- FreeSWITCH-users - FreeSWITCH Mailing Lists

Freeswitch translates "480 Temporarily Unavailable" as 200 OK and then BYE
by Camila Troncoso 15 Mar '13

15 Mar '13

Hi, I have a this call : * PROXY FS DEST* INVITE INVITE 100 Trying 100 Trying 183 SDP 183 Session P.SDP 480 Temporarily Unavailable 200 Ok DP ACK BYE When I receive a “ 480 Temporarily Unavailable” after a “183 Session Progress with SDP” FreeSWITCH tries to hang-up the call normally by sending a 200 ok and then a BYE response. But the call was not effectively established. It is generating problems in my proxy server. Is there a way to relay the 480 response or to change it to some other more accurate response. Regards, *Camila Troncoso **|* Ingeniero de Desarrollo RedVoiss *|*ctroncoso(a)redvoiss.net Santiago - Chile *|* +56 2 2408535 www.redvoiss.net

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SaltStack + deployment techniques
by Cal Leeming [Simplicity Media Ltd] 15 Mar '13

15 Mar '13

Changed topic to be more relevant.. Agreed, saltstack is wonderful.. we're using it in combination with Atlassian Bamboo with a similar outcome, we push a release button for a specific workflow, and salt takes care of all the steps; * Compile stack components into deb packages (for example freeswitch, backend web applications etc) and place on local mirror * Performs unit test in the build steps * Fire up a bare instance with salt taking care of installation and ensuring services are running * SIP balancer (freeswitch) moves traffic between the switch instances to allow for rapid release fail over, and a real staging environment.. if there is a problem with a release, we can quickly switch back to the previous release at the LB.. if there is no problem, the old release instance is terminated * Traffic balancer takes care of moving traffic between the API/MySQL instances * Stack runs uwsgi/django/nginx/freeswitch * We use new relic / pingdom / cacti for monitoring * Code is kept in GIT I'm still not entirely happy with the overall procedure and always looking for new/better ways to improve it.. but SaltStack is a clear winner, leaving puppet/chef in it's dust. Cal On Wed, Mar 13, 2013 at 4:38 AM, Gabriel Gunderson <gabe(a)gundy.org> wrote: > On Tue, Mar 12, 2013 at 6:44 AM, Cal Leeming [Simplicity Media Ltd] > <cal.leeming(a)simplicitymedialtd.co.uk> wrote: > > The second problem is that the build resulted in nearly 100 different > *.deb > > files This also poses somewhat of an annoyance in automated deployment > > environments, for example saltstack, where the configuration would have > to > > list each individual FreeSWITCH module. > > Off topic a bit, but it's great to see another Salt user. We have our > stuff built-out to the point where we pull the trigger on the Salt > Master and 30 mins later we have a full stack (12+ systems depending > on how we configure it) of load balanced and highly available servers > running FreeSWITCH, OpenSIPS, nginx, HAProxy, Django, PostgreSQL, > UWSGI, Postfix, etc. --all layered on top of freshly built virtual > hosts running kvm/libvirt under Ubuntu 12.04 LTS (except for FS, it > runs on iron). I almost get goosebumps when it's done running. > > I know you all do this kinda stuff all day long, but still, I can have > my fun too, right? > > Long live Salt! > > :) > > Gabe > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org >

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Friday FreeFor All
by Ken Rice 15 Mar '13

15 Mar '13

Hey Guys Join us for the FreeSWICH Friday Free For All. Bridge is open now, but we¹re targetting getting started at 3PM EST, 12 PST SIP:888@conference.freeswitch.org -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch

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password for voicemail retrieval
by Archana Venugopan 15 Mar '13

15 Mar '13

Hi, Can anyone please tell me how to set password for voicemail retrieval for multiple users and how? Regards, Archana

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Compile failure - v1.3.16
by Cal Leeming [Simplicity Media Ltd] 15 Mar '13

