June 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] FS & Wikipbx Help SOLVED
by FERNANDO VILLARROEL 09 Jun '09

09 Jun '09

Dear. Excuse some off topic of Wikipbx, but today i solved my problem; wikipbx and FS working now. Best regard. --- On Sun, 5/24/09, Jason White <jason(a)jasonjgw.net> wrote: > From: Jason White <jason(a)jasonjgw.net> > Subject: Re: [Freeswitch-users] FS & Wikipbx Help > To: freeswitch-users(a)lists.freeswitch.org > Date: Sunday, May 24, 2009, 11:17 PM > FERNANDO VILLARROEL <fvillarroel(a)yahoo.com> > wrote: > > > My install of wikipbx was succesfully, but i have > problem for registering a > > Softphone Xlite for testing; i look the next warning > in FS_CLI: > > > > freeswitch@internal> 2009-04-28 06:13:58 [WARNING] > sofia_reg.c:1701 > > sofia_reg_parse_auth() Can't find user > [223(a)192.168.1.150] You must define a > > domain called '192.168.1.150' in your directory and > add a user with the > > id="223" attribute and you must configure your device > to use the proper > > domain in it's authentication credentials. > > > > When i solve this problem i want make a gateway > between FS and Asterisk, for > > sending traffic from FS to Asterisk. > > > > If any could help me to resolve register ATA or > Softphones with wikipbx. > > If WikiPBX has its own mailing list, I think you would > receive better help > there. > > After installing FreeSWITCH, try registering to one of the > extensions (1000 to > 1019) provided in the default configuration. If that works, > which it should, > you should then try creating your own extension, or > modifying an existing one > as required. > > The error message you quote says very explicitly what is > wrong, but without > access to your FreeSWITCH configuration, it is impossible > to help any further, > except to suggest that you start with the default > configuration and work from > there. > > > _______________________________________________ > 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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video transcoding
by Francois Delawarde 09 Jun '09

09 Jun '09

Hello, I'm interested in being able to do video transcoding mainly for bridging 3G mobile and sip networks, and maybe later on some conferencing with FS. Are video codecs planned to be added to FS even in a far future? Are there copyright/patent problems with common video codecs (H.263 / H.264) or with libraries (ffmpeg) that would prevent any of that from happening? Meanwhile, would it be feasible to do some video transcoding using external software (vlc?) with socket connections from-to FS? Thanks, François.

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Re: [Freeswitch-users] Problem with attendant transfer
by Durk de Beer 09 Jun '09

09 Jun '09

Ok I've recieved an error message so if this message is being send a second time my deepest apologies for it. Hello Brian, I observed the problem by a Siemens Gigaset DE380 IPR and A Cisco 7960 and 7965. What happens is this, a call is coming in on Freeswitch and is being bridged to lets say the Siemens on extension 100. Now the person accepting the call on extension 100 wants to transfer the call to an other extension lets say 200. The normal scenario would be extension 100 puts the original call on hold, Freeswitch streams moh to the original call, extension 100 dials 200, the having a conversation, 100 transfers the original call too 200. Now here the problem begins. Normally extension 100 would send a refer sip message to freeswitch, who would then connect the original call to extension 200. The Siemens and Cisco phones do not send a refer sip message first. What the do is putting extension 200 on hold by means of an send only sip message. When Freeswitch is receiving this it streams moh to extension 200. After this the phones are sending the transfer by means of a sip refer message. When Freeswitch is receiving this it can't perform this transfer, that of original call to extension 200, because extension 100 has put extension 200 on receive only and extension 200 is receiving the moh. Resulting in an original call receiving moh and an extension 200 receiving moh. When this situation arises there's no way in connecting these to together. So what I need is a way to detect that there is an transfer by means of an sip refer message to a extension that has being put on hold. If so I need to get freeswitch to break this hold and transfer the original call to this extension. I hope that this will make the problem a little bit clearer. Kind regards Durk > This makes no sense.... Can you try to explain it more? I do attended > transfers with sip phones every day without a single problem. Maybe > i'm missing what you're talking about. > /b > On Jun 8, 2009, at 2:37 AM, Durk de Beer wrote: >> Hello all, >> I have observed an issue on using Freeswitch and some SIP-phones. Ok >> the problem is this. Some phones, when attempting an attendant >> transfer, put the recipient of the transfer on hold. This results in >> Freeswitch starting to stream MOH music to the phone put on hold, if >> implemented. When now the original phone is pasing the transfer, >> Freeswitch is not going to process this transfer because the >> recipient end of it is on hold. It is however terminating the >> connections it has with the phone initiating the transfer. This >> means that the recipient of the transfer is never coming of hold >> again until it terminates the call. >> Is there a way to detect this behaviour, so I can get the recipient >> of hold before Freeswitch is processing the transfer?. >> Kind regards >> Durk > Brian West > brian(a)freeswitch.org

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I need a favor...
by Gerry Hull 09 Jun '09

09 Jun '09

OK, thanks to help on the list have my very cool FreeSwitch app running... Gotta love FS once you get over the learning hump! So, I build FS and got everything running smoothly on my Wndows development box. Great. Then I went to deploy it on a production server. As I figured, no copy-and-run here. I tried building the setup project but it's just not happening for me! Can someone out there build me the Windows MSI for build 13496 or later and provide a link to it? I'm in a bind here to get this up and running. If I can pry a few bux out of the boss, I hope to be a ClueCon and describe to application we have built with FreeSwitch. Regards, Gerry

