July 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Implementing <Form> Tag of VXML in Freeswtich using Javascript
by david.fu＠oocl.com 28 Jul '15

28 Jul '15

Dear Freeswitch experts, I would like to implement something like <Form> and <submit> tag of VXML in Freeswtich using Javascript. The purpose is to sending request back to the application server from Freeswtich. However, I couldn't find the related API(s) in the Freeswitch official web site. Would you please help advise ? Thank you so much. Yours faithfully, David IMPORTANT NOTICE Email from OOCL is confidential and may be legally privileged. If it is not intended for you, please delete it immediately unread. The internet cannot guarantee that this communication is free of viruses, interception or interference and anyone who communicates with us by email is taken to accept the risks in doing so. Without limitation, OOCL and its affiliates accept no liability whatsoever and howsoever arising in connection with the use of this email. Under no circumstances shall this email constitute a binding agreement to carry or for provision of carriage services by OOCL, which is subject to the availability of carrier's equipment and vessels and the terms and conditions of OOCL's standard bill of lading which is also available at http://www.oocl.com.

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how to add mod_event_socket and mod_conference activities to the logfile
by Markus von Arx 28 Jul '15

28 Jul '15

Hallo I'd like to see more details about what's going on in the mod_event_socket and in the mod_conference modules in the FreeSWITCH log files. With the default (vanilla) configuration, nothing gets logged to the freeswitch.log logfile from inside these modules. I scanned all config files and the FS confluence wiki, but could not find any hint on how this can be achieved. The information I'd like to see in the freeswitch.log logfile is e.g. the following: * state changes in the mod_conference conferences, e.g. conference created, conference destroyed, member added, member removed, member muted, member unmuted, ... (including affected conference name and member id of course) * requests, replies & events in the mod_event_socket - if possible "shortened" and not including the whole event texts Any hints how this can be configured? Thanks a lot, Markus

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FreeSWITCH Week in Review (Master Branch) July 18th-July 24th
by Ken Rice 28 Jul '15

28 Jul '15

New Post on freeswitch.org from Kathleen King check it out at http://ift.tt/1U48OzU FreeSWITCH Week in Review (Master Branch) July 18th-July 24th Hello, again. This passed week in the FreeSWITCH master branch we had 46 commits. The new features this week are: the addition of getcputime to retrieve FreeSWITCH process CPU usage, added support for 80 ms, 100 ms, 120 ms packetization to mod_opus, and added H.263 codec support to mod_av. Join us on Wednesdays at 12:00 CT for some more FreeSWITCH fun! And head over to freeswitch.com to learn more about FreeSWITCH support. New features that were added: FS-7848 [mod_opus] Add support for 80 ms, 100 ms, 120 ms packetization FS-7519 FS-7677 [mod_av] Add H.263 codec support FS-7885 Add getcputime to retrieve FreeSWITCH process CPU usage FS-7889 [mod_conference] Move conference chat to use an event channel so messages only go to the right ‘room’ for the conference and move conference chat functionality to use event_channel. Improvements in build system, cross platform support, and packaging: FS-7860 Prevent a switch_rtp header conflict FS-7130 Make /run/freeswitch persistent, so it will start under systemd The following bugs were squashed: FS-7789 [mod_av] Fixed issue with audio dropping out partway through recordings FS-7854 Add task_runtime to tasks table in core database FS-7856 [mod_av] Fix some segfaults and leaks. FS-7866 Fixed a crash when running incorrect var api expansion syntax “eval ${${external_sip_ip}:4}” FS-7861 FS-7862 [mod_conference] Fixed a crash and other issues caused by multi canvas feature FS-7681 [mod_conference] Factor out conference->canvas and allow per canvas record and play FS-7869 [mod_conference] Fixed a deadlock on shutdown after playing video file that will not display video FS-7654 Fixed regressions on eavesdropping on channels playing a file and on channels with unlike rates FS-7872 [mod_verto] Gracefully fail attempting to transfer 1 legged call FS-7874 [mod_conference] Fixed incorrect layout group count FS-7870 [mod_conference] Allow jsonapi commands to pass the string id field to pass special ID’s like “last” FS-7882 [mod_conference] Allow JSON API commands to send third arg for muting FS-7888 [mod_verto] Fixed namespacing problems in javascript library masked by global verto object FS-7811 Use more common format CIF for blank image FS-7902 [mod_local_stream] Fix for queue filling up when you have a mix of video and non video files FS-7891 [mod_spandsp] Allow spandsp dtmf detector to work on rates other than 8k FS-7839 Correct firefox > 38 DTLS behavior to match new EC requirements

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IP-Based Authentication
by Murugan Pandian 28 Jul '15

28 Jul '15

Hi, How i auth agent based on IP Address

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How can I have FS participate in a shared UDP multicast session/conference?
by D D 28 Jul '15

28 Jul '15

Hi, I want to have FS participate in a multicast session with other members. I see there is the esf_page_group application available, but I think it will constantly send rtp and therefor will monopolize the port.I think I would need to use VAD my call into esf_page_group. Is this the right approach? If so, how can I use VAD? Thanks,David

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Freetdm configuration
by Nikolay Zaytsev 28 Jul '15

28 Jul '15

Good day to you. I have a problem with the setting up of the freetdm module. I have sangoma a102de card. The system recognizes the card correctly. Audiocodes mediant 600 with the same trunk configuration works corrctly. But during the startup freeswitch writes [DEBUG] ftdm_io.c:5469 Module sangoma_isdn does not support configuration. and after that [INFO] ftmod_sangoma_isdn_stack_rcv.c:1098 sng_isdn->s2: Link is down, dropping transmit frame. Config files are attached. Please, help me:) Best regards, Nikolay Zaytsev

