Incoming calls from Bandwidth.org provider will not answer

rbevill7 · September 30, 2009, 5:10am

I have trunks from Gizmo5, Voipvoip.com, and Bandwidth.org. Bandwidth is newest addition, because I like their multiple DID with a single trunk package. I wish the other guys would offer the same thing.

The inbound/outbound calls over the Gizmo5 and Voipvoip.com trunks work just fine. So, I don’t believe that Comcast is blocking any of the traffic.

For the Bandwidth.org trunk, outgoing calls work find. However, on the inbound calls, the originator continues to hear ringing, even though the log indicates that the call has been answered. Any ideas?

Thanks,
Bob

[[email protected] asterisk]# asterisk -rv
Asterisk 1.4.26.2, Copyright © 1999 - 2008 Digium, Inc. and others.
Created by Mark Spencer [email protected]
Asterisk comes with ABSOLUTELY NO WARRANTY; type ‘core show warranty’ for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type ‘core show license’ for details.

Connected to Asterisk 1.4.26.2 currently running on localhost (pid = 9454)
Verbosity is at least 3
– Executing [[email protected]:1] Set(“SIP/freepbx-rocks-01c79bc0”, “__FROM_DID=6037220990”) in new stack
– Executing [[email protected]:2] ExecIf(“SIP/freepbx-rocks-01c79bc0”, “0 |Set|CALLERID(name)=6034299779”) in new stack
– Executing [[email protected]:3] Set(“SIP/freepbx-rocks-01c79bc0”, “__CALLINGPRES_SV=allowed_not_screened”) in new stack
– Executing [[email protected]:4] SetCallerPres(“SIP/freepbx-rocks-01c79bc0”, “allowed_not_screened”) in new stack
– Executing [[email protected]:5] Goto(“SIP/freepbx-rocks-01c79bc0”, “from-did-direct|1201|1”) in new stack
– Goto (from-did-direct,1201,1)
– Executing [[email protected]:1] Macro(“SIP/freepbx-rocks-01c79bc0”, “exten-vm|1201|1201”) in new stack
– Executing [[email protected]:1] Macro(“SIP/freepbx-rocks-01c79bc0”, “user-callerid”) in new stack
– Executing [[email protected]:1] Set(“SIP/freepbx-rocks-01c79bc0”, “AMPUSER=6034299779”) in new stack
– Executing [[email protected]:2] GotoIf(“SIP/freepbx-rocks-01c79bc0”, “0?report”) in new stack
– Executing [[email protected]:3] ExecIf(“SIP/freepbx-rocks-01c79bc0”, “1|Set|REALCALLERIDNUM=6034299779”) in new stack
– Executing [[email protected]:4] Set(“SIP/freepbx-rocks-01c79bc0”, “AMPUSER=”) in new stack
– Executing [[email protected]:5] Set(“SIP/freepbx-rocks-01c79bc0”, “AMPUSERCIDNAME=”) in new stack
– Executing [[email protected]:6] GotoIf(“SIP/freepbx-rocks-01c79bc0”, “1?report”) in new stack
– Goto (macro-user-callerid,s,9)
– Executing [[email protected]cro-user-callerid:9] GotoIf(“SIP/freepbx-rocks-01c79bc0”, “0?continue”) in new stack
– Executing [[email protected]:10] Set(“SIP/freepbx-rocks-01c79bc0”, “__TTL=64”) in new stack
– Executing [[email protected]:11] GotoIf(“SIP/freepbx-rocks-01c79bc0”, “1?continue”) in new stack
– Goto (macro-user-callerid,s,18)
– Executing [[email protected]:18] NoOp(“SIP/freepbx-rocks-01c79bc0”, “Using CallerID “+16034299779” <6034299779>”) in new stack
– Executing [[email protected]:2] Set(“SIP/freepbx-rocks-01c79bc0”, “RingGroupMethod=none”) in new stack
– Executing [[email protected]:3] Set(“SIP/freepbx-rocks-01c79bc0”, “VMBOX=1201”) in new stack
– Executing [[email protected]:4] Set(“SIP/freepbx-rocks-01c79bc0”, “EXTTOCALL=1201”) in new stack
– Executing [[email protected]:5] Set(“SIP/freepbx-rocks-01c79bc0”, “CFUEXT=”) in new stack
– Executing [[email protected]:6] Set(“SIP/freepbx-rocks-01c79bc0”, “CFBEXT=”) in new stack
– Executing [[email protected]:7] Set(“SIP/freepbx-rocks-01c79bc0”, “RT=15”) in new stack
– Executing [[email protected]:8] Macro(“SIP/freepbx-rocks-01c79bc0”, “record-enable|1201|IN”) in new stack
– Executing [[email protected]:1] GotoIf(“SIP/freepbx-rocks-01c79bc0”, “1?check”) in new stack
– Goto (macro-record-enable,s,4)
– Executing [[email protected]:4] AGI(“SIP/freepbx-rocks-01c79bc0”, “recordingcheck|20090930-010437|1254287077.0”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20090930-010437|1254287077.0: Inbound recording not enabled
– AGI Script recordingcheck completed, returning 0
– Executing [[email protected]:5] MacroExit(“SIP/freepbx-rocks-01c79bc0”, “”) in new stack
– Executing [[email protected]:9] Macro(“SIP/freepbx-rocks-01c79bc0”, “dial|15|tr|1201”) in new stack
– Executing [[email protected]:1] GotoIf(“SIP/freepbx-rocks-01c79bc0”, “1?dial”) in new stack
– Goto (macro-dial,s,3)
– Executing [[email protected]:3] AGI(“SIP/freepbx-rocks-01c79bc0”, “dialparties.agi”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
dialparties.agi: Starting New Dialparties.agi
== Parsing ‘/etc/asterisk/manager.conf’: Found
== Parsing ‘/etc/asterisk/manager_additional.conf’: Found
== Parsing ‘/etc/asterisk/manager_custom.conf’: Found
== Manager ‘admin’ logged on from 127.0.0.1
dialparties.agi: Caller ID name is ‘+16034299779’ number is '6034299779’
dialparties.agi: Methodology of ring is ‘none’
– dialparties.agi: Added extension 1201 to extension map
– dialparties.agi: Extension 1201 cf is disabled
– dialparties.agi: Extension 1201 do not disturb is disabled
dialparties.agi: ExtensionState: 0
– dialparties.agi: dbset CALLTRACE/1201 to 6034299779
– dialparties.agi: Filtered ARG3: 1201
== Manager ‘admin’ logged off from 127.0.0.1
– AGI Script dialparties.agi completed, returning 0
– Executing [[email protected]:7] Dial(“SIP/freepbx-rocks-01c79bc0”, “SIP/1201|15|tr”) in new stack
– Called 1201
– SIP/1201-01c7fe60 is ringing

