- FreeSWITCH-users - FreeSWITCH Mailing Lists

How to set timeout in bgapi originate
by Leonardo Alves 15 Apr '08

15 Apr '08

How can I set the timeout in this command bgapi originate {ignore_early_media}sofia/gateway/asterisk/number &playback(message) My asterisk is calling but after 2 rings the freeswitch ends the call. I have tried in a lot of ways to set the timeout in the final of the command but I could not get it to work. If someone can help Thks Leonardo

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Callerid and swedish characters
by Jonas Gauffin 15 Apr '08

15 Apr '08

Hello I get a debug assert failure when using swedish characters in caller id name. File: isctype.c Line: 56 Expression: (unsigned)(c + 1) <= 256 The character i tried with is "ö" (o with two dots over if mailinglist cant handle it, html encoding: ö). Provided directory from my (local testing) server: 17:20:22 | RuntimeMethodHandle.InvokeMethodFast => RuntimeMethodHandle._InvokeMethodFast => SwitchController.Freeswitch | Reply: <document type="freeswitch/xml"> <section name="directory"><domain name="192.168.1.110"> <user id="u1000024" mailbox="1203"> <params /> <variables> <variable name="user_context" value="default" /> <variable name="effective_caller_id_name" value="Pelle Lösvan" /> <variable name="effective_caller_id_number" value="1203" /> <variable name="gate_userid" value="24" /> <variable name="gate_siteid" value="1" /> </variables> </user> </domain></section> </document> 17:20:22 | LazyAsyncResult.ProtectedInvokeCallback => LazyAsyncResult.Complete => HttpServerBase.OnRequest | Processing of http://localhost:81/switch/freeswitch/ is done. //Jonas

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Man what a news day!
by Brian West 14 Apr '08

14 Apr '08

http://digg.com/software/HD_Telephony_Hip_or_Hype_2 Digg it! Brian West sip:brian@freeswitch.org

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FreeSWITCH on Digg Again please check it out and Digg it!
by Brian West 14 Apr '08

14 Apr '08

Here you go guys! http://digg.com/software/Freeswitch_Poised_to_Shake_Up_the_Open_Source_VoIP… Thanks, Brian West sip:brian@freeswitch.org

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Freeswitch win32 error (Procedure entry point inet_pton could not be located in the dynamic link library WS2_32.dll)
by B Karthik 14 Apr '08

14 Apr '08

Hi, Over the days, my interest in FreeSwitch is overwhelmingly increasing. I love this software. I am sure that one day i will be competent enough to contribute to this project. I get this error when i compile and run freeswitch using visual studio 2008 express edition. (Procedure entry point inet_pton could not be located in the dynamiclink library WS2_32.dll) This is happening in svn revision 8099. Any help will be deeply appreciated. Thanks in advance. Warm Regards, B Karthik

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Re: [Freeswitch-users] [Freeswitch-dev] bad REGISTER processing
by Anthony Minessale 14 Apr '08

