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[SDP Negotiation Error][Calling through JsSIP]
by Bilal Abbasi 12 Feb '16

12 Feb '16

Hi all, Currently i am facing an CODEC NEGOTIATION ERROR while calling through JsSIP Caller 1001 Callie 1000, Please view the logs, https://pastebin.freeswitch.org/24550 I have enabled PCMA and PCMU in my var.xml, Regards Abbasi

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Dev meeting
by Blake Priddy 11 Feb '16

11 Feb '16

https://www.gofundme.com/freeswitch It's not too late!!

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Two Users Registered ..But calls only going one way
by David Wafula 11 Feb '16

11 Feb '16

Hi all, I have two users who registered in the same domain: user A and B. A can call B just fine. When B tries to call A, there is silence (no ringback)..then after sometime the call goes into voice mail. A never receives the call. Please see the call trace here: http://pastebin.com/gWrrS4zw Am not sure what is causing it. Regards -- David W

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Re: [Freeswitch-users] Two Users Registered ..But calls only going one way
by Ahmed Habiba 11 Feb '16

11 Feb '16

Are you using TLS? is GS configures with nat configuration correctly? > > From: Michael Jerris <mike(a)jerris.com <mailto:mike@jerris.com>> > Subject: Re: [Freeswitch-users] Two Users Registered ..But calls only going one way > Date: February 11, 2016 at 7:21:11 PM GMT+3 > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org <mailto:freeswitch-users@lists.freeswitch.org>> > Reply-To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org <mailto:freeswitch-users@lists.freeswitch.org>> > > > Sip trace would help... is call forwarding turned on on the phone? > >> On Feb 11, 2016, at 3:46 AM, David Wafula <davidwaf(a)gmail.com <mailto:davidwaf@gmail.com>> wrote: >> >> Hi all, >> I have two users who registered in the same domain: user A and B. >> A can call B just fine. When B tries to call A, there is silence (no ringback)..then after sometime the call goes into voice mail. A never receives the call. Please see the call trace here: >> >> >> http://pastebin.com/gWrrS4zw <http://pastebin.com/gWrrS4zw> >> >> Am not sure what is causing it. > > > > > From: mohammed shafeeque <shafeeq.v(a)gmail.com <mailto:shafeeq.v@gmail.com>> > Subject: Re: [Freeswitch-users] Oneway audio issues in freeswitch > Date: February 11, 2016 at 7:28:45 PM GMT+3 > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org <mailto:freeswitch-users@lists.freeswitch.org>> > Reply-To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org <mailto:freeswitch-users@lists.freeswitch.org>> > > > Surprised that no one else experienced this problem. Can anyone give any hint. Really Dont want to move back to 1.4.x > > On Thu, Feb 11, 2016 at 7:44 AM, Luis Daniel Lucio Quiroz <luis.daniel.lucio(a)gmail.com <mailto:luis.daniel.lucio@gmail.com>> wrote: > As a rule of dumb, try turning on rport > Le 10 févr. 2016 8:55 PM, "Ítalo Rossi" <italo(a)freeswitch.org <mailto:italo@freeswitch.org>> a écrit : > You need to look at the sip signaling to see what's going on > > On Wed, Feb 10, 2016 at 5:59 PM, mohammed shafeeque <shafeeq.v(a)gmail.com <mailto:shafeeq.v@gmail.com>> wrote: > Hello All > > We are getting one way audio issues with some softphones and grandstream phones behind nat registerd to our freeswitch server. > > Here is scenario: > Grandstream call any extensions (one way audio) > Any extension call Grandstream ( Audio works just fine) > > We have tried multiple softphones and the result is same. > > Everything was working fine with 1.4.18 and 1.6.2. We were having DTMF issue with 1.6.2, upgraded to 1.6.5 and now one way audio issue started with an upgrade to freeswitch. > > Any help or hint will be much appreciated. > > Thank you, > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org <mailto:consulting@freeswitch.org> > http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/> > > Official FreeSWITCH Sites > http://www.freeswitch.org <http://www.freeswitch.org/> > http://confluence.freeswitch.org <http://confluence.freeswitch.org/> > http://www.cluecon.com <http://www.cluecon.com/> > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users> > http://www.freeswitch.org <http://www.freeswitch.org/> > > > > -- > Ítalo Rossi > italo(a)freeswitch.org <mailto:italo@freeswitch.org> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org <mailto:consulting@freeswitch.org> > http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/> > > Official FreeSWITCH Sites > http://www.freeswitch.org <http://www.freeswitch.org/> > http://confluence.freeswitch.org <http://confluence.freeswitch.org/> > http://www.cluecon.com <http://www.cluecon.com/> > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users> > http://www.freeswitch.org <http://www.freeswitch.org/> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org <mailto:consulting@freeswitch.org> > http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/> > > Official FreeSWITCH Sites > http://www.freeswitch.org <http://www.freeswitch.org/> > http://confluence.freeswitch.org <http://confluence.freeswitch.org/> > http://www.cluecon.com <http://www.cluecon.com/> > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org <mailto:FreeSWITCH-users@lists.freeswitch.org> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users <http://lists.freeswitch.org/mailman/options/freeswitch-users> > http://www.freeswitch.org <http://www.freeswitch.org/> > > > >

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P-asserted-identity
by Jose Serrano 11 Feb '16

11 Feb '16

Hello. My freeswitch by default replace the P-asserted-identity by Remote-Party-ID when routing calls to the gateway. I want to send the P-asserted-identity and not the Remote-Party-ID and for that I tried the following: I have configured in the outbound gateway definition the following parameters: <param name="caller-id-type" value="pid"/> or <param name="cid-type" value="pid"/> but te behavior is the same. The only thing that works is to set up the "sip_cid_type" variable in the dialplan like this: <action application="bridge" data="{sip_cid_type=pid}sofia/gateway/Mygateway/$1"/> Anyone knows how I can send the P-Asserted-identity without having to modify all my dial plan adding the sip_cid_type? Thanks in avanced

