Re: [Freeswitch-users] Presence across several networked FSs

[Jon Bruel]

I have added an example on the wiki illustrating how to propagate presence and 
registrations over a set of networked FSs. Interested? Find it on: 
http://wiki.freeswitch.org/wiki/Mod_event_multicast. /Jon

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[Freeswitch-users] mod_xml_curl and gateways

[Jon Bruel]

I wonder if it is possible to define common gateways (not user specific 
gateways) by xml_curl, and if so, the bindings and syntax to use?

All the best /Jon


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Re: [Freeswitch-users] Presence across several networked FSs

[Jon Bruel]

I have found some ways to get presence, or rather BLF functions to work on Snom 
telephones in a distributed network with several FSs. I'll post a solution on 
the wiki when I have tested it further.

Anyhow, I'm using the mod_event_multicast module with the following 
configuration:


  



  


With this setting on all FSs, the registration table is also automatically 
updated thus listing all sets registered across all FSs. In the table 
sip_registrations (under the database for the profile used), the field status 
has the value: "Registered" if the UA is registered on another FS and the value 
"Registered(UDP)" if the UA is registered on the same FS. The field 
server_host, however, is the ip-address of "local" FS.

Now comes the question: is there any way to let the field server_host show the 
server address of the server actually registered to? Or any other way using the 
existing modules to get the information about which FS the UAs are registered 
to? The information is going to be used for the routing decisions between 
networked FSs.

/Jon

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[Freeswitch-users] Presence across several networked FSs

[Jon Bruel]

I'm working on a setup with several networked FreeSWITCHes based on a central 
FS and one or more satellite FSs. The boxes are connected to each other with 
IAX, SIP over VPN or another protocol. In the choice of the protocol the 
presence issue mentioned below should be considered, and the real life 
practicalities such as routers, NAT and connections with packet loss.

I find the directory facility as a good tool for expressing the topology on 
each server, where the dial-string in the directory can be used to beak out of 
the box into some of the other boxes via the central unit.  When I dial out 
from a phone, the presence information is updated OK on same phone and other 
phones on the same box.

I would like to keep track of the presence of the users on the other boxes as 
well. Any suggestion on to how the presence information scan be propagated to 
all boxes in the network.

Jon Brüel
Consiglia Telecommunications
DK-2960 Rungsted Kyst
Tel: +45 45 16 1000
Mob: +45 26 15 30 60
CVR: 27047882


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Re: [Freeswitch-users] Lua and database access to core_db

[Jon Bruel]

Point taken Anthony. Naturally you are not going to work for me for free.

But I'm a bit confused about the statement that "I'm rude". That's not my 
purpose to be. And I certainly do hope that this is not just a question of a 
cultural clash between an elderly man with a Phd in black holes from a European 
background and a young American FS genius.

But frankly, I did believe that focus regarding changes and new developments 
was somewhat guided by the input you get from the users list, including changes 
which makes the FS easier to access for newbies, but maybe I'm wrong.

That's my last comment, hope we can continue the exchange of views in a good 
spirit.


Jon Brüel
Consiglia Telecommunications
DK-2960 Rungsted Kyst
Tel: +45 45 16 1000
Mob: +45 26 15 30 60
CVR: 27047882


From: freeswitch-users-boun...@lists.freeswitch.org 
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Anthony 
Minessale
Sent: 8. december 2009 17:28
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Lua and database access to core_db

One last bit of free consulting advice for you:

You are again being rude because you want us to work for you for free.
The code is free sir, the support here is voluntary and based on our 
willingness to help and comments like that are all it takes to get us to ignore 
you completely.

On Tue, Dec 8, 2009 at 5:46 AM, Jon Bruel 
mailto:j...@consiglia.dk>> wrote:
I got the combination Lua with direct access to the core Sqlite database to 
work. Hurray, maybe I'm not as stupid as A.M II hints...
The problem was that Lua did not "like":

require "luasql.sqlite"
env = luasql.sqlite()
con = assert(env:connect("/usr/local/freeswitch/db/core.db"))

After changing it to

require "luasql.sqlite3"
env = luasql.sqlite3()
con = assert(env:connect("/usr/local/freeswitch/db/core.db"))

And seeing that there was a symlink in one of the right directories called with 
the name: sqlite3.so, it worked.

Changing the core db into a MySQL via ODBC caused some problems even after it 
seemed to work. For instance, console help caused an error with an error 
description indicating that a SQL SELECT query including the reserved word key 
has been fired.

It this problem likely to be solved if I used another version of the MySQL?

Jon Brüel
Consiglia Telecommunications
DK-2960 Rungsted Kyst
Tel: +45 45 16 1000
Mob: +45 26 15 30 60
CVR: 27047882



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--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_miness...@hotmail.com<mailto:msn%3aanthony_miness...@hotmail.com>
GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com<mailto:paypal%3aanthony.miness...@gmail.com>
IRC: irc.freenode.net<http://irc.freenode.net> #freeswitch

FreeSWITCH Developer Conference
sip:8...@conference.freeswitch.org<mailto:sip%3a...@conference.freeswitch.org>
iax:gu...@conference.freeswitch.org/888<http://iax:gu...@conference.freeswitch.org/888>
googletalk:conf+...@conference.freeswitch.org<mailto:googletalk%3aconf%2b...@conference.freeswitch.org>
pstn:213-799-1400
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Re: [Freeswitch-users] Lua and database access to core_db

[Jon Bruel]

I got the combination Lua with direct access to the core Sqlite database to 
work. Hurray, maybe I'm not as stupid as A.M II hints...
The problem was that Lua did not "like":

require "luasql.sqlite"
env = luasql.sqlite()
con = assert(env:connect("/usr/local/freeswitch/db/core.db"))

After changing it to

require "luasql.sqlite3"
env = luasql.sqlite3()
con = assert(env:connect("/usr/local/freeswitch/db/core.db"))

And seeing that there was a symlink in one of the right directories called with 
the name: sqlite3.so, it worked.

Changing the core db into a MySQL via ODBC caused some problems even after it 
seemed to work. For instance, console help caused an error with an error 
description indicating that a SQL SELECT query including the reserved word key 
has been fired.

It this problem likely to be solved if I used another version of the MySQL?

Jon Brüel
Consiglia Telecommunications
DK-2960 Rungsted Kyst
Tel: +45 45 16 1000
Mob: +45 26 15 30 60
CVR: 27047882


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Re: [Freeswitch-users] Lua and database access to core_db

[Jon Bruel]

The MySQL version is 5.1.37. Well I'm not an expert on every field, and I have 
no skills in the C, include libraries, and the art of compiling. For this I 
have to follow the guidelines. But it wouldn't harm the FS project if it 
generally became more accessible to the race of non-specialists, which I hereby 
represent.

Jon Brüel


From: freeswitch-users-boun...@lists.freeswitch.org 
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Anthony 
Minessale
Sent: 6. december 2009 20:42
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Lua and database access to core_db


Most of this is unfortunatly because you do not have the proper skill to set it 
up because, with the proper skills, all of the ways you tried would have ended 
sucessfully.  I say that beacause I have had many users use each of the 
different methods in your list of failures only they were sucessful.

What you are asking for is possible but would require many hours of coding just 
to help solve your problem.
You would have to wait a really long time until someone had the time to do it 
for free or post a bounty for it.  Probably about 1k in consulting time.  It 
may be cheaper for you to pay a consultant to set up one of the ways known to 
work.  These are your options as I see it.
On Dec 6, 2009 12:20 PM, "Lon Baker" 
mailto:l...@kickasspixels.com>> wrote:

Jon,

What version of MySQL are you using?


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Re: [Freeswitch-users] Lua and database access to core_db

[Jon Bruel]

Dear all

Some feedback regarding using Lua to access core database:

First of all, I did not succeed to get SQLite drivers in Lua or ODBC-drivers in 
Lua to work. The SQLite driver did compile OK, but there was an error when 
loading into Lua. The ODBC driver did also compile OK, did load into Lua, but 
could not connect. Accessing the SQLite from the Linux console using "isql -v 
" worked OK. The problems may be related to the present Linux 
distribution, which is Ubuntu 9.1 server. Unfortunately the public searchable 
information about Lua ODBC driver problems is sparse.

