TIPS CallCentric Setup for FreePBX

Flash9999

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Hello,

I've setup a fresh install of Piaf Green and although everything went smooth after configuring a callcentric trunk since i can't no longer use sipgate the incoming calls are rejected and i get the "the number you have dialed is not in service" message. I've included a screenshot of the log file because i am hesitant to allow anonymous sip calls. Aren't callcentric servers automaically added to the whilelist after setting up incrediblepbx11? Will I have to allow anonymous sip calls?

Please help.
Semper fi
 

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rossiv

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Flash9999 Instead of a screenshot, would you mind posting a full log from beginning to end of the call here, wrapped inside "[ code ]" tags with private information sanitized? It would help a lot.
My initial guess is that something is mis-configured with your Sipgate trunk since Asterisk thinks that the call isn't coming in on the Sipgate trunk.
 

Flash9999

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rossiv thanks for replying the actual trunk is Callcentric since sipgate is out of business atleast here in the US. I'll post a full log I'm a few mins

Thank you
 

rossiv

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Oh, I misread your first post then. In that case, I think in the case of outbound, still check Callcentric's whitelist. And post an log of an attempted outbound call as well please.

As for inbound, there's another thread here that deals with a very similar issue if not the exact same. I'll dig it up.
 

bobkoure

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Something of a newbie here. I'd been using callcentric before I set up an asterisk PBX. I couldn't see any way to make callcentric trunks work without allowing anon access. As I wasn't that familiar with the SIP, I wasn't sure how much of a security problem anon access would have been - so I switched to vitelity (Ward's got a deal). This was some months ago, and callcentric may have changed, but you also may be banging into an "it works that way" wall...
 

Flash9999

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hello Gents,

I've previously had problems getting callcentric calls routed through piaf, this thread was actually started by myself sometime ago describing the problem. Although it is marked solved I am now trying to get this working on a fresh Ubuntu install with no avail. Callcentric calls will not come through unless I enable "Allow Anonymous Inbound SIP Calls" which is something I do not want to do. There has been various tweaks that users here have tried but I want to know what is the right way to get this working properly. I've included the changes I've made to my extentions_custom.conf file along with my trunk settings for review.

trunk settings:

Trunk Name: callcentric

Peer Details:
type=friend
username=1777XXXXXXX
qualify=yes
context=from-trunk
host=callcentric.com
defaultuser=1777XXXXXXX
secret=ThereWasAPasswordHere
fromuser=1777XXXXXXX
fromdomain=callcentric.com
insecure=port,invite
dtmfmode=rfc2833
disallow=all
allow=ulaw
sendrpid=yes
trustrpid=no

No Incoming Settings

Register String: 1777XXXXXXX:[email protected]/1777XXXXXXX

extensions_custom.conf

[from-sip-external-custom]
exten => _.,1,NoOp(Received incoming SIP connection from unknown peer to ${EXTEN})
exten => _.,n,Set(DID=${IF($["${EXTEN:1:2}"=""]?s:${EXTEN})})
exten => _.,n,Goto(s,1)

; CallCentric Check
exten => s,1,GotoIf($["${DID}"="1777number"]?callcentric)

; Regular Check
exten => s,n,GotoIf($["${ALLOW_SIP_ANON}"="yes"]?checklang:noanonymous)

; CallCentric DID Code
exten => s,n(callcentric),Set(Var_FROM_DOMAIN=${CUT(CUT(SIP_HEADER(TO),@,2),>,1)})
exten => s,n,GotoIF($["${Var_FROM_DOMAIN}" = "callcentric.com"]?callcentric-next)
exten => s,n,GotoIF($["${Var_FROM_DOMAIN}" = "ss.callcentric.com"]?callcentric-next)
exten => s,n,GotoIF($["${Var_FROM_DOMAIN}" = "66.193.176.35"]?callcentric-next:checklang)
exten => s,n(callcentric-next),Set(Var_TO_DID=${CUT(CUT(SIP_HEADER(TO),@,1),:,2)})
exten => s,n,GotoIF($["${Var_TO_DID}" = ""]?checklang)
exten => s,n,Set(DID=${Var_TO_DID})

