- FreeSWITCH-users - FreeSWITCH Mailing Lists

sip trunk with edgemarc
by Joli Martinez 09 Jul '15

09 Jul '15

Hello, I have an Edgemarc as an SBC and behind it is a Freeswitch pbx. I have configured the edgemarc as scenario 1 on the link below. So only does topology hiding but Freeswitch does the registration. On the sip-profile external I have my xml file which points to the LAN side of the Edgemarc. I am unable to register to the provider I get 503 service unvail. https://edgewaternetworks.com/wp-content/uploads/2014/07/Edgewater_Configur… Please help.

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Sangoma's FreeTDM installation and setup on CentOS 6.6 with FreeSWITCH 1.4.20
by Andrew Keil 09 Jul '15

09 Jul '15

To all, There are some important bug fixes in Sangoma's FreeTDM and everyone should use the Sangoma updated version. In order to fast-track the changes I have created a tar.gz file on my FTP site and a PDF document (see attached) to setup the latest production release of FreeSWITCH (version 1.4.20) on CentOS 6.6 with the Sangoma updated FreeTDM 1.9.0. It may help anyone using FreeSWITCH with Sangoma hardware for the first time. Sangoma hope to merge FreeTDM (with the fixes) with FreeSWITCH in August/September this year. I will attempt to keep the PDF and FreeTDM tar.gz file up-to-date alongside future production releases of FreeSWITCH. FYI: Sangoma support also have a copy of this PDF file and they are thankful for the efforts I have made to fast-track new FreeSWITCH users to use their hardware. As with any instruction document, Visytel Pty Ltd accepts no liability or loss of earnings from following the attached PDF. Kind Regards, Andrew Keil Visytel Pty Ltd

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park_after_bridge still broken in 1.4.20
by Ali Pey 08 Jul '15

08 Jul '15

Hello, A fix went in 1.4.20 for this: - FS-7636 <https://freeswitch.org/jira/browse/FS-7636> Fixed an issue with transfer_after_bridge and park_after_bridge pre-empting transfers However, park after bridge still doesn't work in 1.4.20. It only works if I downgrade to 1.4.15. Is there a bug open for this? Should I open a new bug or re-open FS-7636? Regards, Ali Pey

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no audio with bypass_media_after_bridge
by stefano 08 Jul '15

08 Jul '15

Hello, I'm trying to use "bypass_media_after_bridge" in my dialplan. In this scenario I have a) softphone client using g729 b) FS version: FreeSWITCH Version 1.5.13b~64bit ( 64bit) c) Dialogic HMP 3.0 (g729 + g711) I'd like to use FS in passthrough mode so calls from a) will be rerouted to c). If I use the "bypass_media_after_bridge" signalling of calls is ok but i have no audio on a). If I use "bypass_media" audio is ok but I don't have the answer 200 OK back to my softphone a) This is my dialplan: <extension name="calls_to_ivr"> <condition field="destination_number" expression="^9999"> <action application="export" data="jitterbuffer_msec=60:200:40"/> <action application="set" data="bypass_media_after_bridge=true"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="ignore_early_media=true"/> <action application="set" data="effective_caller_id_number=1234"/> <action application="bridge" data="{execute_on_answer=start_dtmf_generate}sofia/internal/9000(a)10.1.1.1"/> <action application="pre_answer"/> </condition> </extension> In my sip profile the "inbound-late-negotiation" param is set to true. Is there any mistake in my dialplan? Thanks, Stefano F. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/no-audio-with-bypass-media-af… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Fax problem
by Alexey Melnichuk 08 Jul '15