So I continued to try to get the FS to use MySQL as the core db. A number of 
problem occurred, which I did not find solution for in the FS documents. The 
problems and solutions are described below:

1)   The core database is not automatically created by FS, therefore I 
created it manually.
2)   During startup, the FS test for transaction support, and this test 
failed. To achieved transaction support with MySQL and MyODBC, three things had 
to be changed:
a.   A line was added in my.cnf to force innoDB as the default table: under 
the [mysqld] header, the following line was added: set-variable = 
default-table-type=InnoDB.
b.  The a line under the DNS was added to allow for multiple line statement 
support: option = 67108864. (ODBC version is 3.51).
3)   After these changes the transaction worked, but all the tables in the 
core db were not created, therefore I copied the structure from the SQLite 
tables into tables with the same names in the MySQL database. This exercise 
also showed what the problem was: MySQL could not create tables with many 
VARCHAR type files with a size of 4096 (sound very big?). The size was reduced 
to 255, and most of the tables were created OK. One table still gave problems: 
the interface table. One of the fields is called key, which is a reserved word 
in MySQL, and by backticking the word key in the create statement, it worked.
4)   Finally the FS started up using the MySQL, but errors splashed over 
the screen just after startup. There was a problem creating new records in the 
interface table, the problem was the key field. Changing the insert statement 
in switch.core.sqldb.c file by backticking the key field name and recompiling 
the FS solved that problem.

I guess this will be fixed in later releases and I hope this will assist the 
brave programmers!

I would like to argue for the development of SQLite connectivity in Lua. The 
ODBC core solution is not as clean as a direct database connection, and as long 
as this is limited to SQLite, a direct connection from "recommended script 
language" would be the cleanest solution. Further, it would be nice if 
everything works after having compiled the FS package.

Jon Brüel
Consiglia Telecommunications
DK-2960 Rungsted Kyst
Tel: +45 45 16 1000
Mob: +45 26 15 30 60
CVR: 27047882


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Re: [Freeswitch-users] Lua and database access to core_db

[Jon Bruel]

I have now tested the FS with core db configured using MySql (by modifying the 
switch.conf.xml file). Unfortunately, it does not solve my problem because some 
of the core tables still remain as active SQLite tables.

After restarting the FS in the new configuration (with SQLite database core 
deleted), the following tables are created in MySql and SQLite:

MySQL: aliases, complete, nat and tasks (database starting with no tables prior 
to FS restart).
SQLite: aliases, calls, channels, interfaces, nat and tasks.

As I would like to access the channels table using Lua, the change did not fix 
my problem. I have positive verified that the channels table is active and 
populated during calls.

Are there other places where I should define the usage of the MySql database?


Jon Brüel
Consiglia Telecommunications
DK-2960 Rungsted Kyst
Tel: +45 45 16 1000
Mob: +45 26 15 30 60
CVR: 27047882


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Re: [Freeswitch-users] Lua and database access to core_db

[Jon Bruel]

Anthony, you advised me to use MySQL as the core database in order to access it 
from Lua. I'm testing that as a work-around.

Still, I guess that your choice of SQLite as the default core database have 
been taken from efficiency or stability considerations. Using MySQL through an 
ODBC-connector does not sound as a clean solution. Have you any experience 
about "how bad" it is to use the ODBC MySQL combination in terms of stability, 
memory leaks and efficiency?

Regards

Jon Brüel
Consiglia Telecommunications
DK-2960 Rungsted Kyst
Tel: +45 45 16 1000
Mob: +45 26 15 30 60
CVR: 27047882


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[Freeswitch-users] Lua and database access to core_db

[Jon Bruel]

I am trying to rewrite all my javascript scripts into Lua scripts. I have run 
into the problem of core_db access. This can be achieved with Spidermonkey, but 
apparently not with Lua. I have tried to get the binary for Lua (using apt-get) 
but I get an error when I require the sqlite.so: undefined symbol: 
luaopen_luasql_sqlite, so I'm stuck. So what is a feasible way to manipulate 
the core database from Lua?
I may mention that access to MySQL works perfectly from Lua.
Regards Jon

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[Freeswitch-users] SBC functions in FS

[Jon Bruel]

I'm looking into the first design phase of a setup like this:
PSTN <-> FS <-> Various SIP PBXs or nothing <-> SBC <-> Internet <-> Customer
Where I'm considering using the FS as the SBC and as an alternative to OpenSIPS 
(OpenSER), where the FS has the advantage of a design having only one type of 
switch, e.g. less education to the team supporting the system.
If this should work, the FS should be able to route SIP messages such as 
NOTIFY, SUBSCRIBE etc. Is that possible, and what are the measures to implement 
such a routing function? /Jon


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[Freeswitch-users] How to control to domain used in INVITE From header

[Jon Bruel]

The situation is as follows: An incoming call is processed by the FS and sent 
out to a sip client. I want to control the From header in this outgoing INVITE.
I have tried to set various channel variables, including sip_h_From, in order 
to control to domain used in the INVITE From header, which for instance looks 
like this:
From: "JBS (Soft)" 
Instead of the server IP address, X.X.X.X, I want to set the SIP-domain used 
for the specific customer in a multi tenant setup. This from header is used by 
the phone telephone number list register (at least for the Snom phones), so 
controlling it is important. How is it done? /Jon
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[Freeswitch-users] fifo.conf.xml usage

[Jon Bruel]

I'm happy to see that you can add consumers to queues using the fifo.conf.xml 
configuration file. I have made some tests and I hope it may lead to a more 
universal way of setting up queues for small organisations than the one I have 
described in the wiki, and which includes (too) many javascripts. I have some 
questions to clarify my understanding. Using the fifo.conf.xml, I find:
 - That the consumers continue to ring after the caller has abandoned the 
queue. Is there a way to avoid this?
Further:
 - Is there a way to control the caller_id_name/number presented to the 
consumer?
 - Is there a way to control the ringing tone in the consumers like the one 
which can be used in the dialplan?
 - Can the fifo.conf.xml refer to an ODBC connection in order to get the 
members from a database?
Finally, thanks for all the good work everybody in the FS community has put 
into FS, I truly believe in the possibilities of this product. Checking the 
hits on Google certainly indicates you moving into the right direction. /Jon

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Re: [Freeswitch-users] How to force FS using the hosts entry in outgoing gateway calls

[Jon Bruel]

Thanks Anthony. Using the parameters: 




Returns error 900, and a 'ngrep port port-number' indicates that its doesn't 
try to register at all. I have now let the server look at a local DNS where I 
have added a "wrong" A-record. That solves the issue, but your solution would 
be cleaner. The version is: trunk 10220. /Jon


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[Freeswitch-users] How to force FS using the hosts entry in outgoing gateway calls

[Jon Bruel]

For the configuration of a gateway I need to use a specific proxy domain name 
before the server (Covergence SBC with a BroadWorks Application Server behind) 
accepts calls. The twist is that the right proxy name points the wrong 
IP-address (the voicemail server for the account). I have tried to overrule 
this by adding a host entry (Linux). When I ping to the domain name I get the 
right address (the one from the host table), but the FS uses the address from 
the DNS lookup, not the address from the host table. What can I do to force the 
FS using the entry from the host table? Thanks /Jon.

