Re: [Freeswitch-users] Live Upgrade Techniques

[Michal Bielicki]

Am 11.06.2009 um 05:04 schrieb John Dalgliesh:



Hi,

I am slowly gaining confidence using FreeSWITCH in production, but  
there

is one issue that I'm still wondering about: how are people upgrading
their FreeSWITCH installation binaries without dropping all current  
calls?


So far I have been upgrading in the dead of night, after pausing for 5
minutes then dropping the stragglers, but this is hardly ideal.

What I would like to do is to run an upgraded instance of FreeSWITCH  
on
the same machine, and have it handle all new call packets, whereas  
the old
instance continues to handle the existing call packets, until there  
are no

more old calls left.

I can think of about seven ways to accomplish this, but before I  
dive into

the code I thought I'd better ask what everyone else has been doing :)

(The only standard way I can think of doing this is to have a SIP  
proxy

sitting in front of FS the whole time, just to handle these upgrade
windows. It seems like a bit of a waste.)

So how are you handling your FS software upgrades?

{P^/
John





We use freeswitch on solaris and just upgrade it to a new zfs which  
gets remounted to the old place and freeswitch gracefully restartet.  
On failure we can allways do a rollback, which takes between 2 and 10  
seconds, so the dwntime is pretty acceptable.


Michal Bielicki
Leiter der Niederlassung
HaloKwadrat Sp. z o.o.
Niederlassung Kleinmachnow
Eingetragen im Handelsregister beim Amtsgericht Potsdam, HRB21422P
Ust.Id.: DE261885536
Geschaeftsfuehrer: Aleksander Wiercinski
Meiereifeld 2b, 14532 Kleinmachnow
t. +49 33203 263220
f. +49 33203 263229 sip. i...@halokwadrat.de
e. michal.bieli...@halokwadrat.de | w. www.halokwadrat.de
Hauptgeschäftsstelle:
Halo Kwadrat Sp. z o.o.
ul. Polna 46/14
00-644 Warszawa, Polen
EIngetragen im HRB Warszawa, KRS 153539



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[Freeswitch-users] FIFO through javascript

[Baskar]

Hi,

I have configured inbound in FS SVN Trunk. i have written small program
for inbound call to bridge. i have used session fifo

session.execute( "fifo", "sales_fifo_1 out wait undef
'/usr/local/freeswitch/sounds/en/us/callie/time/8000/tomorrow.wav'" );

session.execute("bridge", "sofia/internal/1003%");

Inbound call pass through JavaScript session and play the voice file but it
did not bridge to the extension 1003. It keep on playing the same voice
file. how can i bridge the call after
session.execute.

session.execute( "fifo", "sales_fifo_1 out wait undef
'/usr/local/freeswitch/sounds/en/us/callie/time/8000/tomorrow.wav'" );

Can any one assist me to resolve the above problem


-- 
Warm Regards,
N.Baskar
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Re: [Freeswitch-users] FIFO through javascript

[Brian West]

You can't... once you execute fifo your script has stopped.  I think  
you have the idea that your script will keep running after you enter  
the fifo...


/b
On Jun 11, 2009, at 7:13 AM, Baskar wrote:

>
> Can any one assist me to resolve the above problem


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[Freeswitch-users] Help understanding DEBUG and INFO log

[Lars Zeb]

In a dialplan, the action sets effective_caller_id_number to a value,
however, in INFO, the displayed value is not the same as the set. Why?

 

http://pastebin.freeswitch.org/9361

 

Thanks, Lars

 

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[Freeswitch-users] Rejecting calls without answering

[Klaus Teller]

Hi Team,

I'm still in need of a way to reject a call without answering it. I very much 
appreciate your help.

Klaus. 
-- 
GMX FreeDSL Komplettanschluss mit DSL 6.000 Flatrate und Telefonanschluss
für nur 17,95 Euro/mtl.!* http://portal.gmx.net/de/go/dsl02

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Re: [Freeswitch-users] Help understanding DEBUG and INFO log

[Brian West]

Are you doing an originate or a bridge?

/b

On Jun 11, 2009, at 8:49 AM, Lars Zeb wrote:

In a dialplan, the action sets effective_caller_id_number to a  
value, however, in INFO, the displayed value is not the same as the  
set. Why?


http://pastebin.freeswitch.org/9361

Thanks, Lars

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Brian West
br...@freeswitch.org

-- Meet us at ClueCon!  http://www.cluecon.com




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Re: [Freeswitch-users] Rejecting calls without answering

[Brian West]

respond will do exactly that...  try just hangup

/b

On Jun 11, 2009, at 9:19 AM, Klaus Teller wrote:


Hi Team,

I'm still in need of a way to reject a call without answering it. I  
very much appreciate your help.


Klaus.


Brian West
br...@freeswitch.org

-- Meet us at ClueCon!  http://www.cluecon.com




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Re: [Freeswitch-users] Rejecting calls without answering

[Raymond Chandler]

Klaus Teller wrote:
> Hi Team,
>
> I'm still in need of a way to reject a call without answering it. I very much 
> appreciate your help.
>
> Klaus. 
>   

-Ray

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Re: [Freeswitch-users] Help understanding DEBUG and INFO log

[Lars Zeb]

Bridge

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian
West
Sent: Thursday, June 11, 2009 7:24 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Help understanding DEBUG and INFO log

 

Are you doing an originate or a bridge?

 

/b

 

On Jun 11, 2009, at 8:49 AM, Lars Zeb wrote:





In a dialplan, the action sets effective_caller_id_number to a value,
however, in INFO, the displayed value is not the same as the set. Why?

 

http://pastebin.freeswitch.org/9361

 

Thanks, Lars

 

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Brian West

br...@freeswitch.org

 

-- Meet us at ClueCon!  http://www.cluecon.com  

 

 

 

 

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Re: [Freeswitch-users] Help understanding DEBUG and INFO log

[Brian West]

make sure you set it before the bridge.

/b

On Jun 11, 2009, at 9:54 AM, Lars Zeb wrote:


Bridge

From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org 
] On Behalf Of Brian West

Sent: Thursday, June 11, 2009 7:24 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Help understanding DEBUG and INFO log

Are you doing an originate or a bridge?

/b


Brian West
br...@freeswitch.org

-- Meet us at ClueCon!  http://www.cluecon.com




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Re: [Freeswitch-users] Web page thoughts

[Anthony Minessale]

One important thing is that if we go around following everything everybody
else says
we become a follower in our field.

I have had numerous people tell me what to do in the code, what to name
things, what to eat for breakfast.
Plain and simple, I will choose what to put on our website, when to put it
there and what it says.

You are welcome to your own opinion.  I have no problem with it.  If you say
something I like we may
even listen.  Feel free to comment on anything else you find when browsing
our community.

BUT,

If you have no sense of humor, you will not make it far in the open source
telecom industry.

If you want a more professional looking site, we do have some guys in suits
on our FreeSWITCH Solutions site.
http://www.freeswitchsolutions.com/




On Wed, Jun 10, 2009 at 7:49 PM,  wrote:

> My company is currently investigating a couple of projects that may take
> me in the direction of FreeSwitch...  In general, our management does
> not often consider open source software for projects such as this, but
> I've been successful in proving to them recently that open source can
> deliver.
>
> FreeSwitch is a *very* professional and polished product, I can tell -
> from the code and from the community.
>
> Unfortunately, I've been hesitant to send people to your webpage lately
> because it went downhill a few weeks ago.  Whenever I think about one of
> our executives going to your webpage (after my recommendation) and
> seeing a picture of people clanking beer glasses, or some idiot tied up
> in phone cables, I cringe.  I know you're advertising for ClueCon, but
> honestly, some of those huge images on your front page really knock your
> product down a peg in professionalism.
>
> Anyway, I'm pretty new to the community and I don't claim to be a web
> designer.  You have an excellent piece of software, but if I didn't
> already know that about FreeSwitch, your webpage would not make a good
> first impression.
>
> Please take that for what it's worth...  I wanted to voice my opinion
> because if I'm thinking it, others may be as well.
> Thoughts anyone?
>
> J
>
> (Have I mentioned how awesome your source browser is though??!!)
>
> ___
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> 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/

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
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Re: [Freeswitch-users] Help understanding DEBUG and INFO log

[Lars Zeb]

It was. You can see the set at line 2 was done before the bridge at line 4.
What am I missing?

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian
West
Sent: Thursday, June 11, 2009 8:06 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Help understanding DEBUG and INFO log

 

make sure you set it before the bridge.

 

/b

 

On Jun 11, 2009, at 9:54 AM, Lars Zeb wrote:





Bridge

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian
West
Sent: Thursday, June 11, 2009 7:24 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Help understanding DEBUG and INFO log

 

Are you doing an originate or a bridge?

 

/b

 

Brian West

br...@freeswitch.org

 

-- Meet us at ClueCon!  http://www.cluecon.com  

 

 

 

 

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Re: [Freeswitch-users] Web page thoughts

[Steve Underwood]

Anthony Minessale wrote:
> One important thing is that if we go around following everything 
> everybody else says
> we become a follower in our field.
>
> I have had numerous people tell me what to do in the code, what to 
> name things, what to eat for breakfast.
> Plain and simple, I will choose what to put on our website, when to 
> put it there and what it says.
>
> You are welcome to your own opinion.  I have no problem with it.  If 
> you say something I like we may
> even listen.  Feel free to comment on anything else you find when 
> browsing our community.
>
> BUT,
>
> If you have no sense of humor, you will not make it far in the open 
> source telecom industry.
>
> If you want a more professional looking site, we do have some guys in 
> suits on our FreeSWITCH Solutions site.
> http://www.freeswitchsolutions.com/
The main reason www.freeswitchsolutions.com 
 looks more professional that 
www.freeswitch.org is not the content of the pictures but their size. 
The pictures at the top of the www.freeswitch.org are too big and in 
your face. They completely dominate the screen when it appears.

Steve


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Re: [Freeswitch-users] Rejecting calls without answering

[Klaus Teller]

Hi Folks,

Here is what i'm observing. When i connect with Xlite (registered device) and 
call the 9444 extension (see below), Freeswitch does hangup as i would like it 
to.

But when i call via gafachi, something weird happens. What i can see is that 
Freeswitch sends a hangup signal (service temporarily not available) to 
Gafachi, but the guys keep sending back the very same call. 


It looks to me like a Gafachi issue. But can anything else be done on the 
Freeswitch side?

I'm attaching the logs for the gafachi call this. All you see in there is just 
one single call. You will see that a new channel is created more than once.

Any thought?

Klaus.

The gafachi respond extension (under conf/dialplan/public/reject.xml):


   

   




The gafachi profile (under conf/sip_profiles/external/gafachi.xml):

  



  


The Xlite respond test extension (in default.xml):

  

  



Any idea?






