Can place but not receive calls?

R

Ronald Raygun

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I should probably state that I am half-successfully running PiaF in a VMWare ESXi server. I say half-successfully because I'm able to place outbound calls but for some reason the PBX is not catching inbound calls.

When watching the panel, I see the incoming call arrive in the sipgate trunk as it should, but it seems the call is dropped or something similar happens. When I place an outbound call, I see the call go to the parking lot the sipgate trunk receive the call and connect to my extension.

Here's a quick video showing what I think is the 'problem' http://www.screencast.com/users/Ron...ng/media/906d0db9-10c1-44e1-874f-f4a41eebb711

While making the video, I had to dial out twice because the first time, I heard the message "please wait while we connect your call" but I didn't hear the hold music after that.


Thanks in advance for your help.

[edit]

I also attached a screenshot of the status screen. I removed the IP address and hostname just in case.
 

Attachments

There are a lot of things to check but I think you'd find it more helpful to be looking at file logs or in realtime you should type:

asterisk -r -vvv

at that root prompt you showed and see whats really happening. You can get all sorts of detail based on how things are set but watching the FOP panel doesn't really tell us much.

Other things would be router, nat, IP (fixed - dynamic)... how is sipgate set up.. show us a scrubbed config etc etc... post that CLI trace from the command above...

Brian
 
Thanks for your help briankelly63

Here is the output of that command you give me whenever I try to call into the asterisk server.

