Re: [Freeswitch-users] How to connect SIP phone to freeswitch
http://wiki.freeswitch.org/wiki/Getting_Started_Guide http://wiki.freeswitch.org/wiki/Interop_List On Nov 25, 2009, at 1:36 AM, ovvenkat wrote: Hi . Could you please tell me, How to connect sip phone (which one is more friendly with freeswitch) to freeswitch. How I can check whether connection is properly established 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] FS compile error under Windows: error LNK2019
It sounds like the platform sdk is set up wrong. This used to be a problem with older versions of express edition. Double check that your compiler works at all with anything else. Mike On Nov 22, 2009, at 11:51 PM, 大泥人 wrote: All, I tried to compile FS source code under Windows while there are lots of errors: Error LNK2019, external _imp_sl...@4 can not be resolved, this function was referred by _tMCRTStartup. Some other more similiar errors detail information attached. Any ideas? Thanks Daniel Zeng ___ 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] Git
I think this one is kept up to date, but we may re-do this at some point soon, so it may get re-built. http://svn.freeswitch.org/freeswitch.git/ Mike On Nov 23, 2009, at 1:22 PM, William Suffill wrote: Just wondering if anyone is keeping an update to date git repo of FreeSwitch? I been using git-svn to keep a copy on my machines but it can be quite time consuming due to the per revision fetching. If there was a repo to clone that would speed up the process considerably. ___ 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] tcp call misses sip message
This looks like a nat issue to me, please re-test this against latest svn trunk and if its still not working pastebin a full sip trace and report the link back here. Mike On Nov 21, 2009, at 6:23 PM, RobertT wrote: Yep, I use proxy media. First it started with 1.0.4 release, then I've updated a week or two ago with the latest svn trunk, not sure what was the rev number. ___ 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 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] User who answer the bridge in a execute_answer
Try running the info app there and see if the info is anywhere in that output .
Mike
On Nov 23, 2009, at 5:36 AM, Albano Daniele Salvatore - Lavoro wrote:
Hi,
i'm writing some dialplan parts that get executed on execute_on_answer. In
this dialplan that get executed i need to make a directory to handle
recordings for record_session and my folder structure is:
USER/YEAR/MONTH/HOUR-MINUTE-SECOND-CALLER_NUMBER.wav
--
action application=system data=mkdir -p
$${base_dir}/recordings/${sip_from_user}/${strftime(%Y)}/${strftime(%m)}/ /
action application=bind_meta_app data=1 a s
record_session::$${base_dir}/recordings/${sip_from_user}/${strftime(%Y)}/${strftime(%m)}/${strftime(%H_%M_%S)}-${caller_id_number}.wav
/
--
The call flow is:
Call from external - IVR - Transfer to Group - Execute on Answer -
system/bind_meta_app
Pratically, i need the number (or better the user) that answered the call:
what variable should i check?
I tried with sip_from_user, callee_id_number and some other.
Thank for your help,
Best Regards,
Daniele
info.vcf___
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Re: [Freeswitch-users] Execute on Answer with JavaScript
This is done automatically when you bridge 2 sessions together. Mike On Nov 23, 2009, at 6:45 AM, Oscav wrote: How can we send the answer to the caller only when the callee answers, in JavaScript?? ___ 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] Question about rtp-timeout-sec variable
Take a look at a pcap of the traffic, I suspect the other side still has media flowing. On Nov 23, 2009, at 7:00 AM, Maciej Aniserowicz wrote: Hello, I have 2 instances of FS: one controlled by my application (making calls with TCP commands, recording sessions, listening to events etc) and one acting as a remote gateway to which all users register. When I leave the default values of rtp-timeout-sec and brutally kill x-lite during conversation, the 'hangup' event with 'media_timeout' cause is obviously sent after the default 5 minutes (and until then, the other leg is still connected to a 'dead' channel). The question is: which FS instance is responsible for terminating the connection after timeout? Only the 'remote' FS instance config seems to work. I thought that the shortest configured value should cause the timeout, but it's not the case. Am I missing something, or is this the correct behavior? ___ 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] Using odbc in FS core
Yes please On Nov 23, 2009, at 6:45 AM, Mike Tkachuk wrote: Hello Anthony, Is clear, thanks, I'll test and will let you know. Should I add 'core-db-dsn' parameter description to Wiki? Maybe we need to add this parameter also to sample conf files? ___ 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] Building in a builddir using --srcdir option but modules still build in srcdir
The Makefile rules that those are built with can all be found in build/modmake.rules.in. I looked them over real quick and they look right, maybe try throwing some debug echo statements in there or build with env var of VERBOSE=1 to see more of what is going on and toss a patch to correct the issue on jira for me. Mike On Nov 23, 2009, at 12:53 PM, Robert Hadley wrote: I am trying to build in a subdirectory off the Freeswitch source. I can configure successfully and have make working for switch files and the libraries, but I am having trouble with the modules in src/mod. They still compile in the src/mod folders. Any ideas? ___ 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_flite sound profiles
Sounds like they don't want your business that much. You can try using mrcp with them , not sure if they have that released on their side or not. I think the build integration for mrcp client just went into the windows build earlier today. To be honest we used to have a pretty good relationship with them but we have had basically no response at all to any technical problems we have had with them in quite some time, so maybe they have decided to move on and not work with open source any more. It would appear so from their actions at least. Mike On Nov 23, 2009, at 1:41 PM, Malay Thakershi wrote: Thank you for your responses. I did follow that web link to ask them as instructed but they declined. They asked me where I want to use it. I told them I wanted it to build FreeSwitch so that I can use Cepstral voices (to be purchased from them with it). Their response was they do not provide trial of the SDK. They do not support FreeSwitch. Malay Thakershi From: Brian West [mailto:br...@freeswitch.org] Sent: Monday, November 23, 2009 12:14 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_flite sound profiles You don't have to buy the SDK... I have had it sent to everyone that has asked me for it... the address is on the wiki for who to contact. If you were using linux the SDK is included already. /b ___ 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] Building in a builddir using --srcdir optionbut modules still build in srcdir
In these builds how is it supposed to work, do generated files like Makefiles get put it builddir or srcdir? Mike On Nov 23, 2009, at 2:54 PM, Robert Hadley wrote: Thanks Mike. modmake.rules is created in the $(switch_builddir)/build. What I see as the problem is in src/mod/Makefile.am There is a statement line 12 that points moddir to the source if test –d “$(switch_srcdir)/src/mod/$$confmoddir” ; then \ moddir = “$(switch_srcdir)/src/mod/$$confmoddir” ; And then the statements starting around line 22 that cd to moddir (in src) and fire off make if test –f “$$moddir/Makefile” ; then \ß Yep, this will be true cd $$moddir … $(MAKE) I’m not sure what to change to get it to build in $(switch_builddir), and getting the source automatically from $(switch_srcdir). My old-fashion brute-force idea is to symlink the source src/mod/subdirs in the build src/mod/subdirs right before line 12, changing line 12 to use $(switch_builddir). Does anybody have a better idea? Thanks, Robert From: Michael Jerris [mailto:m...@jerris.com] Sent: Monday, November 23, 2009 11:16 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Building in a builddir using --srcdir optionbut modules still build in srcdir The Makefile rules that those are built with can all be found in build/modmake.rules.in. I looked them over real quick and they look right, maybe try throwing some debug echo statements in there or build with env var of VERBOSE=1 to see more of what is going on and toss a patch to correct the issue on jira for me. Mike On Nov 23, 2009, at 12:53 PM, Robert Hadley wrote: I am trying to build in a subdirectory off the Freeswitch source. I can configure successfully and have make working for switch files and the libraries, but I am having trouble with the modules in src/mod. They still compile in the src/mod folders. Any ideas? ___ 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 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] Simplest of Conference Setup questions
Default controls are hard coded. If you want to change them you must use a name other than default. Mike On Nov 23, 2009, at 3:42 PM, Phillip Jones wrote: Anthony - setting control action=hangup digits=9/ or control action=hangup digits=event/ does not make a difference, even when the default profile has param name=caller-controls value=default/ un-commented. Looks to me like that default group is ignored even when specifically referred to? As Michael says though, creating a specific group: group name=myConf and adding param name=caller-controls value=myConf/ in the default profile works a charm. I am good - but let me know if you want me to try anything else. Phil ___ 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] Building in a builddir using --srcdiroptionbut modules still build in srcdir
I'll work on this, can you open me up a bug on http://jira.freeswitch.org in regards to this please. Mike On Nov 23, 2009, at 4:19 PM, Robert Hadley wrote: In typical automake builds the configure step takes the Makefile.am from the srcdir and generates the Makefile in the builddir. Most src/mod subdirs are not using automake and/or configure. They just have a simple Makefile in with the source. Robert From: Michael Jerris [mailto:m...@jerris.com] Sent: Monday, November 23, 2009 1:09 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Building in a builddir using --srcdiroptionbut modules still build in srcdir In these builds how is it supposed to work, do generated files like Makefiles get put it builddir or srcdir? Mike On Nov 23, 2009, at 2:54 PM, Robert Hadley wrote: Thanks Mike. modmake.rules is created in the $(switch_builddir)/build. What I see as the problem is in src/mod/Makefile.am There is a statement line 12 that points moddir to the source if test –d “$(switch_srcdir)/src/mod/$$confmoddir” ; then \ moddir = “$(switch_srcdir)/src/mod/$$confmoddir” ; And then the statements starting around line 22 that cd to moddir (in src) and fire off make if test –f “$$moddir/Makefile” ; then \ß Yep, this will be true cd $$moddir … $(MAKE) I’m not sure what to change to get it to build in $(switch_builddir), and getting the source automatically from $(switch_srcdir). My old-fashion brute-force idea is to symlink the source src/mod/subdirs in the build src/mod/subdirs right before line 12, changing line 12 to use $(switch_builddir). Does anybody have a better idea? Thanks, Robert From: Michael Jerris [mailto:m...@jerris.com] Sent: Monday, November 23, 2009 11:16 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Building in a builddir using --srcdir optionbut modules still build in srcdir The Makefile rules that those are built with can all be found in build/modmake.rules.in. I looked them over real quick and they look right, maybe try throwing some debug echo statements in there or build with env var of VERBOSE=1 to see more of what is going on and toss a patch to correct the issue on jira for me. Mike On Nov 23, 2009, at 12:53 PM, Robert Hadley wrote: I am trying to build in a subdirectory off the Freeswitch source. I can configure successfully and have make working for switch files and the libraries, but I am having trouble with the modules in src/mod. They still compile in the src/mod folders. Any ideas? ___ 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 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 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 with mod_local_stream
That rev should have fixed that memory leak, could you test mod_local_stream.c from rev 15430 (http://fisheye.freeswitch.org/browse/~raw,r=15430/FreeSWITCH/src/mod/formats/mod_local_stream/mod_local_stream.c) with your current fs version to confirm this is the cause please? Mike On Nov 23, 2009, at 4:53 PM, Rob Forman wrote: Hey guys, Having a problem with mod_local_stream. I recently did a make current from 15334 to the latest trunk (15630). After restarting, there now appears to be a memory leak. On a test system (CentOS 5.4, 64-bit) with no calls or registrations, Freeswitch gradually consumes all of the host memory (rate of about 200K/second), then swaps out, eventually rendering the system useless. I isolated it to mod_local_stream. If I unload mod_local_stream, the memory use stops climbing. If I re-load mod_local_stream, it starts again. I would submit the logs except they aren't any besides it starting. The system is just sitting there idle. Even valgrind didn't show much (http://pastebin.freeswitch.org/11238). Maybe I'm using it wrong? I ran it: valgrind --tool=memcheck --log-file-exactly=vg.log --leak- check=full --leak-resolution=high --show-reachable=yes .libs/ freeswitch -vg Questions: * has anyone else seen this? * what is the best way I can assist troubleshooting this? I saw a patch to mod_local_stream (rev 15431) a few weeks back. Could that have anything to do with it? Rob ___ 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 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] Requesting testing.
