924ad Caller-Source: mod_sofia Caller-Context: features Caller-RDNIS: 0916331550 Caller-Channel-Name: sofia/internal/888%4010.10.254.236%3A5060 Caller-Profile-Index: 1 Caller-Profile-Created-Time: 1472735051270863 Caller-Channel-Created-Time: 1472735051270863 Caller-Channel-Answered-Time: 1472735052210864 Caller-Channel-Progress-Time: 1472735051510855 Caller-Channel-Progress-Media-Time: 0 Caller-Channel-Hangup-Time: 0 Caller-Channel-Transfer-Time: 0 Caller-Channel-Resurrect-Time: 0 Caller-Channel-Bridged-Time: 1472735052230858 Caller-Channel-Last-Hold: 0 Caller-Channel-Hold-Accum: 0 Caller-Screen-Bit: true Caller-Privacy-Hide-Name: false Caller-Privacy-Hide-Number: false Other-Type: originator Other-Leg-Direction: inbound Other-Leg-Logical-Direction: inbound Other-Leg-Username: 844 Other-Leg-Dialplan: XML Other-Leg-Caller-ID-Name: 844 Other-Leg-Caller-ID-Number: 844 Other-Leg-Orig-Caller-ID-Name: 844 Other-Leg-Orig-Caller-ID-Number: 844 Other-Leg-Callee-ID-Name: Outbound%20Call Other-Leg-Callee-ID-Number: 0916331550 Other-Leg-Network-Addr: 10.10.254.235 Other-Leg-ANI: 844 Other-Leg-Destination-Number: agent_state Other-Leg-Unique-ID: 8cb36612-7044-11e6-b3d8-e558b70924ad Other-Leg-Source: mod_sofia Other-Leg-Context: features Other-Leg-RDNIS: 0916331550 Other-Leg-Channel-Name: sofia/internal/844%4010.10.254.254 Other-Leg-Profile-Created-Time: 0 Other-Leg-Channel-Created-Time: 0 Other-Leg-Channel-Answered-Time: 0 Other-Leg-Channel-Progress-Time: 0 Other-Leg-Channel-Progress-Media-Time: 0 Other-Leg-Channel-Hangup-Time: 0 Other-Leg-Channel-Transfer-Time: 0 Other-Leg-Channel-Resurrect-Time: 0 Other-Leg-Channel-Bridged-Time: 0 Other-Leg-Channel-Last-Hold: 0 Other-Leg-Channel-Hold-Accum: 0 Other-Leg-Screen-Bit: true Other-Leg-Privacy-Hide-Name: false Other-Leg-Privacy-Hide-Number: false variable_direction: outbound variable_is_outbound: true variable_uuid: 934ee67c-7044-11e6-b404-e558b70924ad variable_session_id: 65 variable_sip_profile_name: internal variable_channel_name: sofia/internal/888%4010.10.254.236%3A5060 variable_sip_destination_url: sip%3A888%4010.10.254.236%3A5060%3Bfs_nat%3Dyes variable_sip_cid_type: pid variable_switch_m_sdp: v%3D0%0D%0Ao%3D844%208000%208000%20IN%20IP4%2010.10.253.10%0D%0As%3DSIP%20Call%0D%0Ac%3DIN%20IP4%2010.10.253.10%0D%0At%3D0%200%0D%0Am%3Daudio%2015004%20RTP/AVP%208%2018%0D%0Aa%3Drtpmap%3A8%20PCMA/8000%0D%0Aa%3Drtpmap%3A18%20G729/8000%0D%0Aa%3Dfmtp%3A18%20annexb%3Dno%0D%0Aa%3Dptime%3A20%0D%0A variable_dialed_user: 