Extens work, trunks work, can't dial btwn extns

General Help
#1

Hi there, I posted on another topic, but was advised to open a new topic:

From http://www.freepbx.org/support/documentation/howtos/howto-setup-a-remote-sip-extension

Hi everyone, not sure if I’ve got this in the right place, because my problem is a little different than everyone has here. My asterisk system is in a remote location. I can register my extensions and use my trunks for outgoing calls. It’s just that I can’t call from one extension to another. It errors out with 603.

The extensions show as online in FOP. They will go to voicemail if it’s set up under followme and the audio there works fine, regardless of what extension you’re calling from. Echo tests work great from any individual ext as well.

Any insight would be appreciated. Thanks.

CON’D:

In response to a question posed on the other thread, my setup is the base is ubuntu, with no firewall on the server. It is remote to all extensions. The extensions I use are a linksys 3102, xlite and a linux softphone, and a mobile extension with fring/nokia sip (the nokia client is giving me headaches too, but that’s separate)

As stated above, the extensions register and stay registered. They can call out, but can’t call each other. I am having difficulty having calls come in too, but I haven’t been able to test that as well as should be.

Any help would be great.

#2

When you build your own, you are sort of on your own. What I mean is, if you choose to build a system from scratch using a different “flavor” of Linux than most of us use (CentOS) then you may encounter issues that are unique to your particular installation, and this certainly sounds like one of those. If at all possible, I’d try temporarily installing one of the “all-in-one” distributions (such as Elsatix, PBX in a Flash, Asterisk NOW, Fonica PABX, etc.), even if you do it on a different machine - note it will wipe EVERYTHING off the hard drive, so do it using an older system that you aren’t using for anything else. Set that up and see if you still have the same issue.

But if you want to try a little troubleshooting first, go to the Linux command prompt (from a terminal window) and start asterisk -rvvvv and then try placing a call (that you normally have problems with) while watching the CLI. You may see something in the output that will give you a clue as to the cause of your issue.

One other point, make sure you have not set canreinvite to yes on any of your extensions. If you do that, and any extensions are behind a firewall, they may try to communicate with each other directly without going through the server, and because of the firewall there’s a good chance that won’t work. So, at least while troubleshooting, make sure canreinvite is set to no on all extensions.

#3

could be a lot of things. If all the extensions are remote and they can make calls but can’t call each other, sounds very likely a lack of keepalive on the remote phones keeping their remote firewalls open.

Again though, it could be a lot of things and the information provided is minimal for someone to provide too much help. Also, turn on SIP debugging when making the calls, it will possibly tell you a lot.

#4

It seems to “have resolved”. I don’t like that sort of solution better than anyone else, but I had an experience like this on an install in 2007, which I only remembered after doing it again this time.

I had extensions in the 3000 range. I don’t know if there’s anything special about this range or not, but they wouldn’t work. Then I remembered what fixed it a few years ago, and added an extension in the 1000 range. Bam, done. So far…

Since it doesn’t seem like a real solution it will take some further testing.

I still have issues getting a Nokia SIP client on, and my trunks register, but don’t show for incoming. Ah… :slight_smile:

Thanks for your help everyone.

Dave

#5

Dave,

This may not be the issue, but some firewall routers, like early versions of PFsense had trouble with multiple SIP clients natted behind the same IP. My old antique Smoothwall seems to be immune to this.

#6

your issue has nothing to do with extensions being 1000 vs. 3000 range, it is completely unrelated. Being that you are ‘shooting in the dark’ as well as not providing any information of any value to the people on this board, you will very likely end up continuing to have problems so be prepared…

#7

Thanks for your suggestions everyone.

Philippe, I’m not shooting in the dark at all. I’m telling you about TWO instances of this happening to me. And, as it happens, a friend of mine has had the same issue in the past. I thought it was par for the course, and therefore advised him to follow my path and make an extension in the 1 something range, which also fixed his problem. So that’s three cases.

So, if this is able to help anyone, then I’m glad. I didn’t come to give help. I came to seek it, for which I did receive many ideas. I am really thankful for that.

