Re: [Ekiga-list] adding ekiga as softphone to asterisk 1.6

[Thierry Simonnet <t simonnet esiee fr>]

 Hello,

I use ekiga with Asterisk 1.4 and 1.6 without major troubles.
I use 3 asterisk configuration files for SIP usage : users.conf, sip.con 
and extensions.conf

Ekiga's conf : user, name are set to 6001. Registrar : name or IP adress 
of my PBX.

You can use asterisk CLI command : sip show peers to check if user is 
registered.

Don't hesitate to contact me for more informations.

Best regards

Th. Simonnet
------------------------

users.conf :
[general]
disallow = all
allow = 
ulaw,alaw,gsm,ilbc,speex,g726,adpcm,lpc10,g729,g723,h261,h263,h263p,h264,t140,red

for each user (you can also use macros, LDAP....
[6001]
username = Esiee 1
transfer = yes
mailbox = 6001
call-limit = 100
type = friend
videosupport = yes
;textsupport = yes
fullname = Esiee 1
registersip = no
host = dynamic
callgroup = 1
context = DLPN_ESIEE1
cid_number = 6001
hasvoicemail = yes
vmsecret = 6001
email =
threewaycalling = no
hasdirectory = no
callwaiting = no
hasmanager = no
hasagent = no
hassip = yes
hasiax = no
secret = esiee1
nat = yes
qualify = yes
canreinvite = no
dtmfmode = rfc2833
insecure = port,invite
pickupgroup = 1
autoprov = no
label =
macaddress =
linenumber = 1
LINEKEYS = 1
disallow = all
allow = 
ulaw,alaw,gsm,ilbc,speex,g726,adpcm,lpc10,g729,g723,h261,h263,h263p,h264,t140,red

sip.conf : I only use this file for registering and defining trunks to 
other PBX
for example for registering ekiga.net :
[general]
register => <user>:<passwd>@ekiga.net/ekiga

[ekiga]
type=friend
fromuser=<user>
secret=<passwd>
host=ekiga.net
canreinvite=no
qualify=yes
insecure=invite,port
context = DID_ekiga
nat = yes
disallow = all
allow = 
ulaw,alaw,gsm,ilbc,speex,g726,adpcm,lpc10,g729,g723,h261,h263,h263p,h264,t140,red


extensions.conf :

[general]
static = yes
writeprotect = no
autofallthrough = yes
clearglobalvars = no
priorityjumping = no

[globals]
CONSOLE = Console/dsp
TRUNKMSD = 1
FEATURES =
DIALOPTIONS =
RINGTIME = 20
FOLLOWMEOPTIONS =
PAGING_HEADER = Intercom
CID_6001 = 6001
CID_6002 = 6002
CID_6003 = 6003
CID_6004 = 6004
CID_6005 = 6005
CID_6006 = 6006
CID_6007 = 6007
CID_6008 = 6008
CID_6009 = 6009
CID_6010 = 6010
CID_6011 = 6011
CID_6012 = 6012
ekiga = SIP/ekiga
[DLPN_ESIEE1]
include = CallingRule_ekiga
include = internal
include = default
include = parkedcalls
include = conferences
include = ringgroups
include = voicemenus
include = queues
include = voicemailgroups
include = directory
include = pagegroups
include = page_an_extension

[CallingRule_ekiga]
exten = _4.,1,NoOp()
exten = _4.,n,Dial(SIP/ekiga/${EXTEN:1},30)
exten = _4.,n,Hangup()

[internal]

exten = 6200,1,VoicemailMain()
exten = _#6XXX,1,Set(MBOX=${EXTEN:1} default)
exten = _#6XXX,n,VoiceMail(${MBOX})
exten = a,1,VoicemailMain(${MBOX})
exten => 6900,1,Verbose(1,Echo test application) ;console asterisk
exten => 6900,2,Playback(tt-weasels) ;message a ecouter
exten => 6900,n,Echo() ;je parle et je m'entend
exten => 6900,n,Hangup() ;déconnection

exten = _6XXX,1,NoOp()
exten = _6XXX,n,Dial(SIP/${EXTEN},5)
exten = _6XXX,n,GotoIf(["${DIALSTATUS}" = "BUSY"]?busy:unavail)
exten = _6XXX,n(unavail), Voicemail(${EXTEN} default,u)
exten = _6XXX,n,Hangup()
exten = _6XXX,n(busy), Voicemail(${EXTEN} default,b)
exten = _6XXX,n,Hangup()

[DID_ekiga]
include = internal


Le 04/10/2010 03:50, Tom Poe a écrit :
> Here's /etc/sip.conf file quote:
> ;sip_custom_post.conf If you have extra parameters that are needed for
> a
> ;extension to work to for example, those go here.  So you have
> extension
> ;1000 defined in your system you start by creating a line [1000](+) in
> this
> ;file.  Then on the next line add the extra parameter that is needed.
> ;When the sip.conf is loaded it will append your additions to the end of
> ;that extension.
> ;
> #include sip_custom_post.conf
>
> Here's what I found with ekiga documentation about adding ekiga to
> asterisk:
> Assuming that your Asterisk is in place and functioning, the first step is
> to make Ekiga a client of your Asterisk. You do this by having the
> following lines in sip.conf – one of Asterisk's many configuration files:
>
> [general]
> context=default
> srvlookup=yes
> videosupport=yes
>
> disallow=all   ; First disallow all codecs
> allow=ulaw
> allow=alaw     ; Allow codecs in order of
> allow=ilbc     ; preference
> allow=gsm
> allow=h261
>
> [101]
> type=friend
> secret=welcome
> qualify=yes    ; Qualify peer is not more than 2000 mS away
> nat=no         ; This phone is not natted
> host=dynamic   ; This device registers with us
> canreinvite=no ; Asterisk by default tries to redirect
> context=home
> ;port=5061     ; Uncomment this line if Ekiga and Asterisk
>                 ; are on the same host
> - - - - - - - - -
>
> Has anyone figured out how to handle the situation.  Should I treat the
> instruction that says to add to sip.conf as updated to say the info should
> be added to sip_post_custom.conf?
>
> Any help appreciated.
> Tom
> _______________________________________________
> ekiga-list mailing list
> ekiga-list gnome org
> http://mail.gnome.org/mailman/listinfo/ekiga-list