888 variable_dialed_domain: 192.168.141.60 variable_sip_invite_domain: 192.168.141.60 variable_presence_id: 888%40192.168.141.60 variable_rtp_local_sdp_str: v%3D0%0Ao%3DFreeSWITCH%201472709987%201472709988%20IN%20IP4%2010.10.254.254%0As%3DFreeSWITCH%0Ac%3DIN%20IP4%2010.10.254.254%0At%3D0%200%0Am%3Daudio%2025064%20RTP/AVP%208%2013%0Aa%3Drtpmap%3A8%20PCMA/8000%0Aa%3Dptime%3A20%0Aa%3Dsendrecv%0A variable_sip_outgoing_contact_uri: %3Csip%3Amod_sofia%4010.10.254.254%3A5060%3E variable_sip_req_uri: 888%4010.10.254.236%3A5060 variable_sip_nat_detected: true variable_sofia_profile_name: internal variable_recovery_profile_name: internal variable_sip_local_network_addr: 10.10.254.254 variable_sip_reply_host: 10.10.254.236 variable_sip_reply_port: 5060 variable_sip_network_ip: 10.10.254.236 variable_sip_network_port: 5060 variable_ep_codec_string: PCMA%408000h%4020i%4064000b variable_sip_user_agent: Grandstream%20GXP2100%201.0.7.8 variable_sip_recover_contact: %3Csip%3A888%4010.10.253.11%3A5060%3E variable_sip_full_via: SIP/2.0/UDP%2010.10.254.254%3Brport%3D5060%3Bbranch%3Dz9hG4bK3gv0mrSa3Z1va variable_sip_recover_via: SIP/2.0/UDP%2010.10.254.254%3Brport%3D5060%3Bbranch%3Dz9hG4bK3gv0mrSa3Z1va variable_sip_from_display: 844 variable_sip_full_from: %22844%22%20%3Csip%3A844%40192.168.141.60%3E%3Btag%3DmDe1c5tB3tDrp variable_sip_full_to: %3Csip%3A888%4010.10.254.236%3A5060%3E%3Btag%3D1143493955 variable_sip_from_user: 844 variable_sip_from_uri: 844%40192.168.141.60 variable_sip_from_host: 192.168.141.60 variable_sip_to_user: 888 variable_sip_to_port: 5060 variable_sip_to_uri: 888%4010.10.254.236%3A5060 variable_sip_to_host: 10.10.254.236 variable_sip_contact_user: 888 variable_sip_contact_port: 5060 variable_sip_contact_uri: 888%4010.10.253.11%3A5060 variable_sip_contact_host: 10.10.253.11 variable_sip_to_tag: 1143493955 variable_sip_from_tag: mDe1c5tB3tDrp variable_sip_cseq: 96031461 variable_sip_call_id: 6aad3d60-eae7-1234-4793-f4ce46a93024 variable_switch_r_sdp: v%3D0%0D%0Ao%3D888%208000%208000%20IN%20IP4%2010.10.253.11%0D%0As%3DSIP%20Call%0D%0Ac%3DIN%20IP4%2010.10.253.11%0D%0At%3D0%200%0D%0Am%3Daudio%2015004%20RTP/AVP%208%0D%0Aa%3Drtpmap%3A8%20PCMA/8000%0D%0Aa%3Dptime%3A20%0D%0A variable_rtp_use_codec_string: PCMA%408000h%4020i,PCMA variable_rtp_audio_recv_pt: 8 variable_rtp_use_codec_name: PCMA variable_rtp_use_codec_rate: 8000 variable_rtp_use_codec_ptime: 20 variable_rtp_use_codec_channels: 1 variable_rtp_last_audio_codec_string: PCMA%408000h%4020i%401c