No inbound or outbound calls

I just installed the distro for freepbx. Running 1.8 and 2.10. This is a fresh install. The DAHDI package is installed and the TDM400 card registers on inital load. I can not dial out or get incoming calls. In fact it doesnt even show up on the log files for the incoming calls. Here is the log.

[2012-04-13 01:04:46] VERBOSE[4900] config.c: == Parsing ‘/etc/asterisk/sip_custom.conf’: [2012-04-13 01:04:46] VERBOSE[4900] config.c: == Found
[2012-04-13 01:04:46] WARNING[4923] chan_iax2.c: Error opening firmware directory ‘/var/lib/asterisk/firmware/iax’: No such file or directory
[2012-04-13 01:04:46] NOTICE[4923] iax2-provision.c: No IAX provisioning configuration found, IAX provisioning disabled.
[2012-04-13 01:04:46] VERBOSE[4900] config.c: == Parsing ‘/etc/asterisk/sip_additional.conf’: [2012-04-13 01:04:46] VERBOSE[4923] loader.c: – Reloading module ‘res_rtp_asterisk.so’ (Asterisk RTP Stack)
[2012-04-13 01:04:46] VERBOSE[4900] config.c: == Found
[2012-04-13 01:04:46] VERBOSE[4923] loader.c: – Reloading module ‘codec_dahdi.so’ (Generic DAHDI Transcoder Codec Translator)
[2012-04-13 01:04:46] VERBOSE[4923] loader.c: – Reloading module ‘codec_lpc10.so’ (LPC10 2.4kbps Coder/Decoder)
[2012-04-13 01:04:46] VERBOSE[4923] loader.c: – Reloading module ‘codec_g722.so’ (ITU G.722-64kbps G722 Transcoder)
[2012-04-13 01:04:46] VERBOSE[4923] loader.c: – Reloading module ‘codec_alaw.so’ (A-law Coder/Decoder)
[2012-04-13 01:04:46] VERBOSE[4923] loader.c: – Reloading module ‘cdr_custom.so’ (Customizable Comma Separated Values CDR Backend)
[2012-04-13 01:04:46] VERBOSE[4900] config.c: == Parsing ‘/etc/asterisk/sip_custom_post.conf’: [2012-04-13 01:04:46] ERROR[4923] cdr_custom.c: Unable to load cdr_custom.conf. Not logging custom CSV CDRs.
[2012-04-13 01:04:46] VERBOSE[4923] loader.c: – Reloading module ‘cel_custom.so’ (Customizable Comma Separated Values CEL Backend)
[2012-04-13 01:04:46] VERBOSE[4900] config.c: == Found
[2012-04-13 01:04:46] ERROR[4923] cel_custom.c: Unable to load cel_custom.conf. Not logging CEL to custom CSVs.
[2012-04-13 01:04:46] VERBOSE[4923] loader.c: – Reloading module ‘res_config_curl.so’ (Realtime Curl configuration)
[2012-04-13 01:04:46] VERBOSE[4923] loader.c: – Reloading module ‘res_config_ldap.so’ (LDAP realtime interface)
[2012-04-13 01:04:46] WARNING[4900] chan_sip.c: No valid transports available, falling back to ‘udp’.
[2012-04-13 01:04:46] ERROR[4923] res_config_ldap.c: Cannot load configuration file: res_ldap.conf
[2012-04-13 01:04:46] NOTICE[4923] res_config_ldap.c: Cannot reload LDAP RealTime driver.
