Establishing trunk between asterisk and SIP Server

Hello All,
I've tried this many times but with noe success,
Could anyone tell me If we could establish a trunk between Asterisk and SIP Server like OpenSER ?
I've created the trunk but the call doesn't seem to route to the OpenSER?

Pls. Help!
Thanks

Hi,
We are going to need a little more information.

Does Trixbox show it has registered with the Ser server?
If the Ser server is yours, does it see your Trixbox server?
Have a look at the full Asterisk log, does it show an error, is it actually trying to pass the call to Ser?

How have you configured the inbound and outbound trunk contexts?

Thank you,
Well My Primary concern at present is for outbound calls and hence i have defined an outbound trunk for outgoing calls from asterisk. The asterisk log doesn't give out any error !
The log is as below :

Code:

Executing GotoIf("SIP/603-086b7860", "1?report") in new stack     -- Goto (macro-outbound-callerid,s,22)     -- Executing NoOp("SIP/603-086b7860", "CallerID set to "Rav sip test" <603>") in new stack     -- Executing GotoIf("SIP/603-086b7860", "1?nomax") in new stack     -- Goto (macro-dialout-trunk,s,17)     -- Executing AGI("SIP/603-086b7860", "fixlocalprefix") in new stack     -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix        >  fixlocalprefix: Using pattern 8|.XXX        >  fixlocalprefix: Using pattern 8|.XXX     -- AGI Script fixlocalprefix completed, returning 0     -- Executing Set("SIP/603-086b7860", "OUTNUM=101") in new stack     -- Executing Set("SIP/603-086b7860", "custom=SIP/sip_out") in new stack     -- Executing GotoIf("SIP/603-086b7860", "1?gocall") in new stack     -- Goto (macro-dialout-trunk,s,24)     -- Executing GotoIf("SIP/603-086b7860", "0?customtrunk") in new stack     -- Executing Dial("SIP/603-086b7860", "SIP/sip_out/101|300|") in new stack     -- Called sip_out/101     -- AGI Script fixlocalprefix completed, returning 0     -- Executing Set("SIP/603-086b7860", "OUTNUM=101") in new stack     -- Executing Set("SIP/603-086b7860", "custom=SIP/sip_out") in new stack     -- Executing GotoIf("SIP/603-086b7860", "1?gocall") in new stack     -- Goto (macro-dialout-trunk,s,24)     -- Executing GotoIf("SIP/603-086b7860", "0?customtrunk") in new stack     -- Executing Dial("SIP/603-086b7860", "SIP/sip_out/101|300|") in new stack     -- Called sip_out/101   == Everyone is busy/congested at this time (1:0/0/1)     -- Executing Goto("SIP/603-086b7860", "s-CHANUNAVAIL|1") in new stack     -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)     -- Executing GotoIf("SIP/603-086b7860", "1?noreport") in new stack     -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,3)     -- Executing NoOp("SIP/603-086b7860", "TRUNK Dial failed due to CHANUNAVAIL - failing through to other trunks") in new stack     -- Executing Macro("SIP/603-086b7860", "outisbusy|") in new stack     -- Executing Playback("SIP/603-086b7860", "all-circuits-busy-now|noanswer") in new stack     -- Playing 'all-circuits-busy-now' (language 'en')   == Everyone is busy/congested at this time (1:0/0/1)     -- Executing Goto("SIP/603-086b7860", "s-CHANUNAVAIL|1") in new stack     -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)     -- Executing GotoIf("SIP/603-086b7860", "1?noreport") in new stack     -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,3)     -- Executing NoOp("SIP/603-086b7860", "TRUNK Dial failed due to CHANUNAVAIL - failing through to other trunks") in new stack     -- Executing Macro("SIP/603-086b7860", "outisbusy|") in new stack     -- Executing Playback("SIP/603-086b7860", "all-circuits-busy-now|noanswer") in new stack     -- Playing 'all-circuits-busy-now' (language 'en')     -- Executing Playback("SIP/603-086b7860", "pls-try-call-later|noanswer") in new stack     -- Playing 'pls-try-call-later' (language 'en')     -- Executing Playback("SIP/603-086b7860", "pls-try-call-later|noanswer") in new stack     -- Playing 'pls-try-call-later' (language 'en')     -- Executing Macro("SIP/603-086b7860", "hangupcall") in new stack     -- Executing ResetCDR("SIP/603-086b7860", "w") in new stack     -- Executing NoCDR("SIP/603-086b7860", "") in new stack     -- Executing GotoIf("SIP/603-086b7860", "1?skiprg") in new stack     -- Goto (macro-hangupcall,s,6)     -- Executing GotoIf("SIP/603-086b7860", "1?skipblkvm") in new stack     -- Goto (macro-hangupcall,s,9)     -- Executing GotoIf("SIP/603-086b7860", "1?theend") in new stack     -- Goto (macro-hangupcall,s,11)     -- Executing Hangup("SIP/603-086b7860", "") in new stack   == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/603-086b7860' in macro 'hangupcall'   == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/603-086b7860' in macro 'outisbusy'   == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/603-086b7860'     -- Executing Macro("SIP/603-086b7860", "hangupcall") in new stack     -- Executing ResetCDR("SIP/603-086b7860", "w") in new stack     -- Executing NoCDR("SIP/603-086b7860", "") in new stack     -- Executing GotoIf("SIP/603-086b7860", "1?skiprg") in new stack     -- Goto (macro-hangupcall,s,6)     -- Executing GotoIf("SIP/603-086b7860", "1?skipblkvm") in new stack     -- Goto (macro-hangupcall,s,9)     -- Executing GotoIf("SIP/603-086b7860", "1?theend") in new stack     -- Goto (macro-hangupcall,s,11)     -- Executing Hangup("SIP/603-086b7860", "") in new stack   == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/603-086b7860' in macro 'hangupcall'   == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/603-086b7860' in macro 'outisbusy'   == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/603-086b7860'