variable_read_codec: PCMA variable_original_read_codec: PCMA variable_read_rate: 8000 variable_original_read_rate: 8000 variable_write_codec: PCMA variable_write_rate: 8000 variable_dtmf_type: info variable_local_media_ip: 10.10.254.254 variable_local_media_port: 25064 variable_advertised_media_ip: 10.10.254.254 variable_rtp_use_timer_name: soft variable_rtp_use_pt: 8 variable_rtp_use_ssrc: 4157591195 variable_endpoint_disposition: ANSWER variable_current_application_data: 4%20a%20s%20execute_extension%3A%3Aatt_xfer%20XML%20features variable_current_application: bind_meta_app variable_record_stereo: true variable_default_gateway: example.com variable_default_areacode: 918 variable_transfer_fallback_extension: operator variable_toll_allow: domestic,international,local variable_accountcode: 888 variable_max_calls: 2 variable_limit_exceeded: limit_exceeded variable_user_context: default variable_effective_caller_id_name: Karolina variable_effective_caller_id_number: 888 variable_outbound_caller_id_name: FreeSWITCH variable_outbound_caller_id_number: 0000000000 variable_callgroup: techsupport variable_sip-force-contact: NDLB-connectile-dysfunction variable_max_forwards: 68 variable_originator_codec: PCMA%408000h%4020i variable_originator: 8cb36612-7044-11e6-b3d8-e558b70924ad variable_export_vars: RFC2822_DATE,execute_on_answer_1,execute_on_answer_2 variable_RFC2822_DATE: Thu,%2001%20Sep%202016%2015%3A04%3A00%20%2B0200 variable_execute_on_answer_1: start_dtmf variable_execute_on_answer_2: bind_meta_app%204%20a%20s%20execute_extension%3A%3Aatt_xfer%20XML%20features variable_originate_early_media: true variable_originating_leg_uuid: 8cb36612-7044-11e6-b3d8-e558b70924ad variable_call_uuid: 8cb36612-7044-11e6-b3d8-e558b70924ad variable_last_bridge_to: 8cb36612-7044-11e6-b3d8-e558b70924ad variable_bridge_channel: sofia/internal/844%4010.10.254.254 variable_bridge_uuid: 8cb36612-7044-11e6-b3d8-e558b70924ad variable_signal_bond: 8cb36612-7044-11e6-b3d8-e558b70924ad variable_last_sent_callee_id_name: 844 variable_last_sent_callee_id_number: 844 variable_remote_media_ip_reported: 10.10.253.11 variable_remote_media_ip: 10.10.254.236 variable_remote_media_port_reported: 15004 variable_remote_media_port: 15004 variable_rtp_auto_adjust: true variable_sip_hangup_phrase: OK variable_last_bridge_hangup_cause: NORMAL_CLEARING variable_last_bridge_proto_specific_hangup_cause: sip%3A200 Content-Length: 9672 Content-Type: text/event-plain regards, Tihomir.

