SOLVED Asterisk mistakes call from extension as anonymous

sirdotcom

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Hi,
I'm having an odd problem that only effects the latest Centos AND Ubuntu Incredible 13-13.10. With two different hardpones, I get this when trying to call the demo. But when I use a softphone, it works fine. I can't figure out what's making it turn around and think the call is from-external.

Code:
-- Executing [[email protected]:1] Set("SIP/701-00000000", "MyDomain=192.168.2.175") in new stack
    -- Executing [[email protected]:2] NoOp("SIP/701-00000000", "SIPDOMAIN: mydomain.net") in new stack
    -- Executing [[email protected]:3] GotoIf("SIP/701-00000000", "1?FoundDomain") in new stack
    -- Goto (from-internal,3366,6)
    -- Executing [[email protected]:6] GotoIf("SIP/701-00000000", "0?OutAllRoutes") in new stack
    -- Executing [[email protected]:7] Macro("SIP/701-00000000", "uridial,[email protected]") in new stack
    -- Executing [[email protected]:1] Set("SIP/701-00000000", "[email protected]") in new stack
    -- Executing [[email protected]:2] Set("SIP/701-00000000", "CALLERID(number)=") in new stack
    -- Executing [[email protected]:3] ExecIf("SIP/701-00000000", "1?Set(CALLERID(number)=Anonymous)") in new stack
    -- Executing [[email protected]:4] NoOp("SIP/701-00000000", "Called SIP URI: [email protected]") in new stack
    -- Executing [[email protected]:5] NoOp("SIP/701-00000000", "Calling From  : "701" <Anonymous> ?^?^?") in new stack
    -- Executing [[email protected]:6] Dial("SIP/701-00000000", "SIP/[email protected],60,tr") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
       > 0x7f2dec04f760 -- Strict RTP learning after remote address set to: 127.0.0.1:17256
    -- Called SIP/[email protected]
    -- Executing [[email protected]:1] NoOp("SIP/127.0.0.1-00000002", "Received incoming SIP connection from unknown peer to 3366") in new stack
    -- Executing [[email protected]:2] Set("SIP/127.0.0.1-00000002", "DID=3366") in new stack
    -- Executing [[email protected]:3] Goto("SIP/127.0.0.1-00000002", "s,1") in new stack
    -- Goto (from-sip-external,s,1)
    -- Executing [[email protected]:1] GotoIf("SIP/127.0.0.1-00000002", "1?setlanguage:checkanon") in new stack
    -- Goto (from-sip-external,s,2)
    -- Executing [[email protected]:2] Set("SIP/127.0.0.1-00000002", "CHANNEL(language)=en") in new stack
    -- Executing [[email protected]:3] GotoIf("SIP/127.0.0.1-00000002", "1?noanonymous") in new stack
    -- Goto (from-sip-external,s,5)
    -- Executing [[email protected]:5] Set("SIP/127.0.0.1-00000002", "TIMEOUT(absolute)=15") in new stack
    -- Channel will hangup at 2019-04-03 13:21:21.489 EDT.
    -- Executing [[email protected]:6] Log("SIP/127.0.0.1-00000002", "WARNING,"Rejecting unknown SIP connection from 127.0.0.1"") in new stack
[2019-04-03 13:21:06] WARNING[7514][C-00000001]: Ext. s:6 @ from-sip-external: "Rejecting unknown SIP connection from 127.0.0.1"
The phone registers normally. Has anyone had problems like this? This is the first time an Incredible PBX has done this,

Thanks,
Steve
 

tbrummell

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I don't know what you got going on, but my test call to *43 (Echo Test) does not look like that. I have no 'macro-uridial' entries. I notice your "MyDomain" and "SIPDOMAIN" entries are different than mine when I do a test call.
Code:
[2019-04-04 07:20:11] VERBOSE[27764][C-0000000e] pbx.c: Executing [*[email protected]:1] Set("SIP/212-0000001a", "MyDomain=") in new stack
[2019-04-04 07:20:11] VERBOSE[27764][C-0000000e] pbx.c: Executing [*[email protected]:2] NoOp("SIP/212-0000001a", "SIPDOMAIN: my.FQDN.domaint") in new stack
[2019-04-04 07:20:11] VERBOSE[27764][C-0000000e] pbx.c: Executing [*[email protected]:3] GotoIf("SIP/212-0000001a", "0?FoundDomain") in new stack
[2019-04-04 07:20:11] VERBOSE[27764][C-0000000e] pbx.c: Executing [*[email protected]:4] Set("SIP/212-0000001a", "MyDomain=") in new stack
[2019-04-04 07:20:11] VERBOSE[27764][C-0000000e] pbx.c: Executing [*[email protected]:5] Set("SIP/212-0000001a", "SIPDOMAIN=") in new stack
[2019-04-04 07:20:11] VERBOSE[27764][C-0000000e] pbx.c: Executing [*[email protected]:6] GotoIf("SIP/212-0000001a", "1?OutAllRoutes") in new stack
[2019-04-04 07:20:11] VERBOSE[27764][C-0000000e] pbx_builtins.c: Goto (from-internal,*43,8)
Are you dialing the call as a SIP URI?