#8

Funny that “solution” would work for you and your friend, and virtually no one else on the planet that we’re aware of. Have you considered that maybe it’s a problem with your endpoints? They may have something funky in their dial plan that assumes that all extensions will begin with “1”. In fact, if they were marketed for use in the U.S. or Canada, they may by default assume that all numbers starting with 10 or 11 are extensions, while anything starting with 12 through 19 are the start of a 1+area code+number pattern, while anything starting with 2 through 9 is the start of an area code+number pattern. I’ve seen a couple small installations (well under 200 extensions) where that extension numbering pattern (all are four digits starting with 10 or 11) is used.

#9

In my case I used the range 200. It used the same rage when I tested Trixbox and it was working.

Now I installed asterisk+freepbx on my debian server: witch I am using for other Internet services.
I created the 2 extensions and 2 trunks. One trunk has a number for incoming calls; the other trunk is used when calling external numbers.
I can connect to asterisk trough a softphone. I can call external numbers.

My problem is : I cannot call the other extension! I got a speech "All service is busy now. Please try again later"
I think it is a default configuration issue. I installed the system twice and have the same behaviour!

Below the log (using asterisk -rvvvv) when I call an extension from another:
– Executing [[email protected]:1] Macro(“SIP/202-b5f098d8”, “exten-vm|novm|201”) in new stack
– Executing [[email protected]:1] Macro(“SIP/202-b5f098d8”, “user-callerid”) in new stack
– Executing [[email protected]:1] Set(“SIP/202-b5f098d8”, “AMPUSER=202”) in new stack
– Executing [[email protected]:2] GotoIf(“SIP/202-b5f098d8”, “0?report”) in new stack
– Executing [[email protected]:3] ExecIf(“SIP/202-b5f098d8”, “1|Set|REALCALLERIDNUM=202”) in new stack
– Executing [[email protected]:4] Set(“SIP/202-b5f098d8”, “AMPUSER=202”) in new stack
– Executing [[email protected]:5] Set(“SIP/202-b5f098d8”, “AMPUSERCIDNAME=202”) in new stack
– Executing [[email protected]:6] GotoIf(“SIP/202-b5f098d8”, “0?report”) in new stack
– Executing [[email protected]:7] Set(“SIP/202-b5f098d8”, “AMPUSERCID=202”) in new stack
– Executing [[email protected]:8] Set(“SIP/202-b5f098d8”, “CALLERID(all)=“202” <202>”) in new stack
– Executing [[email protected]:9] Set(“SIP/202-b5f098d8”, “REALCALLERIDNUM=202”) in new stack
– Executing [[email protected]:10] GotoIf(“SIP/202-b5f098d8”, “0?continue”) in new stack
– Executing [[email protected]:11] Set(“SIP/202-b5f098d8”, “__TTL=64”) in new stack
– Executing [[email protected]:12] GotoIf(“SIP/202-b5f098d8”, “1?continue”) in new stack
– Goto (macro-user-callerid,s,19)
– Executing [[email protected]:19] NoOp(“SIP/202-b5f098d8”, “Using CallerID “202” <202>”) in new stack
– Executing [[email protected]:2] Set(“SIP/202-b5f098d8”, “RingGroupMethod=none”) in new stack
– Executing [[email protected]:3] Set(“SIP/202-b5f098d8”, “VMBOX=novm”) in new stack
– Executing [[email protected]:4] Set(“SIP/202-b5f098d8”, “EXTTOCALL=201”) in new stack
– Executing [[email protected]:5] Set(“SIP/202-b5f098d8”, “CFUEXT=”) in new stack
– Executing [[email protected]:6] Set(“SIP/202-b5f098d8”, “CFBEXT=”) in new stack
– Executing [[email protected]:7] Set(“SIP/202-b5f098d8”, “RT=”"") in new stack
– Executing [[email protected]:8] Macro(“SIP/202-b5f098d8”, “record-enable|201|IN”) in new stack
– Executing [[email protected]:1] GotoIf(“SIP/202-b5f098d8”, “1?check”) in new stack
– Goto (macro-record-enable,s,4)
