Asterisk returns that a number is not in service

[jfmays]

Have an asterisk server set up that is handling phone calls from a number of sources. On a new Cisco IAD2400 setup, inbound calling to a phone on the IAD2400 works fine, but when you try to place an outbound call from the phone the asterisk returns that the number is not in service. It does this for any number you call from the phone.

Here is an attempt to call my cell phone from the IAD2400 (216.24.28.250). My cell phone number works fine, and calls from other SIP trunking clients out to the PSTN work fine through this server, but for some reason calls from the IAD all get that the number is not in service.

Code: Select all

216-24-23-90*CLI>
  == Using UDPTL TOS bits 184
  == Using UDPTL CoS mark 5
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- Executing [5023871095@from-sip-external:1] NoOp("SIP/216.24.23.90-000002ee", "Received incoming SIP connection from unknown peer to 5023871095") in new stack
    -- Executing [5023871095@from-sip-external:2] Set("SIP/216.24.23.90-000002ee", "DID=5023871095") in new stack
    -- Executing [5023871095@from-sip-external:3] Goto("SIP/216.24.23.90-000002ee", "s,1") in new stack
    -- Goto (from-sip-external,s,1)
    -- Executing [s@from-sip-external:1] GotoIf("SIP/216.24.23.90-000002ee", "0?checklang:noanonymous") in new stack
    -- Goto (from-sip-external,s,5)
    -- Executing [s@from-sip-external:5] Set("SIP/216.24.23.90-000002ee", "TIMEOUT(absolute)=15") in new stack
Channel will hangup at 2012-04-30 14:30:48.533 EDT.
    -- Executing [s@from-sip-external:6] Answer("SIP/216.24.23.90-000002ee", "") in new stack
    -- Executing [s@from-sip-external:7] Wait("SIP/216.24.23.90-000002ee", "2") in new stack
    -- Executing [s@from-sip-external:8] Playback("SIP/216.24.23.90-000002ee", "ss-noservice") in new stack
    -- <SIP/216.24.23.90-000002ee> Playing 'ss-noservice.ulaw' (language 'en')
216-24-23-90*CLI>


[david55]

Please ask the person who wrote the dialplan. We have no idea what it is trying to do.

[navaismo]

This call is from anonymous sip? im right? And yor distro is using freepbx like system, so you are allowing sip guest but you didnt oveeride that context and for default it will play ss-noservice message.

[jfmays]

This call is from anonymous sip? im right? And yor distro is using freepbx like system, so you are allowing sip guest but you didnt oveeride that context and for default it will play ss-noservice message.

This sounds pretty much right. We are not requiring authentication on connections because we are limiting who can connect by ip number, and in some other ways. But I suppose in the context of the asterisk box these are guest users. We don't have this problem with most ATA's we have configured to talk through the asterisk server, but I am guessing that they are telling something to the asterisk server that allows it to override the guest status for them. I just need to figure out what that is and figure out how to tell the asterisk box to do that.

What did you have in mind when you referred to overriding the context?

[david55]

You are really in the wrong forum. Your issue is with the use of FreePBX, not with Asterisk.