- FreeSWITCH-users - FreeSWITCH Mailing Lists

Freeswitch ignoring bye
by Simon Leck 18 Aug '11

18 Aug '11

Hi Everyone, Does anybody does why freeswitch is not acknowledging or ignoring BYE message and if so how do we fix it? Thanks in advanced to anyone who can tell me how to fix this issue. Thanks again Simon

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Dynamically switching DTMF mode?
by Dmitry Sytchev 18 Aug '11

18 Aug '11

Hi all I'm trying to achieve dynamic DTMF mode on incoming/outgoing calls. I have three profiles - one for outgoing/incoming calls to/from gateways, one for local registrations, and one for transferred call rerouting from bind_meta_app att_xfer. I faced some strange problems when tried to maintain inband DTMF path for call through FS. How should I set initial DTMF modes on profiles to allow changing DTMF mode on inbound calls? Is it enough to set dtmf_type to 'inband' and originate call? On which leg of call I should do this? And what to do with calls going through several profiles? As far as I understand, I need late negotiation turned on to disable automatic RFC2833 negotiation on inbound calls? -- Best regards, Dmitry Sytchev, IT Engineer

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Multiple Registration
by DJB International 18 Aug '11

18 Aug '11

I have some question regarding the multiple registration. When I registered 2 devices, even though both of them got registered and I can make calls from both devices, but when I list, I can only see the last device that got registered on the list from sofia status profile internal. However, when I comment out the multiple-registrations value=true in sofia profile, I can then see 2 devices listed. Questions: 1) Is that normal behavior that I need to make multiple-registrations value=true before I can see 2 devices listed when I run sofia status profile internal ? 2) How come I can only see one device listed even though both devices got registered in the first situation? Thank you, Dorn B.

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Re: [Freeswitch-users] [Freeswitch-dev] Meetup Paris?
by Giovanni Maruzzelli 18 Aug '11

18 Aug '11

I have fond memories of a conference I gave couple of years ago in Hyderabad and the nice reception has had. Certainly there are lot of people and companies that can be interested in FreeSWITCH, and Cluecon India. Most people is still on Asterisk, but I believe they'll be very interested into knowing first hand what are the advantages of using FS, maybe in powerful combination with OpenSips. Let's have the ball rolling, try to gauge what interest you can gather, and which companies can act as sponsors (money matters ;) ). -giovanni On Tue, Aug 16, 2011 at 8:28 AM, Prashant Lamba <prashant.lamba(a)gmail.com> wrote: > Colleagues, > After seeing the success of ClueCon, I have always been keen to promote open > source telephony here in India and possibly organize an "India" leg of > something like ClueCon here in Hyderabad. I can start the ground work of > gathering people, speakers, venue etc. > Hyderabad is probably the best technology hub here in India. Any > suggestions? > Prashant > -- > Prashant Lamba > Phonologies (India) Private Limited > e: prashant(a)phonologies.com > m: +91 9867.22.1975 > > On Fri, Aug 12, 2011 at 10:58 AM, Mathieu Rene <mrene_lists(a)avgs.ca> wrote: >> >> English version follows. >> >> En cette fin de ClueCon, je pars sur Paris demain soir (donc j'arrive >> samedi en matiné, bien décalé), et j'y resterai jusqu'à mercredi. Je >> propose d'organiser un petit meetup afin de joindre l'utile à l'agréable et >> de voir un peu le visage de la communauté freeswitch parisienne. Est-ce >> quelqu'un peut proposer un bar sympa ou on pourrait faire un truc? >> >> --- >> >> ClueCon is over and I'm leaving to Paris tomorrow (landing saturday >> morning, well jetlagged) and staying until Wednesday. I'm proposing to >> organize a little meetup in order to get to know the face of the parisian >> freeswitch community. Does anyone know a bar where we could meet up? >> >> Mathieu Rene >> Avant-Garde Solutions Inc >> Office: + 1 (514) 664-1044 x100 >> Cell: +1 (514) 664-1044 x200 >> mrene(a)avgs.ca >> >> >> >> >> >> _______________________________________________ >> Join us at ClueCon 2011, Aug 9-11, Chicago >> http://www.cluecon.com 877-7-4ACLUE >> >> FreeSWITCH-dev mailing list >> FreeSWITCH-dev(a)lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >> http://www.freeswitch.org > > > _______________________________________________ > Join us at ClueCon 2011, Aug 9-11, Chicago > http://www.cluecon.com 877-7-4ACLUE > > FreeSWITCH-dev mailing list > FreeSWITCH-dev(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618