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[Freeswitch-users] Connection to a QuesCom gateway without password

[Jon Bruel]

I have tried connecting to a SIP GSM gateway made by the French company, 
QuesCom. It does not handle the authentication process properly, and the 
connection cannot be setup because of this. I have tried to omit the password 
parameter but apparently FS does not accept a setup without password. Is there 
any way to circumvent this? /Jon

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[Freeswitch-users] fifo in on connect

[Jon Bruel]

Is there a way to activate an action when a call has entered the "fifo in"? I'm 
thinking of something similar as setting the channel variable execute_on_aswer. 
/Jon

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Re: [Freeswitch-users] Putting javascript to sleep

[Jon Bruel]

Thanks Anthony, the msleep function is really nice, but I have some problems 
getting it to work. Somehow it fucks up the system, which becomes unresponsive 
after I have run it in a javascript. Should I just wait for the next release, 
or do you want a more detailed error report through the appropriate channels? 
/Jon
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[Freeswitch-users] Putting javascript to sleep

[Jon Bruel]

I have worked on making a set of scripts and dialplans for a traditional queue, 
which rings the agents. I have published suggestions and ideas on the WIKI, but 
I'm still not satisfied with the solution. One of the challenges has been to 
emulate a sleep command as known in php and many other languages.  I have 
solved it by creating a session for the sake of this, and run 
session.execute("sleep", 5000), but this solution pivots on the creation of the 
session, and it not very clean in my opinion. So my question is: are there any 
elegant ways of putting javascript to sleep in FS? /Jon
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[Freeswitch-users] Error database is locked

[Jon Bruel]

I have made an application using the fifo application which behaves as
the Asterisk queue: It rings to the members. I use javascript DB to
access the core database in various scripts, f.in the following script:
//This script is used to disconnect agents being rung from a queue when
abandoned by the customer before answered.
use("CoreDB");
var db = new CoreDB("core");
var sql="select uuid from channels where cid_num='"+argv+"' and
state='CS_EXECUTE';";
db.prepare(sql);
while(db.next()) {
rec = db.fetch();
apiExecute("uuid_kill", rec["uuid"]);
}
db.close();

My problem is that I get a message from the CLI: mod_spidermonkey.c:3300
js_api_use() Loading CoreDB, [ERR] mod_spidermonkey_core_db.c:250
db_prepare() Error database is locked.
The error does not occur every time.
I have closed the database in every script using it.
Is the anything I should be aware of to prevent this from happening?

BEGIN:VCARD
VERSION:2.1
N:Bruel;Jon
FN:Jon Bruel
ORG:Consiglia;Udviklingsafdelingen
TITLE:Direktør
TEL;WORK;VOICE:201
TEL;HOME;VOICE:45161001
TEL;CELL;VOICE:26153060
ADR;WORK:;B201
LABEL;WORK:B201
EMAIL;PREF;INTERNET:[EMAIL PROTECTED]
REV:20080627T092557Z
END:VCARD
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Re: [Freeswitch-users] Originate command

[Jon Bruel]

Brian, I assume that the "Originate command" thread is still resting in
your courtyard for a resolution? /Jon



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Re: [Freeswitch-users] Originate command

[Jon Bruel]

Brian, I hope this is what you expect: Testing on CLI, this one works:
Originate sofia/gateway/1500/201,sofia/gateway/1500/205
206 XML internalpreparation
And this one does not:
Originate user/[EMAIL PROTECTED],user/[EMAIL PROTECTED]
1206 XML internalpreparation
Or rather, it gives the same result as this:
Originate user/[EMAIL PROTECTED],user/[EMAIL PROTECTED]
1206 XML internalpreparation
(Last user made identical to first user.) I should remark that the
gateway above loops back to the dialplan with the context
internalpreparation. /Jon



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Re: [Freeswitch-users] Originate command

[Jon Bruel]

Brian, I made another test:
Originate sofia/gateway/1500/201,sofia/gateway/1500/205 206 XML 
internalpreparation
This works as it should. So the issue may be related to the user "channel". /Jon



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Re: [Freeswitch-users] Originate command

[Jon Bruel]

Brian, I have now pastebinned the SIP trace as taken from a Snom phone as a 
response to the command:
Originate user/[EMAIL PROTECTED],user/[EMAIL PROTECTED] 206 XML 
internalpreparation
I also checked it from the server, and verified that no INVITEs are sent to the 
address belonging to 1205. Further, from the console the line:
[DEBUG] switch_ivr_originate.c:700 switch_ivr_originate() variable string 0 = 
[EMAIL PROTECTED],
is repeated twice with some 10-20 debug lines in between. To get some way 
further, I could try to repeat the tests you have done in your lab showing that 
there is no problem!? /Jon



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Re: [Freeswitch-users] Originate command

[Jon Bruel]

Brian, there is a reason for the same address: it's connection to an
Asterisk server. My original question remains: Does the originate
command work as it should ("originate
user/[EMAIL PROTECTED],user/[EMAIL PROTECTED] 206 XML
internalpreparation")? The problem was that the FS only made a call to
one phone, 1205, not to both. Maybe I didn't make it clear
before that when the call was answered by 1205, the phone with
extension 206 rung as expected.  /Jon

 

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Re: [Freeswitch-users] Originate command

[Jon Bruel]

Brian, how wonderful, this is illustration the art of communication. Or
rather, how difficult it is. I was thinking of some kind of directory in
the structure of pastebin, not the FS, when you mentioned the
"directory".
So, regarding the FS user directory, I'm using xml_curl, and I have
pasted a part of it (it has 300 entries). I have also tried to comment
out the xml_curl module and used a static directory, in order to
eliminate the possibility of an undesired interaction with xml_curl, but
the result was the same.
Anyhow, the three bits are now included and sent using pastebin.
Goodnight! /Jon



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Re: [Freeswitch-users] Originate command

[Jon Bruel]

Brian, thanks for the useful info! I have pasted it under my login name,
jbr. I don't know if that you mean by user directory. /Jon



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Re: [Freeswitch-users] Originate command

[Jon Bruel]

Brian, this is an example of what I do: From the CLI, I enter:
Originate user/[EMAIL PROTECTED],user/[EMAIL PROTECTED] 206
XML internalpreparation. 
I have a different phone on each endpoint referred to above, but I can
see that when I enter the command, two calls go to
user/[EMAIL PROTECTED] When I answer, the phone with extension 206
is rung.
I'll send you further information on pastebin.freeswitch.org, but I need
some instructions on how to be authorized accessing it. By the way, a
similar issue has been reported by B Karthik, August 18th.
/Jon



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[Freeswitch-users] Originate command

[Jon Bruel]

I have tried to use the originate command from the API, from javascript
apiExecute and from javascript using newsession.originate. In all cases
I get the same behaviour: when the dialstring includes several
destinations such as:
user/[EMAIL PROTECTED],user/[EMAIL PROTECTED]
it's only the first user dialled. Actually this user is dialled twice.
If I add a third number, the first user is dialled three times. Is this
as per design, or should I report a bug. And if so, which project should
I use? /Jon



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Re: [Freeswitch-users] Load test - performance not even matching Asterisk

[Jon Bruel]

I have been asked by Mark Crane and others to contribute more positively
by joining the community, and I would be glad to do so. As with anyone I
have my own agenda being to create a virtual PBX suite - using FS or
what ever I find most suitable. At this stage, I have added a load
testing page to the wiki, a modest first step to my contribution.  
A few words about my findings regarding load tests: On the surface the
FS compares with the Asterisk: +/- 20% in terms of CPU load. The test
may not credit the FS for possible better resilience. Knowing Asterisk I
also know its instability under load, and with the architecture of the
FS, this may be resolved by design, and I'm looking forward to verifying
that. So I'll - in between the other tasks - continue with load tests
which attempt to simulate real life situations, with complex dial plans
and with erroneous internets. Ideas are welcome!
/Jon


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[Freeswitch-users] Error loading mod_event_multicast

[Jon Bruel]

I have compiled FS with mod_event_multicast uncommented in the
source/modules.conf file, but I get the following error when I start FS:
[ERR] mod_event_multicast.c:220 mod_event_multicast_load() Multicast
Error
[CRIT] switch_loadable_module.c:767 switch_loadable_module_load_file()
Error Loading module /usr/local/freeswitch/mod/mod_event_multicast.so
**Module load routine returned an error**
I run Linux Fedora 9, 64 bit. Any ideas how to resolve it? /Jon


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Re: [Freeswitch-users] Load test - performance not even matching Asterisk

[Jon Bruel]

Resolved: I have made further tests, and my final conclusion is that the
previous stated test results were screwed by the application 'gentones'.
This application does in some cases send more rtp than expected. If I
used:



the expected rtp of 8600 kB/s was transmitted. If I used


.
the rtp was 34600 kB/s, and the memory is heavily consumed. The only
difference being the sequence of the gentones commands. I don't know if
this is the expected behaviour of 'gentones' or not, but it certainly
screwed up the results previously posted. /Jon


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Re: [Freeswitch-users] Load test - performance not even matching Asterisk

[Jon Bruel]

