date:20091209

On Tue, Dec 8, 2009 at 5:42 AM, DJB djbin...@yahoo.com wrote:
 One thing that I forgot to mention, these 2 FreeSWITCH servers are getting
 calls with load balancing from another switch.  Thus, the traffic type are
 pretty much identical and both FSs have exactly the same on configuration.
  Any suggestion would be appreciated.  Thank you.

If you could explain how you are doing the load balancing it would be
really helpful to me. I am trying to do the same thing.

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Re: [Freeswitch-users] CLIP on FXS channels with mod_openzap

2009-12-09 Thread François Legal



I'm still working on this issue, and decided to take a look at the
openzap code. 

First, I figured out that the parameter name for callerid
is enable_callerid rather than enable-callerid. 

I also figured out that
this parameter defaults to TRUE (which is coherent with the observed
behaviour on my FXO span) 

By further checking the code, I figured out
that presenting the callerid on an FXS port might not be implemented yet. I
could see the code for retrieving the callerid from FXO but nothing to send
it. 

Is my asumption (feature not implemented) correct ? 

François 

On
Wed, 2 Dec 2009 13:17:55 -0600, Anthony Minessale  wrote:  

Did you also
update your wanpipe drivers and rebuild openzap again after you upgraded
it?

 On Wed, Dec 2, 2009 at 2:12 AM, François Legal  wrote:

So I did some
tests and still I can not see CLIP on a phone connected to an FXS port.
Whether the call is bridged from SIP UA or from an incoming call on FXO
port does not change anything. Whether the parameter enable-caller-id=true
is present or not in openzap.conf.xml does not change anything too. 

On
that subject, sangoma support team says it must be freeswitch as this
feature is supported and has been tested working. 

However, the good point
is that I did not experience cuts in my call bridged from FXS to FXO with
that new release. 

François

On Tue, 1 Dec 2009 19:02:11 -0600,
Anthony Minessale  wrote:  

upgrading always helps *something* not sure.
but that is where we have to start because we have changed that code
alot.

 On Tue, Dec 1, 2009 at 2:37 AM, François Legal  wrote:

Sure, I'll
try that. I'm just building freeswitch-snapshot that I downloaded from
files.freeswitch.org [4] 

I also experience, when bridging a call from an
FXS to FXO the call is cut after a random time (this does not appear when
bridging SIP to FXO). Might this upgrade fix this problem also ? 

François
  

On Mon, 30 Nov 2009 10:48:26 -0600, Anthony Minessale wrote:  

can
you test svn trunk or latest pre release of 1.0.5

 On Mon, Nov 30, 2009 at
9:36 AM, François Legal  wrote:

Hello, 

I'm using Freeswitch with a
Sangoma A400 card, and I'm having CLIP problems on the FXS ports. 

When I
ring on FXS ports, the connected phone does not display
callerid/callerid-name. 

I tried turning the stuff of in openzap.conf.xml
() but it did not help. 

As a side note, turning this on on the FXO ports
drops the callerid information on incoming calls. 

Running freeswitch
1.0.4 on linux 2.6.27. 

François

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-- 
Anthony Minessale II

FreeSWITCH
http://www.freeswitch.org/ [10]
ClueCon http://www.cluecon.com/ [11]

Twitter: http://twitter.com/FreeSWITCH_wire [12]

AIM:
anthm
MSN:anthony_miness...@hotmail.com [13]

GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com [14]
 IRC: irc.freenode.net
[15] #freeswitch

 FreeSWITCH Developer
Conference
sip:8...@conference.freeswitch.org
[16]
iax:gu...@conference.freeswitch.org/888
[17]
googletalk:conf+...@conference.freeswitch.org
[18]
pstn:213-799-1400

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[19]
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UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
[21]
http://www.freeswitch.org [22]

-- 
Anthony Minessale II

FreeSWITCH
http://www.freeswitch.org/ [23]
ClueCon http://www.cluecon.com/ [24]

Twitter: http://twitter.com/FreeSWITCH_wire [25]

AIM:
anthm
MSN:anthony_miness...@hotmail.com [26]

GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com [27]
 IRC: irc.freenode.net
[28] #freeswitch

 FreeSWITCH Developer
Conference
sip:8...@conference.freeswitch.org
[29]
iax:gu...@conference.freeswitch.org/888
[30]
googletalk:conf+...@conference.freeswitch.org
[31]
pstn:213-799-1400

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[32]
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[34]
http://www.freeswitch.org [35]

-- 
Anthony Minessale II

FreeSWITCH
http://www.freeswitch.org/ [36]
ClueCon http://www.cluecon.com/ [37]

Twitter: http://twitter.com/FreeSWITCH_wire [38]

AIM:
anthm
MSN:anthony_miness...@hotmail.com
[39]
GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com [40]
 IRC:
irc.freenode.net [41] #freeswitch

FreeSWITCH Developer
Conference
sip:8...@conference.freeswitch.org
[42]
iax:gu...@conference.freeswitch.org/888
[43]
googletalk:conf+...@conference.freeswitch.org [44]
pstn:213-799-1400




Links:
--
[1] mailto:de...@thom.fr.eu.org
[2]
mailto:anthony.miness...@gmail.com
[3] mailto:de...@thom.fr.eu.org
[4]

[Freeswitch-users] FS Rocks!!!!!!!!!

Thought I'd send this little hurrah!  As there seems to have been a lot
of negativity on this list lately.

 

From my point of view, having looked at many solutions out there, FS is
still number one with regards to flexibility and performance.  I cannot
imagine doing what I'm using FS for, with any other product.  Yes it's
frustrating at times, but this is largely down to a lack
documentation/samples.  

 

So, if you have a solution to a problem, share it by adding an entry on
the WIKI.

 

Kudos to AM and all the other dev's, as someone said once 'Don't let the
bastards grind you down'

 

Regards,

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Re: [Freeswitch-users] no hang-up on B leg

No doubt, but that's a little difficult as this only happens
occasionally and I have 200 calls going on at the time.  It's needle in
the haystack stuff.

 

Here's what I know.

 

I have an external process listening for DTMF events.  If I detect '*' I
do a kill uuid on the B leg.  On a number of occasions I get an error
saying the B leg doesn't exist, so I now do a double kill on the
associated leg which I get from the event.  I do not get a 'doesn't
exist' message for the A leg, which leads me to believe that process of
tearing down both bridged legs is flawed.

 

The kluge clears the B leg hang issue, so the pressure's off for me, but
when I get a few nano seconds, I'll look at the code to see if there's
anything obvious.

 

Can anyone give me a hint on what module handles bridged calls? (sorry,
being lazy and suffering from a lack of sleep)

 

Regards,

 



From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Michael Jerris
Sent: 08 December 2009 16:16
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] no hangup on B leg

 

We will really need debug logs and sip traces to be able to figure out
what exactly is going on here.

 

Mike

 

On Dec 7, 2009, at 4:06 PM, Nik Middleton wrote:





Sorry no, apart from the fact that I was seeing the hangup.

 

 

I'm wondering if this a bandwidth congestion issue.  Is there anyway on
a bridged call I could trap on dtmf like look for '*' and force a
hangup?  I don't seem to able to see this tone on the B leg though.

 

Regards,



From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Michael Collins
Sent: 07 December 2009 19:12
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] no hangup on B leg

 

 

On Mon, Dec 7, 2009 at 11:01 AM, Nik Middleton
nik.middle...@noblesolutions.co.uk wrote:

Hi all,

 

I'll slowly pulling my hair out on this one.  I had FS successfully
hanging up both legs on a bridge, now today, with nothing changed, I'm
not seeing a hangup of the b leg at all.

 

FS is behind a PIX, so it might be a weird NAT issue, but A leg calls
hangup just fine.  Before when I had an issue with the B leg not closing
the bridge, I was at least getting a hangup event, now it's not being
fired.  Does anyone have an idea what might be causing this?

