January 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

VOPTech gateways
by Ержан Туле пов 15 Jan '13

15 Jan '13

Does anybody know anything about gateways from VOPTech, namely VG1X, VG4X or VG5X? How are they in general, and with FS (inter-operation) in particular? And one more question: there are a couple of companies on the market, which have similar product lines: New Rock Technologies with MX gateways and Real Tone Technologies with their WSS gateways? /Y

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MySQL AutoCommit
by Colin Mason 15 Jan '13

15 Jan '13

I see this in FreeSWITCH logs whenever MySQL is restarted. Is this something to be concerned about? Using: FreeSWITCH Version 1.3.13b+git~20130110T184435Z~f913670e27 (git f913670 2013-01-10 18:44:35Z) 2013-01-14 06:02:51.007527 [CRIT] switch_core_sqldb.c:1565 ERROR [Unable to Set AutoCommit Off--1] 2013-01-14 11:40:35.507511 [CRIT] switch_core_sqldb.c:1565 ERROR [Unable to Set AutoCommit Off--1] 2013-01-14 11:40:44.627524 [CRIT] switch_core_sqldb.c:1565 ERROR [Unable to Set AutoCommit Off--1] 2013-01-14 11:40:44.767535 [CRIT] switch_core_sqldb.c:1565 ERROR [Unable to Set AutoCommit Off--1] 2013-01-14 11:40:47.447532 [CRIT] switch_core_sqldb.c:1565 ERROR [Unable to Set AutoCommit Off--1] 2013-01-14 11:40:50.407523 [CRIT] switch_core_sqldb.c:1565 ERROR [Unable to Set AutoCommit Off--1] 2013-01-14 11:40:53.527518 [CRIT] switch_core_sqldb.c:1565 ERROR [Unable to Set AutoCommit Off--1] 2013-01-14 11:46:32.267507 [CRIT] switch_core_sqldb.c:1565 ERROR [Unable to Set AutoCommit Off--1] 2013-01-14 11:46:41.507535 [CRIT] switch_core_sqldb.c:1565 ERROR [Unable to Set AutoCommit Off--1] 2013-01-14 11:46:41.547531 [CRIT] switch_core_sqldb.c:1565 ERROR [Unable to Set AutoCommit Off--1] 2013-01-14 11:46:47.487519 [CRIT] switch_core_sqldb.c:1565 ERROR [Unable to Set AutoCommit Off--1] 2013-01-14 11:46:50.447521 [CRIT] switch_core_sqldb.c:1565 ERROR [Unable to Set AutoCommit Off--1] 2013-01-14 11:46:53.567526 [CRIT] switch_core_sqldb.c:1565 ERROR [Unable to Set AutoCommit Off--1]

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mod_amr transcoing for testing purposes
by Luis Azedo 15 Jan '13

15 Jan '13

Hi, is there any way to enable mod_amr with transcoding for testing purposes ? maybe with opencore-amr ? thanks

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Delay Round Two...FIGHT!
by Blake Priddy 14 Jan '13

14 Jan '13

Ok everyone, hope your new years went wonderful! So here is the situation again. THE DELAY IS BACK! I have wiped FS and installed the latest git head and the delay is still present. I want to eliminate all FS options that could fix it before I call AT&T. Any help is appreciated! Person from outside the school calls, secretary answers, secretary hears they person saying hello, secretary says hello over and over again for 10 seconds. Then the B-Leg audio finally is heard by the A-LEG. Note* The B-Leg (secretary) can hear the A-leg talking the whole time. Just the A-Leg cannot hear the (secretary) B-Leg until about 7-13 seconds has passed. I see nothing in the FS log after the audio from the B-Leg finally traverses to the A-Leg. Please help our school fellow FS users! LONG LIVE FS!!! :) -- *Blakelund Priddy* Network Systems Engineer Bryant Public School District Bryant, Arkansas 72022 http://www.bryantschools.org p 501-653-5038 f 501-847-5656

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Multiple Network Interfaces
by Steven Schoch 14 Jan '13

