June 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

improper conversation sequence
by Aakash 10 Jun '15

10 Jun '15

Hi All, Freeswitch Calls are recorded in mp3 format in my setup. I experience the improper conversation sequence on the recordings. Any advice and suggestions will be greatly appreciated Thank you, Aakash ----- Regards, Aakash -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/improper-conversation-sequenc… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Got Video?
by Brian West 10 Jun '15

10 Jun '15

Visit 888 us on https://cantina.freeswitch.org/conf/ We are trying to load test it and verify a few latency metrics. SEE YOU THERE! -- *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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Presence Issues with Secure Websockets
by Thomas Quintana 10 Jun '15

10 Jun '15

Hey Guys, We're subscribing to presence events via SIP.js using a secure websocket (WSS) and we can see the subscriptions being received by FreeSWITCH with a 202 Accepted, but never receive any events via the notify event. Alternatively, when we move to a plain websocket (WS) everything seems to work as expected. After digging around we found that a few people were experiencing similar issues with FreeSWITCH and additionally with Kamailio http://lists.sip-router.org/pipermail/sr-users/2014-July/084157.html. The issue was resolved with Kamailio by patching the contact header. In an attempt to achieve the same results we have experimented with the settings below but no success. - <param name="NDLB-force-rport" value="safe"/> - <param name="NDLB-received-in-nat-reg-contact" value="true"/> We also have an open ticket with the folks over at SIP.js https://github.com/onsip/SIP.js/issues/184. Any help would be greatly appreciated. Best Regards, Thomas Quintana Chief Technology Officer Phone: 1 (512) 677-6200 Website: http://www.bettervoice.com

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installing Freeswitch on Windows 7 x86
by cynttia moreno 10 Jun '15

10 Jun '15

Anyone tried installing Freeswitch on Windows 7 x86 I am researching for one job in college and I need the .msi installer the downloaded http://files.freeswitch.org/windows/installer/ it does not work, FreeSwitchConsole.exe opening is closed immediately, try to follow the instructions to create my own executable of https://freeswitch.org/confluence/display/FREESWITCH/Windows but I find it very difficult, I think the instructions are not updated.if someone already installed on Windows 7 x86 could spend the installer or help me do, Please be so kind as to help me? thanks

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Jitter Buffer specifics
by Danny G 10 Jun '15

10 Jun '15

According to the FS wiki the 3 jitterbuffer parameters are: initial buffer, max buffer, max drift. - Length is initial size of jitter buffer What does this do in terms of affecting how packets are handled? What happens during a call that affects length? What happens when it is exceeded? - max length is upper bound for how big jitter buffer can grow What happens when the max length is reached? What is difference between max length and length and how do they interact as params to act on the system? - Max drift controls how much delay What is delay measured by in terms of time it takes for a packet to arrive? How does that then interact with the other two params to act? Thanks for the help.

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ZRTP Not working on Master
by Victor Medina 10 Jun '15

10 Jun '15

Hi guys! First of all good morning! OK. I rebuild Master today. Debian 8. x64. As per instructions on https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video. Also on the same machine I build v1.4. Im testing with: Win8, Win7, 32bits and 64bits. Zoiper 3.6 lite and biz, Zoiper 3.7 lite and biz. Zoiper 3.8RC1 biz. Zoiper 3.8lite. Wireshark on all machines. All internal testing. No nat, direct netwoking, near mint default config. While testing v1.4 I can fully establish a ZRTP session. Master (FreeSWITCH Version 1.7.0+git~20150609T205846Z~2e47abc9e6~64bit (git 2e47abc 2015-06-09 20:58:46Z 64bit)) seems to be happily ignoring ZRTP Hello packages from Zoiper. Also... (and remember this is almost MINT/PLAIN default config) v1.4 seems to work MUCH faster in processing and establishing calls. Fill a jira? -- Víctor E. Medina M. Platform Architect / Chief Infrastructure +58424 291 4561 BB #79A8AFA2 @VMCibersys