08 Jul '15

FreeSWITCH Version 1.4.20+git~20150703T164215Z~b95362f965~32bit (git b95362f 2015-07-03 16:42:15Z 32bit) I think I found bug in spandsp. `t4_image_put_handler_t` defined as (__cdecl) typedef int (*t4_image_put_handler_t)(void *user_data, const uint8_t buf[], size_t len); But `tXX_decode_put` functions defined as `__stdcall`. So when I do s->image_put_handler = (t4_image_put_handler_t) t4_t6_decode_put; ... s->image_put_handler(...) I got AV. After I declare t4_image_put_handler_t as __stdcall problem solved. (Alse need update declaretion to `pre_encoded_put` function) But I start getting heap corruption on `span_free((char *) s->tiff.file);` Problem is that it uses `strdup` function with different allocator. So I just copy function to `alloc.c` SPAN_DECLARE(char *) span_strdup(const char *str) { char *sdup; size_t len = strlen(str) + 1; sdup = (char *) span_alloc(len); memcpy(sdup, str, len); return sdup; } And use it as ds->tiff.file = span_strdup(file);

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FreeSWITCH 1.6 Video + streaming
by Jurijs Ivolga 08 Jul '15

08 Jul '15

Hi, I installed Freeswitch as described here: https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video Additionally I complied and loaded mod_vlc. Now I need to stream video call. I read wiki documentation for module mod_vlc and I was trying following config for streaming: <action application="set" data="record_concat_video=true"/> <action application="record" data="vlc://#standard{access=http,mux=raw,dst= 198.168.0.100:8080/bla}"/> With such config call is failing. Nevertheless I'm able to record video of call with such config: <action application="set" data="record_concat_video=true"/> <action application="set" data="execute_on_answer=record_session $${recordings_dir}/${strftime(%Y%m%d%H%M%S)}_${caller_id_number}.mp4"/> Maybe somebody know how is possible to stream video call? Any help will be appreciated! Thank you!

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DTMF bind digit action not firing
by Martyn Davies 07 Jul '15

07 Jul '15

I have an action set up in my conference call controls like so: <action application="bind_digit_action" data="confi,#*,exec:javascript,/usr/share/freeswitch/scripts/record-my-name.js" /> Which oddly works with some calls but not with others (all the calls are coming via the same operator network - the SIP/RTP comes from a media gateway somewhere in their network). In the case of a good one, we see this in the console: 2015-07-07 14:41:08.597322 [DEBUG] switch_rtp.c:6039 RTP RECV DTMF #:480 2015-07-07 14:41:08.597322 [DEBUG] switch_channel.c:488 RECV DTMF #:480 2015-07-07 14:41:08.937325 [DEBUG] switch_rtp.c:6039 RTP RECV DTMF *:480 2015-07-07 14:41:08.937325 [DEBUG] switch_channel.c:488 RECV DTMF *:480 2015-07-07 14:41:08.937325 [DEBUG] mod_conference.c:4131 Execute app: javascript, /usr/share/freeswitch/scripts/record-my-name.js and in the case of a bad one: 2015-07-07 14:43:13.897322 [DEBUG] switch_rtp.c:6039 RTP RECV DTMF #:480 2015-07-07 14:43:13.897322 [DEBUG] switch_channel.c:488 RECV DTMF #:480 2015-07-07 14:43:14.757320 [DEBUG] switch_rtp.c:6039 RTP RECV DTMF *:480 2015-07-07 14:43:14.757320 [DEBUG] switch_channel.c:488 RECV DTMF *:480 In other words the DTMF are seen, but the bind action doesn't fire. I grabbed a trace of the RTP, and the two cases look indistinguishable, there are start/stop events, with sensible volume and duration (in fact these parameters are the same in the two cases). Can anyone tell me what further diagnostics are available in Freeswitch to tell why the DTMF don't fire the bind action? Regards, Martyn

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FS 1.7 / [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION]
by Igor Olhovskiy 07 Jul '15