Are you just using IPBX, or have you set up Kamilio and all of the other things Ward has posted in the past few weeks?
 

Eliad

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Hi,
I'm having an odd problem that only effects the latest Centos AND Ubuntu Incredible 13-13.10. With two different hardpones, I get this when trying to call the demo. But when I use a softphone, it works fine. I can't figure out what's making it turn around and think the call is from-external.

Code:
-- Executing [[email protected]:1] Set("SIP/701-00000000", "MyDomain=192.168.2.175") in new stack
    -- Executing [[email protected]:2] NoOp("SIP/701-00000000", "SIPDOMAIN: mydomain.net") in new stack
    -- Executing [[email protected]:3] GotoIf("SIP/701-00000000", "1?FoundDomain") in new stack
    -- Goto (from-internal,3366,6)
    -- Executing [[email protected]:6] GotoIf("SIP/701-00000000", "0?OutAllRoutes") in new stack
    -- Executing [[email protected]:7] Macro("SIP/701-00000000", "uridial,[email protected]") in new stack
    -- Executing [[email protected]:1] Set("SIP/701-00000000", "[email protected]") in new stack
    -- Executing [[email protected]:2] Set("SIP/701-00000000", "CALLERID(number)=") in new stack
    -- Executing [[email protected]:3] ExecIf("SIP/701-00000000", "1?Set(CALLERID(number)=Anonymous)") in new stack
    -- Executing [[email protected]:4] NoOp("SIP/701-00000000", "Called SIP URI: [email protected]") in new stack
    -- Executing [[email protected]:5] NoOp("SIP/701-00000000", "Calling From  : "701" <Anonymous> ?^?^?") in new stack
    -- Executing [[email protected]:6] Dial("SIP/701-00000000", "SIP/[email protected],60,tr") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
       > 0x7f2dec04f760 -- Strict RTP learning after remote address set to: 127.0.0.1:17256
    -- Called SIP/[email protected]
    -- Executing [[email protected]:1] NoOp("SIP/127.0.0.1-00000002", "Received incoming SIP connection from unknown peer to 3366") in new stack
    -- Executing [[email protected]:2] Set("SIP/127.0.0.1-00000002", "DID=3366") in new stack
    -- Executing [[email protected]:3] Goto("SIP/127.0.0.1-00000002", "s,1") in new stack
    -- Goto (from-sip-external,s,1)
    -- Executing [[email protected]:1] GotoIf("SIP/127.0.0.1-00000002", "1?setlanguage:checkanon") in new stack
    -- Goto (from-sip-external,s,2)
    -- Executing [[email protected]:2] Set("SIP/127.0.0.1-00000002", "CHANNEL(language)=en") in new stack
    -- Executing [[email protected]:3] GotoIf("SIP/127.0.0.1-00000002", "1?noanonymous") in new stack
    -- Goto (from-sip-external,s,5)
    -- Executing [[email protected]:5] Set("SIP/127.0.0.1-00000002", "TIMEOUT(absolute)=15") in new stack
    -- Channel will hangup at 2019-04-03 13:21:21.489 EDT.
    -- Executing [[email protected]:6] Log("SIP/127.0.0.1-00000002", "WARNING,"Rejecting unknown SIP connection from 127.0.0.1"") in new stack
[2019-04-03 13:21:06] WARNING[7514][C-00000001]: Ext. s:6 @ from-sip-external: "Rejecting unknown SIP connection from 127.0.0.1"
The phone registers normally. Has anyone had problems like this? This is the first time an Incredible PBX has done this,

Thanks,
Steve
I had a similar problem for a local server install. I solved by using an older Centos IncrediblePBX installation script. Is your install in the cloud or local? Apparently cloud install does not give problems according to another user following the thread No outgoing calls.
 

sirdotcom

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Hi,

I'm just using basic IPBX "out-of-the-box". And I'm just dialing 3-3-6-6 to access the Demo IVR. What's so odd is that it works normally with a softphone.

