[asterisk-bugs] [Asterisk 0014921]: Cannot make outbound call through analog trunk

Asterisk Bug Tracker noreply at bugs.digium.com
Thu Apr 30 04:18:03 CDT 2009


A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=14921 
====================================================================== 
Reported By:                colynd
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   14921
Category:                   Channels/chan_dahdi
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.6.0 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
====================================================================== 
Date Submitted:             2009-04-17 02:03 CDT
Last Modified:              2009-04-30 04:18 CDT
====================================================================== 
Summary:                    Cannot make outbound call through analog trunk
Description: 
Cannot make outbound calls. inbound works fine. Also same outbound rules
work with Asterisk 1.4.x and zaptel(using AsteriskNow 1.0.2).

Asterisk version is actually 1.6.0.9

DAHDI Version: 2.1.0.4 Echo Canceller: 

This is what I get in the log when I try to dial out:

[Apr 17 08:48:47] WARNING[5632] chan_dahdi.c: Unable to determine channel
for data trunk_1/0117831508
[Apr 17 08:48:47] WARNING[5632] app_dial.c: Unable to create channel of
type 'DAHDI' (cause 0 - Unknown)

====================================================================== 

---------------------------------------------------------------------- 
 (0103993) ivan77 (reporter) - 2009-04-30 04:18
 http://bugs.digium.com/view.php?id=14921#c103993 
---------------------------------------------------------------------- 
I have same problem. I am using openvox a800p - 4fxo's and i cannot make
outbound calls when i configure Outbound Calling Rules using Asterisk GUI.
It works fine if outbound route is configured manually (in
extensions.conf). i am getting following errors:


    -- Executing [9995 at DLPN_dp_first:1] Macro("SIP/1001-008646a0",
"trunkdial-failover-0.3|DAHDI/trunk_1/9995||trunk_1|") in new stack
    -- Executing [s at macro-trunkdial-failover-0.3:1]
GotoIf("SIP/1001-008646a0", "0?1-fmsetcid|1") in new stack
    -- Executing [s at macro-trunkdial-failover-0.3:2]
GotoIf("SIP/1001-008646a0", "0?1-setgbobname|1") in new stack
    -- Executing [s at macro-trunkdial-failover-0.3:3]
Set("SIP/1001-008646a0", "CALLERID(num)=1001") in new stack
    -- Executing [s at macro-trunkdial-failover-0.3:4]
GotoIf("SIP/1001-008646a0", "0?1-dial|1") in new stack
    -- Executing [s at macro-trunkdial-failover-0.3:5]
Set("SIP/1001-008646a0", "CALLERID(all)=") in new stack
    -- Executing [s at macro-trunkdial-failover-0.3:6]
Goto("SIP/1001-008646a0", "1-dial|1") in new stack
    -- Goto (macro-trunkdial-failover-0.3,1-dial,1)
    -- Executing [1-dial at macro-trunkdial-failover-0.3:1]
Dial("SIP/1001-008646a0", "DAHDI/trunk_1/9995") in new stack
[Apr 30 11:03:01] WARNING[7426]: chan_dahdi.c:8368 dahdi_request: Unable
to determine channel for data trunk_1/9995
[Apr 30 11:03:01] WARNING[7426]: app_dial.c:1237 dial_exec_full: Unable to
create channel of type 'DAHDI' (cause 0 - Unknown)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [1-dial at macro-trunkdial-failover-0.3:2]
GotoIf("SIP/1001-008646a0", "0 > 0 ?1-CHANUNAVAIL|1:1-out|1") in new stack
    -- Goto (macro-trunkdial-failover-0.3,1-out,1)
    -- Executing [1-out at macro-trunkdial-failover-0.3:1]
Hangup("SIP/1001-008646a0", "") in new stack
  == Spawn extension (macro-trunkdial-failover-0.3, 1-out, 1) exited
non-zero on 'SIP/1001-008646a0' in macro 'trunkdial-failover-0.3'
  == Spawn extension (DLPN_dp_first, 9995, 1) exited non-zero on
'SIP/1001-008646a0'
------------------------------------------------------------------------------------
i am using Asterisk 1.4.24.1 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-04-30 04:18 ivan77         Note Added: 0103993                          
======================================================================




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