November 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] media proxy
by Max Kovalenko 08 Nov '12

08 Nov '12

Unfortunately, it isn't well documented, or else I would not ask about :)))))) Those parameters perhaps are exclusive of each other, by mechanics they affects to is almost the same. Or at least, I would call them two sides of ame thing. Theoretically, if you consider the situation of bypass_media=true, that means media flow bypass around Freeswitch, it also automatically annihilates settings of proxy_media variable. It doesn't matter, even you would set proxy_media=true for same channel, it should not provide you "light proxy" of media flow, because your settign of bypass_proxy comprises Freeswitch don't proxy media. Thus I guess in order to set "light" or "hard" proxy the variable bypass_media must be set to FALSE first, for same channel of course. My task requires either bypass media or light/hard proxy to be set for a channel depending of some conditions of a session. In the other words, I need to set up one of three modes per each channel. Perhaps I undestand something not so right - I will appreciate if you will correct me with the following: 1. Default (full media proxy) : bypass_media=false ; proxy_media=false 2. Transparent proxy (mine "light proxy"): bypass_media=false; proxy_media=true 3. Bypass media: bupass_media=true (it's doesn't matter what currect value of proxy_media is) Thanks Max Kovalenko Team Leader VoIP & UC Team Managed Services Dept. CyberVision Inc. ------------------------------------------------- tel. +1 (201) 585-9809 ext. 215 Email: mkovalenko(a)cybervisiontech.com Skype: mkovalenko_cv, panzer_meister WWW: www.cybervisiontech.com ----- Исходное сообщение ----- От: "Ken Rice" <krice(a)freeswitch.org> Кому: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org> Отправленные: Пятница, 2 Ноябрь 2012 г 17:12:11 GMT +02:00 Афины, Бухарест, Стамбул Тема: Re: [Freeswitch-users] media proxy Default, proxy and bybass media are exclusive of each other, you can not combine them on a single call... Proxy media is just that proxy the media Bypass media is just that, bypass freeswitch and send the media direct between the end point... Setting this options is fairly well documented on the wiki... On 11/2/12 9:13 AM, "Max Kovalenko" <mkovalenko(a)cybervisiontech.com> wrote: > Hello, > > There are two modes of media stream to be proxied or bypassed. > > 1. Default - "hard proxy" supplying trans coding, DTMF manipulations, etc. > 2. Proxy - "light proxy" supplying only symmetrical RTP for mostly NAT > traversal purposes. No media stream parsing is supported > 3. Media bypass - RTP streams are bypassing FreeSWITCH at all. > > There are also two channel parameters affecting above modes: bypass_media and > proxy_media. > > - Are these parameters independent? Meaning the combination: bypass_media=true > AND proxy_media=true is possible. What effect will be? > > - Does it means that if I want to turn on "light proxy" I would always need to > set bypass_media=false AND proxy_media=true? > > - How to set "hard proxy" (full media proxy) per channel before to bridge > legs? > > Waiting for your replay ASAP. Thank you in advance. > > Best Regards. > > Max Kovalenko > Team Leader > VoIP & UC Team > Managed Services Dept. > CyberVision Inc. > ------------------------------------------------- > tel. +1 (201) 585-9809 ext. 215 > Email: mkovalenko(a)cybervisiontech.com > Skype: mkovalenko_cv, panzer_meister > WWW: www.cybervisiontech.com > > > > > _________________________________________________________________________ > 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 -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch _________________________________________________________________________ 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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Possible mod_smpp
by Michael Collins 08 Nov '12

08 Nov '12

Hello all, If you are interested in giving monetary support to have a professional software firm create mod_smpp then please contact me off list and I will give you more details. Thanks, Michael

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Newbie routing question
by Jeff Geddes 08 Nov '12

08 Nov '12

I've been able to receive a call from a landline thru a gateway and route it to my softphone. can also make an outbound call from my softphone to a landline. I need to 'tie' the two together, receive an inbound call and using a dialplan route it as an outbound call to the number in the dialplan. what would be the proper approach to do this? thanks jeff

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Re: [Freeswitch-users] GSMOPEN dongle port changes randomly
by farmradio 08 Nov '12

08 Nov '12

Hi Giovanni and Ivan - we would really like to know what this workaround is as we are encountering the same problem using the USB dongle (Huawei E173) with Freedom Fone. Thanks. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/GSMOPEN-dongle-port-changes-r… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Connecting RTP between leg A and B directly
by Víctor Vladimir Chávez Gallardo 08 Nov '12