At the same time I am not able to get any logs at the OpenSER

You do indeed have an error- It returns channel unavailable

This would mean that:-
Trixbox cannot see Ser- check your network settings in sip-nat.conf
The outbound context is wrong-
Or Openser doesn't like your Trixbox settings-

Not of great help I know and I'm no authority on OpenSer

Set the Asterisk context to type=friend
set qualify=yes
check the host setting, If not already done so make sure it is aimed at the IP address.
These will at least allow you to check Trixbox is connecting.

Have you tried getting an ordinary SIP phone to connect to OpenSer. If not, I suggest this may be a good place to start from. Once that works, you will have settings to feed Asterisk

Thank you rgower,

I'm trying to sort out the things what you mentioned in your first portion of your reply,

Yes, I've tried out Xlite to register accounts at OpenSER, It registers all fine and I'm indeed able to make call between the a/c's which suggests that OpenSER is configured OK.

Could you suggest some more options?

Thanks

Hey,
Enable sip debug on the command line and set verbose 11. Post the output (It is going to be quite long)

Thanks

<snip>Please see our Posting Rules and Guidelines - thanks</snip>

Here you are the debug messages at the asterisk verbose = 11

Code:

-- Executing Macro("SIP/603-09a7e220", "dialout-trunk|1|101||") in new stack     -- Executing Set("SIP/603-09a7e220", "DIAL_TRUNK=1") in new stack     -- Executing Set("SIP/603-09a7e220", "DIAL_NUMBER=101") in new stack     -- Executing Set("SIP/603-09a7e220", "ROUTE_PASSWD=") in new stack     -- Executing GotoIf("SIP/603-09a7e220", "1?noauth") in new stack     -- Goto (macro-dialout-trunk,s,6)     -- Executing GotoIf("SIP/603-09a7e220", "0?disabletrunk|1") in new stack     -- Executing Set("SIP/603-09a7e220", "_NODEST=") in new stack     -- Executing Set("SIP/603-09a7e220", "DIAL_TRUNK_OPTIONS=tr") in new stack     -- Executing Set("SIP/603-09a7e220", "GROUP()=OUT_1") in new stack     -- Executing Macro("SIP/603-09a7e220", "user-callerid|SKIPTTL") in new stack     -- Executing NoOp("SIP/603-09a7e220", "user-callerid: device 603") in new stack     -- Executing Set("SIP/603-09a7e220", "AMPUSER=603") in new stack     -- Executing GotoIf("SIP/603-09a7e220", "0?report") in new stack     -- Executing GotoIf("SIP/603-09a7e220", "0?start") in new stack     -- Executing Set("SIP/603-09a7e220", "REALCALLERIDNUM=603") in new stack     -- Executing NoOp("SIP/603-09a7e220", "REALCALLERIDNUM is 603") in new stack     -- Executing Set("SIP/603-09a7e220", "AMPUSER=603") in new stack     -- Executing Set("SIP/603-09a7e220", "AMPUSERCIDNAME=Rav sip test") in new stack     -- Executing GotoIf("SIP/603-09a7e220", "0?report") in new stack     -- Executing Set("SIP/603-09a7e220", "AMPUSERCID=603") in new stack     -- Executing Set("SIP/603-09a7e220", "CALLERID(all)="Rav sip test" <603>") in new stack     -- Executing Set("SIP/603-09a7e220", "REALCALLERIDNUM=603") in new stack     -- Executing NoOp("SIP/603-09a7e220", "TTL:  ARG1: SKIPTTL") in new stack     -- Executing GotoIf("SIP/603-09a7e220", "1?continue") in new stack     -- Goto (macro-user-callerid,s,23)     -- Executing NoOp("SIP/603-09a7e220", "Using CallerID "Rav sip test" <603>") in new stack     -- Executing Macro("SIP/603-09a7e220", "record-enable|603|OUT") in new stack     -- Executing GotoIf("SIP/603-09a7e220", "0?2:4") in new stack     -- Goto (macro-record-enable,s,4)     -- Executing AGI("SIP/603-09a7e220", "recordingcheck|20080108-164054|1199789754.34") in new stack     -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck   recordingcheck|20080108-164054|1199789754.34: Outbound recording not enabled     -- AGI Script recordingcheck completed, returning 0     -- Executing NoOp("SIP/603-09a7e220", "No recording needed") in new stack     -- Executing GotoIf("SIP/603-09a7e220", "0?skipoutcid") in new stack     -- Executing Set("SIP/603-09a7e220", "DIAL_TRUNK_OPTIONS=") in new stack     -- Executing Macro("SIP/603-09a7e220", "outbound-callerid|1") in new stack     -- Executing GotoIf("SIP/603-09a7e220", "1?start") in new stack     -- Goto (macro-outbound-callerid,s,3)     -- Executing NoOp("SIP/603-09a7e220", "REALCALLERIDNUM is 603") in new stack     -- Executing GotoIf("SIP/603-09a7e220", "1?normcid") in new stack     -- Goto (macro-outbound-callerid,s,9)     -- Executing Set("SIP/603-09a7e220", "USEROUTCID=") in new stack     -- Executing Set("SIP/603-09a7e220", "EMERGENCYCID=") in new stack     -- Executing Set("SIP/603-09a7e220", "TRUNKOUTCID=") in new stack     -- Executing GotoIf("SIP/603-09a7e220", "1?trunkcid") in new stack     -- Goto (macro-outbound-callerid,s,16)     -- Executing GotoIf("SIP/603-09a7e220", "1?usercid") in new stack     -- Goto (macro-outbound-callerid,s,18)     -- Executing GotoIf("SIP/603-09a7e220", "1?report") in new stack     -- Goto (macro-outbound-callerid,s,22)     -- Executing NoOp("SIP/603-09a7e220", "CallerID set to "Rav sip test" <603>") in new stack     -- Executing GotoIf("SIP/603-09a7e220", "1?nomax") in new stack     -- Goto (macro-dialout-trunk,s,17)     -- Executing AGI("SIP/603-09a7e220", "fixlocalprefix") in new stack     -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix        >  fixlocalprefix: Using pattern 8|.XXX     -- AGI Script fixlocalprefix completed, returning 0     -- Executing Set("SIP/603-09a7e220", "OUTNUM=101") in new stack     -- Executing Set("SIP/603-09a7e220", "custom=SIP/sip_out") in new stack     -- Executing GotoIf("SIP/603-09a7e220", "1?gocall") in new stack     -- Goto (macro-dialout-trunk,s,24)     -- Executing GotoIf("SIP/603-09a7e220", "0?customtrunk") in new stack     -- Executing Dial("SIP/603-09a7e220", "SIP/sip_out/101|300|") in new stack     -- Called sip_out/101   == Everyone is busy/congested at this time (1:0/0/1)     -- Executing Goto("SIP/603-09a7e220", "s-CHANUNAVAIL|1") in new stack     -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)     -- Executing GotoIf("SIP/603-09a7e220", "1?noreport") in new stack     -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,3)     -- Executing NoOp("SIP/603-09a7e220", "TRUNK Dial failed due to CHANUNAVAIL - failing through to other trunks") in new stack     -- Executing Macro("SIP/603-09a7e220", "outisbusy|") in new stack     -- Executing Playback("SIP/603-09a7e220", "all-circuits-busy-now|noanswer") in new stack     -- Playing 'all-circuits-busy-now' (language 'en')     -- Executing Playback("SIP/603-09a7e220", "pls-try-call-later|noanswer") in new stack     -- Playing 'pls-try-call-later' (language 'en')   == Spawn extension (macro-outisbusy, s, 2) exited non-zero on 'SIP/603-09a7e220' in macro 'outisbusy'   == Spawn extension (macro-outisbusy, s, 2) exited non-zero on 'SIP/603-09a7e220'

Hello,
You are still missing the sip debug portion. (sip set debug) in asterisk 1.4 (sip debug) in 1.2

Thanks
-Thelostpacket

Thanks TheLostPacket,

Ya I had my outgoing trunk misconfigured,
Now that I've configured the calls are routing as wished.

Thank you so very much for the debugging command

anamupota

could you post how you fixed this? im getting the same thing