Unable to call into extensions or from extension to extension

General Help
1

When trying to call into an extension, it goes straight to voicemail. I setup a new extension with no voicemail and it goes straight to congested. I have not idea what happened, but it happened after an update to the core. If you’re able to help me debug, that would be great! Thanks a lot

[Jun 19 17:01:53] – Executing Macro(“SIP/203-12fa”, “exten-vm|novm|204”) in new stack
[Jun 19 17:01:53] – Executing Macro(“SIP/203-12fa”, “user-callerid”) in new stack
[Jun 19 17:01:53] – Executing Set(“SIP/203-12fa”, “AMPUSER=203”) in new stack
[Jun 19 17:01:53] – Executing GotoIf(“SIP/203-12fa”, “0?report”) in new stack
[Jun 19 17:01:53] – Executing ExecIf(“SIP/203-12fa”, “1|Set|REALCALLERIDNUM=203”) in new stack
[Jun 19 17:01:53] – Executing Set(“SIP/203-12fa”, “AMPUSER=203”) in new stack
[Jun 19 17:01:53] – Executing Set(“SIP/203-12fa”, “AMPUSERCIDNAME=foo bar”) in new stack
[Jun 19 17:01:53] – Executing GotoIf(“SIP/203-12fa”, “0?report”) in new stack
[Jun 19 17:01:53] – Executing Set(“SIP/203-12fa”, “AMPUSERCID=203”) in new stack
[Jun 19 17:01:53] – Executing Set(“SIP/203-12fa”, “CALLERID(all)=“foo bar” <203>”) in new stack
[Jun 19 17:01:53] – Executing Set(“SIP/203-12fa”, “REALCALLERIDNUM=203”) in new stack
[Jun 19 17:01:53] – Executing GotoIf(“SIP/203-12fa”, “0?continue”) in new stack
[Jun 19 17:01:53] – Executing Set(“SIP/203-12fa”, “__TTL=64”) in new stack
[Jun 19 17:01:53] – Executing GotoIf(“SIP/203-12fa”, “1?continue”) in new stack
[Jun 19 17:01:53] – Goto (macro-user-callerid,s,19)
[Jun 19 17:01:53] – Executing NoOp(“SIP/203-12fa”, “Using CallerID “foo bar” <203>”) in new stack
[Jun 19 17:01:53] – Executing Set(“SIP/203-12fa”, “RingGroupMethod=none”) in new stack
[Jun 19 17:01:53] – Executing Set(“SIP/203-12fa”, “VMBOX=novm”) in new stack
[Jun 19 17:01:53] – Executing Set(“SIP/203-12fa”, “EXTTOCALL=204”) in new stack
[Jun 19 17:01:53] – Executing Set(“SIP/203-12fa”, “CFUEXT=”) in new stack
[Jun 19 17:01:53] – Executing Set(“SIP/203-12fa”, “CFBEXT=”) in new stack
[Jun 19 17:01:53] – Executing Set(“SIP/203-12fa”, “RT=”) in new stack
[Jun 19 17:01:53] – Executing Macro(“SIP/203-12fa”, “record-enable|204|IN”) in new stack
[Jun 19 17:01:53] – Executing GotoIf(“SIP/203-12fa”, “1?check”) in new stack
[Jun 19 17:01:53] – Goto (macro-record-enable,s,4)
[Jun 19 17:01:53] – Executing AGI(“SIP/203-12fa”, “recordingcheck|20090620-000153|1245456113.7”) in new stack
[Jun 19 17:01:53] – Launched AGI Script /opt/asterisk/agi-bin/recordingcheck
[Jun 19 17:01:53] – AGI Script recordingcheck completed, returning 0
[Jun 19 17:01:53] – Executing MacroExit(“SIP/203-12fa”, “”) in new stack
[Jun 19 17:01:53] – Executing Macro(“SIP/203-12fa”, “dial||tr|204”) in new stack
[Jun 19 17:01:53] – Executing GotoIf(“SIP/203-12fa”, “1?dial”) in new stack
[Jun 19 17:01:53] – Goto (macro-dial,s,3)
[Jun 19 17:01:53] – Executing AGI(“SIP/203-12fa”, “dialparties.agi”) in new stack
[Jun 19 17:01:53] – Launched AGI Script /opt/asterisk/agi-bin/dialparties.agi
[Jun 19 17:01:53] – AGI Script dialparties.agi completed, returning 0
[Jun 19 17:01:53] – Executing NoOp(“SIP/203-12fa”, "Returned from dialparties with no extensions to call and DIALSTATUS: ") in new stack
[Jun 19 17:01:53] – Executing GotoIf(“SIP/203-12fa”, “0?exit|return”) in new stack
[Jun 19 17:01:53] – Executing Set(“SIP/203-12fa”, “SV_DIALSTATUS=”) in new stack
[Jun 19 17:01:53] – Executing GosubIf(“SIP/203-12fa”, “0?docfu|1”) in new stack
[Jun 19 17:01:53] – Executing GosubIf(“SIP/203-12fa”, “0?docfb|1”) in new stack
[Jun 19 17:01:53] – Executing Set(“SIP/203-12fa”, “DIALSTATUS=”) in new stack
[Jun 19 17:01:53] – Executing NoOp(“SIP/203-12fa”, “Voicemail is novm”) in new stack
[Jun 19 17:01:53] – Executing GotoIf(“SIP/203-12fa”, “1?s-|1”) in new stack
[Jun 19 17:01:53] – Goto (macro-exten-vm,s-,1)
[Jun 19 17:01:53] – Executing NoOp(“SIP/203-12fa”, "IVR_RETVM: IVR_CONTEXT: ") in new stack
[Jun 19 17:01:53] – Executing GotoIf(“SIP/203-12fa”, “0?exit|1”) in new stack
[Jun 19 17:01:53] – Executing PlayTones(“SIP/203-12fa”, “congestion”) in new stack
[Jun 19 17:01:53] – Executing Congestion(“SIP/203-12fa”, “10”) in new stack
[Jun 19 17:01:53] == Spawn extension (macro-exten-vm, s-, 4) exited non-zero on ‘SIP/203-12fa’ in macro ‘exten-vm’
[Jun 19 17:01:53] == Spawn extension (macro-exten-vm, s-, 4) exited non-zero on 'SIP/203-12fa’
fromage*CLI>

