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Gateway dependencies
by Stanislav Sinyagin 14 Feb '17

14 Feb '17

did someone work already on the following feature? It should be relatively easy to do via ESL: let's say our FreeSWITCH server is a part of a redundant cluster, and it should stop accepting new calls if its PSTN gateway is down. Ideally, it would stop responding on OPTIONS pings, so that its upstream neighbors recognize that this node is not good for sending calls. So, the solution should define dependencies between gateways (ideally, not only SIP gateways, but other endpoint types too), like: if gateway X is down for more than 10 seconds, shut down gateways Y and Z. If X is up for more than 5 seconds, bring up Y and Z. In a more scalable approach, you would trigger the change if less than a certain percentage in a set of gateways is down.

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binary audiostream
by Tito Cumpen 14 Feb '17

14 Feb '17

Hello Group, I am looking to do some integration with watson for text to speech. Not sure if anyone here is familiar with it but I was wondering if it possible to send a binary stream of the audio within a call to a 3rd party resource. I dont mind build a middle man app that would allow me to implement the following https://www.ibm.com/watson/developercloud/doc/speech-to-text/websockets.sht… It seems watson has to be notified of the audio pt so if that would be acquired somehow that would be helpful. Also I don't wan't to deal with removing rtp headers or decryption DTLS if possible. Please advise

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Broadcasting a phrase every n seconds till session is connected to a callcenter agent
by Babak Yakhchali 14 Feb '17

14 Feb '17

Hi How can I play a phrase to a session using scheduler api every n seconds when it is waiting in a mod_callcenter queue? I know I can use the script in mod_callcenter confluence page but I think it is not scaleable. thanks

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A note from the Lead Developer. FreeSWITCH 1.8 Feature Requests
by Anthony Minessale 14 Feb '17

14 Feb '17

Hi, We are having an annual gathering to decide the roadmap for upcoming FreeSWITCH 1.8 Now is your chance to propose some feature requests in this thread. No guarantees, but this is the best time during our release cycle to daydream about the future. We hope to have a first cut of 1.8 in time for ClueCon 2017 so register early, the price is the lowest it will ever be during the Early Bird pricing offer lasting only a few more weeks. Also, Brian has started a fund to raise money to provide the team with meals and offset travel expenses if possible during the upcoming summit in little over a week from now. Its one really easy way to show your appreciation for another year of open source telephony! https://www.gofundme.com/freeswitch-developer-meeting-2017 -- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>* ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900 https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA

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“failed to set remote sdp” due to 2 SDP answers
by Georgi Nikolaev 13 Feb '17

13 Feb '17

Hi Folks, I am experiencing an issue where remote SDP can not be set due to receiving 2 SDP answers. The setup is webrtc (sipjs)->Freeswitch->SIP Termination, where a call is originated by the sipjs.ua calling a public number. As soon as the remote party answers the call, the call drops due to not being able to gather the remote media path. Worth mentioning is that only some calls fail where the majority are successful. The difference between the successful and the unsuccessful ones is in the SDP messages. The good one receive ‘answer' for UDP/TLS/RTP/SAVPF followed by ‘offer’ for RTP/SAVPF. The one that fail are 2 SDP answers. I have discussed the topic on the webrtc group: https://groups.google.com/forum/?utm_medium=email&utm_source=footer#!msg/di… Also raised this in the sipjs forum: https://groups.google.com/forum/?utm_medium=email&utm_source=footer#!topic/… On both it has been suggested that this is either a setting in Freeswitch or its incorrectly sent SDP messages. Freeswitch version: 1.6.14 SIP.js: 0.7.5 Browser: Chrome - latest version I would appreciate any help on this as its causing me some random call drops affecting the service. Here are the logs as well: FS log: http://pastebin.com/UQW6AdcQ Chrome log: http://pastebin.com/Crk69MrS webrtc-internals-log: http://pastebin.com/BVr7v4EB Any ideas/suggestions/comments are welcome. Thank you in advance. Regards, Georgi Nikolaev

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Cepstral will provide an Allison-8kHz for Linux license to anyone donating $100 or more.
by Brian West 13 Feb '17

13 Feb '17

See that? Cepstral will provide an Allison-8kHz for Linux license to anyone donating $100 or more. You get an Allison 8kHz license for Linux if you donate more than 100 dollars. https://www.gofundme.com/allison-prompts-for-freeswitch Thanks, -- *Brian West* brian(a)freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Allison prompts for FreeSWITCH: *https://www.gofundme.com/allison-prompts-for-freeswitch* <https://www.gofundme.com/allison-prompts-for-freeswitch> Wish to schedule a meeting? http://app.timebridge.com/#/meet/freeswitch Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *Skype:*briankwest

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WebRTC calls failing
by SamyGo 13 Feb '17