 

Regards,

 

Time for SIP traces and debug logs. Also, do you have any logs from when
things seemed to be working so that you can compare?
-MC

 

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Re: [Freeswitch-users] [OpenZAP] Does Dahdi (ex-Zaptel) require Asterisk?

2009-12-09 Thread Fred-145



Michael Jerris wrote:
 Our plan for 1.0.5 is that we will also have rpm and deb packages for many
 distros on our own repo.  Stay tuned.  This has been another major reason
 for the delay in 1.0.5.

Great news. I also prefer to use packages whenever possible, so as to know
what software is installed in a host, and have the package manager handle
conflicts and missing dependencies.

-- 
View this message in context: 
http://old.nabble.com/-OpenZAP--Does-Dahdi-%28ex-Zaptel%29-require-Asterisk--tp26694069p26708848.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.


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Re: [Freeswitch-users] CLIP on FXS channels with mod_openzap

2009-12-09 Thread Michael Jerris

I recall implementing that back when we released openzap, it should be in there 
unless someone chopped it out for some reason.  Look for 
zap_channel_send_fsk_data

Mike

On Dec 9, 2009, at 6:01 AM, François Legal wrote:

 I'm still working on this issue, and decided to take a look at the openzap 
 code.
 
 First, I figured out that the parameter name for callerid is enable_callerid 
 rather than enable-callerid.
 
 I also figured out that this parameter defaults to TRUE (which is coherent 
 with the observed behaviour on my FXO span)
 
  
 By further checking the code, I figured out that presenting the callerid on 
 an FXS port might not be implemented yet. I could see the code for retrieving 
 the callerid from FXO but nothing to send it.
 
  
 Is my asumption (feature not implemented) correct ?
 
  
 François
 
  
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Re: [Freeswitch-users] OT: Spa2102 and call transfer

2009-12-09 Thread Jonas Gauffin

I have the same problem with a HandyTone 502 adapter.

Anyone got any hints to get the flash button to work?

On Wed, Dec 9, 2009 at 11:25 AM, Jonas Gauffin jonas.gauf...@gmail.comwrote:

 Hello,

 I can't get call transfer to work with a SPA2102 adapter.
 I don't think it has something to do with FS, but I'm hoping someone here
 could help me.
 I do not get a new line in the phone (by pressing the R button), all DTMF
 tones are sent as audio to the other connected phone.

 Anyone got it working?

 Thanks,
   Jonas

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Re: [Freeswitch-users] no hang-up on B leg

2009-12-09 Thread Michael Jerris

src/switch_ivr_bridge.c

This could just as well be a glare condition when the call is in process of 
tearing down.

Mike


On Dec 8, 2009, at 6:48 PM, Nik Middleton wrote:

 No doubt, but that’s a little difficult as this only happens occasionally and 
 I have 200 calls going on at the time.  It’s needle in the haystack stuff.
  
 Here’s what I know.
  
 I have an external process listening for DTMF events.  If I detect ‘*’ I do a 
 kill uuid on the B leg.  On a number of occasions I get an error saying the B 
 leg doesn’t exist, so I now do a double kill on the associated leg which I 
 get from the event.  I do not get a ‘doesn’t exist’ message for the A leg, 
 which leads me to believe that process of tearing down both bridged legs is 
 flawed.
  
 The kluge clears the B leg hang issue, so the pressure’s off for me, but when 
 I get a few nano seconds, I’ll look at the code to see if there’s anything 
 obvious.
  
 Can anyone give me a hint on what module handles bridged calls? (sorry, being 
 lazy and suffering from a lack of sleep)
  
 Regards,
  
 From: freeswitch-users-boun...@lists.freeswitch.org 
 [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Michael 
 Jerris
 Sent: 08 December 2009 16:16
 To: freeswitch-users@lists.freeswitch.org
 Subject: Re: [Freeswitch-users] no hangup on B leg
  
 We will really need debug logs and sip traces to be able to figure out what 
 exactly is going on here.
  
 Mike
  
 On Dec 7, 2009, at 4:06 PM, Nik Middleton wrote:
 
 
 Sorry no, apart from the fact that I was seeing the hangup.
  
  
 I’m wondering if this a bandwidth congestion issue.  Is there anyway on a 
 bridged call I could trap on dtmf like look for ‘*’ and force a hangup?  I 
 don’t seem to able to see this tone on the B leg though.
  
 Regards,
 From: freeswitch-users-boun...@lists.freeswitch.org 
 [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Michael 
 Collins
 Sent: 07 December 2009 19:12
 To: freeswitch-users@lists.freeswitch.org
 Subject: Re: [Freeswitch-users] no hangup on B leg
  
  
 
 On Mon, Dec 7, 2009 at 11:01 AM, Nik Middleton 
 nik.middle...@noblesolutions.co.uk wrote:
 Hi all,
  
 I’ll slowly pulling my hair out on this one.  I had FS successfully hanging 
 up both legs on a bridge, now today, with nothing changed, I’m not seeing a 
 hangup of the b leg at all.
  
 FS is behind a PIX, so it might be a weird NAT issue, but A leg calls hangup 
 just fine.  Before when I had an issue with the B leg not closing the bridge, 
 I was at least getting a hangup event, now it’s not being fired.  Does anyone 
 have an idea what might be causing this?
  
 Regards,
  
 Time for SIP traces and debug logs. Also, do you have any logs from when 
 things seemed to be working so that you can compare?
 -MC
  
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Re: [Freeswitch-users] Svar: Re: pfSense with Freeswitch - so far so good, but no calls going through

2009-12-09 Thread Michael Gende

Hey There,

I came in not seeing any former posts of yours, so if this one is unhelpful,
just delete.

I did my FS install using PFsense as well. Its been working famously for a
few months now. Very happy.

I wrote what I discovered for FS on PFsense in this wiki:
http://wiki.freeswitch.org/wiki/Multi_home_tutorial

I assume since you're using PFsense that your computer is functioning as a
firewall AND a phone system (a dual homed host, in other words). That's what
the wiki attempt above is aimed at.

If you follow those instructions, you'll send and receive calls (as we were
and are able to). It is working for us, at least.

One problem in your case: I didn't really like using the PFsense Web
interface when configuring FS (except for installing FS and setting some
system parameters. Its great for PFsense, though).

It helped me more to get in with ssh and vi and make FS work. Having done
that successfully, you'll be more likely to effectively use the PFsense web
interface for FS, as it's really just a short cut for someone that
understands the FS file system, in my opinion.

Good luck,

Mike G.

On Tue, Dec 8, 2009 at 1:20 PM, mailinglist mailingl...@fribert.dk wrote:

Hi All

Ok, after reading a bit more I think I see what I've done wrong, but I
don't know how to fix it properly.
Looking in the Dialplan directory I see the following:
default (dir)
default.xml
features.xml
public (dir)
public.xml

Under the default dir the webinterface has created the 001_musimi.dk.xml
file that I've created.
But as I understand it, it doesn't use it.

How do I make it use it, I would very much like to keep the webinterface
editor, and not have to do it via ssh and vi all the time.

08-12-2009 kl. 07:05 skrev mailinglist mailingl...@fribert.dk i
meddelelsen 4b1dfabc02e10...@mail.fribert.dk:
Hi Mark

Ok, thanks.
Yes I have a gateway placed in external called musimi.dk (or should it be
in public?), and I'll just create the empty XML's in lan to get rid of that
error.

I'll remove the second part of the dialplan, my idea was that it was needed
for calls between sip phones hooked up to the freeswitch.

