Re: [Freeswitch-users] What's problem in SVN ?
Dome Charoenyost d...@tel.co.th wrote: What's problem in SVN ? Not thing update after 23/12/2009 (16055) Surely the FreeSWITCH developers are entitled to spend time with their families/friends after a highly productive year of work. Note that there are holidays in many countries at this time of year. I would like to wish everyone involved in the FreeSWITCH project a pleasant and refreshing holiday, and much success in 2010. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] MacOSX
Ken Gillett k...@ukgb.net wrote: I'd liked to create an install package, but that's a whole new can of worms. Are the FreeSwitch files all installed in a single directory that could be copied to a different machine? Yes, but that isn't a substitute for package management. For example, you typically want to preserve configuration files across package updates while having the opportunity to merge changes from newer configuration files. Package management solves this problem; it also solves dependency issues. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Make error...
Klaus Hochlehnert maili...@kh-dev.de wrote: src/switch_apr.c:899: warning: control reaches end of non-void function Are you on rev. 16032? As of 16032, this function shouldn't generate any such warning unless there's a compiler bug. ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] RTP problems in recent revisions?
Revision 15904 is fine, but after upgrading to revision 16003 I get the following. 1. No problems with a FreeSWITCH to FreeSWITCH call (Celt codec). 2. A PCMU call to a SIP provider is fine for the first 20 to 30 seconds, then the audio breaks up completely. I have ZRTP compiled in, if that makes any difference. Obviously there's a regression somewhere. Let me know if I can provide further help. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] [OT] Re: Scanning my firewall for open UDP ports?
Gabriel Gunderson g...@gundy.org wrote: Funny that you assume his desktop is running Windows (maybe it is). I would have guessed that the average person on this list doesn't run Windows on the desktop. But, what do I know? Some of us on the list have never run Windows on anything. It's Debian on my desktop, by the way, with FreeSWITCH acting as a soft-phone via a USB head set, and also handling my Snom 320 SIP phone. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] embedded freeswitch compatable hardware
Brian May br...@microcomaustralia.com.au wrote: I do have a spare TDM400p card, although as it is full height, suspect this isn't going to help. Have a look at http://www.yawarra.com.au/ Some of their hardware (notably the Soekris Engineering boards: http://www.soekris.com/) has a PCI slot. Disclaimer: in principle this should work well with FreeSWITCH, but I haven't tested it as I don't own the hardware yet. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Skype SIP Beta
Brian West br...@freeswitch.org wrote: They have yet to type make on a 64bit box and build us a binary that is 64bit. Chances are they mucked it up like the BroadVoice codecs were and it just won't work on 64bit just yet... if they would just give us the src we could be done in under two days with it I suspect. Given that they released the codec specification, perhaps someone is writing an independent C implementation? (Not that I'm much interested, but...) ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] IAX? Issues connecting road warriors with SIP?
Tim Uckun timuc...@gmail.com wrote: Yes I get issues quite a bit with the server being behind a firewall. IAX is much nicer in this circumstance. I just set up an IPv6 over IPv4 tunnel and nat goes away. I have native IPv6 over ADSL now, as part of a trial that my ISP is conducting. As a result, one end of the conection doesn't go through a tunnel provider anymore. Given the problems I've had (and still have) with nat, I want to be rid of it as much as possible. Nevertheless, I agree that in a nat scenario, IAX can be easier to configure correctly. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] ATA that supports TLS/SRTP w FS
Mark Campbell-Smith mcampbellsm...@gmail.com wrote: Does the SPA3102 support TLS or only SRTP? I don't know, but supporting only SRTP would be ridiculous, since the keys would then be transmitted in the clear and therefore amenable to interception. SRTP requires the SIP channel to be encrypted by TLS in order to be secure. ZRTP, on the other hand, doesn't have this limitation: it works entirely in RTP. I would be rather surprised were a hardware manufacturer to implement SRTP without TLS for the SIP traffic. On the other hand, we've seen often in this forum that some manufacturers are really clueless... ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] mod_bv16/32 removed. Added mod_bv
Brian West br...@freeswitch.org wrote: It now works.. update and have fun! Unfortunately it fails to compile under Debian: mod_bv.c:33:24: error: broadvoice.h: No such file or directory The header file exists, so I assume the include path specified to gcc just isn't right. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] RTP issues (possibly nat-related)
I can still reproduce this as of rev. 15584. Symptom: 1. I called a test number via my ISP (IPv4, subject to nat). This worked. 2. I placed a second call to the same number 30 seconds later - connected, but no audio received. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] XML config file parsing
Samuel Mukoti samuelmuk...@gmail.com wrote: I'm a new freeswitch user and am wondering what people do when setting options in the freeswitch config files. Do people use special tools, XML editors etc or is it just vi/emacs/Kate? Emacs has an XML editing mode; Vim may have extensions for handling XML as well. However, I have not found it necessary to invoke the XML features of an editor; just treating the configuration files as plain text is sufficient. ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] RTP issues (possibly nat-related)