13 Feb '17

Hi, I've trying to figure out the error that causes the WebRTC based calls to hangup. 2017-02-09 18:57:53.708073 [NOTICE] switch_rtp.c:1275 Auto Changing audio stun/rtp/dtls port from 70.54.102.180:56188 to 70.54.102.180:1572 2017-02-09 18:57:54.688079 [ERR] switch_rtp.c:3165 audio Handshake failure 1 2017-02-09 18:57:54.688079 [INFO] switch_rtp.c:3166 Changing audio DTLS state from HANDSHAKE to FAIL Operating System is: Ubuntu 14.04.5 LTS *OpenSSL version:* OpenSSL 1.0.1f 6 Jan 2014 built on: Mon Jan 30 20:38:38 UTC 2017 platform: debian-amd64 options: bn(64,64) rc4(16x,int) des(idx,cisc,16,int) blowfish(idx) compiler: cc -fPIC -DOPENSSL_PIC -DOPENSSL_THREADS -D_REENTRANT -DDSO_DLFCN -DHAVE_DLFCN_H -m64 -DL_ENDIAN -DTERMIO -g -O2 -fstack-protector --param=ssp-buffer-size=4 -Wformat -Werror=format-security -D_FORTIFY_SOURCE=2 -Wl,-Bsymbolic-functions -Wl,-z,relro -Wa,--noexecstack -Wall -DMD32_REG_T=int -DOPENSSL_IA32_SSE2 -DOPENSSL_BN_ASM_MONT -DOPENSSL_BN_ASM_MONT5 -DOPENSSL_BN_ASM_GF2m -DSHA1_ASM -DSHA256_ASM -DSHA512_ASM -DMD5_ASM -DAES_ASM -DVPAES_ASM -DBSAES_ASM -DWHIRLPOOL_ASM -DGHASH_ASM OPENSSLDIR: "/usr/lib/ssl" Tried couple of Jira Bug links around this and still no progress. Disabling "inbound_late_negotiations" doesn't help either. Regard, Sammy

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Re: [Freeswitch-users] Allison prompt set for FreeSWITCH DONATE TODAY! https://www.gofundme.com/allison-prompts-for-freeswitch
by Brian West 13 Feb '17

13 Feb '17

FreeSWITCHers, I've worked hard to get this opportunity lined up with Allison and our community, She is also a hoot to work with, If you recall back in my Asterisk days we came up with some really awesome prompts all voiced by Allison, So we can bring some of that fun to FreeSWITCH. With every single donation you'll get a license to use FreeSWITCH, You know you want that right? ;) https://www.gofundme.com/allison-prompts-for-freeswitch So lets welcome Allison to FreeSWITCH Thanks, On Wed, Feb 8, 2017 at 10:54 AM, Brian West <brian(a)freeswitch.org> wrote: > FreeSWITCHers, > > Last week we finalized a few things we wanted to accomplish, One was we've > been asked to make an Allison Smith prompt set for FreeSWITCH, She's agreed > to do the entire sound set and now we have to raise the money to pay her > for this. > > I would like to thank everyone that has donated to our dev week... We are > very grateful our community got so involved. > > Here is the link to the GoFundMe for the Allison prompts (full sound set, > including the zrtp words): > > https://www.gofundme.com/allison-prompts-for-freeswitch > > The total price for this was a little over $10,000, but we've had someone > who shall remain nameless donate almost 50% of this total, Now we have to > raise the rest. > > Thanks, > -- *Brian West* brian(a)freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Allison prompts for FreeSWITCH: *https://www.gofundme.com/allison-prompts-for-freeswitch* <https://www.gofundme.com/allison-prompts-for-freeswitch> Wish to schedule a meeting? http://app.timebridge.com/#/meet/freeswitch Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *Skype:*briankwest

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INSERT command failed: ERROR: invalid byte sequence for encoding "UTF8"
by Mimiko 13 Feb '17

13 Feb '17

Hello. This error I get for some cdr log in pg using mod_cdr_pg_csv: 2017-02-13 14:52:52.859613 [CRIT] mod_cdr_pg_csv.c:274 INSERT command failed: ERROR: invalid byte sequence for encoding "UTF8": 0xee646f HINT: This error can also happen if the byte sequence does not match the encoding expected by the server, which is controlled by "client_encoding". I've found that this is related to "caller name" field. In FS "caller name" is populated from a MySQL table in UTF8 using odbc. The cdr is logged to Postgres database in UTF8. No where in configs I specify encoding. The failed log is spewed in xml log file from which I manually take the insert command and pull it in Postgres. The file is encoded in ascii, so I correct the missing letters. If the caller name does not contain any character if code > 128, all works ok. The problem is with characters for other languages. Where is the culprit? -- Mimiko desu.

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Tone detection questions spandsp.conf.xml
by Gonzalo Gasca Meza 13 Feb '17

13 Feb '17

Hi Freeswitchers: I have been trying to detect *India* ringback tone using spandsp.conf.xml My understanding is that* f1* and *f2* are the Frequency and *min* and *max* is the Tone duration value. For example: *Based on ITU-T reference India progress tones are:* ring = 400*25/400,0/200,400*25/400,0/2000 *Questions* a) How to show the Tone frequency received? b) What is the meaning for the values assigned to f1 and f2 in debug mode. I see possible values: - -1 - 1 - 2 c) Is there an option to configure modulation for Tone frequency as specified in ITU standard (*25 above)? d) Sample configuration will be appreciated. *Configuration* <tone name="RING_TONE"> <element freq1="400" freq2="0" min="384" max="400"/> <element freq1="0" freq2="0" min="200" max="224"/> <element freq1="400" freq2="0" min="384" max="400"/> <element freq1="0" freq2="0" min="1968" max="1984"/> </tone> When debugging: I have enabled already debug level to 2. 2017-02-13 02:59:32.080389 [DEBUG] mod_spandsp_dsp.c:705 Tone segment:* f1 = 1, f2 = 2, duration = 400* 2017-02-13 02:59:32.300524 [DEBUG] mod_spandsp_dsp.c:705 Tone segment:* f1 = -1, f2 = -1, duration = 224* 2017-02-13 02:59:32.680369 [DEBUG] mod_spandsp_dsp.c:705 Tone segment:* f1 = 1, f2 = 2, duration = 384* 2017-02-13 02:59:34.680384 [DEBUG] mod_spandsp_dsp.c:705 Tone segment:* f1 = -1, f2 = -1, duration = 1984* http://pastebin.com/xm8xN9R3 Thanks -Gonzalo

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