Now the remaining problem:
When I call ext 1002 from ext 1001 I see this message and get an error, the
same goes for dialing 0 to get an external number:

2009-12-08 07:02:20.522883 [NOTICE] switch_channel.c:602 New Channel
sofia/internal/1...@10.11.12.25 [b2b1253f-bfe3-de11-af59-000c29b7b4cb]
2009-12-08 07:02:20.522883 [INFO] mod_dialplan_xml.c:252 Processing
1001-1002 in context default
2009-12-08 07:02:20.522883 [NOTICE] switch_channel.c:602 New Channel
sofia/external/$1 [199e263f-bfe3-de11-af59-000c29b7b4cb]
2009-12-08 07:02:20.642371 [NOTICE] sofia.c:3770 Hangup sofia/external/$1
[CS_CONSUME_MEDIA] [NO_ROUTE_DESTINATION]
2009-12-08 07:02:20.642371 [INFO] mod_dptools.c:2091 Originate Failed.
Cause: NO_ROUTE_DESTINATION
2009-12-08 07:02:20.642371 [NOTICE] mod_dptools.c:2123 Hangup
sofia/internal/1...@10.11.12.25 [CS_EXECUTE] [NO_ROUTE_DESTINATION]
2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1085 Session 2
(sofia/external/$1) Ended
2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1087 Close
Channel sofia/external/$1 [CS_DESTROY]
2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1085 Session 1 (
sofia/internal/1...@10.11.12.25) Ended
2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1087 Close
Channel sofia/internal/1...@10.11.12.25 [CS_DESTROY]
I don't see any mention of the statements in the Dialplan, so for me it
looks like it haven't registered the Dialplan?

Best regards
Kenneth

08-12-2009 kl. 03:05 skrev Mark Crane mc...@yahoo.com i meddelelsen
659603.29094...@web56408.mail.re3.yahoo.com:

Question --
If I do a reloadxml it gives me this output on the console:
freeswi...@firewall.fribert.dkhttp://us.mc564.mail.yahoo.com/mc/compose?to=freeswi...@firewall.fribert.dk
reloadxml
2009-12-07 21:21:40.381445 [ERR] switch_xml.c:1282 Couldnt open
/usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No
such file or directory)
Error including
/usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No
such file or directory)

I'm not sure if it's a genuine problem,as I can see it, it just complains
that I haven't created any sip_profiles in /lan, but is that necessary?

Response: --
This isn't really a problem. To get rid of the error simply put a blank xml
file into each folder as in the internal and external directories. Dump the
lan directory and lan profile as mentioned earlier.

Question --

Extension Name musimi.dk
Enabled true
Order 001
Description ...

condition ^0(.\d+)$
action bridge sofia/gateway/musimi.dk/$1

Response: --

This is correct as long as you have a gateway that is

Re: [Freeswitch-users] no hang-up on B leg

I would have tended to agree with the glare, however, before I killed
both sides, I was back to my issue of the call not clearing down at all.
(rtp timeout eventually does it)

 

Thanks for the pointer to the source.

 

Regards,

 



From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Michael Jerris
Sent: 09 December 2009 14:01
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] no hang-up on B leg

 

src/switch_ivr_bridge.c

 

This could just as well be a glare condition when the call is in process
of tearing down.

 

Mike

 

 

On Dec 8, 2009, at 6:48 PM, Nik Middleton wrote:





No doubt, but that's a little difficult as this only happens
occasionally and I have 200 calls going on at the time.  It's needle in
the haystack stuff.

 

Here's what I know.

 

I have an external process listening for DTMF events.  If I detect '*' I
do a kill uuid on the B leg.  On a number of occasions I get an error
saying the B leg doesn't exist, so I now do a double kill on the
associated leg which I get from the event.  I do not get a 'doesn't
exist' message for the A leg, which leads me to believe that process of
tearing down both bridged legs is flawed.

 

The kluge clears the B leg hang issue, so the pressure's off for me, but
when I get a few nano seconds, I'll look at the code to see if there's
anything obvious.

 

Can anyone give me a hint on what module handles bridged calls? (sorry,
being lazy and suffering from a lack of sleep)

 

Regards,

 



From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Michael Jerris
Sent: 08 December 2009 16:16
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] no hangup on B leg

 

We will really need debug logs and sip traces to be able to figure out
what exactly is going on here.

 

Mike

 

On Dec 7, 2009, at 4:06 PM, Nik Middleton wrote:






Sorry no, apart from the fact that I was seeing the hangup.

 

 

I'm wondering if this a bandwidth congestion issue.  Is there anyway on
a bridged call I could trap on dtmf like look for '*' and force a
hangup?  I don't seem to able to see this tone on the B leg though.

 

Regards,



From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Michael Collins
Sent: 07 December 2009 19:12
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] no hangup on B leg

 

 

On Mon, Dec 7, 2009 at 11:01 AM, Nik Middleton
nik.middle...@noblesolutions.co.uk wrote:

Hi all,

 

I'll slowly pulling my hair out on this one.  I had FS successfully
hanging up both legs on a bridge, now today, with nothing changed, I'm
not seeing a hangup of the b leg at all.

 

FS is behind a PIX, so it might be a weird NAT issue, but A leg calls
hangup just fine.  Before when I had an issue with the B leg not closing
the bridge, I was at least getting a hangup event, now it's not being
fired.  Does anyone have an idea what might be causing this?

 

Regards,

 

Time for SIP traces and debug logs. Also, do you have any logs from when
things seemed to be working so that you can compare?
-MC

 

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Re: [Freeswitch-users] Wrong RFC2833 in SDP on NEC phones

2009-12-09 Thread Fernando Testa

It worked! 
Tnx!

Em 08/12/2009, às 16:51, Brian West escreveu:

 Best option for you is to use 96 in the sofia profile you're using to  
 talk to these broken devices.
 
 /b
 
 On Dec 8, 2009, at 12:41 PM, Fernando Gregianin Testa wrote:
 
 Dear list,
 
 Some Nec phones sends DTMF RFC2833 with payload 101 during the call,  
 but have negotiated a different one on SDP.
 When integrating with pabx NEC SV8500 and using phones DT700ITL32D-1  
 we notice this phone sends the following INVITE packet and RTP  
 packets: http://pastebin.freeswitch.org/11433
 Whole wireshark capture file is on 
 http://gregianin.org/teste_voice_rfc2833.pcap
 
 Is there any parameter to tweak FS in such a way to force understand  
 101 packets as DTMF?
 Thank you in advance!
 
 Fernando Testa
 PS: On pcap you have the following IPs:
 FS at 10.91.10.210
 Nec Pbx 10.91.10.22
 phone 10.91.10.85
 
 
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[Freeswitch-users] controlling calls handled within a fifo using event_socket

2009-12-09 Thread Luke Graybill

In my FreeSWITCH environment, calls are originated out to customers who are
placed into a fifo upon answer. There are members (x-lite endpoints) in this
fifo who handle those customer calls. I am writing a monitoring application
that uses event_socket to watch the channels involved in this process,
ultimately displaying an interface for each rep that allows them to
interactively drive the calls (playback audio conditionally to the customer,
save information obtained during the call to another database, etc).

Problems arise when attempting to identify which customer channel is
speaking to which rep (consumer) channel. My event_socket application is
inspecting the CHANNEL_ANSWER event, but this event does not appear to
contain enough information to make this determination.

I have identified three distinct uuid values in the CHANNEL_ANSWER headers
on the consumer channel: core uuid, the uuid of the consumer channel, and
another uuid which is not the customer uuid (I'm assuming this is the uuid
of the fifo).

According to the wiki
herehttp://wiki.freeswitch.org/wiki/Mod_fifo#Additional_variables_.28not_yet_documented.29,
I expected the consumer CHANNEL_ANSWER headers to contain variables such as
`fifo_target` with the uuid of the customer channel it is bridged to, but
this variable is not in the headers. Indeed, no channel variables are set
which correspond to the uuid of the customer channel to which the rep is
speaking. After the call has been completed, data posted in the cdr does in
fact contain the `fifo_target` information, but this does not help me during
the call.