Anthony, your last response is surprising me. I appreciate the work you
have done in supporting people doing load testing. But the information
about this is rather unorganized and difficult to access for newbies.
Further, many threads on this and other topics are written in telegram
style language, which increases the learning time. As an example, the
first time I heard about the testing tool sipp was yesterday - through
the users list - thanks. What a great tool! The documentation though, is
not aimed at making the learning curve easy. A "these are the 20 steps
to do test" manual does not exist to my knowledge. Bear in mind that I'm
not a programmer but a generalist with all-round knowledge about
telecoms and virtual PBXs.
I hope you can use this feedback in a productive way in order to improve
the overall level of documentation. And it also pinpoints the need for
the userlist and repetitive questions.
Setting up commercial relations also makes sense when we have reached
the proof of concept and a business plan can be made. The proof of
concept includes a decision about which switch to use, Asterisk is still
an alternative to FS. Asterisk may have some architectural drawbacks,
but it has been tested for a longer time and in more setups than FS.
Further, Asterisk has a bad reputation. 
As you have mentioned before, it may be worth while visiting your team
soon, and I'm preparing for that.  /Jon


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Re: [Freeswitch-users] Load test - performance not even matching Asterisk

[Jon Bruel]

Hi all
An update on the performance measurements:
The measurements I have referred to earlier all involved an Asterisk as
the call generator. Somehow this setup leads to extensive rtp bandwidth
usage. Each channel used around 500 kbps. If a phone is entered into the
loop, this is reduced to the expected 64 kbps. I have not found any
reason for this, but it certainly fouls up the test, and I have changed
the test setup.
Further, and since the earlier tests, the network has been updated to a
Gbits network.
I have now made two new test: 
1) Using WinSIP from Touchstone as a call generator.
2) Using the Asterisk as one component, and setting up a chain of calls
which goes forth and back from the Asterisk and the FS. All call are
started from a real phone, and after 100 loops, where the calls are
answered and sent on by the dial plan, the calls are terminated by an
tone () in the
FS.
The two test show similar top-figures at similar loads.
The first test would be my preferable, but it is limited to 50 calls due
to the trial licence limitations. Using an external non-FS and
non-Asterisk device will eliminate some uncertainties, that's why it
would be preferred.
The other test has been done with 600, 400 and 200 channels (300, 200
and 100 calls), and the results of the top command are:
cpu sy  ni  id  wa  hi  si  total
* 600   10  30  0   33  0   2   25  100
FS600   22  33  0   30  0   0   15  100
0
* 400   7   18  0   67  0   1   7   100
FS400   14  17  0   62  0   0   7   100
0
* 200   3   10  0   84  1   0   2   100
FS200   7   8   0   82  1   0   2   100
The results do not show significant differences between the capacity
behaviour of the Asterisk (*) and the FS. The also show an expected
interrupt load (si) proportional to the square of the call load.
Still the FS does not really outperform the Asterisk - which I find
disappointing. Any comments are welcome.


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Re: [Freeswitch-users] Load test - performance not even matching Asterisk

[Jon Bruel]

I have made some further studies of the performance, and after I have
removed and old PC running 10 Mb/s on the Ethernet, the performance has
been drastically improved. So I tentatively think that there has been a
bottleneck somehow in the combination of Ethernet switch and slow
Ethernet cards.
I'll test further and report back. At this stage, Anthony, I need more
testing before you look at my system. If any others have had similar
experiences, please report back. /Jon


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Re: [Freeswitch-users] Load test - performance not even matching Asterisk

[Jon Bruel]

OK, you can access it directly now, and yes it would be fine if we can
find out what's wrong once for all. Phone is: +45 45 16 1001. I'm on
CET.
Regarding the calling application: It is Asterisk, setting up one leg to
MOH and the other to FS, so audio is certainly going through. 
Regarding the responsiveness of FS when the load is high: it does
respond, but the is a delay of around 1 second.
The load measurement is done by "top". /Jon


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Re: [Freeswitch-users] Load test - performance not even matching Asterisk

[Jon Bruel]

Thanks for the input.

 

I have now upgraded the OS to i86_64 (FC9), and I configured without
libcurl, as recommended by Anthony. The FS loaded without problems apart
from some "database is locked" errors, which may be harmless? I ran the
ulimit as recommended to unlimit all resources.  I started the FS after
the ulimit. The same performance test showed a slightly better result.
Without trying to make this too scientific, the improvement by going
from 32 bit OS to 64 bit OS has been 10-20%, at the most. Still the
performance is around 30-40% of the performance of Asterisk. 

 

With 100 a-leg (no b-leg, as all calls are answered by the echo
application) permanent channels the CPU idle is 10%, with 60%
system-load and 30% CPU-load. So something is rotten in Denmark... /Jon

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Re: [Freeswitch-users] Load test - performance not even matching Asterisk

[Jon Bruel]

The load on the CPU was after the calls were set up, this indicated
that's not a disk access issue.

 

I'm running FC 8, i686, which I understand is 32 bit. Changing to FC 8
or 9 x86_64 is possible on the Intel(r) Core(tm)2 Duo Processor, and if
it's worth it; I'll do it. Does anyone have an idea of what the
performance boost will be? /Jon

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Re: [Freeswitch-users] Load test - performance not even matching Asterisk

[Jon Bruel]

Regarding the 32bit vs. 64bit questions, I'm a bit uncertain about how I
answer it, but here is some imformation, which you may use:

cpu family  : 6
model   : 15
model name  : Intel(R) Pentium(R) Dual  CPU  E2140  @ 1.60GHz
flags   : fpu vme de pse tsc msr pae mce cx8 apic mtrr pge mca
cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe nx lm
constant_tsc arch_perfmon pebs bts pni monitor ds_cpl est tm2 ssse3 cx16
xtpr lahf_lm

uname -ar gives: Linux econel2.everdance.com 2.6.25.9-40.fc8 #1 SMP Fri
Jun 27 16:25:53 EDT 2008 i686 i686 i386 GNU/Linux.

With respect to compiling the FS, I have not been aware of any options,
so I just did the standard: bootstrap.sh, configure, make and make
install. 

I have tried to configure with some 64bit flags: CFLAGS="-m64 -fast"
CXXFLAGS="-m64 -fast" LDFLAGS="-m64" ./configure
--prefix=/opt/freeswitch --enable-core-odbc-support
--enable-core-libedit-support --enable-64 --with-openssl=/usr/sfw, but I
get an error:

 

checking for C compiler default output file name...

configure: error: C compiler cannot create executables

 

so I may need to install some 64bit libraries? If so some hints would be
great! /Jon

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Re: [Freeswitch-users] Load test - performance not even matching Asterisk

[Jon Bruel]

The protocol is SIP. The profiles relevant for the incoming call test
with echo are here:

Dialplan:
*





  



  





  



  



 

Sip Profiles:
*

 



  

  



  

 

  



  

 

  



  

 

  


















(comment: using PCMA)

































  



 

Gateway (one of them):
**



  

  









 









  



 

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[Freeswitch-users] Load test - performance not even matching Asterisk

[Jon Bruel]

I have made some load test, where an Asterisk server was controlled to
make a high number of calls to a FreeSWITCH, which was registered on
Asterisk. Each had its own server. The calls made to the FreeSWITCH were
answered and echoed. The CPU load on the FreeSWITCH server approached
100 % at around 100 calls. Similar test with an Asterisk server would
show 2-3 times less CPU load. Looping back to the Asterisk, thus
avoiding a possible CPU-loading application (echo), did not show a
significantly different result.

The test has been done on a CentOS 4 and a FC8 with similar result. 

Any hints about how to increase the capacity to the level of several
times the level of Asterisk - as reported by others? Are we possibly
dealing with some hardware issues related to the NIC and its drivers.
/Jon

 

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[Freeswitch-users] Sendevent syntax

[Jon Bruel]

I have tried to get an understanding of the possibilities of sendevent,
and with the help from Brian West I had successes with a few, as shown
in the examples below: 
Sendevent  NOTIFY
Event-string: talk
content-length: 2
content-type: application/simple-message-summary
Profile: internal
User: 11000
Host: 10.3.1.11

Sendevent  SEND_MESSAGE
Content-length: 2
content-type: application/simple-message-summary
Profile: internal
User: 11000
Host: 10.3.1.11

Sendevent  MESSAGE_WAITING
MWI-Message-Account: [EMAIL PROTECTED]
MWI-Messages-Waiting: 3
MWI-Voice-Message: 2/1 (1/1)

I lack a general understanding of the syntax. I have tried to hangup a
session using CHANNEL_HANGUP where I made a qualified guess about the
headers, but it did not work (instead I used "api uuid_kill ").
Could I have an overview of the syntax or a hint about which code I
could look at in order to get a more full understanding? /Jon




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Re: [Freeswitch-users] Possible problem in adding channel variables to the bridge

[Jon Bruel]

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Re: [Freeswitch-users] Possible problem in adding channel variables to the bridge

[Jon Bruel]

Anthony, this is a simplified version dialplan, just for an illustration
of the previous posts: 


file default.xml:


  

  
 
 
 
 
  
   

   
  
  
  
   
  


file info.xml:


  
  



  
  
  

  
  


Sorry about my inconsistent use of the terms, the XML about should
explain. /Jon

 

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Re: [Freeswitch-users] Possible problem in adding channel variables to the bridge

[Jon Bruel]

Anthony, I think I understand this part of it, but I'll try to stay
humble. 