07 Jul '15

Hi! Trying to up FS 1.7 (master release from 07.07.2015) with web-based SIP (jsSIP 0.6) (Yes, I know about mod_verto) When making an ORIGINATE request (one of legs is Web-softphone) got this: 2015-07-07 07:35:30.354313 [DEBUG] switch_core_state_machine.c:40 sofia/internal/rlakoni6(a)avkit3h1g7jo.invalid Standard INIT 2015-07-07 07:35:30.354313 [DEBUG] switch_core_state_machine.c:48 (sofia/internal/rlakoni6(a)avkit3h1g7jo.invalid) State Change CS_INIT -> CS_ROUTING 2015-07-07 07:35:30.354313 [DEBUG] switch_core_state_machine.c:516 (sofia/internal/rlakoni6(a)avkit3h1g7jo.invalid) State INIT going to sleep 2015-07-07 07:35:30.354313 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/rlakoni6(a)avkit3h1g7jo.invalid) Running State Change CS_ROUTING 2015-07-07 07:35:30.354313 [DEBUG] sofia.c:6708 Channel sofia/internal/rlakoni6(a)avkit3h1g7jo.invalid entering state [calling][0] 2015-07-07 07:35:30.354313 [DEBUG] switch_core_state_machine.c:532 (sofia/internal/rlakoni6(a)avkit3h1g7jo.invalid) State ROUTING 2015-07-07 07:35:30.354313 [DEBUG] mod_sofia.c:141 sofia/internal/rlakoni6(a)avkit3h1g7jo.invalid SOFIA ROUTING 2015-07-07 07:35:30.354313 [DEBUG] switch_ivr_originate.c:67 (sofia/internal/rlakoni6(a)avkit3h1g7jo.invalid) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2015-07-07 07:35:30.354313 [DEBUG] switch_core_state_machine.c:532 (sofia/internal/rlakoni6(a)avkit3h1g7jo.invalid) State ROUTING going to sleep 2015-07-07 07:35:30.354313 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/rlakoni6(a)avkit3h1g7jo.invalid) Running State Change CS_CONSUME_MEDIA 2015-07-07 07:35:30.354313 [DEBUG] switch_core_state_machine.c:551 (sofia/internal/rlakoni6(a)avkit3h1g7jo.invalid) State CONSUME_MEDIA 2015-07-07 07:35:30.354313 [DEBUG] switch_core_state_machine.c:551 (sofia/internal/rlakoni6(a)avkit3h1g7jo.invalid) State CONSUME_MEDIA going to sleep 2015-07-07 07:35:30.594339 [DEBUG] sofia.c:6708 Channel sofia/internal/rlakoni6(a)avkit3h1g7jo.invalid entering state [proceeding][180] 2015-07-07 07:35:30.594339 [NOTICE] sofia.c:6810 Ring-Ready sofia/internal/rlakoni6(a)avkit3h1g7jo.invalid! 2015-07-07 07:35:30.594339 [DEBUG] switch_channel.c:3332 (sofia/internal/rlakoni6(a)avkit3h1g7jo.invalid) Callstate Change DOWN -> RINGING 2015-07-07 07:35:32.434365 [DEBUG] sofia.c:6708 Channel sofia/internal/rlakoni6(a)avkit3h1g7jo.invalid entering state [completing][200] 2015-07-07 07:35:32.434365 [DEBUG] sofia.c:6718 Remote SDP: v=0 o=- 4629109737787807592 2 IN IP4 127.0.0.1 s=- t=0 0 a=msid-semantic: WMS nIK9U7F3jq79chtGTGAtUOON7Lbekhy8msI2 m=audio 61483 RTP/SAVPF 0 8 13 c=IN IP4 192.168.88.51 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:13 CN/8000 a=rtcp:9 IN IP4 0.0.0.0 a=candidate:708354780 1 udp 2122260223 192.168.88.51 61483 typ host generation 0 a=candidate:1690000940 1 tcp 1518280447 192.168.88.51 0 typ host tcptype active generation 0 a=ice-ufrag:Bb2sJM3lTievaC/+ a=ice-pwd:0wnAgFcoY7QnqfqNg11YYkkA a=fingerprint:sha-256 90:46:2D:15:D0:D6:24:34:F1:8E:8C:8C:20:F9:49:3D:F6:9B:C1:AD:92:16:5F:A9:46:18:30:9C:59:99:41:C0 a=setup:active a=mid:audio a=rtcp-mux a=ssrc:2731247554 cname:d/DW/QebHgyWUgUz a=ssrc:2731247554 msid:nIK9U7F3jq79chtGTGAtUOON7Lbekhy8msI2 c3732e3c-701f-496c-a0df-92bac3364375 a=ssrc:2731247554 mslabel:nIK9U7F3jq79chtGTGAtUOON7Lbekhy8msI2 a=ssrc:2731247554 label:c3732e3c-701f-496c-a0df-92bac3364375 2015-07-07 07:35:32.434365 [DEBUG] sofia.c:6708 Channel sofia/internal/rlakoni6(a)avkit3h1g7jo.invalid entering state [ready][200] 2015-07-07 07:35:32.434365 [DEBUG] switch_core_media.c:4065 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] *2015-07-07 07:35:32.434365 [DEBUG] switch_core_media.c:4120 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match* 2015-07-07 07:35:32.434365 [DEBUG] switch_core_media.c:4065 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2015-07-07 07:35:32.434365 [DEBUG] switch_core_media.c:4065 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[GSM:3:8000:20:13200:1] 2015-07-07 07:35:32.434365 [DEBUG] switch_core_media.c:4065 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2015-07-07 07:35:32.434365 [DEBUG] switch_core_media.c:4065 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] *2015-07-07 07:35:32.434365 [DEBUG] switch_core_media.c:4120 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match* 2015-07-07 07:35:32.434365 [DEBUG] switch_core_media.c:4065 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[GSM:3:8000:20:13200:1] 2015-07-07 07:35:32.434365 [DEBUG] switch_core_media.c:2851 Set Codec sofia/internal/rlakoni6(a)avkit3h1g7jo.invalid PCMU/8000 20 ms 160 samples 64000 bits 1 channels 2015-07-07 07:35:32.434365 [DEBUG] switch_core_codec.c:111 sofia/internal/rlakoni6(a)avkit3h1g7jo.invalid Original read codec set to PCMU:0 2015-07-07 07:35:32.434365 [WARNING] switch_core_media.c:3187 NO candidate ACL defined, Defaulting to wan.auto 2015-07-07 07:35:32.434365 [DEBUG] switch_core_media.c:3216 Save audio Candidate cid: 1 proto: udp type: host addr: 192.168.88.51:61483 *2015-07-07 07:35:32.434365 [DEBUG] switch_core_media.c:3256 Searching for rtp candidate.* *2015-07-07 07:35:32.434365 [DEBUG] switch_core_media.c:3256 Searching for rtcp candidate.* *2015-07-07 07:35:32.434365 [DEBUG] switch_core_media.c:3300 sofia/internal/rlakoni6(a)avkit3h1g7jo.invalid no suitable candidates found.* 2015-07-07 07:35:32.434365 [DEBUG] switch_core_media.c:4345 No 2833 in SDP. Disable 2833 dtmf and switch to INFO 2015-07-07 07:35:32.434365 [NOTICE] sofia.c:7591 Hangup sofia/internal/rlakoni6(a)avkit3h1g7jo.invalid [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION] 2015-07-07 07:35:32.434365 [DEBUG] sofia.c:1355 Channel is already hungup. PCMA and PCMU are in there. But why incompatible dest? I've got ring, but after accept mic sharing, got hangup. On FS 1.5.15 same config, all is ok. -- Best regards, Igor

