sip proxy

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sip proxy

Postby hakimabouchaib » Wed May 23, 2007 8:49 am

hello,
i want to connecte my astersk server with un @ of sip proxy ,
i make
register=> 112233:password@"@sip server"
in sip.conf
and this
[112233]
type=peer
username=112233
fromuser=112233
secret=password
context=incoming-SIP-context-in-extensions.conf
host=@sip server

[1003]
secret=mypassword
username=1003
host=dynamic
auth-plaintext
dtmfmode=inband
context=outgoing-SIP-context-in-extensions.conf
type=friend




in extensions.conf i have make this

[outgoing-SIP-context-in-extensions.conf]
exten=>_00XXXXXXXXXXX,1,Dial(SIP/112233:password@"@sipserver"/${EXTEN})


[incoming-SIP-context-in-extensions.conf]
exten=>_112233,1,Dial(SIP/1003)



in ade fisk i have the account 1003
and when i dial a number i have this message



-- Executing Dial("SIP/1003-0817b880", "SIP/112233:password@"@sip server"/number what i have called") in new stack
May 23 14:44:12 WARNING[14777]: chan_sip.c:1991 create_addr: No such host: @"@sip server"/number what i have calledMay 23 14:44:12 WARNING[14777]: chan_sip.c:1991 create_addr: No such host: @"@sip server"/number what i have called
May 23 14:44:12 NOTICE[14777]: app_dial.c:1056 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
May 23 14:44:12 NOTICE[14777]: app_dial.c:1056 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
== Everyone is busy/congested at this time (1:0/0/1)
== Auto fallthrough, channel 'SIP/1003-0817b880' status is 'CHANUNAVAIL'


is this configuration correct ???
if not can you help me to configure sip.conf and extensions.conf corectelly!!
hakimabouchaib
 
Posts: 57
Joined: Sun Apr 01, 2007 6:37 am

Postby gerski » Wed May 23, 2007 8:54 am

what are you trying to dial? this is asterisk issue...

also there is no server "@sip server"
http://www.perfectnetworkcorp.com
Perfect Network Corporation
gerski
 
Posts: 432
Joined: Fri Jul 14, 2006 6:21 am

Postby hakimabouchaib » Wed May 23, 2007 9:17 am

gerski wrote:what are you trying to dial? this is asterisk issue...

also there is no server "@sip server"

yes this is asterisk
i want to say by sip server an sip proxy
for exemple binfone or free phone , i want just to make a test with a sip proxy that i have his @ip , i want to pass all the call to this sip proxy !!!
hakimabouchaib
 
Posts: 57
Joined: Sun Apr 01, 2007 6:37 am

Postby ramindia » Wed May 23, 2007 9:20 am

Hi

can you post your cli

asterisk > sip show registry


ram
Kindly post your feedback, if this solution works.
so its very usefull for others who join later as a NEWBIE.
ramindia
 
Posts: 688
Joined: Wed Oct 11, 2006 4:06 am
Location: India

Postby hakimabouchaib » Wed May 23, 2007 9:25 am

ramindia wrote:Hi

can you post your cli

asterisk > sip show registry


ram

Host Username Refresh State
@ip:5060 112233 1785 Registered
hakimabouchaib
 
Posts: 57
Joined: Sun Apr 01, 2007 6:37 am


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