> respond will do exactly that...  try just hangup
> 
> /b
> 
> On Jun 11, 2009, at 9:19 AM, Klaus Teller wrote:
> 
> > Hi Team,
> >
> > I'm still in need of a way to reject a call without answering it. I  
> > very much appreciate your help.
> >
> > Klaus.
> 
> Brian West
> br...@freeswitch.org
> 
> -- Meet us at ClueCon!  http://www.cluecon.com
> 
> 
> 
> 

-- 
GMX FreeDSL Komplettanschluss mit DSL 6.000 Flatrate und Telefonanschluss
für nur 17,95 Euro/mtl.!* http://portal.gmx.net/de/go/dsl02

2009-06-11 12:14:34.870820 [NOTICE] switch_channel.c:602 New Channel 
sofia/external/6473671...@sip.gafachi.com [6867d2a1-e26a-4cab-9122-7ecab2b9397f]
2009-06-11 12:14:34.870820 [DEBUG] sofia.c:3100 Channel 
sofia/external/6473671...@sip.gafachi.com entering state [received][100]
2009-06-11 12:14:34.870820 [DEBUG] sofia.c:3107 Remote SDP:
v=0
o=root 273544867 273544867 IN IP4 67.216.37.18
s=session
c=IN IP4 67.216.37.18
t=0 0
m=audio 35116 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=silenceSupp:off - - - -
a=ptime:20

2009-06-11 12:14:34.870820 [DEBUG] sofia_glue.c:3059 Audio Codec Compare 
[PCMU:0:8000:20]/[PCMU:0:8000:20]
2009-06-11 12:14:34.870820 [DEBUG] sofia_glue.c:2017 Set Codec 
sofia/external/6473671...@sip.gafachi.com PCMU/8000 20 ms 160 samples
2009-06-11 12:14:34.870820 [DEBUG] switch_core_state_machine.c:397 
(sofia/external/6473671...@sip.gafachi.com) Running State Change CS_NEW
2009-06-11 12:14:34.874889 [DEBUG] switch_core_state_machine.c:403 
(sofia/external/6473671...@sip.gafachi.com) State NEW
2009-06-11 12:14:34.874889 [DEBUG] sofia_glue.c:3019 Set 2833 dtmf payload to 
101
2009-06-11 12:14:34.874889 [DEBUG] sofia.c:3266 
(sofia/external/6473671...@sip.gafachi.com) State Change CS_NEW -> CS_INIT
2009-06-11 12:14:34.874889 [DEBUG] switch_core_session.c:933 Send signal 
sofia/external/6473671...@sip.gafachi.com [BREAK]
2009-06-11 12:14:34.878827 [DEBUG] switch_core_state_machine.c:397 
(sofia/external/6473671...@sip.gafachi.com) Running State Change CS_INIT
2009-06-11 12:14:34.878827 [DEBUG] switch_core_state_machine.c:480 
(sofia/external/6473671...@sip.gafachi.com) State INIT
2009-06-11 12:14:34.878827 [DEBUG] mod_sofia.c:83 
sofia/external/6473671...@sip.gafachi.com SOFIA INIT
2009-06-11 12:14:34.878827 [DEBUG] mod_sofia.c:111 
(sofia/external/6473671...@sip.gafachi.com) State Change CS_INIT -> CS_ROUTING
2009-06-11 12:14:34.878827 [DEBUG] switch_core_session.c:933 Send signal 
sofia/external/6473671...@sip.gafachi.com [BREAK]
2009-06-11 12:14:34.878827 [DEBUG] switch_core_state_machine.c:480 
(sofia/external/6473671...@sip.gafachi.com) State INIT going to sleep
2009-06-11 12:14:34.878827 [DEBUG] switch_core_state_machine.c:397 
(sofia/external/6473671...@sip.gafachi.com) Running State Change CS_ROUTING
2009-06-11 12:14:34.878827 [DEBUG] switch_core_state_machine.c:483 
(sofia/external/6473671...@sip.gafachi.com) State ROUTING
2009-06-11 12:14:34.878827 [DEBUG] mod_sofia.c:130 
sofia/external/6473671...@sip.gafachi.com SOFIA ROUTING
2009-06-11 12:14:34.878827 [DEBUG] switch_core_state_machine.c:78 
sofia/external/6473671...@sip.gafachi.com Standard ROUTING
2009-06-11 12:14:34.878827 [INFO] mod_dialplan_xml.c:252 Processing 
unknown->18664591152 in context public
Dialplan: sofia/external/6473671...@sip.gafachi.com parsing [public->unloop] 
continue=false
Dialplan: sofia/external/6473671...@sip.gafachi.com Regex (PASS) [unloop] 
${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/external/6473671...@sip.gafachi.com Regex (FAIL) [unloop] 
${sip_looped_call}() =~ /^true$/ break=on-false
Dialplan: sofia/external/6473671...@sip.gafachi.com parsing 
[public->outside_call] continue=true
Dialplan: sofia/external/6473671...@sip.gafachi.com Absolute Condition 
[outside_call]
Dialplan: sofia/external/6473671...@sip.gafachi.com Action 
set(outside_call=true)
Dialplan: sofia/external/6473671...@sip.gafachi.com parsing 
[public->call_debug] continue=true
Dialplan: sofia/ex

Re: [Freeswitch-users] Rejecting calls without answering

[Ken Rice]

Hah they are just retrying the call to see if they get a different answer
the 2nd and 3rd time around... This is common unfortunately since 503 in the
VoIP world is typically interpreted by the PSTN world as a "Temp congestion"
(and rightly so) so they will retry and not fail the call... You can try
responding w/ 486 Busy if you know the call doesn't need to fail somewhere
else...


> From: Klaus Teller 
> Reply-To: 
> Date: Thu, 11 Jun 2009 18:21:30 +0200
> To: 
> Subject: Re: [Freeswitch-users] Rejecting calls without answering
> 
> Hi Folks,
> 
> Here is what i'm observing. When i connect with Xlite (registered device) and
> call the 9444 extension (see below), Freeswitch does hangup as i would like it
> to.
> 
> But when i call via gafachi, something weird happens. What i can see is that
> Freeswitch sends a hangup signal (service temporarily not available) to
> Gafachi, but the guys keep sending back the very same call.
> 
> 
> It looks to me like a Gafachi issue. But can anything else be done on the
> Freeswitch side?
> 
> I'm attaching the logs for the gafachi call this. All you see in there is just
> one single call. You will see that a new channel is created more than once.
> 
> Any thought?
> 
> Klaus.
> 
> The gafachi respond extension (under conf/dialplan/public/reject.xml):
> 
> 
>
> 
>
> 
> 
> 
> 
> The gafachi profile (under conf/sip_profiles/external/gafachi.xml):
> 
>   
> 
> 
> 
>   
> 
> 
> The Xlite respond test extension (in default.xml):
> 
>   
> 
>   
> 
> 
> 
> Any idea?
> 
> 
> 
> 
> 
> 
>> respond will do exactly that...  try just hangup
>> 
>> /b
>> 
>> On Jun 11, 2009, at 9:19 AM, Klaus Teller wrote:
>> 
>>> Hi Team,
>>> 
>>> I'm still in need of a way to reject a call without answering it. I
>>> very much appreciate your help.
>>> 
>>> Klaus.
>> 
>> Brian West
>> br...@freeswitch.org
>> 
>> -- Meet us at ClueCon!  http://www.cluecon.com
>> 
>> 
>> 
>> 
> 
> -- 
> GMX FreeDSL Komplettanschluss mit DSL 6.000 Flatrate und Telefonanschluss
> für nur 17,95 Euro/mtl.!* http://portal.gmx.net/de/go/dsl02
> ___
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Re: [Freeswitch-users] Web page thoughts

[Anthony Minessale]

Ok,
let's try them at half size.


On Thu, Jun 11, 2009 at 11:09 AM, Steve Underwood wrote:

> Anthony Minessale wrote:
> > One important thing is that if we go around following everything
> > everybody else says
> > we become a follower in our field.
> >
> > I have had numerous people tell me what to do in the code, what to
> > name things, what to eat for breakfast.
> > Plain and simple, I will choose what to put on our website, when to
> > put it there and what it says.
> >
> > You are welcome to your own opinion.  I have no problem with it.  If
> > you say something I like we may
> > even listen.  Feel free to comment on anything else you find when
> > browsing our community.
> >
> > BUT,
> >
> > If you have no sense of humor, you will not make it far in the open
> > source telecom industry.
> >
> > If you want a more professional looking site, we do have some guys in
> > suits on our FreeSWITCH Solutions site.
> > http://www.freeswitchsolutions.com/
> The main reason www.freeswitchsolutions.com
>  looks more professional that
> www.freeswitch.org is not the content of the pictures but their size.
> The pictures at the top of the www.freeswitch.org are too big and in
> your face. They completely dominate the screen when it appears.
>
> Steve
>
>
> ___
> 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/

AIM: anthm
MSN:anthony_miness...@hotmail.com 
GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com
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Re: [Freeswitch-users] Rejecting calls without answering

[Klaus Teller]

Excellent! Thank you everybody. Response 486 did the trick.

Klaus.
 Original-Nachricht 
> Datum: Thu, 11 Jun 2009 11:31:13 -0500
> Von: Ken Rice 
> An: freeswitch-users@lists.freeswitch.org
> Betreff: Re: [Freeswitch-users] Rejecting calls without answering

> Hah they are just retrying the call to see if they get a different answer
> the 2nd and 3rd time around... This is common unfortunately since 503 in
> the
> VoIP world is typically interpreted by the PSTN world as a "Temp
> congestion"
> (and rightly so) so they will retry and not fail the call... You can try
> responding w/ 486 Busy if you know the call doesn't need to fail somewhere
> else...
> 
> 
> > From: Klaus Teller 
> > Reply-To: 
> > Date: Thu, 11 Jun 2009 18:21:30 +0200
> > To: 
> > Subject: Re: [Freeswitch-users] Rejecting calls without answering
> > 
> > Hi Folks,
> > 
> > Here is what i'm observing. When i connect with Xlite (registered
> device) and
> > call the 9444 extension (see below), Freeswitch does hangup as i would
> like it
> > to.
> > 
> > But when i call via gafachi, something weird happens. What i can see is
> that
> > Freeswitch sends a hangup signal (service temporarily not available) to
> > Gafachi, but the guys keep sending back the very same call.
> > 
> > 
> > It looks to me like a Gafachi issue. But can anything else be done on
> the
> > Freeswitch side?
> > 
> > I'm attaching the logs for the gafachi call this. All you see in there
> is just
> > one single call. You will see that a new channel is created more than
> once.
> > 
> > Any thought?
> > 
> > Klaus.
> > 
> > The gafachi respond extension (under conf/dialplan/public/reject.xml):
> > 
> > 
> > expression="^866.*$">
> > 
> >
> > 
> > 
> > 
> > 
> > The gafachi profile (under conf/sip_profiles/external/gafachi.xml):
> > 
> >   
> > 
> > 
> > 
> >   
> > 
> > 
> > The Xlite respond test extension (in default.xml):
> > 
> >   
> > 
> >   
> > 
> > 
> > 
> > Any idea?
> > 
> > 
> > 
> > 
> > 
> > 
> >> respond will do exactly that...  try just hangup
> >> 
> >> /b
> >> 
> >> On Jun 11, 2009, at 9:19 AM, Klaus Teller wrote:
> >> 
> >>> Hi Team,
> >>> 
> >>> I'm still in need of a way to reject a call without answering it. I
> >>> very much appreciate your help.
> >>> 
> >>> Klaus.
> >> 
> >> Brian West
> >> br...@freeswitch.org
> >> 
> >> -- Meet us at ClueCon!  http://www.cluecon.com
> >> 
> >> 
> >> 
> >> 
> > 
> > -- 
> > GMX FreeDSL Komplettanschluss mit DSL 6.000 Flatrate und
> Telefonanschluss
> > für nur 17,95 Euro/mtl.!* http://portal.gmx.net/de/go/dsl02
> > ___
> > 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
> 
> 
> 
> ___
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> Freeswitch-users@lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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-- 
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Jetzt freischalten unter http://portal.gmx.net/de/go/maxdome01

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Re: [Freeswitch-users] Web page thoughts

[Diego Viola]

That's what I tried to say, I didn't expressed myself well, sorry.

