How to call extension no. of my old PBX systems

Hello The Expert,

Now I am trying to call extension of my old PBX (NEC) …
I have configured Outbound routes below:

Route Name: 7_oldpbx

Dial Patterns: 7|XXX
Trunk Sequence -> Zap/g1

dial 7300 (my extension no. is 300 on my old PBX systems) still not working …

please help

Thanks

Do you know what a butt set is? They have a switch on the side, off hook and monitor mode. Monitor lets you bridge on the line.

Hi SkykingOH… thanks…

it works I already… this is because I missed the Trunk configurations

Regards

Well, you must have setup Group 1 because it clearly shows the channel being dialed out on:

- Called g1/250
-- DAHDI/1-1 answered SIP/250-00000023

I would hook a butt set up in monitor mode and listen to what is going on.

hmm … sorry, I am very new with this, could you please advise how to change it to monitor mode?

thanks a lot in advance

Regards

thanks,
I am running AsteriskNow 1.7.1 and FreePBX 2.7.0.10

my config are as follow:

cat [[email protected] ~]# cat /proc/dahdi/1
Span 1: WCTDM/4 “Wildcard S400P Prototype Board 5” (MASTER)

1 WCTDM/4/0 FXSKS (In use)
2 WCTDM/4/1 FXSKS (In use) RED
3 WCTDM/4/2 FXOKS (In use)
4 WCTDM/4/3 FXOKS (In use)

my “7_oldpbx” outbound routes:

Dial patterns: 7|XXX
Trunkg Sequence: ZAP/g1

from my extension no. 251 (this is old PBX) I plug in my extension wire into FXO port
by using X-Lite I try to call other extension 300 (old PBX) … X-Lite shown “Call was established” but extension 300 does not ring…

what I missed?

Please help

Thanks & Regards
Winanjaya

I already told you what to look at. Is your FXO port in group 1?

Send output of ‘dahdi show channels’ and a short log clip when you try and dial.

hmm…below what I have:

asterisk*CLI> dahdi show channels
Chan Extension Context Language MOH Interpret Blocked State
pseudo default default In Service
1 from-zaptel default In Service
2 from-zaptel default In Service
3 from-analog default In Service
4 from-analog default In Service

I am trying to call extension 250 (my old PBX), from X-Lite it shown “Call established” but the ext. 250 does not ring

[[email protected] ~]# asterisk -r
Asterisk 1.6.2.11, Copyright © 1999 - 2010 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.