I have done quite a few changes to the build system and correcting build problems and other platform specific problems the last few days. Could everyone on the list please take a little time out of their day and do a clean fresh svn trunk checkout of FreeSWITCH and do a full build and report any errors you encounter (if not already reported) to http://jira.freeswitch.org. We have fixed things for many platforms including bsd, solaris, linux, and especially issues on OS X. Please try these out to make sure all works. Thanks Mike ___ 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] need help !! Problem with freeswitch uniMRCP
Jira is the best, otherwise just mail me the patch and I'll take a look. Also, I just synced lib up to current trunk. Can you take a look at my last patch to the module to make it build please. Mike On Nov 22, 2009, at 1:02 PM, Arsen Chaloyan achalo...@yahoo.com wrote: We discussed build integration related issues a few months ago with Mike and seemed to find a solution which would work for both UniMRCP and FreeSWITCH source trees. Now I've just got a chance to look into this a bit closer trying to further complete VS2008 build integration in FreeSWITCH. So I've got it working, the module is not only being built, but also is getting loaded. Current build integration is not as seamless as I want it to be, but probably we can start with what we have now and then discuss and identify what can be done in the future. This concerns not only build integration but overall integrity. So would you be interested in the patch? Where should I upload it? I thought I had a Jira account, but not sure it exists any more. -- Arsen Chaloyan The author of UniMRCP http://www.unimrcp.org From: Jeff Lenk jl...@frontiernet.net To: freeswitch-users@lists.freeswitch.org Sent: Fri, November 20, 2009 7:59:28 PM Subject: Re: [Freeswitch-users] need help !! Problem with freeswitch uniMRCP That module is not currently being built for Windows. Also the library unimrcp needs build integration work with FS to make that happen under windows. ss1 wrote: Hi Everyone, Please help freeswitch experts... !!! i have been working on freeswitch from last 2 days. i have downloaded freeswitch and unimrcp (server + client) for windows. I tested the unimrcp client and server, which is running fine with the command: run synth and run recog. I got both synth.pcm recog.pcm files. But my objective is to call Freeswitch through x-lite, where freeswitch should call unimrcp client and return the PCM files. I tried it alot, but unable to do it. after lots of reading i found that i do not have mod_unimrcp. i do not know from where to download it and how to merge it into freeswitch. I would be very thankful if you may help. Thanks, ss -- View this message in context: http://n2.nabble.com/need-help-Problem-with-freeswitch-uniMRCP-tp4031590p4038514.html Sent from the freeswitch-users mailing list archive at Nabble.com. ___ 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 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 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 compile error under Windows: error LNK2019
On Nov 22, 2009, at 11:51 PM, 大泥人 qinglan_z...@hotmail.com wrote: ___ 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.spec patch
This was merged into trunk. On Nov 20, 2009, at 12:34 PM, Brian West wrote: Hope on IRC and talk to MikeJ in #freeswitch he can direct you better on what to do vs not do since he maintains the builds system in FreeSWITCH. /b On Nov 20, 2009, at 11:31 AM, Igor Neves wrote: Ok, But how should I proceed? Thanks, ___ 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] change event value
no. On Nov 19, 2009, at 1:36 AM, Eli Hayun wrote: Hi Is there is a way to intercept an event (for example : REGISTER) and change one of its parameters (for example: the extension number) and fire up the corrected event? I need it to set the speedial of the phone value to be **x but to make it register as x ___ 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] Media got stuck after attended transfer...
I think a better approach here is to use spandsp. We already have some groundwork done for this. If you are interested in contributing, please email consult...@freeswitch.org and we can discuss further. Mike On Nov 19, 2009, at 6:54 PM, Klaus Hochlehnert wrote: Hi, one of my customers is willing to contribute for t38 integration. The basic idea is to connect HylaFAX to FS: t38modem - FreeSWITCH - Media Gateway with t38 support All this without media proxy. Another idea might be to implement t38 origination/termination with a class 1 modem input/output for use with HylaFAX. Do you know how much money we need to collect for t38 support? How much time is needed for implementing this? Thanks, Klaus From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Michael Collins Sent: Friday, October 16, 2009 2:10 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Media got stuck after attended transfer... On Thu, Oct 15, 2009 at 11:54 AM, Tihomir Culjaga tculj...@gmail.com wrote: hi, any clue when can t38 be added? Eventually. :) Of course, if we could get more to add to the bounty it might grease the wheels of innovation. http://wiki.freeswitch.org/wiki/Bounty#spanDSP_.2B_t.38_.28origination.2C_termination.2C_.26_gateway.29_in_Freeswitch Of course, I was listening to my A.M radio the other day and they said that there was this new invention called the Internet that would let people send documents to each other electronically. Maybe you should look into that. Next thing you know they'll come up with telephones that people don't have to plug into the wall and can take with them in the car. ;) -MC ___ 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 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 do I know the destination profile name?
check out sofia_contact function. If you use this in combination with binding
profiles together so they are one table I think this should work right.
Mike
On Nov 18, 2009, at 12:36 AM, Eli Hayun wrote:
Brian West wrote:
Why do you need to know the destination profile like that? You get to
pick that on your own so you should already know that before hand.
/b
On Nov 17, 2009, at 3:03 AM, Eli Hayun wrote:
Hi
We have more then one profile. To make a call I have to enter : bridge
sofia/profile/num...@ip
The problem is when I use : ${use_profile} I am getting the caller
profile, and I need the destination profile.
How do I get this information?
Thanks for your answer.
The problem is when I call to that number that the phone hook to other
server, I cannot make the call.
Is there is a variable that can tell me the destination profile?
Lets say the other profile called ph1 I have to dial
sofia/ph1/xx...@host to make the call. Is there other way to do that?
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Re: [Freeswitch-users] Compilation problem
This issue is now fixed in trunk. Mike On Nov 17, 2009, at 9:05 AM, Christopher Z. wrote: Hi, I've got this error after make: http://pastebin.freeswitch.org/11145 Any idea how to fix this error ? ___ 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 FS without spandsp or libtiff
Kristian, catch up with me somewhere that I can get remote access to this build environment so that we can sort this out. Mike On Nov 18, 2009, at 2:02 PM, Kristian Kielhofner wrote: On Wed, Nov 18, 2009 at 12:45 PM, Steve Underwood ste...@coppice.org wrote: Over time more and more of spandsp will be used by Freeswitch, so its most certainly an integral part of FS going forward. In a few months it might be possible to not use libTIFF, depending how things go with some developments. spandsp builds OK for many cross compile setups. make_at_dictionary and make_modem_filters should be built using the host compiler, not the target compiler. This seems to work in the places I've tried it. The problem in your pastebin log seems to be a broken C99 environment, and not a spandsp problem. Steve make_at_dictionary is not build using the host compiler. I had to hack it (manually passing CC and LIBTOOL to make) to get the build to proceed to the next error... This may be specific to the integration with the rest of the FreeSWITCH build system. I'm using uClibc (as are most other embedded environments) and I've had other C99 issues before. -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com ___ 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 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] Anybody interested in helping fix the -srcdir option?
Fixed in svn r15526 and other fixes in svn r15527. mike On Nov 18, 2009, at 11:40 AM, Robert Hadley wrote: Hi All, Anybody interested in helping fix the –srcdir option? I am trying to build in a subdirectory off the Freeswitch source. I am working on it and finding issues. However, being a newbie at autoconf/automake and shell scripting I sometimes struggle at finding fixes. For example, the script command below is in bootstrap.sh, but might need to be moved or duplicated in configure.* to support using configure –srcdir option, as the modules.conf file also needs to be to the build destination folder. if [ ! -f modules.conf ]; then cp build/modules.conf.in modules.conf fi Thanks, Robert ___ 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 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 dptools record problem - hangup channel with invalid file path
Okay, I'll ask the obvious question. Why are you passing record invalid file paths and why should it not fail if you do? Mike On Nov 18, 2009, at 2:38 PM, William Kendi ... wrote: While I was testing the mod dptools record application using invalid file paths, i noted that the mod dptools record application terminated the call. I am currently looking for a way to change this behaviour. Any suggestions? Can this be done? ___ 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 mod_SQL
http://wiki.freeswitch.org/wiki/Mod_xml_curl On Nov 15, 2009, at 11:39 AM, Samuel Mukoti wrote: Hi, I'm a newbie to FS, and I wanted to implement a setup where I provision the sip endpoints though a SQL database like mysql and also manage call routing too? Is this possible since I understand FS uses XML config files. ___ 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 Siemens A580 IP Phones
It doesn't look like your call ever gets setup in this trace, if you enable the sip trace you might see a bit more, but it looks like we are receiving a 480 response from the called phone. Mike On Nov 15, 2009, at 12:42 PM, vedama...@netscape.net wrote: I am FS beginner and I have a basic PBX setup using FS with the Siemens A580 IP Phones. I thought everything was working fine since I could make and receive basic calls without any obvious issues. However, recently I wanted to use more advanced functions in FS and discovered that I could not use any of DTMF based functions (e.g. call transfer/record) during calls with the Siemens IP phones. The same functions work fine when I use a softphone. So, I started looking at the log file and I think there is some problem between the Siemens IP phones and FS (log file attached below). It seems that when a call comes in, FS calls the extensions and then the extensions send back confirmation and SIP status codes. With softphone extensions, I see 180 (Ringing) and 200 (OK) as normal status. However, with Siemens IP phone extensions, I see 480 (Temporarily Unavailable) which seems to cause FS to terminate the session. So, FS log shows there is actually no active session which explains why it does not performs DTMF detection for the call session. However, the call to Siemens IP phones actually continues with ringing when an extension handset answers the call is established with the caller with full voice communication. I don't know how FS works but this seems very strange. I would like to know how to get FS to work properly with Siemens IP phones including the DTMF functions during calls. Any help would be appreciated. ___ 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 configure error using --srcdir option
Patches to make this work would be gladly accepted. Mike On Nov 13, 2009, at 7:56 PM, Brian West wrote: Don't use --srcdir we don't fully support that and the howto guides do not mention it AT ALL. So doing things that are not in the howto aren't really tested nor supported. /b On Nov 13, 2009, at 11:26 AM, Robert Hadley wrote: Hello All On CentOS 5.3, I am trying to build Freeswitch in a different directory and use the –srcdir= option. One reason I want to do this to have Debug and Release build targets from the same source. It doesn’t work, the configure errors when it gets to the first library subdirectory lib/srtp and tries to configure in there. The steps I am doing are: Building as root Unzip freeswitch-1.0.4-tar.gz in /opt cd into /opt/freeswitch-1.0.4 mkdir Debug cd Debug ../configure –srcdir=”..” CFLAGS=”-g –ggdb –O2” After several seconds of configuring I get: === configuring in libs/srtp (/opt/freeswitch-1.0.4/Debug/libs/srtp) configure: running /bin/sh ../../../libs/srtp/configure.gnu --disable-option-checking '--prefix=/usr/local/freeswitch' 'CFLAGS=-g -ggdb -O2' --cache-file=/dev/null --srcdir=../../../libs/srtp ../../../libs/srtp/configure.gnu: line 2: ./configure: No such file or directory configure: error: ../../../libs/srtp/configure.gnu failed for libs/srtp [r...@roberth-c53 Debug]# The file that’s executing is this: [r...@roberth-c53 srtp]# cd libs/srtp; cat ../../../libs/srtp/configure.gnu #! /bin/sh ./configure $@ --disable-shared --with-pic Please tell me if I understood the –srcdir option correctly and if there is a way to do build in a different directory. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] hangup incoming call by Reason: Q.850; cause=1; text=Unallocated (unassigned) number