On Thu, Jun 11, 2009 at 12:09 PM, Steve Underwood wrote:

> Anthony Minessale wrote:
> > One important thing is that if we go around following everything
> > everybody else says
> > we become a follower in our field.
> >
> > I have had numerous people tell me what to do in the code, what to
> > name things, what to eat for breakfast.
> > Plain and simple, I will choose what to put on our website, when to
> > put it there and what it says.
> >
> > You are welcome to your own opinion.  I have no problem with it.  If
> > you say something I like we may
> > even listen.  Feel free to comment on anything else you find when
> > browsing our community.
> >
> > BUT,
> >
> > If you have no sense of humor, you will not make it far in the open
> > source telecom industry.
> >
> > If you want a more professional looking site, we do have some guys in
> > suits on our FreeSWITCH Solutions site.
> > http://www.freeswitchsolutions.com/
> The main reason www.freeswitchsolutions.com
>  looks more professional that
> www.freeswitch.org is not the content of the pictures but their size.
> The pictures at the top of the www.freeswitch.org are too big and in
> your face. They completely dominate the screen when it appears.
>
> Steve
>
>
> ___
> Freeswitch-users mailing list
> Freeswitch-users@lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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> http://www.freeswitch.org
>
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Re: [Freeswitch-users] Rejecting calls without answering

[Metik]

Although a little overkill, RFC3398 also describes some desirable interop 
behavior between ISUP, ISDN and SIP.

(From "7.2.4.1 ISDN Cause Code to Status Code Mapping")

[...]
   ISUP Cause valueSIP response
   
   1  unallocated number   404 Not Found
   2  no route to network  404 Not found
   3  no route to destination  404 Not found
   16 normal call clearing --- (*)
   17 user busy486 Busy here
   18 no user responding   408 Request Timeout
   19 no answer from the user  480 Temporarily unavailable
   20 subscriber absent480 Temporarily unavailable
   21 call rejected403 Forbidden (+)
   22 number changed (w/o diagnostic)  410 Gone
   22 number changed (w/ diagnostic)   301 Moved Permanently
   23 redirection to new destination   410 Gone
   26 non-selected user clearing   404 Not Found (=)
   27 destination out of order 502 Bad Gateway
   28 address incomplete   484 Address incomplete
   29 facility rejected501 Not implemented
   31 normal unspecified   480 Temporarily unavailable

[...]
Resource unavailable

   This kind of cause value indicates a temporary failure.  A 'Retry-After' 
header MAY be added to the response if appropriate.

   ISUP Cause valueSIP response
   
   34 no circuit available 503 Service unavailable
   38 network out of order 503 Service unavailable
   41 temporary failure503 Service unavailable
   42 switching equipment congestion   503 Service unavailable
   47 resource unavailable 503 Service unavailable


- Original Message - 
From: "Ken Rice" 
To: 
Sent: Thursday, June 11, 2009 12:31 PM
Subject: Re: [Freeswitch-users] Rejecting calls without answering


Hah they are just retrying the call to see if they get a different answer
the 2nd and 3rd time around... This is common unfortunately since 503 in the
VoIP world is typically interpreted by the PSTN world as a "Temp congestion"
(and rightly so) so they will retry and not fail the call... You can try
responding w/ 486 Busy if you know the call doesn't need to fail somewhere
else...


> From: Klaus Teller 
> Reply-To: 
> Date: Thu, 11 Jun 2009 18:21:30 +0200
> To: 
> Subject: Re: [Freeswitch-users] Rejecting calls without answering
>
> Hi Folks,
>
> Here is what i'm observing. When i connect with Xlite (registered device) 
> and
> call the 9444 extension (see below), Freeswitch does hangup as i would 
> like it
> to.
>
> But when i call via gafachi, something weird happens. What i can see is 
> that
> Freeswitch sends a hangup signal (service temporarily not available) to
> Gafachi, but the guys keep sending back the very same call.
>
>
> It looks to me like a Gafachi issue. But can anything else be done on the
> Freeswitch side?
>
> I'm attaching the logs for the gafachi call this. All you see in there is 
> just
> one single call. You will see that a new channel is created more than 
> once.
>
> Any thought?
>
> Klaus.
>
> The gafachi respond extension (under conf/dialplan/public/reject.xml):
> 
> 
>
> 
>
> 
> 
>
>
> The gafachi profile (under conf/sip_profiles/external/gafachi.xml):
> 
>   
> 
> 
> 
>   
> 
>
> The Xlite respond test extension (in default.xml):
> 
>   
> 
>   
> 
>
>
> Any idea?
>
>
>
>
>
>
>> respond will do exactly that...  try just hangup
>>
>> /b
>>
>> On Jun 11, 2009, at 9:19 AM, Klaus Teller wrote:
>>
>>> Hi Team,
>>>
>>> I'm still in need of a way to reject a call without answering it. I
>>> very much appreciate your help.
>>>
>>> Klaus.
>>
>> Brian West
>> br...@freeswitch.org
>>
>> -- Meet us at ClueCon!  http://www.cluecon.com
>>
>>
>>
>>
>
> -- 
> GMX FreeDSL Komplettanschluss mit DSL 6.000 Flatrate und Telefonanschluss
> für nur 17,95 Euro/mtl.!* http://portal.gmx.net/de/go/dsl02
> ___
> 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
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Re: [Freeswitch-users] Live Upgrade Techniques

[Anthony Minessale]

or you can put a sip proxy in front of 2 boxes where you can control the
flow of traffic.
when you want to upgrade one, take all the traffic off of it by forcing all
calls to the other box, upgrade it then shift the traffic to the new one.
if that goes well, upgrade the other one too.



On Thu, Jun 11, 2009 at 5:47 AM, Michal Bielicki
wrote:

>
> Am 11.06.2009 um 05:04 schrieb John Dalgliesh:
>
>
>> Hi,
>>
>> I am slowly gaining confidence using FreeSWITCH in production, but there
>> is one issue that I'm still wondering about: how are people upgrading
>> their FreeSWITCH installation binaries without dropping all current calls?
>>
>> So far I have been upgrading in the dead of night, after pausing for 5
>> minutes then dropping the stragglers, but this is hardly ideal.
>>
>> What I would like to do is to run an upgraded instance of FreeSWITCH on
>> the same machine, and have it handle all new call packets, whereas the old
>> instance continues to handle the existing call packets, until there are no
>> more old calls left.
>>
>> I can think of about seven ways to accomplish this, but before I dive into
>> the code I thought I'd better ask what everyone else has been doing :)
>>
>> (The only standard way I can think of doing this is to have a SIP proxy
>> sitting in front of FS the whole time, just to handle these upgrade
>> windows. It seems like a bit of a waste.)
>>
>> So how are you handling your FS software upgrades?
>>
>> {P^/
>> John
>>
>>
>>
>
> We use freeswitch on solaris and just upgrade it to a new zfs which gets
> remounted to the old place and freeswitch gracefully restartet. On failure
> we can allways do a rollback, which takes between 2 and 10 seconds, so the
> dwntime is pretty acceptable.
>
> Michal Bielicki
> Leiter der Niederlassung
> HaloKwadrat Sp. z o.o.
> Niederlassung Kleinmachnow
> Eingetragen im Handelsregister beim Amtsgericht Potsdam, HRB21422P
> Ust.Id.: DE261885536
> Geschaeftsfuehrer: Aleksander Wiercinski
> Meiereifeld 2b, 14532 Kleinmachnow
> t. +49 33203 263220
> f. +49 33203 263229 sip. i...@halokwadrat.de
> e. michal.bieli...@halokwadrat.de | w. www.halokwadrat.de
> Hauptgeschäftsstelle:
> Halo Kwadrat Sp. z o.o.
> ul. Polna 46/14
> 00-644 Warszawa, Polen
> EIngetragen im HRB Warszawa, KRS 153539
>
>
> ___
> 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/

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
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Re: [Freeswitch-users] Live Upgrade Techniques

[Michael Giagnocavo]

Exactly. You probably want to have something like this anyways, so that when 
someone accidentally unplugs the system, or the disks/CPU/RAM crash, you're not 
stuck.

That is, until FreeSWITCH can record its internal state to some inter-machine 
memory so we can have hot failover. ;)

-Michael

From: freeswitch-users-boun...@lists.freeswitch.org 
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Anthony 
Minessale
Sent: Thursday, June 11, 2009 10:55 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Live Upgrade Techniques

or you can put a sip proxy in front of 2 boxes where you can control the flow 
of traffic.
when you want to upgrade one, take all the traffic off of it by forcing all 
calls to the other box, upgrade it then shift the traffic to the new one.
if that goes well, upgrade the other one too.


On Thu, Jun 11, 2009 at 5:47 AM, Michal Bielicki  
wrote:

Am 11.06.2009 um 05:04 schrieb John Dalgliesh:


Hi,

I am slowly gaining confidence using FreeSWITCH in production, but there
is one issue that I'm still wondering about: how are people upgrading
their FreeSWITCH installation binaries without dropping all current calls?

So far I have been upgrading in the dead of night, after pausing for 5
minutes then dropping the stragglers, but this is hardly ideal.

What I would like to do is to run an upgraded instance of FreeSWITCH on
the same machine, and have it handle all new call packets, whereas the old
instance continues to handle the existing call packets, until there are no
more old calls left.

I can think of about seven ways to accomplish this, but before I dive into
the code I thought I'd better ask what everyone else has been doing :)

(The only standard way I can think of doing this is to have a SIP proxy
sitting in front of FS the whole time, just to handle these upgrade
windows. It seems like a bit of a waste.)

So how are you handling your FS software upgrades?

{P^/
John


We use freeswitch on solaris and just upgrade it to a new zfs which gets 
remounted to the old place and freeswitch gracefully restartet. On failure we 
can allways do a rollback, which takes between 2 and 10 seconds, so the dwntime 
is pretty acceptable.

