Re: [Freeswitch-users] Question with curl_xml

Hi, I changed gateway-url to http://localhost/fs/directory.php"; bindings="directory."/> then the error messages go away. However, I am still not able to authenticate user. I can see the response in the /tmp directory, but the result is not right. So, I am suspecting that the

[Freeswitch-users] Question with curl_xml

Hi, I place my xml response generator directory.php under /var/www/fs/directory.php In my xml_curl_conf.xml, I have http://localhost/fs/"; bindings="directory.php"/> 2008-04-18 22:23:17 [ERR] mod_xml_curl.c:251 do_config() Binding has no url! 2008-04-18 22:23:17 [NOTICE] mod_xml

Re: [Freeswitch-users] newbie dialplan question

Hi , The problem is that if I call from the xLite (1001) that is on the same machine as the fs to sip client (1004) on another machine , it can't be connected. If I call from the other sip client to the xlite client, then it works. Don't know why. >From the log, xLite ( 1001) calling sip client

Re: [Freeswitch-users] Hangup Cause: UNALLOCATED

This is the complete log of a call from eyeBeam to SIPp: 2008-04-17 14:53:45 [NOTICE] switch_channel.c:468 switch_channel_set_name() New Chan sofia/freeswitch.mocon.esk.fraunhofer.de/[EMAIL PROTECTED] .de:5060 [5

Re: [Freeswitch-users] newbie dialplan question

Let me guess you're running FS and X-Lite on the same machine? What was your problem again? From the looks of this you have changed the default configs a little bit. /b On Apr 17, 2008, at 11:48 AM, Pete Kay wrote: Hi Brian, Here is the full log. Please kindly take a look: 2008-04-18

Re: [Freeswitch-users] Build problem on Debian slice

segfault right away.. guess you don't get nuthin for free. I'll just play without js for a while I guess. Tony On Thu, Apr 17, 2008 at 9:03 PM, Anthony Knight <[EMAIL PROTECTED]> wrote: > Ok this is weird, > > first off using ./configure --host=x86 and/or build=x86 did not help, > libfreeswitch

Re: [Freeswitch-users] Build problem on Debian slice

Ok this is weird, first off using ./configure --host=x86 and/or build=x86 did not help, libfreeswitch.la would not build. I started from scratch again - deleted all directories, retrieved source using svn - got exactly the same problem. So then I tried editing os_Linux_x86_64.s, and substituted

Re: [Freeswitch-users] Segfault

I say first C then a bit of A on top of that to prevent the situation that leads to that. On Thu, Apr 17, 2008 at 3:51 PM, David Knell <[EMAIL PROTECTED]> wrote: > Hi all - > > Just had an interesting FS segfault. switch_xml_free (called from > mod_xml_cdr) recursed 3500+ times > and, I guess,

[Freeswitch-users] freeswitch as a session border controller

Hi, I have been reading the recent news articles about freeswitch. I am particularly interested in using it to replace a session border controller we have (kagoor voiceflow 1000). Has anyone actually used this in production as a session border controller for NAT traversal? Thanks Robert McNaug

[Freeswitch-users] Segfault

Hi all - Just had an interesting FS segfault. switch_xml_free (called from mod_xml_cdr) recursed 3500+ times and, I guess, the thing ran out of stack space. Having a look at the core dump shows that it's freeing a valid XML CDR structure for a call which wasn't cleared for ages; FS sat in a lo

Re: [Freeswitch-users] Build problem on Debian slice

You can try the --host configure arg to force the platform to 32 Mike On Apr 16, 2008, at 8:41 PM, "Anthony Knight" <[EMAIL PROTECTED]> wrote: I'm having a problem building freeswitch on a xen debian etch slice (Details below). Everything from the standard modules.conf builds except for

Re: [Freeswitch-users] newbie dialplan question

This keeps you from calling your self in some situations. /b Sent from my iPhone On Apr 17, 2008, at 12:00 PM, Daniel Hefti <[EMAIL PROTECTED]> wrote: Added a brief about this to the wiki. I took a quick look through sofia.c to see if other variables are available from the dialplan, but I

Re: [Freeswitch-users] RC3 Callie & Music

On Apr 17, 2008, at 6:15 AM, Michael Jerris <[EMAIL PROTECTED]> wrote: > DOH!! I knew I forgot something. Haha you suck! :) Just kidding. You have done a ton on FS and we appreciate your hard work. -MC > Will roll those this morning. You > can grab the rc2 ones, I think there was only a

Re: [Freeswitch-users] newbie dialplan question

Added a brief about this to the wiki. I took a quick look through sofia.c to see if other variables are available from the dialplan, but I wasn't entirely sure which variables were meant/are to be accessible or how they are used, other than the mentioned header variables that can be set, which