I have upgraded FreeSWITCH several times recently for testing purposes. Also, my router's configuration has changed slightly as I have moved from tunneled IPv6 to a new native IPv6-over-ADSL trial. However, the problem now is related to my ISP's IPv4-only SIP service, and the symptoms are as follows. 1. If I call a test number, sometimes it all works perfectly. 2. On other occasions (with no discernible pattern) the call connects but no audio is received from the remote end. When this occurs, tshark shows that rtp packets are being sent out to the correct IPv4 address of the server. I am using Stun to handle nat, as my router does not support any of the nat configuration protocols. I want to establish whether it's a router issue or a FreeSWITCH problem. The router is going to be replaced eventually with a small form-factor Linux box and an ADSL2+ card from Traverse Technologies (http://www.traverse.com.au/), but given my priorities at the moment, it won't happen until next year. I can compare SIP traces of that would help. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] upgrading to latest SVN
Ujjval Karihaloo ujj...@simplesignal.com wrote: Getting error below..not sure whats wrong..which line number in what file does this refer to? freeswitch/log/freeswitch.xml.fsxml This will be due to a syntax error somewhere in your configuration. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] upgrading to latest SVN
Ujjval Karihaloo ujj...@simplesignal.com wrote: Does svn update try to merge the config files..Need some help, I think it has added some entries in my config files that is causing tag mismatches.. Building and installing FreeSWITCH won't interfere with your configuration. I would suggest using Git or another version control system to keep track of configuration files. I prefer Git. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] RTP issues (possibly nat-related)
Brian West br...@freeswitch.org wrote: I think the fix for this is coming to an SVN repo near you... so give it a few and update. Thanks Brian! I'll watch the svn logs and update when the fix lands. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] fs_cli Error
t...@a2unlimited.com t...@a2unlimited.com wrote: I'm trying to setup fs_cli on a server that is not running the FreeSWITCH server, and I keep getting the following error: error while loading shared libraries: libedit.so.0: cannot open shared object file: No such file or directory For me, under Debian, it's in the libedit2 package. However, fs_cli isn't looking for it, at least not directly. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Configuring freeswitch with voicepulse
Paul Thirumalai paul.thiruma...@gmail.com wrote: I am really new to VOIP and having a hard time with this. I am really not sure how to proceed. Any help would be really appreciated. First, turn on debug logging (in fs_cli, it's /log debug) to obtain more information. The proxy variables in your configuration could be complicating the situation unnecessarily - try removing them and specifying only the realm. I don't think you want two proxy variables here. If you're just new to FreeSWITCH, leave the debug logging level on and read the logs in /opt/freeswitch/log/freeswitch.log to track down problems. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Newbie trying to setup Cisco 7940 phones
Peter J. Zandvoort pe...@cindyandpeter.com wrote: After looking at various asterisk distributions, SipX, 3CX and what-have-you, I've come to the conclusion that FreeSWITCH is by far the most advanced platform out there. Its architecture and performance is literally light years ahead of the rest and I have yet to come up with something that it can't do. But all that comes at a price: The learning curve is like scaling a brick wall. The most flexible and sophisticated tools tend to have this characteristic, the best solution to which is a supportive community and good documentation. FreeSWITCH has the community; the documentation is improving thanks to ongoing efforts to extend, clarify and enhance the wiki. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] SIP Proxy with direct media path
With MySipSwitch it is possible to: - register service with multiple SIP providers as client This is easy once you understand the FreeSWITCH XML configuration. - register your ATA with MySipSwitch That's really a special case of the above. - create smart dial plan (in Ruby) to route calls based on prefixes and/or call source and some other logic FreeSWITCH can do better than that: there are lots of variables which can be matched by Perl-compatible regular expressions, including the destination number. - accept and route calls from all registered providers That's handled in the same way as the previous item. MySipSwitch does not have any media capabilities - RTP is always established directly between call parties (no re-Invites either). FreeSWITCH gives you a choice: you can use proxy media or bypass media modes if you don't want it to handle the media. If this is possible - is there any beginner guide for this? http://wiki.freeswitch.org/ is the best there is. It is being improved by the community over time. You can also take advantage of the IRC channel, the FreeSWITCH conference and of course the mailing list. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] ANNOUNCEMENT: FreeSWITCH 1.0.5pre3 is available for testing!
Craig Askings cr...@overthewire.com.au wrote: Is the plan to run 1.0.5 as a stable branch with bug fix updates or will it be the case of just follow the trunk like it is with any problems you encounter in 1.0.4? I can't speak for the developers, but I would expect the current policy to continue: bug fixes are placed in trunk, and bug reports should be submitted against whatever the current trunk revision is at the time of lodging the bug reports. The maintainers of FreeSWITCH are very effective, in general, at keeping SVN trunk relatively stable. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] can't dial from IPv6 to IPv6
ineya ineya ine...@gmail.com wrote: Sort of. I have different error now: 2009-10-21 08:34:08.446939 [NOTICE] mod_sofia.c:1521 Pre-Answer sofia/internal-ipv6/1...@franta.openstage.net! 2009-10-21 08:34:08.462967 [NOTICE] sofia.c:3925 Hangup sofia/internal-ipv6/sip:1...@[2000:2::1002]:5062 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION] I suspect the codec negotiation. Make sure that both ends are offering a common codec. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] can't dial from IPv6 to IPv6
ineya ineya ine...@gmail.com wrote:
Hmm, I didn't about this.
so in directory/default.xml I have:
param name=dial-string
value={presence_id=${dialed_us...@${dialed_domain}}${sofia_contact(${dialed_us...@${dialed_domain})}/
and the modified version for IPv6 would be ?...
param name=dial-string
value={presence_id=${dialed_us...@${dialed_domain}}${sofia_contact(internal-ipv6/${dialed_us...@${dialed_domain})}/
Yes. Are you suggesting it didn't work?