Michal Bielicki
Leiter der Niederlassung
HaloKwadrat Sp. z o.o.
Niederlassung Kleinmachnow
Eingetragen im Handelsregister beim Amtsgericht Potsdam, HRB21422P
Ust.Id.: DE261885536
Geschaeftsfuehrer: Aleksander Wiercinski
Meiereifeld 2b, 14532 Kleinmachnow
t. +49 33203 263220
f. +49 33203 263229 sip. i...@halokwadrat.de
e. michal.bieli...@halokwadrat.de | w. 
www.halokwadrat.de
Hauptgeschäftsstelle:
Halo Kwadrat Sp. z o.o.
ul. Polna 46/14
00-644 Warszawa, Polen
EIngetragen im HRB Warszawa, KRS 153539


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--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

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
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Re: [Freeswitch-users] Live Upgrade Techniques

[Michael Collins]

On Thu, Jun 11, 2009 at 10:33 AM, Michael Giagnocavo wrote:

>  Exactly. You probably want to have something like this anyways, so that
> when someone accidentally unplugs the system, or the disks/CPU/RAM crash,
> you’re not stuck.
>
>
>
> That is, until FreeSWITCH can record its internal state to some
> inter-machine memory so we can have hot failover. ;)
>
>
>
I think that's going to be in 1.0.5. :)

> -Michael
>
>
>
> *From:* freeswitch-users-boun...@lists.freeswitch.org [mailto:
> freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *Anthony
> Minessale
> *Sent:* Thursday, June 11, 2009 10:55 AM
> *To:* freeswitch-users@lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Live Upgrade Techniques
>
>
>
> or you can put a sip proxy in front of 2 boxes where you can control the
> flow of traffic.
> when you want to upgrade one, take all the traffic off of it by forcing all
> calls to the other box, upgrade it then shift the traffic to the new one.
> if that goes well, upgrade the other one too.
>
>
>  On Thu, Jun 11, 2009 at 5:47 AM, Michal Bielicki
>  wrote:
>
>
> Am 11.06.2009 um 05:04 schrieb John Dalgliesh:
>
>
>
>
> Hi,
>
> I am slowly gaining confidence using FreeSWITCH in production, but there
> is one issue that I'm still wondering about: how are people upgrading
> their FreeSWITCH installation binaries without dropping all current calls?
>
> So far I have been upgrading in the dead of night, after pausing for 5
> minutes then dropping the stragglers, but this is hardly ideal.
>
> What I would like to do is to run an upgraded instance of FreeSWITCH on
> the same machine, and have it handle all new call packets, whereas the old
> instance continues to handle the existing call packets, until there are no
> more old calls left.
>
> I can think of about seven ways to accomplish this, but before I dive into
> the code I thought I'd better ask what everyone else has been doing :)
>
> (The only standard way I can think of doing this is to have a SIP proxy
> sitting in front of FS the whole time, just to handle these upgrade
> windows. It seems like a bit of a waste.)
>
> So how are you handling your FS software upgrades?
>
> {P^/
> John
>
>
>
> We use freeswitch on solaris and just upgrade it to a new zfs which gets
> remounted to the old place and freeswitch gracefully restartet. On failure
> we can allways do a rollback, which takes between 2 and 10 seconds, so the
> dwntime is pretty acceptable.
>
> Michal Bielicki
> Leiter der Niederlassung
> HaloKwadrat Sp. z o.o.
> Niederlassung Kleinmachnow
> Eingetragen im Handelsregister beim Amtsgericht Potsdam, HRB21422P
> Ust.Id.: DE261885536
> Geschaeftsfuehrer: Aleksander Wiercinski
> Meiereifeld 2b, 14532 Kleinmachnow
> t. +49 33203 263220
> f. +49 33203 263229 sip. i...@halokwadrat.de
> e. michal.bieli...@halokwadrat.de | w. www.halokwadrat.de
> Hauptgeschäftsstelle:
> Halo Kwadrat Sp. z o.o.
> ul. Polna 46/14
> 00-644 Warszawa, Polen
> EIngetragen im HRB Warszawa, KRS 153539
>
>
> ___
> Freeswitch-users mailing list
> Freeswitch-users@lists.freeswitch.org
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>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
>
> 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
>
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Re: [Freeswitch-users] Web page thoughts

[jcromes]

Haha, good point on the FreeSwitch Solutions site...  Suits - very 
professional.  =)
Please don't think I'm telling you guys what to do, I know you don't 
need that.  It IS your software and your site, and you've done a HELL of 
a job with it so far.

It was just a thought.  Sorry if I offended.


Anthony Minessale wrote:
> One important thing is that if we go around following everything 
> everybody else says
> we become a follower in our field.
>
> I have had numerous people tell me what to do in the code, what to 
> name things, what to eat for breakfast.
> Plain and simple, I will choose what to put on our website, when to 
> put it there and what it says.
>
> You are welcome to your own opinion.  I have no problem with it.  If 
> you say something I like we may
> even listen.  Feel free to comment on anything else you find when 
> browsing our community.
>
> BUT,
>
> If you have no sense of humor, you will not make it far in the open 
> source telecom industry.
>
> If you want a more professional looking site, we do have some guys in 
> suits on our FreeSWITCH Solutions site.
> http://www.freeswitchsolutions.com/
>

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Re: [Freeswitch-users] Web page thoughts

[Michael Collins]

On Thu, Jun 11, 2009 at 10:43 AM,  wrote:

> Haha, good point on the FreeSwitch Solutions site...  Suits - very
> professional.  =)
> Please don't think I'm telling you guys what to do, I know you don't
> need that.  It IS your software and your site, and you've done a HELL of
> a job with it so far.
>
> It was just a thought.  Sorry if I offended.
>
No offense taken. We DO appreciate feedback, unsolicited or otherwise, but
we don't always agree with it. Definitely show your bosses the FSS site. :)

-MC
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Re: [Freeswitch-users] Live Upgrade Techniques

[John Dalgliesh]

OK thanks that is what I thought the general way of doing it would be. But 
it seems a bit wasteful to have that SIP proxy there the whole time 
especially when I am using FS in the role of an SBC.


The problem with the graceful restart of course is that you have to wait 
for the calls count to get to zero, which may never happen. It's 3:30am 
here in Sydney now and I just checked FS: 20 calls in progress still!


So what I plan to do is add a '--upgrade' cmd line arg to FS. This will 
make the new instance contact the old one on a unix socket and receive a 
dup of its SIP socket fd(s) via a SCM_RIGHTS sendmsg. It will use those 
for sending and the unix socket for receiving. Meanwhile the old instance 
will pass any packets with unknown Call-Ids over the unix socket to the 
new instance, instead of handling them itself. When the old instance has 
no calls left, it shuts down. The new instance detects the unix socket is 
closed and switches to reading from the SIP socket (which would have 
buffered any unread packets - so nothing is lost).


Sound good? I realise this will be 90% in libsofia but I've read teh code 
and it seems very do-able. Anyone interested in my changes will of course 
be most welcome to them.


The runner-up approach I considered was to make a kernel module that 
extends iptables with a filter that can extract the Call-Id and look it up 
in a table that is somehow populated from FS. Maybe this exists already? 
Kind of a SIP proxy lite that can be enabled on the server machine when 
needed. Anyway that lost out as it's more work and even less portable.


{P^/
John

On Thu, 11 Jun 2009 at 11:54 -0500, Anthony Minessale wrote:


or you can put a sip proxy in front of 2 boxes where you can control the
flow of traffic.
when you want to upgrade one, take all the traffic off of it by forcing all
calls to the other box, upgrade it then shift the traffic to the new one.
if that goes well, upgrade the other one too.



On Thu, Jun 11, 2009 at 5:47 AM, Michal Bielicki
wrote:



Am 11.06.2009 um 05:04 schrieb John Dalgliesh:



Hi,

I am slowly gaining confidence using FreeSWITCH in production, but there
is one issue that I'm still wondering about: how are people upgrading
their FreeSWITCH installation binaries without dropping all current calls?

So far I have been upgrading in the dead of night, after pausing for 5
minutes then dropping the stragglers, but this is hardly ideal.

What I would like to do is to run an upgraded instance of FreeSWITCH on
the same machine, and have it handle all new call packets, whereas the old
instance continues to handle the existing call packets, until there are no
more old calls left.

I can think of about seven ways to accomplish this, but before I dive into
the code I thought I'd better ask what everyone else has been doing :)

(The only standard way I can think of doing this is to have a SIP proxy
sitting in front of FS the whole time, just to handle these upgrade
windows. It seems like a bit of a waste.)

So how are you handling your FS software upgrades?

{P^/
John





We use freeswitch on solaris and just upgrade it to a new zfs which gets
remounted to the old place and freeswitch gracefully restartet. On failure
we can allways do a rollback, which takes between 2 and 10 seconds, so the
dwntime is pretty acceptable.

Michal Bielicki
Leiter der Niederlassung
HaloKwadrat Sp. z o.o.
Niederlassung Kleinmachnow
Eingetragen im Handelsregister beim Amtsgericht Potsdam, HRB21422P
Ust.Id.: DE261885536
Geschaeftsfuehrer: Aleksander Wiercinski
Meiereifeld 2b, 14532 Kleinmachnow
t. +49 33203 263220
f. +49 33203 263229 sip. i...@halokwadrat.de
e. michal.bieli...@halokwadrat.de | w. www.halokwadrat.de
Hauptgeschäftsstelle:
Halo Kwadrat Sp. z o.o.
ul. Polna 46/14
00-644 Warszawa, Polen
EIngetragen im HRB Warszawa, KRS 153539


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--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
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googletalk:conf+...@conference.freeswitch.org
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[Freeswitch-users] Dialplan extension using Caller-ID-Name not matching condition

[Larry Marshall]

http://pastebin.freeswitch.org/9365

 

I do not know what I am doing wrong. I am trying to set the
effective_caller_id_name and _number depending on the originating extension.

 

I tried:



and



and



 

But each got substituted with the name of the extension in the log:

Dialplan: sofia/internal/1...@192.168.10.29 Regex (FAIL) [Long Distance -
flowroute] () =~ /^100[09]$/ break=on-true

where the extension looks like:

 

 

Info from the log shows

 

variable_sip_from_user: [1000]

Caller-Caller-ID-Number: [1000]

 

Can anyone help? Thanks, Lars

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Re: [Freeswitch-users] Live Upgrade Techniques

[John Dalgliesh]

I assume he's talking about hardware failures here :P

But to answer the question: crashes are easy to deal with. With a crash 
you have lost the calls that are in progress anyway; you don't have to 
manage a gradual transition.

Currently, since FS is quite quick to start up, I am just relaunching it 
immediately.

But when I have a second box up and running what I'll do is just add the 
IP of the dead machine as another IP of the second box, and then it will 
take all the old machine's traffic. That is the plan anyway. I've seen 
some commercial boxes that use a similar trick.