14 Jan '13

(Don't hesitate to tell me that I'm doing this the wrong way and what is the right way to do it.) I have 3 network cards on our Freeswitch box (which will be used as the office PBX, as soon as I can figure out how to get this all working.) One card is on our office LAN. It's address is 192.168.1.17. The default route is 192.168.1.9, which is our PepLink Balance 20 router, which knows how to route VoIP, and balances the outgoing network traffic via 2 WANs: an AT&T DSL and one from tlinkbroadband.com. The second card is connected to a PoE switch that will connect to all the desktop phones. It's address is 192.168.4.1. The FS machine runs DHCP, TFTP, FTP, and of course FS on this interface. The third card is not yet configured with an address. It will be an outside address, so that users can connect from home using SIP to check their voicemail, make office calls, etc. It may also be used as a failsafe interface on case the PepLink goes down. The phones connect successfully to FS over the 192.168.4.1 address. FS connects to sip.flowroute.com over the default route. Outgoing calls work. (When I get a DID assigned to our flowroute.com account, and get the auto-attendant working, then I expect it will work correctly in that direction. What doesn't work is SIP FAX software that I just installed on a PC on the LAN. I attempted to connect it using 192.168.1.17, but that didn't work, so I used 192.168.4.1, which is routed by the router to the FS box. The SIP registration was successful, and the test call was actually placed and rang on the receiving FAX machine, but then it failed. I have a suspicion that the audio got dropped somewhere. I also think that if I can get FS to accept a SIP registration on the other network card, that it might work. I can't find anything in the wiki about multiple network cards (except a page that was removed) -- Steve

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FreeSWITCH Weekly News and Notes
by Michael Collins 14 Jan '13

14 Jan '13

Greetings! It has been a light week when it comes to news items. However, the FreeSWITCH team has been working hard on bug fixes and various projects. More information on those will come out once the details are available. Last week we had a community discussion on the weekly conference call<http://wiki.freeswitch.org/wiki/FS_weekly_2013_01_09>, mostly discussing the ways that we can all help out with getting Jira's handled. Although we did not formally adopt the "Jira karma" idea we decided that we (as a project and community) would do our best to reward those who help out with bug marshaling, verifying, documenting, etc. Every little bit helps. To that end this week<http://wiki.freeswitch.org/wiki/FS_weekly_2013_01_16>we will be spending some time as a group going over a few selected bugs and as a group trying our best to figure what's going on. Ken and Brian will supply us with a few bugs to tackle. Oh, and if you use Cisco phones, please be very careful<http://arstechnica.com/security/2013/01/hack-turns-the-cisco-phone-on-your-…> ! -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org

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Re: [Freeswitch-users] How to pause and resume play an audio file in conference?
by 王迪 14 Jan '13