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Re: [Freeswitch-users] freeswitch french community
Diego Viola diego.vi...@gmail.com wrote: I propose that we start a new documentation from the ground up, we really need a better wiki engine that supports inline translation and a better search engine. I'm open to do all the hard work like porting the entire documentation to it. Is there a good wiki engine which allows the documentation to be edited as files in a file system as well as via a Web interface - something like Ikiwiki which uses Git or another version control system to store the files, perhaps? On occasion I have wished I could add to or update the wiki easily, but there are two problems. 1. The current wiki requires a captcha to be completed in order to register an account. I can't see the captcha - this is an accessibility issue. 2. Editing anything through a Web interface is painful. If I could edit the content as files in Emacs or Vim it would be much easier, and I would be more likely to do it instead of deciding that the difficulty and inconvenience are simply too great. Obviously this is only one person's perspective: other people will have very different needs from mine. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Subject:, Re: FreeSWITCH-users Digest, Orien L. answer to Mike G.
Russell Mosemann russell.mosem...@cune.org wrote: Yup, I got the slashes wrong, and I was staring right at the commands when I wrote the message. I'll blame it on how late it is here. :-) It should be /quit, /bye, /exit and /help. ctrl-d is also recognized and it's easier to type. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Brazilians (Off-Topic)
Rudá Cunha r...@ruda.com.br wrote: Hello, I wanted to create a forum so we can discuss FreeSWITCH. You're writing to one now. It can be accessed as a mailing list, on the Web at gmane.org, and via NNTP at gmane.org. I think there's a Web archive somewhere else, too. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Brazilians (Off-Topic)
Itamar Reis Peixoto ita...@ispbrasil.com.br wrote: instead of creating thousand of small sites, why don't put the content in only place. freeswitch.org already exist, no need for a new domain. This mailing list, the development mailing list, the IRC channel and the FreeSWITCH conference all exist - no need for another forum. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Brazilians (Off-Topic)
Diego Viola diego.vi...@gmail.com wrote: I'm with Moises and with the other people supporting this initiative. I'm not Brazilian, but they should be able to do whatever they want, after all, that's how open source works, if you can do it go ahead and do it. Correct. We have enough of them as far as English-language fora are concerned; other languages are a different question altogether, though. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] failed in bridging with SIPp
Erwin Davis davis.er...@gmail.com wrote: action application=bridge data=sofia/internal/profile1/$ 0...@192.168.1.36/ The above syntax won't work. data=user/$...@192.168.1.36 (assuming that your FreeSWITCH domain is 192.168.1.36) would be better. If the other end doesn't require authentication (i.e., it isn't registering with FS) you can use sofia/internal/$...@domain ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] failed in bridging with SIPp
Erwin Davis davis.er...@gmail.com wrote:
I prefer the second case. How to define the domain? Thanks,
It's the domain of FreeSWITCH itself. You could just specify $${domain} if you
aren't using multiple domains.
user/6...@$${domain}
By default, the domain is the IPv4 address of the machine running FreeSWITCH,
which you can change by setting it in vars.xml (assuming that you're using the
default configuration; it will be different if you totally rewrite the
configuration, of course.)
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Re: [Freeswitch-users] dingaling: Destination out of order
Mark Campbell-Smith mcampbellsm...@gmail.com wrote: When I dial (which is to call my gtalk user), I get the following in the console: [snip] Could you turn on debug logging in the console and post the output? ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Re corded file as voicemail.
Seven Du dujinf...@gmail.com wrote:
originate {ignore_early_media=true,RECORD_STEREO=true}sofia/gateway/xx/xx
bridge(sofia/gateway/yy/yy)
Shouldn't that be record_stereo=false for mono recording?
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Re: [Freeswitch-users] Status of ubuntu/debian packages.
for those who are working on the Debian packages, the following ITP bug may be of interest. http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=513606 which also refers to an older entry. Thanks are due to all who are contributing to the packaging effort. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] mod_fax compile fails
Mark Campbell-Smith mcampbellsm...@gmail.com wrote: Thanks Rob, Is this a fault in the svn update process? Usually, no, it's rather that if you build in a directory that you've checked out and then run svn update, not everything gets cleaned up properly the next time you build. I find it's faster to build in a tmpfs file system anyway, e.g., mkdir /tmp/fs sudo mount -t tmpfs tmpfs /tmpfs svn export . /tmp/fs then go into /tmp/fs and run the compilation process, or the package building command, or whatever you need. This way, my original svn tree is never altered and it will always update cleanly. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Re corded file as voicemail.