(I've only seen one crash that wasn't my fault. Something to do with 
terminating a bridge: when the first leg gets a hangup it hangs up the 
other leg on its own thread... which can cause problems if the other leg 
was doing something funky at the time. Leads to a heap corruption. Doesn't 
happen with MALLOC_CHECK_ set so I'm just leaving it set for now :)

{P^/

On Thu, 11 Jun 2009 at 00:41 -0400, Mathieu Rene wrote:
>
> By reporting it on Jira so it doesn't crash anymore :D
>
>
> On 11-Jun-09, at 12:04 AM, Michael Giagnocavo wrote:
>
>> How are you handling your FS box crashing?
>>
>> -Original Message-
>> From: freeswitch-users-boun...@lists.freeswitch.org 
>> [mailto:freeswitch-users-boun...@lists.freeswitch.org
>> ] On Behalf Of John Dalgliesh
>> Sent: Wednesday, June 10, 2009 9:04 PM
>> To: freeswitch-users@lists.freeswitch.org
>> Subject: [Freeswitch-users] Live Upgrade Techniques
>>
>>
>> Hi,
>>
>> I am slowly gaining confidence using FreeSWITCH in production, but
>> there
>> is one issue that I'm still wondering about: how are people upgrading
>> their FreeSWITCH installation binaries without dropping all current
>> calls?
>>
>> So far I have been upgrading in the dead of night, after pausing for 5
>> minutes then dropping the stragglers, but this is hardly ideal.
>>
>> What I would like to do is to run an upgraded instance of FreeSWITCH
>> on
>> the same machine, and have it handle all new call packets, whereas
>> the old
>> instance continues to handle the existing call packets, until there
>> are no
>> more old calls left.
>>
>> I can think of about seven ways to accomplish this, but before I
>> dive into
>> the code I thought I'd better ask what everyone else has been doing :)
>>
>> (The only standard way I can think of doing this is to have a SIP
>> proxy
>> sitting in front of FS the whole time, just to handle these upgrade
>> windows. It seems like a bit of a waste.)
>>
>> So how are you handling your FS software upgrades?
>>
>> {P^/
>> John
>>
>> ___
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Re: [Freeswitch-users] Live Upgrade Techniques

[John Dalgliesh]

On Thu, 11 Jun 2009 at 10:42 -0700, Michael Collins wrote:

On Thu, Jun 11, 2009 at 10:33 AM, Michael Giagnocavo wrote:


 Exactly. You probably want to have something like this anyways, so that
when someone accidentally unplugs the system, or the disks/CPU/RAM crash,
you’re not stuck.



That is, until FreeSWITCH can record its internal state to some
inter-machine memory so we can have hot failover. ;)




I think that's going to be in 1.0.5. :)


I'm still too much of a noob to be certain that's a joke :) ... but FS 
core already does record much of its internal state... to a DB, right? It 
just has to not clear that out on startup and problem solved!


OTOH there will be a bit of trouble getting the internal state out of all 
those modules and libraries... in particular sofia :D


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Re: [Freeswitch-users] Can't hear outbound calls

[Lars Zeb]

Michael,

 

Removing everything between the  tag in
sip_profiles/internal/example.xml did the trick - no error message on FS
startup. I'm running 13723.

 

2009-06-11 07:21:03.609317 [INFO] switch_event.c:564 Activate Eventing
Engine.

2009-06-11 07:21:03.612274 [DEBUG] switch_event.c:552 Create event dispatch
thread 0

2009-06-11 07:21:03.995025 [INFO] switch_nat.c:159 Scanning for NAT

2009-06-11 07:21:03.995436 [DEBUG] switch_nat.c:127 Checking for PMP 1/5

2009-06-11 07:21:04.245056 [DEBUG] switch_nat.c:127 Checking for PMP 2/5

2009-06-11 07:21:04.246056 [DEBUG] switch_nat.c:127 Checking for PMP 3/5

2009-06-11 07:21:04.745950 [DEBUG] switch_nat.c:127 Checking for PMP 4/5

2009-06-11 07:21:05.745725 [DEBUG] switch_nat.c:127 Checking for PMP 5/5

2009-06-11 07:21:07.745256 [DEBUG] switch_nat.c:164 Checking for UPnP

2009-06-11 07:21:12.221251 [DEBUG] switch_nat.c:77 No InternetGatewayDevice,
using first entry as default.

2009-06-11 07:21:12.234867 [INFO] switch_nat.c:174 No PMP or UPnP NAT
detected!

 

Lars

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Wednesday, June 10, 2009 5:18 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Can't hear outbound calls

 

 

On Wed, Jun 10, 2009 at 4:54 PM, Lars Zeb  wrote:

Rupa,

 

I think the console log has information in it that log/freeswitch.log does
not.

 

Console:

[r...@fs bin]# ../freeswitch 

2009-06-10 16:12:50.771529 [INFO] switch_event.c:564 Activate Eventing
Engine.

2009-06-10 16:12:50.773760 [DEBUG] switch_event.c:552 Create event dispatch
thread 0

2009-06-10 16:12:50.894819 [ERR] switch_xml.c:1282 Couldnt open
/usr/local/freeswitch/conf/autoload_configs/../sip_profiles/internal/*.xml
(No such file or directory)

Error including
/usr/local/freeswitch/conf/autoload_configs/../sip_profiles/internal/*.xml
(Invalid or incomplete multibyte or wide character)

2009-06-10 16:12:51.140903 [INFO] switch_nat.c:159 Scanning for NAT

2009-06-10 16:12:51.141324 [DEBUG] switch_nat.c:127 Checking for PMP 1/5

2009-06-10 16:12:51.391941 [DEBUG] switch_nat.c:127 Checking for PMP 2/5

2009-06-10 16:12:51.891822 [DEBUG] switch_nat.c:127 Checking for PMP 3/5

2009-06-10 16:12:52.891593 [DEBUG] switch_nat.c:127 Checking for PMP 4/5

2009-06-10 16:12:54.891125 [DEBUG] switch_nat.c:127 Checking for PMP 5/5

2009-06-10 16:12:54.892134 [DEBUG] switch_nat.c:164 Checking for UPnP

2009-06-10 16:12:58.891233 [INFO] switch_nat.c:174 No PMP or UPnP NAT
detected!

 

log/freeswitch.log:

2009-06-10 16:12:09.337973 [DEBUG] switch_loadable_module.c:570 Write lock
interface 'console' to wait for existing references. (from previous
Freeswitch invocation)

2009-06-10 16:12:58.964463 [NOTICE] switch_loadable_module.c:208 Adding
Dialplan 'enum'

2009-06-10 16:12:58.964535 [NOTICE] switch_loadable_module.c:248 Adding
Application 'enum'

2009-06-10 16:12:58.964599 [NOTICE] switch_loadable_module.c:270 Adding API
Function 'enum'

2009-06-10 16:12:58.964662 [NOTICE] switch_loadable_module.c:270 Adding API
Function 'enum_auto'

2009-06-10 16:12:58.964974 [DEBUG] mod_cdr_csv.c:314 Adding default
template.

2009-06-10 16:12:58.965016 [DEBUG] mod_cdr_csv.c:359 Adding template sql.

2009-06-10 16:12:58.965032 [DEBUG] mod_cdr_csv.c:359 Adding template sql2.

2009-06-10 16:12:58.965047 [DEBUG] mod_cdr_csv.c:359 Adding template
example.

2009-06-10 16:12:58.965063 [DEBUG] mod_cdr_csv.c:359 Adding template snom.

2009-06-10 16:12:58.965079 [DEBUG] mod_cdr_csv.c:359 Adding template
linksys.

2009-06-10 16:12:58.965095 [DEBUG] mod_cdr_csv.c:359 Adding template
asterisk.

 

I disabled the DLink router UPnP mode, so no 0.0.0.0 appears in the console
log. However, the disk log begins at 16:12:58, whereas the console log
starts at 16:12:50. The console log finishes its NAT and UPnP reporting
before the disk log begins, so I wouldn't see any 0.0.0.0 if it were
present.

 

The [ERR] was due to me removing example.xml from sip_profiles/internal. I
put it back after this. I don't understand the following command in
conf/sofia.conf.xml.

 



I think this is just a cosmetic error. You could probably put an empty xml
file in sip_profiles/internal and be done with it. Or possibly have just an
empty include node, like ""

Try it out and report back - we're dying to know what happens! ;)

-MC 

 

Lars

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Rupa
Schomaker
Sent: Tuesday, June 09, 2009 6:05 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Can't hear outbound calls

 

if you haven't changed your logging, then it is probably ok.  The 0.0.0.0
thing is logged at error level, so will show up in the logs.  How did you
search?  Grep?  

grep '0\.0\.0\.0' freeswitch.log

On Tue, Jun 9, 2009 at 7:35 PM, Lars Zeb  wrote:

Rupa,

 

What options do I have for setting up logging

Re: [Freeswitch-users] Live Upgrade Techniques

[Michael Collins]

On Thu, Jun 11, 2009 at 11:24 AM, John Dalgliesh  wrote:

> On Thu, 11 Jun 2009 at 10:42 -0700, Michael Collins wrote:
>
>> On Thu, Jun 11, 2009 at 10:33 AM, Michael Giagnocavo > >wrote:
>>
>>>
>>>  Exactly. You probably want to have something like this anyways, so that
>>> when someone accidentally unplugs the system, or the disks/CPU/RAM crash,
>>> you’re not stuck.
>>>
>>>
>>>
>>> That is, until FreeSWITCH can record its internal state to some
>>> inter-machine memory so we can have hot failover. ;)
>>>
>>>
>>>
>>>  I think that's going to be in 1.0.5. :)
>>
>
> I'm still too much of a noob to be certain that's a joke :) ... but FS core
> already does record much of its internal state... to a DB, right? It just
> has to not clear that out on startup and problem solved!
>
It was a joke. :)
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Re: [Freeswitch-users] Live Upgrade Techniques

[Michael Jerris]

On Jun 11, 2009, at 2:24 PM, John Dalgliesh wrote:

> On Thu, 11 Jun 2009 at 10:42 -0700, Michael Collins wrote:
>> On Thu, Jun 11, 2009 at 10:33 AM, Michael Giagnocavo > >wrote:
>>>
>>> Exactly. You probably want to have something like this anyways, so  
>>> that
>>> when someone accidentally unplugs the system, or the disks/CPU/RAM  
>>> crash,
>>> you’re not stuck.
>>>
>>> That is, until FreeSWITCH can record its internal state to some
>>> inter-machine memory so we can have hot failover. ;)
>>>
>> I think that's going to be in 1.0.5. :)
>
> I'm still too much of a noob to be certain that's a joke :) ... but  
> FS core already does record much of its internal state... to a DB,  
> right? It just has to not clear that out on startup and problem  
> solved!
>
> OTOH there will be a bit of trouble getting the internal state out  
> of all those modules and libraries... in particular sofia :D

We have talked quite some about this, its a major job, easily months  
of work for multiple programmers.  We would love to do it but its not  
on any roadmaps at this time.

Mike


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Re: [Freeswitch-users] Dialplan extension using Caller-ID-Name not matching condition

[Brian West]

try destination_number

/b

On Jun 11, 2009, at 1:04 PM, Larry Marshall wrote:


http://pastebin.freeswitch.org/9365

I do not know what I am doing wrong. I am trying to set the  
effective_caller_id_name and _number depending on the originating  
extension.


I tried:
expression="^100[09]$" break="on-true">

and
expression="^100[09]$" break="on-true">

and
expression="^100[09]$" break="on-true">




Brian West
br...@freeswitch.org

-- Meet us at ClueCon!  http://www.cluecon.com




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Re: [Freeswitch-users] Dialplan extension using Caller-ID-Name not matching condition

[Mathieu Rene]

Your syntax is also wrong, it should be



and NOT field=${varname}$

Math

On 11-Jun-09, at 2:07 PM, Brian West wrote:


try destination_number

/b

On Jun 11, 2009, at 1:04 PM, Larry Marshall wrote:


http://pastebin.freeswitch.org/9365

I do not know what I am doing wrong. I am trying to set the  
effective_caller_id_name and _number depending on the originating  
extension.