-- SIP/1201-01c7fe60 answered SIP/freepbx-rocks-01c79bc0


-- Executing [[email protected]:1] Macro("SIP/freepbx-rocks-01c79bc0", "hangupcall") in new stack
-- Executing [[email protected]:1] GotoIf("SIP/freepbx-rocks-01c79bc0", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,4)
-- Executing [[email protected]:4] GotoIf("SIP/freepbx-rocks-01c79bc0", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,7)
-- Executing [[email protected]:7] GotoIf("SIP/freepbx-rocks-01c79bc0", "1?theend") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [[email protected]:9] Hangup("SIP/freepbx-rocks-01c79bc0", "") in new stack

== Spawn extension (macro-hangupcall, s, 9) exited non-zero on ‘SIP/freepbx-rocks-01c79bc0’ in macro ‘hangupcall’
== Spawn h extension (macro-dial, h, 1) exited non-zero on ‘SIP/freepbx-rocks-01c79bc0’
== Spawn extension (macro-dial, s, 7) exited non-zero on ‘SIP/freepbx-rocks-01c79bc0’ in macro ‘dial’
== Spawn extension (macro-exten-vm, s, 9) exited non-zero on ‘SIP/freepbx-rocks-01c79bc0’ in macro ‘exten-vm’
== Spawn extension (from-did-direct, 1201, 1) exited non-zero on 'SIP/freepbx-rocks-01c79bc0’
localhost*CLI> exit
Executing last minute cleanups
[[email protected] asterisk]#

colinwright · November 2, 2009, 6:09pm

Bandwidth has been a very solid (but expensive) provider. 2 simple things I would check 1st

make sure you have a catch all inbound route. Create this by leaving the DID Number and Caller ID Number fields blank in the Inbound Route setup.
Bandwidth.com passes the number in the format +1NXXXXXXXXX. So make sure the DID in you inbound route starts with +1. For example one of my inbound DIDs is +1949271**** (numbers hidden).

system · June 4, 2014, 6:54pm