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Re: [Freeswitch-users] SIP proxy collect DTMF using FS
by Sam Govind 18 Aug '11

18 Aug '11

Thanks for taking your time Gohar, I've looked at the link and this indeed seems like helping. Brain, can you suggest any other possibilities for this? I'd definitely like to go for the simple solution. On Wed, Aug 17, 2011 at 9:54 AM, Gohar Ahmed <gohar.ahmed(a)vopium.com> wrote: > Hey, **** > > ** ** > > Don’t know if this will do the job or not but do take a look at deflect > application: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_deflect > **** > > This will remove the FS from the call path and tell the originator to go > somewhere else. **** > > ** ** > > I hope this helps, maybe some other gurus here suggest you anything better > than this.**** > > ** ** > > Regards,**** > > Gohar A.**** > > ** ** > > *From:* freeswitch-users-bounces(a)lists.freeswitch.org [mailto: > freeswitch-users-bounces(a)lists.freeswitch.org] *On Behalf Of *Sam Govind > *Sent:* Monday, August 15, 2011 10:58 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] SIP proxy collect DTMF using FS**** > > ** ** > > Thanks Brian for showing concern. I'm always open for ideas. What I'm > trying to achieve is collect DTMF from user and then have my SIP Proxy > verify if a particular caller is allowed to dial that destination(input as > DTMF).**** > > ** ** > > Obviously I could've done the same checks at FS, BUT I'm *required *to let > SIP Proxy verify instead. FreeSWITCH is only required to get input, release > the call and if Proxy allows the call only then call be routed to any other > FreeSWITCH (Pool of FS LoadBalanced).**** > > ** ** > > This is supposed to simplify the operations of FS , decrease the load > volume on FS, and increase the call capacity.**** > > ** ** > > If you've any better Ideas do share.**** > > ** ** > > On Sun, Aug 14, 2011 at 8:39 AM, Brian West <brian(a)freeswitch.org> wrote:* > *** > > What are you trying to do exactly because it sounds like you have selected > the most painful way to accomplish it?**** > > ** ** > > /b**** > > ** ** > > On Aug 9, 2011, at 2:11 AM, Sam Govind wrote:**** > > > > **** > > Hi guys, > > I'm looking to establish a scenario like this, any idea how to do it, if > its > possible. > > 1- SIP proxy send call to FS where DTMF will be collected. (I'm thinking of > using PlayAndGetDigits) > 2- DTMF collected be sent back to SIP proxy while FS ends the call > 3- Call at SIP proxy end keeps running for some other processing. > > basically I just need FS to collect DTMF and send those back to SIP Proxy. > > Any ideas are welcome. > > Thanks.**** > > ** ** > > > _______________________________________________ > Join us at ClueCon 2011, Aug 9-11, Chicago > http://www.cluecon.com 877-7-4ACLUE > > 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**** > > ** ** > > > 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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Bad port when using rport
by Spencer Thomason 17 Aug '11