I tried:
expression="^100[09]$" break="on-true">

and
expression="^100[09]$" break="on-true">

and
expression="^100[09]$" break="on-true">




Brian West
br...@freeswitch.org

-- Meet us at ClueCon!  http://www.cluecon.com




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Re: [Freeswitch-users] Live Upgrade Techniques

[Kristian Kielhofner]

That's exactly what I do.

Between dispatcher and FLAGS/GFLAGS this is easy to do in OpenSIPS/SER.

On Thu, Jun 11, 2009 at 12:54 PM, Anthony
Minessale wrote:
> or you can put a sip proxy in front of 2 boxes where you can control the
> flow of traffic.
> when you want to upgrade one, take all the traffic off of it by forcing all
> calls to the other box, upgrade it then shift the traffic to the new one.
> if that goes well, upgrade the other one too.
>

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
http://www.voalte.com

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[Freeswitch-users] Dialplan XML phone number match

[Lars Zeb]

I have a match expression for outbound calls as "\d{10}". It's fine for
unformatted numbers. Not knowing any better, I created another extension to
handle numbers formatted like XXX-XXX-, which is easier to read and
exists in one hard phone's phonebook.

 

It looks like: "^1?(\d{3})-(\d{3})-(\d{4})$". But I can see making many
extensions for different formats.

 

There's got to be a better way. Any suggestions?

 

Thanks, Lars

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Re: [Freeswitch-users] Dialplan XML phone number match

[Brian West]

Don't do it!  Doing that stuff is highly silly.

/b

On Jun 11, 2009, at 2:49 PM, Lars Zeb wrote:


There’s got to be a better way. Any suggestions?


Brian West
br...@freeswitch.org

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[Freeswitch-users] re direct calls

[NOx-WHV]

Hello Freeswitch User!

I am using FS since a few weeks. My intent is to have clients who uses TLS
and SRTP for a full encrypted call.

I just managed it, that calls are encrypted with TLS and SRTP. My second aim
ist to redirect this calls for reduce the processing of the server. I only
use the FS for calls between users of this freeswitch. I just tested a
"redirect" in the dialplan (without TLS and SRTP) but it doesn´t work. How i
have to configure the dialplan for redirect the call to the other user.

I use a SNOM hardphone and a phonerlite softphone. 

Thanks for sour help!

NOX
-- 
View this message in context: 
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Re: [Freeswitch-users] re direct calls

[Brian West]

On Jun 11, 2009, at 3:04 PM, NOx-WHV wrote:



Hello Freeswitch User!

I am using FS since a few weeks. My intent is to have clients who  
uses TLS

and SRTP for a full encrypted call.

I just managed it, that calls are encrypted with TLS and SRTP. My  
second aim
ist to redirect this calls for reduce the processing of the server.  
I only

use the FS for calls between users of this freeswitch. I just tested a
"redirect" in the dialplan (without TLS and SRTP) but it doesn´t  
work. How i
have to configure the dialplan for redirect the call to the other  
user.


You can't.  Its not possible because we are a b2bua and you have  
already negotiated the keys between the endpoints and FreeSWITCH and  
when you redirect the media neither phone can decrypt the packets  
correctly.




I use a SNOM hardphone and a phonerlite softphone.

Thanks for sour help!


Brian West
br...@freeswitch.org

-- Meet us at ClueCon!  http://www.cluecon.com




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Re: [Freeswitch-users] Live Upgrade Techniques

[Michael Giagnocavo]

Well, if you're running multiple machines, waiting for it to drainstop isn't 
that big of a deal unless you're in some sort of hurry, right? Give it an hour 
or so to drainstop, then kill 'em. 

Would it not be simpler to try to do something with re-invites or REFER, 
assuming your endpoints support it?

-Michael

-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org 
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of John 
Dalgliesh
Sent: Thursday, June 11, 2009 12:14 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Live Upgrade Techniques


I assume he's talking about hardware failures here :P

But to answer the question: crashes are easy to deal with. With a crash 
you have lost the calls that are in progress anyway; you don't have to 
manage a gradual transition.

Currently, since FS is quite quick to start up, I am just relaunching it 
immediately.

But when I have a second box up and running what I'll do is just add the 
IP of the dead machine as another IP of the second box, and then it will 
take all the old machine's traffic. That is the plan anyway. I've seen 
some commercial boxes that use a similar trick.

(I've only seen one crash that wasn't my fault. Something to do with 
terminating a bridge: when the first leg gets a hangup it hangs up the 
other leg on its own thread... which can cause problems if the other leg 
was doing something funky at the time. Leads to a heap corruption. Doesn't 
happen with MALLOC_CHECK_ set so I'm just leaving it set for now :)

{P^/

On Thu, 11 Jun 2009 at 00:41 -0400, Mathieu Rene wrote:
>
> By reporting it on Jira so it doesn't crash anymore :D
>
>
> On 11-Jun-09, at 12:04 AM, Michael Giagnocavo wrote:
>
>> How are you handling your FS box crashing?
>>
>> -Original Message-
>> From: freeswitch-users-boun...@lists.freeswitch.org 
>> [mailto:freeswitch-users-boun...@lists.freeswitch.org
>> ] On Behalf Of John Dalgliesh
>> Sent: Wednesday, June 10, 2009 9:04 PM
>> To: freeswitch-users@lists.freeswitch.org
>> Subject: [Freeswitch-users] Live Upgrade Techniques
>>
>>
>> Hi,
>>
>> I am slowly gaining confidence using FreeSWITCH in production, but
>> there
>> is one issue that I'm still wondering about: how are people upgrading
>> their FreeSWITCH installation binaries without dropping all current
>> calls?
>>
>> So far I have been upgrading in the dead of night, after pausing for 5
>> minutes then dropping the stragglers, but this is hardly ideal.
>>
>> What I would like to do is to run an upgraded instance of FreeSWITCH
>> on
>> the same machine, and have it handle all new call packets, whereas
>> the old
>> instance continues to handle the existing call packets, until there
>> are no
>> more old calls left.
>>
>> I can think of about seven ways to accomplish this, but before I
>> dive into
>> the code I thought I'd better ask what everyone else has been doing :)
>>
>> (The only standard way I can think of doing this is to have a SIP
>> proxy
>> sitting in front of FS the whole time, just to handle these upgrade
>> windows. It seems like a bit of a waste.)
>>
>> So how are you handling your FS software upgrades?
>>
>> {P^/
>> John
>>
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[Freeswitch-users] Orphaned calls

[Nik Middleton]

 

Ok, so I did a mere 86,000 calls today, but when it was all over, I had
6 sessions remaining like the one below (number and ISP changed)

 

Anyone have an idea why these 6 sessions remain?   I also had 120 calls
that I didn't get a hang-up for, but that might be me not processing the
events fast enough.

 

That said, FS was handling a steady concurrent call level of around 350
which was awesome !!

 

Regards

 

 

UP 0 years, 0 days, 9 hours, 39 minutes, 10 seconds, 533 milliseconds,
400 microseconds

86913 session(s) since startup

 

f47e08a2-04b7-4faf-86bc-a57410f07676,inbound,2009-06-11
18:38:23,1244741903,sofia/external/0x...@gk.myisp.net,CS_NEW
,

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Re: [Freeswitch-users] re direct calls

[NOx-WHV]

Thanks for your answer.

Can you just announce b2bua. 



Brian West-3 wrote:
> 
> 
> On Jun 11, 2009, at 3:04 PM, NOx-WHV wrote:
> 
>>
>> Hello Freeswitch User!
>>
>> I am using FS since a few weeks. My intent is to have clients who  
>> uses TLS
>> and SRTP for a full encrypted call.
>>
>> I just managed it, that calls are encrypted with TLS and SRTP. My  
>> second aim
>> ist to redirect this calls for reduce the processing of the server.  
>> I only
>> use the FS for calls between users of this freeswitch. I just tested a
>> "redirect" in the dialplan (without TLS and SRTP) but it doesn´t  
>> work. How i
>> have to configure the dialplan for redirect the call to the other  
>> user.
> 
> You can't.  Its not possible because we are a b2bua and you have  
> already negotiated the keys between the endpoints and FreeSWITCH and  
> when you redirect the media neither phone can decrypt the packets  
> correctly.
> 
>>
>> I use a SNOM hardphone and a phonerlite softphone.
>>
>> Thanks for sour help!
> 
> Brian West
> br...@freeswitch.org
> 
> -- Meet us at ClueCon!  http://www.cluecon.com
> 
> 
> 
> 
> 
> ___
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> 

-- 
View this message in context: 
http://www.nabble.com/redirect-calls-tp23982889p23989162.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.


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Re: [Freeswitch-users] Caller id when doing transfers

[John Wehle]

> It appears from some limited testing that the original caller id is always
> shown when the call is transfered.  Is there some way to have the person
> making the transfer show up as the caller id?

To answer my own question it appears that the information is available
in the sip_h_Referred-By variable.  E.g.:

  
 
 

allows the station id making the transfer to be known when a call is
transfered to *5.  The station id can then be used to park the call in
the proper fifo.

-- John
-
|   Feith Systems  |   Voice: 1-215-646-8000  |  Email: j...@feith.com  |
|John Wehle| Fax: 1-215-540-5495  | |
-


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Re: [Freeswitch-users] Orphaned calls

[Michael Collins]

On Thu, Jun 11, 2009 at 2:07 PM, Nik Middleton <
nik.middle...@noblesolutions.co.uk> wrote:

>
>
> Ok, so I did a mere 86,000 calls today, but when it was all over, I had 6
> sessions remaining like the one below (number and ISP changed)
>
>
>
> Anyone have an idea why these 6 sessions remain?   I also had 120 calls
> that I didn’t get a hang-up for, but that might be me not processing the
> events fast enough.
>
>
>
Do they show on "show calls"? Or do they show up on "show channels" only?
Just curious to see if they were bridged or not.
-MC

> That said, FS was handling a steady concurrent call level of around 350
> which was awesome !!
>
>
>
> Regards
>
>
>
>
>
> UP 0 years, 0 days, 9 hours, 39 minutes, 10 seconds, 533 milliseconds, 400
> microseconds
>
> 86913 session(s) since startup
>
>
>
> f47e08a2-04b7-4faf-86bc-a57410f07676,inbound,2009-06-11
> 18:38:23,1244741903,sofia/external/0x...@gk.myisp.net
> ,CS_NEW,
>
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Re: [Freeswitch-users] Dialplan XML phone number match

[Michael Collins]

On Thu, Jun 11, 2009 at 12:49 PM, Lars Zeb  wrote:

>  I have a match expression for outbound calls as “\d{10}”. It’s fine for
> unformatted numbers. Not knowing any better, I created another extension to
> handle numbers formatted like XXX-XXX-, which is easier to read and
> exists in one hard phone’s phonebook.
>
>
>
> It looks like: “^1?(\d{3})-(\d{3})-(\d{4})$”. But I can see making many
> extensions for different formats.
>

Out of curiosity, what benefit does having all these formats get you?
-MC
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Re: [Freeswitch-users] Caller id when doing transfers

[Michael Collins]

On Thu, Jun 11, 2009 at 2:17 PM, John Wehle  wrote:

> > It appears from some limited testing that the original caller id is
> always
> > shown when the call is transfered.  Is there some way to have the person
> > making the transfer show up as the caller id?
>
> To answer my own question it appears that the information is available
> in the sip_h_Referred-By variable.  E.g.:
>
>  
> 
>  expression="^
>
> allows the station id making the transfer to be known when a call is
> transfered to *5.  The station id can then be used to park the call in
> the proper fifo.