– Executing [[email protected]:4] AGI(“SIP/202-b5f098d8”, “recordingcheck|20090802-131622|asterisk-1249211782.95”) in new stack
– Launched AGI Script /usr/share/asterisk/agi-bin/recordingcheck
== recordingcheck|20090802-131622|asterisk-1249211782.95: Failed to execute ‘/usr/share/asterisk/agi-bin/recordingcheck’: No such file or directory
– Executing [[email protected]:5] MacroExit(“SIP/202-b5f098d8”, “”) in new stack
– Executing [[email protected]:9] Macro(“SIP/202-b5f098d8”, “dial||tr|201”) in new stack
– Executing [[email protected]:1] GotoIf(“SIP/202-b5f098d8”, “1?dial”) in new stack
– Goto (macro-dial,s,3)
– Executing [[email protected]:3] AGI(“SIP/202-b5f098d8”, “dialparties.agi”) in new stack
– Launched AGI Script /usr/share/asterisk/agi-bin/dialparties.agi
== dialparties.agi: Failed to execute ‘/usr/share/asterisk/agi-bin/dialparties.agi’: No such file or directory
– Executing [[email protected]:4] NoOp(“SIP/202-b5f098d8”, "Returned from dialparties with no extensions to call and DIALSTATUS: ") in new stack
– Executing [[email protected]:10] GotoIf(“SIP/202-b5f098d8”, “0?exit|return”) in new stack
– Executing [[email protected]:11] Set(“SIP/202-b5f098d8”, “SV_DIALSTATUS=”) in new stack
– Executing [[email protected]:12] GosubIf(“SIP/202-b5f098d8”, “0?docfu|1”) in new stack
– Executing [[email protected]:13] GosubIf(“SIP/202-b5f098d8”, “0?docfb|1”) in new stack
– Executing [[email protected]:14] Set(“SIP/202-b5f098d8”, “DIALSTATUS=”) in new stack
– Executing [[email protected]:15] NoOp(“SIP/202-b5f098d8”, “Voicemail is novm”) in new stack
– Executing [[email protected]:16] GotoIf(“SIP/202-b5f098d8”, “1?s-|1”) in new stack
– Goto (macro-exten-vm,s-,1)
– Executing [[email protected]:2] Goto(“SIP/202-b5f098d8”, “|return|1”) in new stack
– Goto (from-internal,return,1)
– Executing [[email protected]:1] Macro(“SIP/202-b5f098d8”, “user-callerid|SKIPTTL|”) in new stack
– Executing [[email protected]:1] Set(“SIP/202-b5f098d8”, “AMPUSER=202”) in new stack
– Executing [[email protected]:2] GotoIf(“SIP/202-b5f098d8”, “0?report”) in new stack
– Executing [[email protected]:3] ExecIf(“SIP/202-b5f098d8”, “0|Set|REALCALLERIDNUM=202”) in new stack
– Executing [[email protected]:4] Set(“SIP/202-b5f098d8”, “AMPUSER=202”) in new stack
– Executing [[email protected]:5] Set(“SIP/202-b5f098d8”, “AMPUSERCIDNAME=202”) in new stack
– Executing [[email protected]:6] GotoIf(“SIP/202-b5f098d8”, “0?report”) in new stack
– Executing [[email protected]:7] Set(“SIP/202-b5f098d8”, “AMPUSERCID=202”) in new stack
– Executing [[email protected]:8] Set(“SIP/202-b5f098d8”, “CALLERID(all)=“202” <202>”) in new stack
– Executing [[email protected]:9] Set(“SIP/202-b5f098d8”, “REALCALLERIDNUM=202”) in new stack
– Executing [[email protected]:10] GotoIf(“SIP/202-b5f098d8”, “1?continue”) in new stack
– Goto (macro-user-callerid,s,19)
– Executing [[email protected]:19] NoOp(“SIP/202-b5f098d8”, “Using CallerID “202” <202>”) in new stack
– Executing [[email protected]:2] Set(“SIP/202-b5f098d8”, “_NODEST=”) in new stack
– Executing [[email protected]:3] Macro(“SIP/202-b5f098d8”, “record-enable|202|OUT|”) in new stack
– Executing [[email protected]:1] GotoIf(“SIP/202-b5f098d8”, “1?check”) in new stack
– Goto (macro-record-enable,s,4)
– Executing [[email protected]:4] AGI(“SIP/202-b5f098d8”, “recordingcheck|20090802-131622|asterisk-1249211782.95”) in new stack
– Launched AGI Script /usr/share/asterisk/agi-bin/recordingcheck
== recordingcheck|20090802-131622|asterisk-1249211782.95: Failed to execute ‘/usr/share/asterisk/agi-bin/recordingcheck’: No such file or directory
– Executing [[email protected]:5] MacroExit(“SIP/202-b5f098d8”, “”) in new stack
