Can't receive phone call on my extension

ledude · September 12, 2013, 9:17pm

Hi all.

I’m running out of hair to pull here. I have my freepbx installed and configured ok at the beginning. I was testing it with Google Voice account. I can call out and receive call. I’m sure I must did something when I was trying to configure a new DID from Voip.ms. Suddenly my sip phone can’t dial out. Tried to call extension to extension and having the same problem also. Please help?

My Phone (Info Scrubbed): 2065551212
Destination Dial: 2065551234
FreePbx 2.11.0.10
Asterisk 11
OS: Centos 6.4 64Bit


Here is my outbound setting.
Route Name: mygvoicegmail.com

dial patterns:

prepend + prefix | match pattern /callerid

empty + empty | 1NXXNXXXXXX /2065551212

1 + empty | NXXNXXXXXX /2065551212

Trunk Sequence:

GVM_2065551212

Optional Destination on Congestion: Normal Congestion

Here is the log on what’s happening when I tried to call out.


[2013-09-12 14:04:43] VERBOSE[10674][C-00000006] netsock2.c: == Using SIP RTP CoS mark 5

[2013-09-12 14:04:43] VERBOSE[10880][C-00000006] pbx.c: – Executing [[email protected]:1] Macro(“SIP/8881-00000006”, “user-callerid,LIMIT,”) in new stack

[2013-09-12 14:04:43] VERBOSE[10880][C-00000006] pbx.c: – Executing [[email protected]:1] Set(“SIP/8881-00000006”, “TOUCH_MONITOR=1379019883.6”) in new stack

[2013-09-12 14:04:43] VERBOSE[10880][C-00000006] pbx.c: – Executing [[email protected]:2] Set(“SIP/8881-00000006”, “AMPUSER=8881”) in new stack

[2013-09-12 14:04:43] VERBOSE[10880][C-00000006] pbx.c: – Executing [[email protected]:3] GotoIf(“SIP/8881-00000006”, “0?report”) in new stack

[2013-09-12 14:04:43] VERBOSE[10880][C-00000006] pbx.c: – Executing [[email protected]:4] ExecIf(“SIP/8881-00000006”, “1?Set(REALCALLERIDNUM=8881)”) in new stack

[2013-09-12 14:04:43] VERBOSE[10880][C-00000006] pbx.c: – Executing [[email protected]:5] Set(“SIP/8881-00000006”, “AMPUSER=8881”) in new stack

[2013-09-12 14:04:43] VERBOSE[10880][C-00000006] pbx.c: – Executing [[email protected]:6] Set(“SIP/8881-00000006”, “AMPUSERCIDNAME=MyHomeExt”) in new stack

[2013-09-12 14:04:43] VERBOSE[10880][C-00000006] pbx.c: – Executing [[email protected]:7] GotoIf(“SIP/8881-00000006”, “0?report”) in new stack

[2013-09-12 14:04:43] VERBOSE[10880][C-00000006] pbx.c: – Executing [[email protected]:8] Set(“SIP/8881-00000006”, “AMPUSERCID=8881”) in new stack

[2013-09-12 14:04:43] VERBOSE[10880][C-00000006] pbx.c: – Executing [[email protected]:9] Set(“SIP/8881-00000006”, “__DIAL_OPTIONS=Ttr”) in new stack

[2013-09-12 14:04:43] VERBOSE[10880][C-00000006] pbx.c: – Executing [[email protected]:10] Set(“SIP/8881-00000006”, "CALLERID(all)=“MyHomeExt” ") in new stack

[2013-09-12 14:04:43] VERBOSE[10880][C-00000006] pbx.c: – Executing [[email protected]:11] GotoIf(“SIP/8881-00000006”, “0?limit”) in new stack

[2013-09-12 14:04:43] VERBOSE[10880][C-00000006] pbx.c: – Executing [[email protected]:12] ExecIf(“SIP/8881-00000006”, “1?Set(GROUP(concurrency_limit)=8881)”) in new stack

[2013-09-12 14:04:43] VERBOSE[10880][C-00000006] pbx.c: – Executing [[email protected]:13] ExecIf(“SIP/8881-00000006”, “0?Set(CHANNEL(language)=)”) in new stack

[2013-09-12 14:04:43] VERBOSE[10880][C-00000006] pbx.c: – Executing [[email protected]:14] GosubIf(“SIP/8881-00000006”, “7?sub-ccss,s,1(from-internal,12065551234)”) in new stack