Take a look at the freeswitch debug log, it should tell you exactly why it hung up. Mike On Nov 12, 2009, at 10:01 AM, Lei Tang wrote: Hi, I'm running a ivr script on FS, the call is from a softswitch to extenal sip endpoint of FS. I added two dialplan in public dialplan xml file. as flow: extension name=ivr_demo2 condition field=destination_number expression=^8$ action application=lua data=../ivr/test.lua/ /condition /extension extension name=ivr_demo2 condition field=destination_number expression=^\*114$ action application=lua data=../ivr/test.lua/ /condition /extension Every thing is ok when call to number 8. but when I call the second number *114, fs hangup after accept and answer the call, I captured the sip packets and found FS sent a bye packet after answer the call. the cause is Reason: Q.850;cause=1;text=Unallocated (unassigned) number. But as the fs console log show, the call is answered and the correct ivr script is runned. Why FS hangup the call? Does somebody have any idea about this problem? sip packets=== invite msg from softswitch INVITE sip:*...@10.37.143.6:5060;user=phone SIP/2.0 Contact: sip:xx...@10.4.35.17:5061 Content-Type: application/sdp To: sip:*...@10.37.143.6:5060;user=phone From: xsip:xx...@10.4.35.17:5061;user=phone;tag=949132463135364198E42500 P-Asserted-Identity: sip:xx...@10.4.35.17:5061;user=phone Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,COMET,UPDATE,PRACK,REFER,SUBSCRIBE,NOTIFY,MESSAGE Supported: 100rel,timer,replaces,diversion Expires: 155 Session-Expires: 1800 Min-SE: 90 Call-ID: 01fd10d1bd8140010...@sip-3 Max-Forwards: 70 CSeq: 1 INVITE Timestamp: 58520 Via: SIP/2.0/UDP 10.4.35.17:5061;branch=z9hG4bK5C0F524645A70C943998751419749696 Content-Length: 150 v=0 o=- 54000602557 1258015146 IN IP4 10.4.35.59 s=SDP Data c=IN IP4 10.4.35.59 t=0 0 m=audio 3 RTP/AVP 8 a=rtpmap:8 PCMA/8000 a=ptime:20 **FS ack SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.4.35.17:5061;branch=z9hG4bK5C0F524645A70C943998751419749696 From: x sip:xx...@10.4.35.17:5061;user=phone;tag=949132463135364198E42500 To: sip:*...@10.37.143.6:5060;user=phone Call-ID: 01fd10d1bd8140010...@sip-3 CSeq: 1 INVITE Timestamp: 58520 0.00 User-Agent: FreeSWITCH-mod_sofia/1.0.4-14460 Content-Length: 0 *FS answer the call (in lua script, I called session:answer() ) SIP/2.0 200 OK Via: SIP/2.0/UDP 10.4.35.17:5061;branch=z9hG4bK5C0F524645A70C943998751419749696 From: x sip:xx...@10.4.35.17:5061;user=phone;tag=949132463135364198E42500 To: sip:*...@10.37.143.6:5060;user=phone;tag=UjZcZUKZXjHcQ Call-ID: 01fd10d1bd8140010...@sip-3 CSeq: 1 INVITE Contact: sip:*...@10.37.143.6:5060;transport=udp User-Agent: FreeSWITCH-mod_sofia/1.0.4-14460 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO Require: timer Supported: timer, precondition, path, replaces Allow-Events: talk, refer Session-Expires: 1800;refresher=uac Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 245 v=0 o=FreeSWITCH 1257988835 1257988836 IN IP4 10.37.143.6 s=FreeSWITCH c=IN IP4 10.37.143.6 t=0 0 m=audio 24890 RTP/AVP 8 120 a=rtpmap:8 PCMA/8000 a=rtpmap:120 telephone-event/8000 a=fmtp:120 0-16 a=silenceSupp:off - - - - a=ptime:20 ACK sip:*...@10.37.143.6:5060;transport=udp SIP/2.0 CSeq: 1 ACK To: sip:*...@10.37.143.6:5060;user=phone;tag=UjZcZUKZXjHcQ From: xsip:xx...@10.4.35.17:5061;user=phone;tag=949132463135364198E42500 Call-ID: 01fd10d1bd8140010...@sip-3 Max-Forwards: 70 Timestamp: 58520 Via: SIP/2.0/UDP 10.4.35.17:5061;branch=z9hG4bK0CC4AE6EE59CA15F69429CDB97848C21 Content-Length: 0 ***FS hangup the call BYE sip:*...@10.37.143.6:5060;transport=udp SIP/2.0 Reason: Q.850;cause=1;text=Unallocated (unassigned) number To: sip:*...@10.37.143.6:5060;user=phone;tag=UjZcZUKZXjHcQ From: xsip:xx...@10.4.35.17:5061;user=phone;tag=949132463135364198E42500 Call-ID: 01fd10d1bd8140010...@sip-3 Max-Forwards: 70 CSeq: 2 BYE Timestamp: 58521 Via: SIP/2.0/UDP 10.4.35.17:5061;branch=z9hG4bKBE2D7D86B44CA171A5D374ECAA99A1DB Content-Length: 0 ___ 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] Request: Notify sip messages from Freeswitch to UserAgent
I have asked you before to please not cross post to both mailing lists. Please refrain from this in the future. Mike On Nov 9, 2009, at 6:36 PM, srinivasula reddy wrote: Hi, From Freeswitch there is continuously Request: Notify (Messages- waiting) requests are comming, i didnt subscribe from Freeswith and pjsip(ua). any body know how to stop those requests from Freeswitch. Thanks-- Srinivasula Reddy K ___ 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 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] PortAudio needs work on Mac OS X 10.6
If you can figure out a clean way for us to do this with proper ifdefs in tree in a way that will not break others that would be the most preferred. Mike On Nov 8, 2009, at 1:03 PM, Bruce Fletcher wrote: OK, I'll ignore that MacPorts patch for now and try to find a better approach. I'll look into this further tonight, but this morning I found a more recent promising patch on the PortAudio site: http://www.portaudio.com/trac/changeset/1418 It seems to push some data types to 32 bit regardless of platform, which might work better than the MacPorts approach of migrating some data structures to 64 bit. At any rate, this patch being on the PortAudio site suggests it might be a more approved fix. I'll keep plugging at this in my free time and report any significant progress back to the list. Thanks, - Bruce ___ 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] Extension: No audio
You don't have ext-rtp-ip set in your config. Mike On Nov 8, 2009, at 6:59 AM, Mark Campbell-Smith wrote: Hi! I have FS natted and am connecting with an 'external' extension that is registered to FS. ie the extension 2000 is registered on the internet with a public IP through my router to FS (192.168.1.120 IP address). uPnP works and I see that the extension is registered successfully. The problem is that I do not get any audio When looking at the SIP trace, I see the INVITE but do not see a TRYING or RINGING message. The extension is actually ringing. I modified the RTP port range on the remote end to match the RTP ports of freeswitch. I have put a sip trace in the pastebin at http://pastebin.freeswitch.org/11035 If anyone has an idea what needs to be set to get audio, help appreciated. Thanks! ___ 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] Extension: No audio
Your packet traces would disagree with the statements below. It is sending your internal address in rtp, so its not set correctly on whatever profile your using to call out, MIke On Nov 8, 2009, at 4:59 PM, Mark Campbell-Smith wrote: Hi Mike, I should have put that in also. I do have external_rtp_ip set in my config. I have it set to my domain name: X-PRE-PROCESS cmd=set data=external_rtp_ip=host:mydomainname/ I should also mention that if I use flaphone.com (which registers with an external IP address), then I get audio. In sofia, I see my IP addresses: = = = = = = = = = = = = = = = = = = = = = = = = = = = == Nameinternal Domain Name N/A DBName sofia_reg_internal Pres Hosts DialplanXML Context public Challenge Realm auto_from RTP-IP 192.168.1.120 Ext-RTP-IP 124.xxx.xxx.xxx SIP-IP 192.168.1.120 Ext-SIP-IP 124.xxx.xxx.x URL sip:mod_so...@192.168.1.120:5060 BIND-URLsip:mod_so...@192.168.1.120:5060 HOLD-MUSIC silence OUTBOUND-PROXY N/A CODECS G726-32,G722,PCMU,PCMA TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEGfalse PROXY-MEDIA false AGGRESSIVENAT true STUN-ENABLEDtrue STUN-AUTO-DISABLE false On Mon, Nov 9, 2009 at 7:28 AM, Michael Jerris m...@jerris.com wrote: You don't have ext-rtp-ip set in your config. Mike On Nov 8, 2009, at 6:59 AM, Mark Campbell-Smith wrote: Hi! I have FS natted and am connecting with an 'external' extension that is registered to FS. ie the extension 2000 is registered on the internet with a public IP through my router to FS (192.168.1.120 IP address). uPnP works and I see that the extension is registered successfully. The problem is that I do not get any audio When looking at the SIP trace, I see the INVITE but do not see a TRYING or RINGING message. The extension is actually ringing. I modified the RTP port range on the remote end to match the RTP ports of freeswitch. I have put a sip trace in the pastebin at http://pastebin.freeswitch.org/11035 If anyone has an idea what needs to be set to get audio, help appreciated. Thanks! ___ 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 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 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] leg_delay_start
Those vars were not even available in 1.0.3. I can't recall if they were in 1.0.4 or if you will need to use the latest 1.0.5 pre-release. Mike On Nov 7, 2009, at 9:26 AM, Steven Brown wrote: Hi I've been trying to experiment with leg_delay_start when bridging to two mobiles via a gateway, however regardless of settings both legs are bridged immediately. I noticed a previous post on problems with leg_delay_start which seemed to go unanswered, just wondered if there is a known issue or if its something I'm doing wrong. Using FS 1.0.3 Dialplan extract as follows : action application=bridge data=sofia/gateway/SIPGATE_JF/ 07x1,[leg_delay_start=20]sofia/gateway/SIPGATE_JF/07x2/ Any pointers appreciated ___ 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] Events in mod_perl
You can use EventConsumer class for this, I am afraid its not very
documented, but I do recall either a sample or discussion on the
mailing list that you should be able to find.
Mike
On Nov 7, 2009, at 12:38 AM, lakshmanan ganapathy wrote:
Ya. I have done that event processing with ESL. But I wanted to
know, whether in mod_perl, we can get the events and process it or
not. I've seen function's like events_get etc.. But I don't know how
to use those things.
In mod_perl if I'm able to get the events, then it will be easier
for me.
Is it possible!!!
2009/11/6 João Mesquita jmesqu...@freeswitch.org
I don't know what you are trying to do exactly but I think that you
might need to you ESL instead.
Why don't you take a look at all the examples inside ${SVNROOT}/libs/
esl and see if that fits you? I have a hunch that it would.
JM
On Fri, Nov 6, 2009 at 4:29 AM, lakshmanan ganapathy lakindi...@gmail.com
wrote:
Hi all,
Is there any way to receive events while running a perl program
with the help of mod_perl??
I've seen some functions related to sending and receiving events in
the mod_perl wiki. But I don't know how to use that.
Any help!!!
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Re: [Freeswitch-users] mod_shout.so: undefined symbol: ogg_sync_wrote
looks like ogg devel packages are installed but ogg lib is not? On Nov 7, 2009, at 3:59 AM, Sean Ferguson wrote: FreeSWITCH seems to be unable to read MP3 files, citing that it's an unknown format. Looking through the log, I found this during startup: 2009-11-07 02:43:45.749328 [CRIT] switch_loadable_module.c:871 Error Loading module /usr/local/freeswitch/mod/mod_shout.so **/usr/local/freeswitch/mod/mod_shout.so: undefined symbol: ogg_sync_wrote** There don't seem to be any compile-time errors, yet I can't seem to eliminate this issue. Any help would be appreciated. ___ 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] Dialpan: try.. finally analogs
It cleans up after itself fine, but it is an indication of some issue in the code we need to address. if you can reproduce this in svn trunk, please file a bug on jira.freeswitch.org with details how to reproduce. mike On Nov 5, 2009, at 12:44 PM, Artem Shiyanov wrote: Hello! I have to deal with classic problem: Leaking stream handle in FS console. I also know the reason - firstly channel is sent to the extension with playback and later it is transfered to another extensions with execute_extension or, another trouble-case - channel is bridged to some addres. I do not ask (but I wish to) why FS doesn't close stream automatically when channel is gone. I ask whether it is possible to use some try.. finally construction in diaplan? If yes then I can simply stop playback in the finally block.. Any thoughs? ___ 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 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] Does OpenZap support CTR21?
This would be specific to the zaptel driver for that card, not openzap. mike On Nov 5, 2009, at 1:43 PM, Fred-145 wrote: Hello As an alternative to more expensive alternatives like OpenVox or Sangoma, I'd like to order an X100P clone from www.x100p.com for use in France. According to a PDF on the site, the reason this card gets bad reviews is that the Silicon labs Si3012/Si3035 DAA chip used in the original Digium X100P card and low cost X100P clone cards only supports FCC mode. However, the Si3014/Si3034 DAA chip used on the X100P SE supports global line standards. As for software, the Silicon labs Si3014/Si3034 DAA chip used in the X100P SE supports 600 Ohm impedance and complex impedance to meet CTR21 line standards. However, the Zaptel wcfxo driver only supports CTR21 mode with 600 Ohm AC termination, which may or may not be the correct setting depending on the country and the phone system in use. So... does someone know if OpenZap, which is apparently required in addition to Zaptel/Dahdi for FreeSwitch to work PCI TDM cards, supports CTR21? ___ 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/2.0 503 Maximum Calls In Progress
Call loop? On Nov 4, 2009, at 10:25 AM, Diego Viola wrote: Hello, I tried to help Roy with this issue yesterday, I saw that calls couldn't go through and then I made a sofia profile internal siptrace on. Then I found a message like SIP/2.0 503 Maximum Calls In Progress and saw he had like 800 sessions. I thought it was an ACL issue but it wasn't, it seems like he reached a session limit, when I restarted his FS the problem went away. Best Regards, Diego ___ 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] Gateway Error
It means you need to go change the setting from the broken defaults, thats all. Mike On Nov 4, 2009, at 1:48 PM, Dave Stevenson wrote: Michael et al - and specifically, the FS Developers, this is all the more annoying given the fact that the SPA-3102 was bought specifically to run with FreeSwitch following a recommendation here in the UK. It was just unwrapped this afternoon :-( (http://robsmart.co.uk/2009/06/02/freeswitch_linksys3102/). I am setting up a VOIP system at home, and this device sounded like the ideal gateway to the PSTN. What does the error message actually mean - is this device a non- starter or are there work-arounds or fixes to the code in progress ? Surely the device can't be as broken as the message - or am I just being too hopeful ? Regards Dave - Original Message - From: Michael Collins To: freeswitch-users@lists.freeswitch.org Sent: Wednesday, November 04, 2009 6:23 PM Subject: Re: [Freeswitch-users] Gateway Error On Wed, Nov 4, 2009 at 10:03 AM, Dave Stevenson steve...@primrosebank.net wrote: Hi, I am trying to set up FreeSwitch with a new Linksys SPA-3102 Voice Gateway and am seeing the following error :- [WARNING] mod_sofia.c:810 We were told to use ptime 30 but what they meant to say was 20 This issue has so far been identified to happen on the following broken platforms/devices: Linksys/Sigura aka Cisco ShoreTel Sonus/L3 We will try to fix it but some of the devices on this list are so broken who know what will happen.. Having just bought the Gateway specifically for FS, that was a bit of a rude awakening ! Does anyone know of a fix in the pipeline, or am I sc***ed already ? The cynical among us will say that you were hosed the moment you paid for a Linksys device. :) It's very sad but the FS devs find this kind of thing all the time. They've literally got all sorts of checks in the code to make sure that devices aren't saying one thing and doing something else. Cisco is not the only one to do stupid things like this. In any case, just be aware of it. If you want suggestions then list to the others here who can offer their experiences with various devices they have in production. -MC ___ 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 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 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] Get error 415 Unsupported Media Type whenreceiving call from softswitch
That is correct. Mike On Nov 2, 2009, at 4:24 AM, Lei Tang wrote: Hi all, The problem is solved. I ask the softswitch to send only sdp in INVITE message, then It works. I think sofia doesn't support multipart content currently. is it 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] no REINVITE on Blind Transfer with bypass_media
Please re-try with latest svn trunk. Mike On Nov 2, 2009, at 9:36 AM, Humberto Quintana wrote: Thanks for you answers guys, I test the parameters you suggested but still no audio due to the lack of reINVITE. By the way I'm using 1.0.4 but I also tried 1.0.5pre3. One particular condition is that there is no on-hold before the Blind Transfer. Regards, Humberto param name=media-option value=resume-media-on-hold/ param name=media-option value=bypass-media-after-att-xfer/ ___ 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 Overlap support?
