Hi Alex Thank you for responding

I''ve checked my error.log file and seems to be an exception like "table phoneServer doesn''t exist" or something like that, im sorry but I have not acces to the log file rigth now because im in home.

Im using a MySQL database installed according to the online docuementation and there are some tables with the prefix "phone" but nothig related with server.  I hope you can help me, if you need more information about my configuration or servers please let me know

Thank you  !!!

I just see that is a new version of the wildfire''s beta , I''ll try it next monday and let you know how did it work, Thank you !!!

The new beta version of wildfire should not need any manual updates of the database rigth?

greetings

Hi Alex !!! (And everybody)

It works !! I installed the new wildfire''s beta version and my asterisk phone server is now registered in wildfire admin console, however, I''ve could not make a call using spark yet, I have read in the forum how to do it but it is not working for me yet, I hope that you can help me, here is the infomation:

my asterisk server is in the same pc that my wildfire server and im triying to make a call between two clients using spark, this clients are in two diferents pc''s

The information of my  asterisk''s manager.conf file is:

enabled = yes

port = 5038

bindaddr = 127.0.0.1

displayconnects = yes

secret = my_pass

deny=0.0.0.0/0.0.0.0

permit=10.x.x.x/255.255.255.0

permit=10.x.x.x/255.255.255.0

permit=127.0.0.1/255.255.255.0

read = system,call,log,verbose,command,agent,user

write = system,call,log,verbose,command,agent,use

______________________________

my sip.conf file is:

general

context=default

realm=localhost

bindport=5060

bindaddr=0.0.0.0

srvlookup=yes

disallow=all

allow=ulaw           &nbs p;   

allow=alaw

allow=speex

allow=gsm

allow=h261

allow=h263

allow=h263p

type=friend

secret=user1

qualify=yes

nat=no

host=dynamic

canreinvite=no

context=internal

type=friend

secret=user2

qualify=yes

nat=no

host=dynamic

canreinvite=no

context=internal

______________________________

my extensions.conf file is:

internal

exten=>100,1,Dial(SIP/user1)

exten=>101,1,Dial(SIP/user2)

exten=>611,1,Echo();

______________________________

My asterisk server registration data in the wildfire admin console is:

server name: localhost

server address: 10.x.x.x

port:5038

username: my_user

password: my_pass

______________________________

so, what should i use in the Phone Mappings form to make a call from the user1 to user2 using spark?, the fields requeired in the form are:

username

phone

extension

caller ID

primary: yes/no

what else sould I configure? im really really confused about this part.

Thank you, and if you need more information about my configuration please let me know

Hi Alex

yes i''ve already configure my sip.conf file but i forgot to put the configuration complete in my last post, Im Sorry. I have edited the post, can you check it again please?

Thank you for helping me.

Greetings from Mexico !

nop, there is anything y my error log. the questions are:

What should i put in the phone Mappings form to enable the phone calls throug spark client?

and

are my asterisk''s configuration files rigth to do what im trying to do?

Thank you !!!

hi Again !!

I''ve been testing my asterisk-im configuration but it''s still not working, I''ve registered the users in the phone mapping form just like you  said and it works but when I start sesion in spark a dont see any call button enabled, the only way to make calls is doing rigth click on the name of the user that im trying to call, when i try , asterisk''s console shows me this message:

"Oct  3 09:20:05 NOTICE[3815]: channel.c:2491 __ast_request_and_dial: Unable to request channel SIP/user2"

All my users and extensions are registered just like a posted before And i dont know what is going on? can you help me?... typing "sip show settings" in the asterisk consoles shows this:

Global Settings:

-

  SIP Port:            &n bsp;  5060

  Bindaddress:            0.0.0.0

  Videosupport:           Yes

  AutoCreatePeer:         No

  Allow unknown access:   Yes

  Promsic. redir:         No

  SIP domain support:     No

  Call to non-local dom.: Yes

  URI user is phone no:   No

  Our auth realm          localhost

  Realm. auth:            No

  Always auth rejects:    No

  User Agent:             Asterisk PBX

  MWI checking interval:  10 secs

  Reg. context:           (not set)

Caller ID:            &nbs p; asterisk

  From: Domain:

  Record SIP history:     Off

  Call Events:            Off

  IP ToS:            &nb sp;    0x0

  OSP Support:            No

  SIP realtime:           Disabled

Global Signalling Settings:

-

  Codecs:                  ulaw,alaw,speex,gsm,h261,h263,h263p

  Relax DTMF:             No

  Compact SIP headers:    No

  RTP Timeout:            0 (Disabled)

  RTP Hold Timeout:       0 (Disabled)

  MWI NOTIFY mime type:   application/simple-message-summary

  DNS SRV lookup:         Yes

  Pedantic SIP support:   No

  Reg. max duration:      3600 secs

  Reg. default duration:  120 secs

  Outbound reg. timeout:  20 secs

  Outbound reg. attempts: 0

  Notify ringing state:   Yes

Default Settings:

-

  Context:           &nbsp ;    default

  Nat:            &nb sp;       RFC3581

  DTMF:            &n bsp;      rfc2833

  Qualify:           &nbsp ;    0

  Use ClientCode:         No

  Progress inband:        Never

  Language:           &nbs p;   (Defaults to English)

  Musicclass:           &n bsp; default

  Voice Mail Extension:   asterisk

Thank you and I hope that this can be useful fore someone else too

Greetings !!