[2012-04-13 01:04:46] VERBOSE[4923] loader.c: – Reloading module ‘codec_gsm.so’ (GSM Coder/Decoder)
[2012-04-13 01:04:46] VERBOSE[4923] loader.c: – Reloading module ‘chan_dahdi.so’ (DAHDI Telephony Driver w/PRI & SS7 & MFC/R2)
[2012-04-13 01:04:46] VERBOSE[4900] netsock2.c: == Using SIP TOS bits 96
[2012-04-13 01:04:46] VERBOSE[4923] config.c: == Parsing ‘/etc/asterisk/chan_dahdi.conf’: [2012-04-13 01:04:46] VERBOSE[4900] netsock2.c: == Using SIP CoS mark 4
[2012-04-13 01:04:46] VERBOSE[4923] config.c: == Found
[2012-04-13 01:04:46] VERBOSE[4923] config.c: == Parsing ‘/etc/asterisk/dahdi-channels.conf’: [2012-04-13 01:04:46] VERBOSE[4923] config.c: == Found
[2012-04-13 01:04:46] VERBOSE[4923] config.c: == Parsing ‘/etc/asterisk/chan_dahdi_additional.conf’: [2012-04-13 01:04:46] VERBOSE[4923] config.c: == Found
[2012-04-13 01:04:46] WARNING[4923] chan_dahdi.c: Ignoring any changes to ‘signalling’ (on reload) at line 12.
[2012-04-13 01:04:46] WARNING[4923] chan_dahdi.c: Ignoring any changes to ‘rxwink’ (on reload) at line 13.
[2012-04-13 01:04:46] WARNING[4923] chan_dahdi.c: Ignoring any changes to ‘signalling’ (on reload) at line 12.
[2012-04-13 01:04:46] VERBOSE[4900] config.c: == Parsing ‘/etc/asterisk/sip_notify.conf’: [2012-04-13 01:04:46] WARNING[4923] chan_dahdi.c: Attempt to configure channel 1 with signaling Unknown signalling -1 ignored because it is already configured to be FXS Kewlstart.
[2012-04-13 01:04:46] VERBOSE[4900] config.c: == Found
[2012-04-13 01:04:46] VERBOSE[4923] chan_dahdi.c: – Reconfigured channel 1, FXS Kewlstart signalling
[2012-04-13 01:04:46] VERBOSE[4900] config.c: == Parsing ‘/etc/asterisk/sip_notify_custom.conf’: [2012-04-13 01:04:46] DEBUG[4923] chan_dahdi.c: Channel ‘1’ configured.
[2012-04-13 01:04:46] VERBOSE[4900] config.c: == Found
[2012-04-13 01:04:46] WARNING[4923] chan_dahdi.c: Ignoring any changes to ‘signalling’ (on reload) at line 22.
[2012-04-13 01:04:46] WARNING[4923] chan_dahdi.c: Attempt to configure channel 2 with signaling Unknown signalling -1 ignored because it is already configured to be FXS Kewlstart.
[2012-04-13 01:04:46] VERBOSE[4900] config.c: == Parsing ‘/etc/asterisk/sip_notify_additional.conf’: [2012-04-13 01:04:46] WARNING[4923] chan_dahdi.c: Detected alarm on channel 2: Red Alarm
[2012-04-13 01:04:46] VERBOSE[4900] config.c: == Found
[2012-04-13 01:04:46] VERBOSE[4923] chan_dahdi.c: – Reconfigured channel 2, FXS Kewlstart signalling
[2012-04-13 01:04:46] DEBUG[4923] chan_dahdi.c: Channel ‘2’ configured.
[2012-04-13 01:04:46] WARNING[4923] chan_dahdi.c: Ignoring any changes to ‘signalling’ (on reload) at line 32.
[2012-04-13 01:04:46] WARNING[4923] chan_dahdi.c: Attempt to configure channel 3 with signaling Unknown signalling -1 ignored because it is already configured to be FXO Kewlstart.
[2012-04-13 01:04:46] VERBOSE[4923] chan_dahdi.c: – Reconfigured channel 3, FXO Kewlstart signalling
[2012-04-13 01:04:46] DEBUG[4923] chan_dahdi.c: Channel ‘3’ configured.
[2012-04-13 01:04:46] WARNING[4923] chan_dahdi.c: Ignoring any changes to ‘signalling’ (on reload) at line 44.