14 Jan '13

"uuid_fileman" is for channel, not for a conference. If I used "uuid_fileman" to pause a channel, and I can not talk to this channel. I wanna play file in a conference, and pause the playing and talk, then stop talking and resume the playing. 发件人： freeswitch-users-request 发送时间： 2013-01-10 22:12:37 收件人： freeswitch-users 抄送：主题： FreeSWITCH-users Digest, Vol 79, Issue 54 Send FreeSWITCH-users mailing list submissions to freeswitch-users(a)lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to freeswitch-users-request(a)lists.freeswitch.org You can reach the person managing the list at freeswitch-users-owner(a)lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..." Today's Topics: 1. Re: How to pause and resume play an audio file in conference? (Avi Marcus) 2. Re: Best practices question about SIP registration (Steven Ayre) 3. Re: mod_com_g729 transcoding (Steven Ayre) 4. Early media without bridge (Tamas.Cseke ) 5. Loopback Endpoint (Jon Sch?pzinsky) 6. Re: mod_directory menu-top? (Abaci) ------------------------------------------------------------ From: Avi Marcus <avi(a)avimarcus.net> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Subject: Re: [Freeswitch-users] How to pause and resume play an audio filein conference? Date: Thu10 Jan 2013 11:31:49 +0200 It looks like you can do it via api command: http://wiki.freeswitch.org/wiki/Mod_commands#uuid_fileman It seems the conference has stop-talking and start-talking events, so you can start the file and then have an ESL app that listens for the events and stops/starts the playback. Let us know how that goes. -Avi On Thu, Jan 10, 2013 at 3:39 AM, 王迪 <wangd(a)alongtechnology.com.cn> wrote: > ** > How to pause and resume play an audio file in conference? For example: > I create a conference room with some members, and play an audio file to > them. When someone start-talking, I pause the playing. When nobody talking, > I resume play the audio file. > > BTW, In freeswitch-1.2rc, I tryed play,pause,resume commands, but pause > and resume is for recording, not for my needs. > > > _________________________________________________________________________ > 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 > > ------------------------------------------------------------ ------------------------------------------------------------ From: Steven Ayre <steveayre(a)gmail.com> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Subject: Re: [Freeswitch-users] Best practices question about SIP registration Date: Thu10 Jan 2013 09:31:29 +0000 > There IS a wrinkle to this special case: If you have a URL of the form <sip:user@example.com:5055> the client should NOT look for a NAPTR or SRV, but instead go straight to looking for an A or AAAA for the host The reason being that a SRV record includes the port (note that means it's a very easy way to add/change additional ports as well as addresses without having to change customer configs). So obviously a host:port isn't compatible with a SRV, since it wouldn't make sense to override the port, so the assumption is that it's specifying an A/AAAA record instead and port on that IP instead. -Steve On 8 January 2013 23:35, Lawrence Conroy <lconroy(a)insensate.co.uk> wrote: > Hi Michael, folks, > if you're going to codify it, I did slightly simplify things: > (full disclosure -- I disagreed with it at the time which is probably why I forgot to mention it -- honest). > > There IS a wrinkle to this special case: If you have a URL of the form <sip:user@example.com:5055> the client should NOT look for a NAPTR or SRV, but instead go straight to looking for an A or AAAA for the host example.com (i.e., there's a machine called example.com, and it is handling SIP traffic for that domainpart). > Basically, if you see a colon in the domainpart, you're looking for a machine -- otherwise you're looking for a NAPTR (and/or a SRV at _sip._udp.<sipdomain>). > > I'd put that before the paragraph starting "However, relying on ..." > > Curiously enough, the old 2543-compliant servers did hunt the SRV rather than giving up and looking for an A, so this was a change. Such fun was had re-writing implementations (plural) and testing them yet again. Sigh. > > all the best, > Lawrence > > > On 8 Jan 2013, at 22:56, Michael Collins wrote: > >> Lawrence, >> >> Thanks for this explanation. It was very well written. I'm looking for a >> place to codify this on the wiki so that it gets preserved... :) >> >> -MC >> >> On Tue, Jan 8, 2013 at 1:56 PM, Lawrence Conroy <lconroy(a)insensate.co.uk>wrote: >> >>> Hi there, >>> at the risk of butting in on someone else's party ... >>> Nope; your interpretations is NOT best practice. >>> I have some sympathy, as the term domain is overloaded within fS. >>> >>> A sip address consists of a userpart and a domain part -- e.g., >>> <sip:user@sipdomain> >>> The sip domain is similar to an email domain -- e.g., <mailto: >>> user@maildomain> >>> With email, you need to do a lookup of the MX record in DNS to find the >>> FQDN of the machine that handles mail for the domain. >>> With SIP (see RFC 3263), you do a lookup on the SRV record (at >>> _sip._udp.<sipdomain>) to find the machine that handles SIP >>> registrations/incalls for the domain. That also gives you the port on >>> which that machine is listening. >>> (Yup, you can also have a NAPTR record in the domain to tell you where the >>> SRV record is, but many folks don't bother -- for Best Practice, you >>> should, but ...) >>> >>> There IS a "get out" clause in the SIP specs for RFC 2543 (AKA legacy) >>> support that means most SIP clients will look for the SRV and, if it can't >>> be found (or there's an IP address rather than a DNS -style domain, in >>> which case the SIP client won't bother hunting the SRV), the client will >>> guess that the domain has a machine (i.e. it will look for an A or AAAA >>> record), and also guess it's listening on 5060 (the default port). >>> Email is the same (mail to fred(a)example.com, and strictly the sender will >>> do a check for a MX and then look for an A record for example.com, and >>> try there). >>> >>> However, relying on that default "get out" clause is definitely NOT what >>> you should do for BCP. >>> Using the hostname as the sip domain is a kludge -- the FQDN with A record >>> usually works, but it's not what you want to do. >>> >>> SO ... get yourself a domain, put a D2U NAPTR at that domain, put a SRV at >>> _sip._udp.