Nagalenoj nagale...@gmail.com wrote: No, When I do voicemail_inject and check through voicemail, it is not playing the file instead reporting the following error. 'Stereo is currently not supported, please downsample to mono.' This has been discussed on the list before. I think the solution was to use sox to convert the files to mono. You might need a script to do this before injecting them into the voicemail. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] mod_xml_curl
srinivasula reddy srinivas.ksvre...@gmail.com wrote: Thank you very much for your reply, i have gone through the link what u have sent me, they have given how to access remote system through webserver, but they have not given how to access from local system, On that page, there is an example of how to use mod_curl to access a local system. Do you know what localhost means? You can't use a file:// URL, you have to use an HTTP URL and set up a (small) HTTP server on localhost. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] mod_opal - call charged before H.225 connect
Brian West br...@freeswitch.org wrote: From what I have been told h323plus is a based/fork of OpenH323 which OPAL is just a continuation of OpenH323. From a quick search of gmane.org, the situation seems a little more complicated. http://article.gmane.org/gmane.comp.telephony.openh323.general/10930/match=h+460+opal ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] problem with compiling freeswith
Woody Dickson woodydick...@gmail.com wrote: Is this just me who is having this problem? I can't compile the latest freeswitch source code and here is the error: Try starting with a fresh checkout from the repository. If the problem persists, please report the operating system and version thereof, so that someone with access to a similar environment can try to reproduce the issue. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] mod_opal - call charged before H.225 connect
Tihomir Culjaga tculj...@gmail.com wrote: I understand your financial point of view, but anyhow while the entire world is wants sip and trying to move to sip, the reality is quite different. The majority of voice traffic exchanged via IP is still H323. Is there any evidence in support of the above assertion (e.g., survey results of VoIP traffic)? I've heard of H323 but I don't know anyone who uses it, or any phones that implement it. The lack of interest in this forum and the absence of financial support to improve the H323 support in FreeSWITCH suggest that the level of demand for this is quite low, relative to SIP. Of course, improvements are always welcome, so if you're interested in funding better H323 support, or helping with the module I'm sure the FreeSWITCH community would welcome your efforts. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Minimum audio length for uuid_record
Dan Le dule.maill...@gmail.com wrote: We're running into a problem with the minimum file size when recording using uuid_record. It seems if the audio is too short it deletes the audio file. Is there a way to override that? Yes. It was discussed on the list recently. I suggest searching the list archives. Someone may have documented it on the wiki by now also. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] conference participant from behind NAT
RobertT siniy...@gmail.com wrote: Where is the problem? Is it NAT, closing RTP port after some silence period from client? It could be a time-out, i.e., the nat router isn't keeping the port translation alive. I don't like nat at all. As more people migrate to IPv6 the problem will gradually go away. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] SILK speech codec, Celt, FreeSWITCH
Brian West br...@freeswitch.org wrote: SILK will NEVER take off if they don't stop jerking developers around. I have a silk binary but its 32bit only... You CAN NOT link a 32bit .a file into a 64bit .so just won't work. And I emailed about this fact and I got brushed off .. it takes only a few seconds to compile a 64bit version yet they seem to not be interested. The submission to the IETF changes this situation in that, subject to patent issues, someone could write their own implementation. CELT will win this battle. Its Open it works NOW. I hope Celt succeeds; I like its audio quality. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] [Freeswitch-dev] Status of ubuntu/debian packages.
[Just catching up on this thread.] William King quentus...@gmail.com wrote: I would be more than happy to share the code I use. Here is the git repo: http://git.home.quentustech.com:9191/cgit.cgi/freeswitch-ubuntu/ When you would like your changes to the Debian build infrastructure in FreeSWITCH to be tested on Debian Sid, I'll gladly volunteer, excluding any modules that depend on proprietary software that I don't have and don't want, e.g., Skype. I would also like to see these changes integrated into the FreeSWITCH repository to replace what is currently in the debian directory, once you have a version that is well tested. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Mod_perl $session in not hangup
lakshmanan ganapathy lakindi...@gmail.com wrote: Thanks for your replay. I don't know what is latest trunk. Is it latest version? I'm using freeswitch 1.0.4. It's the latest version from the svn repository. Use svn checkout, then compile it as documented on the wiki. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Debian, libtool2 and mod_portaudio
Michael Jerris m...@jerris.com wrote: What issues are there with libtool 2 under debian? Libtool 2 issues that I am aware of were all sorted out quite some time ago. With libtool2, mod_portaudio fails to link to the Alsa sound library, hence fails to load due to unresolved symbols. I can't test this just now, due to the Ogg vorbis issue mentioned recently in this thread. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] CALL FOR VOLUNTEERS: Assisting With FreeSWITCH Subprojects
Michael Jerris m...@jerris.com wrote: A couple people have taken on major work on packages for ubuntu. Most of that work will translate directly back to debian, we should just need people to do testing of debian pacakges once their work is done. I'm volunteering. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] A real puzzler for you guys... (perpetual registration causes denial of service.)
Karl Vesterling k...@ken-ton.com wrote: No penguin is perfect... There's issues w/ 2.6.X - 2.6.27.X with respect to timing for things like packet shaping, which is a requirement for me. Two suggestions: 1. Your distribution's bug tracker. 2. http://ltp.sourceforge.net/ (If they get test coverage of the relevant interfaces there will be quicker detection of problems and, we hope, prevention of regressions.) ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Build problems with Shoutcast module under Debian
While trying to build FreeSWITCH rev. 14913, compilation failed with the following. the operating system is Debian Sid. Ogg development files are installed, but libogg.la does not exist anywhere. I'm still using libtool 1.5.26, because the build problems with FreeSWITCH and libtool 2 under Debian haven't been resolved. As soon as someone takes over the Debian packaging I'll gladly help out with testing and fixes - I'm far too busy at the moment to work on it intensively. ranlib .libs/libshout.a rm -fr .libs/libshout.lax creating libshout.la /bin/sed: can't read /usr/lib/libogg.la: No such file or directory libtool: link: `/usr/lib/libogg.la' is not a valid libtool archive make[10]: *** [libshout.la] Error 1 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Build problems with Shoutcast module under Debian
It turns out that Debian recently removed the libogg.la file, deliberately, from the package. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] How to Process Invalid extension in FS
Ahmed Munir ahmedmunir...@gmail.com wrote: I'm newbie in FS. I want to know how to process invalid extension in FS? Because I want to prompt the IVR if invalid extension is dialled. You could write an entry at the end of the dial-plan that matches any extension and invokes the IVR. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] A real puzzler for you guys... (perpetual registration causes denial of service.)
