February 2017 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Strip and use SIP headers in freeswitch dialplan configuration
by vivek＠advaitamtech.com 15 Feb '17

15 Feb '17

Hi All, Am trying to fetch "Proxy-Authorization" header present in the SIP INVITE request through dialplan xml. Since there is no Built-in variable to fetch this header, is there a way to find if this header is present in the SIP INVITE request?. I browsed to find any possible solution but wasnt successful. Thanks, Vivek

3 3

spandsp_start_fax_detect sometimes miss fax detection
by Yehavi Bourvine 15 Feb '17

15 Feb '17

Hello, I am using the above application to detect an incoming fax. My dialplan has the following code: <action application="set" data="execute_on_answer=spandsp_start_fax_detect transfer 'fax_${dest}' XML ${context}' 10 cng"/> When using the same fax machine to send, sometimes it detects the fax, sometimes not. This mainly happens when the call lands into voicemail; when answering the call the success rate is much higher (almost 100%). Is there any way to "tune" this detection? Or shall I use something different than "execute_on_answer" for voicemail? Thanks! __Yehavi:

1 0

incoming call going to the wrong context
by David Villasmil 14 Feb '17

14 Feb '17

Hello guys, So i have a gw in external profile, this gw's context is "default" which doesn't exist, still when a call comes in, fs sends it to public context (which is the context on external.xml), why is that? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

1 0

Freeswitch sends 2 SDP answers which causes webrtc client to drop the call
by Georgi Nikolaev 14 Feb '17

14 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. Best Regards, Georgi Nikolaev

2 2

Playing the loudest participant in a conference and muting the others.
by Oivvio Polite 14 Feb '17

14 Feb '17

Is there a way to only hear the currently loudest speaker in a conference? I know about "energy-level" that let's me suppress silent channels that don't reach a certain level. But that's not quite what I want. Rather I want the output of each channel to be dependent on the current level of the other channels. Oivvio

2 1

FreeSwitch originate api + early_media timeout
by Vyronas Tsingaras 14 Feb '17

14 Feb '17

Hi, Is there a way to use bridge_answer_timeout when using the originate API? For example something like: api originate {origination_uuid=XXXX,ignore_early_media=true,answer_ timeout=5}[]sofia/gateway/ to timeout 5 seconds AFTER going into Callstate EARLY. Kind Regards, * Vyronas Tsingaras * Network & Systems Administrator vtsingaras(a)amdtelecom.net www.amdtelecom.net

1 0

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.

2 1

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

1 0

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

2 2

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

18 35