Hi there,
I am using asterisk as an H323 GW through the chan_h323 so that
SIP terminals can interact with an H323 network managed by gnugk.
When a SIP phone calls an H323 phone, everything works just fine. Except that
the caller-id is not sent correctly.
Now, what I need to do is to stablish a call from SIP to SIP authenticating
through the H323 Gatekeeper.
Is this possible?
I have tried it, and it did not work. These are my config files:
/***************************************************
h323.conf
/***************************************************
[general]
port = 1720
bindaddr = 0.0.0.0
disallow=all
allow=g729
allow=G723.1
allow=ulaw ; Allow codecs in order of preference
allow=alaw
gatekeeper = 192.168.0.103
context=h323
[1005] ; When this line and the context [1004] lines are set
type=h323 ; the caller id 1004 is always sent. I don't know why.
e164=011005 ; In case, this lines are not set, the GS phones receives
context=default ; "Error" as the caller id, and the H323 phone receives
; "asterisk" as the caller-id
[1004]
type=h323
e164=011004
context=default
[asterisk]
type=h323
prefix=01
context=h323
/**************************************************
extensions.conf (Just a few lines, the rest is the standar extensions.conf
file)
/**************************************************
;(...)
[default]
;(...)
exten => 021005,1,Dial(h323/011005)
exten => 021004,1,Dial(h323/011004)
;exten => _03XXX,1,Dial(h323/${EXTEN:2})
exten => 301,1,Dial(h323/301)
[h323]
exten => 011005,1,Dial(SIP/1005)
exten => 011004,1,Dial(SIP/1004)
/**************************************************
sip.conf
/**************************************************
[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind SIP channel to
context = default ; Default context for incoming calls
disallow=all ; Disallow all codecs
allow=g729
allow=G723.1
allow=ulaw ; Allow codecs in order of preference
allow=alaw
[1004]
type=friend
username=1004
secret=123
host=dynamic
canreinvite=no
[1005]
type=friend
username=1005
secret=123
host=dynamic
canreinvite=no
/*****************************************************/
/*****************************************************/
Also, find attached an ethereal file with the call flow for a call from SIP to
SIP through my Gatekeeper
My scenario is the following:
SIP Phones: Grandstream phones
H323 Phone: Planet IP phone
Gatekeeper: gnugk 1.0.7
Asterisk: version 0.9.0
As you can see, asterisk doesn't inform the calling partie that the called
partie has picked up the phone, so no RTP conexion is stablished between
asterisk and the calling partie. This happens always, without the matter of
who is the calling partie(1004 or 1005)
I am really in a hurry here. Please, somebody help me.
Pablo Salinas
Attachment:
sip-sip-through-gk
Description: Binary data
