- FreeSWITCH-users - FreeSWITCH Mailing Lists

Recommend an IP Phone for use behind NAT
by Alan Schmitz 25 Feb '12

25 Feb '12

I am looking into changing my VOIP server to FreeSwitch. My current system uses Polycom SoundPoint IP 335 phones. The server has a public IP, but several of the phones are behind a variety of DSL routers that do NAT. These remote users have reported intermittent problems with one way audio. What IP desk phone would you recommend for a remote user behind a NAT router with a dynamic IP? I'm looking for something of comparable audio and build quality to the Polycoms I currently have. Thanks, Alan

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RECOVERY_ON_TIMER_EXPIRE
by Ira Tessler 24 Feb '12

24 Feb '12

Can someone tell me what a hang up cause of RECOVERY_ON_TIMER_EXPIRE means? I have a customer complaining of dropped calls, and this seems to be the hang up cause. Thanks, Ira Tessler ConnectMe (732) 490-9007 x2 www.connectmevoice.com

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Forcing ptime for A Leg and B Leg call
by Arafath-uz-zaman khan 24 Feb '12

24 Feb '12

Hello Can anyone guide me on following Client----120ms----->FS----20ms---->GW Client<----50ms------FS<----20ms----GW Sip Client will use G729 codec with 120 ptime, Freeswitch will send the call to GW using G279@20 ptime. Freeswitch will send receive call from RTP with G729@20 ptime and will send it to Sip Client using G279@50ptime. Sip Client will be registered at FS using internal SIP Profile. I will use Sangoma transcoding card for it. Can anyone help me to tell the way what i have to do in sip profile and dialplan. With Regards -- Arafath-uz-zaman khan

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Wiki article: XML User Directory Guide and "alphanumeric to numeric user mapping"
by Anton Kvashenkin 24 Feb '12

24 Feb '12

I think that there is a mistake at http://wiki.freeswitch.org/index.php?title=XML_User_Directory_Guide&diff=26… revision. It should be <action application="bridge" data="user/${user_data(${destination_number}@${domain_name} attr id)}@${domain_name}"/> and NOT <action application="bridge" data="user/${user_data(${destination_number}@${domain_name} attr id)}"/> i.e. with @${domain_name} freeswitch@internal> user_data 1002(a)XXX.XXX.XXX.XXX attr id 1002 freeswitch@internal> user_data 1000(a)XXX.XXX.XXX.XXX attr id jugatsu freeswitch@internal> <user id="jugatsu" number-alias="1000"> <user id="1002">

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Speech Phrases
by Brian Foster 24 Feb '12

24 Feb '12

I'm trying to figure out how phrases works in freeswitch, and I have a set of my own voices to use instead of the default ones. I've read http://wiki.freeswitch.org/wiki/Speech_Phrase_Management but I guess I'm not understanding correctly. Situation (might be easier to understand this way): Basically I have a set of prompts (they are actually in my voice) that I plan on using to read off addresses and such from a SIP message. Think "automated dispatching" for fire departments. A script might be: (fire tones) "Indianapolis Engine 123, Indianapolis Medic 123, 928 West Banta Road, Vehicle Fire" x2 "Coordinates 3700 North, 1200 East, Assigned to Marion Ops 1" ...so that information would be set in some variables, like responding_units, address, incident_type, coordinates_1, coordinates_1_direction, coordinates_2, coordinates_2_direction, etc... So the million dollar question is, how do I (temporarily) get freeswitch to use my voice prompts instead of the default prompts when using the say command in the dialplan? Also, how should I organize those prompts so that freeswitch can find them? -- Brian D. Foster Endigo Computer LLC Email: bdfoster(a)endigotech.com Phone: 317-429-1069 Indianapolis, Indiana, USA This message contains confidential information and is intended for those listed in the "To:", "CC:", and/or "BCC:" fields of the message header.If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version.

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Brian West quick update
by Ken Rice 24 Feb '12

24 Feb '12

Hey guys, wanted to pass on some info about Brian aka BKW He's still sitting in the hospital in Tulsa, tomorrow, he will go under the knife of a surgeon to remove the offending cyst. so here we are going on another week with Brian away... He'll probably be there are another week... (Docs are saying 5 to 7 days after the surgery before he goes home) Keep i mind the VoiceNetworks offer, I believe he's not even 1/2 way there yet. so everyone fire up those paypal accounts and hit him up... if you wish to send him something in another way, see the Jira Ticket BKW-1 via http://jira.freeswitch.org/browse/bkw-1 all the good info is there He is rapidly running out of sick leave and it looks like they are going to put him on Leave of Absence from work... this means his salary stops... so lets help him out guys K

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xmlcdrd start
by Miha Zoubek 24 Feb '12

24 Feb '12

Hi, I have installed modul xmlcdrd (http://wiki.freeswitch.org/wiki/Xmlcdrd)with Nazim help. Can some one please help me how to stat this modul, what I must do after installed and plugin configuration? Thanks! miha

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Long call duration - Hangup issue
by Balamurugan Mahendran 24 Feb '12

24 Feb '12

HI all, While dialing-out (outbound call) if the call is terminated by the calling party(other side), then the call will never hang-up that reflects long call duration in CDR. Please let me know if Iam missing anything. Thanks, Bala

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can i get gtalk account online status on mod_dingaling?
by 熊子奇 24 Feb '12

24 Feb '12

HI All, im using mod_dingaling to connect to google talk, and the call from freeswitch to remote gtalk account has been done successfully. now here raise a question, can i get the remote account online/offline status before i call it?thanks for your attention. if any reply, im really appreciate . regards.--xiong

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Re: [Freeswitch-users] Overhead paging
by George Sapak 23 Feb '12

23 Feb '12

I have taken a look at that, the client is a non profit, we would like to do it on the cheap, we do have some ht286, just though that could be hacked instead of spending almost $400 CAN Thank You, George. ----- Original Message ----- From: "Tim Nelson" <tnelson(a)rockbochs.com> To: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org> Sent: Thursday, February 23, 2012 9:44:55 AM Subject: Re: [Freeswitch-users] Overhead paging Yes, the product referenced would allow your VoIP implementation (Freeswitch) to work with your existing paging system. Did you read the info on the link I sent? --Tim ----- Original Message ----- > there is already a paging system in place. > > Thanks, George. > > > ----- Original Message ----- > From: "Tim Nelson" <tnelson(a)rockbochs.com> > To: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org> > Sent: Thursday, February 23, 2012 8:39:14 AM > Subject: Re: [Freeswitch-users] Overhead paging > > ----- Original Message ----- > > what is the easiest way to set this up, just need a way to pickup a > > set and have an amp output going from the switch going to the paging > > system, I have read somewhere that you could use HT503 for this, > > sounds confusing. > > > > You could use a VoIP to analog paging gateway, such as those made by > Cyberdata. See here: > > http://www.cyberdata.net/products/voip/legacyanalog/index.html > > I've used their products quite a bit recently, and have had fantastic > results. In the rare event I required support, the experience was top > notch.*** > > --Tim > > ***Please keep in mind I'm not affiliated with them in any way, but I > find the need to 'toot my horn' about a product/company that exceeds > my expectations as it is becoming a rarer thing these days. > > _________________________________________________________________________ > 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 > > _________________________________________________________________________ > 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 _________________________________________________________________________ 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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