This may be possible with tcp, how could this work on udp? Can you provide an rfc reference on this? Mike On Oct 24, 2009, at 8:13 AM, Dennis wrote: ok, as written, i come back after some tests with fs and a thomson cirpack. it did not work - at least in our tests. we are using socket outbound and when a call comes in, it starts the socket of fs. the number may be 123456. fs sends the respond 484 and our carrier receives this information. but fs ends the call with hangup_cause = invalid_number_format. the carrier has one more digit for the phone number and sends 1234567 and the above mentioned behavior repeats. the behavior we want and expected is, that the call stays in the socket after response 484, so that the carrier can send the 1234567 into the same socket. the management, when fs should send response 484 and when fs should be answered would be programmed by us. it also important, that fs keeps the call in the socket, so we can tell fs, to answer the call after x seconds anyway. any ideas, what we could do? ___ 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-users Digest, Vol 40, Issue 179
You actually can use these in conditions. Just need to be careful
that the var you are conditioning on is already set.
Mike
On Oct 22, 2009, at 1:54 PM, Michael Collins wrote:
On Thu, Oct 22, 2009 at 5:51 AM, Rupa Schomaker r...@rupa.com wrote:
cond would be helpful here? I updated the wiki on this one just now
with a bit more detail. It is a api call. so, you'd use it like:
${cond(eval ? trueval : falseval)}
so to get a value of ERR if the var my myvar is 15 you could:
${cond(${myvar} 15 ? ERR : OK)}
If both sides of the comparison operator are numeric then it does
numeric comparison otherwise it does lexical string comparison.
Rupa,
Yes, you can do the set/cond API trick but you can only do it in the
action or anti-action tags, not in the condition tags. I'm sure you
know that but I want all those reading this thread to make the
connection.
-MC
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Re: [Freeswitch-users] ANNOUNCEMENT: FreeSWITCH 1.0.5pre3 is available for testing!
We still do plan on branching 1.0 into bugfix only. This has not yet happened but may happen at some point after 1.0.5. In the mean time, the vast majority of the work lately has been fixes with small feature improvements, most all of this would stay in a 1.0 branch even if we were already branched. Trunk remains for the most part very stable with new features mostly coming in new modules. Mike On Oct 28, 2009, at 8:40 PM, Craig Askings 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? 2009/10/29 Michael Collins m...@freeswitch.org Hello FreeSWITCHers, The latest FreeSWITCH version is now available for download on the files site. The announcement story is in the main FreeSWITCH site. Please download and test, and then test some more. We need your feedback. The sooner we get your feedback, the more quickly we can roll the official 1.0.5. -- Craig Askings Network Engineer | Over the Wire Pty Ltd cr...@overthewire.com.au | www.overthewire.com.au Phone: 07 3847 9292 | Fax: 07 3847 9696 | Mobile: 0404 019 365 ___ 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 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] Setting up Conference with Moderator
Have you answered the call? On Oct 30, 2009, at 11:34 AM, Rob Forman wrote: Hm, strange. I haven't seen that before. Can you pastebin your logs at debug level? On Oct 30, 2009, at 9:43 AM, Ujjval Karihaloo wrote: It's strange... a tcpdump tells me that there is no DTMF from my provider when using IVR, but when I call into a TN that goes directly into the Conference App, I see DTMF from the provider. ___ 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_t38gateway
This is a non working module, just a shell for development. Mike On Oct 30, 2009, at 5:52 PM, Tihomir Culjaga wrote: does anybody know how does it work and how to use it in a dialplan? ___ 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-users Digest, Vol 40, Issue 179
see rupa's explanation below.
On Nov 1, 2009, at 1:24 AM, Michael Collins wrote:
How would you do an expression like: if $x 24 in a condition tag?
Just curious. I would like to make sure that is properly documented.
-MC
On Thu, Oct 22, 2009 at 5:51 AM, Rupa Schomaker r...@rupa.com
wrote:
${cond(${myvar} 15 ? ERR : OK)}
If both sides of the comparison operator are numeric then it does
numeric comparison otherwise it does lexical string comparison.
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Re: [Freeswitch-users] error in loading spidermonkey
The libcurl is broken on your distro. You can fix this by configuring with --without-libcurl which will use our working in tree copy instead of the broken one from your distro. Mike On Oct 28, 2009, at 1:39 PM, Erwin Davis wrote: Hi, I got an error in loading mod_spidermonkey. my fs in ver 1.04 runs on fedora 8 VM. Any clue? thanks, [CRIT] switch_loadable_module.c:871 Error Loading module /usr/local/ freeswitch/mod/mod_spidermonkey.so **/usr/lib/libnss3.so: undefined symbol: PR_UnloadLibrary** ___ 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 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_xml_open_root
You probably should not be calling that function, what are you trying to do? Mike On Oct 27, 2009, at 8:48 AM, srinivasula reddy wrote: Hi, when i am calling switch_xml_open_root(1,err) . i am getting this warning message. HEAP[FreeSwitch.exe]: Invalid Address specified to RtlFreeHeap ( 0016, 100E9FD0 ). can any know. please help me. ___ 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_xml_open_root
just api execute reloadxml Mike On Oct 28, 2009, at 12:37 AM, srinivasula reddy wrote: Hi mike, thank for your reply. i am trying to call that function from swig.cs. its working fine first time with the warning information, then onwards it is not working. is there other way can i call reload function from freeswitch.managed code. any 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] Downloaded tar vs latest SVN - 200 OK has more headers
New sofia profile param as follows: !-- set this param to false if your gateway for some reason hates X- headers that is is supposed to ignore-- !--param name=pass-callee-id value=false/-- On Oct 26, 2009, at 12:16 PM, Anthony Minessale wrote: Thus perpetuating the wild-west of sip where you can't do anything according to spec because you have to worry about stupid things not keeping up. Sounds like the education system where I live too. I'll see what I can do. It's always the other end that ppl pay for that drive the free stuff to change its code. On Mon, Oct 26, 2009 at 11:05 AM, Tihomir Culjaga tculj...@gmail.com wrote: On Mon, Oct 26, 2009 at 4:41 PM, Anthony Minessale anthony.miness...@gmail.com wrote: The headers are used to pass the callee-id info back to the other side so you have the id of who you called. The standards have failed us in this case as everything does it differently to the point that there is no standard thus we have invented our own way to carry this across from one FreeSWITCH box to another, but of course we can never make anybody happy. =/ I agree with you, X headers should be ignored by the equipment normally. Anyhow Kristian has a point here; there will be a lot of complains because of broken SIP stack on many vendor equipments So, can you consider some customizable a config option for such headers? T. ___ 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org pstn:213-799-1400 ___ 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 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] Core Dump question!
I did test this on trunk and it seems to work right: freeswi...@default sofia_gateway_data -ERR Parameter missing Mike On Oct 22, 2009, at 3:58 PM, Michael Collins wrote: What SVN rev of FS? What operating system? If you're not on the latest then do a make current and get to the latest SVN and see if you can replicate the issue. -MC On Thu, Oct 22, 2009 at 12:45 PM, Ujjval Karihaloo ujj...@simplesignal.com wrote: freeswi...@ss_freeswitch sofia_gateway_data Segmentation fault (core dumped) Just ran the gateway command above w/o any parameters,,, and it core dumped.. I am sure mistakes like that happen…but I not sure if it should core dump and shutdown….. ___ 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 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
This appears to be some sort of ice implementation? We don't support sip ice at this time. Mike On Oct 21, 2009, at 7:58 AM, ineya ineya wrote: Codecs are fine. I spent much time experimenting with codecs and completely missed, that freeswitch is modifiyng the SDP record. When phone A is making a call the SDP contains candidate media attributes: a=candidate:123abc 1 UDP 9 2000:2::1001 5012 typ host a=candidate:123abc 1 UDP 8 10.80.62.92 5010 typ host But when freeswitch makes the INVITE on phone B, these 2 are missing and phone is looking for it, so the INVITE gets rejected by phone with 448 Not acceptable here So the question is, how can I make the freeswitch to pass these candidate media attributes? On Wed, Oct 21, 2009 at 8:58 AM, Jason White ja...@jasonjgw.net wrote: 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 ___ 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] sched_api doesn't get launched
The syntax is different, but the api is the same as lua: So you need an API object in order to use it. I don't know the syntax for creating an api obj in Java but in Lua it goes like this: api = freeswitch.API(); res = api:execute(sched_api,+300 none my_api my_api_args) create the API object and use the execute method of it. Mike On Oct 21, 2009, at 5:44 AM, Henry Huang wrote: I can't seem to find the right thing to use in mod_java to execute api commands, only api_after_bridge 2009-10-21 17:42:46.593094 [NOTICE] mod_sofia.c:1509 Pre-Answer sofia/internal/1688...@192.168.1.66! # # A fatal error has been detected by the Java Runtime Environment: # # SIGSEGV (0xb) at pc=0x004e4480, pid=1927, tid=16116624 # # JRE version: 6.0_16-b01 # Java VM: Java HotSpot(TM) Client VM (14.2-b01 mixed mode linux-x86 ) # Problematic frame: # C [libc.so.6+0x6f480] strcpy+0x10 # # An error report file with more information is saved as: # /usr/local/freeswitch/bin/hs_err_pid1927.log 2009-10-21 17:42:59.883729 [ERR] switch_core_session.c:1374 Invalid Application sched_api 2009-10-21 17:42:59.883729 [NOTICE] switch_core_session.c:1375 Hangup sofia/internal/1688...@192.168.1.66 [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER] # # If you would like to submit a bug report, please visit: # http://java.sun.com/webapps/bugreport/crash.jsp # The crash happened outside the Java Virtual Machine in native code. # See problematic frame for where to report the bug. On Sat, Oct 17, 2009 at 3:37 AM, Michael Collins m...@freeswitch.org wrote: On Fri, Oct 16, 2009 at 11:53 AM, Henry Huang red.rain.se...@gmail.com wrote: So how would you trigger it from a script dialplan? The only time it seemed to work is when I did setVariable(api_after_bridge, sched_api blah blah blah); but then it gets executed after the channel's been teared down. I thought api_after_bridge means right after the call gets connected. I need something to execute an api command right before or right after the call gets bridged. api_after_bridge is a channel variable, so using setVariable works just fine. If you need to sched_api is an API only. Check these out: http://wiki.freeswitch.org/wiki/Mod_commands#Misc._Commands So you need an API object in order to use it. I don't know the syntax for creating an api obj in Java but in Lua it goes like this: api = freeswitch.API(); res = api:execute(sched_api,+300 none my_api my_api_args) Remember, if the method you are using isn't found in the dial plan tools then it isn't a dial plan application. Make sure it's on the list: http://wiki.freeswitch.org/wiki/Mod_dptools On the other hand, API commands are listed here: http://wiki.freeswitch.org/wiki/Mod_commands dptools require a session object, api commands require an api object... -MC ___ 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 -- Henry Huang UniC Solution - Communication Unified VoIP Open Source software Consultant ___ 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 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] Calls dropping on SIP timer expiry due to UPDATE's being ignored.
This should now be fixed in latest svn trunk. Mike On Oct 21, 2009, at 12:45 PM, Keith Laaks wrote: Hi, Hope someone knows how I am able to get around this one. Here goes... Did an upgrade to trunk (from a July vintage build) last week and noticed calls out to a provider were now failing after about 30 seconds or so - post answer. Tried latest (15183) - same thing. Analysing, I see that I have multiple UPDATE messages now being sent to the provider, but no response being sent back to FS. So FS times out and eventually kills the call. Interestingly, it only drops the A-leg; the B-leg remains up till the B party hangs up. I cant recall seeing these UPDATE messages before... The intent of the UPDATE seems to be to send the callee name number to the B-leg. If its the provider's sip stack that's broken w.r.t. handling UPDATE - is there any way to get around it by doing something in my config to ensure these UPDATE's are not 'triggered' ? ___ 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] CS_REPORTING Channel event state
REPORTING is the state that it writes to CDR. If you have calls stuck in this state, take one and try to use uuid_kill on it and see if it goes away, then get a core off of it and pastebin the thread apply all bt (with no other calls up). What modules are you using for cdr and with what configuration? Mike On Oct 20, 2009, at 11:47 AM, Dome Charoenyost wrote: Dear All What's CS_REPORTING state ? I found many channels not hang up ans state is CS_REPORTING e264f84a-bd87-11de-9a90-2320c02172de,outbound,2009-10-20 21:50:26,1256050226,sofia/external/x...@xxx.xxx.xx. 191:7050,CS_REPORTING,FreeSWITCH,,xx.xxx.xxx.xxx, 7050,,,XML,public, ___ 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] Connect PHP SOAP Web Server with SQLite database of FS
If you really want to access this information outside I would strongly recommend using odbc instead of the internal sqlite db, it does not handle locking contention well. If you need access to things in the core db (like show calls and show channels information) you will need to write a small daemon that listens on events socket and puts that information into a database. Mike On Oct 20, 2009, at 2:23 PM, Chris Burns wrote: If you really wanted: http://php.net/manual/en/book.sqlite.php But I would recommend you make use of ODBC to use a client/server RDBMS. Here's some good reading: http://www.sqlite.org/cvstrac/wiki?p=WhenToUseSqlite On October 20, 2009 10:53:01 am homqua wrote: Now I am building a PHP SOAP Web Service to access the database of FS. Anyone has idea about how to access sqlite database of FS through PHP ? I have read about socket event in FS, but I don't know whether it can response with the query of database or not. Thanks for your 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] NAT problems migrating from Version 1.0.trunk (13168M) to Version 1.0.trunk (15166)
we added some params for new better automatic nat handling, grep the
new defailt configs for localnet and you will find what you are missing.
Mike
On Oct 18, 2009, at 11:14 PM, Chris Fowler wrote:
I've tried all sorts of debug and parameter changes over the weekend,
but still can't figure out the correct solution.
If I disable timers on the sip profile then all works fine.
param name=enable-timer value=false/
But that seems like a hack; not a correct solution. With the build
13168M (which is pre the new NAT functionality) everything worked
fine.
