March 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

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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Re: [Freeswitch-users] FreeTDM configuration
by ashish gautam 14 Mar '13

14 Mar '13

Hi Michael, I get this error " 2013-03-14 12:33:19.476237 [NOTICE] switch_ivr_originate.c:2636 Cannot create outgoing channel of type [freetdm] cause: [DESTINATION_OUT_OF_ORDER] 2013-03-14 12:33:19.476237 [DEBUG] switch_ivr_originate.c:3601 Originate Resulted in Error Cause: 27 [DESTINATION_OUT_OF_ORDER] " On Thu, Mar 14, 2013 at 12:19 PM, < freeswitch-users-request(a)lists.freeswitch.org> wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. FreeTDM configuration (ashish gautam) > 2. Re: SaltStack + deployment techniques (Gabriel Gunderson) > 3. Re: Recording channels input and output voice in two separate > files. (Michael Collins) > 4. Re: FreeTDM configuration (Michael Collins) > 5. Re: SaltStack + deployment techniques (Henry Huang) > 6. Re: Recording channels input and output voice in two separate > files. (Nikhitha) > > > ---------- Forwarded message ---------- > From: ashish gautam <gautamashish09(a)gmail.com> > To: freeswitch-users(a)lists.freeswitch.org > Cc: > Date: Thu, 14 Mar 2013 10:47:06 +0530 > Subject: [Freeswitch-users] FreeTDM configuration > Hi, > > I have setup freetdm-dahdi-libpri stack. When I run ftdm list command it > does not show any output. I guess the channels have not been configured > properly. I am not able to figure out what is happening. Please help me > out. > > I am using a Digium single span E1 card. Dahdi and libpri have been > properly installed and are working fine. > > -- > REGARDS > ============================================ > *Ashish Gautam* > (+918802865008) > > > > ---------- Forwarded message ---------- > From: Gabriel Gunderson <gabe(a)gundy.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Wed, 13 Mar 2013 23:47:58 -0600 > Subject: Re: [Freeswitch-users] SaltStack + deployment techniques > On Wed, Mar 13, 2013 at 7:59 AM, Cal Leeming [Simplicity Media Ltd] > <cal.leeming(a)simplicitymedialtd.co.uk> wrote: > > 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. > > Sounds like you're getting close to being happy :) > > When we do some Salt Modules and States specific to OpenSIP and > FreeSWITCH, we'll be sure to open source them. > > Happy hacking! > > > Gabe > > > > > ---------- Forwarded message ---------- > From: Michael Collins <msc(a)freeswitch.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Wed, 13 Mar 2013 22:50:42 -0700 > Subject: Re: [Freeswitch-users] Recording channels input and output voice > in two separate files. > You can use an api_hangup_hook to call system and do two sox commands: > > sox infile.wav outfile.left.wav remix 1 > sox infile.wav outfile.right.wav remix 2 > > -MC > > On Wed, Mar 13, 2013 at 9:52 PM, Nikhitha <nikhitha.voxta(a)gmail.com>wrote: > >> Thanks for the reply... >> In the path we are specifying the path where the recorded file should be >> stored, here we are only specifying a single wav file path then how two >> files will get recorded one as inbound and one as outbound?? >> >> >> >> -- >> View this message in context: >> http://freeswitch-users.2379917.n2.nabble.com/Recording-channels-input-and-… >> Sent from the freeswitch-users mailing list archive at Nabble.com. >> >> _________________________________________________________________________ >> 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 >> > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > > ---------- Forwarded message ---------- > From: Michael Collins <msc(a)freeswitch.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Wed, 13 Mar 2013 22:54:10 -0700 > Subject: Re: [Freeswitch-users] FreeTDM configuration > You need to look at the log output when mod_freetdm is being loaded. There > are several ways to do that, but the quickest is probably just to rotate > log files, restart freeswitch, then rotate log files again. You'll have a > file like: > freeswitch.log.2013-03-14-10-45-30.1 > > Look in there for errors when mod_freetdm is loading. Sometimes those > errors point out the obvious, like it can't find the card or the DAHDI > drivers aren't loading/responding, etc. > > -MC > > On Wed, Mar 13, 2013 at 10:17 PM, ashish gautam <gautamashish09(a)gmail.com>wrote: > >> Hi, >> >> I have setup freetdm-dahdi-libpri stack. When I run ftdm list command it >> does not show any output. I guess the channels have not been configured >> properly. I am not able to figure out what is happening. Please help me >> out. >> >> I am using a Digium single span E1 card. Dahdi and libpri have been >> properly installed and are working fine. >> >> -- >> REGARDS >> ============================================ >> *Ashish Gautam* >> (+918802865008) >> >> >> _________________________________________________________________________ >> 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 >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > > ---------- Forwarded message ---------- > From: Henry Huang <red.rain.seven(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Wed, 13 Mar 2013 23:36:27 -0700 > Subject: Re: [Freeswitch-users] SaltStack + deployment techniques > It's the first time I hear about SaltStack. Can you briefly explain why is > it better than Puppet or Chef. I would pick Chef out of the 2 because it's > using Ruby natively and is being adapted by Amazon AWS. So if you ever need > some kind of hybrid architecture to run production or development servers > on AWS, you will spend minimal effort for deploying those given that you > can reuse your Chef cookbooks. > > Henry > > > On Wed, Mar 13, 2013 at 10:47 PM, Gabriel Gunderson <gabe(a)gundy.org>wrote: > >> On Wed, Mar 13, 2013 at 7:59 AM, Cal Leeming [Simplicity Media Ltd] >> <cal.leeming(a)simplicitymedialtd.co.uk> wrote: >> > 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. >> >> Sounds like you're getting close to being happy :) >> >> When we do some Salt Modules and States specific to OpenSIP and >> FreeSWITCH, we'll be sure to open source them. >> >> Happy hacking! >> >> >> 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 >> > > > > ---------- Forwarded message ---------- > From: Nikhitha <nikhitha.voxta(a)gmail.com> > To: freeswitch-users(a)lists.freeswitch.org > Cc: > Date: Wed, 13 Mar 2013 23:49:35 -0700 (PDT) > Subject: Re: [Freeswitch-users] Recording channels input and output voice > in two separate files. > Without using sox ,do we have any thing that we can directly do with > freeswitch to record them in seperate files? > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Recording-channels-input-and-… > Sent from the freeswitch-users mailing list archive at Nabble.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 > > -- REGARDS ============================================ *Ashish Gautam* (+918802865008)