14 Apr '08

The main reason we write the port in that case is because the idea of that nat hack is to make the contact be the exact ip/port the register came from. Because of NAT it's some random port so putting the port in the contact is the only way we can get the packet back to the phone from our scope of the SIP stack. Any more advanced techniques need to be applied by a router such as openser or from the sofia stack itself. You can visit them in #sofia-sip on the same irc server as ours. First of all I would like to complement us on our ability to keep up with you. Just as a hint, you are starting to abuse our help by asking for an average of 3 incidents a day this week. Do people you pay money even help you that much? Let me give you a few pointers on how to get along with us. You may or may not have done all of these things but I am listing them for your information. I am not telling you to stop bringing up issues but be careful about dominating our time. You may want to also use http://jira.freeswitch.org as a more formal way to track your problems. 1) Please do not take the extra time to provide any justification to why you think we *should* do something just to help your case. Feel free to ask but do not use what someone else does as an excuse. If everyone in Europe was jumping off a cliff, should we too? This includes used car salesman techniques like using statements like "I have this simple app and it doesn't work how I want. If this were a real soft-switch it would do this....." 2) Do not use RFC as a bible. When someone wants something they tend to stand up on a soapbox and wave it in the air. Meanwhile when something else inconvenient happens like NAT where breaking the RFC usually fixes it, then it's ok. We all know we should try to work as close to the RFC as possible. We also all know it's impossible to actually work right at 100% RFC compliance unless you live on a commune full of SIP purists. So avoid quoting the RFC to prove your point. Save it for rare occasions. The Sofia SIP stack does a lot of work to comply and they do a good job you should thank them. 3) Keep it in mind that FreeSWITCH is developed primarily by me and I only have so many hours in a day. I am more than happy to answer questions and help people but be sure to give others a turn too. Try asking more questions on the users list or the irc channel to give others a chance to help too. 4) When you feel yourself crossing the line from testing the waters to going in production, consider getting a support contract or the visiting the donation and/or wishlists so you can give back to the project and put smiles on underprivileged developer's faces for pennies a day. On Fri, Apr 11, 2008 at 2:22 AM, kokoska rokoska <kokoska.rokoska(a)post.cz> wrote: > > Just for completion of my previous e-mail: > > May be I miss something significant and all my thesis are completly > wrong. If yes, I apologize and ask you for correction of my thoughts :-) > Thank you. > > Best regards, > > kokoska.rokoska > > > kokoska rokoska napsal(a): > > Thank you very much, Michael, for your answer! > > > > My comments are included in the e-mail body: > > > > Michael Jerris napsal(a): > >> The NDLB-connectile-dysfunction is a paramater to explicitly break the > >> rfc > > > > My be, but only internaly - i.e. all your public communication could be > > RFC compliant if you want. > > > >> and re-write our behavior as if we got a contact of the same > >> address that we received the packet from. > > > > Yes, in the same manner as all VoIP TSP (at least all bigger VoIP TSP in > > Europe) do :-) > > > >> If it comes from 5060, it > >> will have 5060 in the contact. > > > > And this brakes the RFC and UAC should refuse your 200 OK. There are a > > lot of broken clients which accepts this but not all of them :-) > > > >> If you would like to handle nat and > >> still be rfc compliant, then you need to use a client that follows the > >> rfc's > > > > I'm sorry but I don't know what is RFC noncomplient if UAC behind NAT > > sends private IP in contact. Could you point me to relevant part of some > > RFC, please? It will be very helpful for me... > > > > > >> so you don't have to use non compliant hacks. > >> > > > > Like I wrote before, I can't remember any real (and not marginal) VoIP > > telco provider in Europe which doesn't internaly rewrite contact URI in > > case of UAC behind NAT. Even thou, a lot of them rewrite EVERY contact > > URI and don't try to detect NAT, because a lot of SOHO routers have > > SIP-ALG support (god dammed) which makes thing even complicated. > > > > But there is - IMO - no reason why to rewrite contact URI in 200 OK > > response to REGISTER request. > > > > ---------------- > > > > All at all, from my point of view there is a big drawback if I couldn't > > use FreeSWITCH for UACs behind NAT. > > It forces me to use OpenSER as registrar (what I want to avid to because > > of performance and harder setup to keep good interoperability) which has > > no problem with clients behind NAT. > > > > I believe that improvement in NAT handling could help FreeSWITCH users a > > lot. Thus I will be glad to help you as much as I can do. > > Please let me know if I can do something handy (other than rewrite FS by > > myself :-). > > > > Thanks once more, Michael, for your answer! > > > > Best regards, > > > > kokoska.rokoska > > > > > > _______________________________________________ > > Freeswitch-dev mailing list > > Freeswitch-dev(a)lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > > http://www.freeswitch.org > > > > > _______________________________________________ > Freeswitch-dev mailing list > Freeswitch-dev(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale(a)hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale(a)gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888(a)conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888(a)conference.freeswitch.org> pstn:213-799-1400

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Problem playing media
by Nicolas Brenner 13 Apr '08