This is what I want: I want to have several destinations in my
dialstring, and I want to pass different variables on to the script
which runs after answer.

I also wanted to control the A-number caller ID, and that works as it
should, by placing [originate_caller_id_number=] in front of the
destination. So let's forget about that.

I have tried:



In order to pass the variable NEWVARIABLE on to the ONANSWER script.

I have let the ONANSWER XML script make an "execute_extension call" to
another XML script. For the sake of clarity, lets call it the INFO
script. In the INFO script I check if variable in the dial string has
been given a value. The ONANSWER script is fired off two times: when the
a-leg answers and when the b-leg answers. Apparently that's the way
"execute_on_answer" works, when its set using the export facility.

When it is the a-leg which fires off the ONANSWER script, the
NEWVARIABLE exists in the following INFO script fired off by the
b-leg-answer event. But if the b-leg which fires off the ONANSWER script
first, the variable has no value in INFO script. In some tests, I get a
populated variable in 3 out of 10 calls.

Hope this clarified the issue, if not, I may call you on the phone. /Jon

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Re: [Freeswitch-users] Possible problem in adding channel variables to the bridge

[Jon Bruel]

I have made some further testing, and even without an
"execute_on_answer" in the ONANSWER-code, the variable set in the
dialstring appears now and then. And it only happens, when the ONANSWER
code for the a-leg is executed first. On my system, this happens in
30-50% of the tests. If I move the ONANSWER code to another file, where
this is the only condition, it does not work, but entering some dummy
conditions before, makes it work now and then. So this may be a
concurrency issue?

Jon

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Re: [Freeswitch-users] Possible problem in adding channel variables to the bridge

[Jon Bruel]

Thanks Anthony. I have been confused because of the semantics:
execute_on_anwer implies that the code pointed at is re-executed.
Anyhow, including the application execute_extension in the
ONANSWER-code, helps but not every time: In 20-30% of all the instances,
the variable is passed on to the b-leg part of the ONANSWER. In other
cases the variable is still undefined.

When the b-leg of the ONANSWER runs before the a-leg part, it is
undefined, and vice versa. The code executed by execute_extension is:



  
  



  
  
  
  

  
  


Jon

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Re: [Freeswitch-users] Possible problem in adding channel variables to the bridge

[Jon Bruel]

For a test, I have used the following show sample dialstring:



  
  
  
  
  
  

  



  
  

  

  

  

  

The two extensions Call_Answered and Ringing are just test extensions to
check if the variable NEWVARIABLE has been passed. If I just make normal
call, it is not passed on. Using export for the execute_on_answer, the
code runs twice. I have tried other "execute_on_XXX" but I can't see the
new variable. Again this may be related to my lack of understanding the
fundamentals of FS. /Jon

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Re: [Freeswitch-users] Execute on answer only executes once?

[Jon Bruel]

I have solved the problem. The variable execute_on_answer must be set as
an inherited variable, e.g. by "export" instead of by "set". I hope my
many "beginners questions" will give someone food for thought for a
really pedagogical bundle of documentation. /Jon

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Re: [Freeswitch-users] Execute on answer only executes once?

[Jon Bruel]

Thank for the many answers. And Anthony, you are right about
understanding the paradigm. I need something written by a person, who is
new to it. Many articles require pre-knowledge about programming details
- and I'm not there yet.

Anyhow, I think this case is straight forward. Again to repeat: A (ext.
1000) calls B (ext. 1002) which answers. A (ext. 1000) makes an
unattended transfer to C (ext. 1001) which answers. The unattended
transfer goes through the same dialplan as before. After the transfer, B
is connected to C. And as I understand it: no dialplan has told B to go
anywhere after answer, so nothing happens. So maybe I should rephrase my
problem: What should I do in this case to obtain the desired result:
that a specific dialplan code is executed after B connects to C? /Jon

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Re: [Freeswitch-users] Possible problem in adding channel variables to the bridge

[Jon Bruel]

Well what I really should say is that adding many variables to the
bridge data, such as this:

 

data="[origination_caller_id_number=45161061,my_var1=blah,my_other_var=w

hatever]sofia/gateway/45161061/$1"/>

 

would be very nice. My problem is that I cannot verify that the
variables are passed by the info application in an execute_on_answer
code done when the call has been answered. The variable is not defined
at that stage? Is it lost? /Jon

 

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[Freeswitch-users] Execute on answer only executes once?

[Jon Bruel]

These setting are as listed (without that I could not make a "moved
temporary" from the B-phone). The unattended transfer works as it
should, the only thing is that the aexecute-on.answer script isn't
fired.

 

Check your profile.

 

 

 

 

Comment out those two lines.

 

/n

 

On Sep 23, 2008, at 7:49 AM, Jon Bruel wrote:

 

> I have set up the internal extensions to execute on answer. When I 

> from A call B, it works OK. If I make an unattended transfer from B to


> C, thus setting up a connection between A and C, it does not execute 

> when C answers. It this intentional, and is there a way around it to 

> get it to execute every time?  /Jon

> 

> 

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Re: [Freeswitch-users] Possible problem in adding channel variables to the bridge

[Jon Bruel]

Resolved:

The problem has been solved after updating to the last trunk: 1.0.1,
9624.

The  works, but it would be nice to be able to add more variables than this
only one.

Jon 

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Re: [Freeswitch-users] Problems with configure script

[Jon Bruel]

Resolved.

Thanks, Michael, for the help. For the sake of the community the problem
likely was erroneous trunk download. /Jon.

 

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Re: [Freeswitch-users] Problems with configure script

[Jon Bruel]

I have tried several times to bootstrap. The result is the same every
time. Could it be some errors related to different copies of the
programs used - some under /usr/local/share, some under /usr/local? 

And from other users: have you experienced the same problem? /Jon

 

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[Freeswitch-users] Execute on answer only executes once?

[Jon Bruel]

I have set up the internal extensions to execute on answer. When I from
A call B, it works OK. If I make an unattended transfer from B to C,
thus setting up a connection between A and C, it does not execute when C
answers. It this intentional, and is there a way around it to get it to
execute every time?  /Jon

 

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Re: [Freeswitch-users] Problems with configure script

[Jon Bruel]

I do have c++ installed, and the release version 1.0.1 did install OK.
On the other hand 1.0.1 came with the configure script, in this version
configure was build by bootstrap.

I can't answer what the difference of the trunk version compared to
version 1.0.x is. Anyhow I have the same problem with the snapshot from
20-09-2008.

Somehow I think the AX_COMPILER_VENDOR is undefined when called, but I'm
not at all into the autoconf-world, and I waste time trying to figure
out what's going on.. /Jon

 

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[Freeswitch-users] Problems with configure script

[Jon Bruel]

I have got the trunk version of FreeSWITCH, have run bootstrap.sh, but I
get get stuck in configure, where I get the following message, which I
don't know how to act on:

./configure: line 13837: AX_COMPILER_VENDOR: command not found
configure: error: Please update configure.in with SOLINK values for your
compiler

Versions: Freeswitch: build 9609, automake: 1.10.1, autoconf: 2.61,
libtool: 1.514 or 1.5.24. Operating system is CentOS, uname -r =
2.6.9.-67.0.4.plus.c4 or 2.6.25.9-40.fc8 (same problem on two different
systems). 

What should I do?