[2013-09-12 14:04:43] VERBOSE[10880][C-00000006] pbx.c: – Executing [[email protected]:1] ExecIf(“SIP/8881-00000006”, “0?Return()”) in new stack

[2013-09-12 14:04:43] VERBOSE[10880][C-00000006] pbx.c: – Executing [[email protected]:2] Set(“SIP/8881-00000006”, “CCSS_SETUP=TRUE”) in new stack

[2013-09-12 14:04:43] VERBOSE[10880][C-00000006] pbx.c: – Executing [[email protected]:3] GosubIf(“SIP/8881-00000006”, “0?monitor_config,1(from-internal,12065551234):monitor_default,1(from-internal,12065551234)”) in new stack

[2013-09-12 14:04:43] VERBOSE[10880][C-00000006] pbx.c: – Executing [[email protected]:1] GotoIf(“SIP/8881-00000006”, “0?is_exten”) in new stack

[2013-09-12 14:04:43] VERBOSE[10880][C-00000006] pbx.c: – Executing [[email protected]:2] StackPop(“SIP/8881-00000006”, “”) in new stack

[2013-09-12 14:04:43] VERBOSE[10880][C-00000006] pbx.c: – Executing [[email protected]:3] Return(“SIP/8881-00000006”, “FALSE”) in new stack

[2013-09-12 14:04:43] VERBOSE[10880][C-00000006] pbx.c: – Executing [[email protected]:15] GotoIf(“SIP/8881-00000006”, “1?continue”) in new stack

[2013-09-12 14:04:43] VERBOSE[10880][C-00000006] pbx.c: – Goto (macro-user-callerid,s,28)

[2013-09-12 14:04:43] VERBOSE[10880][C-00000006] pbx.c: – Executing [[email protected]:28] Set(“SIP/8881-00000006”, “CALLERID(number)=8881”) in new stack

[2013-09-12 14:04:43] VERBOSE[10880][C-00000006] pbx.c: – Executing [[email protected]:29] Set(“SIP/8881-00000006”, “CALLERID(name)=MyHomeExt”) in new stack

[2013-09-12 14:04:43] VERBOSE[10880][C-00000006] pbx.c: – Executing [[email protected]:30] Set(“SIP/8881-00000006”, “CDR(cnum)=8881”) in new stack

[2013-09-12 14:04:43] VERBOSE[10880][C-00000006] pbx.c: – Executing [[email protected]:31] Set(“SIP/8881-00000006”, “CDR(cnam)=MyHomeExt”) in new stack

[2013-09-12 14:04:43] VERBOSE[10880][C-00000006] pbx.c: – Executing [[email protected]:32] Set(“SIP/8881-00000006”, “CHANNEL(language)=en”) in new stack

[2013-09-12 14:04:43] VERBOSE[10880][C-00000006] pbx.c: – Executing [[email protected]:2] NoCDR(“SIP/8881-00000006”, “”) in new stack

[2013-09-12 14:04:43] VERBOSE[10880][C-00000006] pbx.c: – Executing [[email protected]:3] Progress(“SIP/8881-00000006”, “”) in new stack

[2013-09-12 14:04:43] VERBOSE[10880][C-00000006] pbx.c: – Executing [[email protected]:4] Wait(“SIP/8881-00000006”, “1”) in new stack

[2013-09-12 14:04:44] VERBOSE[10880][C-00000006] pbx.c: – Executing [[email protected]:5] Progress(“SIP/8881-00000006”, “”) in new stack

[2013-09-12 14:04:44] VERBOSE[10880][C-00000006] pbx.c: – Executing [[email protected]:6] Playback(“SIP/8881-00000006”, “silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer”) in new stack

[2013-09-12 14:04:44] VERBOSE[10880][C-00000006] file.c: – <SIP/8881-00000006> Playing ‘silence/1.ulaw’ (language ‘en’)

[2013-09-12 14:04:45] VERBOSE[10880][C-00000006] file.c: – <SIP/8881-00000006> Playing ‘cannot-complete-as-dialed.ulaw’ (language ‘en’)

[2013-09-12 14:04:47] VERBOSE[10880][C-00000006] pbx.c: == Spawn extension (from-internal, 12065551234, 6) exited non-zero on ‘SIP/8881-00000006’