<domain>, and you're done. No need for an A record at that domain >>> at all. >>> >>> (RFC 3263 is not too hard to read, for a change -- it's certainly shorter >>> than RFC 3261, and it even has an ASCII art diagram :). >>> >>> all the best, >>> Lawrence >>> >>> On 8 Jan 2013, at 21:05, Steven Schoch wrote: >>> >>>> On Fri, Dec 28, 2012 at 8:47 PM, Tim St. Pierre < >>>> fs-list(a)communicatefreely.net> wrote: >>>> >>>>> Hi Steven, >>>>> >>>>> I would recommend using a proper domain name as much as possible. For >>>>> one, it looks >>>>> nicer! A SIP URI is supposed to be user@domain like an e-mail address >>>>> is, and I hope that >>>>> one day URI dialing will be common place, so we might as well do it >>> right >>>>> the first time. >>>>> >>>> >>>> What you're saying is that "domain" should really be a fully-qualified >>> host >>>> name that points via DNS to the actual host on which FreeSwitch is >>> running. >>>> That is, the domain should be "pbx.example.com" instead of just " >>>> example.com", as the last example would most likely point to a web >>> server, >>>> not the SIP server. Do I have that right? >>>> >>>> Next, in the configuration for Polycom phones (for example), there are 2 >>>> fields that both have the userid. In the example in >>>> http://wiki.freeswitch.org/wiki/Polycom_configuration it has: >>>> >>>> reg.1.auth.userId="1000" >>>> >>>> and >>>> >>>> reg.1.address="1000(a)fs.domain.local" >>>> >>>> How is the "address" value used? Is that sent in the SIP registration >>>> message? If that's the case, what does Freeswitch do with it? >>>> >>>> -- >>>> Steve >>>> _________________________________________________________________________ >>>> 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 >>> >> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.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 ------------------------------------------------------------ ------------------------------------------------------------ From: Steven Ayre <steveayre(a)gmail.com> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Subject: Re: [Freeswitch-users] mod_com_g729 transcoding Date: Thu10 Jan 2013 09:38:52 +0000 A more complete debug-level log could be useful to get more context. Is box 2 running any other calls at the same time that might be using the license? Is box 2 doing anything else with the call? Anything like eavesdrop, recording, start_dtmf / start_dtmf_generate etc which uses the media will use a license, and I *think* that license only gets released until the end of the call. -Steve On 10 January 2013 07:09, Colin Mason <cmason(a)frontiernetworks.ca> wrote: > I have phone A connected to freeswitch box 1 and phone B connected to > freeswitch box 2. > > > > Phone A wants to dial phone B using G729. Codec is always G729. > > The path RTP follows is: > > > > > > > > Phone A -------> FreeSWITCH 1 -------> FreeSWITCH 2 -------> Phone B > > (g729) (g729) > (g729) > > > > > > > > My question is, why is it that the freeswitch box receiving the call uses up > an encoder/decoder when the codec is G729 along the path? If I reverse the > call and call Phone A from Phone B, FreeSWITCH Box 1 uses up 1 license. > > if I dial the PSTN to a carrier who supports G729, I don’t use up a license. > Any thoughts? Maybe this is normal. > > > > FreeSWITCH Box 1: > > 2013-01-10 01:50:14.195277 [DEBUG] sofia_glue.c:3327 AUDIO RTP > [sofia/mpls/I888_1(a)172.17.17.17] 172.17.17.17 port 32014 -> 10.253.200.6 > port 16466 codec: 18 ms: 20 > > 2013-01-10 01:50:14.195277 [DEBUG] sofia_glue.c:3327 AUDIO RTP > [sofia/transport/2996] 172.17.17.17 port 27276 -> 172.16.16.16 port 17056 > codec: 18 ms: 20 > > freeswitch@internal> g729_info > > Permitted G729 channels: 40 > > Encoders in use: 0 > > Decoders in use: 0 > > > > FreeSWITCH Box 2: > > 2013-01-10 01:50:14.184757 [DEBUG] sofia_glue.c:3351 AUDIO RTP > [sofia/transport/3888(a)172.17.17.17] 172.16.16.16 port 17056 -> 172.17.17.17 > port 27276 codec: 18 ms: 20 > > 2013-01-10 01:50:15.664733 [DEBUG] sofia_glue.c:3351 AUDIO RTP > [sofia/mpls/sip:C996_1@10.253.200.10:5060] 172.16.16.16 port 18966 -> > 10.253.200.10 port 16486 codec: 18 ms: 20 > > freeswitch@internal> g729_info > > Permitted G729 channels: 40 > > Encoders in use: 1 > > Decoders in use: 1 > > > > > > > > Thanks in advance guys. > > Colin > > > _________________________________________________________________________ > 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 > ------------------------------------------------------------ ------------------------------------------------------------ From: "Tamas.Cseke " <cstomi.levlist(a)gmail.com> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Subject: [Freeswitch-users] Early media without bridge Date: Thu10 Jan 2013 12:24:00 +0100 Hello, We would like to hear early media without CHANNEL_BRIDGE event These are failed calls and callers would like to hear the message that the provider plays Because the caller thinks the call is answered if originate returns. as far as I understand: -early media makes the originate return -if we ignore early media the bridge won't return, but we don't hear it we would like both of them, is it possible somehow? I 'm not sure I fully understand all of the ignore_early_media options but I haven't find solution for this, Could you please advise me one, if there is any? I'm thinking about we maybe need a new ignore_early_media option like "consume" but sending the media to the caller instead of dropping it If there isn't already a solution I also would appreciate if you let me know your opinion about this idea Thanks advance, Tamas ------------------------------------------------------------ ------------------------------------------------------------ From: Jon_Sch鴓zinsky<jos(a)firstcom.dk> To: "freeswitch-users(a)lists.freeswitch.org"<freeswitch-users(a)lists.freeswitch.org> Subject: [Freeswitch-users] Loopback Endpoint Date: Thu10 Jan 2013 14:36:06 +0100 Hello List, I can see that the "this will destroy the world and this may kill your pets" warning has been removed from the documentation for the Loopback endpoint. Is this an indication that it has become safer to use? I have a specific problem where I essentially have to dial a dialplan for each user I am trying to reach, in parallel. Is it correctly understood that loopback would be the best/only way of implementing this, or is there another way? Kind Regards Jon Schøpzinsky ------------------------------------------------------------ ------------------------------------------------------------ From: Abaci <abaci64(a)gmail.com> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Subject: Re: [Freeswitch-users] mod_directory menu-top? Date: Thu10 Jan 2013 09:11:09 -0500 use 'execute_extension' to start the directory application so that you get back to the ivr when you exit the directory application. On 1/9/2013 6:41 PM, Phillip Warner wrote: > Hi, is there a parameter in mod_directory to have it transfer (back) to an ivr if the user decides not to search by directory instead of having to hang-up and call again? > > For example: ivr --> directory --> user changes mind about dialling by name and wants to return to ivr --> ivr > > Thanks. Phil. > _________________________________________________________________________ > 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 ------------------------------------------------------------ _______________________________________________ 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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Bridging an incoming call to outbound gateway
by Meditel 14 Jan '13