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Re: [Freeswitch-users] Calls drop immediately when terminator forc es G.729 Codec.
by Jim Burke 09 Jun '09

09 Jun '09

Yes, the terminator is Linksys so will change it and test. Noticed there is a list of mime types associated with FS and G729a was not listed, does this have anything to do with the root cause? - original message - Subject: Re: [Freeswitch-users] Calls drop immediately when terminator forc es G.729 Codec. From: Mathieu Rene <mrene_lists(a)avgs.ca> Date: 05/06/2009 23:56 Linksys still uses G729a in their sdp, but you can change it in the admin panel (if thats what you have) Math On 5-Jun-09, at 7:53 PM, Anthony Minessale wrote: > it is using G729a which is not the correct RFC value that goes with > payload 18 so it's moving it to 96 as if it's some other > codec. > > the only thing you can do is get them to stop using invalid data in > their sdp > or hack it to replace "G729a" with "G729 " before it's too late. > > > On Fri, Jun 5, 2009 at 6:25 PM, Jim Burke <jim(a)evolutiontel.net> > wrote: > Yup sure did, same result :( > > - original message - > Subject: Re: [Freeswitch-users] Calls drop immediately when > terminator forc es G.729 Codec. > From: Brian West <brian(a)freeswitch.org> > Date: 05/06/2009 14:10 > > DId you update to trunk? > > /b > > On Jun 5, 2009, at 3:06 AM, Jim Burke wrote: > > > should open a jira on this? Did you guys need any more info? > > Brian West > brian(a)freeswitch.org > > -- Meet us at ClueCon! 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 > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > > AIM: anthm > MSN:anthony_minessale@hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888@conference.freeswitch.org > iax:guest@conference.freeswitch.org/888 > googletalk:conf+888@conference.freeswitch.org > pstn:213-799-1400 > _______________________________________________ > 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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Caller id when doing transfers
by John Wehle 09 Jun '09

09 Jun '09

Consider the following sequence: 1) Outside caller (OC) calls Ext 1001 Caller id shows OC 2) Ext 1001 transfers the call to Ext 1002 In some cases we want the caller id to shows OC, in other cases we want the caller id to show Ext 1001. It appears from some limited testing that the original caller id is always shown when the call is transfered. Is there some way to have the person making the transfer show up as the caller id? Our application is I want to setup an extension (*5) which automatically places calls into a fifo corresponding to the Extension number of the person transferring the call. This will provide park capability similar to that of our old System 25 PBX. I'll then setup an extension (*8<fifo>) which picks up the call at the front of the fifo "fifo". -- John ------------------------------------------------------------------------- | Feith Systems | Voice: 1-215-646-8000 | Email: john(a)feith.com | | John Wehle | Fax: 1-215-540-5495 | | -------------------------------------------------------------------------

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Problems subscribing to outbound socket events
by Nik Middleton 08 Jun '09

08 Jun '09

I've put some c++ test code together to let the outbound socket control the call, all works as expected, apart from the event subscription Sending myevents\n\n gives the channel events However sending event text all\n\n doesn't give me any events apart from the channel events. Anyone care to suggest what I might be doing wrong? Regards,

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Re: [Freeswitch-users] Routing 911 calls
by Chris 08 Jun '09

08 Jun '09

Looking for a reasonable DID/e911 provider. Any suggestions. p.s. Sorry to hijack the thread. --- On Mon, 6/8/09, Michael Collins <msc(a)freeswitch.org> wrote: From: Michael Collins <msc(a)freeswitch.org> Subject: Re: [Freeswitch-users] Routing 911 calls To: freeswitch-users(a)lists.freeswitch.org Date: Monday, June 8, 2009, 1:03 PM On Mon, Jun 8, 2009 at 11:31 AM, Tim B <timb0311(a)hotmail.com> wrote: If I have 1 inbound DID and "client software" that calls this DID from multiple locations (states, city, etc), how would I go about placing and routing a 911 / emergency calls? Keep in mind each client does have their own existing phone line (DID) with 911/e911 attached to that DID. Could I route by setting outbound caller id and number to the incoming, and then place the outbound 911 call? You can try it, and then make a test 911 call and tell the operator that you are testing a new phone system install. The 911 op can then verify the information he/she sees on the screen with where you are. If you're in New York when you call but the 911 op sees "Jerkwater, Alabama" then you know it doesn't work... -MC -----Inline Attachment Follows----- _______________________________________________ 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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Sounds for 1.0.4 release and zRTP sound files.
by Brian West 08 Jun '09

08 Jun '09

I have a deal from GM Voices for this order at a major discount but its still $650 USD, If you wish to donate to this order please let me know... brian(a)freeswitch.org is my paypal. Thanks, Brian West brian(a)freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com

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Routing 911 calls
by Tim B 08 Jun '09

08 Jun '09

If I have 1 inbound DID and "client software" that calls this DID from multiple locations (states, city, etc), how would I go about placing and routing a 911 / emergency calls? Keep in mind each client does have their own existing phone line (DID) with 911/e911 attached to that DID. Could I route by setting outbound caller id and number to the incoming, and then place the outbound 911 call? Tim _________________________________________________________________ Insert movie times and more without leaving Hotmail®. http://windowslive.com/Tutorial/Hotmail/QuickAdd?ocid=TXT_TAGLM_WL_HM_Tutor…

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