[2013-09-12 14:04:47] VERBOSE[10880][C-00000006] pbx.c: – Executing [[email protected]:1] Hangup(“SIP/8881-00000006”, “”) in new stack

[2013-09-12 14:04:47] VERBOSE[10880][C-00000006] pbx.c: == Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/8881-00000006’

ElevateNet · February 18, 2014, 5:52am

Thanks for the work ledude and SkykingOH. I had a similar issue
after watching the same video ledude did to set things up. Once I
deleted the numbers in CID on outbound - “Ok, you are confused. In the outbound route you show that you have a value in the callerid field next to match pattern that has to be blank (at least in this case).” all of my systems work now for ext to ext, inbound and outbound. That is one thing to be careful of if someone is watching “http://www.youtube.com/watch?v=G1dXb85Bzts” and going step by step. I
am not sure why they put that in there but my CID works fine without it. Next steps will be to allow the different extensions to call different lines and get
an auto attendant and management system in place. How have things been going with you setup ledude?

ledude · September 13, 2013, 4:30pm

No one really has seen this problem?

SkykingOH · September 13, 2013, 4:59pm

Looks to me like you don’t have a route for that pattern.

ledude · September 16, 2013, 12:25am

Thanks SkyKingOH. So is it more configuration screw up or something else because I have tried many combinations on my inbound. and nothing seems to work. Voip.ms is contacting the previous vendor who we lnp from to make sure that it’s not the lnp that screw up.

SkykingOH · September 16, 2013, 12:48am

I thought we were working on outbound?

If you look at the log you will see call never goes to trunk.

How is your outbound route pattern configured?

ledude · September 16, 2013, 1:58am

Sorry Skyking. Yeah it’s for both inbound and outbound. Too many problems. Here is the route pattern I have for outbound.

prepend + prefix | match pattern /callerid
empty + empty | 1NXXNXXXXXX /2065551212
1 + empty | NXXNXXXXXX /2065551212
empty + empty | +1NXXNXXXXXX /2065551212

Is this what you were looking for? As for inbound, I got confirmation from Voip.ms that everything is fixed and should be ok but when I test it this time, the message said something like “The number you dialed is not in service, please check your number and try again” Many thanks again for your help SkykingOH.

ledude · September 16, 2013, 2:11am

Here is my dial out log.