14 Jan '13

hi Michael, sip traces : http://pastebin.freeswitch.org/20209 gsm is ringing and no audio in both ways.. Best regards

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Re: [Freeswitch-users] mod_amr transcoing for testing purposes
by Luis Azedo 14 Jan '13

14 Jan '13

i know there is no licensed version of the module available yet. i have a dialer that supports amr and want to see if it can be used in freeswitch with transcoding. like in here http://sourceforge.net/projects/asterisk-amr/ but for freeswitch does anyone have mod_amr with transcoding running in freeswitch ? -- Luis Azedo

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Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 79, Issue 73
by royj 14 Jan '13

14 Jan '13

try <param name="inbound-zrtp-passthru" value="true"/> > > I need to set up a passthrough zrtp in freeswitch. > I followed this wiki: http://wiki.freeswitch.org/wiki/ZRTP and setup two > Jitsi softphones. > > MiTM: works well but with different SAS in the softphones. It is ok because > of a missing zrtp enrollment process. > Passthrough: no SAS at all and Jitsi says the media is not encrypted > I tried to achieve passthrough by uncommenting > sip_profiles/internal.xml: > <param name="inbound-proxy-media" value="true"/> > > What is the proper way to set up passthrough zrtp and having an end-to-end > zrtp session with the same SAS on both softphones? > > > Thank you, > Peter regards royj

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