Asterisk Incoming Call

Hi,
Actually I am new for Asterisk , I tested Asterisk Outgoing call by changing sip.conf and extensions.conf .In Asterisk CLI I am getting

sip show peers : Monitored

sip show registry : Registered trunk

Now I am unable to test incoming call . What changes I needed to made for inbound call testing with DID . Any tutorial link for this will be helpful .

I followed for inbound with DID : callcentric.com/support/device/did_routing

Kindly help me .

it is really simple setup CallCentric with Asterisk. Post your sip.conf and extensions.conf please

Do you see anything coming on Asterisk CLI when a call on DID is made? Post Asterisk CLI output.

–Satish Barot
satish4asterisk@gmail.com

Today I started to test is again and now I am not able to make outbound call too :frowning:

There are CLI Output (with [color=#004000]verbosity set to 5[/color]) has been given below:

– Executing [8600051@default:1] Dial(“Local/8600051@default-05b7,2”, "SIP/8600051@voiptrunk stack
[Mar 19 11:12:15] – Called 8600051@voiptrunk
[Mar 19 11:12:16] == Parsing ‘/etc/asterisk/manager.conf’: [Mar 19 11:12:16] Found
[Mar 19 11:12:16] == Manager ‘sendcron’ logged on from 127.0.0.1
[Mar 19 11:12:16] – Executing [58600051@default:1] Dial(“Local/58600051@default-18af,2”, "SIP/58600051@voiptrunk new stack
[Mar 19 11:12:16] – Called 58600051@voiptrunk
[Mar 19 11:12:16] – Got SIP response 603 “Declined” back from XXX.XXX.XX.XXX
[Mar 19 11:12:16] – SIP/voiptrunk-00000003 is busy
[Mar 19 11:12:16] == Everyone is busy/congested at this time (1:1/0/0)
[Mar 19 11:12:16] == Auto fallthrough, channel ‘Local/58600051@default-18af,2’ status is ‘BUSY’
[Mar 19 11:12:16] > Channel Local/58600051@default-18af,1 was never answered.
[Mar 19 11:12:16] == Manager ‘sendcron’ logged off from 127.0.0.1
[Mar 19 11:12:16] == Parsing ‘/etc/asterisk/manager.conf’: [Mar 19 11:12:16] Found
[Mar 19 11:12:16] == Manager ‘sendcron’ logged on from 127.0.0.1
[Mar 19 11:12:20] > Channel SIP/8002-00000004 was answered.
[Mar 19 11:12:20] == Manager ‘sendcron’ logged off from 127.0.0.1
[Mar 19 11:12:20] – Executing [8600051@default:1] Dial(“SIP/8002-00000004”, “SIP/8600051@voiptrunk”) in n
[Mar 19 11:12:20] – Called 8600051@voiptrunk
[Mar 19 11:12:20] – Got SIP response 603 “Declined” back from XXX.XXX.XX.XXX
[Mar 19 11:12:20] – SIP/voiptrunk-00000005 is busy
[Mar 19 11:12:20] == Everyone is busy/congested at this time (1:1/0/0)
[Mar 19 11:12:20] == Auto fallthrough, channel ‘SIP/8002-00000004’ status is ‘BUSY’
[Mar 19 11:12:33] – SIP/voiptrunk-00000002 is circuit-busy
[Mar 19 11:12:33] == Everyone is busy/congested at this time (1:0/1/0)
[Mar 19 11:12:33] == Auto fallthrough, channel ‘Local/8600051@default-05b7,2’ status is ‘CONGESTION’
[Mar 19 11:12:33] > Channel Local/8600051@default-05b7,1 was never answered.

[color=#000080]There are two status in Autofallthrough a). ‘BUSY’ b). ‘CONGESTION’[/color]

When I have dialed the DID number from Mobile (and there is no SIP Phone )

I got the CLI output :

[Mar 19 11:55:39] – Executing [s@trunkinbound:1] Goto(“SIP/voiptrunk-0000002d”, “incoming|s|1”) in new stack
[Mar 19 11:55:39] – Goto (incoming,s,1)
[Mar 19 11:55:39] – Executing [s@incoming:1] Set(“SIP/voiptrunk-0000002d”, “Var_TO=sip:s@XXX.XXX.XX.XXX:49152”) in new stack
[Mar 19 11:55:39] – Executing [s@incoming:2] GotoIf(“SIP/voiptrunk-0000002d”, “0?extension1|s|1:3”) in new stack
[Mar 19 11:55:39] – Goto (incoming,s,3)
[Mar 19 11:55:39] == Auto fallthrough, channel ‘SIP/voiptrunk-0000002d’ status is ‘UNKNOWN’

Kindly help me to solve it .

Thanks !

First part:

Got SIP response 603 “Declined” back from XXX.XXX.XX.XXX

Second part:

Looks like you don’t have a priority 3 in your dial plan. You haven’t told us what condition was being evaluated by the GotoIf.

Thanks for the clue Mr. david55.

I have made a small mistake I replaced the priority with ‘n’ that was the error in my case . For test I removed the gotoif conditional jump (that was actually not needed in my case). The Test was successful.