Code: 
  == Parsing '/etc/asterisk/asterisk.conf': Found
  == Parsing '/etc/asterisk/extconfig.conf': Found
Connected to Asterisk 1.4.33 currently running on pbx (pid = 3475)
Verbosity is at least 3
    -- Executing [4156396476@from-trunk:1] NoOp("SIP/sipgate-00000004", "Catch-All DID Match - Found 4156396476 - You probably want a DID for this.") in new stack
    -- Executing [4156396476@from-trunk:2] Goto("SIP/sipgate-00000004", "ext-did|s|1") in new stack
    -- Goto (ext-did,s,1)
    -- Executing [s@ext-did:1] Set("SIP/sipgate-00000004", "__FROM_DID=s") in new stack
    -- Executing [s@ext-did:2] Gosub("SIP/sipgate-00000004", "cidlookup|cidlookup_3|1") in new stack
  == Parsing '/etc/asterisk/manager.conf': Found
  == Parsing '/etc/asterisk/manager_additional.conf': Found
  == Parsing '/etc/asterisk/manager_custom.conf': Found
  == Manager 'admin' logged on from 127.0.0.1
  == Manager 'admin' logged off from 127.0.0.1
    -- Executing [cidlookup_3@cidlookup:1] Set("SIP/sipgate-00000004", "CALLERID(name)=4156396476") in new stack
    -- Executing [cidlookup_3@cidlookup:2] Return("SIP/sipgate-00000004", "") in new stack
    -- Executing [s@ext-did:3] Gosub("SIP/sipgate-00000004", "app-blacklist-check|s|1") in new stack
    -- Executing [s@app-blacklist-check:1] LookupBlacklist("SIP/sipgate-00000004", "") in new stack
    -- Executing [s@app-blacklist-check:2] GotoIf("SIP/sipgate-00000004", "0?blacklisted") in new stack
    -- Executing [s@app-blacklist-check:3] Set("SIP/sipgate-00000004", "CALLED_BLACKLIST=1") in new stack
    -- Executing [s@app-blacklist-check:4] Return("SIP/sipgate-00000004", "") in new stack
    -- Executing [s@ext-did:4] ExecIf("SIP/sipgate-00000004", "0 |Set|CALLERID(name)=7025226448") in new stack
    -- Executing [s@ext-did:5] Set("SIP/sipgate-00000004", "FAX_RX=system") in new stack
    -- Executing [s@ext-did:6] Set("SIP/sipgate-00000004", "FAX_RX_EMAIL=root@localhost") in new stack
    -- Executing [s@ext-did:7] Answer("SIP/sipgate-00000004", "") in new stack
    -- Executing [s@ext-did:8] PlayTones("SIP/sipgate-00000004", "ring") in new stack
  == Spawn extension (ext-did, s, 9) exited non-zero on 'SIP/sipgate-00000004'
pbx*CLI>
pbx*CLI> root@pbx:~ $ asterisk -r -vvv
No such command 'root@pbx:~ $ asterisk -r -vvv' (type 'help root@pbx:~ $' for other possible commands)
pbx*CLI> Asterisk 1.4.33, Copyright (C) 1999 - 2010 Digium, Inc. and others.
No such command
pbx*CLI>   == Parsing '/etc/asterisk/asterisk.conf': Found
No such command 
No such command '  == Parsing '/etc/asterisk/asterisk.conf': Found' (type 'help == Parsing' for other possible commands)
pbx*CLI>   == Parsing '/etc/asterisk/extconfig.conf': Found
No such command '  == Parsing '/etc/asterisk/extconfig.conf': Found' (type 'help == Parsing' for other possible commands)
pbx*CLI> Connected to Asterisk 1.4.33 currently running on pbx (pid = 3475)
pbx*CLI> Verbosity is at least 3
No such command 'Connected to Asterisk 1.4.33 currently running on pbx (pid = 3475)' (type 'help Connected to' for other possible commands)
No such command 'Verbosity is at least 3' (type 'help Verbosity is' for other possible commands)
pbx*CLI>     -- Executing [4156396476@from-trunk:1] NoOp("SIP/sipgate-00000004", "Catch-All DID Match - Found 4156396476 - You probably want a DID for this.") in new stack
pbx*CLI>     -- Executing [4156396476@from-trunk:2] Goto("SIP/sipgate-00000004", "ext-did|s|1") in new stack
No such command '    -- Executing [4156396476@from-trunk:1] NoOp("SIP/sipgate-00000004", "Catch-All DID Match - Found 4156396476 - You probably want a DID for this.") in new stack' (type 'help -- Executing' for other possible commands)
No such command '    -- Executing [4156396476@from-trunk:2] Goto("SIP/sipgate-00000004", "ext-did|s|1") in new stack' (type 'help -- Executing' for other possible commands)
pbx*CLI>     -- Goto (ext-did,s,1)
No such command '    -- Goto (ext-did,s,1)' (type 'help -- Goto' for other possible commands)
pbx*CLI>     -- Executing [s@ext-did:1] Set("SIP/sipgate-00000004", "__FROM_DID=s") in new stack
No such command '    -- Executing [s@ext-did:1] Set("SIP/sipgate-00000004", "__FROM_DID=s") in new stack' (type 'help -- Executing' for other possible commands)
pbx*CLI>     -- Executing [s@ext-did:2] Gosub("SIP/sipgate-00000004", "cidlookup|cidlookup_3|1") in new stack
No such command '    -- Executing [s@ext-did:2] Gosub("SIP/sipgate-00000004", "cidlookup|cidlookup_3|1") in new stack' (type 'help -- Executing' for other possible commands)
pbx*CLI>   == Parsing '/etc/asterisk/manager.conf': Found
