July 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

dtmf in Skype
by luixsansan＠hotmail.com 09 Jul '10

09 Jul '10

Hello, I have FS ver. 1.0.6, Windows Vista Ultimate and Skype 4.2.0.169 for Windows. (user1) On a laptop I have Windows Vista Home and Skype 4.1.0.179 (user2) When I make a call from user2 to user1 I listen the IVR menu and see the activity on FS console. The problem is that when I click on the skype's phone keyboard I can not listen to any DTMF signal and user1 is not receiving DTMF signals ( the IVR menu is not interrupted). Is there a configuration change I have to do in order to make skype send/receive dtmf signals or do I need another type of skype client? The info in the document for mod_skypopen is not very clear. Thank you. Luis.

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Two FreeSWITCH Box IVR Call Transfer
by Jigar Thakkar 09 Jul '10

09 Jul '10

Hi Guys, I am a beginner to FreeSWITCH. I have few questions. 1. What is the difference between BRIDGE, ORIGINATE and TRANSFER. 2. I have two FreeSWITCH box. BoxA and BoxB 1. User will make a call to BoxA; BoxA's IVR collect some information i.e. userid and pin. 2. Now BoxA will forward/transfer/bridge call to BoxB's IVR ( here i want to pass userid as channel variable) 3. Box B will ask for another pin i.e. pin2, verifies the same and pass whether pin2 is correct or not to BoxA (also transfers the call to boxA's ivr) 4. Box A shall play message to User whether this process is successful or not. And then call shall be hanged up. My work till date is 1. setup the two freeswitch ivr, box a and box b. 2. I wrote .net code to handle the individual role. i.e. to ask user id and pin 3. I facing the problem in forwarding a call from boxA to boxB along with the userID Please suggest the best way to implement it. I hope I have explained the scenario properly. Thanks & Regards, Jigs.

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FS 1.0.6 on EC2 - no audio on outgoing calls
by paul gore 09 Jul '10

09 Jul '10

Hi there, I am experimenting with FS on EC2, I like results, but stuck on weird audio issue - I followed FreeSwitch EC2 wiki article and modified internal profile and vars.xml accordingly, but unfortunately still cannot get it working. Incoming and outgoing calls made using a SIP phone to FS extensions work just fine. As well as calls to FS from PSTN. But calls to PSTN via gateways result in no audio at all, no ring, nothing, SIP signaling goes through OK. Sofia status profile shows correct values for Ext-RTP-IP for both profiles - my static public IP, RTP-IP shows local IP. Any thoughts on that? Anybody can share working profile configuration may be? Please help, I really need to get this going. Thanks.

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Contact Header Modification
by Roger Salloum 09 Jul '10

09 Jul '10

Is it possible to modify the contact header such that it is the caller id? I noticed sip_contact_user but it does not seen to effect the outbound leg of the call. The documentation specifically mentions that it is only for the internal SIP contact so I don't believe it does what I was hoping it would. I know you are able to set it to a static value, but I was hoping to be able to set it on a call per call basis. Thanks

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Re: [Freeswitch-users] enforcing RTP payload numbers for codecs
by Anthony Minessale 08 Jul '10

08 Jul '10

No, you can't force codecs in the dynamic range and if properly implemented, it won't matter what number either side chooses. On Jul 8, 2010 8:34 AM, "Tzury Bar Yochay" <tzury.by(a)reguluslabs.com> wrote: Hi, I just download the latest, build and brought up all is fine. However, our clients are connected via IP over GPRS(UMTS) and thus we are using Speex codec. Is there a way I can enforce speex to have 97 or 98 instead of 99? Currently as it looks in the vars.xml speex is 99. see dump (taken from default vars.xml) thanks Tzury RTP Dynamic Payload Numbers currently used in FreeSWITCH and what for. 96 - AMR 97 - iLBC (30) 98 - iLBC (20) 99 - Speex 8kHz, 16kHz, 32kHz 100 - _______________________________________________ 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] enforcing RTP payload numbers for codecs
by Tzury Bar Yochay 08 Jul '10