2

Try removing the extensions and re entering them.

3

I’ve done that, as I’ve tried this with a brand new extension. Any other ideas?

4

Any other ideas? This has basically left me with an non operational Asterisk since this upgrade to FreePBX.

Thanks in advance.

5

wprater,

While you have provided a call trace you ahve not provided a lot of other details for us to go on. You said you upgraded FreePBX but that statment is no good to us. So how about providing from what version to what version? What version of asterisk are you using? Are the extensions showing as registered in asterisk? Anything else upgraded in the process? Is this a distro built box or a hand built one?

The more information you give us the more we are going to be able to help.

6

I upgraded to FreePBX 2.5.1.7 core which was a minor revision.
Asterisk version: asterisk-1.2.7.1-solvoip-143

Yes, the extensions are registered to asterisk and can make outgoing calls, but one cannot call into the messages. As you can see in the trace, the DIALSTATUS is an empty string. Maybe that info is helpful?

Im running on a solaris box and needed to make some very minor tweaks to get everything up and running. Perhaps there was an error in the upgrade process? I wonder if there is a way to run a few of the FreePBX upgrade scripts manually?

7

Well if what you wrote is not a typo (happens to the best of us), asterisk version 1.2.7 is a VERY VERY old version. There have been hundreds and hundreds (probably thousands) of asterisk fixes applied between then and the more current releases like 1.2.26.1 It is strongly recommended that you upgrade asterisk to a much newer version as this could be asterisk issue. I know the newer versions works well with the later asterisk 1.2.x versions.

I know that the latest FreePBX code works just find with 1.2.26.1 as I have a box at that version.

8

That is the only version of asterisk that I could find that will run on Solaris. It used to work fine, so there must be a configuration issue. Can you help me diagnose ?

9

I’d love to but this is not the weekend for me to help in the next few days. I’m heading out the door for several days and will not be back till Mid Monday.

First thing I’d check is at asterisk forums to see if there is a newer version for Solaris (there has to be). If you can get on the freenode.net IRC channels, go to #asterisk and ask there. There is a large collection of very smart asterisk people (just do not mention FreePBX) and ask how to get a newer version of 1.2 or even 1.4 (say around the 1.4.20-21 range) for Solaris. I’m sure somebody will know where to find it.

10

Thanks for you willingness to help! I just figured it out, the path to php was incorrect in the dialparties.agi.

FreePBX should really use the env trick to use php in their shell scripts.

For example:
#!/usr/bin/env php