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Presence question
by Kamil Nigmatullin 21 Nov '14

21 Nov '14

One of our clients wants to switch on presence feature, but they have 140 lines of FXS that does not support Publish/Subscribe function. Is there a way to achive this using data in freeswitch (if Registered or Busy)? I know that it is a silly question but they asked me to ask anyway, so will anybody just confirm that it is not possible? -- Kamil Nigmatullin Skype: kamil.nigmatullin

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Conference recording
by Oleg Stolyar 21 Nov '14

21 Nov '14

Hi, I am planing to record conference using the auto-record param. My question is this: If the machine runs out of disk space, will FS just stop recording or will bad things like crash happen? Same question for logging. Thank you

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mod_conference stereo? record stereo wav?
by Stephen Dame 20 Nov '14

20 Nov '14

Question, we are successfully connecting chrome/firefox with opus48, 2 channel stereo to mod_conference user master build. Recording the conference seems to only record 1 channel Couple questions, did not see any documentation on stereo in wiki. 1) Do we need to add a param to conference.conf.xml profile to have the conference mix stereo? 2) What is the proper way to record the conference in stereo wav file? Thanks in advance. Regards, Stephen

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Re: [Freeswitch-users] mod_conference stereo? record stereo wav? (Stephen Dame)
by Danny Gershman 20 Nov '14

20 Nov '14

I'm interested in the same thing as well.

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Opus Codec Error
by David Shemesh 20 Nov '14