– Executing [[email protected]:4] Macro(“SIP/202-b5f098d8”, “dialout-trunk|5|return||”) in new stack
– Executing [[email protected]:1] Set(“SIP/202-b5f098d8”, “DIAL_TRUNK=5”) in new stack
– Executing [[email protected]:2] GosubIf(“SIP/202-b5f098d8”, “0?sub-pincheck|s|1”) in new stack
– Executing [[email protected]:3] GotoIf(“SIP/202-b5f098d8”, “0?disabletrunk|1”) in new stack
– Executing [[email protected]:4] Set(“SIP/202-b5f098d8”, “DIAL_NUMBER=return”) in new stack
– Executing [[email protected]:5] Set(“SIP/202-b5f098d8”, “DIAL_TRUNK_OPTIONS=tr”) in new stack
– Executing [[email protected]:6] Set(“SIP/202-b5f098d8”, “OUTBOUND_GROUP=OUT_5”) in new stack
– Executing [[email protected]:7] GotoIf(“SIP/202-b5f098d8”, “1?nomax”) in new stack
– Goto (macro-dialout-trunk,s,9)
– Executing [[email protected]:9] GotoIf(“SIP/202-b5f098d8”, “0?skipoutcid”) in new stack
– Executing [[email protected]:10] Set(“SIP/202-b5f098d8”, “DIAL_TRUNK_OPTIONS=”) in new stack
– Executing [[email protected]:11] Macro(“SIP/202-b5f098d8”, “outbound-callerid|5”) in new stack
– Executing [[email protected]:1] ExecIf(“SIP/202-b5f098d8”, “0|SetCallerPres|”) in new stack
– Executing [[email protected]:2] ExecIf(“SIP/202-b5f098d8”, “0|Set|REALCALLERIDNUM=202”) in new stack
– Executing [[email protected]:3] GotoIf(“SIP/202-b5f098d8”, “1?normcid”) in new stack
– Goto (macro-outbound-callerid,s,6)
– Executing [[email protected]:6] Set(“SIP/202-b5f098d8”, “USEROUTCID=”) in new stack
– Executing [[email protected]:7] Set(“SIP/202-b5f098d8”, “EMERGENCYCID=”) in new stack
– Executing [[email protected]:8] Set(“SIP/202-b5f098d8”, “TRUNKOUTCID=”) in new stack
– Executing [[email protected]:9] GotoIf(“SIP/202-b5f098d8”, “1?trunkcid”) in new stack
– Goto (macro-outbound-callerid,s,12)
– Executing [[email protected]:12] ExecIf(“SIP/202-b5f098d8”, “0|Set|CALLERID(all)=”) in new stack
– Executing [[email protected]:13] ExecIf(“SIP/202-b5f098d8”, “0|Set|CALLERID(all)=”) in new stack
– Executing [[email protected]:14] ExecIf(“SIP/202-b5f098d8”, “0|SetCallerPres|prohib_passed_screen”) in new stack
– Executing [[email protected]:12] ExecIf(“SIP/202-b5f098d8”, “0|AGI|fixlocalprefix”) in new stack
– Executing [[email protected]:13] Set(“SIP/202-b5f098d8”, “OUTNUM=return”) in new stack
– Executing [[email protected]:14] Set(“SIP/202-b5f098d8”, “custom=SIP/voipbuster”) in new stack
– Executing [[email protected]:15] ExecIf(“SIP/202-b5f098d8”, “0|Set|DIAL_TRUNK_OPTIONS=M(setmusic^)”) in new stack
– Executing [[email protected]:16] Macro(“SIP/202-b5f098d8”, “dialout-trunk-predial-hook|”) in new stack
– Executing [[email protected]:1] MacroExit(“SIP/202-b5f098d8”, “”) in new stack
– Executing [[email protected]:17] GotoIf(“SIP/202-b5f098d8”, “0?bypass|1”) in new stack
– Executing [[email protected]:18] GotoIf(“SIP/202-b5f098d8”, “0?customtrunk”) in new stack
– Executing [[email protected]:19] Dial(“SIP/202-b5f098d8”, “SIP/voipbuster/return|300|”) in new stack
– Called voipbuster/return
– SIP/voipbuster-09e4fd48 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
– Executing [[email protected]:20] Goto(“SIP/202-b5f098d8”, “s-CONGESTION|1”) in new stack
– Goto (macro-dialout-trunk,s-CONGESTION,1)
– Executing [[email protected]:1] GotoIf(“SIP/202-b5f098d8”, “1?noreport”) in new stack
– Goto (macro-dialout-trunk,s-CONGESTION,3)
– Executing [[email protected]:3] NoOp(“SIP/202-b5f098d8”, “TRUNK Dial failed due to CONGESTION - failing through to other trunks”) in new stack
– Executing [[email protected]:5] Macro(“SIP/202-b5f098d8”, “dialout-trunk|2|return||”) in new stack
– Executing [[email protected]:1] Set(“SIP/202-b5f098d8”, “DIAL_TRUNK=2”) in new stack