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Background noise on SIP 2 SIP calls
by Don Hawkins 02 Sep '16

02 Sep '16

Hey everyone! When I place a call from one extension to another (SIP account -> SIP account / both on same FreeSwitch server) there is a LOT of background noise, I guess I would describe it as white noise. On the flip side, when calls are placed through one of our gateways to a landline or mobile the call is crystal clear without any background noise. I've tried different SIP phones in all kinds of ways, from my mobile (using SIP app), from Grandstream ATA, from X-Lite, they all have a lot of background noise on SIP 2 SIP calls. Any suggestions? A setting perhaps?

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Re: [Freeswitch-users] Event Socket Outbound - Playback Sequence
by Luiz Fernando Softov 02 Sep '16

02 Sep '16

Now I'm having another problem. Let's say I have some audio files to playback file_1.mp3 -> hello, you have pendencies file_2.mp3 -> to inform payment file_3.mp3 -> please press file_4.mp3 -> one Then, i do execute playback, without LOCK (because I need to garb DTMF and send BREAK to stop playback) playback="file_1.mp3!file_2.mp3!file_3.mp3!file_4.mp3" FS must say -> hello, you have pendencies, to inform payment, please press, one But FS is saying -> hello, you have ..... payment, please ... one The scenario is 2001 --> [FS A] -- gateway --> [FS B] --> playfiles If the scenario is 2001 --> [FS A] --> playfiles The files are played correct This is happen ing with the same codec in all legs I already enable and disable transcoding, without success There is the log of FS A https://pastebin.freeswitch.org/view/b783fc14 A nd log of FS B https://pastebin.freeswitch.org/view/95aa3580 2016-09-01 19:43 GMT-04:00 Luiz Fernando Softov <fernando(a)softov.com.br>: > Now I'm having another problem. > > Let's say > I > have > some > audio files to playback > > file_1.mp3 -> > > hello, you have pendencies > file_2.mp3 -> to inform payment > file_3.mp3 -> please press > file_4.mp3 -> one > > Then, i do execute playback, without LOCK (because I need to garb DTMF > and send BREAK to stop playback) > > > playback="file_1.mp3!file_2.mp3!file_3.mp3!file_4.mp3" > > FS must say > -> hello, you have pendencies, to inform payment, please press, one > > But FS is saying > -> hello, you have > > ..... > > payment, please ... one > > The scenario is > > 2001 --> [FS A] -- gateway --> [FS B] --> playfiles > > If the scenario is > > 2001 --> [FS A] --> playfiles > > The files are played correct > > > > This > is > happen > ing > with the same codec in all legs > > > I already enable and disable transcoding, > without success > > > There is the log of FS A > > https://pastebin.freeswitch.org/view/b783fc14 > > A > nd log of > FS B > > > https://pastebin.freeswitch.org/view/95aa3580 > > > > > > 2016-08-27 17:41 GMT-04:00 Luiz Fernando Softov <fernando(a)softov.com.br>: > >> Thanks, this do the job... >> >> 2016-08-26 8:01 GMT-04:00 Dave Horton <daveh(a)beachdognet.com>: >> >>> Can't you play all the files in a single command using a playback >>> delimiter? >>> >>> Sent from my iPhone >>> >>> On Aug 26, 2016, at 12:34 AM, Luiz Fernando Softov < >>> fernando(a)softov.com.br> wrote: >>> >>> Hi all, >>> >>> I'm using FS with Event Socket Outbound. >>> >>> Eg: The user calls to XXX and the calls goes to socket. >>> >>> <extension name="VoiprExt"> >>> <condition> >>> <action >>> >>> application="socket" data="127.0.0.1:3012 async full"/> >>> <action application="hangup"/> >>> </condition> >>> </extension> >>> >>> I receive a call, and play some files (sometimes http:// or shout://). >>> Using command action playback. >>> >>> sendmsg <uuid> >>> call-command: execute >>> execute-app-name: playback >>> execute-app-arg: >>> >>> >>> /sounds/system/chr_ >>> 1 >>> .mp3 >>> >>> >>> The problem is, I'm sending commands to playback files 1, 2, 3 >>> And the system are playing 3, 2, 1 >>> >>> I already tried to use event-lock. >>> S >>> ometimes, >>> >>> if user press any DTMF >>> , >>> I need to stop playback >>> I >>> f I use >>> event-lock >>> : true. I >>> can't do anything until the files are done. >>> The same, using command break (break only show up when files are done). >>> >>> Is