20 Nov '14

Hope someone can help with this: I am experiencing a strange issue using Opus. The call is from a registered client to a landline through a sip Trunk. Freeswitch is transcoding From Opus to PCMU and after about ~1min the call drops with "switch_core_io.c:1282 Codec OPUS (STANDARD) decoder error!" and the call gets disconnected. It looks like this happens only when i enabling the jitterbuffer. I was able to reproduce this using various Clients (Csipsimple, Jitsi) & Sip trunks. it seems like its has something to do with freeswitch's implementation of OPUS. FreeSWITCH Version 1.5.15b+git~20141112T004420Z~dd629c1516~64bit Dialplan: <extension name="out" > <condition field="destination_number" expression="^9(\d+)$" > <action application="set" data="rtp_jitter_buffer_during_bridge=true" /> <action application="set" data="rtp_jitter_buffer_plc=true" /> <action application="set" data="jitterbuffer_msec=60:200:20" /> <action application="set" data="call_direction=outbound" /> <action application="set" data="hangup_after_bridge=true" /> <action application="set" data="effective_caller_id_name=${outbound_caller_id_name}" /> <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}" /> <action application="set" data="inherit_codec=true" /> <action application="set" data="continue_on_fail=true" /> <action application="bridge" data="sofia/gateway/out01/$1" /> <action application="bridge" data="sofia/gateway/ou02/$1" /> </condition> </extension> Freeswitch Log: 2014-11-18 10:51:25.876680 [DEBUG] switch_core_media.c:3389 Audio Codec Compare [PCMU:0:8000:10:64000]/[opus:116:48000:20:0] 2014-11-18 10:51:25.876680 [DEBUG] switch_core_media.c:3389 Audio Codec Compare [PCMU:0:8000:10:64000]/[PCMU:0:8000:20:64000] 2014-11-18 10:51:25.876680 [DEBUG] switch_core_media.c:3424 Audio Codec Compare [PCMU:0:8000:20:64000] is saved as a near-match 2014-11-18 10:51:25.876680 [DEBUG] switch_core_media.c:3389 Audio Codec Compare [PCMU:0:8000:10:64000]/[GSM:3:8000:20:13200] 2014-11-18 10:51:25.876680 [DEBUG] switch_core_media.c:3389 Audio Codec Compare [PCMU:0:8000:10:64000]/[PCMA:8:8000:20:64000] 2014-11-18 10:51:25.876680 [DEBUG] switch_core_media.c:3389 Audio Codec Compare [PCMU:0:8000:10:64000]/[G722:9:8000:20:64000] 2014-11-18 10:51:25.876680 [DEBUG] switch_core_media.c:3389 Audio Codec Compare [PCMU:0:8000:10:64000]/[AMR:96:8000:20:12200] 2014-11-18 10:51:25.876680 [DEBUG] switch_core_media.c:3389 Audio Codec Compare [PCMU:0:8000:10:64000]/[iLBC:97:8000:30:13330] 2014-11-18 10:51:25.876680 [DEBUG] switch_core_media.c:3389 Audio Codec Compare [PCMU:0:8000:10:64000]/[SPEEX:99:8000:20:24600] 2014-11-18 10:51:25.876680 [DEBUG] switch_core_media.c:3389 Audio Codec Compare [PCMU:0:8000:10:64000]/[SPEEX:99:16000:20:42200] 2014-11-18 10:51:25.876680 [DEBUG] switch_core_media.c:3389 Audio Codec Compare [PCMU:0:8000:10:64000]/[SPEEX:99:32000:20:44000] 2014-11-18 10:51:25.876680 [DEBUG] switch_core_media.c:3315 Set telephone-event payload to 101 2014-11-18 10:51:25.876680 [DEBUG] switch_core_media.c:3494 Substituting codec PCMU@10i@8000h 2014-11-18 10:51:25.876680 [DEBUG] switch_core_media.c:2343 Set Codec sofia/external/123456789 PCMU/8000 10 ms 80 samples 64000 bits 2014-11-18 10:51:25.876680 [DEBUG] switch_core_codec.c:111 sofia/external/123456789 Original read codec set to PCMU:0 2014-11-18 10:51:25.876680 [DEBUG] switch_core_media.c:3623 Set 2833 dtmf send payload to 101 2014-11-18 10:51:25.876680 [DEBUG] switch_core_media.c:4847 AUDIO RTP [sofia/external/123456789] 10.0.50.160 port 23644 -> 212.179.176.54 port 60036 codec: 0 ms: 10 2014-11-18 10:51:25.876680 [DEBUG] switch_rtp.c:3334 Starting timer [soft] 80 bytes per 10ms 2014-11-18 10:51:25.876680 [DEBUG] switch_core_media.c:5194 Set 2833 dtmf send payload to 101 2014-11-18 10:51:25.876680 [DEBUG] switch_core_media.c:5200 Set 2833 dtmf receive payload to 101 2014-11-18 10:51:25.876680 [DEBUG] switch_core_media.c:5228 Set comfort noise payload to 13 2014-11-18 10:51:25.876680 [NOTICE] sofia_media.c:92 Pre-Answer sofia/external/123456789! 2014-11-18 10:51:25.876680 [DEBUG] switch_channel.c:3389 Send signal sofia/internal/1000(a)192.168.10.150 [BREAK] 2014-11-18 10:51:25.876680 [DEBUG] switch_channel.c:3393 (sofia/external/123456789) Callstate Change RINGING -> EARLY 2014-11-18 10:51:25.896682 [DEBUG] switch_core_codec.c:246 sofia/internal/1000(a)192.168.10.150 Restore previous codec opus:116. 2014-11-18 10:51:25.896682 [DEBUG] switch_ivr_originate.c:3551 Originate Resulted in Success: [sofia/external/123456789] 2014-11-18 10:51:25.896682 [DEBUG] switch_core_session.c:907 Send signal sofia/external/123456789 [BREAK] 2014-11-18 10:51:25.896682 [DEBUG] switch_core_session.c:907 Send signal sofia/internal/1000(a)192.168.10.150 [BREAK] 2014-11-18 10:51:25.896682 [DEBUG] switch_ivr_bridge.c:1460 (sofia/external/123456789) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA 2014-11-18 10:51:25.896682 [DEBUG] switch_core_session.c:1387 Send signal sofia/external/123456789 [BREAK] 2014-11-18 10:51:25.896682 [DEBUG] switch_core_state_machine.c:467 (sofia/external/123456789) Running State Change CS_EXCHANGE_MEDIA 2014-11-18 10:51:25.896682 [DEBUG] switch_core_state_machine.c:533 (sofia/external/123456789) State EXCHANGE_MEDIA 2014-11-18 10:51:25.896682 [DEBUG] mod_sofia.c:592 SOFIA EXCHANGE_MEDIA 2014-11-18 10:51:26.056685 [DEBUG] switch_rtp.c:5555 Correct ip/port confirmed. 2014-11-18 10:51:26.056685 [NOTICE] switch_core_io.c:1230 Activating write resampler 2014-11-18 10:51:26.056685 [DEBUG] switch_core_io.c:1458 Engaging Write Buffer at 160 bytes to accommodate 320->160 2014-11-18 10:51:26.136684 [ERR] switch_core_io.c:1282 Codec OPUS (STANDARD) decoder error! 2014-11-18 10:51:26.136684 [DEBUG] switch_ivr_bridge.c:578 sofia/external/123456789 ending bridge by request from write function 2014-11-18 10:51:26.136684 [DEBUG] switch_ivr_bridge.c:659 BRIDGE THREAD DONE [sofia/internal/1000(a)192.168.10.150] 2014-11-18 10:51:26.136684 [DEBUG] switch_ivr_bridge.c:689 Send signal sofia/external/123456789 [BREAK] 2014-11-18 10:51:26.136684 [DEBUG] switch_ivr_bridge.c:659 BRIDGE THREAD DONE [sofia/external/123456789] 2014-11-18 10:51:26.136684 [DEBUG] switch_ivr_bridge.c:689 Send signal sofia/internal/1000(a)192.168.10.150 [BREAK] 2014-11-18 10:51:26.136684 [NOTICE] switch_ivr_bridge.c:751 Hangup sofia/external/123456789 [CS_EXCHANGE_MEDIA] [ORIGINATOR_CANCEL] 2014-11-18 10:51:26.136684 [DEBUG] switch_channel.c:3215 Send signal sofia/external/123456789 [KILL] 2014-11-18 10:51:26.136684 [DEBUG] switch_core_session.c:1387 Send signal sofia/external/123456789 [BREAK] 2014-11-18 10:51:26.136684 [DEBUG] switch_core_state_machine.c:533 (sofia/external/123456789) State EXCHANGE_MEDIA going to sleep 2014-11-18 10:51:26.136684 [DEBUG] switch_core_state_machine.c:467 (sofia/external/123456789) Running State Change CS_HANGUP 2014-11-18 10:51:26.136684 [DEBUG] switch_core_state_machine.c:730 (sofia/external/123456789) Callstate Change EARLY -> HANGUP 2014-11-18 10:51:26.136684 [DEBUG] switch_core_state_machine.c:732 (sofia/external/123456789) State HANGUP 2014-11-18 10:51:26.136684 [DEBUG] mod_sofia.c:413 Channel sofia/external/123456789 hanging up, cause: ORIGINATOR_CANCEL Anyone have any idea whats going on ? Thanks David