08 Nov '12

Hi, i have an spidermonkey script, the script place two outbound calls, but i need to set the RTP directly between the legs (a-b) but i dont know hoy to set the parameter bypass_media i tried: lega_session.setVariable('bypass_media', 'true'); bridge(lega_session,legb_session); also lega_session.setVariable('bypass_media', 'true'); legb_session.setVariable('bypass_media', 'true'); bridge(lega_session,legb_session); and also setting another variable bypass_media_after_bridge lega_session.setVariable('bypass_media_after_bridge', 'true'); legb_session.setVariable('bypass_media_after_bridge', 'true'); lega_session.setVariable('bypass_media', 'true'); legb_session.setVariable('bypass_media', 'true'); bridge(lega_session,legb_session); but i dont have the RTP in the correct way, instead my freeswitch it's behind, the rtp flow: LEG-A -----> FREESWITCH -------> LEG-B any idea?

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Connecting delays too much sometims
by Luis Daniel Lucio Quiroz 08 Nov '12

08 Nov '12

I have FS 1.2.3 with updated 1 week ago vbilling installed, some destinations delays to much to start connecting, others are too quick. Wondering if you tell me some topics i shall review to find the bootleneck. Using mod_memcache will help? Regards, LD

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Problem with digest authentication
by Gunnlaugur Máni Hrólfsson 07 Nov '12

07 Nov '12

Hello all, I'm having a problem with a particular device, that I'm hoping some one on this list can help me with. The device is a Telsey IAD. At first the device registers just fine. But when the registration expires and the device re-registers, freeswitch responds with a 403 Forbidden. I realized that this must be because either the device og Freeswitch wasn't calculating the digest response correctly. So I tried to calculate it myself. This is a sample sent from the device, that Freeswitch responded with a 403 to: REGISTER sip:test.sip.is SIP/2.0 Call-ID: 5f880-1485c-aab33cd(a)10.171.32.9 From: <sip:4151664@test.sip.is;user=phone>;tag=2f5c0-1485c To: <sip:4151664@test.sip.is;user=phone> CSeq: 104 REGISTER Via: SIP/2.0/UDP 10.171.32.9;branch=z9hG4bK-2f5c0-101315 Contact: <sip:4151664@10.171.32.9;user=phone> Max-Forwards: 70 User-Agent: Telsey ver4_49_25 Expires: 30 Supported: timer Authorization: Digest username="4151664",realm="test.sip.is",uri="sip: test.sip.is ",response="c3b5cd8147fb883fbbe9acb183838717",algorithm=MD5,nonce="9202dbb4-292d-11e2-bac2-399d85a4015c",qop=auth,cnonce="0002f818",nc=00000001 Content-Length: 0 Assuming a password of "1234", the digest response should be 041323244fc4bdac8d49872bf29d9a3a. But the device seems to increment the nc value to 00000002 and calculates the response as c3b5cd8147fb883fbbe9acb183838717. The problem is that the device sends nc=00000001 so freeswitch uses that value to calculate the digest, while the device uses 00000002. I think it is obviously the device that's broken but not Freeswitch. But, in this case I need to get this device to work as is. Is there any way I can have freeswitch authenticate the device without using digest method that requires a nonce count, like rfc2069? Best regards, Mani Hrolfsson

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Bad connection diagnostics?
by Sean Devoy 07 Nov '12

07 Nov '12

Hi, I have a client who has been working fine for months, but as of late they are reporting connections where the audio only works in one direction or neither and "terrible echo". I know they had some issues with their local Cable Connection, but they are supposedly resolved. Results from "ping -n 50 -l 256 " and "tracert" are virtually identical to mine from here and I have no issues. Are there some diagnostics I could run to try and pin this down. The phones are all cisco 504Gs, the server is a VPS from synapseglobal. Thanks for any ideas you may have. Sean

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I could really use help with my ITSP outage, translate asterisk to freeswitch
by tehfink 07 Nov '12

07 Nov '12

Hi Mario, what Freeswitch settings did you use to restore service? I agree with you, the FS wiki shows a parm to turn DNSRV off so I am pretty sure it's on by default. Anyway, we're back up. Thanks for the note!

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Calling number lookup
by Archana Venugopan 07 Nov '12

07 Nov '12

Hi, Which script will lookup the destination number? Say for example if we are dialling 750**** number, which script will lookup corresponding database and dial this number Regards, Archana

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