Karl Vesterling k...@ken-ton.com wrote: What Kernel Bug: It's a kernel bug that corrupted the sqlite database. This caused Freeswitch to refuse the phones registration request. Please take this up with your Linux distribution as a bug report related to the kernel, and persist with it until it's sorted out. The more that users do this, the more kernel bugs will get fixed. We're all responsible to some extent for the quality of our free/open-source operating systems. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] how can i tell freeswitch listen on 0.0.0.0 and a fixed ip
jun yang yj13535428...@gmail.com wrote: when freeswitch start ,it auto bind to the pubic ip, so the lan user cann't connect to freeswitch use lan ip. i have setting X-PRE-PROCESS cmd=set data=bind_server_ip=0.0.0.0/ but have no effect, freeswitch also auto bind to the public ip. any help is thanks. Set local_ip_v4 in vars.xml to your desired IP address. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] filter in fs_cli
João Mesquita jmesqu...@freeswitch.org wrote: No can do. There are better tools to do that. tshark, wireshark and all other variants can do that for you. I would recommend learning about read filters in tshark/wireshark, which support a very flexible filtering language that is suitable for capturing SIP traffic. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] example configs for FS outside of NAT?
Nandy Dagondon nandy1...@gmail.com wrote: for outside clients to register w/ the internal profile, the router has to forward port 5060 to FS. am i correct? Yes, but by default the internal profile doesn't handle nat, which is why (if I recall correctly) it has been recommended that the external profile be used to register clients that are not on the local network when nat is involved. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] switch_core_io.c:118 sofia/internal/XXXXX has no read codec.
Yehavi Bourvine yehavi.bourv...@gmail.com wrote: I have a problem when trying to put a call on hold: I get the above message and the call is disconnected. Any idea where to look for the source of the problem? My next step in your situation would be to obtain a Sip trace and post relevant details from it to the list. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] No dial-string available error
Leon de Rooij l...@scarlet-internet.nl wrote: For examplle, right now, when I have two profiles where users can register with the same domain (for example one with a v4 and one with a v6 address), what should I do ? I can't alias the same domain to two profiles, so that means I have to call sofia_contact twice, once for each profile ? The latter partly works, but as several colleagues and I discovered, doing this with the | syntax (i.e., two sofia_contact function calls separated by |) breaks the group call functionality of FreeSWITCH. There needs to be a solution that allows a user to register over v4 or v6 and to be contacted, without breaking group calls. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Conferencing setup with FS
Ujjval Karihaloo ujj...@simplesignal.com wrote: I cannot seem to find a Document online for setting up conferencingon FreeSwitch. Can someone point me to one? Have a look at http://wiki.freeswitch.org/ and search for conference. There's a document describing mod_conference there. Also look at the default conference configuration supplied with FreeSWITCH. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
Ken Rice kr...@freeswitch.org wrote: I know there has already been some discussion on several fronts of atleast getting the core and several other pieces to where they need to be for stateful failover and I'm not sure if its been mentioned here, but sofia is going to be a bit of work and estimates run in the 100K USD range. Now if we could get a get a couple of corporate sponsors to help here it would be great. I agree. Perhaps some of those corporations which are saving money over proprietary solutions by using FreeSWITCH, and who also want the high-availability functionality, would be ideal sponsors for the work. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] What is the difference between preAnswer and ring_ready?
Max Bridgewater max.bridgewa...@gmail.com wrote: Hi, Assuming an inbound call, I have trouble understanding what the supposed difference between the following two set of instructions is: session.execute(ring_ready); session.set(set, ringback=/home/ring.wav); Ring_ready sends a SIP ringing message and will use your device's default ringback sound - at least, that has been my experience. I should remember the message number in the SIP protocol, but I don't (the SIP experts here will undoubtedly have the details at the ready). and session.preAnswer(); session.streamFile(/home/ring.wav) I think that will play the file as early media. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
Pete Mueller p...@privateconnect.com wrote: There is a need for ensuring that calls do not drop, but we must balance that with the cost of making the system redundant. We took some small, inexpensive measures, to improve our odds, but we could spend a lot more, for basically nothing more than giving some client a warm fuzzy. I think this is one area where, as indicated earlier in the thread, a lot of development effort would be needed to obtain that extra degree of reliability. From a broader perspective, the question is whether, over the next decade or two, VoIP can compete with the PSTN in reliability. My (limited) understanding is that PSTN equipment typically achieves 99.9% uptime, and if VoIP systems are going to play in that arena, it would be desirable for free/open-source software to do so. If FreeSWITCH itself is working correctly, all you need is a hardware failure or a kernel panic or a network outage to drop that up-time substantially, not to mention dropping the calls as well, which I've never experienced as a user of the PSTN due to equipment at the telephone exchange. I have, however, experienced some rather low-quality PSTN calls over international lines, which have the added disadvantage of being expensive to use. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] memory leak