The short version of my question is this: how do I programmatically
determine which channel uuid the consumer channel in a fifo is connected to?

Any help here would be greatly appreciated :) Thanks!
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Re: [Freeswitch-users] controlling calls handled within a fifo using event_socket

fifo list issue this API and get the fifo XML and get the caller's  
uuid out of the list.

/b

On Dec 9, 2009, at 10:50 AM, Luke Graybill wrote:

 The short version of my question is this: how do I programmatically  
 determine which channel uuid the consumer channel in a fifo is  
 connected to?


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[Freeswitch-users] Multiple registration: Registration1 cannot call Registration2 in attended_transfer

2009-12-09 Thread Peter P GMX

Hello,

in our dialplan we have enabled multiple-registrations, so 2 phones can
register on a single directory entry.
param name=multiple-registrations value=true/
Both phones are registered, both phones can be called and each phone can
call the other phone.
However in an attended_transfer mode calls cannot be transferred to the
other phone with the same number.
Attended_transfer in this case is needed when you take a call on your
main SIP phone and and then want to transfer it to your mobile DECT/SIP
phone, because you may have to check something in another room.
I did a SIP trace and see the following:

* A invites B(phone 1) = ok
* B(phone 1) places call on hold = ok
* B(phone 1) dials number B(phone 2 DECT) on second line
* Freeswitch send Invite to B(phone 1) = ok
* Freeswitch send Invite to B(phone 2 DECT)
* B(phone 2 DECT) sends Ringing to Freeswitch = ok
* B(phone 1) sends Busy to Freeswitch
* B(phone 1) displays Busy and hangs up the second line

Is there any way to overcome this? Is there a way to ignore the Busy
from phone 1 when phone 2 answers Ringing?


Best regards
Peter

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Re: [Freeswitch-users] esl for Mac OS X 10.4

2009-12-09 Thread EdPimentl

Regarding Mac OSX 10.5/6 can you point me where the latest FS binary file
is?

Thanks in advance,
-E
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Re: [Freeswitch-users] FS Rocks!!!!!!!!!

2009-12-09 Thread Michael Collins

On Tue, Dec 8, 2009 at 3:59 PM, Nik Middleton 
nik.middle...@noblesolutions.co.uk wrote:

  Thought I’d send this little hurrah!  As there seems to have been a lot
 of negativity on this list lately.



I hereby multiply all the negative comments by -1. :P
-MC
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Re: [Freeswitch-users] Multiple registration: Registration1 cannot call Registration2 in attended_transfer

2009-12-09 Thread João Mesquita

That is more dependent on the endpoint than on the switch itself. I guess
you can always use mod_limit to come up with some crazy key to identify one
endpoint or the other but still it seems overly complicated for something
that is not supposed to be working this way.

You can also park the call instead of transferring, can't ya?

JM

On Wed, Dec 9, 2009 at 3:13 PM, Peter P GMX prometheus...@gmx.net wrote:

 Hello,

 in our dialplan we have enabled multiple-registrations, so 2 phones can
 register on a single directory entry.
param name=multiple-registrations value=true/
 Both phones are registered, both phones can be called and each phone can
 call the other phone.
 However in an attended_transfer mode calls cannot be transferred to the
 other phone with the same number.
 Attended_transfer in this case is needed when you take a call on your
 main SIP phone and and then want to transfer it to your mobile DECT/SIP
 phone, because you may have to check something in another room.
 I did a SIP trace and see the following:

* A invites B(phone 1) = ok
* B(phone 1) places call on hold = ok
* B(phone 1) dials number B(phone 2 DECT) on second line
* Freeswitch send Invite to B(phone 1) = ok
* Freeswitch send Invite to B(phone 2 DECT)
* B(phone 2 DECT) sends Ringing to Freeswitch = ok
* B(phone 1) sends Busy to Freeswitch
* B(phone 1) displays Busy and hangs up the second line

 Is there any way to overcome this? Is there a way to ignore the Busy
 from phone 1 when phone 2 answers Ringing?


 Best regards
 Peter

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-- 
João Mesquita
FreeSWITCH™ Solutions
t: +1 (646) 4959927
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Re: [Freeswitch-users] FS Rocks!!!!!!!!!

2009-12-09 Thread Fred-145



Nik Middleton wrote:
 I cannot imagine doing what I'm using FS for, with any other product.  Yes
 it's frustrating at times, but this is largely down to a lack
 documentation/samples. 

Speaking of which... would this layout be good for a book on Freeswitch?

Preface
1. VoIP, Freeswitch, FS vs. Asterisk, softswitch vs. PBX, etc.
2. Choosing hardware options (server, phones, gateways)
3. Setting up FS
4. Configuring FS (SIP, profiles/contexts, VoIP providers, SIP/POTS
gateways, etc.)
5. Administering FS (CLI and GUI)
6. Customizing dialplan (adding SIP accounts, voice-mail, etc.)
7. Performance, sound quality, other issues
8. Writing scripts (LUA, etc.), connecting to databases
9. Real-life examples (Gino's Pizza, etc.)
Conclusion
Index
-- 
View this message in context: 
http://old.nabble.com/FS-Rocks%21%21%21%21%21%21%21%21%21-tp26708755p26716612.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.


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Re: [Freeswitch-users] FS Rocks!!!!!!!!!

Looks good, but you've missed out billing and the key one, the event
socket which could be a chapter in it's self.

Do you have a publisher for it yet?

Regards

-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Fred-145
Sent: 09 December 2009 19:55
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] FS Rocks!



Nik Middleton wrote:
 I cannot imagine doing what I'm using FS for, with any other product.
Yes
 it's frustrating at times, but this is largely down to a lack
 documentation/samples. 

Speaking of which... would this layout be good for a book on Freeswitch?

Preface
1. VoIP, Freeswitch, FS vs. Asterisk, softswitch vs. PBX, etc.
2. Choosing hardware options (server, phones, gateways)
3. Setting up FS
4. Configuring FS (SIP, profiles/contexts, VoIP providers, SIP/POTS
gateways, etc.)
5. Administering FS (CLI and GUI)
6. Customizing dialplan (adding SIP accounts, voice-mail, etc.)
7. Performance, sound quality, other issues
8. Writing scripts (LUA, etc.), connecting to databases
9. Real-life examples (Gino's Pizza, etc.)
Conclusion
Index
-- 
View this message in context:
http://old.nabble.com/FS-Rocks%21%21%21%21%21%21%21%21%21-tp26708755p267
16612.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.


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Re: [Freeswitch-users] Svar: Re: pfSense with Freeswitch - so far so good, but no calls going through

2009-12-09 Thread mailinglist

Yes, I have two extensions.
I can even make them join a group, and if I call the group, the two extensions 
will ring.
 


 08-12-2009 kl. 23:45 skrev Nandy Dagondon nandy1...@gmail.com i 
 meddelelsen 7d0bfd8c0912081445v124dd6cs9174a201eb109...@mail.gmail.com:

have you created Extension 1002? 
-nandy


On Wed, Dec 9, 2009 at 3:20 AM, mailinglist mailingl...@fribert.dk wrote:


Hi All
Ok, after reading a bit more I think I see what I've done wrong, but I don't 
know how to fix it properly.
Looking in the Dialplan directory I see the following:
default (dir)
default.xml
features.xml
public (dir)
public.xml
Under the default dir the webinterface has created the 001_musimi.dk.xml file 
that I've created.
But as I understand it, it doesn't use it.

How do I make it use it, I would very much like to keep the webinterface 
editor, and not have to do it via ssh and vi all the time.