08 Jul '10

If I could have enforce FS to transcode that session between those two clients it would have work properly (currently experiencing half duplex). On Thu, Jul 8, 2010 at 5:31 PM, Tzury Bar Yochay <tzury.by(a)reguluslabs.com>wrote: > BTW, > > Off topic though, I would be glad to hire you for our FS works. I was > looking for that for long enough and glad you brought that up ;-). > > > On Thu, Jul 8, 2010 at 5:28 PM, Tzury Bar Yochay <tzury.by(a)reguluslabs.com > > wrote: > >> Brian and Anthony, >> Dears, >> >> I am afraid it is a bug in teh FS >> See, if during SDP the client declare on payload 98 and fs declare on 99. >> Then client expect receiving 99 while sending 98. That is fine. >> However, when both clients are the same (as in our case) since FS is not >> trans-coding >> clients receives 98 in >> given that same sip client cannot work with previous versions of FS >> where there Speex was 98 (and 97 in older) >> >> >> On Thu, Jul 8, 2010 at 5:08 PM, Anthony Minessale < >> anthony.minessale(a)gmail.com> wrote: >> >>> No, you can't force codecs in the dynamic range and if properly >>> implemented, it won't matter what number either side chooses. >>> >>> On Jul 8, 2010 8:34 AM, "Tzury Bar Yochay" <tzury.by(a)reguluslabs.com> >>> wrote: >>> >>> Hi, >>> >>> I just download the latest, build and brought up all is fine. >>> However, our clients are connected via IP over GPRS(UMTS) and thus we >>> are using Speex codec. >>> >>> Is there a way I can enforce speex to have 97 or 98 instead of 99? >>> Currently as it looks in the vars.xml speex is 99. >>> see dump (taken from default vars.xml) >>> >>> thanks >>> Tzury >>> RTP Dynamic Payload Numbers currently used in FreeSWITCH and what >>> for. >>> >>> 96 - AMR >>> 97 - iLBC (30) >>> 98 - iLBC (20) >>> 99 - Speex 8kHz, 16kHz, 32kHz >>> 100 - >>> >>> _______________________________________________ >>> 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 >>> >>> >> >> >> -- >> Tzury Bar Yochay >> >> tzury.by(a)reguluslabs.com >> + 972 52 5133399 >> twitter.com/tzury >> > > > > -- > Tzury Bar Yochay > > tzury.by(a)reguluslabs.com > + 972 52 5133399 > twitter.com/tzury > -- Tzury Bar Yochay tzury.by(a)reguluslabs.com + 972 52 5133399 twitter.com/tzury

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enforcing RTP payload numbers for codecs
by Tzury Bar Yochay 08 Jul '10

08 Jul '10

Hi, I just download the latest, build and brought up all is fine. However, our clients are connected via IP over GPRS(UMTS) and thus we are using Speex codec. Is there a way I can enforce speex to have 97 or 98 instead of 99? Currently as it looks in the vars.xml speex is 99. see dump (taken from default vars.xml) thanks Tzury RTP Dynamic Payload Numbers currently used in FreeSWITCH and what for. 96 - AMR 97 - iLBC (30) 98 - iLBC (20) 99 - Speex 8kHz, 16kHz, 32kHz 100 -

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More questions migrating from Asterisk to FS
by mazilo 08 Jul '10

08 Jul '10

I recently managed to get FS to run on my Seagate DockStar device. So far, I have no problem to configure my FS with a single Gizmo5 account, except losing the 1st two seconds at the beginning of a call connection (I am sure there is a fix to this, but haven't found one, yet). My first step is to move all my Gizmo5 accounts to my new FS system so that I can decommission my Asterisk PBX system. The question I have is if FS has a support for template with inherritance (like asterisk) so that I don't have to copy the same parameters for each Gizmo5 account, except the username/password. I apologize if this question had been asked before, but a search through mailing list couldn't did not seem to yield such a topic. Thank you. ----- don't and stop are the ONLY two 4-letter words considered offensive to men, but not when used together. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/More-questions-migrating-from… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Re: [Freeswitch-users] hangup does not break the Lua loop
by Anthony Minessale 08 Jul '10

08 Jul '10

Also session:ready will return false if the call is being transferred. Bottom line is you should always be checking session:ready on any loops and periodicly throughout your script and exit asap if it returns false. On Jul 8, 2010 7:13 AM, "Javier Aristizábal" <javieraristizabal(a)gmail.com> wrote: Tony, you can try: while (var and session:ready) do ........ end On Thu, Jul 8, 2010 at 4:37 AM, Steven Ayre <steveayre(a)gmail.com> wrote: > > session:ready - checks... -- Javier Aristizábal _______________________________________________ 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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hangup does not break the Lua loop
by Tony Tin 08 Jul '10

08 Jul '10

Hi, I'm writing dialplan with Lua script. There is a while loop in the dialplan, I found that the loop is not broke out even the call has been hung up. I'm wondering whether this is a normal behavior. Thanks. Regards, Tony

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