John,

Would you be willing to add this wonderful knowledge to the wiki? :) Let me
know if you have any questions about where/how to add it and we'll come up
with something.
-MC
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Re: [Freeswitch-users] re direct calls

[Brian West]

what?

On Jun 11, 2009, at 4:16 PM, NOx-WHV wrote:


Can you just announce b2bua.


Brian West
br...@freeswitch.org

-- Meet us at ClueCon!  http://www.cluecon.com




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Re: [Freeswitch-users] Caller id when doing transfers

[Brian West]

It does if you do a blind transfer... if you're talking attended  
transfers thats a whole different ball of wax...


/b

On Jun 11, 2009, at 4:17 PM, John Wehle wrote:

It appears from some limited testing that the original caller id is  
always
shown when the call is transfered.  Is there some way to have the  
person

making the transfer show up as the caller id?


To answer my own question it appears that the information is available
in the sip_h_Referred-By variable.  E.g.:

 




allows the station id making the transfer to be known when a call is
transfered to *5.  The station id can then be used to park the call in
the proper fifo.


Brian West
br...@freeswitch.org

-- Meet us at ClueCon!  http://www.cluecon.com




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Re: [Freeswitch-users] Orphaned calls

[Brian West]

You could also attach to it with GDB and see if its hanging somewhere  
else.


/b

On Jun 11, 2009, at 4:20 PM, Michael Collins wrote:

Do they show on "show calls"? Or do they show up on "show channels"  
only? Just curious to see if they were bridged or not.

-MC


Brian West
br...@freeswitch.org

-- Meet us at ClueCon!  http://www.cluecon.com




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Re: [Freeswitch-users] Orphaned calls

[Anthony Minessale]

it may be a race in the sql event handler where the delete comes before the
insert on a really short call.
how many sessions did "status" report were in use?

On Thu, Jun 11, 2009 at 4:07 PM, Nik Middleton <
nik.middle...@noblesolutions.co.uk> wrote:

>
>
> Ok, so I did a mere 86,000 calls today, but when it was all over, I had 6
> sessions remaining like the one below (number and ISP changed)
>
>
>
> Anyone have an idea why these 6 sessions remain?   I also had 120 calls
> that I didn’t get a hang-up for, but that might be me not processing the
> events fast enough.
>
>
>
> That said, FS was handling a steady concurrent call level of around 350
> which was awesome !!
>
>
>
> Regards
>
>
>
>
>
> UP 0 years, 0 days, 9 hours, 39 minutes, 10 seconds, 533 milliseconds, 400
> microseconds
>
> 86913 session(s) since startup
>
>
>
> f47e08a2-04b7-4faf-86bc-a57410f07676,inbound,2009-06-11
> 18:38:23,1244741903,sofia/external/0x...@gk.myisp.net
> ,CS_NEW,
>
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>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

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
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Re: [Freeswitch-users] re direct calls

[NOx-WHV]

back to back user agent! :-)

Thanks! I just ask google!



NOx-WHV wrote:
> 
> Thanks for your answer.
> 
> Can you just announce b2bua. 
> 
> 
> 
> Brian West-3 wrote:
>> 
>> 
>> On Jun 11, 2009, at 3:04 PM, NOx-WHV wrote:
>> 
>>>
>>> Hello Freeswitch User!
>>>
>>> I am using FS since a few weeks. My intent is to have clients who  
>>> uses TLS
>>> and SRTP for a full encrypted call.
>>>
>>> I just managed it, that calls are encrypted with TLS and SRTP. My  
>>> second aim
>>> ist to redirect this calls for reduce the processing of the server.  
>>> I only
>>> use the FS for calls between users of this freeswitch. I just tested a
>>> "redirect" in the dialplan (without TLS and SRTP) but it doesn´t  
>>> work. How i
>>> have to configure the dialplan for redirect the call to the other  
>>> user.
>> 
>> You can't.  Its not possible because we are a b2bua and you have  
>> already negotiated the keys between the endpoints and FreeSWITCH and  
>> when you redirect the media neither phone can decrypt the packets  
>> correctly.
>> 
>>>
>>> I use a SNOM hardphone and a phonerlite softphone.
>>>
>>> Thanks for sour help!
>> 
>> Brian West
>> br...@freeswitch.org
>> 
>> -- Meet us at ClueCon!  http://www.cluecon.com
>> 
>> 
>> 
>> 
>> 
>> ___
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>> 
>> 
> 
> 

-- 
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Re: [Freeswitch-users] Orphaned calls

[Nik Middleton]

It was the output from show channels.  I've rebooted the server now, so
I can't run show calls.  I'll see what happens tomorrow.  Certainly
running status showed 6 sessions

 

All calls are initiated using and 'Originate' from an inbound socket

 

Regards

 

 

 



From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Michael Collins
Sent: 11 June 2009 22:20
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Orphaned calls

 

 

On Thu, Jun 11, 2009 at 2:07 PM, Nik Middleton
 wrote:

 

Ok, so I did a mere 86,000 calls today, but when it was all over, I had
6 sessions remaining like the one below (number and ISP changed)

 

Anyone have an idea why these 6 sessions remain?   I also had 120 calls
that I didn't get a hang-up for, but that might be me not processing the
events fast enough.

 

Do they show on "show calls"? Or do they show up on "show channels"
only? Just curious to see if they were bridged or not.
-MC 

That said, FS was handling a steady concurrent call level of
around 350 which was awesome !!

 

Regards

 

 

UP 0 years, 0 days, 9 hours, 39 minutes, 10 seconds, 533
milliseconds, 400 microseconds

86913 session(s) since startup

 

f47e08a2-04b7-4faf-86bc-a57410f07676,inbound,2009-06-11
18:38:23,1244741903,sofia/external/0x...@gk.myisp.net,CS_NEW
,


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Re: [Freeswitch-users] Orphaned calls

[Michael Jerris]

If they were still showing in status, can you use gcore to dump a core  
next time this happens, leave it running somewhere we can get to it  
and post a thread apply all bt to Jira.


Mike

On Jun 11, 2009, at 5:40 PM, "Nik Middleton" > wrote:


It was the output from show channels.  I’ve rebooted the server now, 
 so I can’t run show calls.  I’ll see what happens tomorrow.  Certai 
nly running status showed 6 sessions




All calls are initiated using and ‘Originate’ from an inbound socket



Regards







From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org 
] On Behalf Of Michael Collins

Sent: 11 June 2009 22:20
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Orphaned calls





On Thu, Jun 11, 2009 at 2:07 PM, Nik Middleton > wrote:




Ok, so I did a mere 86,000 calls today, but when it was all over, I  
had 6 sessions remaining like the one below (number and ISP changed)




Anyone have an idea why these 6 sessions remain?   I also had 120  
calls that I didn’t get a hang-up for, but that might be me not proc 
essing the events fast enough.




Do they show on "show calls"? Or do they show up on "show channels"  
only? Just curious to see if they were bridged or not.

-MC

That said, FS was handling a steady concurrent call level of around  
350 which was awesome !!




Regards





UP 0 years, 0 days, 9 hours, 39 minutes, 10 seconds, 533  
milliseconds, 400 microseconds


86913 session(s) since startup



f47e08a2-04b7-4faf-86bc-a57410f07676,inbound,2009-06-11  
18:38:23,1244741903,sofia/external/ 
0x...@gk.myisp.net,CS_NEW,



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[Freeswitch-users] Status Event

[Nik Middleton]

Not sure where enhancement requests should be posted, but here it is
anyway

 

 

I would dearly love to be able to send a status event that returns an
event style output that provides machine readable output rather than the
wordy human readable response. (I hate parsing)

 

Is there such an event already?

 

Regards

 

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Re: [Freeswitch-users] Orphaned calls

[Nik Middleton]

Will do

 

Regards,

 



From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Michael Jerris
Sent: 11 June 2009 22:51
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Orphaned calls

 

If they were still showing in status, can you use gcore to dump a core
next time this happens, leave it running somewhere we can get to it and
post a thread apply all bt to Jira.

 

Mike

On Jun 11, 2009, at 5:40 PM, "Nik Middleton"
 wrote:

It was the output from show channels.  I've rebooted the server
now, so I can't run show calls.  I'll see what happens tomorrow.
Certainly running status showed 6 sessions

 

All calls are initiated using and 'Originate' from an inbound
socket

 

Regards

 

 

 





From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Michael Collins
Sent: 11 June 2009 22:20
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Orphaned calls

 

 

On Thu, Jun 11, 2009 at 2:07 PM, Nik Middleton <

nik.middle...@noblesolutions.co.uk> wrote:

 

Ok, so I did a mere 86,000 calls today, but when it was all
over, I had 6 sessions remaining like the one below (number and ISP
changed)

 

Anyone have an idea why these 6 sessions remain?   I also had
120 calls that I didn't get a hang-up for, but that might be me not
processing the events fast enough.

 

Do they show on "show calls"? Or do they show up on "show
channels" only? Just curious to see if they were bridged or not.
-MC 

That said, FS was handling a steady concurrent call
level of around 350 which was awesome !!

 

Regards

 

 

UP 0 years, 0 days, 9 hours, 39 minutes, 10 seconds, 533
milliseconds, 400 microseconds

86913 session(s) since startup

 

f47e08a2-04b7-4faf-86bc-a57410f07676,inbound,2009-06-11
18:38:23,1244741903,sofia/external/ 
0x...@gk.myisp.net,CS_NEW,


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Re: [Freeswitch-users] Dialplan XML phone number match

[Lars Zeb]

The users entering numbers into their phonebooks are able to recognize the
number more easily.

 

I will tell them to forget it and make the phone numbers numeric only.

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Thursday, June 11, 2009 2:21 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Dialplan XML phone number match

 

 

On Thu, Jun 11, 2009 at 12:49 PM, Lars Zeb  wrote:

I have a match expression for outbound calls as "\d{10}". It's fine for
unformatted numbers. Not knowing any better, I created another extension to
handle numbers formatted like XXX-XXX-, which is easier to read and
exists in one hard phone's phonebook.

 

It looks like: "^1?(\d{3})-(\d{3})-(\d{4})$". But I can see making many
extensions for different formats.


Out of curiosity, what benefit does having all these formats get you?
-MC

 

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Re: [Freeswitch-users] Dialplan XML phone number match

[Brian West]

If the phone sends them with dashes in them the phone IS BROKEN and  
should be smashed with a hammer.


/b

On Jun 11, 2009, at 5:57 PM, Lars Zeb wrote:

The users entering numbers into their phonebooks are able to  
recognize the number more easily.