 08-12-2009 kl. 07:05 skrev mailinglist mailingl...@fribert.dk i 
 meddelelsen 4b1dfabc02e10...@mail.fribert.dk:

Hi Mark
Ok, thanks.
Yes I have a gateway placed in external called musimi.dk (or should it be in 
public?), and I'll just create the empty XML's in lan to get rid of that error.
I'll remove the second part of the dialplan, my idea was that it was needed for 
calls between sip phones hooked up to the freeswitch.
Now the remaining problem:
When I call ext 1002 from ext 1001 I see this message and get an error, the 
same goes for dialing 0 to get an external number:
2009-12-08 07:02:20.522883 [NOTICE] switch_channel.c:602 New Channel 
sofia/internal/1...@10.11.12.25 [b2b1253f-bfe3-de11-af59-000c29b7b4cb]
2009-12-08 07:02:20.522883 [INFO] mod_dialplan_xml.c:252 Processing 1001-1002 
in context default
2009-12-08 07:02:20.522883 [NOTICE] switch_channel.c:602 New Channel 
sofia/external/$1 [199e263f-bfe3-de11-af59-000c29b7b4cb]
2009-12-08 07:02:20.642371 [NOTICE] sofia.c:3770 Hangup sofia/external/$1 
[CS_CONSUME_MEDIA] [NO_ROUTE_DESTINATION]
2009-12-08 07:02:20.642371 [INFO] mod_dptools.c:2091 Originate Failed. Cause: 
NO_ROUTE_DESTINATION
2009-12-08 07:02:20.642371 [NOTICE] mod_dptools.c:2123 Hangup 
sofia/internal/1...@10.11.12.25 [CS_EXECUTE] [NO_ROUTE_DESTINATION]
2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1085 Session 2 
(sofia/external/$1) Ended
2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1087 Close Channel 
sofia/external/$1 [CS_DESTROY]
2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1085 Session 1 
(sofia/internal/1...@10.11.12.25) Ended
2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1087 Close Channel 
sofia/internal/1...@10.11.12.25 [CS_DESTROY]
I don't see any mention of the statements in the Dialplan, so for me it looks 
like it haven't registered the Dialplan?
Best regards
Kenneth

 08-12-2009 kl. 03:05 skrev Mark Crane mc...@yahoo.com i meddelelsen 
 659603.29094...@web56408.mail.re3.yahoo.com:



Question --
If I do a reloadxml it gives me this output on the console:
freeswi...@firewall.fribert.dk ( 
http://us.mc564.mail.yahoo.com/mc/compose?to=freeswi...@firewall.fribert.dk ) 
reloadxml
2009-12-07 21:21:40.381445 [ERR] switch_xml.c:1282 Couldnt open 
/usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No such 
file or directory)
Error including 
/usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No such 
file or directory)

I'm not sure if it's a genuine problem,as I can see it, it just complains that 
I haven't created any sip_profiles in /lan, but is that necessary?

Response: --
This isn't really a problem. To get rid of the error simply put a blank xml 
file into each folder as in the internal and external directories. Dump the lan 
directory and lan profile as mentioned earlier.

Question --

Extension Name musimi.dk
Enabled true
Order 001
Description ...
condition ^0(.\d+)$
action bridge sofia/gateway/musimi.dk/$1

Response: --

This is correct as long as you have a gateway that is registered called 
musimi.dk

Question --
Extension Name 10.11.12.25
Enabled true
Order 002
Description ...
action bridge sofia/internal/$

Response: --

No idea what this is for its not needed as far as I can tell.


Now please summarize what you still need help on.


Mark J Crane
http://fusionpbx.com
pfSense FreeSWITCH package developer


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Re: [Freeswitch-users] Svar: Re: pfSense with Freeswitch - so far so good, but no calls going through

2009-12-09 Thread mailinglist

This is a new install, but it's grabbed from a pfSense repository.


 09-12-2009 kl. 10:28 skrev Mark Crane mc...@yahoo.com i meddelelsen 
 187489.95329...@web56408.mail.re3.yahoo.com:


Is this a new install of the FreeSWITCH package or is it an upgrade from and 
earlier package?

Mark J Crane
mc...@yahoo.com

--- On Tue, 12/8/09, Nandy Dagondon nandy1...@gmail.com wrote:



From: Nandy Dagondon nandy1...@gmail.com
Subject: Re: [Freeswitch-users] Svar: Re: pfSense with Freeswitch - so far so 
good, but no calls going through
To: freeswitch-users@lists.freeswitch.org
Date: Tuesday, December 8, 2009, 3:45 PM

have you created Extension 1002? 
-nandy


On Wed, Dec 9, 2009 at 3:20 AM, mailinglist mailingl...@fribert.dk ( 
/mc/compose?to=mailingl...@fribert.dk ) wrote:


Hi All
 
Ok, after reading a bit more I think I see what I've done wrong, but I don't 
know how to fix it properly.
Looking in the Dialplan directory I see the following:
default (dir)
default.xml
features.xml
public (dir)
public.xml
 
Under the default dir the webinterface has created the 001_musimi.dk.xml file 
that I've created.
But as I understand it, it doesn't use it.

How do I make it use it, I would very much like to keep the webinterface 
editor, and not have to do it via ssh and vi all the time.

 08-12-2009 kl. 07:05 skrev mailinglist mailingl...@fribert.dk ( 
 /mc/compose?to=mailingl...@fribert.dk ) i meddelelsen 
 4b1dfabc02e10...@mail.fribert.dk ( 
 /mc/compose?to=4b1dfabc02e10...@mail.fribert.dk ):

Hi Mark
 
Ok, thanks.
Yes I have a gateway placed in external called musimi.dk (or should it be in 
public?), and I'll just create the empty XML's in lan to get rid of that error.
 
I'll remove the second part of the dialplan, my idea was that it was needed for 
calls between sip phones hooked up to the freeswitch.
 
Now the remaining problem:
When I call ext 1002 from ext 1001 I see this message and get an error, the 
same goes for dialing 0 to get an external number:
 
2009-12-08 07:02:20.522883 [NOTICE] switch_channel.c:602 New Channel 
sofia/internal/1...@10.11.12.25 ( 
/mc/compose?to=sofia/internal/1...@10.11.12.25 ) 
[b2b1253f-bfe3-de11-af59-000c29b7b4cb]
2009-12-08 07:02:20.522883 [INFO] mod_dialplan_xml.c:252 Processing 1001-1002 
in context default
2009-12-08 07:02:20.522883 [NOTICE] switch_channel.c:602 New Channel 
sofia/external/$1 [199e263f-bfe3-de11-af59-000c29b7b4cb]
2009-12-08 07:02:20.642371 [NOTICE] sofia.c:3770 Hangup sofia/external/$1 
[CS_CONSUME_MEDIA] [NO_ROUTE_DESTINATION]
2009-12-08 07:02:20.642371 [INFO] mod_dptools.c:2091 Originate Failed.  Cause: 
NO_ROUTE_DESTINATION
2009-12-08 07:02:20.642371 [NOTICE] mod_dptools.c:2123 Hangup 
sofia/internal/1...@10.11.12.25 ( 
/mc/compose?to=sofia/internal/1...@10.11.12.25 ) [CS_EXECUTE] 
[NO_ROUTE_DESTINATION]
2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1085 Session 2 
(sofia/external/$1) Ended
2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1087 Close Channel 
sofia/external/$1 [CS_DESTROY]
2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1085 Session 1 
(sofia/internal/1...@10.11.12.25 ( 
/mc/compose?to=sofia/internal/1...@10.11.12.25 )) Ended
2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1087 Close Channel 
sofia/internal/1...@10.11.12.25 ( 
/mc/compose?to=sofia/internal/1...@10.11.12.25 ) [CS_DESTROY]
I don't see any mention of the statements in the Dialplan, so for me it looks 
like it haven't registered the Dialplan?
 