[2013-09-15 19:04:46] DEBUG[1725][C-0000004d] sip/sdp_crypto.c: Crypto line: a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:7z7rwzpFJIEqVvVQpm/SdAdCl2DSz17zDrz2t9mn
[2013-09-15 19:04:46] VERBOSE[12071][C-0000004d] pbx.c: -- Executing [[email protected]:1] Macro("SIP/8890-00000052", "user-callerid,LIMIT,") in new stack
[2013-09-15 19:04:46] VERBOSE[12071][C-0000004d] pbx.c: -- Executing [[email protected]:1] Set("SIP/8890-00000052", "TOUCH_MONITOR=1379297086.82") in new stack
[2013-09-15 19:04:46] VERBOSE[12071][C-0000004d] pbx.c: -- Executing [[email protected]:2] Set("SIP/8890-00000052", "AMPUSER=8890") in new stack
[2013-09-15 19:04:46] VERBOSE[12071][C-0000004d] pbx.c: -- Executing [[email protected]:3] GotoIf("SIP/8890-00000052", "0?report") in new stack
[2013-09-15 19:04:46] VERBOSE[12071][C-0000004d] pbx.c: -- Executing [[email protected]:4] ExecIf("SIP/8890-00000052", "1?Set(REALCALLERIDNUM=8890)") in new stack
[2013-09-15 19:04:47] VERBOSE[12071][C-0000004d] pbx.c: -- Executing [[email protected]:5] Set("SIP/8890-00000052", "AMPUSER=8890") in new stack
[2013-09-15 19:04:47] VERBOSE[12071][C-0000004d] pbx.c: -- Executing [[email protected]:6] Set("SIP/8890-00000052", "AMPUSERCIDNAME=MyCaller ID 2065551212") in new stack
[2013-09-15 19:04:47] VERBOSE[12071][C-0000004d] pbx.c: -- Executing [[email protected]:7] GotoIf("SIP/8890-00000052", "0?report") in new stack
[2013-09-15 19:04:47] VERBOSE[12071][C-0000004d] pbx.c: -- Executing [[email protected]:8] Set("SIP/8890-00000052", "AMPUSERCID=8890") in new stack
[2013-09-15 19:04:47] VERBOSE[12071][C-0000004d] pbx.c: -- Executing [[email protected]:9] Set("SIP/8890-00000052", "__DIAL_OPTIONS=Ttr") in new stack
[2013-09-15 19:04:47] VERBOSE[12071][C-0000004d] pbx.c: -- Executing [[email protected]:10] Set("SIP/8890-00000052", "CALLERID(all)="MyCaller ID 2065551212" ") in new stack
[2013-09-15 19:04:47] VERBOSE[12071][C-0000004d] pbx.c: -- Executing [[email protected]:11] GotoIf("SIP/8890-00000052", "0?limit") in new stack
[2013-09-15 19:04:47] VERBOSE[12071][C-0000004d] pbx.c: -- Executing [[email protected]:12] ExecIf("SIP/8890-00000052", "1?Set(GROUP(concurrency_limit)=8890)") in new stack
[2013-09-15 19:04:47] VERBOSE[12071][C-0000004d] pbx.c: -- Executing [[email protected]:13] ExecIf("SIP/8890-00000052", "0?Set(CHANNEL(language)=)") in new stack
[2013-09-15 19:04:47] VERBOSE[12071][C-0000004d] pbx.c: -- Executing [[email protected]:14] GosubIf("SIP/8890-00000052", "7?sub-ccss,s,1(from-internal,12065551234)") in new stack
[2013-09-15 19:04:47] VERBOSE[12071][C-0000004d] pbx.c: -- Executing [[email protected]:1] ExecIf("SIP/8890-00000052", "0?Return()") in new stack
[2013-09-15 19:04:47] VERBOSE[12071][C-0000004d] pbx.c: -- Executing [[email protected]:2] Set("SIP/8890-00000052", "CCSS_SETUP=TRUE") in new stack
[2013-09-15 19:04:47] VERBOSE[12071][C-0000004d] pbx.c: -- Executing [[email protected]:3] GosubIf("SIP/8890-00000052", "0?monitor_config,1(from-internal,12065551234):monitor_default,1(from-internal,12065087342)") in new stack
[2013-09-15 19:04:47] VERBOSE[12071][C-0000004d] pbx.c: -- Executing [[email protected]:1] GotoIf("SIP/8890-00000052", "0?is_exten") in new stack
[2013-09-15 19:04:47] VERBOSE[12071][C-0000004d] pbx.c: -- Executing [[email protected]:2] StackPop("SIP/8890-00000052", "") in new stack