The SIP trace shows the phones and FreeSWITCH happily exchanging
NOTIFY
and 200 OK messages. Audio's working - just calls timeout after 100
seconds with RECOVERY_ON_TIMER_EXPIRE. Is enforcement of this timer
new
functionality - and really just exposing a problem I've always had
before?
The config is (50 Polycom Phones - NAT - Internet - Amazon EC2)
I would really appreciate some pointers on what to look for;
additional
trace that might reveal something.
Thanks, Chris.
On Fri, 16 Oct 2009 20:06:52 -0700, Chris Fowler ch...@fowler.cc
said:
Hi,
We've been using 13168M in production for some time now (works
great).
I want to get us onto the latest build but am having problems getting
NAT to work.
Phones can register; can dial test #, but after 100 seconds the
call is disconnected with error:
2009-10-16 19:52:26.936618 [NOTICE] sofia.c:4038 Hangup
sofia/internal/1...@myhost.mydomain.com [CS_EXECUTE]
[RECOVERY_ON_TIMER_EXPIRE]
I took the standard internal.xml and vars.xml files from the new
build
and made the following modifications - which worked previously:
modify conf/vars.xml and update
X-PRE-PROCESS cmd=set data=domain=myhost.mydomain.com/
X-PRE-PROCESS cmd=set data=bind_server_ip=1.2.3.4/
X-PRE-PROCESS cmd=set data=external_rtp_ip=1.2.3.4/
X-PRE-PROCESS cmd=set data=external_sip_ip=1.2.3.4/
Modify conf/sip_profiles/internal.xml
param name=aggressive-nat-detection value=true/ param
name=ext-rtp-ip value=$${external_rtp_ip}/ param
name=ext-sip-ip value=$${external_sip_ip}/ param
name=NDLB-received-in-nat-reg-contact value=true/ param
name=NDLB-force-rport value=true/ param
name=NDLB-broken-auth-hash value=true/
The big difference I note is that on PRODUCTION (which works) sofia
status profile internal yields:
URL sip:mod_so...@1.2.3.4:5060
BIND-URL
sip:mod_so...@1.2.3.4:5060;maddr=10.250.35.224
But on Test I see:
URL sip:mod_so...@10.250.66.210:5060
BIND-URLsip:mod_so...@10.250.66.210:5060
Any ideas?
Thanks, Chris.
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Re: [Freeswitch-users] uuid_send_dtmf fails (was: conference call)
There is an event you can send as well to switch them, it your trying to switch it via event socket, that should be better, its not on the wiki, but a session message with eavesdrop-command header with data as the same as dtmf should do the trick Mike On Oct 16, 2009, at 11:54 AM, Nikita Belov wrote: Yes, it is what I need. But now I have problem with sending dtmf. Here what I've done: [r...@centos4-4-vm ~]# telnet localhost 8021 Trying 127.0.0.1... Connected to localhost.localdomain (127.0.0.1). Escape character is '^]'. Content-Type: auth/request auth ClueCon Content-Type: command/reply Reply-Text: +OK accepted api originate user/1...@master.agent.starpoundtech.net park() Content-Type: api/response Content-Length: 41 +OK bba3b45a-4cc1-48af-a15d-1052d5f11371 SendMsg bba3b45a-4cc1-48af-a15d-1052d5f11371 call-command: execute execute-app-name: eavesdrop execute-app-arg: cd99f999-9b47-457e-8439-1d366e015b8c Content-Type: command/reply Reply-Text: +OK Here I had started to hear A and B. Here what I saw in FS log: 2009-10-18 03:22:47 [DEBUG] switch_core_session.c:706 switch_core_session_queue_private_event() Send signal sofia/internal/sip:1...@172.26.10.64:5060;fs_nat=yes [BREAK] 2009-10-18 03:22:47 [DEBUG] switch_ivr.c:540 switch_ivr_parse_event() sofia/internal/sip:1...@172.26.10.64:5060;fs_nat=yes Command Execute eavesdrop(cd99f999-9b47-457e-8439-1d366e015b8c) 2009-10-18 03:22:47 [DEBUG] switch_core_media_bug.c:297 switch_core_media_bug_add() Attaching BUG to sofia/internal/1...@master.agent.starpoundtech.net 2009-10-18 03:22:47 [DEBUG] switch_core_io.c:234 switch_core_session_read_frame() sofia/internal/1...@master.agent.starpoundtech.net receive message [TRANSCODING_NECESSARY] 2009-10-18 03:22:47 [DEBUG] switch_core_io.c:234 switch_core_session_read_frame() sofia/internal/sip:1...@172.26.10.64:5060;fs_nat=yes receive message [TRANSCODING_NECESSARY] Then I run command: api uuid_send_dtmf bba3b45a-4cc1-48af-a15d-1052d5f11371 1 Content-Type: api/response Content-Length: 14 -ERR no reply Log: 2009-10-18 03:24:01 [DEBUG] switch_core_io.c:1190 switch_core_session_send_dtmf_string() sofia/internal/sip:1...@172.26.10.64:5060;fs_nat=yes send dtmf digit=1 ms=250 samples=2000 2009-10-18 03:24:01 [DEBUG] switch_rtp.c:1282 do_2833() Send start packet for [1] ts=2241760 dur=160/160/2000 seq=21346 2009-10-18 03:24:01 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet for [1] ts=2241760 dur=320/320/2000 seq=21347 2009-10-18 03:24:01 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet for [1] ts=2241760 dur=480/480/2000 seq=21348 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet for [1] ts=2241760 dur=640/640/2000 seq=21349 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet for [1] ts=2241760 dur=800/800/2000 seq=21350 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet for [1] ts=2241760 dur=960/960/2000 seq=21351 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet for [1] ts=2241760 dur=1120/1120/2000 seq=21352 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet for [1] ts=2241760 dur=1280/1280/2000 seq=21353 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet for [1] ts=2241760 dur=1440/1440/2000 seq=21354 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet for [1] ts=2241760 dur=1600/1600/2000 seq=21355 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet for [1] ts=2241760 dur=1760/1760/2000 seq=21356 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet for [1] ts=2241760 dur=1920/1920/2000 seq=21357 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send end packet for [1] ts=2241760 dur=2080/2080/2000 seq=21358 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send end packet for [1] ts=2241760 dur=2080/2080/2000 seq=21359 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send end packet for [1] ts=2241760 dur=2080/2080/2000 seq=21360 But both A and B couldn't hear me. Btw, after I had send dtmf 1 manually from my phone. B started to hear me. There was this record in log: 2009-10-18 03:47:55 [DEBUG] switch_rtp.c:1767 switch_rtp_dequeue_dtmf () RTP RECV DTMF 1:2240 Does anybody know, what had I done wrong? ___ Thanks, Nikita -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch- users-boun...@lists.freeswitch.org] On Behalf Of Rupa Schomaker Sent: Thursday, October 15, 2009 4:04 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] conference call The wiki has a pretty good explanation of how eavesdrop works. Enabling a talk path to A or B or both A and B requires dtmf. So, if C hits the 1 button on the phone they can talk to the UUID you bound
Re: [Freeswitch-users] Freeswitch.managed
Try starting out reading this. http://wiki.freeswitch.org/wiki/Mod_managed Mike On Oct 19, 2009, at 9:14 AM, srinivasula reddy wrote: Hi, How can i use freeswitch.managed project. what are the parameters for calling Execute method? and how can i call? any 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] sofia gateways and linux multipath routing
You need a sofia profile for each identity, if your using multiple external ip addresses, you will need a profile for each. If you are using bgp or something of the sort and only using one external ip, you can use a single profile and route using standard routing. Mike On Oct 13, 2009, at 6:09 AM, François Delawarde wrote: Hello all, I'm interested in using mod_sofia with multiple Internet connections (configured as a unique load-balancing route using multipath). One solution would be to define a different profile for each connection, but it would be more practical having a unique external profile that would automatically handle everything (detecting multiple public IPs and selecting the right one for a call, being able to select the router for gateway registration...). Routing table: 192.168.10.0/24 dev eth0 proto kernel scope link src 192.168.10.1 192.168.1.0/24 dev eth1 proto kernel scope link src 192.168.1.2 192.168.2.0/24 dev eth2 proto kernel scope link src 192.168.2.2 default proto static nexthop via 192.168.1.1 dev eth1 weight 1 nexthop via 192.168.2.1 dev eth2 weight 1 Both default routers (192.168.1.1 and 192.168.2.1) would have a distinct public IP. Several questions cross my mind: - Can a unique sofia profile be bound to multiple IPs (not 0.0.0.0)? - How would FS behave with a unique external profile in that situation? * Would FS reply to an incoming call using the same router it came from forcing packet source address? * Would FS stick to a unique router for all flows of an outgoing call (SIP, RTP, UDPTL)? * Can I force a gateway to use a given router (for calls, registration, ...)? * Would the NAT system (using stun or auto-nat) work in that situation, or does it assume only one default router (and a unique public IP) exists per profile? - Knowing the above, would it be necessary to use a different profile for each router/interface, and define the same gateway in each of these? - Tricky question: What if multiple routers are on the same network/interface (192.168.1.1, 192.168.1.2, ...)? Thanks in advance, François. ___ 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 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] TLS client authentification
I can't recall if we ever exposed an option for this, take a look at sofia-sip and see if they have a tag to enable this, if so it would probably be a fairly simple patch to add. Mike On Oct 15, 2009, at 3:20 PM, Alexandre Savard wrote: Hi, Does Freeswitch support TLS Client-Authenticated handshake. Openssl does, but it has to be enabled in order to send the certificate request to the client. I tested simple TLS hanshaking and it wotks well. ___ 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] validating dtmf digits received
inline is new, it won't work unless your using recent trunk. That
being said, read is not being run inline, so the set is actually being
run before digits_dialed is set. You will most likely need to use
transfer in this situation.
Mike
On Oct 19, 2009, at 12:53 AM, Mark Campbell-Smith wrote:
Hi!
I simply want to validate the dtmf digits I read from a user.From
the wiki, it appears I need to use inline=true when setting the
variable so it can be used directly within the same extension.
What have I done wrong below? I have tried many different
alternatives, but the second condition field, which is meant to match
the dtmf digits received (in this case ) is never matched, and the
anti-action is called instead.
:
some code here
action application=read data=1 10
ivr/ivr-please_enter_pin_followed_by_pound.wav res 1 9/
action application=phrase data=spell,${res}/
action inline=true application=set data=code=$
{digits_dialed}/
!-- action inline=true application=set data=code=$
{res}/ --
/condition
condition field=digits_dialed expression=^$
!-- condition field=${code} expression=^$ --
!-- condition field=${res} expression=^$ --
some code here
anti-action application=hangup/
Thanks!