Best regards
Kenneth

 08-12-2009 kl. 03:05 skrev Mark Crane mc...@yahoo.com ( 
 /mc/compose?to=mc...@yahoo.com ) i meddelelsen 
 659603.29094...@web56408.mail.re3.yahoo.com ( 
 /mc/compose?to=659603.29094...@web56408.mail.re3.yahoo.com ):



Question --
If I do a reloadxml it gives me this output on the console:
freeswi...@firewall.fribert.dk ( 
http://us.mc564.mail.yahoo.com/mc/compose?to=freeswi...@firewall.fribert.dk ) 
reloadxml
2009-12-07 21:21:40.381445 [ERR] switch_xml.c:1282 Couldnt open 
/usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No such 
file or directory)
Error including 
/usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No such 
file or directory)

I'm not sure if it's a genuine problem,as I can see it, it just complains that 
I haven't created any sip_profiles in /lan, but is that necessary?

Response: --
This isn't really a problem. To get rid of the error simply put a blank xml 
file into each folder as in the internal and external directories. Dump the lan 
directory and lan profile as mentioned earlier.

Question --

Extension Name  musimi.dk
Enabled true
Order 001
Description  ...
 
condition ^0(.\d+)$
action bridge sofia/gateway/musimi.dk/$1

Response: --

This is correct as long as you have a gateway that is registered called 
musimi.dk

Re: [Freeswitch-users] Svar: Re: pfSense with Freeswitch - so far so good, but no calls going through

2009-12-09 Thread mailinglist

Hi Michael
 
Thankyou for the excellent wiki article, yes, I did follow your guide there, 
all the way except to 'dialplan' and it seems that's the problem at the moment.
I would very much like to create the dialplan in the webinterface, and not in 
the public.xml file.
But at the moment it only uses the public.xml file :-(
 
Great writeup you made, and it has brought me a long way.
 
BR
Fribert


 09-12-2009 kl. 16:16 skrev Michael Gende mge...@gendesign.com i 
 meddelelsen a87d43fd0912090716m689f6690m105264b874f12...@mail.gmail.com:

Hey There,

I came in not seeing any former posts of yours, so if this one is unhelpful, 
just delete.

I did my FS install using PFsense as well. Its been working famously for a few 
months now. Very happy. 

I wrote what I discovered for FS on PFsense in this wiki: 
http://wiki.freeswitch.org/wiki/Multi_home_tutorial

I assume since you're using PFsense that your computer is functioning as a 
firewall AND a phone system (a dual homed host, in other words). That's what 
the wiki attempt above is aimed at.

If you follow those instructions, you'll send and receive calls (as we were and 
are able to). It is working for us, at least. 

One problem in your case: I didn't really like using the PFsense Web interface 
when configuring FS (except for installing FS and setting some system 
parameters. Its great for PFsense, though). 

It helped me more to get in with ssh and vi and make FS work. Having done that 
successfully, you'll be more likely to effectively use the PFsense web 
interface for FS, as it's really just a short cut for someone that 
understands the FS file system, in my opinion.

Good luck,

Mike G.

On Tue, Dec 8, 2009 at 1:20 PM, mailinglist mailingl...@fribert.dk wrote:


Hi All
Ok, after reading a bit more I think I see what I've done wrong, but I don't 
know how to fix it properly.
Looking in the Dialplan directory I see the following:
default (dir)
default.xml
features.xml
public (dir)
public.xml
Under the default dir the webinterface has created the 001_musimi.dk.xml file 
that I've created.
But as I understand it, it doesn't use it.

How do I make it use it, I would very much like to keep the webinterface 
editor, and not have to do it via ssh and vi all the time.

 08-12-2009 kl. 07:05 skrev mailinglist mailingl...@fribert.dk i 
 meddelelsen 4b1dfabc02e10...@mail.fribert.dk:

Hi Mark
Ok, thanks.
Yes I have a gateway placed in external called musimi.dk (or should it be in 
public?), and I'll just create the empty XML's in lan to get rid of that error.
I'll remove the second part of the dialplan, my idea was that it was needed for 
calls between sip phones hooked up to the freeswitch.
Now the remaining problem:
When I call ext 1002 from ext 1001 I see this message and get an error, the 
same goes for dialing 0 to get an external number:
2009-12-08 07:02:20.522883 [NOTICE] switch_channel.c:602 New Channel 
sofia/internal/1...@10.11.12.25 [b2b1253f-bfe3-de11-af59-000c29b7b4cb]
2009-12-08 07:02:20.522883 [INFO] mod_dialplan_xml.c:252 Processing 1001-1002 
in context default
2009-12-08 07:02:20.522883 [NOTICE] switch_channel.c:602 New Channel 
sofia/external/$1 [199e263f-bfe3-de11-af59-000c29b7b4cb]
2009-12-08 07:02:20.642371 [NOTICE] sofia.c:3770 Hangup sofia/external/$1 
[CS_CONSUME_MEDIA] [NO_ROUTE_DESTINATION]
2009-12-08 07:02:20.642371 [INFO] mod_dptools.c:2091 Originate Failed. Cause: 
NO_ROUTE_DESTINATION
2009-12-08 07:02:20.642371 [NOTICE] mod_dptools.c:2123 Hangup 
sofia/internal/1...@10.11.12.25 [CS_EXECUTE] [NO_ROUTE_DESTINATION]
2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1085 Session 2 
(sofia/external/$1) Ended
2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1087 Close Channel 
sofia/external/$1 [CS_DESTROY]
2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1085 Session 1 
(sofia/internal/1...@10.11.12.25) Ended
2009-12-08 07:02:20.663502 [NOTICE] switch_core_session.c:1087 Close Channel 
sofia/internal/1...@10.11.12.25 [CS_DESTROY]
I don't see any mention of the statements in the Dialplan, so for me it looks 
like it haven't registered the Dialplan?
Best regards
Kenneth

 08-12-2009 kl. 03:05 skrev Mark Crane mc...@yahoo.com i meddelelsen 
 659603.29094...@web56408.mail.re3.yahoo.com:



Question --
If I do a reloadxml it gives me this output on the console:
freeswi...@firewall.fribert.dk ( 
http://us.mc564.mail.yahoo.com/mc/compose?to=freeswi...@firewall.fribert.dk ) 
reloadxml
2009-12-07 21:21:40.381445 [ERR] switch_xml.c:1282 Couldnt open 
/usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No such 
file or directory)
Error including 
/usr/local/freeswitch/conf/autoload_configs/../sip_profiles/lan/*.xml (No such 
file or directory)

I'm not sure if it's a genuine problem,as I can see it, it just complains that 
I haven't created any sip_profiles in /lan, but is that necessary?

Response: --

Re: [Freeswitch-users] FS Rocks!!!!!!!!!


 Preface
 1. VoIP, Freeswitch, FS vs. Asterisk, softswitch vs. PBX, etc.
 2. Choosing hardware options (server, phones, gateways)
 3. Setting up FS
 4. Configuring FS (SIP, profiles/contexts, VoIP providers, SIP/POTS
 gateways, etc.)
 5. Administering FS (CLI and GUI)
 6. Customizing dialplan (adding SIP accounts, voice-mail, etc.)
 7. Performance, sound quality, other issues
 8. Writing scripts (LUA, etc.), connecting to databases
 9. Real-life examples (Gino's Pizza, etc.)
 Conclusion
 Index
 --

I found the rosetta stone useful though woefully lacking in volume.

I guess that's true overall with the project.

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Re: [Freeswitch-users] FS Rocks!!!!!!!!!

Visit the friday meetings and we can help if you document it.  ;)

/b

On Dec 9, 2009, at 3:56 PM, Tim Uckun wrote:

 I found the rosetta stone useful though woefully lacking in volume.

 I guess that's true overall with the project.


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Re: [Freeswitch-users] FS Rocks!!!!!!!!!

2009-12-09 Thread Michael Collins

 I found the rosetta stone useful though woefully lacking in volume.

 I guess that's true overall with the project.