[2013-09-15 19:04:47] VERBOSE[12071][C-0000004d] pbx.c: -- Executing [[email protected]:3] Return("SIP/8890-00000052", "FALSE") in new stack
[2013-09-15 19:04:47] VERBOSE[12071][C-0000004d] pbx.c: -- Executing [[email protected]:15] GotoIf("SIP/8890-00000052", "1?continue") in new stack
[2013-09-15 19:04:47] VERBOSE[12071][C-0000004d] pbx.c: -- Goto (macro-user-callerid,s,28)
[2013-09-15 19:04:47] VERBOSE[12071][C-0000004d] pbx.c: -- Executing [[email protected]:28] Set("SIP/8890-00000052", "CALLERID(number)=8890") in new stack
[2013-09-15 19:04:47] VERBOSE[12071][C-0000004d] pbx.c: -- Executing [[email protected]:29] Set("SIP/8890-00000052", "CALLERID(name)=MyCaller ID 2065551212") in new stack
[2013-09-15 19:04:47] VERBOSE[12071][C-0000004d] pbx.c: -- Executing [[email protected]:30] Set("SIP/8890-00000052", "CDR(cnum)=8890") in new stack
[2013-09-15 19:04:47] VERBOSE[12071][C-0000004d] pbx.c: -- Executing [[email protected]:31] Set("SIP/8890-00000052", "CDR(cnam)=MyCaller ID 2065551212") in new stack
[2013-09-15 19:04:47] VERBOSE[12071][C-0000004d] pbx.c: -- Executing [[email protected]:32] Set("SIP/8890-00000052", "CHANNEL(language)=en") in new stack
[2013-09-15 19:04:47] VERBOSE[12071][C-0000004d] pbx.c: -- Executing [[email protected]:2] NoCDR("SIP/8890-00000052", "") in new stack
[2013-09-15 19:04:47] VERBOSE[12071][C-0000004d] pbx.c: -- Executing [[email protected]:3] Progress("SIP/8890-00000052", "") in new stack
[2013-09-15 19:04:47] DEBUG[12071][C-0000004d] sip/sdp_crypto.c: Crypto line: a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:7z7rwzpFJIEqVvVQpm/SdAdCl2DSz17zDrz2t9mn
[2013-09-15 19:04:47] VERBOSE[12071][C-0000004d] pbx.c: -- Executing [[email protected]:4] Wait("SIP/8890-00000052", "1") in new stack
[2013-09-15 19:04:48] VERBOSE[12071][C-0000004d] pbx.c: -- Executing [[email protected]:5] Progress("SIP/8890-00000052", "") in new stack
[2013-09-15 19:04:48] VERBOSE[12071][C-0000004d] pbx.c: -- Executing [[email protected]:6] Playback("SIP/8890-00000052", "silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer") in new stack
[2013-09-15 19:04:48] VERBOSE[12071][C-0000004d] file.c: --  Playing 'silence/1.ulaw' (language 'en')
[2013-09-15 19:04:49] VERBOSE[12071][C-0000004d] file.c: --  Playing 'cannot-complete-as-dialed.ulaw' (language 'en')
[2013-09-15 19:04:51] VERBOSE[12071][C-0000004d] file.c: --  Playing 'check-number-dial-again.ulaw' (language 'en')
[2013-09-15 19:04:53] VERBOSE[12071][C-0000004d] pbx.c: -- Executing [[email protected]:7] Wait("SIP/8890-00000052", "1") in new stack
[2013-09-15 19:04:54] VERBOSE[12071][C-0000004d] pbx.c: -- Executing [[email protected]:8] Congestion("SIP/8890-00000052", "20") in new stack
[2013-09-15 19:04:54] WARNING[12071][C-0000004d] channel.c: Prodding channel 'SIP/8890-00000052' failed
[2013-09-15 19:04:54] VERBOSE[12071][C-0000004d] pbx.c: == Spawn extension (from-internal, 12065551234, 8) exited non-zero on 'SIP/8890-00000052'
[2013-09-15 19:04:54] VERBOSE[12071][C-0000004d] pbx.c: -- Executing [[email protected]:1] Hangup("SIP/8890-00000052", "") in new stack
[2013-09-15 19:04:54] VERBOSE[12071][C-0000004d] pbx.c: == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/8890-00000052'

