[Asterisk-Users] Dial command nor progressing on Zap channels

Eric Bishop asterisk.eric at gmail.com
Fri Aug 26 14:03:55 MST 2005


Hi all,

Our Asterisk box sends calls outbound via either SIP (through our VoIP
provider) or an E1 PRI (directly connected via a TE410P). When we dial
a number that is engaged via our VoIP provider we get the following on
the Asterisk console (numbers and IP addresses changed to protect the
innocent):

 -- Called 12345678 at sip-outbound
  -- Got SIP response 486 "Busy here" back from 123.123.123.123
  -- SIP/sip-outbound-af71 is busy
 == Everyone is busy/congested at this time

This is what we want as it then send the call to priority n+101 and we
can handle it any way we want from there. However if the outbound call
is made via the PRI (Zap channel) to an enaged number it simply plays an enaged
signal to the caller and never progresses past the Dial priority. I
know for a fact the call is not actually being answered, because I get
the following onthe console.

Executing Dial("SIP/1001-270b", "Zap/g1/123456789") in new stack
    -- Called g1/123456789

pri debug span 1 gives me:

< Protocol Discriminator: Q.931 (8)  len=5
< Call Ref: len= 2 (reference 32809/0x8029) (Terminator)
< Message type: RELEASE (77)
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Release Request
> Protocol Discriminator: Q.931 (8)  len=9
> Call Ref: len= 2 (reference 41/0x29) (Originator)
> Message type: RELEASE COMPLETE (90)
> [08 02 81 90]
> Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0) 0: 0   Location: Private network serving the local user (1)
>                  Ext: 1  Cause: Normal Clearing (16), class = Normal Event (1) ]
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null
NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null
    -- Executing Macro("SIP/1001-8cdc", "Dial_Telco_ISDN|123456789")
in new stack
    -- Executing SetAccount("SIP/1001-8cdc", "TELCO-ISDN") in new stack
    -- Executing NoOp("SIP/1001-8cdc", "") in new stack
    -- Executing Dial("SIP/1001-8cdc", "Zap/g1/123456789") in new stack
-- Making new call for cr 32810
> Protocol Discriminator: Q.931 (8)  len=51
> Call Ref: len= 2 (reference 42/0x2A) (Originator)
> Message type: SETUP (5)
> [04 03 80 90 a3]
> Bearer Capability (len= 5) [ Ext: 1  Q.931 Std: 0  Info transfer capability: Speech (0)
>                              Ext: 1  Trans mode/rate: 64kbps, circuit-mode (16)
>                              Ext: 1  User information layer 1: A-Law (35)
> [18 03 a9 83 81]
> Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0
>                        ChanSel: Reserved
>                       Ext: 1  Coding: 0   Number Specified   Channel Type: 3
>                       Ext: 1  Channel: 1 ]
> [28 09 41 6c 6c 61 6e 20 44 69 62]
> Display (len= 9) [ Eric Bishop ]
> [6c 09 21 81 33 30 30 31 30 30 31]
> Calling Number (len=11) [ Ext: 0  TON: National Number (2)  NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1)
>                           Presentation: Presentation permitted, user number passed network screening (1) '3001001' ]
> [70 0b 80 30 33 39 35 37 30 32 37 30 38]
> Called Number (len=13) [ Ext: 1  TON: Unknown Number Type (0)  NPI: Unknown Number Plan (0) '123456789' ]
> [a1]
> Sending Complete (len= 1)
    -- Called g1/123456789


Why is the Dial command not progeessing?



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