 Documentation is neither easy nor glamorous. The woefully lacking
documentation has been provided by a little group of people who've done a
big bit of documenting and a big group of people who've done a little bit of
documenting. If ever there was an aspect of this project that could use more
volunteers it is documentation and bug testing.

If anyone wants to help on either of these fronts please email me off list.
-MC
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[Freeswitch-users] Even socket question.

Hey All. I am trying to get freeswitch to route to my socket handler
and am having a problem.

I am running freeswitch inside a virtualbox VM for testing purposes.
The vitualbox communicates with my host via the host only adapter.
The VM IP address is 192.168.56.3 and the laptop has the iP
192.168.56.1

I have set up both an outbound and an inbound socket handlers. The
inbound one works fine, the outbound is not working . The inbound
merely logs the event name. The outbound logs the connection and hangs
up.

I have set up an extension like this

include
  extension name=8084
condition field=destination_number expression=^8084$
  action application=set data=continue_on_fail=true / !--
we still need this to continue if bridging times out --
  action application=set data=call_timeout=5 /
  action application=socket data=192.168.56.1:8084 sync full/
/condition
  /extension
/include


When I dial 8084 I get a lot of events being logged but the oubound
never gets the calls and never logs the call.

I have added the fs_cli output below. It looks to me like it's sending
the output to the other IP address of my laptop instead of the one I
specified in my extension but I could just be misreading that.   I
have set the external IP of the freeswitch to the 56.3 address.

Here is the LSOF output

freeswitc 2468   root   31u IPv4   5785
TCP ubuntuvm01:5080 (LISTEN)
freeswitc 2468   root   33u IPv6   5791
TCP localhost:5060 (LISTEN)
freeswitc 2468   root   36u IPv4   5804
TCP 192.168.56.3:5060 (LISTEN)
freeswitc 2468   root   48u IPv4   5910
TCP 192.168.56.3:8021 (LISTEN)
freeswitc 2468   root   50u IPv4   5912
TCP *:8080 (LISTEN)


Here is the output from the fs_cli

2009-12-09 14:31:53.255579 [DEBUG] sofia.c:5224 0 acls to check for proxy
2009-12-09 14:31:53.255579 [DEBUG] sofia.c:5242 network ip is a proxy [0]
2009-12-09 14:31:53.255579 [DEBUG] sofia.c:5270 IP 192.168.56.1
Rejected by acl domains. Falling back to Digest auth.
2009-12-09 14:31:53.357865 [DEBUG] sofia.c:5224 0 acls to check for proxy
2009-12-09 14:31:53.357865 [DEBUG] sofia.c:5242 network ip is a proxy [0]
2009-12-09 14:31:53.357865 [DEBUG] sofia.c:5270 IP 192.168.56.1
Rejected by acl domains. Falling back to Digest auth.
2009-12-09 14:31:53.420949 [NOTICE] switch_channel.c:613 New Channel
sofia/internal/1...@192.168.56.3
[2fbcf6fe-b35e-4c40-92a6-9f21de3102fa]
2009-12-09 14:31:53.422090 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/1...@192.168.56.3) Running State Change CS_NEW
2009-12-09 14:31:53.422090 [DEBUG] switch_core_state_machine.c:320
(sofia/internal/1...@192.168.56.3) State NEW
2009-12-09 14:31:53.422090 [DEBUG] sofia.c:3727 Channel
sofia/internal/1...@192.168.56.3 entering state [received][100]
2009-12-09 14:31:53.422090 [DEBUG] sofia.c:3738 Remote SDP:
v=0
o=Z 0 0 IN IP4 218.101.6.157
s=Z
c=IN IP4 218.101.6.157
t=0 0
m=audio 8000 RTP/AVP 3 110 98 8 0 101
a=rtpmap:3 GSM/8000
a=rtpmap:110 speex/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=30
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

2009-12-09 14:31:53.422090 [DEBUG] sofia_glue.c:3305 Audio Codec
Compare [GSM:3:8000:20]/[G7221:115:32000:20]
2009-12-09 14:31:53.422090 [DEBUG] sofia_glue.c:3305 Audio Codec
Compare [GSM:3:8000:20]/[G7221:107:16000:20]
2009-12-09 14:31:53.422090 [DEBUG] sofia_glue.c:3305 Audio Codec
Compare [GSM:3:8000:20]/[G722:9:8000:20]
2009-12-09 14:31:53.422090 [DEBUG] sofia_glue.c:3305 Audio Codec
Compare [GSM:3:8000:20]/[PCMU:0:8000:20]
2009-12-09 14:31:53.422090 [DEBUG] sofia_glue.c:3305 Audio Codec
Compare [GSM:3:8000:20]/[PCMA:8:8000:20]
2009-12-09 14:31:53.422090 [DEBUG] sofia_glue.c:3305 Audio Codec
Compare [GSM:3:8000:20]/[GSM:3:8000:20]
2009-12-09 14:31:53.422090 [DEBUG] sofia_glue.c:2143 Set Codec
sofia/internal/1...@192.168.56.3 GSM/8000 20 ms 160 samples
2009-12-09 14:31:53.423898 [DEBUG] sofia_glue.c:3261 Set 2833 dtmf
payload to 101
2009-12-09 14:31:53.423898 [DEBUG] sofia.c:3885
(sofia/internal/1...@192.168.56.3) State Change CS_NEW - CS_INIT
2009-12-09 14:31:53.423898 [DEBUG] switch_core_session.c:999 Send
signal sofia/internal/1...@192.168.56.3 [BREAK]
2009-12-09 14:31:53.423898 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/1...@192.168.56.3) Running State Change CS_INIT
2009-12-09 14:31:53.423898 [DEBUG] switch_core_state_machine.c:338
(sofia/internal/1...@192.168.56.3) State INIT
2009-12-09 14:31:53.423898 [DEBUG] mod_sofia.c:83
sofia/internal/1...@192.168.56.3 SOFIA INIT
2009-12-09 14:31:53.423898 [DEBUG] mod_sofia.c:111
(sofia/internal/1...@192.168.56.3) State Change CS_INIT - CS_ROUTING
2009-12-09 14:31:53.423898 [DEBUG] switch_core_session.c:999 Send
signal sofia/internal/1...@192.168.56.3 [BREAK]
2009-12-09 14:31:53.423898 [DEBUG] switch_core_state_machine.c:338
(sofia/internal/1...@192.168.56.3) State INIT going to sleep
2009-12-09 14:31:53.423898 [DEBUG] switch_core_state_machine.c:314

Re: [Freeswitch-users] FS Rocks!!!!!!!!!

That is what is nice about our community I'm more than willing to  
answer the questions if you document them... as are many others in the  
core team...we just have a lot to do and I think the best repayment is  
documentation! ;)


/b

On Dec 9, 2009, at 4:39 PM, Tim Uckun wrote:

On Thu, Dec 10, 2009 at 11:07 AM, Brian West br...@freeswitch.org  
wrote:

Visit the friday meetings and we can help if you document it.  ;)



I would be willing to lend a hand with the documentation but I know so
little (a complete freeswitch noob). For example I was trying to
figure out how to tell if an extension was set up show dialplan in
asterisk.  I could not find this anywhere. If I find out I would be
happy to add it to the rosetta stone.

I am currently working on getting outbound socket working. Once I get
it going I would be happy to add it to the relevant section of the
wiki (in this case ruby).


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[Freeswitch-users] FreeSWITCH 1.0.4 Bug Reports...

Dear FreeSWITCHers,

As of Friday Dec. 11th we will NOT accept any more bug reports on  
1.0.4.  You need to be on a 1.0.5pre or SVN trunk.  1.0.4 is over 6  
months old and I really suspect your issues in 1.0.4 are already  
fixed.  We will release a new pre every monday morning till 1.0.5 is  
released please keep up to date if possible.

We are working hard to get 1.0.5 out and be as stable as possible and  
its more stable than 1.0.4... their might be some edge or corner cases  
that aren't accounted for so we need you to please download SVN trunk  
in your test labs and try it out... report issues and help us make the  
best FreeSWITCH release possible.