; Regular script continues
exten => s,n(checklang),GotoIf($["${SIPLANG}"!=""]?setlanguage:from-trunk,${DID},1)
exten => s,n(setlanguage),Set(CHANNEL(language)=${SIPLANG})
exten => s,n,Goto(from-trunk,${DID},1)
exten => s,n(noanonymous),Set(TIMEOUT(absolute)=15)
exten => s,n,Log(WARNING,"Rejecting unknown SIP connection from ${CHANNEL(recvip)}")
exten => s,n,Answer
exten => s,n,Wait(2)
exten => s,n,Playback(ss-noservice)
exten => s,n,Playtones(congestion)
exten => s,n,Congestion(5)
exten => h,1,Hangup
exten => i,1,Hangup
exten => t,1,Hangup
LogFile:

[2014-08-11 19:06:01] VERBOSE[7111][C-00000001] pbx.c: -- Executing [s@from-sip-external:1] GotoIf("SIP/66.193.176.35-00000001", "0?checklang:noanonymous") in new stack
[2014-08-11 19:06:01] VERBOSE[7111][C-00000001] pbx.c: -- Goto (from-sip-external,s,5)
[2014-08-11 19:06:01] VERBOSE[7111][C-00000001] pbx.c: -- Executing [s@from-sip-external:5] Set("SIP/66.193.176.35-00000001", "TIMEOUT(absolute)=15") in new stack
[2014-08-11 19:06:01] VERBOSE[7111][C-00000001] func_timeout.c: -- Channel will hangup at 2014-08-11 19:06:16.817 EDT.
[2014-08-11 19:06:01] VERBOSE[7111][C-00000001] pbx.c: -- Executing [s@from-sip-external:6] Log("SIP/66.193.176.35-00000001", "WARNING,"Rejecting unknown SIP connection from 204.11.192.163"") in new stack
[2014-08-11 19:06:01] WARNING[7111][C-00000001] Ext. s: "Rejecting unknown SIP connection from 204.11.192.163"
[2014-08-11 19:06:01] VERBOSE[7111][C-00000001] pbx.c: -- Executing [s@from-sip-external:7] Answer("SIP/66.193.176.35-00000001", "") in new stack
[2014-08-11 19:06:02] VERBOSE[7111][C-00000001] pbx.c: -- Executing [s@from-sip-external:8] Wait("SIP/66.193.176.35-00000001", "2") in new stack
[2014-08-11 19:06:04] VERBOSE[7111][C-00000001] pbx.c: -- Executing [s@from-sip-external:9] Playback("SIP/66.193.176.35-00000001", "ss-noservice") in new stack
[2014-08-11 19:06:04] VERBOSE[7111][C-00000001] file.c: -- <SIP/66.193.176.35-00000001> Playing 'ss-noservice.gsm' (language 'en')
[2014-08-11 19:06:09] VERBOSE[7111][C-00000001] pbx.c: -- Executing [s@from-sip-external:10] PlayTones("SIP/66.193.176.35-00000001", "congestion") in new stack
[2014-08-11 19:06:09] VERBOSE[7111][C-00000001] pbx.c: -- Executing [s@from-sip-external:11] Congestion("SIP/66.193.176.35-00000001", "5") in new stack
[2014-08-11 19:06:14] VERBOSE[7111][C-00000001] pbx.c: == Spawn extension (from-sip-external, s, 11) exited non-zero on 'SIP/66.193.176.35-00000001'
[2014-08-11 19:06:14] VERBOSE[7111][C-00000001] pbx.c: -- Executing [h@from-sip-external:1] Hangup("SIP/66.193.176.35-00000001", "") in new stack
[2014-08-11 19:06:14] VERBOSE[7111][C-00000001] pbx.c: == Spawn extension (from-sip-external, h, 1) exited non-zero on 'SIP/66.193.176.35-00000001'

wardmundy
 

john p

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I have seen this problem when the underlying OS IP name lookup is failing. As a result, the OS passes the IP address of the incoming SIP call (204.11.192.163) rather than the name and the PBX is expecting a registered name (callcentric.com) so it considers this anonymous. I'd check if the machine can ping the name & properly resolve the IP. Hope this helps.
 

wardmundy

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CallCentric has "a different way" of doing things than virtually every other SIP provider on the planet. Remember the story of the mother going to watch her son march in the parade? As her son passes the reviewing stand, she turns to her friend and says, "Look. My little boy's the only one in step." Meet CallCentric!