Jay Binks jaybi...@gmail.com wrote: Reason I ask ... I personally only have a preference for debian, but others may have policy mandated Os's For their companies, and it would be great to have some info about this. The only problem I've had with FreeSWITCH under Debian Squeeze and Sid involves TLS-related segmentation faults that appear to be related to something in the version of libssl supplied with Debian. The same problem can't be reproduced on Fedora, for example, but it does occur under Debian Lenny as well as Debian Squeeze and Sid (i.e., testing and unstable, respectively). besides this, all of the issues that I have encountered turned out to be (usually short-lived) bugs in FreeSWITCH or one of the libraries included in the source tree - they're mostly FreeSWITCH issues. I should point out that the FreeSWITCH developers are very good at avoiding the introduction of bugs into their code and that known bugs get fixed. It appears to be an unchanging fact about programming that with a large and complex project, even given highly knowledgeable, experienced, committed and talented developers (as we have with FreeSWITCH), sometimes, bugs do slip through. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Inboud Call Queue
Diego Viola diego.vi...@gmail.com wrote: I was wondering if some of you run FreeSWITCH on a call center environment, I ask this because I plan to do that soon and I was wondering how well mod_fifo works for queues, etc. This was mentioned on the list once before, and it might be what you want: http://wiki.opencsm.org/wiki/index.php/Spice_Telephony (Spice Telephony). ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] can't pass full sip url to dialplan
Henry Huang red.rain.se...@gmail.com wrote: It that case, the example of dialing sip_uri in the dialplan/default.xml should be removed to prevent confusion. Because according to what you said, one can never be able to hit this extension: It is entirely possible to reach this extension, but notice that the sip: prefix is removed before the rest of the URI is used in calling the bridge application. If you don't understand why this dial-plan entry works, go back and read about regular expressions and the format of destinations used with the bridge application. There are examples and explanations on the wiki. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Transporting SIP over TCP
Tzury Bar Yochay tzury...@reguluslabs.com wrote: Brian/Bakko, Would you please tell me which softphone are you using? As you know, my own one is not working and when I tried tcp with xlite (providing transport=tls) I see in wireshark that it is still transporting it over udp(!) I've successfully used TLS with FreeSWITCH at both ends (yes, that's with FreeSWITCH itself as the softphone). Snom 320 phones are known to work as well if you set up SRV records for the TLS. I haven't tried other softphones because, basically, FreeSWITCH is a better phone than anything else I can find. To debug this, try wireshark or tshark to find out whether your softphone is trying to connect over the TLS port at all. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Transporting SIP over TCP
Tzury Bar Yochay tzury...@reguluslabs.com wrote: Well, as I said at the beginning of this thread, TLS works fine for me. The problem is when using TCP (transport=tcp and not transport=tls) I'm not sure whether that's supposed to use TLS. I suspect not. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Transporting SIP over TCP
Tzury Bar Yochay tzury...@reguluslabs.com wrote: I think I confused you with this TLS/TCP thing. For the sake of clarification, I am talking about TCP and _not_ about TLS. That is simply transporting the signaling packets over TCP instead of UDP. No TLS should be involved at this stage. It is simply communication layer matter. I can confirm that transport=tcp works fine here from one FreeSWITCH to another. (I'm doing it over IPv6, but that shouldn't make any difference). ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Not receiving DTMF
Nicolas Brenner nico...@medularis.com wrote: My question is: since my provider is not doing RC2833 dtmf (even though they say they do), is there another way to get dtmf to work? You can try info and tone detection just in case one of those is being used, but it appears from your message that you have attempted both of these already. I think it's time to find a more reliable provider, or try to persuade your current provider to fix it (the latter is probably a waste of time and effort, however). ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] files.freeswitch.org resets connection.
Diego Viola diego.vi...@gmail.com wrote: Resolving files.freeswitch.org... failed: Temporary failure in name resolution. It must be a problem at your end. ja...@jdc:~$ host files.freeswitch.org files.freeswitch.org is an alias for filessync.freeswitch.netdna-cdn.com. filessync.freeswitch.netdna-cdn.com has address 69.174.57.101 Note that I am running my own Bind 9 daemon on this host. The record wasn't cached, as it has been a few weeks since I last upgraded FreeSWITCH. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] New to Freeswitch - some help needed
Alan Chandler a...@chandlerfamily.org.uk wrote: I want to set this up as a small private voice network, so anyone can ring anyone else. I will add fancy facilities such as conferencing and voicemail later - I just want to get the basics working first. I have a similar arrangement operating here which involves friends and colleagues in the U.S., as well as a local VoIP provider that gives me access to the PSTN. To eliminate NAT issues, we are using IPv6: each of us has an IPv6 over IPv4 tunnel configured to provide access to the IPv6 Internet. NAT and all the problems associated with it go away. Another option, although I don't know how well real-time communication works in this setting, would be to create a VPN using, for example, OpenVPN so that the clients and server all appear to be on the same lan. Alternatively, you could play with port forwarding and FreeSWITCH settings in an attempt to work around the nat issues - good luck! I can't answer any questions about MacOS or Windows softphones - there are no MacOS or Windows machines in my life. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Numeric Value Ranges Expressions in dialplan
Dome Charoenyost d...@tel.co.th wrote: Is posible to check numeric range in dialplan (expression). example i got balance vaiable from somewhere and want to check 0 or not before call bridge application. ( I don't want to call scripts) Can you write a regular expression to match it? ^[1-9]\d*$ for example, might be a good start to identify non-zero integers. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Enable sip communication between two Freeswitch servers
Gregory Charles gregory.char...@sogeti.com wrote:
2009-07-29 17:20:18 [ERR] sofia_glue.c:568
sofia_glue_ext_address_lookup() STUN Failed!
stun.freeswitch.org:3478 [Timeout]
Set external_sip_ip and external_rtp_ip to something reasonable, e.g.,
$${local_ip_v4} in vars.xml, or change it in your external.xml SIP profile.
Either way, get rid of stun:stun.freeswitch.org on both machines and try
again.