No such command '  == Parsing '/etc/asterisk/manager.conf': Found' (type 'help == Parsing' for other possible commands)
pbx*CLI>   == Parsing '/etc/asterisk/manager_additional.conf': Found
No such command '  == Parsing '/etc/asterisk/manager_additional.conf': Found' (type 'help == Parsing' for other possible commands)
pbx*CLI>   == Parsing '/etc/asterisk/manager_custom.conf': Found
No such command '  == Parsing '/etc/asterisk/manager_custom.conf': Found' (type 'help == Parsing' for other possible commands)
pbx*CLI>   == Manager 'admin' logged on from 127.0.0.1
No such command '  == Manager 'admin' logged on from 127.0.0.1' (type 'help == Manager' for other possible commands)
pbx*CLI>   == Manager 'admin' logged off from 127.0.0.1
No such command '  == Manager 'admin' logged off from 127.0.0.1' (type 'help == Manager' for other possible commands)
pbx*CLI>     -- Executing [cidlookup_3@cidlookup:1] Set("SIP/sipgate-00000004", "CALLERID(name)=4156396476") in new stack
No such command '    -- Executing [cidlookup_3@cidlookup:1] Set("SIP/sipgate-00000004", "CALLERID(name)=4156396476") in new stack' (type 'help -- Executing' for other possible commands)
pbx*CLI>     -- Executing [cidlookup_3@cidlookup:2] Return("SIP/sipgate-00000004", "") in new stack
No such command '    -- Executing [cidlookup_3@cidlookup:2] Return("SIP/sipgate-00000004", "") in new stack' (type 'help -- Executing' for other possible commands)
pbx*CLI>     -- Executing [s@ext-did:3] Gosub("SIP/sipgate-00000004", "app-blacklist-check|s|1") in new stack
No such command '    -- Executing [s@ext-did:3] Gosub("SIP/sipgate-00000004", "app-blacklist-check|s|1") in new stack' (type 'help -- Executing' for other possible commands)
pbx*CLI>     -- Executing [s@app-blacklist-check:1] LookupBlacklist("SIP/sipgate-00000004", "") in new stack
No such command '    -- Executing [s@app-blacklist-check:1] LookupBlacklist("SIP/sipgate-00000004", "") in new stack' (type 'help -- Executing' for other possible commands)
pbx*CLI>     -- Executing [s@app-blacklist-check:2] GotoIf("SIP/sipgate-00000004", "0blacklisted") in new stack
No such command '    -- Executing [s@app-blacklist-check:2] GotoIf("SIP/sipgate-00000004", "0blacklisted") in new stack' (type 'help -- Executing' for other possible commands)
pbx*CLI>     -- Executing [s@app-blacklist-check:3] Set("SIP/sipgate-00000004", "CALLED_BLACKLIST=1") in new stack
No such command '    -- Executing [s@app-blacklist-check:3] Set("SIP/sipgate-00000004", "CALLED_BLACKLIST=1") in new stack' (type 'help -- Executing' for other possible commands)
pbx*CLI>     -- Executing [s@app-blacklist-check:4] Return("SIP/sipgate-00000004", "") in new stack
No such command '    -- Executing [s@app-blacklist-check:4] Return("SIP/sipgate-00000004", "") in new stack' (type 'help -- Executing' for other possible commands)
pbx*CLI>     -- Executing [s@ext-did:4] ExecIf("SIP/sipgate-00000004", "0 |Set|CALLERID(name)=7025226448") in new stack
No such command '    -- Executing [s@ext-did:4] ExecIf("SIP/sipgate-00000004", "0 |Set|CALLERID(name)=7025226448") in new stack' (type 'help -- Executing' for other possible commands)
pbx*CLI>     -- Executing [s@ext-did:5] Set("SIP/sipgate-00000004", "FAX_RX=system") in new stack
No such command '    -- Executing [s@ext-did:5] Set("SIP/sipgate-00000004", "FAX_RX=system") in new stack' (type 'help -- Executing' for other possible commands)
pbx*CLI>     -- Executing [s@ext-did:6] Set("SIP/sipgate-00000004", "FAX_RX_EMAIL=root@localhost") in new stack
No such command '    -- Executing [s@ext-did:6] Set("SIP/sipgate-00000004", "FAX_RX_EMAIL=root@localhost") in new stack' (type 'help -- Executing' for other possible commands)
pbx*CLI>     -- Executing [s@ext-did:7] Answer("SIP/sipgate-00000004", "") in new stack
No such command '    -- Executing [s@ext-did:7] Answer("SIP/sipgate-00000004", "") in new stack' (type 'help -- Executing' for other possible commands)
pbx*CLI>     -- Executing [s@ext-did:8] PlayTones("SIP/sipgate-00000004", "ring") in new stack
pbx*CLI>   == Spawn extension (ext-did, s, 9) exited non-zero on 'SIP/sipgate-00000004'
No such command '    -- Executing [s@ext-did:8] PlayTones("SIP/sipgate-00000004", "ring") in new stack' (type 'help -- Executing' for other possible commands)
No such command '  == Spawn extension (ext-did, s, 9) exited non-zero on 'SIP/sipgate-00000004'' (type 'help == Spawn' for other possible commands)
pbx*CLI> pbx*CLI>
No such command 'pbx*CLI>' (type 'help pbx*CLI>' for other possible commands)
pbx*CLI>
I don't really fully understand what I'm seeing here, but I suspect the following is the culprit.