Thank you,
Brian West


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Re: [Freeswitch-users] Question regarding running FreeSWITCH with high priority enabled.

2009-12-09 Thread DJB

Load sharing feature is coming off our Lucent Telica switch.

From: Tim Uckun timuc...@gmail.com
To: freeswitch-users@lists.freeswitch.org
Sent: Wed, December 9, 2009 2:26:41 AM
Subject: Re: [Freeswitch-users] Question regarding running FreeSWITCH with high 
priority enabled.

On Tue, Dec 8, 2009 at 5:42 AM, DJB djbin...@yahoo.com wrote:
 One thing that I forgot to mention, these 2 FreeSWITCH servers are getting
 calls with load balancing from another switch.  Thus, the traffic type are
 pretty much identical and both FSs have exactly the same on configuration.
  Any suggestion would be appreciated.  Thank you.

If you could explain how you are doing the load balancing it would be
really helpful to me. I am trying to do the same thing.

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Re: [Freeswitch-users] Even socket question.

2009-12-09 Thread Anthony Minessale

do you have something listening on 8084 ?


On Wed, Dec 9, 2009 at 4:35 PM, Tim Uckun timuc...@gmail.com wrote:

 Hey All. I am trying to get freeswitch to route to my socket handler
 and am having a problem.

 I am running freeswitch inside a virtualbox VM for testing purposes.
 The vitualbox communicates with my host via the host only adapter.
 The VM IP address is 192.168.56.3 and the laptop has the iP
 192.168.56.1

 I have set up both an outbound and an inbound socket handlers. The
 inbound one works fine, the outbound is not working . The inbound
 merely logs the event name. The outbound logs the connection and hangs
 up.

 I have set up an extension like this

 include
  extension name=8084
condition field=destination_number expression=^8084$
  action application=set data=continue_on_fail=true / !--
 we still need this to continue if bridging times out --
  action application=set data=call_timeout=5 /
  action application=socket data=192.168.56.1:8084 sync full/
/condition
  /extension
 /include


 When I dial 8084 I get a lot of events being logged but the oubound
 never gets the calls and never logs the call.

 I have added the fs_cli output below. It looks to me like it's sending
 the output to the other IP address of my laptop instead of the one I
 specified in my extension but I could just be misreading that.   I
 have set the external IP of the freeswitch to the 56.3 address.

 Here is the LSOF output

 freeswitc 2468   root   31u IPv4   5785
 TCP ubuntuvm01:5080 (LISTEN)
 freeswitc 2468   root   33u IPv6   5791
 TCP localhost:5060 (LISTEN)
 freeswitc 2468   root   36u IPv4   5804
 TCP 192.168.56.3:5060 (LISTEN)
 freeswitc 2468   root   48u IPv4   5910
 TCP 192.168.56.3:8021 (LISTEN)
 freeswitc 2468   root   50u IPv4   5912
 TCP *:8080 (LISTEN)


 Here is the output from the fs_cli

 2009-12-09 14:31:53.255579 [DEBUG] sofia.c:5224 0 acls to check for proxy
 2009-12-09 14:31:53.255579 [DEBUG] sofia.c:5242 network ip is a proxy [0]
 2009-12-09 14:31:53.255579 [DEBUG] sofia.c:5270 IP 192.168.56.1
 Rejected by acl domains. Falling back to Digest auth.
 2009-12-09 14:31:53.357865 [DEBUG] sofia.c:5224 0 acls to check for proxy
 2009-12-09 14:31:53.357865 [DEBUG] sofia.c:5242 network ip is a proxy [0]
 2009-12-09 14:31:53.357865 [DEBUG] sofia.c:5270 IP 192.168.56.1
 Rejected by acl domains. Falling back to Digest auth.
 2009-12-09 14:31:53.420949 [NOTICE] switch_channel.c:613 New Channel
 sofia/internal/1...@192.168.56.3
 [2fbcf6fe-b35e-4c40-92a6-9f21de3102fa]
 2009-12-09 14:31:53.422090 [DEBUG] switch_core_state_machine.c:314
 (sofia/internal/1...@192.168.56.3) Running State Change CS_NEW
 2009-12-09 14:31:53.422090 [DEBUG] switch_core_state_machine.c:320
 (sofia/internal/1...@192.168.56.3) State NEW
 2009-12-09 14:31:53.422090 [DEBUG] sofia.c:3727 Channel
 sofia/internal/1...@192.168.56.3 entering state [received][100]
 2009-12-09 14:31:53.422090 [DEBUG] sofia.c:3738 Remote SDP:
 v=0
 o=Z 0 0 IN IP4 218.101.6.157
 s=Z
 c=IN IP4 218.101.6.157
 t=0 0
 m=audio 8000 RTP/AVP 3 110 98 8 0 101
 a=rtpmap:3 GSM/8000
 a=rtpmap:110 speex/8000
 a=rtpmap:98 iLBC/8000
 a=fmtp:98 mode=30
 a=rtpmap:8 PCMA/8000
 a=rtpmap:0 PCMU/8000
 a=rtpmap:101 telephone-event/8000
 a=fmtp:101 0-15

 2009-12-09 14:31:53.422090 [DEBUG] sofia_glue.c:3305 Audio Codec
 Compare [GSM:3:8000:20]/[G7221:115:32000:20]
 2009-12-09 14:31:53.422090 [DEBUG] sofia_glue.c:3305 Audio Codec
 Compare [GSM:3:8000:20]/[G7221:107:16000:20]
 2009-12-09 14:31:53.422090 [DEBUG] sofia_glue.c:3305 Audio Codec
 Compare [GSM:3:8000:20]/[G722:9:8000:20]
 2009-12-09 14:31:53.422090 [DEBUG] sofia_glue.c:3305 Audio Codec
 Compare [GSM:3:8000:20]/[PCMU:0:8000:20]
 2009-12-09 14:31:53.422090 [DEBUG] sofia_glue.c:3305 Audio Codec
 Compare [GSM:3:8000:20]/[PCMA:8:8000:20]
 2009-12-09 14:31:53.422090 [DEBUG] sofia_glue.c:3305 Audio Codec
 Compare [GSM:3:8000:20]/[GSM:3:8000:20]
 2009-12-09 14:31:53.422090 [DEBUG] sofia_glue.c:2143 Set Codec
 sofia/internal/1...@192.168.56.3 GSM/8000 20 ms 160 samples
 2009-12-09 14:31:53.423898 [DEBUG] sofia_glue.c:3261 Set 2833 dtmf
 payload to 101
 2009-12-09 14:31:53.423898 [DEBUG] sofia.c:3885
 (sofia/internal/1...@192.168.56.3) State Change CS_NEW - CS_INIT
 2009-12-09 14:31:53.423898 [DEBUG] switch_core_session.c:999 Send
 signal sofia/internal/1...@192.168.56.3 [BREAK]
 2009-12-09 14:31:53.423898 [DEBUG] switch_core_state_machine.c:314
 (sofia/internal/1...@192.168.56.3) Running State Change CS_INIT
 2009-12-09 14:31:53.423898 [DEBUG] switch_core_state_machine.c:338
 (sofia/internal/1...@192.168.56.3) State INIT
 2009-12-09 14:31:53.423898 [DEBUG] mod_sofia.c:83
 sofia/internal/1...@192.168.56.3 SOFIA INIT
 2009-12-09 14:31:53.423898 [DEBUG] mod_sofia.c:111
 (sofia/internal/1...@192.168.56.3) State Change CS_INIT - CS_ROUTING
 2009-12-09 14:31:53.423898 [DEBUG] switch_core_session.c:999 Send
 signal sofia/internal/1...@192.168.56.3 [BREAK]

Re: [Freeswitch-users] Even socket question.