Re: [Freeswitch-users] newbie dialplan question

Hi Brian, Here is the full log. Please kindly take a look: 2008-04-18 08:51:44 [DEBUG] switch_core_session.c:730 switch_core_session_thread() Session 14 (sofia/default/ [EMAIL PROTECTED]:5061) Locked, Waiting on external entities 2008-04-18 08:51:44 [NOTICE] switch_core_session.c:748 switch_core

Re: [Freeswitch-users] newbie dialplan question

Excellent, that was exactly what I was looking for. I don't know why I didn't see that. (I was close, but no cigar. Guess I need to brush up my dialplan skills.) -Dan From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian West Sent: Thursday, April 17, 2008 10:15 AM To: freesw

Re: [Freeswitch-users] newbie dialplan question

Please include more logs. Did you remove any files? Something isn't right here since I see it hunting the public context Did you include the user_context var on the user in the directory? /b On Apr 17, 2008, at 10:38 AM, Pete Kay wrote: Hi, I comment the line at default.xml

Re: [Freeswitch-users] newbie dialplan question

Hi, I comment the line at default.xml But, it still does not work. 2008-04-18 07:38:32 [INFO] mod_dialplan_xml.c:223 dialplan_hunt() Processing 1002->[EMAIL PROTECTED] 2008-04-18 07:38:32 [INFO] switch_ivr_async.c:1357 switch_ivr_bind_dtmf_meta_session() Bound: 1 execute_extension::

Re: [Freeswitch-users] Need help with a gateway problem

Hi, I created another callwithus.xml under outbound and it is working now. Thanks alot. Pete On Thu, Apr 17, 2008 at 11:27 PM, UV <[EMAIL PROTECTED]> wrote: > Try: > > Sofia status gateway callwithus > > > > (maybe someone should add this to the wiki page > http://wiki.freeswitch.org/wiki/Sofi

Re: [Freeswitch-users] Need help with a gateway problem

Try: Sofia status gateway callwithus (maybe someone should add this to the wiki page HYPERLINK "http://wiki.freeswitch.org/wiki/Sofia"http://wiki.freeswitch.org/wiki/Sofia ) _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Pete Kay Sent: Friday, April 18, 2008 1:08 A

Re: [Freeswitch-users] RC3 Callie & Music

Done. /b On Apr 17, 2008, at 10:21 AM, Peder @ NetworkOblivion wrote: > I see that callie is there for rc3 now, but not music. Are you > going to > do that one too? > > Michael Jerris wrote: >> DOH!! I knew I forgot something. Will roll those this morning. You >> can grab the rc2 ones, I t

Re: [Freeswitch-users] newbie dialplan question

The error is in place: There aren’t any XML files in this directory. It is like that for some reason in the example as there are few sample *.noload files there if you DO like to have one (I think). There’s nothing to fix. Your reloadxml command worked. _ From: [EMAIL PROTECTED]

Re: [Freeswitch-users] RC3 Callie & Music

I see that callie is there for rc3 now, but not music. Are you going to do that one too? Michael Jerris wrote: > DOH!! I knew I forgot something. Will roll those this morning. You > can grab the rc2 ones, I think there was only a couple small changes. > > On Apr 17, 2008, at 6:55 AM, "Pede

Re: [Freeswitch-users] newbie dialplan question

$$ is expanded at config parse/load and $ is expanded every time its encountered. In the example of sofia the domain usually won't change every call where the extension will. /b On Apr 17, 2008, at 10:16 AM, Pete Kay wrote: Hi Brian, What is the difference between $$ and $ in FS? Thanks

Re: [Freeswitch-users] Hangup Cause: UNALLOCATED

the beginning of the first line in your trace is missing some of the data can you include at least 10 lines before where you started from in this post. On Thu, Apr 17, 2008 at 10:07 AM, Brian West <[EMAIL PROTECTED]> wrote: > Can you include the log prior to this? > /b > > On Apr 17, 2008, at 9:4

Re: [Freeswitch-users] newbie dialplan question

Again remove /b On Apr 17, 2008, at 3:16 AM, Pete Kay wrote: Hi, I am working on some test on seeing how I can port my exist Asterisk stuff to Freeswitch. I am just getting started and I am hoping someone can give me some help to get started. I installed with all the default config

Re: [Freeswitch-users] newbie dialplan question

Hi Brian, What is the difference between $$ and $ in FS? Thanks, Pete On Thu, Apr 17, 2008 at 11:11 PM, Brian West <[EMAIL PROTECTED]> wrote: > You could also use user/[EMAIL PROTECTED] > > in addition you could do sofia/$${domain}/${dialed_ext} > > /b > > > > On Apr 17, 2008, at 9:45 AM, UV wr

Re: [Freeswitch-users] Need help with a gateway problem

Well I don't personally put them in the config like that. I put then in the directory. You can check out the .noload files I put in SVN to demonstrate that. In addition you should put this on the profile called outbound you do not want a provider on a profile you auth-calls on usually.