– Executing [[email protected]:2] GosubIf(“SIP/202-b5f098d8”, “0?sub-pincheck|s|1”) in new stack
– Executing [[email protected]:3] GotoIf(“SIP/202-b5f098d8”, “0?disabletrunk|1”) in new stack
– Executing [[email protected]:4] Set(“SIP/202-b5f098d8”, “DIAL_NUMBER=return”) in new stack
– Executing [[email protected]:5] Set(“SIP/202-b5f098d8”, “DIAL_TRUNK_OPTIONS=tr”) in new stack
– Executing [[email protected]:6] Set(“SIP/202-b5f098d8”, “OUTBOUND_GROUP=OUT_2”) in new stack
– Executing [[email protected]:7] GotoIf(“SIP/202-b5f098d8”, “1?nomax”) in new stack
– Goto (macro-dialout-trunk,s,9)
– Executing [[email protected]:9] GotoIf(“SIP/202-b5f098d8”, “0?skipoutcid”) in new stack
– Executing [[email protected]:10] Set(“SIP/202-b5f098d8”, “DIAL_TRUNK_OPTIONS=”) in new stack
– Executing [[email protected]:11] Macro(“SIP/202-b5f098d8”, “outbound-callerid|2”) in new stack
– Executing [[email protected]:1] ExecIf(“SIP/202-b5f098d8”, “0|SetCallerPres|”) in new stack
– Executing [[email protected]:2] ExecIf(“SIP/202-b5f098d8”, “0|Set|REALCALLERIDNUM=202”) in new stack
– Executing [[email protected]:3] GotoIf(“SIP/202-b5f098d8”, “1?normcid”) in new stack
– Goto (macro-outbound-callerid,s,6)
– Executing [[email protected]:6] Set(“SIP/202-b5f098d8”, “USEROUTCID=”) in new stack
– Executing [[email protected]:7] Set(“SIP/202-b5f098d8”, “EMERGENCYCID=”) in new stack
– Executing [[email protected]:8] Set(“SIP/202-b5f098d8”, “TRUNKOUTCID=”) in new stack
– Executing [[email protected]:9] GotoIf(“SIP/202-b5f098d8”, “1?trunkcid”) in new stack
– Goto (macro-outbound-callerid,s,12)
– Executing [[email protected]:12] ExecIf(“SIP/202-b5f098d8”, “0|Set|CALLERID(all)=”) in new stack
– Executing [[email protected]:13] ExecIf(“SIP/202-b5f098d8”, “0|Set|CALLERID(all)=”) in new stack
– Executing [[email protected]:14] ExecIf(“SIP/202-b5f098d8”, “0|SetCallerPres|prohib_passed_screen”) in new stack
– Executing [[email protected]:12] ExecIf(“SIP/202-b5f098d8”, “0|AGI|fixlocalprefix”) in new stack
– Executing [[email protected]:13] Set(“SIP/202-b5f098d8”, “OUTNUM=return”) in new stack
– Executing [[email protected]:14] Set(“SIP/202-b5f098d8”, “custom=SIP/freephonie”) in new stack
– Executing [[email protected]:15] ExecIf(“SIP/202-b5f098d8”, “0|Set|DIAL_TRUNK_OPTIONS=M(setmusic^)”) in new stack
– Executing [[email protected]:16] Macro(“SIP/202-b5f098d8”, “dialout-trunk-predial-hook|”) in new stack
– Executing [[email protected]:1] MacroExit(“SIP/202-b5f098d8”, “”) in new stack
– Executing [[email protected]:17] GotoIf(“SIP/202-b5f098d8”, “0?bypass|1”) in new stack
– Executing [[email protected]:18] GotoIf(“SIP/202-b5f098d8”, “0?customtrunk”) in new stack
– Executing [[email protected]:19] Dial(“SIP/202-b5f098d8”, “SIP/freephonie/return|300|”) in new stack
– Called freephonie/return
– SIP/freephonie-09e5e4c0 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
– Executing [[email protected]:20] Goto(“SIP/202-b5f098d8”, “s-CONGESTION|1”) in new stack
– Goto (macro-dialout-trunk,s-CONGESTION,1)
– Executing [[email protected]:1] GotoIf(“SIP/202-b5f098d8”, “1?noreport”) in new stack
– Goto (macro-dialout-trunk,s-CONGESTION,3)
– Executing [[email protected]:3] NoOp(“SIP/202-b5f098d8”, “TRUNK Dial failed due to CONGESTION - failing through to other trunks”) in new stack
– Executing [[email protected]:6] Macro(“SIP/202-b5f098d8”, “dialout-trunk|1|return||”) in new stack
– Executing [[email protected]:1] Set(“SIP/202-b5f098d8”, “DIAL_TRUNK=1”) in new stack
– Executing [[email protected]:2] GosubIf(“SIP/202-b5f098d8”, “0?sub-pincheck|s|1”) in new stack
– Executing [[email protected]:3] GotoIf(“SIP/202-b5f098d8”, “0?disabletrunk|1”) in new stack