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Re: [Freeswitch-users] SIP trunk (PBX - FreeSwitch) routing problem.
julien jgonza...@sqli.com wrote: It was not exactly at the bottom but before X-PRE-PROCESS cmd=include data=default/*.xml/ Why not put it in the default directory, from which it will be included by the above line? If necessary, you could comment out any entries in default.xml that might be matched first. I've debugged this kind of problem before, and the best solution has always been to read the logs carefully to see which extensions matched (or didn't match). Also, if necessary, check out freeswitch/log/freeswitch.xml.fsxml to see where your extension ends up in the final dial plan. ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Dial plan contexts
Has anything changed in the handling of dial plan contexts recently? As of rev. 14363, the context setting in the Sofia profile seems to be overriding the context setting in the user's definition in the directory. As per the default configuration, I have user-context set to public in my internal profile, my user has its context set to default, but calls made from the phone registered to that user ID end up in public context when they reach the dial plan. Either something has changed or there's something wierd in my configuration that I haven't tracked down. I haven't made any changes to any of the profiles or users recently, though, and it was working under an older revision. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Multiple gateways: Priority the first bridge with prefix
Edmar Cruz darklio...@yahoo.com wrote: Not working just the same both of them are running Do you have them as separate extensions in the dial plan? ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Dial plan contexts
From the profile: param name=context value=public/ From the user's entry in the directory: variable name=user_context value=default/ but under rev. 14363 when the phone registered to that user makes a call, the dial plan is searched in public context. I hope this helps to clarify. I tried resetting my configuration using Git to a known good state, but with no change to the above behaviour. I'm going to rebuild with the latest from svn soon. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Dial plan contexts
With apologies to all, it was something that sneaked into my configuration that I'm still tracking down. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Multiple gateways: Priority the first bridge with prefix
Edmar Cruz darklio...@yahoo.com wrote: Yes, I actually just want to not be able to communicate with the other bridges. I have already this extension name = sample-1. Freeswitch gets the first extension the 2nd also trigger it. When the calls finds the match it suits perfectly but I just want that I do not want to view the bridges with CS_DESTROY or hangup_after_false if not found. The above text is absolutely incoherent and incomprehensible, so I don't understand what you are trying to say. Try setting action application=set data=continue_on_fail=true/ on the first extension and see whether that does what you want. I hope this help. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] A stun server lookup
velusamy velu velu.techni...@gmail.com wrote: I commented the stun configurations in vars.xml.conf file eventhough I am receiving the same error. Pleas any one give solution to solve this error Edit vars.xml, change the variables that use Stun to be wahtever you want your ext-sip-ip and ext-rtp-ip addresses to be, then restart the external profile sofia profile external restart reloadxml or restart FreeSWITCH. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Problem in Adding another user in default directory
velusamy velu velu.techni...@gmail.com wrote: How to create another user agent like 1000 to 1919 in internal profile. Copy one of the existing files, edit it, and make all of the obvious changes. Then edit your dial plan so that the extension can be called. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] SIP Trace Option at Runtime
Muhammad Shahzad shaherya...@googlemail.com wrote: Is there any CLI command to enable / disable SIP packet trace at runtime. sofia profile profilename siptrace on sofia profile profilename siptrace off sofia help would have answered your question. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Error in default Sofia profile checking
velusamy velu velu.techni...@gmail.com wrote: When I register my Softphone(Twinkle) with predefined sofia registration(1000 with password 1234). I have got the following error in FreeSWITCH console. 2009-07-11 09:37:16 [ERR] sofia_reg.c:1135 sofia_reg_handle_sip_i_register() NO CONTACT! Activate sip tracing on the profile (e.g., sofia profile internal siptrace on), try to register again and save the trace. This should help you to solve the problem. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Error in default Sofia profile checking
Mathieu Rene mrene_li...@avgs.ca wrote: Chances are the registering UA didnt provide a Contact header (required by rfc3261) Just what I thought, hence the suggestion to obtain a sip trace. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] How to modify the Subject and Body when sending voicemail??
Brad Tuan brad.t...@gmail.com wrote: As title, How to custom the Subject and Body and ... of the mail ?? Have a look at the notify-voicemail.tpl and voicemail.tpl files, and the template parameters in voicemail.conf.xml in the default FreeSWITCH configuration to see how it all works and to decide what to edit. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] dp tools record to 1 channel but uuid_record record to 2 channels which cannot been playback
seven dujinf...@gmail.com wrote: 1) is it the default behavior that uuid_record record with 2 channels Yes. 2) is it reasonable that FS can record to 2 channels but cannot playback? Could you explain what happens when you play back the files? 3) do I need to set RECORD_STEREO=false before uuid_record? That depends on whether you want two-channel output files or not. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] dp tools record to 1 channel but uuid_record record to 2 channels which cannot been playback
seven dujinf...@gmail.com wrote: yes, it's here: http://pastebin.freeswitch.org/9641 Judging by the error message, it's a known limitation. You are welcome to work on a fix, or pay the develoeprs to fix it, or offer a bounty that might encourage someone to work on it, or wait until it gets fixed. Meanwhile, convert the file to mono and try again. Sox should be able to do this, for example. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] -ERR SERVICE_NOT_IMPLEMENTED on mobiles and landlines
Edmar Cruz darklio...@yahoo.com wrote: I have a GSM gateway. The issue is sometimes the calls failed what is the cause of the error this is my logs? The cause of the error is that you are searching the dial plan for 639273642511 in context public, and no dial plan entry matches, so FreeSWITCH terminates the call. It's your task to look at your configuration and work out why this is happening, since clearly it isn't what you intended. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] -ERR SERVICE_NOT_IMPLEMENTED on mobiles and landlines
Edmar Cruz darklio...@yahoo.com wrote: I think the problem is on the bridge No, it's in the fact that FreeSWITCH fails to match the destination number in the public context. If you've placed the user that is making the call in the public context, and the dial plan entry that you want to match the destination number is in the default context, it isn't going to work. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Could this be a bug in the SIP registry?