Jon 

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Re: [Freeswitch-users] Possible problem in adding channel variables to the bridge

[Jon Bruel]

The FREESWITCH version is the 1.0.1 release version. I have updated to
the snapshot from the 20th of September. But I get problems with the
./configure. The bootstrap.sh works OK after I have updated autoconf to
version 2.63 and automake to version 1.10.1. The error from ./configure
after half a page of lines is:

 

./configure: line 4693: syntax error near unexpected token
`build_libtool_libs,'

./configure: line 4693: `_LT_DECL(build_libtool_libs, enable_shared,
0,

 

Operating system is CentOS, uname -r = 2.6.9.-67.0.4.plus.c4.

Jon 

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Re: [Freeswitch-users] Possible problem in adding channel variables to the bridge

[Jon Bruel]

Both suggestions work: If I include the effective_caller_id_number in
the dialplan before bridging, or if I include
{origination_caller_id_number=45161061} or
[origination_caller_id_number=45161061] as a part of the first parameter
after a originate command. 

But skipping the "set" command in the dialplan, and trying to use:

 



 

or:

 



 

does not work. Same thing when using curled brackets.

I need to be able to pass parameters individually to each of the bridge
destinations, not just for the A-number, but also for other stuff to be
used for post processing of the call. An alternative would be to emulate
the Asterisk chan_local by making some kind of loopback to the dialplan,
using IAX or SIP, but this is a inelegant way of solving it. 

Jon 

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Re: [Freeswitch-users] Possible problem in adding channel variables to the bridge

[Jon Bruel]

Yes I can, but somehow I start to repeat myself. An INVITE is sent
allright every time, but one of the headers differs:
Works:
Remote-Party-ID:
;screen=yes;privacy=off
Doesn't work: 
Remote-Party-ID: "Extension 1000"
;screen=yes;privacy=off
I have tried with different variable names in the bridge data: 

as well as: origination_caller_id_number
data="[origination_caller_id_number=45161061]sofia/gateway/45161061/$1"/
>
the latter combined with: .
But the only thing which works is setting the effective_caller_id_number
in the dialplan before the bridge. That would be fine, unless I want to
bridge to several destinations. Every time the call is rejected, the
Remote-Party-ID-header hasn't the right value.
I have tried to execute another dialplan at bridge, testing with other
variables, and the info application doesn't reveal the variable I try to
include in square brackets in the bridge data string. So maybe its not
passed over to the B-leg.
If you need more data, let me know, and I'll package it during the
weekend.
Jon 


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Re: [Freeswitch-users] Possible problem in adding channel variables to the bridge

[Jon Bruel]

No the far side is BroadWorks, and as I said in the first mail, the only
difference between an accepted and a rejected INVITE is in the contents
of the Remote-Party-ID header. Generally in many European countries, the
operators limit the A-number to the actual number of the line dialled
out on.




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Re: [Freeswitch-users] Possible problem in adding channel variables to the bridge

[Jon Bruel]

None of the suggestions worked, and I still can't control the A-number
individually when bridging to multiple destinations. 
I have tried to change the dialplan using
 [origination_caller_id_number=1234] 
As a part of the string in the bridge data, but an info after bridge did
not show the variable. I also tried with an unknown variable such as
new_var=1234, but it did not show up after the bridge using the info
application. Do we have a bug, or has the format been changed..




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Re: [Freeswitch-users] Possible problem in adding channel variables to the bridge

[Jon Bruel]

Mike, you answered: effective_caller_id_number is meant to be set on the
a leg, not the b-leg.
Well I need to control the "effective_caller_id_number" (or whatever it
is called in the B-leg) individually for each location when I bridge to
many destinations. In my case, I need to have different identities
(A-numbers) for local calls and for external calls. I have followed the
WIKI:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridgecall#Calling_
multiple_destinations, though. Anyhow how is this solved? Could it be
possible to loopback to the dialplan as in Asterisk local channel or to
go through some steps before dial attemps are made, but after the bridge
application?
Jon



--

Message: 4
Date: Fri, 19 Sep 2008 09:22:51 -0400
From: Michael Jerris <[EMAIL PROTECTED]>
Subject: Re: [Freeswitch-users] dialpaln
To: freeswitch-users@lists.freeswitch.org
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset="us-ascii"


On Sep 19, 2008, at 9:18 AM, Gopal krishnan wrote:

> Hi,
>
>   Since I am not able to make the outbound call, when I use this  
> command  in the console, I used to get the all the 31  
> channels , for a refrence I am posing one channel block here,
>
> span_id: 1
> chan_id: 31
> physical_span_id: 1
> physical_chan_id: 31
> type: B
> state: DOWN
> last_state: DOWN
> cid_date:
> cid_name:
> cid_num:
> ani:
> aniII:
> dnis:
> rdnis:
> cause: NONE
>
> In this I am seeing that the state seems to be DOWN, but when I try  
> wantouer status, it show as connected like ,
>


in openzap, state down is like on-hook... we should change that name  
maybe.


> Wanrouter Status:
>
> Device name | Protocol | Station | Status|
> wanpipe1| AFT HDLC | N/A | Connected |
> wanpipe2| AFT HDLC | N/A | Connecting|
>
> where I am using is Sangoma A102D card with hardware echo  
> cancellation.  And when I try to dial from the softphone in the  
> console I get this as,
>
> [EMAIL PROTECTED]> 2008-09-19 18:38:58 [NOTICE]  
> switch_channel.c:538 switch_channel_set_name() New Channel
sofia/internal/[EMAIL PROTECTED] 
>  [a75028c4-554a-473e-a8cc-8c52d7f72df4]
> 2008-09-19 18:38:58 [INFO] mod_dialplan_xml.c:232 dialplan_hunt()  
> Processing FreeSwitch->99841799874 in context default
> 2008-09-19 18:38:58 [NOTICE] switch_channel.c:538  
> switch_channel_set_name() New Channel OpenZAP/1:1/9841799874  
> [bd977e0d-e078-4193-a005-d751bdb26db8]
> 2008-09-19 18:39:28 [NOTICE] sofia.c:2705 sofia_handle_sip_i_state()  
> Hangup sofia/internal/[EMAIL PROTECTED] [CS_EXECUTE]  
> [ORIGINATOR_CANCEL]
> 2008-09-19 18:39:28 [NOTICE] switch_ivr_originate.c:1321  
> switch_ivr_originate() Hangup OpenZAP/1:1/9841799874  
> [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL]
> 2008-09-19 18:39:28 [INFO] mod_dptools.c:1814  
> audio_bridge_function() Originate Failed.  Cause: ORIGINATOR_CANCEL
> 2008-09-19 18:39:28 [NOTICE] switch_core_session.c:812  
> switch_core_session_thread() Session 1
(sofia/internal/[EMAIL PROTECTED] 
> ) Ended
> 2008-09-19 18:39:28 [NOTICE] switch_core_session.c:814  
> switch_core_session_thread() Close Channel
sofia/internal/[EMAIL PROTECTED] 
>  [CS_HANGUP]
> 2008-09-19 18:39:28 [NOTICE] switch_core_session.c:812  
> switch_core_session_thread() Session 2 (OpenZAP/1:1/9841799874) Ended
> 2008-09-19 18:39:28 [NOTICE] switch_core_session.c:814  
> switch_core_session_thread() Close Channel OpenZAP/1:1/9841799874  
> [CS_HANGUP]
>
> So where i am wrong, can you please correct me.
>


It looks like the hang-up is coming from the sip side of things, can  
you take a look at the sip trace and confirm there is really a cancel  
on sip.

Mike

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--

Message: 5
Date: Fri, 19 Sep 2008 18:53:27 +0530
From: "Gopal krishnan" <[EMAIL PROTECTED]>
Subject: Re: [Freeswitch-users] Asiterisk Dialplan for Freeswitch
To: freeswitch-users@lists.freeswitch.org
Message-ID:
<[EMAIL PROTECTED]>
Content-Type: text/plain; charset="iso-8859-1"

Hi,

 Basically I just want to test outbound alone with freeswitch, so I can
use
extensions.conf in the conf directory rite?
-- 
Thank you with regards,
Gopal,
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Message: 6
Date: Fri, 19 Sep 2008 19:04:43 +0530
From: "Gopal krishnan" <[EMAIL PROTECTED]>
Subject: Re: [Freeswitch-users] dialpaln
To: freeswitch-users@lists.freeswitch.org
Message-ID:
<[EMAIL PROTECTED]>
Content-Type: text/plain; charset="iso-8859-1"

Hi Mike,

   I changed the name in openzap.conf and also in default.xml, but the
same
thing persisting, this hangup I terminated from the softphone, its not
like
coming from the sip 

[Freeswitch-users] Possible problem in adding channel variables to the bridge destinations

[Jon Bruel]

I have tested the option of adding channel variables to the bridge
string, and it does not work.
This dialplan works:







While this one does not work, it is rejected by the gateway:





The INVITE Remote-Party-ID header differs. In the firs case it is:
"Extension 1000"
;screen=yes;privacy=off
And in the second case it is: 
"Extension 1000" ;screen=yes;privacy=off

Is this an error on my behalf, or do we have a bug?