17 Aug '11

Hello all, Our FS server is on a public IP on a dedicated server at one location and all clients are behind nat or vpn from other locations. We have a few "road warrior" clients when our employees travel. I'm seeing the following from one of the client devices. I'm not sure of any of the details of the router (or have anyway to control it) but for some reason the rport is sent with an invalid port number. Notice the 8 at the end of the port number is duplicated. I'm seeing the same behavior from both a Bria client and Zoiper. Is there any way to work around this? An excerpt from Wireshark: 466 2011-08-17 11:36:04.432912 80.34.4.95 206.125.40.171 SIP Request: REGISTER sip:x.x.x.x Internet Protocol, Src: 80.34.4.95 (80.34.4.95), Dst: 206.125.40.171 (x.x.x.x) User Datagram Protocol, Src Port: 56238 (56238), Dst Port: sip (5060) Session Initiation Protocol Request-Line: REGISTER sip:x.x.x.x SIP/2.0 Message Header Via: SIP/2.0/UDP 80.34.4.95:562388;rport;branch=z9hG4bKPjSAbzV32ywqJ8xWjWLx81Ml6oQcNlrM.s Max-Forwards: 70 From: "XXXX" <sip:1001@x.x.x.x>;tag=rPFQBJFIKAQpE88mHeiZ7ic59suQY70w To: "XXXX" <sip:1001@x.x.x.x> Call-ID: awr-KAMoJpMYEt2mqkOkknkRrCXUj.Jv CSeq: 27137 REGISTER User-Agent: Bria iPhone 1.2.12 Contact: "XXXX" <sip:1001@80.34.4.95:562388> Expires: 900 Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Content-Length: 0

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Inband DTMF when rfc2833 negotiated?
by Avi Marcus 17 Aug '11

17 Aug '11

Why is this call using inband when the SDP says rfc2833? I got complaints that the DTMF wasn't working, despite the dtmf numbers in the log being correct. Trace on leg B: http://pastebin.freeswitch.org/17053 Thanks, -Avi p.s. I think I've seen this with toll free gateway, too.

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Asterisk to FreeSWITCH migration guide
by Nestor A Diaz 17 Aug '11

17 Aug '11

Hi Guys. I am starting to use FreeSWITCH, i am an asterisk user since the 1.0.7 release appears as a package on the debian distribution, at the beginning i was amazed by the fact i can build a PBX for my own business and i did, later i began to install this system for my customers and sooner i meet the problems, however being the software open source i always find a way to fix things using patchs from others, sometimes i felt how my life was at risk when the system stops working and that usually happens when i have to use queues and dealing with digium hardware. Fixing those problems either by applying patches or by changing the hardware where the digium cards were supposed to be installed helps me, but that was to much stress for me and seeking for a balance that will let me invest more time on services, configuration and hoping to have better hardware options brings me to freeswitch. I agree with freeswitch philosophy that instead of having thousands of modules that don't work fine i prefer just a few that works the way it should be, a rock solid system for a corporate pbx and a call center is what i want. So here i am trying to begin the conversion, and i hope the information we can transcript in this list will help others that want to try another alternative to asterisk. First of all i think the saner for a migration is to have the two systems running either on the same machine or different and use the stable features of each one. So could you please freeswitch users help me with this rosetta stone migration guide in order to post it to voip-info.org or freeswitch wiki (i list only the ones i currently use ): *Technology* *Asterisk* *Freeswitch* PSTN Connectivity (Digium / Sangoma) dahdi freetdm IAX2 mod_iax ?? none stable yet. Use Asterisk to forward traffic via SIP. Enable Hardware HPET for IAX2 trunk if card not available for Asterisk Bluetooth Channel chan_mobile ?? Use asterisk via SIP Skype Skypeforasterisk (no longer for sale) mod_skypeopen CDR Stadistics Arternic cdr-stats ?? Queue Statistics Asteriskguru queue-stats ?? Web Management Freepbx ?? IVR AGI / AMI Event Socket Codec G.729 Transcodind Cards G.729 licenses Free G.729 (Intel IPP) Transcodind Cards G.729 licenses fsg729 Intel IPP(any experience with it ? ) Fax Handling Iaxmodem with Hylafax ?? Iaxmodem via asterisk to FS via SIP ? SIP chan_sip sofia ACD app_queue mod_callcenter Thank you all -- Nestor A. Diaz Ingeniero de Sistemas Tel. +57 1-485-3020 x 211 Cel. +57 316-227-3593 Tel. SIP: sip:211@tiendalinux.com Email/MSN: nestor(a)tiendalinux.com http://www.tiendalinux.com/ Bogota, Colombia