15 Mar '13

Got a compile failure for git tag v1.3.16 checking target system type... x86_64-unknown-linux-gnu /tmp/buildd/freeswitch-1.3.16/src/libs/apr-util/xml/expat/configure: line 3359: syntax error near unexpected token `lt_decl_varnames,' /tmp/buildd/freeswitch-1.3.16/src/libs/apr-util/xml/expat/configure: line 3359: `lt_if_append_uniq(lt_decl_varnames, SED, , ,' # lsb_release --a No LSB modules are available. Distributor ID: Debian Description: Debian GNU/Linux 6.0.7 (squeeze) Release: 6.0.7 Codename: squeeze Wasn't sure if build failures should have a JIRA bug posted for them.. if so, let me know and I'll file a bug. Cal

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How to restrict codec negotiation to ULaw only on incoming leg
by Adelia C. 15 Mar '13

15 Mar '13

What are my options when I am looking for a quick fix for the following situation: My in/out traffic is on a few different carriers. Carrier A is sending this SDP on first INVITE, offering PCMA first and PCMU second: v=0 o=Sonus_UAC 17329 9183 IN IP4 206.165.95.165 s=SIP Media Capabilities c=IN IP4 206.165.95.166 t=0 0 m=audio 24330 RTP/AVP 8 0 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=sendrecv a=maxptime:20 Freeswitch does the right thing by RFC 3264 and selects PCMA but then negotiates PCMU on the outgoing leg. Then, it drops out of the data path and I have another B2BUA in the middle, that can't translate. Can I make FreeSwitch reply with PCMU? [ Am in North America, with pretty much all my traffic on ULaw] I found this suggestion: http://wiki.freeswitch.org/wiki/Codec_negotiation#Late_Negotiation_.28requi… but I am looking for a quick and dirty fix, where I restrict all to ULaw. Thank you. A.C.

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FreeTDM PRI Error
by ashish gautam 15 Mar '13

15 Mar '13

Hi, I have configured PRI with freeswitch using ftdm-dahdi-libpri stack. After loading freetdm module I get this error: "2013-03-15 15:56:13.776242 [ERR] ftmod_libpri.c:1950 [s1c31][1:31] -- T316 timed out, channel reached restart attempt limit '3' and is suspended" Kindly help me out. I have tried to do many changes in freetdm.conf.xml -- REGARDS ============================================ *Ashish Gautam* (+918802865008)

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Call Recording has poor sound quality
by Alex Lake 15 Mar '13

15 Mar '13

Just started playing with call recording on Freeswitch Saving as WAV and the sound quality is really poor - very muffled and possibly chunks of sound missing. The actual call is using PCMA 8KHz. The recording is Wave PCM signed 16 bit, 8000 Hz, 256 kbps, stereo Any idea what's going on? Are there any CPU constraints?

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Re: [Freeswitch-users] Status of mod_skinny? is it being maintained?
by Nathan Neulinger 14 Mar '13