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Re: [Freeswitch-users] How to get the content of SIP headers

[Jon Bruel]

As I see it, the NOTIFY methods mentioned are used when there is a call,
i.e. at the earliest when the phone has responded with a RINGING or a
similar response to the INVITE. By the way, the response to the INVITE
also includes a list of events (Header: Allow-Events) which the phone
responds to. I have not seen other instances where the NOTIFY To-header
needs to be tagged, but my experience is limited and not bound to the
RFCs, so don't rely on me!

On Sep 18, 2008, at 4:53 AM, Jon Bruel wrote:

> Thank you for your replies. In checking the SIP messaging sequence,  
> I saw that you are right about the source of the tag to the To- 
> header: it is the phone, which as a response to the INVITE, adds the  
> tag (and not, as I wrote, the FreeSWITCH). Still, we need a way to  
> read it, and not the least to send NOTIFYs with the right tag. I  
> have raised two issues on the Jira, maybe the category is wrong.
>
> One of the tricky things to sort out is how to handle it in the case  
> of bridging to a list of devices, and bridging to a user with  
> multiple registrations. The NOTIFY method should include measures to  
> send a NOTIFY to a specific phone among the many phones responding.
>
> Thanks again for your excellent responsiveness ? it adds the trust  
> of the FreeSWITCH.   Jon.
>
So we would only have the to tag in the case of a device with a call  
in progress, what about for just a registration, how would you  
determine between the devices?

Mike


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[Freeswitch-users] Accessing the variable shown by the application "info"?

[Jon Bruel]

I have tried to access the variables shown by the application info, and
the variable name to be used is generally not the same as the name
listed by info.
I have listed nearly 100 variables below, where I guessed the "real"
names, but a number are still missing, and I would like to know what the
"real" names are or how I derive the right names. 
Some of the problem-variables, though, for instance
Channel-Read-Codec-Name are also found further down in the list as
variable_read_codec, which can be read alright, and the presence of two
variable names which may have the same value seems confusing for a dummy
as me. 
The info-names have been taken - as an example - from the info result of
a call that has been diverted by the phone.
The list below shows my guess (via a javascript), and when the third
column shows FALSE, the guess need to be corrected. In some cases, the
reason for the "FALSE" may be related to the fact that the variable is
unassigned:
Info-name---Guessed name---Result
Channel-State---Channel_State---FALSE
Channel-State-Number---Channel_State_Number---FALSE
Channel-Name---Channel_Name---OK
Unique-ID---Unique_ID---FALSE
Call-Direction---Call_Direction---FALSE
Answer-State---Answer_State---FALSE
Channel-Read-Codec-Name---Channel_Read_Codec_Name---FALSE
Channel-Read-Codec-Rate---Channel_Read_Codec_Rate---FALSE
Channel-Write-Codec-Name---Channel_Write_Codec_Name---FALSE
Channel-Write-Codec-Rate---Channel_Write_Codec_Rate---FALSE
Caller-Username---Username---OK
Caller-Dialplan---Dialplan---OK
Caller-Caller-ID-Name---Caller_ID_Name---OK
Caller-Caller-ID-Number---Caller_ID_Number---OK
Caller-Network-Addr---Network_Addr---OK
Caller-Destination-Number---Destination_Number---OK
Caller-Unique-ID---Unique_ID---FALSE
Caller-Source---Source---OK
Caller-Context---Context---OK
Caller-RDNIS---RDNIS---OK
Caller-Channel-Name---Channel_Name---OK
Caller-Profile-Index---Profile_Index---OK
Caller-Channel-Created-Time---Channel_Created_Time---FALSE
Caller-Channel-Answered-Time---Channel_Answered_Time---FALSE
Caller-Channel-Hangup-Time---Channel_Hangup_Time---FALSE
Caller-Channel-Transfer-Time---Channel_Transfer_Time---FALSE
Caller-Screen-Bit---Screen_Bit---FALSE
Caller-Privacy-Hide-Name---Privacy_Hide_Name---FALSE
Caller-Privacy-Hide-Number---Privacy_Hide_Number---FALSE
variable_sip_received_ip---sip_received_ip---OK
variable_sip_received_port---sip_received_port---OK
variable_sip_authorized---sip_authorized---OK
variable_sip_mailbox---sip_mailbox---OK
variable_sip_auth_username---sip_auth_username---OK
variable_sip_auth_realm---sip_auth_realm---OK
variable_mailbox---mailbox---OK
variable_user_name---user_name---OK
variable_domain_name---domain_name---OK
variable_record_stereo---record_stereo---OK
variable_accountcode---accountcode---OK
variable_user_context---user_context---OK
variable_effective_caller_id_name---effective_caller_id_name---OK
variable_effective_caller_id_number---effective_caller_id_number---OK
variable_caller_domain---caller_domain---OK
variable_sip_from_user---sip_from_user---OK
variable_sip_from_uri---sip_from_uri---OK
variable_sip_from_host---sip_from_host---OK
variable_sip_from_user_stripped---sip_from_user_stripped---OK
variable_sip_from_tag---sip_from_tag---OK
variable_sofia_profile_name---sofia_profile_name---OK
variable_sofia_profile_domain_name---sofia_profile_domain_name---OK
variable_sip_req_params---sip_req_params---OK
variable_sip_req_user---sip_req_user---OK
variable_sip_req_uri---sip_req_uri---OK
variable_sip_req_host---sip_req_host---OK
variable_sip_to_params---sip_to_params---OK
variable_sip_to_user---sip_to_user---OK
variable_sip_to_uri---sip_to_uri---OK
variable_sip_to_host---sip_to_host---OK
variable_sip_contact_params---sip_contact_params---OK
variable_sip_contact_user---sip_contact_user---OK
variable_sip_contact_port---sip_contact_port---OK
variable_sip_contact_uri---sip_contact_uri---OK
variable_sip_contact_host---sip_contact_host---OK
variable_channel_name---channel_name---OK
variable_sip_call_id---sip_call_id---OK
variable_sip_user_agent---sip_user_agent---OK
variable_sip_via_host---sip_via_host---OK
variable_sip_via_port---sip_via_port---OK
variable_sip_via_rport---sip_via_rport---OK
variable_presence_id---presence_id---OK
variable_sip_h_P-Key-Flags---sip_h_P_Key_Flags---FALSE
variable_switch_r_sdp---switch_r_sdp---OK
variable_remote_media_ip---remote_media_ip---OK
variable_remote_media_port---remote_media_port---OK
variable_write_codec---write_codec---OK
variable_write_rate---write_rate---OK
variable_endpoint_disposition---endpoint_disposition---OK
variable_dialed_ext---dialed_ext---OK
variable_transfer_ringback---transfer_ringback---OK
variable_call_timeout---call_timeout---OK
variable_hangup_after_bridge---hangup_after_bridge---OK
variable_continue_on_fail---continue_on_fail---OK
variable_dialed_user---dialed_user---OK
variable_dialed_domain---dialed_domain---OK
variable_sip_redirect_contact_user_0---sip_redirect_contact_user_0---OK
variable_sip_redirect_contact_host_0---si

Re: [Freeswitch-users] How to get the content of SIP headers and sdp in dialplan?

[Jon Bruel]

Thank you for your replies. In checking the SIP messaging sequence, I
saw that you are right about the source of the tag to the To-header: it
is the phone, which as a response to the INVITE, adds the tag (and not,
as I wrote, the FreeSWITCH). Still, we need a way to read it, and not
the least to send NOTIFYs with the right tag. I have raised two issues
on the Jira, maybe the category is wrong.

One of the tricky things to sort out is how to handle it in the case of
bridging to a list of devices, and bridging to a user with multiple
registrations. The NOTIFY method should include measures to send a
NOTIFY to a specific phone among the many phones responding.