Code: 
 Spawn extension (ext-did, s, 9) exited non-zero on 'SIP/sipgate-00000004'
 
If my googling skills are worth anything, "Spawn Extension ... exited non-zero" means the call was disconnected/dropped or something like that.

How do I troubleshoot?
 
Did you open the correct ports on the router?

It's easy to make a call out with the ports closed but you won't be able to make a call in with the SIP and RTP ports closed.
 
I had a second look through the Incredible PBX install instruction set (http://nerdvittles.com/?p=677) and there was no mention of opening ports for Sipgate.

Which ports do I need to forward to my machine?

Google search says these ones. Can you please confirm?

Port 5004 UDP
Port 5060 UDP
Port 8000 - 8012 UDP
Port 10000 STUN
Port 3478 UDP & TCP
 
A couple of things... if you do a little searching you can check out the whole issue of open ports. One of the features in I-PBX is the idea of security. If you have a proper registration to the VOIP provider the port thing should be taken care of for you by Asterisk and the router. We don't really know if thats the case but since its getting to your switch we have to assume something is working. You could purposely forward the ports to the IP of you I-PBX server for testing but you shouldn't have to leave them open.

To exit out of CLI just press ctrl C or type exit... it looked like you had some trouble with that...

Perhaps someone who is more familiar with using FAX on I-PBX could address the CLI lines that are answering the call and playing tones. I'm not using the I-PBX build but it looks like the call is being answered, ringtone is being played while it listens for any fax tones but then the call is dropped.

On your inbound route that handles this trunk do you you have fax detection set to none? If its not set to none then also check the line below it which is the pause after answer setting.

You could post your trunk and route settings for this path altering any private or security information...

Brian
 
I read Ward's primer on security and was curious whether or not those ports really needed to be opened--seems they don't need to be open.

These are my incoming routes. I'll list everything that isn't blank.