SkykingOH · September 16, 2013, 2:16am

Why do you have the called ID field filled in the outbound routes? That’s not what it is for. Clear it out and you will be fine on outbound.

ledude · September 16, 2013, 2:17am

Here is my inbound log.


[2013-09-15 19:03:13] VERBOSE[12066][C-0000004c] pbx.c: -- Executing [[email protected]:1] NoOp("SIP/50.23.160.51-00000051", "Received incoming SIP connection from unknown peer to 2065551212") in new stack
[2013-09-15 19:03:13] VERBOSE[12066][C-0000004c] pbx.c: -- Executing [[email protected]:2] Set("SIP/50.23.160.51-00000051", "DID=2065551212") in new stack
[2013-09-15 19:03:13] VERBOSE[12066][C-0000004c] pbx.c: -- Executing [[email protected]:3] Goto("SIP/50.23.160.51-00000051", "s,1") in new stack
[2013-09-15 19:03:13] VERBOSE[12066][C-0000004c] pbx.c: -- Goto (from-sip-external,s,1)
[2013-09-15 19:03:13] VERBOSE[12066][C-0000004c] pbx.c: -- Executing [[email protected]:1] GotoIf("SIP/50.23.160.51-00000051", "0?checklang:noanonymous") in new stack
[2013-09-15 19:03:13] VERBOSE[12066][C-0000004c] pbx.c: -- Goto (from-sip-external,s,5)
[2013-09-15 19:03:13] VERBOSE[12066][C-0000004c] pbx.c: -- Executing [[email protected]:5] Set("SIP/50.23.160.51-00000051", "TIMEOUT(absolute)=15") in new stack
[2013-09-15 19:03:13] VERBOSE[12066][C-0000004c] func_timeout.c: -- Channel will hangup at 2013-09-15 19:03:28.047 PDT.
[2013-09-15 19:03:13] VERBOSE[12066][C-0000004c] pbx.c: -- Executing [[email protected]:6] Answer("SIP/50.23.160.51-00000051", "") in new stack
[2013-09-15 19:03:13] VERBOSE[12066][C-0000004c] pbx.c: -- Executing [[email protected]:7] Wait("SIP/50.23.160.51-00000051", "2") in new stack
[2013-09-15 19:03:15] VERBOSE[12066][C-0000004c] pbx.c: -- Executing [[email protected]:8] Playback("SIP/50.23.160.51-00000051", "ss-noservice") in new stack
[2013-09-15 19:03:15] VERBOSE[12066][C-0000004c] file.c: --  Playing 'ss-noservice.ulaw' (language 'en')
[2013-09-15 19:03:20] VERBOSE[12066][C-0000004c] pbx.c: -- Executing [[email protected]:9] PlayTones("SIP/50.23.160.51-00000051", "congestion") in new stack
[2013-09-15 19:03:20] VERBOSE[12066][C-0000004c] pbx.c: -- Executing [[email protected]:10] Congestion("SIP/50.23.160.51-00000051", "5") in new stack
[2013-09-15 19:03:20] VERBOSE[12066][C-0000004c] pbx.c: == Spawn extension (from-sip-external, s, 10) exited non-zero on 'SIP/50.23.160.51-00000051'
[2013-09-15 19:03:20] VERBOSE[12066][C-0000004c] pbx.c: -- Executing [[email protected]:1] Hangup("SIP/50.23.160.51-00000051", "") in new stack
[2013-09-15 19:03:20] VERBOSE[12066][C-0000004c] pbx.c: == Spawn extension (from-sip-external, h, 1) exited non-zero on 'SIP/50.23.160.51-00000051'
[2013-09-15 19:04:46] VERBOSE[1725][C-0000004d] netsock2.c: == Using SIP VIDEO TOS bits 136
[2013-09-15 19:04:46] VERBOSE[1725][C-0000004d] netsock2.c: == Using SIP VIDEO CoS mark 6
[2013-09-15 19:04:46] VERBOSE[1725][C-0000004d] netsock2.c: == Using SIP RTP TOS bits 184
[2013-09-15 19:04:46] VERBOSE[1725][C-0000004d] netsock2.c: == Using SIP RTP CoS mark 5

ledude · September 16, 2013, 2:27am

Thanks SkykinOH. That’s strange. Because the callerID field you mentioned is actually a Display name on the extension 8890. I cleared the callerID Field and it still shown AMPUSERCIDNAME and CALLERID name from the display name on extension. Am I missing something?

SkykingOH · September 16, 2013, 2:38am

Ok, you are confused. In the outbound route you show that you have a value in the callerid field next to match pattern that has to be blank (at least in this case).

As far as inbound “Received incoming SIP connection from unknown peer” your trunk does not match the call you received. You can turn on anonymous SIP and you will see your inbound route works. I would not leave it on, but it is useful for testing.

ledude · September 16, 2013, 3:31am

Thanks SkykingOH. You are absolutely correct. I’m a very confused dude. I guess the confusion on the callerid next to the match pattern comes from setting up google voice inbound because that’s how they require to make it work.
FYI. I removed the callerid next to the match pattern and it sure works. However, the callerid shown that it comes from a strange number that I don’t recognize. I thought I should define the callerid in the trunk for that numbers (which I did define it) but it didn’t show the caller ID I want.

As for incoming, I tried to allow anonymous SIP and the incoming now just ring but not to my extension. So is there something I need to set it in the inbound routes so that the trunk match the call I received?

Many thanks for all of your help again SkykingOH. This is definitely becoming “drinking from the fire hose” moment. But at least I have someone like you to help guide me to the right direction though.

SkykingOH · September 16, 2013, 4:32am

For inbound route you just need “2065551212” in the DID field and everything else blank.

You should be able to set CID in extension or in trunk. What your provider does with it is up to them.

Our SIPStation trunks interoperate perfectly and you are helping support development of your favorite FreePBX.

ledude · September 16, 2013, 4:55am

Thanks SkykingOH.

I did that for the inbound route in the DID field and left everything else blank. But I keep getting the “The number you dialed is not in service, please check your number and try again” when I called in. Very strange

As for outbound. I may need to check with my provider because it’s not using my callerid setup either in the trunk or extension. All it shows was their CallerID. Strange.

