RECOMMENDATIONS Flowroute - no audio, but only on SOME calls!?!

MacNix

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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..

PBX in a Flash Version = 1.7.5.5 Running on *HARDWARE*
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?
 

MacNix

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A quick followup on this, with regards to Flowroute.. Got a *very* knowledgeable guy on the line there this am....

As soon as I DMZ'd my Piaf box, EVERYTHING worked for them.. Not exactly a long-term solution, but I traced back to a problem in porting - needed to open up the port width a bit more than normal to handle their unusual RTP process...

As was mentioned elsewhere, flowroute is not a 'normal' SIP company, in that they hand-off the call to secondary carriers whenever the call is connected. This reduces hop-count, theoretically usually resulting in higher signal quality, lower latency, blah blah blah...

The challenge is though, that apparently they tend to use a different pack of ports for most calls, which can result in general unhappiness on the firewall level. This is what I was experiencing with these calls. CERTAIN calls went thru most of the time (perhaps different ports were being utilized which happened to make it thru my firewall?), but other calls hit a wall....

They (flowroute) recommended opening ports 5k-30k, as some signaling happens lower with them.

After opening up from 5k-30k, ALMOST all my calls are going thru without issues from that box. So apparently it's a bit outside the normal port range (10k-20k).....
 

AndyInNYC

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Can you post/summarize exactly what the solution entailed (even post your -A INPUT command changes)?

This is a problem others have had and will likely have in the future. Assuming we avoid another site crash/loss, it would be good to have this info posted here for posterity.

Andrew
 

MacNix

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sure - it was pretty simple:

haha... as simple as anything in the complex world of VOIP could be.....

I use Untangle routers/firewalls for most locations.. As per Ward/others, I've never been willing to put a Piaf box out "open" to the world, as it's just WAY too problematic. I had a RentPBX up for a week without permaban on, and it damn near sunk the ship with the amount of attacks... So if you're trying to run your box without a decent firewall. Well, best of luck... :beatdeadhorse5:

For testing, I went into Untangle, and turned off the Firewall.. :eek: This immediately cleared up said issue on Flowroute..

Knowing the firewall was blocking things let me know it was simple a port issue. So I a) bypassed some port traffic, and b) opened port traffic directly to the server only.

In Untangle:
Network>Config>Advanced(right side)> Bypass Rules, and made entry for all SIP traffic (ports 5000-30000). There is also a "System Bypass Rules" to "Bypass VOIP Traffic (SIP)" an another for IAX2 - I had both on already (didn't help with this)

Firewall>Settings>Enable Rules, created rules passing 5k-30k ports (destined to my Piaf box IP#) thru...

After applying the Firewall rules (to pass all RTP port traffic destined ONLY to the Piaf box IP#), flowroute is happy as can be.


As to IPTable changes - I didn't add flowroute's trunk IP in - actually they didn't want to give me IPs for them, as they have a couple failovers... but since i'm handling all thru a firewall anyway, mute point...

to whitelist an IP, you would to:

....vi / etc/sysconfig/iptables...
...insert ("i")
then insert a line in the "-A INPUT" area:

-A INPUT -s xx.xx.xx.xx -j ACCEPT​
(with xx.xx... being your whitelisted IP server)

Then be sure to save:
hit "escape"​
then
:w (to write/save)​
and
:q (to quit/exit)​
or
if you're in a rush, ":wq!" does all three at once (and guarantees that your accidental fatfingered entry is stored forever)....​
and as always, restart IPTables to make it permanent....
"iptables-restart"


Hope this helps...

if i'm wrong on something, somebody let me know - will happily update

firewall.jpg
 

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