Mitchel Constantin mythical...@weavver.com wrote: 5. My phones now register using the correct domain name (i.e. weavver.com) instead of the IP address (205.134.225.20) as the domain. 6. Now the problem... My originate command no longer works using the new syntax: originate sofia/internal/mythicalbox%weavver.comsofia/internal/johndoe% weavver.com The phones do show up as registered when I type sofia status profile internal: What happens if you use the following syntax? originate user/ph...@domain extension e.g. originate user/1...@example.com 3000 to connext u...@example.com to extension 3000. My other advice would be to read the FreeSWITCH log files carefully. Also, use the sofia_contact command to find out how the registered users will be called when the syntax mentioned above is used. Make sure that everything will go where you want it. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Any advances on T.38 support for FS?
François Delawarde fdelawa...@wirelessmundi.com wrote: Is there any work planned for T.38 termination (in mod_fax)? Yes, as discussed on the mailing list recently. If you're volunteering to help, I'm sure the FreeSWITCH developers would appreciate contributions of code. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Could this be a bug in the SIP registry?
Jason White ja...@jasonjgw.net wrote: originate user/1...@example.com 3000 to connext u...@example.com to extension 3000. That should read to connect 1...@example.com to extension 3000. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] How to know my gateway registering is successed??
Brad Tuan brad.t...@gmail.com wrote: As title ,Does FS keep the status of gateways?? sofia status gateway gateway-name ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] How to know my gateway registering is successed??
Peter P GMX prometheus...@gmx.net wrote: or simply sofia status for all gateways and, from the shell, fs_cli -x help helpfile fs_cli -x sofia help helpfile and any others you need so as to obtain synopses of all the commands that you might need. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] How to know my gateway registering is successed??
Brad Tuan brad.t...@gmail.com wrote: Another question, Where does FS keep these information?? In *.db or somewhere?? It's a hash table in memory. See sofia_reg_find_gateway__ in sofia_reg.c for the code that performs the hash table lookup and returns a pointer to the structure with all of the fields in it. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] How to remove the IP from the SIP caller id number
Mitchel Constantin mythical...@weavver.com wrote: I'm working on configuring my FreeSWITCH and would like to set the caller id number like this in dialplan/default.xml: action application=set data=effective_caller_id_name=John Doe/ action application=set data=effective_caller_id_number= john...@weavver.com/ I wonder if this is a problem with eyeBeam.. When the call is received the CID is like this: John Doe john...@weavver.com@205.134.225.20 205.134.225.20 is the EXT IP of the switch I suspect the other end (whatever device you are calling from FreeSWITCH) is adding the IP address to the caller id. However, I am no SIP expert and may be wrong, but you can confirm this by doing a SIP trace on the device that receives the call (or on its local network via packet capture) to discover what FreeSWITCH is sending out. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] ANNOUNCEMENT: FreeSWITCH 1.0.4pre9
Saeed Ahmad saeedahmad1...@gmail.com wrote: What is the best way to update to latest version if we are already running an older stable version? You did ask for the best way, which is to build packages for your operating system, then use your operating system's package manager to install them and keep track of different versions. This way, you can be sure that the right files are installed, that old versions are cleanly deleted (unless there's an error in the post-installation script, in which case it's a bug) and you can use the package manager to find out what files are installed and where they reside. You can then upgrade or downgrade simply by installing a different version of the package. FreeSWITCH supports building Debian packages, and there is also support for Centos and Fedora. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] ANNOUNCEMENT: FreeSWITCH 1.0.4pre9
Saeed Ahmad saeedahmad1...@gmail.com wrote: Can you give me more info with CentOS. I am more comfortable with SVN trunks, can i do the same with SVN trunks? Yes. There is a spec file in the source tree for building packages. There should be instructions on the wiki explaining how to use it - if not, someone who is more familiar with rpm-based distributions than I am will be able to refer you to instructions on how to build an rpm package. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Is there any license G729?
Edmar Cruz darklio...@yahoo.com wrote: Is there any available license G729 for freeswitch? Yes. It was announced here a few days ago - see the list archives. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Is there any license G729?
Brian West br...@freeswitch.org wrote: Everyone has this need for lower bandwidth calls... I tend to march the other way. 48kHz baby! (btw you can do 48kHz in the same bandwidth as a single ulaw call) 48khz Celt (c...@48000 in your codec preferences) sounds wonderful with FreeSWITCH. To test, run two FreeSWITCH instances, both with mod_portaudio. This also works well in 48khz conferences. I wouldn't use G.729 even if it weren't encumbered by patents - it's G.711, G.722, G.722.1 and (my current favourite) Celt all the way. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Is there any license G729?
SP spr...@gmail.com wrote: FreeSWITCH runs on a Mac and can be configured as a UA... also supports WB and UWB! Correct. It's especially good for those of us who prefer to avoid WIMP user interfaces. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Hardware phones that do 48Khz Celt: Was Re: Is there any license G729?
Craig Askings cr...@overthewire.com.au wrote: Are there any hardware phones that support 48 Khz Celt and automated/mass deployment? Actually... FreeSWITCH in a phone could be a very good project. The main obstacles are: 1. Someone would need to design and build the hardware, or find existing hardware that would be suitable. 2. A script would have to be written to control i/o so that the keyboard and display of the phone could be used to make calls. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