Thanks again for your excellent responsiveness - it adds the trust of
the FreeSWITCH.   Jon.

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Re: [Freeswitch-users] How to get the content of SIP headers and sdp in dialplan?

[Jon Bruel]

Mike, in my test setup, no tags are added to the To-header by the
FreeSWITCH. The reason why I want to control it? Well, many phones uses
the tag to relate an existing call to a NOTIFY message (send after the
INVITE has been sent) with the same headers (and the same to-tag) as the
INVITE. I'm not aware of how this functionality is described in the RFC,
though.

My aim is the check if I can use the FreeSWITCH to replicate a function
known in BroadWorks. In this system, there are a number of plug-ins and
web-interfaces which are used to control the SIP phone: Answer, put on
hold and resume. The plug-in controls the application server through a
XML interface, which in turn sends the NOTIFY to the phone. Thus the
phone can be controlled without connectivity between the PC and the
phone. These three states are controlled by a Notify message sent from
the application server. So it would be useful to find a way of sending a
Notify message to the phone from the FreeSWITCH. Other NOTIFY messages
could be used for other useful things such as resetting the phone.

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[Freeswitch-users] How to get the content of SIP headers and sdp in dialplan?

[Jon Bruel]

Well, let me be more precise: In general, I want to be able to modify
the SIP headers to my liking. I would like to add a tag to the To-header
sent to the B-phone. This tag information can be used at a later stage
to put the phone on/off hold from the switch by sending a NOTIFY with an
header: Event: hold or Event:talk.

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[Freeswitch-users] How to get the content of SIP headers and sdp in dialplan?

[Jon Bruel]

How do I read the contents of the SIP headers sdp in a dialplan?

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[Freeswitch-users] How to use Lua standard libraries

[Jon Bruel]

I want to use a string function "strfind" in a Lua script. What should I
do to access this and other standard Lua libraries?
Thanks, Jon




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[Freeswitch-users] Possible to use fresh channelvariables in conditions?

[Jon Bruel]

I have tried this: 




It runs through the conditions, but the channel variable SHOWANUMBER is
not initiated at that time. When it comes to the code above, it chooses
the anti-action. It looks as if the condition-mechanism cannot be used
if the channel variable is not defined when the system goes through the
conditions in the first run? Are there any alternative ways - except for
using a script language?
Thanks, Jon




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[Freeswitch-users] Can I use the dialplan to execute specific tasks after an incoming all has been bridged?

[Jon Bruel]

I wish to do some variable bookkeeping and database manipulation after
an incoming call has been bridged - but before hangup. If possible, I
would prefer to keep the dialplan lean and not to use the javascript
application, but I havn't figured out how. In terms of Asterisk, I would
like to emulate the dial M(macroname) option. What are the options for
doing that? And are there special consideration with respect to
accessing channel-A vs. channel-B variables?




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[Freeswitch-users] How to obtain information about undocumented applications?

[Jon Bruel]

A number of applications and other functions are not documented in the
WIKI. For example, the application displace, is not documented. Where
can I search to find the documentation regarding this and other
applications?




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[Freeswitch-users] mod_spidermonkey_odbc.so does not exist

[Jon Bruel]

Thanks Michael Jerris, that worked:
"there is a file call modules.conf in the root of the freeswitch source
dir.  Open that file and uncomment mod_spidermonkey_odbc then again run
make and make install".




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[Freeswitch-users] mod_spidermonkey_odbc.so does not exist

[Jon Bruel]

Thanks for your answer, Mashudi. I do have the same configuration, but
the problem is that the file mod_spidermonkey_odbc.so does not exist. I
trie to search for it on the server, and appently it does not exist
anywhere on the server. By - for me - unknown reasons, it is not
compiled.

Jon

 

 

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[Freeswitch-users] mod_spidermonkey_odbc.so does not exist

[Jon Bruel]

I have problems with mod_spidermonkey_odbc. Following the instructions,
I have created the ODBC environment and compiled it after running
./configure --enable-core-odbc-support.

I do not get errors.

Still the file mod_spidermonkey_odbc.so does not exist in the right
directory and the freeswitch starts up with an error when it tries to
load the module.

I get the same error on two different Linux distributions: Fedore 8 and
CentOS 4.3.

The FreeSWITCH version is 1.0.1.

Any idea of what's wrong?

Jon

 

 

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[Freeswitch-users] HA clustering solution?

[Jon Bruel]

Hi everyone,

I would like to join the choir of Tom Warren: HA with state
synchronisation would not just be a killer, it will pave the road for
the FreeSWITH all the way to the entrance of the Telco's.

As a former co-owner of a hosted PBX Telco, I would like to share my
experience about customers: They do accept a disconnection as long as it
does not happen "often", and as long as the call can be re-established
right after the disconnection. The period of "often" depends on the type
of customer, and it ranges from 1 month to 12 months. Mind you,
customers are used to the quality of mobile phones.

Therefore, I suggest that focus is put on making the core stable and
scalable as the first priority, and, as the second HA-priority, to
create the means required to ensure uninterrupted calls during 'hiccups'
in the box having the call.

Jon Bruel

Consiglia Telecommunications
Denmark

 

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[Freeswitch-users] Project "Commercial PBX using FreeSWITCH", question 3

[Jon Bruel]

I have a dream, said M. L. King.

And so do I: To create a commercial alternative to Avaya, Cisco BroadSoft and 
others based on open source components.

At this stage, I'm investigating different possibilities including the 
FreeSWITCH platform before a prospect is with the aim of raising 2-5 mio. USD 
for the initial steps to create ready-to-go-IP-PBX-package for big enterprises 
and telcos. 

I have a number of questions regarding the possibilities of the FreeSWITCH, and 
I have chosen to use this community to share the questions.

Question 3:

My experience with Asterisk has proven that the ability to monitor DTMF after 
calls have been bridges is very useful. With this facility you can transfer a 
call by pressing DTMF # or another predefined digit from any type of phone.

Is this possible with FreeSWITCH? And how?

Further if it is possible, what is the cost in terms of CPU?

 

Thanks

Jon

 

Consiglia Telecommunications
Hulsøvang 16
2960 Rungsted Kyst

Denmark

45 16 1000

CVR: 27 04 78 82

www.consiglia.dk

 

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[Freeswitch-users] Project "Commercial PBX using FreeSWITCH", question 1

[Jon Bruel]

I have a dream, said M. L. King.

And so do I: To create a commercial alternative to Avaya, Cisco BroadSoft and 
others based on open source components.

At this stage, I'm investigating different possibilities including the 
FreeSWITCH platform before a prospect is with the aim of raising 2-5 mio. USD 
for the initial steps to create ready-to-go-IP-PBX-package for big enterprises 
and telcos. 

I have a number of questions regarding the possibilities of the FreeSWITCH, and 
I have chosen to use this community to share the questions.

Question 1:

Well, as this project is likely to be quite expensive, why create needles 
competition? So if anybody is working on a similar project it might be better 
to cooperate than compete! So please let me hear if you know about any similar 
project!

 

Thanks

Jon

 

Consiglia Telekommunikation
Hulsøvang 16
2960 Rungsted Kyst

45 16 1000

CVR: 27 04 78 82

www.consiglia.dk

 

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[Freeswitch-users] Project "Commercial PBX using FreeSWITCH", question 2

[Jon Bruel]

I have a dream, said M. L. King.

And so do I: To create a commercial alternative to Avaya, Cisco BroadSoft and 
others based on open source components.

At this stage, I'm investigating different possibilities including the 
FreeSWITCH platform before a prospect is with the aim of raising 2-5 mio. USD 
for the initial steps to create ready-to-go-IP-PBX-package for big enterprises 
and telcos. 

I have a number of questions regarding the possibilities of the FreeSWITCH, and 
I have chosen to use this community to share the questions.

Question 2:

What are the possibilities to establish carrier grade system using the 
FreeSWITCH?

In practice this requirement will need a setup, where there is a redundant 
server, which takes over, if the prime server fails, including all the 
management software which reacts correctly on failure signals from the prime 
server!

 

Thanks

Jon

 

Consiglia Telecommunications
Hulsøvang 16
2960 Rungsted Kyst

45 16 1000

CVR: 27 04 78 82

www.consiglia.dk

 

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