Connected to Asterisk 1.6.2.11 currently running on asterisk (pid = 2713)
Verbosity is at least 3
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
– Executing [[email protected]:1] Macro(“SIP/250-00000023”, “user-callerid,SKIPTTL,”) in new stack
– Executing [[email protected]:1] Set(“SIP/250-00000023”, “AMPUSER=250”) in new stack
– Executing [[email protected]:2] GotoIf(“SIP/250-00000023”, “0?report”) in new stack
– Executing [[email protected]:3] ExecIf(“SIP/250-00000023”, “1?Set(REALCALLERIDNUM=250)”) in new stack
– Executing [[email protected]:4] Set(“SIP/250-00000023”, “AMPUSER=250”) in new stack
– Executing [[email protected]:5] Set(“SIP/250-00000023”, “AMPUSERCIDNAME=winanjaya”) in new stack
– Executing [[email protected]:6] GotoIf(“SIP/250-00000023”, “0?report”) in new stack
– Executing [[email protected]:7] Set(“SIP/250-00000023”, “AMPUSERCID=250”) in new stack
– Executing [[email protected]:8] Set(“SIP/250-00000023”, “CALLERID(all)=“winanjaya” <250>”) in new stack
– Executing [[email protected]:9] ExecIf(“SIP/250-00000023”, “0?Set(CHANNEL(language)=)”) in new stack
– Executing [[email protected]:10] GotoIf(“SIP/250-00000023”, “1?continue”) in new stack
– Goto (macro-user-callerid,s,19)
– Executing [[email protected]:19] Set(“SIP/250-00000023”, “CALLERID(number)=250”) in new stack
– Executing [[email protected]:20] Set(“SIP/250-00000023”, “CALLERID(name)=winanjaya”) in new stack
– Executing [[email protected]:21] NoOp(“SIP/250-00000023”, “Using CallerID “winanjaya” <250>”) in new stack
– Executing [[email protected]:2] Set(“SIP/250-00000023”, “_NODEST=”) in new stack
– Executing [[email protected]:3] Macro(“SIP/250-00000023”, “record-enable,250,OUT,”) in new stack
– Executing [[email protected]:1] GotoIf(“SIP/250-00000023”, “1?check”) in new stack
– Goto (macro-record-enable,s,4)
– Executing [[email protected]:4] ExecIf(“SIP/250-00000023”, “0?MacroExit()”) in new stack
– Executing [[email protected]:5] GotoIf(“SIP/250-00000023”, “0?Group:OUT”) in new stack
– Goto (macro-record-enable,s,15)
– Executing [[email protected]:15] GotoIf(“SIP/250-00000023”, “0?IN”) in new stack
– Executing [[email protected]:16] ExecIf(“SIP/250-00000023”, “1?MacroExit()”) in new stack
– Executing [[email protected]:4] Macro(“SIP/250-00000023”, “dialout-trunk,2,250,”) in new stack
– Executing [[email protected]:1] Set(“SIP/250-00000023”, “DIAL_TRUNK=2”) in new stack
– Executing [[email protected]:2] GosubIf(“SIP/250-00000023”, “0?sub-pincheck,s,1”) in new stack
– Executing [[email protected]:3] GotoIf(“SIP/250-00000023”, “0?disabletrunk,1”) in new stack
– Executing [[email protected]:4] Set(“SIP/250-00000023”, “DIAL_NUMBER=250”) in new stack
– Executing [[email protected]:5] Set(“SIP/250-00000023”, “DIAL_TRUNK_OPTIONS=tr”) in new stack
– Executing [[email protected]:6] Set(“SIP/250-00000023”, “OUTBOUND_GROUP=OUT_2”) in new stack
– Executing [[email protected]:7] GotoIf(“SIP/250-00000023”, “1?nomax”) in new stack
– Goto (macro-dialout-trunk,s,9)
– Executing [[email protected]:9] GotoIf(“SIP/250-00000023”, “0?skipoutcid”) in new stack
– Executing [[email protected]:10] Set(“SIP/250-00000023”, “DIAL_TRUNK_OPTIONS=”) in new stack
– Executing [[email protected]:11] Macro(“SIP/250-00000023”, “outbound-callerid,2”) in new stack
– Executing [[email protected]:1] ExecIf(“SIP/250-00000023”, “0?Set(CALLERPRES()=)”) in new stack
– Executing [[email protected]:2] ExecIf(“SIP/250-00000023”, “0?Set(REALCALLERIDNUM=250)”) in new stack
– Executing [[email protected]:3] GotoIf(“SIP/250-00000023”, “1?normcid”) in new stack
– Goto (macro-outbound-callerid,s,6)
– Executing [[email protected]:6] Set(“SIP/250-00000023”, “USEROUTCID=”) in new stack
– Executing [[email protected]:7] Set(“SIP/250-00000023”, “EMERGENCYCID=”) in new stack
– Executing [[email protected]:8] Set(“SIP/250-00000023”, “TRUNKOUTCID=g1”) in new stack
– Executing [[email protected]:9] GotoIf(“SIP/250-00000023”, “1?trunkcid”) in new stack
– Goto (macro-outbound-callerid,s,12)
– Executing [[email protected]:12] ExecIf(“SIP/250-00000023”, “1?Set(CALLERID(all)=g1)”) in new stack
– Executing [[email protected]:13] ExecIf(“SIP/250-00000023”, “0?Set(CALLERID(all)=)”) in new stack
– Executing [[email protected]:14] ExecIf(“SIP/250-00000023”, “0?Set(CALLERID(all)=)”) in new stack
– Executing [[email protected]:15] ExecIf(“SIP/250-00000023”, “0?Set(CALLERPRES()=prohib_passed_screen)”) in new stack
– Executing [[email protected]:12] ExecIf(“SIP/250-00000023”, “1?AGI(fixlocalprefix)”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
– <SIP/250-00000023>AGI Script fixlocalprefix completed, returning 0
– Executing [[email protected]:13] Set(“SIP/250-00000023”, “OUTNUM=250”) in new stack
– Executing [[email protected]:14] Set(“SIP/250-00000023”, “custom=DAHDI/g1”) in new stack
– Executing [[email protected]:15] ExecIf(“SIP/250-00000023”, “0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^))”) in new stack
– Executing [[email protected]:16] Macro(“SIP/250-00000023”, “dialout-trunk-predial-hook,”) in new stack
– Executing [[email protected]:1] MacroExit(“SIP/250-00000023”, “”) in new stack
– Executing [[email protected]:17] GotoIf(“SIP/250-00000023”, “0?bypass,1”) in new stack
– Executing [[email protected]:18] GotoIf(“SIP/250-00000023”, “0?customtrunk”) in new stack
– Executing [[email protected]:19] Dial(“SIP/250-00000023”, “DAHDI/g1/250,300,”) in new stack
– Called g1/250
– DAHDI/1-1 answered SIP/250-00000023
– Executing [[email protected]:1] Macro(“SIP/250-00000023”, “hangupcall,”) in new stack
– Executing [[email protected]:1] GotoIf(“SIP/250-00000023”, “1?skiprg”) in new stack
– Goto (macro-hangupcall,s,4)
– Executing [[email protected]:4] GotoIf(“SIP/250-00000023”, “1?skipblkvm”) in new stack
– Goto (macro-hangupcall,s,7)
– Executing [[email protected]:7] GotoIf(“SIP/250-00000023”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,9)
– Executing [[email protected]:9] Hangup(“SIP/250-00000023”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on ‘SIP/250-00000023’ in macro ‘hangupcall’
– Hungup ‘DAHDI/1-1’
== Spawn extension (macro-dialout-trunk, s, 19) exited non-zero on ‘SIP/250-00000023’ in macro ‘dialout-trunk’
== Spawn extension (from-internal, 7250, 4) exited non-zero on 'SIP/250-00000023’
asterisk*CLI>

That sure is not much information. Why g1? Did you change your DAHDI config for the channels to group 1?

Is g1 a PRI? Is the interface on PBX configured for 3 digit feed?

At a more simple basis are the trunks in service to the NEC?

Lastly, please include the proper information when requesting help such as how the system was installed (bare metal or a distro, if a distro specify). All software versions must be supplied.