14 Mar '13

ring_ready worked! Why is the behavior different here vs. sip target phone? both are coming in from the same external caller. Does the sofia endpoint send back that ring ready status automatically? -- Nathan On 03/14/2013 03:50 PM, Anthony Minessale wrote: > The inbound SIP leg cancels the call before the ring time timeout. So the box calling this box [10.20.0.26] is > canceling it after 10 seconds. so it has a shorter timeout than the box you are debugging and you don't last long > enough to make it to VM. > > Try adding the ring_ready app to your dial-plan right before you bridge to skinny. > If that doesn't work try answer app, just to prove if you answer it that the down-stream leg will not cancel. > > > > > On Thu, Mar 14, 2013 at 3:43 PM, Nathan Neulinger <nneul(a)mst.edu <mailto:nneul@mst.edu>> wrote: > > Sorry, new to the diagnostics... redoing now. > > http://pastebin.freeswitch.__org/20692 <http://pastebin.freeswitch.org/20692> > > > On 03/14/2013 03:39 PM, Anthony Minessale wrote: > > the sip trace is not in there still. I assume this was the actual log file and not a console cap. > In that case you either need to add console level to the log map or just do sofia tracelevel debug so the traces > are at > debug level instead of console. > > > > On Thu, Mar 14, 2013 at 3:23 PM, Nathan Neulinger <nneul(a)mst.edu <mailto:nneul@mst.edu> <mailto:nneul@mst.edu > <mailto:nneul@mst.edu>>> wrote: > > > http://pastebin.freeswitch.____org/20691 <http://pastebin.freeswitch.__org/20691 > <http://pastebin.freeswitch.org/20691>> > > > This is with a call coming in from an external sip trunk. If you want me to test from a sip from directly > attached > to freeswitch I can do that as well. > > -- Nathan > > > On 03/14/2013 03:06 PM, Anthony Minessale wrote: > > Can you repeat that log with "sofia global siptrace on" Something odd is happening with the SIP leg on the > other side > that I can't see, > > > > On Thu, Mar 14, 2013 at 2:24 PM, Nathan Neulinger <nneul(a)mst.edu <mailto:nneul@mst.edu> > <mailto:nneul@mst.edu <mailto:nneul@mst.edu>> <mailto:nneul@mst.edu <mailto:nneul@mst.edu> > > <mailto:nneul@mst.edu <mailto:nneul@mst.edu>>>> wrote: > > I know, but I definitely didn't hang up on the calling side. > > For the SIP phones, I can let them ring right up to the full timeout. With the Cisco, it's > stopping after > 10s every time > and not doing any rollover to next step in dialplan. > > > -- > ------------------------------____----------------------------__-- > > Nathan Neulinger nneul(a)mst.edu <mailto:nneul@mst.edu> <mailto:nneul@mst.edu <mailto:nneul@mst.edu>> > Missouri S&T Information Technology (573) 612-1412 <tel:%28573%29%20612-1412> <tel:%28573%29%20612-1412> > > System Administrator - Architect > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH___wire <http://twitter.com/FreeSWITCH_wire> > > AIM: anthm > MSN:anthony_minessale@hotmail.__com <mailto:MSN%3Aanthony_minessale@hotmail.com> > <mailto:MSN%3Aanthony___minessale@hotmail.com <mailto:MSN%253Aanthony_minessale@hotmail.com>> > GTALK/JABBER/PAYPAL:anthony.__minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> > <mailto:PAYPAL%3Aanthony.__minessale@gmail.com <mailto:PAYPAL%253Aanthony.minessale@gmail.com>> > IRC: irc.freenode.net <http://irc.freenode.net> <http://irc.freenode.net> #freeswitch > > FreeSWITCH Developer Conference > > sip:888@conference.freeswitch.__org <mailto:sip%3A888@conference.freeswitch.org> > <mailto:sip%3A888@conference.__freeswitch.org <mailto:sip%253A888@conference.freeswitch.org>> > googletalk:conf+888@__conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> > <mailto:googletalk%3Aconf%__2B888@conference.freeswitch.__org > <mailto:googletalk%253Aconf%252B888@conference.freeswitch.org>> > pstn:+19193869900 <tel:%2B19193869900> > > > -- > ------------------------------__------------------------------ > Nathan Neulinger nneul(a)mst.edu <mailto:nneul@mst.edu> > Missouri S&T Information Technology (573) 612-1412 <tel:%28573%29%20612-1412> > System Administrator - Architect > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> > GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> > IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch > > FreeSWITCH Developer Conference > sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> > googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> > pstn:+19193869900 -- ------------------------------------------------------------ Nathan Neulinger nneul(a)mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect

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Status of mod_skinny? is it being maintained?
by Nathan Neulinger 14 Mar '13

14 Mar '13

Started looking around at the git log for mod_skinny after putting in a jira issue on it, and noticed that it hasn't been directly touched since around Dec 2011. Is it being worked on at all or is the lack of changes just due to "nothing really broke, but no new development taking place"? I'm looking at a possible large (1600+) phone deployment (replacement of old CCM deployment) using almost all SCCP based Cisco phones and just want to know what the status is since it looks like SCCP development on Asterisk is actively ongoing. I started with Freeswitch based on others recommendations, but if the core support for cisco isn't actively maintained, it would probably be a mistake for us to go that direction. -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul(a)mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect

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