there another way, or a correct way to playback multiple sequential >>> files, in order >>> , using >>> >>> application="socket"? >>> This is a BUG? >>> >>> >>> I don't if I made myself clear here. >>> >>> In resume, I can't wait until the current file is done, to send another >>> command to play the next file, since I have one list (options), with >>> several list of files to play (files). And, using asynchronous events >>> (kqueue), the list(s) can change in the middle of process. >>> >>> It's like >>> * Option 1: >>> ** file: wellcome.mp3 >>> ** file: file_ >>> abc >>> 1 >>> .mp3 >>> >>> * Option >>> 2 >>> : >>> ** file: protocol.mp3 >>> >>> ** file: >>> digit >>> _ >>> 8 >>> .mp3 >>> >>> >>> ** file: >>> digit_ >>> 7 >>> .mp3 >>> >>> >>> ** file: >>> digit_ >>> 1 >>> .mp3 >>> >>> >>> >>> ** file: >>> digit_ >>> 2 >>> .mp3 >>> >>> >>> >>> ** file: >>> digit_ >>> 3 >>> .mp3 >>> >>> >>> >>> >>> * Option >>> 3 >>> : >>> ** file: >>> sales >>> .mp3 >>> >>> >>> And, after this, the files to be played are >>> >>> wellcome.mp3 >>> file_ >>> abc >>> 1 >>> .mp3 >>> protocol.mp3 >>> digit >>> _ >>> 8 >>> .mp3 >>> digit_ >>> 7 >>> .mp3 >>> digit_ >>> 1 >>> .mp3 >>> >>> digit_ >>> 2 >>> .mp3 >>> >>> digit_ >>> 3 >>> .mp3 >>> >>> sales >>> .mp3 >>> >>> But, the files are not played in sequence. >>> >>> >>> >>> >>> >>> [DEBUG] switch_channel.c:3770 (sofia/profile_1/6001(a)192.168.142.12) >>> Callstate Change RINGING -> ACTIVE >>> [DEBUG] sofia.c:6858 Channel sofia/profile_1/6001(a)192.168.142.12 >>> entering state [completed][200] >>> [DEBUG] switch_ivr.c:623 sofia/profile_1/6001(a)192.168.142.12 Command >>> Execute playback(/sounds/ivr/ivr_intro.mp3) >>> EXECUTE sofia/profile_1/6001(a)192.168.142.12 >>> playback(/sounds/ivr/ivr_intro.mp3) >>> [DEBUG] switch_ivr_play_say.c:1467 Codec Activated L16@8000hz 1 >>> channels 20ms >>> [DEBUG] sofia.c:6858 Channel sofia/profile_1/6001(a)192.168.142.12 >>> entering state [ready][200] >>> [DEBUG] switch_rtp.c:6706 Correct audio ip/port confirmed. >>> [DEBUG] switch_rtp.c:1845 rtcp_stats_init: audio ssrc[1125545576] >>> base_seq[28137] >>> [DEBUG] switch_ivr_play_say.c:1910 done playing file >>> /sounds/ivr/ivr_intro.mp3 >>> [DEBUG] switch_ivr.c:623 sofia/profile_1/6001(a)192.168.142.12 Command >>> Execute playback(/sounds/ivr/ivr_menu.mp3) >>> EXECUTE sofia/profile_1/6001(a)192.168.142.12 >>> playback(/sounds/ivr/ivr_menu.mp3) >>> [DEBUG] switch_ivr_play_say.c:1467 Codec Activated L16@8000hz 1 >>> channels 20ms >>> [DEBUG] switch_ivr.c:623 sofia/profile_1/6001(a)192.168.142.12 Command >>> Execute playback(/sounds/system/transfer_request.mp3) >>> EXECUTE sofia/profile_1/6001(a)192.168.142.12 >>> playback(/sounds/system/transfer_request.mp3) >>> [DEBUG] switch_ivr_play_say.c:1467 Codec Activated L16@8000hz 1 >>> channels 20ms >>> [DEBUG] switch_ivr.c:623 sofia/profile_1/6001(a)192.168.142.12 Command >>> Execute playback(/sounds/system/chr_1.mp3) >>> EXECUTE sofia/profile_1/6001(a)192.168.142.12 >>> playback(/sounds/system/chr_1.mp3) >>> [DEBUG] switch_ivr_play_say.c:1467 Codec Activated L16@8000hz 1 >>> channels 20ms >>> [DEBUG] switch_ivr.c:623 sofia/profile_1/6001(a)192.168.142.12 Command >>> Execute playback(/sounds/system/chr_2.mp3) >>> EXECUTE sofia/profile_1/6001(a)192.168.142.12 >>> playback(/sounds/system/chr_2.mp3) >>> [DEBUG] switch_ivr_play_say.c:1467 Codec Activated L16@8000hz 1 >>> channels 20ms >>> [DEBUG] switch_ivr.c:623 sofia/profile_1/6001(a)192.168.142.12 Command >>> Execute playback(/sounds/system/chr_3.mp3) >>> EXECUTE sofia/profile_1/6001(a)192.168.142.12 >>> playback(/sounds/system/chr_3.mp3) >>> [DEBUG] switch_ivr_play_say.c:1467 Codec Activated L16@8000hz 1 >>> channels 20ms >>> [DEBUG] switch_ivr_play_say.c:1910 done playing file >>> >>> /sounds/system/chr_3.mp3 >>> [DEBUG] switch_ivr_play_say.c:1910 done playing file >>> /sounds/system/chr_2.mp3 >>> [DEBUG] switch_ivr_play_say.c:1910 done playing file >>> /sounds/system/chr_1.mp3 >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting(a)freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users(a)lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting(a)freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users(a)lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >

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link my freeswitch to external database
by Ahmed Alzaeem 02 Sep '16

02 Sep '16

Dear Dev , I’m newbies to freeswitch & fusionpbx i have : ============== CentOS <http://centos.org/> v7 Freeswitch v1.6 FusionPBX v4 mysql external database to be linked my with fusionpbx . ========== I’m really confused with the option from where do i tell fusionpbx to use the external db ? i already added that db from the gui during the installation but it didn’t work or at least wasn’t chosen to be the db of my freeswitch . what i have modified was the file : /etc/freeswitch/autoload_configs/switch.conf.xml i added line <!-- <param name="core-db-dsn" value="odbc://DRIVER=mysql;SERVER=112.32.33.4;UID=lsfusion;PWD=4424243!;DATABASE=fusionpbx;OPTION=67108864"> —> but seems didn’t work . I’m really don’t know where to tell my server to use the external mysql db hope to help cheers

2 1

Hangup hooks in Python
by Trever L. Adams 02 Sep '16

02 Sep '16

I am working on adding functionality to an external project that already supports Asterisk. I have used hook events with lua to capture call data on hangup and hangup complete before. Unfortunately, I need authenticated json rpc or xmlrpc. The original code I am working from is also in Python. Lua xml and json rpc modules do not seem to do authenticated requests. Since I already have Python, it seems to be easier. https://wiki.freeswitch.org/wiki/Mod_python#Hangup_hook_.2B_SQLAlchemy_cras…, it says that hangup hooks are very buggy and should be avoided. Is this still the case in the last year or so (recent versions)? If so, what is the best way to do this? If not, can someone post a simple equivalent of event:getHeader in a simple handler for hangups so that I can start working on this? Do I use def fsapi(session, stream, env, args) or def handler(session, args)? Thank you for help getting started, Trever

2 3

variable_DIALSTATUS reset by pickup endpoint
by Podrigal, Aron 01 Sep '16

01 Sep '16

Hi, I was relying on *variable_DIALSTATUS* to check for the originate error cause when dialing a user, it works for error cases like RECOVERY_ON_TIMER_EXPIRE, BUSY, however when the cause is USER_NOT_REGISTERED freeswitch overrides the DIALSTATUS with the pickup endpoint's error cause so I'm unable to detect the reason properly. How can I reliably retrieve the originate cause? Is this a bug or intended? 016-09-01 23:17:35.561182 [NOTICE] switch_ivr_originate.c:2759 Cannot create outgoing channel of type [error] cause: [USER_NOT_REGISTERED] 2016-09-01 23:17:35.561182 [DEBUG] mod_dptools.c:3709 () State Change CS_NEW -> CS_ROUTING 2016-09-01 23:17:35.561182 [DEBUG] switch_core_session.c:1397 Send signal N/A [BREAK] 2016-09-01 23:17:35.561182 [NOTICE] switch_channel.c:1077 New Channel pickup/221(a)example.com [446024c6-709a-11e6-a2c0-4d1647ad42e6] 2016-09-01 23:17:35.581187 [NOTICE] switch_ivr_originate.c:794 Hangup pickup/221(a)example.com [CS_ROUTING] [NO_PICKUP] 2016-09-01 23:17:35.581187 [DEBUG] switch_channel.c:3244 Send signal pickup/ 221(a)example.com [KILL] 2016-09-01 23:17:35.581187 [DEBUG] switch_core_session.c:1397 Send signal pickup/221(a)example.com [BREAK] And those are the related variables afterwards. variable_originated_legs: [446024c6-709a-11e6-a2c0-4d1647ad42e6;Outbound Call;221] variable_last_bridge_hangup_cause: [NO_PICKUP] variable_originate_causes: [446024c6-709a-11e6-a2c0-4d1647ad42e6;NO_PICKUP] variable_originate_disposition: [NO_PICKUP] variable_DIALSTATUS: [NO_PICKUP] Thanks -- Aron Podrigal - '1000001', '1110010', '1101111', '1101110' '1010000', '1101111', '1100100', '1110010', '1101001', '1100111', '1100001', '1101100' P: '2b', '31', '33', '34', '37', '34', '35', '38', '36', '30', '39', '39'

1 0