  • any DID / CID
    • Music On Hold: Default
    • Privacy Manager: No
    • Fax Extension: System
    • Fax Email: root@localhost
    • Pause After Answer: 5
    • Source: Caller ID Superfecta
    • Set Destination: Day Night Mode (1) Code1
  • e164 / any CID
    • DID Number: e164
    • Music On Hold: Default
    • Privacy Manager: No
    • Fax Extension: FreePBX default
    • Fax Detection Type: None
    • Source: Caller ID Superfecta
    • Set Destination: Day Night Mode (1) Code1
  • gv-incoming / any CID
    • DID Number: gv-incoming
    • Music On Hold: Default
    • Privacy Manager: No
    • Fax Extension: FreePBX default
    • Fax Detection Type: None
    • Source: Caller ID Superfecta
    • Set Destination: Ring Groups RingAll <700>
  • gv-ringback ########## / xxxxxxxxxx
    • Description: gv-ringback
    • DID Number: ##########
    • Caller ID Number: xxxxxxxxxx
    • Music On Hold: Default
    • Privacy Manager: No
    • Fax Extension: FreePBX default
    • Fax Detection Type: None
    • Source: None
    • Set Destination: Custom Destinations Custom GV-Park
Here are the trunks. All inputs with a value are listed here.
  • local/$OUTNUM$@ (custom)
    • Dial Rules: X. *X.
    • Custom Dial String: local/$OUTNUM$@freenum
  • local/$OUTNUM$@ (custom)
    • Dial Rules: X. *X.
    • Custom Dial String: local/$OUTNUM$@freenum
  • SIP/$OUTNUM$@12 (custom)
  • ENUM
    • Outbound Caller ID: xxxxxxxxxx
    • Maximum Channels: 2
    • Dial Rules: 1+NXXNXXXXXX
    • Dial Rules: 1NXXNXXXXXX
  • PIAF-USA (iax)
    • Outbound Caller ID: yyyyyyyyyy
    • Maximum Channels: 2
    • Dial Rules: 1+NXXNXXXXXX
    • Dial Rules: 1NXXNXXXXXX
    • Peer Details: host=iaxUSA.PBXinaFlash.net
    • Peer Details: qualify=yes
    • Peer Details: secret=aaaaaaaaaa
    • Peer Details: sendrpid=yes
    • Peer Details: trustrpid=yes
    • Peer Details: type=peer
    • Peer Details: username=yyyyyyyyyy
  • ipkall (iax)
  • vitel-outbound (sip)
    • Maximum Channels: 2
    • Peer Details:disallow=all
    • Peer Details:allow=ulaw
    • Peer Details:canreinvite=no
    • Peer Details:context=from-pstn
    • Peer Details:fromuser=yourusername
    • Peer Details:host=outbound1.vitelity.net
    • Peer Details:secret=yourpassword
    • Peer Details:sendrpid=yes
    • Peer Details:trustrpid=yes
    • Peer Details:type=friend
    • Peer Details:username=yourusername
  • vitel-inbound (sip)
    • Maximum Channels: 2
    • Peer Details:disallow=all
    • Peer Details:allow=ulaw
    • Peer Details:canreinvite=no
    • Peer Details:context=from-pstn
    • Peer Details:fromuser=yourusername
    • Peer Details:host=outbound1.vitelity.net
    • Peer Details:secret=yourpassword
    • Peer Details:sendrpid=yes
    • Peer Details:trustrpid=yes
    • Peer Details:type=friend
    • Peer Details:username=yourusername
  • sipgate (sip)
    • Peer Details:
      • type=peer
      • username=[My Username]
      • fromuser=[My Username]
      • secret=[Sipgate Password]
      • context=from-trunk
      • host=sipgate.com
      • fromdomain=sipgate.com
      • insecure=very
      • caninvite=no
      • canreinvite=no
      • nat=yes
      • disallow=all
      • allow=ulaw&alaw
 
First, one of the base principals for the Incredible PBX was, that by using Sipgate, NO PORTS would need to be opened! Since the pbx registers with Sipgate, that establishes the "relationship" that should allow the router to pass the SIP session as well as the RTP sessions as well. (Re-reading the Incredible PBX write-up may b appropriate at this point).

As I read the log, the call is arriving from Sipgate, but the call is not being identified in incoming routes (note the "catch-all" message).

I would suggest that you add an inbound route (or change the "incoming-gv") for the Sipgate DID. I don't recall if this listed as an action in the instructions but without some work in the extensions-custom.conf the DID will never be "incoming-gv". I know I changed mine to the Sipgate DID.

If you simply change the "incoming-gv" entry in that inbound route to your Sipgate DID I think your calls will arrive properly.

Jeff
 
It turns out there were two problems.

1) As Jeffmac figured out, the the Incoming CID needed to be changed from gv-incoming to my sipgate phone number.

After I did that change, calls were going into voicemail. Which leads to the second problem.

2) I was using x-Lite's softphone as one of my endpoints (saving up for an aastra 57i) and in the sip account settings, in the section marked 'domain proxy' the checkbox marked "register with domain and receive incoming calls" was unchecked. I checked that box and set outbound via domain and I was able to receive calls.

I have no idea if the second fix compromises the security of my pbx system, I doubt it but I figure it's better to document it here.


As an aside, I'm guessing other people who have installed Incredible PBX on their computers have been able to place calls without adjusting the DID as I have?
 
Possible solution

"PBX -> PBX Settings -> Inbound Routes > Click on the Route In Question ->

Under "Edit Incoming Route" there are 3 fields:


Description: Main-Inbound-Number
DID Number: 6013094718
Caller ID Number: DO NOT PUT THE PHONE NUMBER HERE!!!!!!!!!!!!!!!!!


As bizarre as this might sound, do NOT put the phone number
in the "Caller ID Number" field!


By deleting this entry, the inbound calls were directed to the
proper context and the IVR."


This resolved my issue of not being able to receive calls. I merely deleted my google voice number from the gv inbound route caller ID entry. Worked!
 
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