13 Apr '08

On Wed, Apr 9, 2008 at 7:36 AM, Brian West <brian(a)freeswitch.org> wrote: > > On Apr 9, 2008, at 1:17 AM, Nicolas Brenner wrote: > > > Hello everyone, > > > > I'm having some trouble with FS :( apparently with mod_shout. I want > > to play an mp3 file after answering a call so I compiled mod_shout > > following the wiki, then configured an extension to answer a call and > > play an mp3 file I uploaded to the server. The thing is, FS supposedly > > plays the file, but I can't hear it on the softphone with which I'm > > calling, and also, after playing the file, FS seems to freeze and then > > I get a lot of: > > > > sofia_event_callback() event [nua_r_bye] status [408][Request Timeout] > > session: n/a > > sofia_event_callback() event [nua_i_state] status [408][to BYE] > > session: n/a > > > > lines on the console/log. > > Well I can't see enough of the log with the two lines you posted to > many any kinda of educated guess. > Ok, attached is a console log with sip traces as well. What I did was: - start freeswitch - register softphone (xlite) with extension 1000 - call 9998: FS answered the call and played the tetris sound, although I was only able to hear it for 3 secs - hanged up: FS received the hang_up and terminated the call - call 9998 again: FS answered the call and supposedly played the tetris sound, but I didn't hear anything on my side - hanged up: FS did not recieve the hang_up signal, and kept the call 'open' - tried calling 9998 again: FS didn't show any sip packets being received or anything, the call could not be made - shutdown FS > > > Also, after I try to call once, the softphone does not work anymore, > > and I have to make it register again with FS. After all this, when I > > shutdown FS, it takes some time, and while it's trying to shutdown it > > prints a lot of the same lines as above. > > I can only guess you're using x-lite or eyebeam. > Yes, I'm using x-lite > > > I thought it was mod_shout, so I commented it from > > autoload_configs/modules.conf.xml so it does not load, and replaced > > the mp3 file with a wav file (FS has permission to read both of them), > > but I'm getting the same behaviour. Now I'm even getting that > > behaviour when dialing 9999 or 9998 (default dialplan pre-configured > > extensions). > > Is this a real server or virtual server? ie vmware or xen? > This is a virtual server I think, made with virtuozzo. Is a dedicated virtual server hosted by mediatemple (www.mediatemple.net) > > > Now I can't get FS to work right, everytime I try to make a call, > > either initiating it from the console, the softphone or js, I get the > > same weird behaviour. Any clues? > > I'm sorry if this is trivial or a known issue, but I haven't been able > > to figure it out, thanks a lot for your time and help. > > Your best bet at this point is to join the IRC channel and ask for > help in realtime... Have you tried "make current" to ensure you don't > have any code skew? > I tried make current several times. For the tests above (for which the attachment log file is), I removed /usr/local/freeswwitch, then checked out FS source from svn and compiled with default options (did not add mod_shout), and used default config files with no modifications (also installed audio and music files). Additionally, the server has a public IP address, and I'm connected directly to Internet (not nat involved whatsoever). > > -- > > Nicolás Brenner > > > > _______________________________________________ > > 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 > > Brian West > sip:brian@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 > The files attached are: - consolelog.txt: What I saw on the FS console while doing the test - siptrace.log: Output of sip trace to a file using TPORT_DUMP env var - freeswitch.log: log/freeswitch.log on FS folder for this test My email was rejected by the list moderator, the files are here (user/pass: freeswitch/mailing): - http://www.medularis.com/fs/freeswitch.log - http://www.medularis.com/fs/consolelog.txt - http://www.medularis.com/fs/siptrace.log Thanks for your help! -- Nicolás Brenner

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TBCT Followup
by Michael Collins 12 Apr '08

12 Apr '08

FYI, I spoke with my primary carrier - TWTC - and they report that there are no additional costs for setting up TBCT functionality. However, they require a provisioning request (aka paperwork) so I can't set it up right away. I'll probably get it set up in the next few months so that I can play with it using my NEC 2400 or possibly my IAT dialer... When I go to sign the paperwork that's when I'll get all the nitty-gritty details like how many concurrent calls, etc. I'll report back to the lists when I have that info. -MC

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woomera
by Josip Djuricic 12 Apr '08

12 Apr '08

Hi there, I know mod_woomera is deprecated, but is there any way to make it work with current stable release of fs? I would need it for simplest task, nothing too fancy... just to be sure incoming and outgoing calls work stable... I'm currently using yate as a sip-to-h323 converter, because my as5300 has some problems talking with freeswitch (when we are making incoming calls), and now I have incoming fax problems as mentioned yesterday. It's definetely not FS issue, now I just ain't sure anymore if it's a yate or as5300 issue, will try direct h323 to patton interconnect to see it more closely. Best regards, Josip

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Lucent Max TNT
by Brandon Shiers 12 Apr '08

12 Apr '08

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