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Re: [Freeswitch-users] getting disconnect cause for a leg after bridge in Lua
by Mi Ke 17 Aug '11

17 Aug '11

I see ... Thanks, Michael ----- Original Message ----- From: Michael Collins Sent: 08/17/11 09:33 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] getting disconnect cause for a leg after bridge in Lua Well, it's possible if you're using an event socket program, however if you are using a dialplan script you simply cannot bend the laws of physics to magically know what's going on with the b leg when your program is "inside" the a leg. I recommend that you use a socket-based program if you want this level of control. A socket-based program gives you the perspective that you need: you can "oversee" multiple call legs. Just a thought... -MC On Wed, Aug 17, 2011 at 7:51 AM, Mi Ke < mi.ke(a)null.net > wrote: Thanks for replying, Michael. Meanwhile if session creation fails, such session can also be considered as disconnected and gone, however its disconnect cause can be read from session:hangupCause(). To my opinion, it would be very useful if disconnect cause could be always readable in Lua after session completion since such mechanism already exists for failed sessions. Thanks / Mike ----- Original Message ----- From: Michael Collins Sent: 08/16/11 03:03 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] getting disconnect cause for a leg after bridge in Lua No, you can't do this because the session you are checking is "gone" as soon as the call leg is disconnected. You are better off using a hangup hook or an event socket application if you need to get that value in realtime. Dialplan scripts are good for connecting endpoints and doing simple logic but they are absolutely not what you want for doing any kind of billing or reporting. -MC On Sat, Aug 13, 2011 at 2:46 AM, Mi Ke < mi.ke(a)null.net > wrote: Hi All, Is there any way to get a real disconnection cause for leg B in the following script ? if (session_a:ready() and session_b:ready()) then freeswitch.bridge(session_a,session_b) -- session_b gets disconnect here ... local session_b_hangup_cause = session_b:hangupCause() session_b_hangup_cause is always "SUCCESS" after debridging while log and CDR shows correct value - can get it to my script ? Thanks / Mike _______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE 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 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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Re: [Freeswitch-users] getting disconnect cause for a leg after bridge in Lua
by Mi Ke 17 Aug '11

17 Aug '11

Thanks for replying, Michael. Meanwhile if session creation fails, such session can also be considered as disconnected and gone, however its disconnect cause can be read from session:hangupCause(). To my opinion, it would be very useful if disconnect cause could be always readable in Lua after session completion since such mechanism already exists for failed sessions. Thanks / Mike ----- Original Message ----- From: Michael Collins Sent: 08/16/11 03:03 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] getting disconnect cause for a leg after bridge in Lua No, you can't do this because the session you are checking is "gone" as soon as the call leg is disconnected. You are better off using a hangup hook or an event socket application if you need to get that value in realtime. Dialplan scripts are good for connecting endpoints and doing simple logic but they are absolutely not what you want for doing any kind of billing or reporting. -MC On Sat, Aug 13, 2011 at 2:46 AM, Mi Ke < mi.ke(a)null.net > wrote: Hi All, Is there any way to get a real disconnection cause for leg B in the following script ? if (session_a:ready() and session_b:ready()) then freeswitch.bridge(session_a,session_b) -- session_b gets disconnect here ... local session_b_hangup_cause = session_b:hangupCause() session_b_hangup_cause is always "SUCCESS" after debridging while log and CDR shows correct value - can get it to my script ? Thanks / Mike _______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE 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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