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Re: [Freeswitch-users] How to enforce freeswitch replying to the source port instead of to the one specified in v/m header parametes (Symmetric NAT)
This is what force-rport is supposed to do. That being said, I can't tell from your trace where it is actually going to, just what it says in the packet, which can be different. Mike On Oct 19, 2009, at 3:23 AM, Tzury Bar Yochay wrote: Hi, I am struggling with a cellular operator which removes the rport from the SIP messages sent by the client. During the troubleshooting process I have been playing with the NDLB parameters I found mentioned at the fs-wiki param name=NDLB-force-rport value=true/ param name=NDLB-received-in-nat-reg-contact value=true/ as well as with sip-force-contact at the extension xml config file. variables variable name=sip-force-contact value=NDLB-connectile- dysfunction/ /variables I Want to make freeswitch replying to the source port instead of the port supplied by the client, that is to port 12543 instead of to the 7608 as you can see below some pcap dumps of the case. I am not sure if that would solve thep roblem and am open to other suggestions as well. Thanks in advance, Tzury pcap dumps below: ### REGISTER ### User Datagram Protocol, Src Port: 12543 (12543), Dst Port: sip (5060) REGISTER sip:example.net SIP/2.0 v: SIP/2.0/UDP 212.154.128.222:7608;branch=z9hG4bKPj41fa0abd044c35c75598d561b2f93167 Max-Forwards: 70 f: sip:1...@example.net;tag=9194adecbf8e8179530eb589e83631db t: sip:1...@example.net i: ebf2e3c254517938f90f33d8fd89326d CSeq: 17355 REGISTER m: sip:1...@212.154.128.222:7608 Expires: 300 l: 0 ### 401 UNAUTHORIZED ### SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 212.154.128.222:7608;branch=z9hG4bKPj41fa0abd044c35c75598d561b2f93167 From: sip:1...@example.net;tag=9194adecbf8e8179530eb589e83631db To: sip:1...@example.net;tag=166HraN0B149g Call-ID: ebf2e3c254517938f90f33d8fd89326d CSeq: 17355 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces WWW-Authenticate: Digest realm=example.net, nonce=f2f1370c-bc7c-11de-b000-fbe81e221ab4, algorithm=MD5, qop=auth Content-Length: 0 ___ 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 1.0.4 problems with music on hold
Try out trunk and see if this issue is resolved please. mike On Oct 19, 2009, at 3:11 AM, Durk de Beer wrote: Hello, This is something I came across on Freeswitch 1.0.4 First let me explain what I'm trying to do. I want Free-Switch to behave as a proxy so in the settings section of Sofia.conf.xml I use param name=media-option value=bypass-media-after-att-xfer/ and param name=inbound-bypass-media value=true/ As fare as I am able to follow the RTP stream it is passing by Freeswitch making it a proxy. So far so good. Now I want to serv music on hold if a call is put on hold. The only way I am able to find for Freeswitch to serv music on hold (MOH) is for it to be in media. Ok we've got a command for this situation namely param name=media-option value=resume-media-on-hold/ Ok if I put the last parram command into the Sofia.conf.xml Freeswitch is starting to behave funny. Let me try to explain this by an example. Let suppose Alice is calling Bob without the last command, the “resume media on hold”. If Bob puts Alice on hold al is silent on Bob's and Alice's phone. If Bob takes Alice of hold there's two way audio. Ok no I put the command in sofia.onf.xml, the “resume media on hold”. Now Alice is calling Bob again and Bob is putting her on hold again. You are expecting MOH on this moment but it isn't, all there is are the sounds of silence. Ok mistake made checking local streams, and they are there?. Yes it is. Ok other mistake, didn't put the “hold-music”command in sofia.conf.xml. Noop it's in there so there must be MOH. Codecs then?. Noop there in vars.xml. Ok for me it is time to throw in the towel, I'll find it out later. Ok Bob is now taking Alice of hold and surprise surprise there is only one way audio, between Alice and Bob. I didn't have the time to look with Wireshark where al the steams go so sorry for that. Did anyone came across the same problem with Freeswitch 1.0.4. I've a 1.0.2 running and that is doing everything a ok. ___ 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] uuid_send_dtmf fails (was: conference call)
You will want to use sendevent with a unique-id header and a eavesdrop- command header. Also please note you will want to use svn revision 15175 or later, I just fixed a segfault in that code. Mike On Oct 19, 2009, at 11:11 AM, Nikita Belov wrote: Thanks, Mike, for idea. But what is the syntax for this session message? I tried this: sendmsg e8e4f0ed-a0cc-4dff-b7e1-09eeade5df05 eavesdrop-command: 1 but it doesn't work. -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch- users-boun...@lists.freeswitch.org] On Behalf Of Michael Jerris Sent: Monday, October 19, 2009 5:19 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] uuid_send_dtmf fails (was: conference call) There is an event you can send as well to switch them, it your trying to switch it via event socket, that should be better, its not on the wiki, but a session message with eavesdrop-command header with data as the same as dtmf should do the trick Mike On Oct 16, 2009, at 11:54 AM, Nikita Belov wrote: Yes, it is what I need. But now I have problem with sending dtmf. Here what I've done: [r...@centos4-4-vm ~]# telnet localhost 8021 Trying 127.0.0.1... Connected to localhost.localdomain (127.0.0.1). Escape character is '^]'. Content-Type: auth/request auth ClueCon Content-Type: command/reply Reply-Text: +OK accepted api originate user/1...@master.agent.starpoundtech.net park() Content-Type: api/response Content-Length: 41 +OK bba3b45a-4cc1-48af-a15d-1052d5f11371 SendMsg bba3b45a-4cc1-48af-a15d-1052d5f11371 call-command: execute execute-app-name: eavesdrop execute-app-arg: cd99f999-9b47-457e-8439-1d366e015b8c Content-Type: command/reply Reply-Text: +OK Here I had started to hear A and B. Here what I saw in FS log: 2009-10-18 03:22:47 [DEBUG] switch_core_session.c:706 switch_core_session_queue_private_event() Send signal sofia/internal/sip:1...@172.26.10.64:5060;fs_nat=yes [BREAK] 2009-10-18 03:22:47 [DEBUG] switch_ivr.c:540 switch_ivr_parse_event () sofia/internal/sip:1...@172.26.10.64:5060;fs_nat=yes Command Execute eavesdrop(cd99f999-9b47-457e-8439-1d366e015b8c) 2009-10-18 03:22:47 [DEBUG] switch_core_media_bug.c:297 switch_core_media_bug_add() Attaching BUG to sofia/internal/1...@master.agent.starpoundtech.net 2009-10-18 03:22:47 [DEBUG] switch_core_io.c:234 switch_core_session_read_frame() sofia/internal/1...@master.agent.starpoundtech.net receive message [TRANSCODING_NECESSARY] 2009-10-18 03:22:47 [DEBUG] switch_core_io.c:234 switch_core_session_read_frame() sofia/internal/sip:1...@172.26.10.64:5060;fs_nat=yes receive message [TRANSCODING_NECESSARY] Then I run command: api uuid_send_dtmf bba3b45a-4cc1-48af-a15d-1052d5f11371 1 Content-Type: api/response Content-Length: 14 -ERR no reply Log: 2009-10-18 03:24:01 [DEBUG] switch_core_io.c:1190 switch_core_session_send_dtmf_string() sofia/internal/sip:1...@172.26.10.64:5060;fs_nat=yes send dtmf digit=1 ms=250 samples=2000 2009-10-18 03:24:01 [DEBUG] switch_rtp.c:1282 do_2833() Send start packet for [1] ts=2241760 dur=160/160/2000 seq=21346 2009-10-18 03:24:01 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet for [1] ts=2241760 dur=320/320/2000 seq=21347 2009-10-18 03:24:01 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet for [1] ts=2241760 dur=480/480/2000 seq=21348 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet for [1] ts=2241760 dur=640/640/2000 seq=21349 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet for [1] ts=2241760 dur=800/800/2000 seq=21350 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet for [1] ts=2241760 dur=960/960/2000 seq=21351 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet for [1] ts=2241760 dur=1120/1120/2000 seq=21352 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet for [1] ts=2241760 dur=1280/1280/2000 seq=21353 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet for [1] ts=2241760 dur=1440/1440/2000 seq=21354 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet for [1] ts=2241760 dur=1600/1600/2000 seq=21355 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet for [1] ts=2241760 dur=1760/1760/2000 seq=21356 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send middle packet for [1] ts=2241760 dur=1920/1920/2000 seq=21357 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send end packet for [1] ts=2241760 dur=2080/2080/2000 seq=21358 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send end packet for [1] ts=2241760 dur=2080/2080/2000 seq=21359 2009-10-18 03:24:02 [DEBUG] switch_rtp.c:1221 do_2833() Send end packet for [1] ts=2241760 dur=2080/2080/2000 seq=21360 But both A and B couldn't hear me. Btw, after I had send dtmf 1 manually from my phone. B started to hear me. There was this record in log: 2009-10-18 03:47:55 [DEBUG] switch_rtp.c:1767
Re: [Freeswitch-users] Brazilians (Off-Topic)
we do have a license for this, people didn't seem to like it last time we looked at it, I can't recall why. On Oct 19, 2009, at 4:24 PM, Roberto Martins wrote: what about http://www.atlassian.com/software/confluence/ they give free licenses to open source project, and FS is using JIRA. roberto ___ 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] validating dtmf digits received
inline is run when the dialplan in parsed, everything else is run
later. So read sets digits dialed after it is finished parsing the
dialplan, if you transfer to another extensions after the read you can
then condition on that value.
Mike
On Oct 19, 2009, at 5:40 PM, Mark Campbell-Smith wrote:
Thanks Mike,
I have a lateish trunk and inline seems to work okay.
Does the inline statement below set variable ${code} to be used
directly or does it require transfer also? ie is digits_dialed
available for use right after a read statement (action
application=read data=1 10
ivr/ivr-please_enter_pin_followed_by_pound.wav res 1 9/ in my
case) or is it not 'set' until after the transfer?
action inline=true application=set data=code=$
{digits_dialed}
Thanks!
On Tue, Oct 20, 2009 at 12:32 AM, Michael Jerris m...@jerris.com
wrote:
inline is new, it won't work unless your using recent trunk. That
being said, read is not being run inline, so the set is actually
being
run before digits_dialed is set. You will most likely need to use
transfer in this situation.
Mike
On Oct 19, 2009, at 12:53 AM, Mark Campbell-Smith wrote:
Hi!
I simply want to validate the dtmf digits I read from a user.
From
the wiki, it appears I need to use inline=true when setting the
variable so it can be used directly within the same extension.
What have I done wrong below? I have tried many different
alternatives, but the second condition field, which is meant to
match
the dtmf digits received (in this case ) is never matched, and
the
anti-action is called instead.
:
some code here
action application=read data=1 10
ivr/ivr-please_enter_pin_followed_by_pound.wav res 1 9/
action application=phrase data=spell,${res}/
action inline=true application=set data=code=$
{digits_dialed}/
!-- action inline=true application=set data=code=$
{res}/ --
/condition
condition field=digits_dialed expression=^$
!-- condition field=${code} expression=^$ --
!-- condition field=${res} expression=^$ --
some code here
anti-action application=hangup/
Thanks!
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Re: [Freeswitch-users] EXCHANGE_ROUTING_ERROR -- how to diagnose?
FreeSWITCH debug level logs should help tell you exactly what is killing the call. On Oct 18, 2009, at 10:25 AM, Matthew Fong wrote: I'm still having this issue with random EXCHANGE_ROUTING_ERROR disconnects for users utilizing my inbound DID to connect to my FreeSWITCH server. It's a predictive dialing application, with one agent session being bridged with multiple calls and transfered back and forth between extensions in my dial plan. After a random number of bridging and transferring, FreeSWITCH suddenly sends a BYE to my DID provider citing an EXCHANGE_ROUTING_ERROR. It does not happen at any one-point in my dial plan, or applications--it just randomly disconnects when a call that the Agent is bridged to hangs-up or is disconnected. It seems to only happen when two external sip profiles are being bridged together, and not when an internal and external profile is being bridged. I turned sip trace on and sofia loglevel all 9 below is the the snippet. I've posted the entire Agent session at the following pastebin http://pastebin.freeswitch.org/10756 tport_deliver(0x18413c0): msg 0x7faeb818ea30 (304 bytes) from udp/ 208.76.18.254:5080/sip next=(nil) nta: received 200 OK for BYE (121818983) nta: 200 OK is going to a transaction nta_outgoing: RTT is 84.409 ms tport_release(0x18413c0): 0x1a15cc0 by 0x1a16a00 with 0x7faeb818ea30 nua(0x1ad6fb0): event r_bye 200 OK nua(0x1ad6fb0): call state changed: terminating - terminated nua(0x1ad6fb0): event i_state 200 to BYE nua: nua_application_event: entering nua(0x1ad6fb0): event i_terminated 200 to BYE nua: nua_handle_magic: entering nua(0x1ad6fb0): removing session usage soa_destroy(static::0x1b5ae90) called nua: nua_application_event: entering nta_leg_destroy(0x1b594a0) nua: nua_handle_magic: entering nua: nua_handle_bind: entering nua: nua_application_event: entering nua: nua_handle_magic: entering nua: terminated session 0x1ad6fb0 nua: nua_handle_destroy: entering nua(0x1ad6fb0): recv signal r_destroy nta_leg_destroy((nil)) nua(0x1ad6fb0): sent signal r_destroy nta: timer set next to 28 ms nta: timer E fired, retransmit BYE (121818989) tport_release(0x18413c0): 0x1b5c9b0 by 0x7faeb817d830 with (nil) tport_tsend(0x18413c0) tpn = */209.216.2.211:5060 tport_resolve addrinfo = 209.216.2.211:5060 tport_by_addrinfo(0x18413c0): not found by name */209.216.2.211:5060 tport_vsend(0x18413c0): 862 bytes of 862 to udp/209.216.2.211:5060 tport_vsend returned 862 send 862 bytes to udp/[209.216.2.211]:5060 at 14:04:11.753690: BYE sip:199.173.100.16:5060;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 67.220.216.146:5080;rport;branch=z9hG4bK02jNX8a4HrNyN Route: sip:209.216.2.211;ftag=4adc7-13c4-1ab03-71ce3705-1ab03;lr=on Route: sip:65.211.120.237:5060;lr Route: sip:63.110.102.239;lr Max-Forwards: 70 From: sip: +12133304...@63.110.102.239:5060;user=phone;tag=cgBe054jZrt3a To: sip: + 14158867717 @199.173.100.16:5060;user=phone;tag=4adc7-13c4-1ab03-71ce3705-1ab03 Call-ID: a0f656a01064adc713c41ab036840746ee20ca11c06b2d8-0440-5124 CSeq: 121818989 BYE Contact: sip:+12133304...@67.220.216.146:5080;transport=udp User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-15135 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, REGISTER, REFER, UPDATE, NOTIFY Supported: timer, precondition, path, replaces Reason: Q.850;cause=25;text=EXCHANGE_ROUTING_ERROR Content-Length: 0 Thanks. --matt On Sat, Aug 8, 2009 at 12:42 PM, Michael Jerris m...@jerris.com wrote: http://wiki.freeswitch.org/wiki/Sofia#Debugging_Sofia-SIP turn the logging all the way up and see what it says. Mike On Aug 4, 2009, at 2:51 PM, Matthew Fong wrote: Hi Mathieu, thanks for the reply. I enabled sip trace logging and got the logs below, but I am still at a loss at being able to identify the error or reproduce it consistently. The below log indicates to me that my FS server is initiating sending 2 BYE message to my DID provider (didforsale.com). Is there a way I can look further inside FreeSWITCH to see why it is sending this BYE packet? sent 882 bytes to udp/[209.216.2.211]:5060 at 17:15:44.679208: BYE sip:199.173.100.144:5060;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 66.197.142.69:5080;rport;branch=z9hG4bKD0UKrDB508mDD Route: sip:209.216.2.211;ftag=dc7-13c4-2401b7-46dea593-2401b7;lr=on Route: sip:65.217.40.210:5060;lr Route: sip:63.110.102.238;lr Max-Forwards: 70 From: sip: +1212381x...@63.110.102.238:5060;user=phone;tag=Ztr5ycrv3QZ1g To: sip: + 1909635 @199.173.100.144:5060;user=phone;tag=dc7-13c4-2401b7-46dea593-2401b7 Call-ID: a22bffb89064adc713c42401b78ca6b689e90b32fdbf612c0-0487-7441 CSeq: 118584736 BYE Contact: sip:+1212381x...@66.197.142.69:5080;transport=udp User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417M Allow: INVITE, ACK, BYE, CANCEL, OPTIONS
Re: [Freeswitch-users] Brazilians (Off-Topic)
I think we strongly lean towards using a sub-domain (if necessary) and maintaining other language content in the same wiki in alternate pages. If the wiki software we are using is not effective to create multiple languages we should find a way to do it all in one. We can set up an additional mailing list if necessary but also we have no issue with going across multiple languages in both the mailing list and in irc. Mike On Oct 17, 2009, at 4:01 PM, Roberto Martins wrote: sub-domain seem to be a very wise choice... ptbr On Oct 17, 2009, at 1:25 PM, Anthony Minessale wrote: or someone could just ask us for pg.freeswitch.org or whatever other sub-domain to avoid trademark infringement of registering 40 domains with freeswitch in the name for no reason. On Sat, Oct 17, 2009 at 6:06 AM, Rudá Cunha r...@ruda.com.br wrote: João Mesquita, Could check with Jeremiah or contact me through, so I talk to him about the forum? He could redirect the domain to my hosting server and we can put there the wiki and forums and we can communicate! Rudá Cunha 2009/10/17 João Mesquita jmesqu...@freeswitch.org Just a heads up, I have talked to Jeremias from Khomp today and he is setting up the wiki. I will personally be adding contents to the that wiki if it ever picks up. Regards, jm On Sat, Oct 17, 2009 at 12:52 AM, Jason White ja...@jasonjgw.net wrote: 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 ___ 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 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org pstn:213-799-1400 ___ 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 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 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] Dingaling: using a hostname instead of stun for rtp
If you don't have working stun, jingle is not going to work very well. It is a required part of the protocol. You need to be able to determine your external ports for media on each call, using a host name will not do this for you. Mike On Oct 16, 2009, at 10:48 AM, Brian West wrote: If you setup your own stun server it wouldn't do that But the hostlookup only solves half the problem .. getting the external IP vs poking holes for RTP which is what stun will do. /b On Oct 15, 2009, at 10:35 PM, Mark Campbell-Smith wrote: Thanks Brian. Is this something that is planned to be implemented? The workaround is to set the stun server also in the dingaling configuration, but as I said, for some reason the stun times for me out occasionally with dingaling. Thanks! ___ 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] sched_api doesn't get launched
sched_api is a fsapi command not a dialplan application, I believe
sched_hangup is both.
