February 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] Freeswitch for Wholesale routing
by David Villasmil 22 Feb '12

22 Feb '12

I did a front-end to manage OpenSIPs via web a long time ago, but never published it... gotta look for it. Looks like the following (pasted pages on top of pages), it's been on production about 4 year, not one restart... of course, low traffic: Cdrs are not rated, but shouldn't but difficult considering they're already in a table... [image: Inline image 2] On Tue, Feb 21, 2012 at 2:32 PM, Abid Naseem <bidnaseem(a)gmail.com> wrote: > Can we integrate opensips and viking? > > > On 21-Feb-2012, at 4:18 PM, David Villasmil < > david.villasmil.work(a)gmail.com> wrote: > > If there's no NAT involved, that's what i'd use... OpenSIPs, not kamailio > (in my case) > > On Mon, Feb 20, 2012 at 4:45 PM, Abid Naseem < <bidnaseem(a)gmail.com> > bidnaseem(a)gmail.com> wrote: > >> Hi David >> >> Lot of people recommend kamailio for wholesale routing. You think thats >> better thn freeswitch? >> >> Thanks, >> Abid. >> >> >> >> On 20-Feb-2012, at 7:43 PM, David Villasmil <<david.villasmil.work(a)gmail.com> >> david.villasmil.work(a)gmail.com> wrote: >> >> Hello Abid, >> >> 100,000+ CONCURRENT? >> >> David >> >> On Sun, Feb 19, 2012 at 4:24 PM, Abid Naseem < <bidnaseem(a)gmail.com><bidnaseem(a)gmail.com> >> bidnaseem(a)gmail.com> wrote: >> >>> Hi David! >>> >>> I am researching on some open source softswitches for a wholesale voip >>> operation (buy and sell routes); Basically I am planning to buy direct >>> routes in bulk and sell to resellers with small requirements. >>> It would be great help if you can tell me about the following: >>> >>> - Number of concurrent calls it can handle (freeswitch + viking) >>> - hardware requirements for 100,000+ concurrent calls (with media >>> by-pass) >>> >>> thanks for your time! keep inspiring. >>> >>> -- >>> Regards, >>> *Abid Naseem* >>> >>> +44 (793) 741 5043 >>> +1 (415) 562 6333 >>> >>> >>> >>> >>> >>> >> >

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mod_callcenter problem
by Wagner 22 Feb '12

22 Feb '12

Hello, I've set some groups in mod_callcenter, but when a call goes to a queue, the call keep playing the music on hold as if there are no agent available and I got this message in the log mod_callcenter.c:1049 Updated Agent 1001@default set status = Logged Out mod_callcenter.c:1049 Updated Agent 1001@default set uuid = mod_callcenter.c:1697 Agent 1001@default Origination Canceled : NONE mod_callcenter.c:1049 Updated Agent 1001@default set state = Waiting why the call isn't tranffered to the agent? Thanks

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Anyone else getting email scams on this list?
by Mario G 22 Feb '12

22 Feb '12

I have over 50 email addresses, and 1 just for the FS mailing list. It is the only one that gets scam emails. If anyone else is getting these I wonder if the list has been skimmed? Anyone else concerned by this? They are all trying to get you to click on links. They are fake but are from the alerts(a)bbb.org (6 in the last 2 days), 1 wanting tax ID information: "INTUIT INC." <onlinebanking(a)ealerts.bankofamerica.com>, multiple ones from Chine, etc. Mario G

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Notify of a missed call (no voicemail left)
by Ryan How 22 Feb '12

22 Feb '12

Hi, Is there any way to send an email notification of a missed call, no voicemail left?. Or even if a voicemail is left I guess, that isn't a biggy... just so if someone rings and hangs up, or rings and voicemail answers but no message. I'm thinking something that goes in here like: <action application="email" data="something here"/> <action application="bridge" data="user/${dialed_extension}@${domain_name}"/> <insert in here> <action application="answer"/> <action application="sleep" data="1000"/> <action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/> But that wouldn't capture when they hang up before voicemail answering would it ? I guess I could do something like application="socket" and then write an event socket program that sends an email. If there isn't already something out there that does it. Otherwise I'll leave this for another day coz it is a bit beyond my current magic ability, I need to level up :P Thanks again! Ryan

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Park an incoming call without answering it?
by Chris Martineau 22 Feb '12

22 Feb '12

Hi, Is it possible to somehow park or just make the call wait on the system without bridging it or answering it. I would like the outside party to still be in ringing state so I can the pick it up. Need something like a virtual ringing destination but cannot see how to implement? Any ideas Many thanks Chris

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Modify INVITE format
by liu rush 22 Feb '12

22 Feb '12

Dear all, Does anyone know how to modify the INVITE format of FS? Regard, Rush

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multi-FS setup and sip presence
by Vik Killa 22 Feb '12

22 Feb '12

So it appears i'm not the only person having issues with presence in a multi-FS setup... http://jira.freeswitch.org/browse/FS-3758 Currently presence is only works when a call comes to a UA on the FS server that UA is registered to... It looks like this patch was submitted to GIT but there is no clear instruction on how to implement it. Should a sofia parameter be used like <param name="shared-subscriptions" value="true"/> ? Also, I've been told i need to set this parameter: <param name="dbname" value="shared_presence"/> My question is what should the value of that parameter be? or does that not matter because FS automatically creates the SCHEMA? All my FS systems are connected to the same ODBC (in sofia.conf.xml and switch.conf.xml) And each FS system has a unique hostname. I'm told that if I set the hostname to the same on all FS boxes, presence will work, but at the same time I'm told setting the hostname of all FS boxes to the same is not good practice in a clustered setup. Also, I have the presence_id setup in the directory like <param name="dial-string" value="{sip_invite_domain=${dialed_domain},presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/> Any help would be much appreciated, thanks.

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(no subject)
by hjz51 22 Feb '12

22 Feb '12

Who have mod_sofia.c and switch_rtp.c source’s development documents. Can you give me to look at, thanks！

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mod_mongo usage
by ovvenkat 22 Feb '12

22 Feb '12

Hi to all, Any one help me to connect mongodb through mod_mongo module. thanks in advance. -- If you have come to help me, you are wasting your time. If you have come to because your liberation is bound up in mine, we can work together. Regards Venkatesan OV.

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Where to best get help with freeswitch on gentoo?
by Marcus Mülbüsch 22 Feb '12

22 Feb '12

Hello all, I make my first steps with freeswitch, and following the guide from http://wiki.freeswitch.org/wiki/Installation:Gentoo I try to get freeswitch running. However, depending on the USE-Flags set, or the FREESWITCH_MODULES used the compile fails; but I cannot find out why. For example, when I try the module "shout", compile fails with * ERROR: net-misc/freeswitch-9999 failed (compile phase): * failed to build FreeSWITCH * * Call stack: * ebuild.sh, line 85: Called src_compile * environment, line 6473: Called die * The specific snippet of code: * emake MONO_SHARED_DIR="${T}" || die "failed to build FreeSWITCH"; Googling around did not really lead me to anything useful. Is gentoo using the freeswitch ebuild still a (more or less) recommended way to run freeswitch? Can I ask more specific questions here, or somewhere else? Marcus

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