Re: [Freeswitch-users] newbie dialplan question

Remove this line from the default config: /b On Apr 17, 2008, at 9:57 AM, Daniel Hefti wrote: A related problem I have is if you have multiple sip accounts on the same machine (registering from the same ip address), they can’t call each other. It may stem from the problem above, but it

Re: [Freeswitch-users] newbie dialplan question

You could also use user/[EMAIL PROTECTED] in addition you could do sofia/$${domain}/${dialed_ext} /b On Apr 17, 2008, at 9:45 AM, UV wrote: > I’m not sure which build you’re using, but I had that problem on sub > version 7946 (RC1). > > There’s a “bug” in the conf/dialplan/default.xml exampl

Re: [Freeswitch-users] newbie dialplan question

No thats not a bug. You have no files in there to include yet. That is setup so you can drop in stuff to include into the default context without having to edit the default.xml ie config sniplets. /b On Apr 17, 2008, at 10:05 AM, Pete Kay wrote: > Hi, > > When I do reloadxml, I keep on get

Re: [Freeswitch-users] Hangup Cause: UNALLOCATED

Can you include the log prior to this? /b On Apr 17, 2008, at 9:46 AM, Alois Komenda wrote: Hello, I'm trying to place a call between two SIPp instances over FreeSWITCH. Before the connection is set up completeley FS sends a BYE to the caller. In the logs is see the hangup cause is "UNA

Re: [Freeswitch-users] newbie dialplan question

Hi, When I do reloadxml, I keep on getting this error: [EMAIL PROTECTED]> reloadxml Error including /usr/local/freeswitch/conf/dialplan/extensions/*.xml API CALL [reloadxml()] output: Is this another bug in the sample files? If so , how do i fix it? Thanks, Pete __

Re: [Freeswitch-users] Need help with a gateway problem

Hi, I tried "sofia status" and it does not show my gateway, [EMAIL PROTECTED]> sofia status API CALL [sofia(status)] output: Name Type Data State = 192.16

Re: [Freeswitch-users] newbie dialplan question

I have noticed this, too. I can open up ekiga on the machine FS is installed on, and it can accept calls, but, if I remember correctly, FS doesn't show that the call is being routed, meaning ekiga is just accepting all the calls itself. What I also found, in relation to this, is that if you hav

Re: [Freeswitch-users] Need help with a gateway problem

Try this instead: http://sip.callwithus.com"sip.callwithus.com"/> _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Pete Kay Sent: Thursday, April 17, 2008 11:02 PM To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] Need help with a gateway problem

Re: [Freeswitch-users] newbie dialplan question

I’m not sure which build you’re using, but I had that problem on sub version 7946 (RC1). There’s a “bug” in the conf/dialplan/default.xml example where under the “Local_Extension” section is says: Where it should actually be: Check HYPERLINK "http://wiki.freeswi

[Freeswitch-users] Hangup Cause: UNALLOCATED

Hello, I'm trying to place a call between two SIPp instances over FreeSWITCH. Before the connection is set up completeley FS sends a BYE to the caller. In the logs is see the hangup cause is "UNALLOCATED". That happens only if there is a pause after sending the ACK. Hangup sofia/freeswitch.moc

Re: [Freeswitch-users] Missing bootstrap

Oh, well that could explain why I had problems initially setting it up using rc’s, because .configure (as well as everything after, of course) didn’t work. The only way I found to fix that was using the bootstrap script I found in the snapshots prior to anything else. So, I hope the bootstrap

[Freeswitch-users] Build problem on Debian slice

I'm having a problem building freeswitch on a xen debian etch slice (Details below). Everything from the standard modules.conf builds except for mod_spidermonkey. Here are the details of the build error. making all mod_spidermonkey cd config; make -j1 export cd pr; make -j1 export cd include; ma

Re: [Freeswitch-users] RC3 Callie & Music

DOH!! I knew I forgot something. Will roll those this morning. You can grab the rc2 ones, I think there was only a couple small changes. On Apr 17, 2008, at 6:55 AM, "Peder @ NetworkOblivion" <[EMAIL PROTECTED] > wrote: > OK, I got through the make and install with rc3, but now I am trying

[Freeswitch-users] Need help with a gateway problem

Hi, I am still not yet able to get one sip phone to call the other due to the problem I posted. So, I used the working sip client to try to call an outside number by setting up a gateway. I tried to set up fastswitch to route call to my voip provider, I am getting channel error but can't know wh

[Freeswitch-users] RC3 Callie & Music

OK, I got through the make and install with rc3, but now I am trying "make sounds-install" and "make sounds-music-8000-install" and it is failing. It appears to be trying to download the following files which don't exist yet: http://files.freeswitch.org/freeswitch-sounds-en-us-callie-8000-1.0.

[Freeswitch-users] newbie dialplan question

Hi, I am working on some test on seeing how I can port my exist Asterisk stuff to Freeswitch. I am just getting started and I am hoping someone can give me some help to get started. I installed with all the default config and xml setting. Then, I bring up two SiP clients - one in the same mach