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How i can disable tcp port in freeswitch.?
by Aqs Younas 20 Nov '14

20 Nov '14

Hi, Who can i disable tcp ports in freeswitch?

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problem with eavesdrop() sending null RTP for 15-30 seconds
by Nathan Neulinger 20 Nov '14

20 Nov '14

Seeing an interesting issue with sending an eavesdrop where the outbound call gets established, but no useful audio is transmitted until around 15-30 seconds into the call. Prior to that point, FS is transmitting all identical RTP frames with the payload containing: 24 1c 61 7a 7f 55 55 55 50 06 83 ff ff ff ff ff ff ff ff ff ff ff ff aa aa aa aa aa aa aa aa aa aa aa aa aa aa aa aa aa aa aa aa aa aa af ff ..... ff This is minimal version of command that was being issued: originate sofia/internal/6679(a)voice-dev.mst.edu &eavesdrop(ca887b3b-013f-4972-aa7c-37a7b132bf52) Reproduced with head of master as of a few minutes ago. Restarting FS and trying again, the FIRST attempted eavesdrop seems to not have the long delay, but the second one does. The 'dummy' payload appears to have the same content in subsequent requests. Interestingly, if I let the outbound call ring longer before I pick up, the delay appears to be reduced by the period of time I let the phone ring for. if I pick up immediately, 25-30 second audio delay approx, if I pick up after 25 seconds of ringing, I get only a couple second delay. Any idea where to start looking for where this buffering might be getting introduced? For the "only eavesdrop" case it's not as big of an issue, but when you switch over to barge, can't really have multi-second audio delay. -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul(a)mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect

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Trying to originate a call using ES.
by Pavel 20 Nov '14

20 Nov '14

Hello Everybody! I'm stuck trying to originate a call using event socket. Somewhere on the internet I've found some Perl script involving ESL. So I changed it a bit so I get an incoming call to outbound socket, look for destination number and trying to set up outgoing call using originate like so: $con->sendRecv("api originate user/1000 &park()") This works. The question is how can I control a unanswered timeout of this originate, say to playback some message to initial caller after 5 seconds ? I've tried setting the following variables: originate_timeout=5 leg_timeout=5 call_timeout=5 like so: $con->sendRecv("api originate {originate_timeout=5}user/1000 &park()") None of this working. The originated call keeps ringing. Please advise. Thanks. Regards, Pavel.

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Moving Voicemails
by Joel White 20 Nov '14

20 Nov '14

I initially configured my FreeSWITCH with an IP address, but now for scalability and high availability I am changing to a domain name. I successfully changed the system over, but was wondering how do I move the voicemails to the new domain folder under voicemail? The obvious answer is to move the files, I already did that and restarted the system. The existing voicemails and greetings are not visible to FreeSWITCH when I call voicemail. How do I transfer them over? Thank you in advance for any light you may shed on this predicament

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Can't INVITE message to jitsi if length effective_caller_id_name >8
by Nguyễn Văn Nghĩa Em 20 Nov '14

20 Nov '14

Hello, I have one problem of make call to sip exension which registered on jitsi. In dialplan if I set effective_caller_id_name a string value which its length > 8, the sip endpoint which use jitsi can't receive the call. But if I register that extension to other phones devices (AudioCodes 310HD, CSipSimple), these endpoint will receive the call (i used sofia_contact to ring multiple sip endpoint simultaneously). If the length of effective_caller_id_name <=8, jitsi sip endpoint will receive the call. I have capture sip messages on computer which install jitsi, I find that there are no INVITE message. But when caller cancel the call, I saw CANCEL message on this pc. I don't know that is the problem? Please help me. My FS version is 1.4.6 which run on RedHat 6 64 bits OS Thanks. -- Best regards, ------------------------------------------------------------------------------------------------------------- EM NGUYEN Mobile: +84 949669075 Yahoo & Skype: nghiaembt Website: http://freeswitch.vn --------------------------------------------------------------------------------------------------------------

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