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[WARNING] sofia_reg.c:2837 Cant register a pointer.
by Aqs Younas 10 Jun '15

10 Jun '15

Hi, users. I am trying to register my softphone with freeswitch using xml_curl *(https://github.com/intralanman/fs_curl <https://github.com/intralanman/fs_curl>*) for fetching user's data from database. My freeswitch is on following instances. tcp 0 0 127.0.0.1:8021 0.0.0.0:* LISTEN 2247/freeswitch tcp 0 0 192.168.1.50:6080 0.0.0.0:* LISTEN 2247/freeswitch tcp 0 0 192.168.1.50:6060 0.0.0.0:* LISTEN 2247/freeswitch udp 0 0 192.168.1.50:6060 0.0.0.0:* 2247/freeswitch udp 0 0 192.168.1.50:6080 0.0.0.0:* 2247/freeswitch Since my internal profile is running on port 6060, i added following " 192.168.1.50:6060" as my domain in database and 192.168.1.50:6060 in my softphone domain. When i try to register my softphone, i encounter with this error. *[WARNING] sofia_reg.c:2837 Cant register a pointer.* But if i change my domain from 192.168.1.50:6060 to 192.168.1.50 in database but in softphone same "192.168.1.50:6060" user is successfully registered. Though xml_curl response is same in both cases expect domain name. For 192.168.1.50:6060 response is*,* <?xml version="1.0" encoding="UTF-8"?> <document type="freeswitch/xml"> <section name="directory" description="FreeSWITCH Directory"> <domain name="192.168.1.50:6060"> <params> <param name="default_gateway" value="errors" /> </params> <variables /> <groups> <group name="default"> <users> <user id="1000" cacheable="0" mailbox="1000"> <params> <param name="password" value="1234" /> <param name="vm-password" value="861000" /> </params> <variables> <variable name="numbering_plan" value="US" /> </variables> </user> </users> </group> </groups> </domain> <domain name="192.168.1.50:6060"> <params /> <variables /> <groups> <group name="default"> <users /> </group> </groups> </domain> </section>       </document> For *192.168.1.50*, response is . <?xml version="1.0" encoding="UTF-8"?> <document type="freeswitch/xml"> <section name="directory" description="FreeSWITCH Directory"> <domain name="192.168.1.50"> <params> <param name="default_gateway" value="errors" /> </params> <variables /> <groups> <group name="default"> <users> <user id="1000" cacheable="0" mailbox="1000"> <params> <param name="password" value="1234" /> <param name="vm-password" value="861000" /> </params> <variables> <variable name="numbering_plan" value="US" /> </variables> </user> </users> </group> </groups> </domain> <domain name="192.168.1.50"> <params /> <variables /> <groups> <group name="default"> <users /> </group> </groups> </domain> </section>       </document> What does that mean *Cant register a pointer*? If i does not enter port in domain, users is successfully register, but not with port why? Any pointer would be greatly appreciated. Thanks

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French sound files and FreeSwitch release
by Yannick 10 Jun '15

10 Jun '15

Hello, I am using FreeSwicth 1.2.3 and I would like to know which release of French sound files should I use ? The same ones as for US, i.e. 1.0.18 ? Thanks for any help, Yannick -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/French-sound-files-and-FreeSw… Sent from the freeswitch-users mailing list archive at Nabble.com.

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g729* offer > g729a answer
by Dmitry Saratsky 10 Jun '15

10 Jun '15

Yes, I'm talking about SDP. 1. If I receive in SDP call with g729ab, can I reply with g729a and then setup the call in g729? Sent from my iPhone

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sip,js with no audio
by anton vojlenko 10 Jun '15

10 Jun '15

Hello all, I 'm using sip.js 0.7.0 and firefox 38.0.5, with this library i can register with the freeswitch without any problem. Unfortunately after successful SIP negotiation i receive no audio call. I have the same issue with google chrome 43.0.2357.81. Here is full log for a problem call: https://pastebin.freeswitch.org/24249 Who can help me with this issue? Anton Vojlenko

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