August 2016 - FreeSWITCH-users - FreeSWITCH Mailing Lists

play_and_get_digits questions
by Anonim Stefan 02 Aug '16

02 Aug '16

Hi, While in a group call, I'm using "play_and_get_digits" to dial out a number as shown in example [1], triggered via a DTMF key: "<control action="execute_application" digits="*1" data="execute_extension ASK_FOR_NUMBER_447279569204 XML default"/>" The issues I'm facing is that when the user forms the number, DTMF caller controls get triggered(like mute, vol up, etc..) as defined in conference.conf.xml. 1. Is there a way to not trigger those DTMF caller controls while forming the number when using play_and_get_digits? 2. Is there a way to get some details in the ASK_FOR_NUMBER_(\d+) extension, about the one that triggered the DTMF? (i.e. something like user_name or user_number) Thanks, Stefan [1] https://wiki.freeswitch.org/wiki/Conference_Add_Call_Example

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save voicemail via http
by jungle Boogie 02 Aug '16

02 Aug '16

Hi All, Has anyone successfully saved voicemail off of their freeswitch install using http? I know you can mount a nfs share and save the files the share. Is that what people usually do? If not, how do folks make voicemail accessible when using a backup freeswitch server? Is something like couchDB used? Does the voicemail code support http put/post methods? Thanks! -- ------- inum: 883510009027723 sip: jungleboogie(a)sip2sip.info

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Custom text in hangup code
by Yu Boot 02 Aug '16

02 Aug '16

How to change default text in hangup SIP messages? For example I want hangup with "503 Call limit exceeded" instead of just "503 Service unavailable" for call limit.

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rtcp-mux on B-leg but there is no signaling in SDP
by Richard Chan 02 Aug '16

02 Aug '16

Hi list, FreeSWITCH 1.6.9 and the default configurations: FreeSWITHCH is using rtcp-mux on the B-leg without signalling in SDP (i.e. no rtcp-mux attribute). If I change rtcp_audio_interval_msec to passthru or say 2000 then there is explicit rtcp-mux in the INVITE SDP. The UAC I'm using, sends a few packets to RTP+1, FreeSWITCH sends RTCP with source port the same as RTP, and the UAC auto-adjusts after 10 or so packets. This UAC always includes an a=rtcp line even if the RTCP port is RTP+1. Is this a "bug"? I would expect FreeSWITCH to signal rtcp-mux on the B-leg. (On the A-leg, FreeSWITCH doesn't use rtcp-mux and includes an explicit rtcp port in SDP). -- Richard Chan

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random delay after correct group_confirm_key
by Benedikt Fraunhofer 02 Aug '16

02 Aug '16

Hello Everyone, we're expriencing random delay after a correct group_confirm_key press with 1.6.9-16-d574870~64bit ac20683c 21:12:58.321694 [DEBUG] mod_dptools.c:1519 SET sofia/external/+XXX [group_confirm_key]=[1] ac20683c 21:12:58.321694 [DEBUG] mod_dptools.c:1519 SET sofia/external/+XXX [group_confirm_read_timeout]=[1000] ac20683c 21:12:58.341607 [DEBUG] mod_dptools.c:1519 SET sofia/external/+XXX [group_confirm_file]=[/etc/freeswitch-qa//prompts/new-layout/de/press-one.wav] [...] f4a97f39 21:13:00.281608 [DEBUG] switch_ivr_play_say.c:1467 Codec Activated L16@8000hz 1 channels 20ms f4a97f39 21:13:00.341654 [DEBUG] switch_rtp.c:6711 Correct audio ip/port confirmed. f4a97f39 21:13:01.301658 [DEBUG] switch_ivr_play_say.c:1910 done playing file /etc/freeswitch-qa//prompts/new-layout/de/press-one.wav f4a97f39 21:13:01.301658 [DEBUG] switch_ivr.c:1325 Codec Activated L16@8000hz 1 channels 20ms [...] f4a97f39 21:13:05.341669 [DEBUG] switch_ivr_play_say.c:1910 done playing file /etc/freeswitch-qa//prompts/new-layout/de/press-one.wav f4a97f39 21:13:05.341669 [DEBUG] switch_ivr.c:1325 Codec Activated L16@8000hz 1 channels 20ms f4a97f39 21:13:06.241632 [DEBUG] switch_rtp.c:6931 RTP RECV DTMF 1:800 f4a97f39 21:13:06.241632 [INFO] switch_channel.c:515 RECV DTMF 1:800 ac20683c 21:13:10.281620 [DEBUG] switch_core_codec.c:246 sofia/external/+XXX Restore previous codec PCMA:8. note the 4s delay after the "1" has been recognized before it bridges the legs. The delay is quite random between one and 4 seconds. I already played around with group_confirm_read_timeout and replaced the prompt with a shorter version because initially I thought it was always continuing to play the prompt to it's end before proceeding but that's not true. Any tips and/or is this worth filing a bug? Thx in advance Benedikt

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What kind of event do we get on event socket upon silence detection
by Naveen Tamanam 01 Aug '16

01 Aug '16

Hi, What kind of event do we get on event socket upon silence detection. What dialplan application I can use for silence detection ? I found wait_for_silence Is there any other dialplan application which is helpful for silence detection ? If silence is detected what event can be expected to receive on event socket ? -- Thanks & Regards, Naveen Tamanam