I will tell them to forget it and make the phone numbers numeric only.


Brian West
br...@freeswitch.org

-- Meet us at ClueCon!  http://www.cluecon.com




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Re: [Freeswitch-users] Status Event

[João Mesquita]

Nik, I am a noobie and all, but most API responses can come as xml just by
adding "as xml" at the end of the call.

jmesquita

On Thu, Jun 11, 2009 at 6:57 PM, Nik Middleton <
nik.middle...@noblesolutions.co.uk> wrote:

>  Not sure where enhancement requests should be posted, but here it is
> anyway
>
>
>
>
>
> I would dearly love to be able to send a status event that returns an event
> style output that provides machine readable output rather than the wordy
> human readable response. (I hate parsing)
>
>
>
> Is there such an event already?
>
>
>
> Regards
>
>
>
> ___
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>
>
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Re: [Freeswitch-users] Status Event

[Brian West]

Only if they have an as xml modifier

/b

On Jun 11, 2009, at 6:25 PM, João Mesquita wrote:

Nik, I am a noobie and all, but most API responses can come as xml  
just by adding "as xml" at the end of the call.


jmesquita


Brian West
br...@freeswitch.org

-- Meet us at ClueCon!  http://www.cluecon.com




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Re: [Freeswitch-users] Dialplan XML phone number match

[Lars Zeb]

It's a SNOM 320.

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian
West
Sent: Thursday, June 11, 2009 4:05 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Dialplan XML phone number match

 

If the phone sends them with dashes in them the phone IS BROKEN and should
be smashed with a hammer.

 

/b

 

On Jun 11, 2009, at 5:57 PM, Lars Zeb wrote:





The users entering numbers into their phonebooks are able to recognize the
number more easily.

 

I will tell them to forget it and make the phone numbers numeric only.

 

Brian West

br...@freeswitch.org

 

-- Meet us at ClueCon!  http://www.cluecon.com  

 

 

 

 

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Re: [Freeswitch-users] Dialplan XML phone number match

[Brian West]

What firmware?

/b

On Jun 11, 2009, at 7:30 PM, Lars Zeb wrote:


It’s a SNOM 320.



Brian West
br...@freeswitch.org

-- Meet us at ClueCon!  http://www.cluecon.com




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Re: [Freeswitch-users] Dialplan XML phone number match

[Lars Zeb]

snom320-SIP 6.5.17.

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian
West
Sent: Thursday, June 11, 2009 5:40 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Dialplan XML phone number match

 

What firmware?

 

/b

 

On Jun 11, 2009, at 7:30 PM, Lars Zeb wrote:





It's a SNOM 320.

 

 

Brian West

br...@freeswitch.org

 

-- Meet us at ClueCon!  http://www.cluecon.com  

 

 

 

 

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Re: [Freeswitch-users] Dialplan XML phone number match

[Brian West]

You should be running 7.1.35 or higher.

/b

On Jun 11, 2009, at 8:34 PM, Lars Zeb wrote:


snom320-SIP 6.5.17.


Brian West
br...@freeswitch.org

-- Meet us at ClueCon!  http://www.cluecon.com




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Re: [Freeswitch-users] Dialplan XML phone number match

[Lars Zeb]

snom320-SIP 7.3.14 14953 fixed it. The phonebook stores XXX-XXX- but
delivers XX to FS.

 

Thanks Brian

 

From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian
West
Sent: Thursday, June 11, 2009 6:41 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Dialplan XML phone number match

 

You should be running 7.1.35 or higher.

 

/b

 

On Jun 11, 2009, at 8:34 PM, Lars Zeb wrote:





snom320-SIP 6.5.17.

 

Brian West

br...@freeswitch.org

 

-- Meet us at ClueCon!  http://www.cluecon.com  

 

 

 

 

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Re: [Freeswitch-users] Dialplan XML phone number match

[Brian West]

See I knew that was a bit of crack :P, Good to hear its working like  
it SHOULD now!


/b

On Jun 11, 2009, at 9:21 PM, Lars Zeb wrote:

snom320-SIP 7.3.14 14953 fixed it. The phonebook stores XXX-XXX-  
but delivers XX to FS.


Thanks Brian


Brian West
br...@freeswitch.org

-- Meet us at ClueCon!  http://www.cluecon.com




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Re: [Freeswitch-users] Live Upgrade Techniques

[John Dalgliesh]

Hi,

On Thu, 11 Jun 2009 at 16:33 -0400, Michael Giagnocavo wrote:
>
> Well, if you're running multiple machines, waiting for it to drainstop 
> isn't that big of a deal unless you're in some sort of hurry, right? 
> Give it an hour or so to drainstop, then kill 'em.

Yes that's exactly what I'm trying to do. The problem is some people will 
only try one IP address.

> Would it not be simpler to try to do something with re-invites or REFER, 
> assuming your endpoints support it?

That was actually plan A. I already added a property in sip_profile called 
failover_redirect, which specifies another server to try if FS can't 
allocate any more sessions (e.g. too busy, paused, shutdown asap, etc.), 
by sending back a SIP 302 Moved Temporarily response, instead of 503 Max 
Calls In Progress.

Turns out not all my endpoints support it :(

I considered REFER too but there seems to be even less support for that.

If I can't get the socket-sharing upgrade working then I will fall back to 
this - and peers which don't support the 302 response (or more likely, 
don't authorise it) will just get no service during the upgrade.

> -Michael

{P^/

> -Original Message-
> From: freeswitch-users-boun...@lists.freeswitch.org 
> [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of John 
> Dalgliesh
> Sent: Thursday, June 11, 2009 12:14 PM
> To: freeswitch-users@lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Live Upgrade Techniques
>
>
> I assume he's talking about hardware failures here :P
>
> But to answer the question: crashes are easy to deal with. With a crash
> you have lost the calls that are in progress anyway; you don't have to
> manage a gradual transition.
>
> Currently, since FS is quite quick to start up, I am just relaunching it
> immediately.
>
> But when I have a second box up and running what I'll do is just add the
> IP of the dead machine as another IP of the second box, and then it will
> take all the old machine's traffic. That is the plan anyway. I've seen
> some commercial boxes that use a similar trick.
>
...

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Re: [Freeswitch-users] Live Upgrade Techniques

[Brian West]

On Jun 11, 2009, at 10:35 PM, John Dalgliesh wrote:



Hi,

On Thu, 11 Jun 2009 at 16:33 -0400, Michael Giagnocavo wrote:


Well, if you're running multiple machines, waiting for it to  
drainstop

isn't that big of a deal unless you're in some sort of hurry, right?
Give it an hour or so to drainstop, then kill 'em.


Yes that's exactly what I'm trying to do. The problem is some people  
will

only try one IP address.


Clients that don't properly implement SRV/NAPTR and fail over need to  
be smacked.  :)  (not customers but software that fails to do that)





Would it not be simpler to try to do something with re-invites or  
REFER,

assuming your endpoints support it?


That was actually plan A. I already added a property in sip_profile  
called

failover_redirect, which specifies another server to try if FS can't
allocate any more sessions (e.g. too busy, paused, shutdown asap,  
etc.),
by sending back a SIP 302 Moved Temporarily response, instead of 503  
Max

Calls In Progress.


You can't send a 302 to a call thats already established.



Turns out not all my endpoints support it :(


AKA broken endpoints.  :)



I considered REFER too but there seems to be even less support for  
that.


ACK really?  thats sad!



If I can't get the socket-sharing upgrade working then I will fall  
back to

this - and peers which don't support the 302 response (or more likely,
don't authorise it) will just get no service during the upgrade.


-Michael




Brian West
br...@freeswitch.org

-- Meet us at ClueCon!  http://www.cluecon.com




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Re: [Freeswitch-users] Live Upgrade Techniques

[Michael Giagnocavo]

>> Well, if you're running multiple machines, waiting for it to drainstop 
>> isn't that big of a deal unless you're in some sort of hurry, right? 
>> Give it an hour or so to drainstop, then kill 'em.
>
>Yes that's exactly what I'm trying to do. The problem is some people will 
>only try one IP address.

Right, so if you have a proxy in front that is handling this, it should be no 
problem. 

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[Freeswitch-users] MPL Confusion

[Muhammad Shahzad]

Hi,

I have some confusion about FreeSWITCH's Mozilla Public License 1.1. I do
understand that me or any one can change provided code according to our
customization needs and we are not bound to share our changes as long as we
are not distributing it, right?

Now, i have been doing R&D on MSN and Yahoo voice chat services, I have now
completed by research and now would like to write up FS modules to
communicate with these servers. But as you all know both MSN and Yahoo
provide SIP based VOIP services, however they are not using standard SIP
stack and have their own versions of customized SIP stack. So, in order to
write an endpoint for these servers, instead of writing everything from
scretch, i can using existing mod_sofia endpoint and customize it to make it
compatible with MSN and Yahoo SIP stack. So here are my questions,

1. Is it possible under MPL, that i make a copy of mod_sofia as say mod_msn
and develop it to work with MSN, similarly mod_yahoo for Yahoo voice chat
service?
2. If yes, how can i mention my role in these modules development, i.e. as
developer or as contributor?

Also i wish to include my work, once completed, in FreeSWITCH, can you
provide me the guidelines and / or eligibility criteria to do so, any link
on FS site etc.?

Thank you.


-- 
Muhammad Shahzad
---
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +92 334 422 40 88
MSN: shari_78...@hotmail.com
Email: shaherya...@googlemail.com
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Re: [Freeswitch-users] Live Upgrade Techniques

[John Dalgliesh]

On Thu, 11 Jun 2009 at 22:57 -0500, Brian West wrote:
> On Jun 11, 2009, at 10:35 PM, John Dalgliesh wrote:
>> On Thu, 11 Jun 2009 at 16:33 -0400, Michael Giagnocavo wrote:
>>> 
>>> Well, if you're running multiple machines, waiting for it to drainstop
>>> isn't that big of a deal unless you're in some sort of hurry, right?
>>> Give it an hour or so to drainstop, then kill 'em.
>> 
>> Yes that's exactly what I'm trying to do. The problem is some people will
>> only try one IP address.
>
> Clients that don't properly implement SRV/NAPTR and fail over need to be 
> smacked.  :)  (not customers but software that fails to do that)

Yes I'm sure much of their software can do this but it has been set up for 
static numeric IPs. And getting the IP changed is a week-long process for 
some customers!

>>> Would it not be simpler to try to do something with re-invites or REFER,
>>> assuming your endpoints support it?
>> 
>> That was actually plan A. I already added a property in sip_profile called
>> failover_redirect, which specifies another server to try if FS can't
>> allocate any more sessions (e.g. too busy, paused, shutdown asap, etc.),
>> by sending back a SIP 302 Moved Temporarily response, instead of 503 Max
>> Calls In Progress.
>
> You can't send a 302 to a call thats already established.

Yes and I don't want to touch established calls - those calls can stay 
there until they drop. This is sent to new requests when 
switch_core_session_request fails in mod_sofia.

>> Turns out not all my endpoints support it :(
>
> AKA broken endpoints.  :)

Some are broken. Some just have this feature disabled. For 'security 
reasons'. You know the drill.


{P^/
John

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