March 2017 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Using cisco ip communicator 8.6 with freeswitch
by Babak Yakhchali 20 Mar '17

20 Mar '17

Hi Has anyone used cisco ip communicator 8.6 with freeswitch? It has some problems with tftp but even if you solve that problem by modifying tftp code it, after loading it's xml config file it never sends any packet to freeswitch.

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Please check jira
by Sergey Safarov 20 Mar '17

20 Mar '17

Jira servers is offline. Please check

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group_confirm_key multiple digits
by Matt Broad 20 Mar '17

20 Mar '17

Hi, is there anyway of setting the group_confirm_key to be any key? For example, the group_confirm_file will play a file that will go something like "press any key to accept this call". thanks Matt

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Fax T.38: How to deal with NSF (NSS)?
by Peter Steinbach 20 Mar '17

20 Mar '17

Hello, we have a number of fax machines attached via a Grandstream ATA to a Freeswitch. Fax devices are Brother MFC. Freeswitch works in t38-passthru mode. In the last weeks we see (maybe due to software update?), that the fax devices detect, that they are siblings with extended feature sets, and then one side switches to "non standard facilities" mode by sending an NSF signal (resp NSS in wireshark). The opposite side then answers with "failed to train" (FTT). This always breaks the fax transmission. We also asked Brother to send us an advice, how to disable NSF in the fax device, but these codes did not work. The fax devices are continuing to send NSF. Besides that we have no control of the opposite fax device, who are initiating NSF. So I am asking here: How are you dealing with NSF? How did you solve this issue? -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de

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Is Freeeswitch support Media Server Markup Language(MSML)
by Uday kumar 20 Mar '17

20 Mar '17

Hi All, Is Freeswitch support Media Server Markup Language (MSML) or any module developed for same? -- Thanks & Regard Uday Site:- www.shareyourknowledge.in Mobile:- +91-9377579349 <093775%2079349>

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Disable speex for build?
by Chad Phillips 19 Mar '17

19 Mar '17

I’m doing a custom build of FreeSWITCH, pretty stripped down. I’ve been able to skip building most things, but I don’t see a way to disable the requirement for libspeex/libspeexdsp. I did poke around configure.ac a bit for clues, but didn’t see any. Is there a way to disable building with speex support? Chad

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Longest-idle-agent or round-robin in group call
by Murugan Pandian 19 Mar '17

19 Mar '17

Hi, Any way i can achieve round-robin or Longest-idle-agent ringing in group call event through XML or lua

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ftdm openr2 status interpretation
by Nandy Dagondon 18 Mar '17

18 Mar '17

Hello folks, Please confirm IF my interpretations are correct. First, the output of "ftdm r2 status <span>": Tx CAS is the AB bits sent TO the opposite end Rx CAS is the AB bits rcvd FROM the opposite end Then, if this is the Called end, then ... Tx CAS is sending the Backward line signals to the Calling party Rx CAS is the Forward line signals received from the Calling party And, I want to make sure, if this is the Calling end, then Tx CAS is sending the Forward line signals; and Rx CAS is receiving the Backward lines signals Thanks in advance. /Nandy

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Convert Sip to Webrtc-like call
by Kamil Nigmatullin 18 Mar '17

18 Mar '17

Maybe it is a silly question, but cannot find anything. I have the opensips that handles auth and all routing, and freeswitch is responsible for handling media, prepaid and so on. Now, I want to implement browser clinent based on sip.js and therefore i pass all sip signalling through opensips's implementetion for websockets. For outgoing invites from browser clients this works. But for calls that are directed to browser clients it doesn't. I beleive that this is due to simlpe sip call does not have ice-candidates. So, the question is how to make freeswitch convert simple sip call to web-rtc oritenteated. I didn't implement mod_verto. Thanks in advance -- Kamil Nigmatullin Skype: kamil.nigmatullin

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Change Hangup Cause on Socket Error!
by Markus Bönke 17 Mar '17

17 Mar '17

Hello, In our dialplan we are using an external service to to handle calls. So our dialplan looks like: <extension name="dialplanserver"> <condition field="destination_number" expression="^(\\+\d{5,20})$"> <action application="info"/> <action application="socket" data="192.168.1.23:8040 async full"/> </condition> </extension> In the case the external service which is posted by the socket application is down, I can see a "Socket Error!“ error message in the log. The call ends with 480 - "Temporarily Unavailable". ... [ERR] mod_event_socket.c:454 Socket Error! ... [DEBUG] mod_sofia.c:549 Responding to INVITE with: 480 ... Is it somehow possible to force hangup cause 503 - "Service Unavailable" instead of 480 - "Temporarily Unavailable" in case of a Socket Error? Thanks and regards Markus

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