June 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] DTMF Passthrough on different legs
by JRichey 02 Jun '12

02 Jun '12

What CODECs are being used? In-band DTMF won't work correctly with compressed CODECs like G.729. -----Original Message----- From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org]On Behalf Of Michael Lutz Sent: Thursday, May 31, 2012 2:12 AM To: FreeSWITCH Users Help Subject: [Freeswitch-users] DTMF Passthrough on different legs Hi Guys, I am stuck with a bit of a problem I cannot figure out where it is going wrong.. Here's my situation, I have an inbound call, comming from an external provider, they don't support RFC2833 so I turn on inband detection using 'start_dtmf' in the dialplan. This works ok, for the incomming channel and IVR provided (via Lua) to the caller. (Though sometimes it seems to misinterpret and returns wrong keys..) (Session:A) After that, this call is bridged (via Lua to a a new inbound session on the same switch) (without specifying start_dtmf) so RC2388 is used by default (?) (Session:B) This new sessions is playing soem audio (Lua) and is heard by the original caller (the two sessions are bridged). Seperatly I have an inbound ESL connection where I do an origanate using a originate {execute_on_answer=start_dtmf, .... etc) &park() to setup an new external call via the same gateway (they don't support RFC2388 so I activate the inband detection on the b-leg to receive the digits. Let's call this leg Session:C. After the orignated call is answered I bridge the parked call with the previous session (Session:B), so at the end Session:A and Session:B can talk to each other. Al this works perfectly. [Incommming call] Session:A ----- INBAND ------> Bridge(FS.Lua) ---> Session:B ---- RFC2388 ----> FS.Bridge ---> Session:C --- INBAND ------> ESL destination number My symptoms are... 1. I have a customer who had a IVR on his own side, (Session:C) and requires input from Session:A. The weird thing is that it seems the input is wrong. Keys entered do not correspond to what his IVR is receiving. As if newly dtmfs are generated somewhere. When I switch to another inbound provider, making Session:A using rfc2388 and not inband, but do dialout via the provider using inband detection. It seems to work fine. 2. Not being able to get the dtmf's from the ESL Destination number (Session:C). I'm reading this by subscribing to events via ESL for that particulair uuid. When I have a connection (inbound) without inband detection it seems to get the DTMF's from Session:C without any problems. Which makes it weird for me. note: I know the setup is a bit odd, but Session A cannot exit it's Lua script, so I needed to do a workaround to be able to keep te lua alive and being able to control the call (audio) via ESL. I realy need some help on this one, becuase we have a lot of customers complaining about not responding to dtmf's or receving invalid input in the (FS) IVR. Thanks for you help, Regards, Mike. _________________________________________________________________________ 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 Join Us At ClueCon - Aug 7-9, 2012 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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mod_soundtouch
by Yuriy Nasida 02 Jun '12

02 Jun '12

Hi guys. I test mod_soundtouch. All work fine but how can I reduce delay of transmiting of voice ? Now I wait about 5 sec and I would like to have realtime. Please adviseThanks.

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NAT issues and "best practices"
by Sean Devoy 01 Jun '12

01 Jun '12

HI All, I have a customer location that has just been a nightmare to implement. I am just learning that they "may" have multiple NAT routers in sequence at their location. I think I fully understand what NAT is trying to accomplish. There seem to be different levels and approaches. The most basic NAT setup (to me) is a HOME LAN with multiple PCs where NAT allows multiple devices to share a single routable IP address on the WAN side from multiple local devices the LAN side. Note I said OUTBOUND initiated connections. Even FTP can have trouble with this level. Almost all inbound traffic is blocked for security. Clearly for FS we need the switch to be able to punch through from the WAN to specific local IPs on the LAN to reach specific phones. This is INBOUND NAT and brings up many security issues for people. Even on devices where you get this "working" you may only be able to support one line per phone or a single inbound connection at a time. I understand NAT has PMP and UPnP protocols and FS "supports" both. What I can't find is where someone says "Here is a great setup that works with cheap, available "commodity" hardware from Cisco/Linksys that supports all the NAT you need for FS." I don't care if it is PMP or UPnP and I might not even care why you pick one over the other, although it is probably a "good read". Can someone just stand up say "FS works GREAT with the XYZ router in ABC mode from MY COMPANY using NAT to Cisco phones"? I have seen some articles about Freeware/Shareware firmware in this devices, but as a novice I want to limit the unknowns until I get more up to speed. My specific issue now is that I cannot get SCA to work at the NAT location. I issued: sofia_contact 220(a)mydomain.com sofia/external/sip:200@<ip address>:44234,sofia/external/sip:200@<same ip>:1024 Only one phone rings on inbound and the line indicator light does not change when either is picked up. Same configuration is working on our LAN with the switch. I am absolutely ready to by a router to fix these issues, I don't want to lose this customer. Thanks for your thoughts, Sean