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Advice on scalable design pattern
by Cal Leeming [Simplicity Media Ltd] 14 Mar '13

14 Mar '13

Hello all, I'm currently looking at the various different ways you can deploy FreeSWITCH in a scalable manner, but struggle a little bit on design. The sweet spot I'm trying to find is one where I can scale out capacity by simply throwing more servers at it. In an ideal world, this would mean support for; * Have multiple users from multiple domains to be spread over multiple servers... a single domain should not be restricted to a single FreeSWITCH instance * Have no single point of failure within the structure * Have no single point of bottleneck within the structure * Should not use OpenSIPS.. (I suspect this might get me a lot of flack, but seriously, I'd rather write my own in python or ZXTM traffic script than use OpenSIPS lol). So far, the best option I can come up with is (although I'm not sure if it's the best available); * Proxy sitting in front of all backend FreeSWITCH instances, acting in a media proxy fashion only (dual pair of proxies in active/passive mode) * Proxy tracks registrations to the appropriate backend instance, and makes their session sticky * If backend instance needs to make a call to another user in the same domain, it bridges to the call to back to the proxy, the proxy then determines which other FreeSWITCH instance has the user then routes the request accordingly. If the call is to an external destination, the proxy will route it to the traffic aggregation switches (which is basically another pair of FreeSWITCH instances), which then gets routed to the upstream provider.. this means you only have to maintain 2 sets of trunk configuration.. so when you need to scale out your freeswitch backends, it doesn't require putting in a request to your upstream providers for an additional set of trunks. * The bottleneck within these clusters is the dual proxies in active/passive mode.. you could fix this by allocating customers to a specific cluster (rather than instance), thus controlling which customers go to which proxy.. if an entire cluster dies, you can re-route that clusters traffic to a different cluster. The other simpler option is to allocate domains to a specific backend instance.. but this really doesn't feel clean.. it means a customer cannot scale past the floor limit of a single instance, it has less redundancy, and overall just feels wrong. Any general thoughts/comments on this would be much appreciated. Thanks Cal

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