I don't know what you got going on, but my test call to *43 (Echo Test) does not look like that. I have no 'macro-uridial' entries. I notice your "MyDomain" and "SIPDOMAIN" entries are different than mine when I do a test call.
Code:
[2019-04-04 07:20:11] VERBOSE[27764][C-0000000e] pbx.c: Executing [*[email protected]:1] Set("SIP/212-0000001a", "MyDomain=") in new stack
[2019-04-04 07:20:11] VERBOSE[27764][C-0000000e] pbx.c: Executing [*[email protected]:2] NoOp("SIP/212-0000001a", "SIPDOMAIN: my.FQDN.domaint") in new stack
[2019-04-04 07:20:11] VERBOSE[27764][C-0000000e] pbx.c: Executing [*[email protected]:3] GotoIf("SIP/212-0000001a", "0?FoundDomain") in new stack
[2019-04-04 07:20:11] VERBOSE[27764][C-0000000e] pbx.c: Executing [*[email protected]:4] Set("SIP/212-0000001a", "MyDomain=") in new stack
[2019-04-04 07:20:11] VERBOSE[27764][C-0000000e] pbx.c: Executing [*[email protected]:5] Set("SIP/212-0000001a", "SIPDOMAIN=") in new stack
[2019-04-04 07:20:11] VERBOSE[27764][C-0000000e] pbx.c: Executing [*[email protected]:6] GotoIf("SIP/212-0000001a", "1?OutAllRoutes") in new stack
[2019-04-04 07:20:11] VERBOSE[27764][C-0000000e] pbx_builtins.c: Goto (from-internal,*43,8)
Are you dialing the call as a SIP URI?

Are you just using IPBX, or have you set up Kamilio and all of the other things Ward has posted in the past few weeks?
 

sirdotcom

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Hi,
I am using a cloud install. The odd thing is that this doesn't happen with softphones, only the two hardphones I've tried (which are all on the same home network, and seen by IPBX as an allowed IP.)

I had a similar problem for a local server install. I solved by using an older Centos IncrediblePBX installation script. Is your install in the cloud or local? Apparently cloud install does not give problems according to another user following the thread No outgoing calls.
 

stoneman68

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Same problem here. Softphones can call one another. Softphones can call my Polycom and Grandstream phones. But, Polycom & Grandstream phones can't call anyone, including *43 and 3366.

When the phones try to make a call to other devices on the LAN, the CLI shows:
Ext. s:6 @ from-sip-external: "Rejecting unknown SIP connection from 172.16.xxx.xxx"
(where 172.16.xxx.xxx is the PBX's LAN ip address)

As a noob, I haven't set up any trunks to the outside world yet.

Incredible PBX 13.0.195.28
 

Eliad

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Same problem here. Softphones can call one another. Softphones can call my Polycom and Grandstream phones. But, Polycom & Grandstream phones can't call anyone, including *43 and 3366.

When the phones try to make a call to other devices on the LAN, the CLI shows:
Ext. s:6 @ from-sip-external: "Rejecting unknown SIP connection from 172.16.xxx.xxx"
(where 172.16.xxx.xxx is the PBX's LAN ip address)

As a noob, I haven't set up any trunks to the outside world yet.

Incredible PBX 13.0.195.28
I tried Yealink T21 and same problem, cant make outgoing calls when using the latest IncrediblePBX install script. Yate softphone does dial out.
I am a true NOOB,, what linux command you used to get this error log?
 

wardmundy

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Yeah. The SIP URI mod is just causing too many issues for some folks. We're going to disable it by default moving forward. But here's how to disable it:
Code:
sed -i '\:// BEGIN SIP URI Mod1:,\:// END SIP URI Mod1:d' /etc/asterisk/extensions_custom.conf
asterisk -rx "dialplan reload"
 

stoneman68

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I tried Yealink T21 and same problem, cant make outgoing calls when using the latest IncrediblePBX install script. Yate softphone does dial out.
I am a true NOOB,, what linux command you used to get this error log?
Hello fellow noob. From the linux prompt, I use this to access the Asterisk CLI (command line interface).
asterisk -rvv
The -rvv is somewhat verbose. -rvvv would be even more verbose.

When I make calls, the CLI spits out a bunch of stuff, including the "Rejecting unknown SIP connection from 172.16.xxx.xxx"
 

stoneman68

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I tried Yealink T21 and same problem, cant make outgoing calls when using the latest IncrediblePBX install script. Yate softphone does dial out.
I am a true NOOB,, what linux command you used to get this error log?
Hello fellow noob. From the linux prompt, I use this to access the Asterisk CLI (command line interface).
asterisk -rvv
The -rvv is somewhat verbose. -rvvv would be even more verbose.

When I make calls, the CLI spits out a bunch of stuff, including the "Rejecting unknown SIP connection from 172.16.xxx.xxx"
Yeah. The SIP URI mod is just causing too many issues for some folks. We're going to disable it by default moving forward. But here's how to disable it:
Code:
sed -i '\:// BEGIN SIP URI Mod1:,\:// END SIP URI Mod1:d' /etc/asterisk/extensions_custom.conf
asterisk -rx "dialplan reload"
Fixed! That did it for me. Thanks Ward and Dr. K!
 

jreming

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Yeah. The SIP URI mod is just causing too many issues for some folks. We're going to disable it by default moving forward. But here's how to disable it:
Code:
sed -i '\:// BEGIN SIP URI Mod1:,\:// END SIP URI Mod1:d' /etc/asterisk/extensions_custom.conf
asterisk -rx "dialplan reload"

Glad I wasn't going crazy, noticed the only time I could dial from inside was when I set the outbound proxy to my servers external fqdn/ip.

Now I just need to figure out why yate stopping initiating the MoH/On Hold

Thanks
 

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