BUG Anveo - Auto Failover

yumbaman

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I set a call limit on anveo to .005. I would like my localphone trunk to pickup the call and send it out. But once anveo sends me a 503 error, pbiaf does not send it to the next trunk. I have failover checked on the trunks and my outbound routes have several outgoing trunks. What am I doing wrong to get asterisk to send the call to the next trunk during a failed call message in the CDR? Also, after looking at the logs, when it fails, the next attempt sends it out the same anveo trunk. I would like it to move the call onto the next trunk.
 

yumbaman

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Code:
Called SIP/[email protected]
[2013-09-06 14:08:29] VERBOSE[1578][C-0000037e] chan_sip.c: -- Got SIP response 503 "Route not found" back from 50.22.101.14:5060
[2013-09-06 14:08:29] VERBOSE[1306][C-0000037e] app_dial.c: -- SIP/sbc.anveo.com-0000073d is circuit-busy
[2013-09-06 14:08:29] VERBOSE[1306][C-0000037e] app_dial.c: == Everyone is busy/congested at this time (1:0/1/0)
[2013-09-06 14:08:29] VERBOSE[1306][C-0000037e] pbx.c: -- Executing [s@macro-dialout-trunk:31] NoOp("SIP/711-0000073c", "Dial failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 34") in new stack
[2013-09-06 14:08:29] VERBOSE[1306][C-0000037e] pbx.c: -- Executing [s@macro-dialout-trunk:32] GotoIf("SIP/711-0000073c", "1?continue,1:s-CONGESTION,1") in new stack
[2013-09-06 14:08:29] VERBOSE[1306][C-0000037e] pbx.c: -- Goto (macro-dialout-trunk,continue,1)
[2013-09-06 14:08:29] VERBOSE[1306][C-0000037e] pbx.c: -- Executing [continue@macro-dialout-trunk:1] NoOp("SIP/711-0000073c", "TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 34 - failing through to other trunks") in new stack
[2013-09-06 14:08:29] VERBOSE[1306][C-0000037e] pbx.c: -- Executing [continue@macro-dialout-trunk:2] Set("SIP/711-0000073c", "CALLERID(number)=711") in new stack
[2013-09-06 14:08:29] VERBOSE[1306][C-0000037e] pbx.c: -- Auto fallthrough, channel 'SIP/711-0000073c' status is 'CONGESTION'

Then afterwards, it starts the same call process and trys to send to anveo again and fails the same way.
 

wardmundy

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Is the failover trunk you want to use in the same outbound route?? FreePBX won't jump from one outbound route to another one when a trunk fails.

yumbaman: Please don't double-post.
 

yumbaman

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Yea, sorry about the double post. I felt I selected the wrong forum for the first post and no way to delete. Yes, I have selected additional trunks in the same outbound route.
0QGOdJR.png
 

atsak

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What version of FPBX are you using? 2.10 had this issue, but 2.11 doesn't.
 

atsak

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Did you tick the "continue if busy" checkbox in the trunk configuration?
 

atsak

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Maybe it's because you're using a custom trunk. Try setting up a SIP trunk instead and see if that works. It's still a bug with the custom trunk of sorts if so, but interesting to see if it works.
 

yumbaman

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I would like to test this scenario, but not sure how to transfer the dial string into a regular sip trunk.
 

atsak

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I would like to test this scenario, but not sure how to transfer the dial string into a regular sip trunk.

Do you mean the prefix Anveo requests you put in? You can put that in the prepend field in freepbx.
 

yumbaman

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I was told on the freepbx irc chat that optional destination in outbound routes was added by a module. Could this be a module causing this problem. Also, I noticed that the second attempt is an attempt by the user and not freepbx or asterisk. And to followup on atsak's question, I know I can add a prefix, but what should the outgoing trunk settings be? I used custom route in following Ward's tutorial for anveo.
 

yumbaman

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I was looking at module admin, and I did install schoomze's extension routing module which I use. And at one time I was testing the trunk balance from possa. I am just trying to narrow down the problem.
 

atsak

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I was told on the freepbx irc chat that optional destination in outbound routes was added by a module. Could this be a module causing this problem. Also, I noticed that the second attempt is an attempt by the user and not freepbx or asterisk. And to followup on atsak's question, I know I can add a prefix, but what should the outgoing trunk settings be? I used custom route in following Ward's tutorial for anveo.


I use this in the outbound settings. So far works perfectly on about a dozen systems.

host=sbc.anveo.com
type=peer
canreinvite=yes
insecure=port,invite
trustrpid=yes
sendrpid=yes
context=from-trunk
qualify=yes
port=5060
 

yumbaman

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These trunk settings work great for outbound, but I still get a busy signal on the yealink t46G and the logs are all the same. Calls less than $.005 (my rate cap) all connect.
 

yumbaman

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Ok, so I finally made progress. Not sure if I have system setup in an incorrect manner, but I don't see why I would have a problem. For each extension, I send a CID num alias that is the same as my outgoing CID in the route. I did this for a number of reasons. One, when family members call by extension, the other persons did shows as the calling party. I match outbound routes based on callerid. After removing CID num alias in extensions, I now have to change outbound routes to match 3 digit extension. So, for whatever reason, once I received a busy signal from anveo, it did not see another way to move the call to the second trunk/carrier. This could be correct functioning of the pbx, but it doesn't make sense to me regardless.
 

yumbaman

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Making these changes seems to work for now. For extension 701, I cleared CID num alias and changed the outbound route to match the extension number. It seems to be working and rolling over to localphone when a rate cap is hit.
 

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