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audio Handshake failure 1
by Alex Polischuk 28 Jul '15

28 Jul '15

Hi All, I use FreeSwitch for transcoding SRTP from WebRTC to SIP with RTP and vice versa. When Firefox version 38 add requires for Perfect Forward Secrecy (PFS) I have problem with Handshake in FS when Firefox Browser terminate a call. I found that that problem was fixed in the version 1.4.19 https://freeswitch.org/jira/browse/FS-7425 I have the same problem in the version 1.4.20 eff635a6-33a0-11e5-ac73-891cb7dea826 2015-07-26 14:16:46.564182 [INFO] switch_rtp.c:2924 Changing audio DTLS state from HANDSHAKE to SETUP eff635a6-33a0-11e5-ac73-891cb7dea826 2015-07-26 14:16:46.564182 [INFO] switch_rtp.c:2832 audio Fingerprint Verified. eff635a6-33a0-11e5-ac73-891cb7dea826 2015-07-26 14:16:46.564182 [INFO] switch_rtp.c:3384 Activating Audio Secure RTP SEND eff635a6-33a0-11e5-ac73-891cb7dea826 2015-07-26 14:16:46.564182 [INFO] switch_rtp.c:3362 Activating Audio Secure RTP RECV eff635a6-33a0-11e5-ac73-891cb7dea826 2015-07-26 14:16:46.564182 [INFO] switch_rtp.c:2872 Changing audio DTLS state from SETUP to READY eff635a6-33a0-11e5-ac73-891cb7dea826 2015-07-26 14:16:46.604193 [DEBUG] switch_rtp.c:1937 rtcp_stats_init: ssrc[-1374535478] base_seq[6179] f01c46d8-33a0-11e5-ac83-891cb7dea826 2015-07-26 14:16:46.644210 [DEBUG] switch_rtp.c:5884 Correct ip/port confirmed. f02184f4-33a0-11e5-ac91-891cb7dea826 2015-07-26 14:16:47.264283 [ERR] switch_rtp.c:2917 audio Handshake failure 1 f02184f4-33a0-11e5-ac91-891cb7dea826 2015-07-26 14:16:47.264283 [INFO] switch_rtp.c:2918 Changing audio DTLS state from HANDSHAKE to FAIL f02184f4-33a0-11e5-ac91-891cb7dea826 2015-07-26 14:16:47.264283 [NOTICE] switch_rtp.c:2899 Hangup sofia/external/992555500004(a)voip.webrtc.jajah.com [CS_EXCHANGE_MEDIA] [DESTINATION_OUT_OF_ORDER] what additional definition I should make to resolve this problem? Thanks, Alex Polischuk

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Incorrect Caller-Context in the events
by Dmitri Muntean 28 Jul '15

28 Jul '15

Hey guys, I'm running freeswitch 1.4.20 and 1.4.13 with the number of conferences setup on them and have 3 profiles and contexts defined, none of those are called "default". I read the events via inbound event socket, and most of the events happening inside those conferences have correct Caller-Context set to the profile that the call came through. But sometimes (very rarely) the Caller-Context is set to "default" - I was wondering if anyone seen such behaviour. I don't have "default" profile or context defined anywhere in my configuration. The events I am reading off freeswitch are Event-Name: DTMF and Event-Name: CUSTOM with Event-Subclass: conference::maintenance. When the Caller-Context is "default" it appears so in both type of events. Thanks

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Should vpn address space be defined as part of local network?
by Rajil Saraswat 28 Jul '15

28 Jul '15

Hello all, I am trying to get my head around the nat.auto and localnet.auto acls. I have a VPN server using the 10.8.0.0/24 address space with gateway on 10.8.0.1. The PBX is on the local lan (172.16.5.0/24) with ip 172.16.5.5. When freeswitch starts i see it builds the following acls nat.auto Created ip list nat.auto default (deny) Adding 172.16.5.5/255.255.255.0 (deny) to list nat.auto Adding 10.0.0.0/8 (allow) [] to list nat.auto Adding 172.16.0.0/12 (allow) [] to list nat.auto localnet.auto Created ip list localnet.auto default (deny) Adding 172.16.5.5/255.255.255.0 (allow) to list localnet.auto Do i need to move my vpn address space (10.8.0.0/16) from nat.auto to the localnet.auto so that it not natted? Something like this: nat.auto 172.16.5.5/255.255.255.0 (deny) 10.0.0.0/8 (allow) 172.16.0.0/12 (allow) 10.8.0.0/16 (deny) localnet.auto 172.16.5.5/255.255.255.0 (allow) 10.8.0.0/16 (allow) Thanks

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Fwd:
by Nikolay Zaytsev 27 Jul '15

27 Jul '15

Good day to you. I have a problem with the setting up of the freetdm module. I have sangoma a102de card. The system recognizes the card correctly. Audiocodes mediant 600 with the same trunk configuration works corrctly. But during the startup freeswitch writes [DEBUG] ftdm_io.c:5469 Module sangoma_isdn does not support configuration. and after that [INFO] ftmod_sangoma_isdn_stack_rcv.c:1098 sng_isdn->s2: Link is down, dropping transmit frame. Config files are attached. Please, help me:) Best regards, Nikolay Zaytsev

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