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- Jun 21, 2011
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So I've been using Flowroute in certain trunks, to save some money..
Some calls (like, to my cell phone, and certain other cell phones) works GREAT! Calls to other PBXs have no audio (neither direction)!
WTF am I doing to cause that? I know the trunk is ok - have definitively checked it and confirmed it's working. But no audio, only on certain calls? Color me confused..
if it's any help, here's a spitout of CLI
a call to my cell, which works perfectly well:
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called flowrouteb/1XXXXXXXXXX
== Begin MixMonitor Recording SIP/715-000000c0
-- SIP/flowrouteb-000000c1 is making progress passing it to SIP/715-000000c0
-- SIP/flowrouteb-000000c1 is ringing
-- SIP/flowrouteb-000000c1 is making progress passing it to SIP/715-000000c0
-- SIP/flowrouteb-000000c1 answered SIP/715-000000c0
> doing dnsmgr_lookup for 'sip.flowroute.com'
> doing dnsmgr_lookup for 'sip.didlogic.com'
-- Executing [h@macro-dialout-trunk:1] Macro("SIP/715-000000c0", "hangupcall,") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/715-000000c0", "1?theend") in new stack
-- Goto (macro-hangupcall,s,3)
-- Executing [s@macro-hangupcall:3] Hangup("SIP/715-000000c0", "") in new stack
== Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'SIP/715-000000c0' in macro 'hangupcall'
== Spawn extension (macro-dialout-trunk, s, 20) exited non-zero on 'SIP/715-000000c0' in macro 'dialout-trunk'
== Spawn extension (from-internal, XXXXXXX, 6) exited non-zero on 'SIP/715-000000c0'
== MixMonitor close filestream
== End MixMonitor Recording SIP/715-000000c0
Here's a call to another PBX box 6 states away... no audio...
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called flowrouteb/1XXXXXXXXXX
== Begin MixMonitor Recording SIP/715-000000bc
-- SIP/flowrouteb-000000bd is ringing
-- SIP/flowrouteb-000000bd is making progress passing it to SIP/715-000000bc
-- SIP/flowrouteb-000000bd answered SIP/715-000000bc
-- Executing [h@macro-dialout-trunk:1] Macro("SIP/715-000000bc", "hangupcall,") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/715-000000bc", "1?theend") in new stack
-- Goto (macro-hangupcall,s,3)
-- Executing [s@macro-hangupcall:3] Hangup("SIP/715-000000bc", "") in new stack
== Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'SIP/715-000000bc' in macro 'hangupcall'
== Spawn extension (macro-dialout-trunk, s, 20) exited non-zero on 'SIP/715-000000bc' in macro 'dialout-trunk'
== Spawn extension (from-internal, XXXXXXX, 6) exited non-zero on 'SIP/715-000000bc'
== MixMonitor close filestream
== End MixMonitor Recording SIP/715-000000bc
As to trunk configs:
basically I copy/pasted their entire FreePBX pre-baked strings into my fields:
type=friend
secret=xxxxxxxxxxxx
username=xxxxxxxxx
host=sip.flowroute.com
dtmfmode=rfc2833
context=from-trunk
canreinvite=no
allow=ulaw&g729
insecure=port,invite
fromdomain=sip.flowroute.com
Reg string: USERNAME:[email protected]
So why would SOME calls work perfectly (namely to SOME but not *all* cell phones), while other calls (local bank, several distant Piaf boxes, several other boxes) NOT connect?
Some calls (like, to my cell phone, and certain other cell phones) works GREAT! Calls to other PBXs have no audio (neither direction)!
WTF am I doing to cause that? I know the trunk is ok - have definitively checked it and confirmed it's working. But no audio, only on certain calls? Color me confused..
PBX in a Flash Version = 1.7.5.5 Running on *HARDWARE*
FreePBX Version = 2.9.0.14
FreePBX Version = 2.9.0.14
Running Asterisk Version = 1.8.4.1
Asterisk Source Version = 1.8.4.1
Dahdi Source Version = 2.4.1.2+2.4.1
Libpri Source Version = 1.4.11.5
IP Address = 10.x.x.x on eth0
Operating System = CentOS release 5.6 (Final)
Kernel Version = 2.6.18-194.26.1.el5 - 32 Bit
if it's any help, here's a spitout of CLI
a call to my cell, which works perfectly well:
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called flowrouteb/1XXXXXXXXXX
== Begin MixMonitor Recording SIP/715-000000c0
-- SIP/flowrouteb-000000c1 is making progress passing it to SIP/715-000000c0
-- SIP/flowrouteb-000000c1 is ringing
-- SIP/flowrouteb-000000c1 is making progress passing it to SIP/715-000000c0
-- SIP/flowrouteb-000000c1 answered SIP/715-000000c0
> doing dnsmgr_lookup for 'sip.flowroute.com'
> doing dnsmgr_lookup for 'sip.didlogic.com'
-- Executing [h@macro-dialout-trunk:1] Macro("SIP/715-000000c0", "hangupcall,") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/715-000000c0", "1?theend") in new stack
-- Goto (macro-hangupcall,s,3)
-- Executing [s@macro-hangupcall:3] Hangup("SIP/715-000000c0", "") in new stack
== Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'SIP/715-000000c0' in macro 'hangupcall'
== Spawn extension (macro-dialout-trunk, s, 20) exited non-zero on 'SIP/715-000000c0' in macro 'dialout-trunk'
== Spawn extension (from-internal, XXXXXXX, 6) exited non-zero on 'SIP/715-000000c0'
== MixMonitor close filestream
== End MixMonitor Recording SIP/715-000000c0
Here's a call to another PBX box 6 states away... no audio...
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called flowrouteb/1XXXXXXXXXX
== Begin MixMonitor Recording SIP/715-000000bc
-- SIP/flowrouteb-000000bd is ringing
-- SIP/flowrouteb-000000bd is making progress passing it to SIP/715-000000bc
-- SIP/flowrouteb-000000bd answered SIP/715-000000bc
-- Executing [h@macro-dialout-trunk:1] Macro("SIP/715-000000bc", "hangupcall,") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/715-000000bc", "1?theend") in new stack
-- Goto (macro-hangupcall,s,3)
-- Executing [s@macro-hangupcall:3] Hangup("SIP/715-000000bc", "") in new stack
== Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'SIP/715-000000bc' in macro 'hangupcall'
== Spawn extension (macro-dialout-trunk, s, 20) exited non-zero on 'SIP/715-000000bc' in macro 'dialout-trunk'
== Spawn extension (from-internal, XXXXXXX, 6) exited non-zero on 'SIP/715-000000bc'
== MixMonitor close filestream
== End MixMonitor Recording SIP/715-000000bc
As to trunk configs:
basically I copy/pasted their entire FreePBX pre-baked strings into my fields:
type=friend
secret=xxxxxxxxxxxx
username=xxxxxxxxx
host=sip.flowroute.com
dtmfmode=rfc2833
context=from-trunk
canreinvite=no
allow=ulaw&g729
insecure=port,invite
fromdomain=sip.flowroute.com
Reg string: USERNAME:[email protected]
So why would SOME calls work perfectly (namely to SOME but not *all* cell phones), while other calls (local bank, several distant Piaf boxes, several other boxes) NOT connect?