IF you are using the latest Ubuntu build of Incredible PBX, then the IPtables firewall is configured to only allow access from WhiteListed IP addresses so there is minimal risk in allowing anonymous calls. You still will have to make sure ALL of the CallCentric IP addresses are added to your WhiteList using /root/add-ip: 204.11.192.160 to 204.11.192.171.

Here's another way of doing it (still a Pain in the Ass!) that doesn't require setting Allow Anonymous Calls to Yes: http://community.freepbx.org/t/callcentric-setup-guide/17500/24
 

Flash9999

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CallCentric has "a different way" of doing things than virtually every other SIP provider on the planet. Remember the story of the mother going to watch her son march in the parade? As her son passes the reviewing stand, she turns to her friend and says, "Look. My little boy's the only one in step." Meet CallCentric!

IF you are using the latest Ubuntu build of Incredible PBX, then the IPtables firewall is configured to only allow access from WhiteListed IP addresses so there is minimal risk in allowing anonymous calls. You still will have to make sure ALL of the CallCentric IP addresses are added to your WhiteList using /root/add-ip: 204.11.192.160 to 204.11.192.171.

Here's another way of doing it (still a Pain in the Ass!) that doesn't require setting Allow Anonymous Calls to Yes: http://community.freepbx.org/t/callcentric-setup-guide/17500/24


wardmundy

This method worked as far as getting calls in (the phone actually rings now) but once I answer the calls on my piaf systems it keeps ringing on the caller side (my cellphone). I've read somewhere in a post to add "session-timers=refuse" but that did not help at all. So as we speak i can get calls to sorta come through and ring the piaf extension but cant answer them.
 

wardmundy

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HINT: Vitelity is the corporate sponsor for the PIAF and Nerd Vittles projects... and you won't ever have to wrestle with this again. :idea:
 

Jay Deal

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A suggested way to overcome this without worrying about any settings in Asterisk, just get a hold of an Obi100, Obi110 or Obi202 and set it up one of the SP's as an extension on PIAF and register one of the other SP's to Callcentric. You can redirect all the incoming CC calls to PIAF and set up a dial plan that handles outgoing. This will work seamlessly. That way the connection to CC is completely separated from Asterisk and you have no worries about any settings other than on the Obi.
 

Mango

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... she turns to her friend and says, "Look. My little boy's the only one in step." Meet CallCentric!

I feel it prudent to point out that you could apply the same analogy to PIAF.

My Asterisk server continues to operate flawlessly with my Callcentric account.
 

wardmundy

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Mango: Wasn't suggesting that CallCentric couldn't be made to work. As for the little boy in the parade, your observation is correct. PIAF and Incredible PBX are the only available aggregations that build Asterisk and FreePBX from source during the installation procedure. The advantage to the end-user or developer is that you can modify the source to meet your own requirements either as part of the installation process or afterwards. Recompiling only takes a few minutes and the source is always on your server unlike "the competition."

Now clue us in on how the CallCentric design improves anything other than being an enormous hassle for the customer to configure.
 

Mango

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Their multiple SBCs allow for load balancing and failover.
 

wardmundy

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Lots of providers include load balancing and failover using a simple SIP registration with a single IP address.
 

Mango

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How is this done? Let's say I have multiple SIP switches and a single IP address, what do I do to load balance my clients between them all?
 

wardmundy

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Our focus and my comments pertain to ease of use for PBX in a Flash and Incredible PBX deployments, not soft switches.
 

Mango

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In other words, my question is: how, technically, do you think Callcentric should implement their multiple server setup such that it would be easier for you?
 

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