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How to detect North America RING_TONE using spandsp
by Naveen Tamanam 01 Aug '16

01 Aug '16

Hi, I'm trying to detect North America RING_TONE using spandsp Using default settings as mentioned follows <tone name="RING_TONE" description="North America ring"> <element freq1="440" freq2="480" min="1200" max="0"/> </tone> But I'm not getting consistent tone detections. Is there any thing that I need to change to have a proper tone detection. As per the tone/spectrum analysis using audacity. This tone is having two frequencies 440 and 480 a mentioned in conf, but spandsp is not detecting them. To have a proper and consistent tone detection how much duration I should use? Please refer spectrum analysis of tone representing 440Hz & 480Hz frequencies. As per the doc min and max values represents duration. How to determine these durations of any give tone? What is the meaning of having max value 0 ? -- Thanks & Regards, Naveen Tamanam

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Typo in Lua API confluence page
by Joel Serrano 01 Aug '16

01 Aug '16

Hi guys, There is a little typo in this page: https://freeswitch.org/confluence/display/FREESWITCH/Lua+API+Reference session:execute session:execute(app, data) local mySound = "/usr/local/freeswitch/sounds/music/16000/partita-no-3-in-e-major-bwv-1006-1-preludio.wav" session:execute("playback", mySound) NOTE: Callbacks (DTMF and friends) CAN NOT EXECUTE during an execute. session:executeString session:execute(api_string) *<<<< Shouldn't this be executeString???* local mySound = "/usr/local/freeswitch/sounds/music/16000/partita-no-3-in-e-major-bwv-1006-1-preludio.wav" session:executeString("playback "..mySound) Note: Callbacks (DTMF and friends) CAN NOT EXECUTE during an execute. Can anybody update it please? Can lead to confusion. Thanks! Joel.

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Disable IPV6
by Agustí Ubalde 01 Aug '16

01 Aug '16

Hi all, Anyone knows if is possible to disable ipv6 from configuration files? Thanks, *PRESENCE TECHNOLOGY* *Agustí Ubalde Bellot* Chief Developer C/ Comte Urgell 240 3A Barcelona 08036 aubalde(a)presenceco.com Ph: +34 93 10 10 300 Fx: +34 93 10 10 333 *www.presenceco.com* <http://www.presenceco.com/> *Follow us on:* *[image: tw]* <http://twitter.com/presencetech>*[image: yt]* <http://www.youtube.com/PresenceTechnology>*[image: in]* <http://www.linkedin.com/groups?mostPopular=&gid=2898558>*[image: ss]* <http://www.slideshare.net/PresenceTech>*[image: fb]* <http://www.facebook.com/presencetech> For additional information, please visit our website *www.presenceco.com* <http://www.presenceco.com/> -- *Presence Technology - DisclaimerThis message, its content and any file attached thereto is for the intended recipient only and is confidential and /or privileged. If you have received this e-mail in error or had access to it, you should note that the information in it is private and any use thereof is unauthorized. In such an event please notify us by e-mail or by telephone (+ 34 93 10 10 300). Any reproduction of this e-mail by whatsoever means and any transmission or dissemination thereof to other persons is prohibited. It should be deleted immediately from your system. Presence Technology reserves the right to take legal action against any persons unlawfully gaining access to the content of any external message it has emitted.* *For additional information, please visit our website **www.presenceco.com <http://www.presenceco.com/>*

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freeswitch wan.auto ipv6 problem
by Sangram Rath 01 Aug '16

01 Aug '16

Hi Group, We have an issue here where we are not able to make inbound/outbound calls using a websocket application that uses sipjs. We think that wan.auto is not excluding IPv6. This problem is related to ICE. There was a bug on this that was fixed in 1.4.19. Additional info: - The platform is Kazoo - FS version: 1.4.23 (also tried upgrading to 1.4.26) - IPv6 is disabled in the OS We have the below in logs: [WARNING] switch_core_media.c:2790 NO candidate ACL defined, Defaulting to wan.auto [DEBUG] switch_core_media.c:5178 AUDIO RTP [sofia/sipinterface_1/ DPRD025488(a)somedomain.com] x.x.x.x port 24092 -> 2001::5ef5:79fd:207f:3d43:3f57:fe92 port 61975 codec: 8 ms: 20 [DEBUG] switch_rtp.c:3588 Not using a timer [INFO] switch_core_media.c:5352 Activating Audio ICE [NOTICE] switch_rtp.c:4029 Activating RTP audio ICE: bveNJceaOHdVG0Tq:tThuJX7NS3O8dtZ1 2001::5ef5:79fd:207f:3d43:3f57:fe92:61975 [INFO] switch_core_media.c:5395 Activating RTCP PORT 61975 [DEBUG] switch_rtp.c:3929 RTCP send rate is: 10000 and packet rate is: 20000 Remote Port: 61975 [DEBUG] switch_rtp.c:2349 Setting RTCP remote addr to :61975 [INFO] switch_core_media.c:5403 Skipping RTCP ICE (Same as RTP) [INFO] switch_rtp.c:3104 Activate RTP/RTCP audio DTLS client It would be great to have ideas on what could be the reason. Thanks, Sangram

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