Mike
On Oct 13, 2009, at 6:14 AM, Henry Huang wrote:
Hi:
I am using mod_java. And in my script I was able to achieve using:
execute(sched_hangup, +300 alloted_timeout);
However, when I try to run sched_api in the same way, system log
returns that it's an invalid application. I have also tried to
trigger it with many conditional channel variable api calls , but
non of them seemed to execute the api command (because I turned on
the highest level of debugging and see no where the sched_api is
being called.
The closest thing I got was by using api_after_bridge like the
following, but it only launches when the bridge is teared down(which
is not what I want). I originally thought after bridge means right
after the 2 party is connected. All I want is to be able to play
some message to leg A at certain time.
setVariable(api_after_bridge, sched_api +10 none uuid_displace $
{uuid} start /path/to/some.wav 20 mux);
I have been struggling with different combination for a week now..
Please shed some light if you know something.
Thanks,
--
Henry Huang
UniC Solution - Communication Unified
VoIP Open Source software Consultant
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Re: [Freeswitch-users] Building Freeswitch with VPATH (src and obj directories are different)?
I would love to see this work in tree, but i am pretty sure it has never worked. I would gladly accept patches that implement this. Mike On Oct 14, 2009, at 2:33 AM, Simon J Mudd wrote: br...@freeswitch.org (Brian West) writes: You shouldn't have to make clean usually ... doing so might break your tree... Why? In any case on my mac (Leopard) neither make clean or distclean fully cleans up afterwards. I would prefer to get a completely untouched source tree after doing this. If that's supposed to work and I should post a bug or report the problem in more detail then I can do that. You can usually get by with make current that will ensure the critical things are cleaned and built correctly... every now and they you'll hit a snafu but we'll usually tell you about it. ok. That's one of the reasons for liking to use VPATH builds. It's also why many BSD system builds use this so that you can mount the source tree readonly (perhaps from another server) and build the binaries in a separate directory independently. If that's not supported then that's fine. I was trying to build the latest svn version of freeswitch (Revision: 15138) and this didn't build. As I'm not too familiar with compiling at least on the Mac I thought I'd first try to see if a VPATH build would work better. I'll have to try and investigate further. ___ 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] T.38 via UPDATE request
There was just a bunch of work on UPDATE, can you confirm this is the same behavior with trunk? On Oct 14, 2009, at 6:55 AM, Peter P GMX wrote: Hello, we have the following problem. 2 Fax machines are communicating via Freeswitch. One is externally attached via a Telco who is able to handle T.38. The other one is attached locally. When 2 Fax machines start syncing each other, the Telco sends a SIP UPDATE message with T.38 SDP, as it detects fax during the fax negociations. Freeswitch answers with an SIP OK message back to the telco, and I can see the T.38 SDP on the debug console of freeswitch. Then nothing happens any more until one of fax machines detects timeout. We have set proxy-media to true. However is was done during call setup when both machines communicated with G711 SDP. The UPDATE message was commited by FS to the telco, but was not sent to the other fax, so I think in this case Freeswitch is supposed to transcode between T.38 and G711 which it cannot do, as we know. How can I overcome this scenario? Is this a defect, should freeswitch send the UPDATE message to the other fax? Or is there a workaround? Best regards Peter ___ 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] TLS Audiocodes
Try turning up all the sofia debug to 9. Mike On Oct 14, 2009, at 2:16 AM, Szasz Szabolcs wrote: Hi, Did anybody set up TLS between Freeswitch and Audiodes MP11X ? I got to work TLS between freeswitch and a softphone (phonerlite), but I have problem with Audiocodes during the TLS authentication. I've loaded the certification but it still doesn't work. Can I debug the tls in freeswitch? Please help me. ___ 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] Sending an Event to a Session for onInput
It updates the display on a phone if the phone supports this. This works on some sip phones right now including polycom and snom. Mike On Oct 12, 2009, at 2:11 AM, Matthew Fong wrote: Hi Mike, I'm just trying to send it an event with some custom event headers, just so an external program can communicate with a session without having to transfer the session to a different program. I'm curious what uuid_display does...the wiki only gives a brief description and my Google'ing could not find any examples. Thanks for the help. --matt http://www.hellohunter.com On Mon, Oct 12, 2009 at 5:04 AM, Michael Jerris m...@jerris.com wrote: We don't have session messages directly exposed, except for things like display, respond, and deflect. What specifically are you trying to send ? Mike On Oct 9, 2009, at 3:34 PM, Matthew Fong wrote: I'm used to using the onInput callbacks inside lua and javascript to listen for dtmf and other events and perform a task accordingly. I'm wondering if there is a way to send an event to a session or channel that can be caught using the setInputCallback inside lua from outside the session program. Maybe an API command that can generate an event for a specific UUID. Does a mechanism exist to do this that I'm over looking? Thanks. ___ 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 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 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] Fwd: Groups information in sqllite
Group information is not stored in sqlite, it is pulled from the xml registry (switch_xml_locate_group function can find them) . Also, please do not cross post between lists. http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide#Groups http://wiki.freeswitch.org/wiki/Mod_commands#in_group http://wiki.freeswitch.org/wiki/Mod_commands#group_call Mike On Oct 13, 2009, at 2:02 AM, srinivasula reddy wrote: can any know where group information is exactly stored in sqllite database, i have seen sip_registration here i can find the registered users, in the same way how i can i find the group information, and which user belongs to which user? any help would be great. ___ 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_sofia.c registered calls how to know
On Oct 7, 2009, at 10:48 AM, srinivasula reddy wrote: Hi can any please tell me where registered calls are stored, so when incoming call came to mod_sofia.c how it will check it is registered or not?\\ Calls are not registered and calls have nothing to do with registration. Users are registered so that you may send calls to them. Registration data is stored either in a sqlite database, or optionally if you setup odbc, in another database of your choice. If you try to send a call to an unregistered user in the dialplan using the proper syntax to send calls to registered users (see the wiki for more details), and that user is not registered, the bridge app will fail, optionally letting you continue on in the dialplan based on variables such as continue_on_fail and hangup_after_bridge. You can use the sofia_contact function to see if there is anyone registered to a specific user. Mike ___ 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 Slide deck?
On Oct 9, 2009, at 2:41 AM, Gabriel Gunderson wrote: On Fri, Oct 9, 2009 at 12:08 AM, Michael Collins m...@freeswitch.org wrote: Thanks for reporting back. Please let all the Asterisk users know that they are welcome to join us in #freeswitch on irc.freenode.net and that they will not be abused like people do in other less friendly IRC channels. Funny you mention this. Many people report that the way the FS community refers to Asterisk in docs/wikis/irc/whatever makes the FS camp seem *less* welcoming to them. After all, they identify as Asterisk Users and take the criticism as being kinda harsh. Most of them acknowledge the shortcomings of Asterisk but are put off when someone else points them out. It's crazy, I know. The thing is, I remember thinking that too. After getting to know FS better, I didn't notice it as much. Nobody likes to hear their baby is ugly --even if they know it is. At our session, and in general, I've noticed people are more interested in hearing about FS when you don't make direct comparisons to Asterisk. Besides, FS stands on it's own merit. Just what I've observed *and* my 2 additional cents. I have certainly seen this on irc in the past and we should do our best to avoid this, I have not seen this in the docs or wiki, do you know of any specifics you can point me to so we can correct this issue. Mike ___ 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] gateway FS informs it's client FS about users hanged up with a long delay
On Oct 9, 2009, at 10:40 AM, Maciej Aniserowicz wrote:
Hello,
The issue is resolved. I feel stupid, because Michael Jerris was
right the first time. Setting external_rtp_ip and external_sip_ip to
$${local_ip_v4} made it work.
But the strange thing is: it SOMETIMES worked before without any
delay, which 'should not be possible', because the original IP was
my external ip and the BYE message was sent straight to it. And
there is no way it could reach the target 'internal' FS, because it
runs on virtual machine, and no ports are forwarded on my router.
Any thoughts? Why this could (rarely) work even with the previous
config?
Thanks to both of you for your answers.
MA
It could work at least in regards to rtp if you had something else on
the other side that adjusted to incorrect rtp ip like freeswitch does,
the bye probably never really worked.
Mike
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Re: [Freeswitch-users] On the handling of SIP headers
There is this endless push and pull on this topic, those who want them
assume it should be default, those who don't assume that should be
default. This probably needs a configuration option defaulting to
pass them (those who don't want to pass them are usually a bit more
educated and would find the option better than the other way around).
Mike
On Oct 9, 2009, at 3:10 PM, Kristian Kielhofner wrote:
Hello everyone,
In using FS for various scenarios I've noticed some behavior that
I'm not sure is completely proper. Given that this probably lives
in mod_sofia who knows what's really proper. It is SIP after all...
So the issue comes up when using FreeSWITCH as a B2BUA and bridging
between endpoints (very common). Should FreeSWITCH copy the X-
headers (possibly others) as it does now? I'd like to think it
shouldn't by default and the behavior should be one of:
1) Don't pass X-* (or anything else, really) from one leg to another.
If you want to pass specific X- headers (or anything else), set them
explicitly on the outbound leg.
2) Make the behavior configurable with a channel variable and/or
sofia config option:
{sip_pass_headers=all|none|X-MyCustomHeaderByName}
Thoughts?