– Executing [[email protected]:4] Set(“SIP/202-b5f098d8”, “DIAL_NUMBER=return”) in new stack
– Executing [[email protected]:5] Set(“SIP/202-b5f098d8”, “DIAL_TRUNK_OPTIONS=tr”) in new stack
– Executing [[email protected]:6] Set(“SIP/202-b5f098d8”, “OUTBOUND_GROUP=OUT_1”) in new stack
– Executing [[email protected]:7] GotoIf(“SIP/202-b5f098d8”, “1?nomax”) in new stack
– Goto (macro-dialout-trunk,s,9)
– Executing [[email protected]:9] GotoIf(“SIP/202-b5f098d8”, “0?skipoutcid”) in new stack
– Executing [[email protected]:10] Set(“SIP/202-b5f098d8”, “DIAL_TRUNK_OPTIONS=”) in new stack
– Executing [[email protected]:11] Macro(“SIP/202-b5f098d8”, “outbound-callerid|1”) in new stack
– Executing [[email protected]:1] ExecIf(“SIP/202-b5f098d8”, “0|SetCallerPres|”) in new stack
– Executing [[email protected]:2] ExecIf(“SIP/202-b5f098d8”, “0|Set|REALCALLERIDNUM=202”) in new stack
– Executing [[email protected]:3] GotoIf(“SIP/202-b5f098d8”, “1?normcid”) in new stack
– Goto (macro-outbound-callerid,s,6)
– Executing [[email protected]:6] Set(“SIP/202-b5f098d8”, “USEROUTCID=”) in new stack
– Executing [[email protected]:7] Set(“SIP/202-b5f098d8”, “EMERGENCYCID=”) in new stack
– Executing [[email protected]:8] Set(“SIP/202-b5f098d8”, “TRUNKOUTCID=”) in new stack
– Executing [[email protected]:9] GotoIf(“SIP/202-b5f098d8”, “1?trunkcid”) in new stack
– Goto (macro-outbound-callerid,s,12)
– Executing [[email protected]:12] ExecIf(“SIP/202-b5f098d8”, “0|Set|CALLERID(all)=”) in new stack
– Executing [[email protected]:13] ExecIf(“SIP/202-b5f098d8”, “0|Set|CALLERID(all)=”) in new stack
– Executing [[email protected]:14] ExecIf(“SIP/202-b5f098d8”, “0|SetCallerPres|prohib_passed_screen”) in new stack
– Executing [[email protected]:12] ExecIf(“SIP/202-b5f098d8”, “0|AGI|fixlocalprefix”) in new stack
– Executing [[email protected]:13] Set(“SIP/202-b5f098d8”, “OUTNUM=return”) in new stack
– Executing [[email protected]:14] Set(“SIP/202-b5f098d8”, “custom=ZAP/g0”) in new stack
– Executing [[email protected]:15] ExecIf(“SIP/202-b5f098d8”, “0|Set|DIAL_TRUNK_OPTIONS=M(setmusic^)”) in new stack
– Executing [[email protected]:16] Macro(“SIP/202-b5f098d8”, “dialout-trunk-predial-hook|”) in new stack
– Executing [[email protected]:1] MacroExit(“SIP/202-b5f098d8”, “”) in new stack
– Executing [[email protected]:17] GotoIf(“SIP/202-b5f098d8”, “0?bypass|1”) in new stack
– Executing [[email protected]:18] GotoIf(“SIP/202-b5f098d8”, “0?customtrunk”) in new stack
– Executing [[email protected]:19] Dial(“SIP/202-b5f098d8”, “ZAP/g0/return|300|”) in new stack
== Everyone is busy/congested at this time (1:0/1/0)
– Executing [[email protected]:20] Goto(“SIP/202-b5f098d8”, “s-CONGESTION|1”) in new stack
– Goto (macro-dialout-trunk,s-CONGESTION,1)
– Executing [[email protected]:1] GotoIf(“SIP/202-b5f098d8”, “1?noreport”) in new stack
– Goto (macro-dialout-trunk,s-CONGESTION,3)
– Executing [[email protected]:3] NoOp(“SIP/202-b5f098d8”, “TRUNK Dial failed due to CONGESTION - failing through to other trunks”) in new stack
– Executing [[email protected]:7] Macro(“SIP/202-b5f098d8”, “outisbusy|”) in new stack
– Executing [[email protected]:1] Playback(“SIP/202-b5f098d8”, “all-circuits-busy-now|noanswer”) in new stack
– <SIP/202-b5f098d8> Playing ‘all-circuits-busy-now’ (language ‘en’)
– Executing [[email protected]:2] Playback(“SIP/202-b5f098d8”, “pls-try-call-later|noanswer”) in new stack
– <SIP/202-b5f098d8> Playing ‘pls-try-call-later’ (language ‘en’)
== Spawn extension (macro-outisbusy, s, 2) exited non-zero on ‘SIP/202-b5f098d8’ in macro ‘outisbusy’
== Spawn extension (macro-outisbusy, s, 2) exited non-zero on ‘SIP/202-b5f098d8’