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Custom Registration
by alberto Villa 01 Jun '12

01 Jun '12

Hello, I'm using FreeSWITCH in a very custom system and I need to setup this kind of registration: There are two SIP phones A and B: when each SIP phone have its own unique ID the registration to freeswitch goes as usual and no change is needed, but when A and B share the same ID I would like to reject the registration for the last SIP phone. So, if A and B have the same ID and A is registered first, I would like to reject the subsequent registration of B. I tried to change the values of some configuration parameters such as "sip-allow-multiple-registrations" and "max-registrations-per-extension" but it didn't work (by the way I'm not shure to have edited properly this parameters). How can I achive this kind of registration? Thank you Alberto -- Dr. Villa Alberto Sw Engineer SeleTech srl via Collodi, 8 20052 Monza (MI) tel: +39 039 5962000 fax: +39 039 9716905 email: a.villa(a)seletech.com web: www.seletech.com www.seletech.eu

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FreeTDM CallerID Detection with DTMF
by Antonio Modesto 01 Jun '12

01 Jun '12

Hi, I have a Digium TDM410P card, I am using it here in Brazil, where the signaling is not FSK, it's DTMF. It is working, though I am not receiving the callerid: Caller-Caller-ID-Number: [0000000000] Here is my freetdm.conf: [span zt FXO1] fxo-channel => 1 [span zt FXO2] fxo-channel => 2 [span zt FXO3] fxo-channel => 3 Here is one section of my autoload_configs/freetdm.conf.xml: <span name="FXO1">    <param name="tonegroup" value="br"/> <param name="digit-timeout" value="2000"/> <param name="max-digits" value="11"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="enable-callerid" value="true"/>     </span> Is it possible to enable the callerid detection in these conditions, or is it a hardware/driver limitation? Regards.

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Spanish prompt: "Para espanol, oprima el numero dos" or similar
by Michael Collins 01 Jun '12

01 Jun '12

Hey, does anyone have a nice little prompt in a female voice that says something like "for Spanish, press..."? I'm working on some generic IVR stuff and I'd love to be able to mix in something like that. Thanks, MC

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FSClient a Windows FreeSWITCH softphone Update Released
by Mitch Capper 01 Jun '12

01 Jun '12

An update had been posted for FSClient is a windows .NET softphone that uses libfreeswitch at the core. Info page at: http://wiki.freeswitch.org/wiki/FSClient Direct binary download: http://files.freeswitch.org/windows/installer/x86/FSClient.zip And source is in contrib repo under MitchCapper/FSClient Some of the changes include: Built against FS 1.2 RC isac codec support for contacts you can now right click and choose what account to call from if sofia goes offline update the accounts status Jabra headset improvements to avoid some crash conditions background task improvements to avoid the phone breaking without you knowing Feedback always welcome. With ZRTP into core now the next version may have some ZRTP related features. ~Mitch

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Re: [Freeswitch-users] Access to Cause filed in Reason header (Michael Collins)
by Javier Gallart 01 Jun '12

01 Jun '12

Thanks Michael I did exactly that, an "info" application is executied right ater the bride and the variable is not there. I also tried firing an event when B-leg ends and none of the headers showed the contents of the Reason Header. Regards Javi > ---------- Forwarded message ---------- > From: Michael Collins <msc(a)freeswitch.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Thu, 31 May 2012 11:16:46 -0700 > Subject: Re: [Freeswitch-users] Access to Cause filed in Reason header > I believe that information is available in other channel variables. I'd do > an info dump right after the bridge and see what shakes out. > -MC > > On Thu, May 31, 2012 at 8:51 AM, Javier Gallart <jgallartm(a)gmail.com>wrote: > >> Hello list >> >> some of our providers include a Reason header in negative replies to an >> INVITE. For instance, they might include a cause=65 inside a Sip 503 reply. >> We need to react upon some of those ISDN causes: in this particular case >> with if Reason header were absent we would simply do nothing but let FS to >> send the response upstream; but if cause=65 in present we want to resend >> the call to the same provider with a different set of offered codecs. Is >> there any way to access to the content of that header?. Our applicatiion >> listens to the event socket in inbound mode but I haven't found a way to do >> this. Any help would be appreciated. >> >> Regards >> >> Javi >> >> > > >

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calls to australian geo numbers for load test
by Chris Mylonas 01 Jun '12

01 Jun '12

hi fs users, is it possible to bounce signalling from an overseas provider and keep media local? i need to find reasonable rates for load testing to australian geo numbers - concurrent channels + rate per minute not fussy about which country the calls originate. i was thinking to work my way through this list of providers http://wiki.freeswitch.org/wiki/SIP_Provider_Examples to tally up costs probably won't happen til july/august but just getting some notes down whilst it's raining and there's nothing for me to do except dev... cheers chris

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