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Re: [Freeswitch-users] Sending an Event to a Session for onInput
We don't have session messages directly exposed, except for things like display, respond, and deflect. What specifically are you trying to send ? Mike On Oct 9, 2009, at 3:34 PM, Matthew Fong wrote: I'm used to using the onInput callbacks inside lua and javascript to listen for dtmf and other events and perform a task accordingly. I'm wondering if there is a way to send an event to a session or channel that can be caught using the setInputCallback inside lua from outside the session program. Maybe an API command that can generate an event for a specific UUID. Does a mechanism exist to do this that I'm over looking? Thanks. ___ 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] Bad sound quality while eavesdropping
maciej.aniserow...@gmail.com wrote: It's the same on the trunk (the last rev I used was not so old anyway). Codecs are the same on both legs: read codec/read rate: PCMU 8000 write codec/write rate: PCMU8000 MA Michael Jerris wrote: What codecs are all the call legs using, also, please try current svn trunk. Mike On Oct 7, 2009, at 3:39 AM, Maciej Aniserowicz wrote: Sorry about posting several questions at once, I wasn't aware it's rude. Let's concentrate on this issue then. I use FS rev 14994. Phones on extensions: 1) x-lite 2) cisco sip phone 3) audio played by fs to the extension being eavesdropped I did not change any codec configuration, I just use the standard one that comes with both FS and the phones. Some time ago someone on FS irc channel told me that this is just how FS eavesdropping works... from your response I understand that this is not entirely true? Maciej Aniserowicz Anthony Minessale wrote: That's is a somewhat vague position. You did not mention which version of FreeSWITCH you are running, the phones being used in your example, your configuration, the codecs in use etc. BTW, I think you should only ask one question at a time on this list. The list is run by volunteers and it's sort of rude to expect 3 or 4 threads to be tended to concerning the same one individual. 2009/10/5 Maciej Aniserowicz maciej.aniserow...@gmail.com Hello, When I use eavesdropping in FreeSWITCH, the sound quality is really bad. Is there any way to improve it? Is this a known problem? Br/ Maciej Aniserowicz ___ 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 -- View this message in context: http://n2.nabble.com/Bad-sound-quality-while-eavesdropping-tp3768542p3788019.html Sent from the freeswitch-users mailing list archive at Nabble.com. ___ 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com MSN %3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.comPAYPAL %3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip %3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3Aconf %2b...@conference.freeswitch.org pstn:213-799-1400 ___ 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 -- View this message in context: http://n2.nabble.com/Bad-sound-quality-while-eavesdropping-tp3768542p3799274.html Sent from the freeswitch-users mailing list archive at Nabble.com. ___ 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 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_socket: custom timeout for HEARTBEAT event
On Oct 11, 2009, at 5:44 PM, Diego Viola wrote: Maybe you want enable_heartbeat or sched_heartbeat from mod_dptools? You can pass your parameters in second to these two. Example: action application=enable_heartbeat data=1/ action application=sched_heartbeat data=1/ Where 1 in this case is the number of heartbeats per seconds. Number of seconds between hearbeats, not hearbeats per second. You can use that example on the Dialplan XML but you can also use it on mod_event_socket outbound, etc. ___ 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 compiling esl for use with freepbx v3
I am still working on the new build system for esl, stay tuned for more info soon, it should be in 1.0.5. Mike On Oct 11, 2009, at 5:36 PM, Herman Griffin wrote: Although probably not the best solution, I figured out a way to make it compile and install: I removed all of the -Werror instances in PATH_TO_FREESWITCH_SOURCE/ libs/esl/Makefile If I was a hardcore c/c++ programmer, I'd figure out the real problem. Herman aka frek818 On Sun, Oct 11, 2009 at 12:12 PM, frek818 herman.grif...@gmail.com wrote: Did anyone find a solution to this problem? I too would like to install the esl module for PHP. ___ 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] apr_queue
On Oct 9, 2009, at 7:58 AM, srinivasula reddy wrote: Hi all, does any know about How apr_queue is maintaing and retriving all registered and all stuff parse error ___ 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] Bad sound quality while eavesdropping
What codecs are all the call legs using, also, please try current svn trunk. Mike On Oct 7, 2009, at 3:39 AM, Maciej Aniserowicz wrote: Sorry about posting several questions at once, I wasn't aware it's rude. Let's concentrate on this issue then. I use FS rev 14994. Phones on extensions: 1) x-lite 2) cisco sip phone 3) audio played by fs to the extension being eavesdropped I did not change any codec configuration, I just use the standard one that comes with both FS and the phones. Some time ago someone on FS irc channel told me that this is just how FS eavesdropping works... from your response I understand that this is not entirely true? Maciej Aniserowicz Anthony Minessale wrote: That's is a somewhat vague position. You did not mention which version of FreeSWITCH you are running, the phones being used in your example, your configuration, the codecs in use etc. BTW, I think you should only ask one question at a time on this list. The list is run by volunteers and it's sort of rude to expect 3 or 4 threads to be tended to concerning the same one individual. 2009/10/5 Maciej Aniserowicz maciej.aniserow...@gmail.com Hello, When I use eavesdropping in FreeSWITCH, the sound quality is really bad. Is there any way to improve it? Is this a known problem? Br/ Maciej Aniserowicz ___ 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] Recording creates a 388-byte long file and deletes it
switch_ivr_async.c:480 On Oct 5, 2009, at 3:16 AM, Maciej Aniserowicz wrote: Hi, When I record a call in FS, it only creates a 388-byte-long wav file. The conversation is no written there, and FS deletes the file when the session finishes. What can cause this strange behavior? ___ 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] gateway FS informs it's client FS about users hanged up with a long delay
Incorrect NAT configuration so one of the boxes is not actually
getting a BYE.
On Oct 5, 2009, at 3:13 AM, Maciej Aniserowicz wrote:
Hi,
When I use two FreeSWITCH instances ('internal' and 'external'), all
users register to the 'external' instance which acts as a gateway by
'internal' instance (which in turn is controlled by my applicaiton
with commands sent by socket).
When user hangs up, the 'hanged up' event is propagated to the
'internal' instance after a long time (~3 minutes) instead of being
propagated immediately.
What can cause this issue?
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Re: [Freeswitch-users] Bridge application with shared lines
On Oct 6, 2009, at 4:14 AM, Yehavi Bourvine wrote:
Hello,
We have Polycom and SNOM phones running with FreeSwitch. The
Polycoms have shared lines defined and the SNOMs have both shared
lines and BLFs (defined as extensions in the phone config). I've
tried supporting both, but have some incompatibility:
When calling the Bridge application with data parameter of sofia/
profile-name/num...@domain the BLF works ok, but not the shared
lines (i.e only one of the phones rings).
When calling the Bridge application with data parameter of $
{sofia_contact(/profile-name/num...@domain)} shared lines work ok
but BLF doesn't fire up.
How do I support both? Is there a way to know whether the
destination is a shared one and then chose one of the above formats?
You should probably always be using the second method or using a %
instead of the @ in the first method to get the registered contact.
can you provide more information about why the BLF doesn't fire up .
Mike
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Re: [Freeswitch-users] Basic compile question.
As I said in the duplicate thread, the voip codecs issue has been resolved in trunk, I had a change 1/2 done waiting for testing and it is now complete. Mike On Oct 6, 2009, at 12:30 AM, David Clark wrote: No I found the one header. I added it to the include list for the project. It included something else, added that. etc. Basically I think I am going to need the VC 2008 compiler and to use the other project file. At 04:41 PM 10/5/2009, Brian West wrote: Have you updated today? /b On Oct 5, 2009, at 3:49 PM, David Clark wrote: Any idea what is up? ___ 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] stun not working in fs 1.0.4?
I am not sure what you mean, do you think that fixes from today should somehow go somewhere else before we do a release? On Oct 6, 2009, at 3:21 AM, Vladimir Elizarov wrote: Brian West пишет: Because TRUNK is stable... its only fixes going in usually and if things do break they don't stay broken for long. Ask anyone our trunk is more table then most commercial products. This separation of the branches a very bad influence on the packaging. That is gathered deb-package trunk 15094. Man found in the trunk bug. Must again rebuild the package from the new trunk... /b On Oct 5, 2009, at 12:50 PM, Vladimir Elizarov wrote: Ok. Brian, why fs no two branches of the stable and trunk? ___ 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 http POST is inconsistent/bug
Could you open a bug on jira.freeswitch.org as a feature request to make this a configurable param. (patches that do it even better) Mike On Oct 6, 2009, at 12:55 PM, Christian Damianidis wrote: I’ve tested this and making the change from ANY to BASIC worked. Thanks for the help. It no longer sends the initial post without auth. From: Anthony Minessale [mailto:anthony.miness...@gmail.com] Sent: Tuesday, October 06, 2009 11:02 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_xml_curl http POST is inconsistent/bug My guess is that we configure the curl to support the full range of http auth methods. Some of them like Digest require a challenge and realm etc so it's probably asking without auth header because it cannot create one until it gets that data. In the case of Basic you can send the login and pass right away but it does not know in advance that it will be basic. Here is a snippet from the libcurl api docs: - Both these options allow you to set multiple types (by ORing them together), to make libcurl pick the most secure one out of the types the server/proxy claims to support. This method does however add a round-trip since libcurl must first ask the server what it supports: curl_easy_setopt(easyhandle, CURLOPT_HTTPAUTH, CURLAUTH_DIGEST| CURLAUTH_BASIC); - So my guess is that if we set it to only support basic, then it would work how you expect so if you want to test it for me I can make it into a parameter. edit: /usr/src/freeswitch.trunk/src/mod/xml_int/mod_xml_curl/ mod_xml_curl.c line 220 change curl_easy_setopt(curl_handle, CURLOPT_HTTPAUTH, CURLAUTH_ANY); to curl_easy_setopt(curl_handle, CURLOPT_HTTPAUTH, CURLAUTH_BASIC); If this works i'll think about exposing the auth methods so you can choose them in the config. ___ 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] Detecting a fax
Fax tones are not played by the remote machine until after answer, the
tone_detect application starts a media bug that listens for the tone,
can you confirm the tone is happening at all. Maybe the issue here is
the timeout, try making that longer, or doing the tone_detect in
execute_on_answer
Mike
On Oct 5, 2009, at 6:28 AM, Mark Campbell-Smith wrote:
Thanks for the response Mike,
I read that page and this one (among others)
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_fax_detect, but
I'm still lost. This is an extract of my dialplan
extension name=Local
condition field=destination_number expression=^(10[01][0-9])
$
action application=set data=dialed_extension=$1/
action application=export data=dialed_extension=$1/
action application=set data=ringback=${au-ring}/
action application=fax_detect/
action application=tone_detect data=fax 1100 r +5000
transfer fax XML features /
action application=set data=hangup_after_bridge=true/
action application=set data=continue_on_fail=true/
action application=bridge data=user/${dialed_extensi...@$
{domain}/
I would assume that on detecting a fax, the dialplan 'fax' is called
in context features. This never happens.
When is the fax tone detected? Is it while the call is ringing or
can it be detected after the call is answered? My goal is to be able
to have the same extension for a voice and fax call. i assume that
the fax 'tones' are standardised and the ones on the wiki are correct?
Also, I guess this doesn't work with media bypass (which I don't
use).
Thanks!
On Mon, Oct 5, 2009 at 9:56 AM, Michael Jerris m...@jerris.com
wrote:
check out http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_tone_detect
Note, you can't just have tone_detect as your last iten in the
dialplan as the call will just get hung up.
Mike
On Oct 4, 2009, at 9:03 AM, Mark Campbell-Smith wrote:
Hi
I was hoping someone could help me to setup the fax detection / tone
detection application.
I want to be able to transfer an incoming fax to a specific
extension.
In my default.xml file, I have the following (extracted):
extension name=1000
condition field=destination_number expression=^(10[01]
[0-9])
$
action application=fax_detect/
action application=tone_detect data=fax 1100 r +5000
transfer fax XML features /
I can't get the fax to be detected and transferred. Is there any
way
this can be done?
Thanks!
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Re: [Freeswitch-users] Re corded file as voicemail.
http://wiki.freeswitch.org/wiki/Mod_voicemail#voicemail_inject On Oct 5, 2009, at 9:46 AM, Nagalenoj wrote: Is it possible to treat a recorded voice as voice mail? Assume that, I've recorded a conversation and I want this recorded file to be treated like voicemail. So, I could check it like 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] overriding conference preference
On Oct 5, 2009, at 10:49 AM, Woody Dickson wrote: Is is possible to override any of the setting specified in the conference profile? Just the flags you can pass per user such as pin and mute What I want to do is to have a default profile, and be able to modify certain fields if necessary in the dialplan. Alternatively, I would prefer to have a dynamic profile setting for the conference to obtain those parameters from odbc. you can do this with mod_xml_curl Mike ___ 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] UPDATED: Basic compile question.
voip codecs is fixed, ptlib I can't recall if we ever did full build integration or if you needed to manually download the libraries, can someone who has done mod_opal build on windows comment? Mike On Oct 5, 2009, at 5:14 PM, David Clark wrote: Ok I found spandsp.h. It is a case of the project file being out of date. No surprise. ptlib.h is still not found. Ok using windows xp x64 here. I download from the trunk as expected. I fire up VS 2005 and I open the VS 2005 solution. Yes it does say unsupported. But I get two missing header files: freeswitch\src\mod\endpoints\mod_opal\mod_opal.h(33) : fatal error C1083: Cannot open include file: 'ptlib.h': No such file or directory .\mod_voipcodecs.c(36) : fatal error C1083: Cannot open include file: 'spandsp.h': No such file or directory Even if the project file is wrong or out of date I should be able to find the include files some place in the fileset. I can't find either file in the freeswitch directory or below 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] Outbound INVITE rejected with 480 Temp Unavail, Reason MANDATORY_IE_MISSING
there is a profile param to enable 3pcc. It should be documented in the default configs. Mike On Sep 29, 2009, at 5:22 PM, Jerry Richards wrote: Hello All, I have an internal extension that needs to send an INVITE without SDP body (Content Length 0). Freeswitch is replying with 480 Temporarily Unavailable with reason MANDATORY_IE_MISSING. Would anyone know what I need to do to enable this? ___ 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 subscription expire
This sounds like a bug in the snom to me, we keep changing the expire on to the future so it should never expire in the first place. You will have to look at a longer running sip trace to see what exactly is going on. Mike On Oct 1, 2009, at 4:52 AM, Helmut Kuper wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello, it seems exired subsciptions are never cleared in FS. A look into sofia_presence.c confirms explains this /* negative in exptime means keep bumping up sub time to avoid a snafu where every device has it's own rules about subscriptions that somehow barely resemble the RFC not that I blame them because the RFC MAY be amibiguous and SHOULD be deleted. So to avoid the problem we keep resetting the expiration date of the subscription so it never expires. Eybeam completely ignores this option and most other subscription-state: directives from rfc3265 and still expires. Polycom is happy to keep upping the subscription expiry back to the original time on each new notify. The rest ... who knows...? */ For some reasons subscriptions created by Snom phones are filling up the sip_subscriptions table over time. This leads to some kind of DOS by FS against the subscribing phone ... The subscribtions are differentiate by call-id. This can be explained by RFC 3842 chapter 3.6 where expired subscriptions must be renewed with a NEW call-id. Because there is no hint about unsubscribing the old subscription I guess the clean up process has to be done by FS. Any way to get FS to do this job? Since there is no creation date or expire value which represents the expire as a timestamp I have no way to clean up the table manually via sql and cronjob - except cleaning the whole table ... A further (but background) question is, why do the subscriptions expire in snom phones 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