Thanks for your help

#10

I’m sorry if this is no enough information… I have the same problem: it’s a new installation. I’ve removed my test extensions and created just two new. As far as I can tell, the calling extension is searching the other extension via the openvox d130p… This is what a I get:

#asterisk -rvvvv
Asterisk 1.8.11-cert7, Copyright © 1999 - 2012 Digium, Inc. and others.
Created by Mark Spencer [email protected]
Asterisk comes with ABSOLUTELY NO WARRANTY; type ‘core show warranty’ for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type ‘core show license’ for details.

== Parsing ‘/etc/asterisk/asterisk.conf’: == Found
== Parsing ‘/etc/asterisk/extconfig.conf’: == Found
Connected to Asterisk 1.8.11-cert7 currently running on asterisk (pid = 5131)
Verbosity was 3 and is now 4
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
– Executing [[email protected]:1] Dial(“SIP/1234-00000001”, “dahdi/g0/1200”) in new stack
[2012-10-26 08:22:41] WARNING[6464]: channel.c:5621 ast_request: No channel type registered for ‘dahdi’
[2012-10-26 08:22:41] WARNING[6464]: app_dial.c:2218 dial_exec_full: Unable to create channel of type ‘dahdi’ (cause 66 - Channel not implemented)
== Everyone is busy/congested at this time (1:0/0/1)
– Executing [[email protected]:2] Hangup(“SIP/1234-00000001”, “”) in new stack
== Spawn extension (from-internal, 1200, 2) exited non-zero on ‘SIP/1234-00000001’
– Executing [[email protected]:1] Hangup(“SIP/1234-00000001”, “”) in new stack
== Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/1234-00000001’
– Unregistered SIP ‘1234’

#11