SkykingOH · September 16, 2013, 5:09am

Please post a trace with anonymous calling turned on.

ledude · September 16, 2013, 5:21am

As you can see, it keeps on ringing and the extension is not picking it up. When I turned off the anonymous sip, it becomes, “The number you have called is not in service”. Here is the log.


– Executing [[email protected]:1] NoOp(“SIP/50.23.160.51-0000009 5”, “Received incoming SIP connection from unknown peer to 2065551212”) in new s tack

– Executing [[email protected]:2] Set("SIP/50.23.160.51-00000095 ", “DID=2065551212”) in new stack

– Executing [[email protected]:3] Goto(“SIP/50.23.160.51-0000009 5”, “s,1”) in new stack

– Goto (from-sip-external,s,1)

– Executing [[email protected]:1] GotoIf(“SIP/50.23.160.51-00000095”, “0? checklang:noanonymous”) in new stack

– Goto (from-sip-external,s,5)

– Executing [[email protected]:5] Set(“SIP/50.23.160.51-00000095”, “TIMEO UT(absolute)=15”) in new stack

– Channel will hangup at 2013-09-15 22:18:44.351 PDT.

– Executing [[email protected]:6] Answer(“SIP/50.23.160.51-00000095”, “”) in new stack

– Executing [[email protected]:7] Wait(“SIP/50.23.160.51-00000095”, “2”) in new stack

– Executing [[email protected]:8] Playback(“SIP/50.23.160.51-00000095”, “ss-noservice”) in new stack

– <SIP/50.23.160.51-00000095> Playing ‘ss-noservice.ulaw’ (language ‘en’)

– Executing [[email protected]:9] PlayTones(“SIP/50.23.160.51-00000095”, “congestion”) in new stack

– Executing [[email protected]:10] Congestion(“SIP/50.23.160.51-00000095”, “5”) in new stack

== Spawn extension (from-sip-external, s, 10) exited non-zero on ‘SIP/50.23.160.51-00000095’

– Executing [[email protected]:1] Hangup(“SIP/50.23.160.51-00000095”, “”) in new stack

== Spawn extension (from-sip-external, h, 1) exited non-zero on ‘SIP/50.23.160.51-00000095’

SkykingOH · September 16, 2013, 7:15am

It’s clear from the log your extension is unreachable. Why do you have a raw SIP dial string? What are you trying to send a call to?

You should test to a phone on the same LAN as the server:

Dial(“SIP/50.23.160.51-0000008a”, “SIP/8890,Ttr”) in new stack
[2013-09-15 22:10:52] WARNING[13173][C-00000086]: app_dial.c:2437 dial_exec_full: Unable to create channel of type ‘SIP’ (cause 20 - Subscriber absent)

ledude · September 16, 2013, 7:50am

Thanks for the input SkykingOH. Pardon my noob but I have no idea what you are talking about. All I want to do is very simple. Just a call comes in, go to the extension, ring the phone and that’s it. Perhaps one day, I’ll add the ivr to it. So I have no idea if I somehow choose a raw SIP dial string because I didn’t configure it that way. Just a simple standard example that I found in the documentation and from googling. Is there anything I can provide to you so that you can pinpoint what I did wrong in the configuration?

I tried extension to extension and it’s working. Google voice inbound and outbound, works just fine. IpKall, same issue with voip.ms. Very puzzled.

ledude · September 16, 2013, 6:38pm

@SkykingOH. First all, thanks for all of your help. Here is what I did to make it works.

Old Settings on the trunk Peer Details.


canreinvite=nonat
;nat=yes 
context=from-trunk
host=provider.hostaddress.com
secret=secretpassword
type=peer
[email protected]
disallow=all
allow=ulaw
allow=g722
; allow=g729 ; uncomment if you purchased g.729 from Digium
[email protected]
trustrpid=yes
sendrpid=yes
insecure=invite
qualify=yes

working settings:


context=from-trunk
host=provider.hostaddress.com
secret=secretpassword
type=friend
username=160976_8889899688
disallow=all
allow=ulaw&g722
; allow=g729 ; uncomment if you purchased g.729 from Digium
[email protected]
insecure=port,invite

Too much information that I didn’t understand and gathered from Google leads to all the confusion. Many thanks again everyone.