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Jessie: Unable to locate package
by Dominik Steinbrecher 11 Feb '16

11 Feb '16

Hi, I want to install freeswitch 1.6 on a freshly installed Debian 8.3 jessie. I followed the guide from: https://freeswitch.org/confluence/display/FREESWITCH/Debian+8+Jessie When I execute apt-get install I got the following errors: root@freeswitch:~# apt-get update && apt-get install -y freeswitch-all freeswitch-all-dbg gdb Hit http://security.debian.org jessie/updates InRelease Ign http://ftp.de.debian.org jessie InRelease Hit http://ftp.de.debian.org jessie-updates InRelease Hit http://ftp.de.debian.org jessie Release.gpg Hit http://security.debian.org jessie/updates/main Sources Hit http://ftp.de.debian.org jessie Release Hit http://security.debian.org jessie/updates/main i386 Packages Hit http://security.debian.org jessie/updates/main Translation-en Hit http://ftp.de.debian.org jessie-updates/main Sources Get:1 http://ftp.de.debian.org jessie-updates/main i386 Packages/DiffIndex [367 B] Hit http://ftp.de.debian.org jessie-updates/main Translation-en Hit http://ftp.de.debian.org jessie/main Sources Hit http://ftp.de.debian.org jessie/main i386 Packages Hit http://ftp.de.debian.org jessie/main Translation-en Hit http://files.freeswitch.org jessie InRelease Hit http://files.freeswitch.org jessie/main i386 Packages Ign http://files.freeswitch.org jessie/main Translation-en_US Ign http://files.freeswitch.org jessie/main Translation-en Fetched 367 B in 3s (102 B/s) Reading package lists... Done Reading package lists... Done Building dependency tree Reading state information... Done E: Unable to locate package freeswitch-all E: Unable to locate package freeswitch-all-dbg I searched for the freeswitch-all package with aptitude but there’s no such package. Any ideas, what I am doing wrong? Is there anything I could try or check? The debian jessie 8.3 64bit is running as VM under VirtualBox. The host computer is a Mac Mini with OS X 10.11.1. It’s a freshly installed VM just for testing freeswitch. Thanks a lot Dominik

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Does freeswitch forks his processes?
by Vladislav Ivanov 11 Feb '16

11 Feb '16

Hey guys, I have a question about freeswitch process/threading usage. So far that I haven't noticed freeswitch to fork himself, I have only 1 freeswitch instance. http://i.imgur.com/bdbYOwp.png But then I found screenshot of htop with freeswitch and noticed that there is multiple freeswitch processes being run: http://i.imgur.com/VNpl55z.jpg I'm having issues with "loading" the freeswitch after 50 cps in any cpu/ram configuration. Be it physical or virtual environment I cant pass the 50 cps mark. I have strange issue with CPU usage on same CPS: http://i.imgur.com/8BdQWVL.png http://i.imgur.com/mWRnoGr.png I timeload test freeswitch with 50cps for 5+ hours, and seems like there is some kind of leak somewhere. I have tested configuration on: Debian 8 2 core/8 gb ram 4 core/8 gb ram (graphs are from here) 8 core/32 gb ram and in all the tests I were not able to send more than 50 cps without CPU dropping to 0 with all system starting to respond really laggy. Test is: sipp -> freeswitch -> sipp Just 1 dialpeer with bridge action. No gateways. Just simple dialplan and 1 profile... Any advice? Thank you all

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Mod_callcenter ODBC vs File
by Jai Rangi 11 Feb '16

11 Feb '16

Hello, FS version: FreeSWITCH Version 1.4.18+git~20150312T185523Z~4eed221b69~64bit (git 4eed221 2015-03-12 18:55:23Z 64bit) I found a very strange behavior on mod_call center when used with odbc. This was easy to reproduce when the agents are busy and new calls come in. New member goes in abandoned state immediately and stay in same state forever. Even if previous call is complete and Agent is in Waiting state. Infact some times new members go in abandoned state immediately even if the agents are available. Everything works perfect once I changed the mod_callcenter.conf.xml config to file only mode, new calls go in Waiting state and call is connected to agent as soon as the agent is free. Any idea why the new member goes in abandoned state randomly in place of being in waiting state? Can it be due to too small timeout in call center module as it query the DB every time? Can that be changed? Anyone else had smiler problem, any experience or any recommendations will be much appreciated. Thank you, -Jai

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Hit that wallet now ! FreeSWITCH core lunch and dinner !
by Giovanni Maruzzelli 10 Feb '16

10 Feb '16

Hey fellow FreeSWITCHers! How much we all made in 2015 thanks to FreeSWITCH? How much we made thanks to core devs (Anthony, Mike, Ken, Brian, William, etc) answering our difficult questions? Let's send them MONEY so they can have a LUXURIOUS lunch and dinner on us ! On top right in http://www.freeswitch.org page there is a "Donate" Paypal button that will directly translate in gluttony for the Core FreeSWITCH Team. Core devs are having the annual development meeting days. If not now, when? Because! -giovanni -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618

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mod_verto and a1-hash
by S.Mohammad Emami Razavi 10 Feb '16

10 Feb '16

Hello, I'm generating hashed password in the domain and with the username in md5. But after setting it as a1-hash param in user directory, verto user can not authenticate correctly and an error is returned from mod_verto. After all authenticating with plain username and password and without a1-hash set in user directory param, is excellent in mod_verto. Can anybody help?!

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