sip and iax2 trunk
Posted:
Tue Oct 31, 2006 10:25 am
by rudra_ach
Hi,
I'm trying use a sip provider.They don't support sip trunking.They have given me userid password and host ip address.Now How I can set the termination.
Anather thing is when createing sip and iax trunk in a single server they both are not working.Please suggest how to do it.
regards
rudra
Posted:
Tue Oct 31, 2006 10:38 am
by ramindia
Hi
just register with the user name password
give by your provider example
sip.conf
register => account:password@ipaddress of the provider
[account]
authuser=account
fromdomain=providerip
fromuser=account
host=providerip
insecure=very
nat=yes
qualify=yes
secret=password
sendrpid=yes
type=peer
username=account@providerip
extensions.conf
exten => _1.,2,Dial(SIP/account/${EXTEN})
you should able to make calls out
its working config of mine
let me know if the solution works
Ram
Posted:
Tue Oct 31, 2006 10:49 am
by rudra_ach
Hi,
following erroe I'm seeing when dialing through sip
-- Executing AGI("IAX2/100-8", "call_log.agi|60018775111876") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/call_log.agi
-- AGI Script call_log.agi completed, returning 0
-- Executing Dial("IAX2/100-8", "SIP/useridXX:passwordXXXXXX@203.196.128.56/0018775111876|30|r") in new stack
Oct 31 21:16:25 WARNING[6533]: chan_sip.c:1989 create_addr: No such host: 203.196.128.56/0018775111876
Oct 31 21:16:25 WARNING[6533]: chan_sip.c:1989 create_addr: No such host: 203.196.128.56/0018775111876
Oct 31 21:16:25 NOTICE[6533]: app_dial.c:1056 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
Oct 31 21:16:25 NOTICE[6533]: 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)
-- Executing Hangup("IAX2/100-8", "") in new stack
== Spawn extension (default, 60018775111876, 3) exited non-zero on 'IAX2/100-8'
-- Executing DeadAGI("IAX2/100-8", "call_log.agi|h") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/call_log.agi
-- AGI Script call_log.agi completed, returning 0
-- Executing DeadAGI("IAX2/100-8", "VD_hangup.agi|h") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/VD_hangup.agi
-- AGI Script VD_hangup.agi completed, returning 0
-- Hungup 'IAX2/100-8'
Posted:
Tue Oct 31, 2006 12:58 pm
by rudra_ach
Thankx ramindia
This setup worked for me.call is going with charm.
regards
rudra
ramindia wrote:Hi
just register with the user name password
give by your provider example
sip.conf
register => account:password@ipaddress of the provider
[account]
authuser=account
fromdomain=providerip
fromuser=account
host=providerip
insecure=very
nat=yes
qualify=yes
secret=password
sendrpid=yes
type=peer
username=account@providerip
extensions.conf
exten => _1.,2,Dial(SIP/account/${EXTEN})
you should able to make calls out
its working config of mine
let me know if the solution works
Ram
Posted:
Tue Oct 31, 2006 12:58 pm
by rudra_ach
Thankx ramindia
This setup worked for me.call is going with charm.
regards
rudra
ramindia wrote:Hi
just register with the user name password
give by your provider example
sip.conf
register => account:password@ipaddress of the provider
[account]
authuser=account
fromdomain=providerip
fromuser=account
host=providerip
insecure=very
nat=yes
qualify=yes
secret=password
sendrpid=yes
type=peer
username=account@providerip
extensions.conf
exten => _1.,2,Dial(SIP/account/${EXTEN})
you should able to make calls out
its working config of mine
let me know if the solution works
Ram