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ClueCon 2015, Hotel Cutoff date is July 15th, Register NOW!
by Brian West 07 Jul '15

07 Jul '15

[image: Description: HDD:Users:anthm:Downloads:ccxx.jpg] *Register NOW!* August 3rd – August 6th 2015 877-742-2583 • marketing(a)cluecon.com Register NOW! · $899 Staying at the Hotel · $1199 Staying Elsewhere · First 100 Gino’s PIZZA PARTY *IMPORTANT DATES* *July 15th* Hotel room cut off date, First come first serve on hotel rooms after this date. Conference ticket prices go up. $1299 Staying at the Hotel $1599 Staying Elsewhere *August 3rd* Conference in Session *Contact Us* https://cluecon.com marketing(a)cluecon.com *Coder Games and Pizza, Register before July 15th and SAVE!* Coder Games have been expanded, IPv6 Round Table discussion, Monday August 3rd. Visit www.cluecon.com for more info! New Sponsors *16bit Sponsor* - Tropo <https://www.tropo.com/?clueconemail>: Think. Build. Connect. Learn more about Tropo <https://www.tropo.com/?clueconemail> ! *8bit Sponsor* - Zoiper <http://www.zoiper.com/?clueconemail>: High Quality, Cross Platform, Secure VoIP, Each attendee will receive a free Zoiper <http://www.zoiper.com/?clueconemail> license. *8bit Sponsor* - TruPhone <https://www.truphone.com/?clueconemail>: International calling is made easy with the Truphone SIM. You can also make free internet calls using the Truphone app. *Still interested in Sponsoring? Call today!* FreeSWITCH Training Friday we will be hosting a FreeSWITCH training session for anyone interested in learning the basics of FreeSWITCH from a skilled instructor. Call 877-742-2583 for pricing and availability. *Great Speakers, Coder Games, WebRTC And MUCH MORE at ClueCon 2015!* [image: codergames] Be sure to register as soon as possible for the upcoming ClueCon 2015 Developers Conference. Not only will it give you piece of mind, the sooner you register, the more opportunities you will get to win prizes! You’ll also get more drink coupons for the Gigabit Reception Tuesday Night! [image: Description: mac1]The grand prize is a laser engraved commemorative FreeSWITCH 1.6 Edition dual-core 13" Retina MacBook Pro! See the Important Dates Section for Registration details! Stay tuned for more exciting announcements! *Why I Think You MUST COME To ClueCon!* [image: Description: kk]Hi, I’m Kathleen. I’m the FreeSWITCH and ClueCon Social Media Correspondent. I’ve been working hard all year keeping you all up to date on what’s going on with FreeSWITCH. Today I’m here to let you know more about the upcoming ClueCon 2015 Conference! This year we are adding an optional day on Monday with an all-day Hack-A-Thon with great coding contests, game shows and kick-off fun! If you are interested in WebRTC, Voice over IP or Open Source projects like FreeSWITCH, ClueCon is the greatest opportunity you have to gain exposure to the most knowledge and technology in one place. Also, it’s the most fun you can possibly have while still getting a ton of work done! I really look forward to seeing you all there and enjoying the amazing talks, the Epic Annual Kick-Off Pizza Party, The Gigabit Reception and so much more. Make sure you register today so you can reserve your place among the attendees! Be sure to follow us on Facebook and Twitter to get my latest updates in info! [image: Inline image 1] <https://facebook.com/cluecon> [image: Inline image 2] <https://twitter.com/cluecon> [image: Inline image 3] <https://www.youtube.com/watch?v=NLaDpGQuZDA> [image: Inline image 4] <https://www.youtube.com/watch?v=9XXgW34t40s> Unsubscribe: http://lists.freeswitch.org -- *Brian West* brian(a)freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com ClueCon 2015 Call for Speakers <https://www.cluecon.com/call-for-speakers/> | Register <https://freeswitch.com/cart.php?gid=1> TODAY! | Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest

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Check session ready after session transfer
by Zvi Agmon 07 Jul '15

07 Jul '15

Hello, Need some help regarding session state validation. In documentation I see this: session:ready - checks whether the session is still active (true anytime between call starts and hangup) - also session:ready will return false if the call is being transferred. Bottom line is you should always be checking session:ready on any loops and periodically throughout your script and exit asap if it returns false. My experience is that after transfer the return value of session:ready() is is not consistent - what is the correct way to check session state in that case? Thanks Zvi Agmon Lexifone <http://www.lexifone.com/> email: zvi(a)lexifone.com Office: +972-4-6817711 Cell: +972-54-4505109

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