[2012-04-13 01:04:46] WARNING[4923] chan_dahdi.c: Attempt to configure channel 4 with signaling Unknown signalling -1 ignored because it is already configured to be FXO Kewlstart.
[2012-04-13 01:04:46] VERBOSE[4923] chan_dahdi.c: – Reconfigured channel 4, FXO Kewlstart signalling
[2012-04-13 01:04:46] DEBUG[4923] chan_dahdi.c: Channel ‘4’ configured.
[2012-04-13 01:04:46] VERBOSE[4923] loader.c: – Reloading module ‘codec_adpcm.so’ (Adaptive Differential PCM Coder/Decoder)
[2012-04-13 01:04:46] VERBOSE[4923] loader.c: – Reloading module ‘res_clialiases.so’ (CLI Aliases)
[2012-04-13 01:04:46] ERROR[4923] res_clialiases.c: res_clialiases configuration file ‘cli_aliases.conf’ not found
[2012-04-13 01:04:46] VERBOSE[4923] loader.c: – Reloading module ‘chan_mgcp.so’ (Media Gateway Control Protocol (MGCP))
[2012-04-13 01:04:46] VERBOSE[4915] chan_mgcp.c: Reloading MGCP
[2012-04-13 01:04:46] NOTICE[4915] chan_mgcp.c: Unable to load config mgcp.conf, MGCP disabled

This is what shows up when trying to make an outside call.

[2012-04-13 01:10:17] WARNING[5379] app_dial.c: Unable to create channel of type 'DAHDI' (cause 0 - Unknown)
[2012-04-13 01:10:17] VERBOSE[5379] app_dial.c: == Everyone is busy/congested at this time (1:0/0/1)
[2012-04-13 01:10:17] VERBOSE[5379] pbx.c: -- Executing [[email protected]:23] NoOp("SIP/100-00000001", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 0") in new stack
[2012-04-13 01:10:17] VERBOSE[5379] pbx.c: -- Executing [[email protected]:24] Goto("SIP/100-00000001", "s-CHANUNAVAIL,1") in new stack
[2012-04-13 01:10:17] VERBOSE[5379] pbx.c: -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
[2012-04-13 01:10:17] VERBOSE[5379] pbx.c: -- Executing [[email protected]:1] Set("SIP/100-00000001", "RC=0") in new stack
[2012-04-13 01:10:17] VERBOSE[5379] pbx.c: -- Executing [[email protected]:2] Goto("SIP/100-00000001", "0,1") in new stack
[2012-04-13 01:10:17] VERBOSE[5379] pbx.c: -- Goto (macro-dialout-trunk,0,1)
[2012-04-13 01:10:17] VERBOSE[5379] pbx.c: -- Executing [[email protected]:1] Goto("SIP/100-00000001", "continue,1") in new stack
[2012-04-13 01:10:17] VERBOSE[5379] pbx.c: -- Goto (macro-dialout-trunk,continue,1)
[2012-04-13 01:10:17] VERBOSE[5379] pbx.c: -- Executing [[email protected]:1] GotoIf("SIP/100-00000001", "1?noreport") in new stack
[2012-04-13 01:10:17] VERBOSE[5379] pbx.c: -- Goto (macro-dialout-trunk,continue,3)
[2012-04-13 01:10:17] VERBOSE[5379] pbx.c: -- Executing [[email protected]:3] NoOp("SIP/100-00000001", "TRUNK Dial failed due to CHANUNAVAIL HANGUPCAUSE: 0 - failing through to other trunks") in new stack
[2012-04-13 01:10:17] VERBOSE[5379] pbx.c: -- Executing [[email protected]:4] Set("SIP/100-00000001", "CALLERID(number)=100") in new stack
[2012-04-13 01:10:17] VERBOSE[5379] pbx.c: -- Executing [[email protected]:7] Macro("SIP/100-00000001", "outisbusy,") in new stack
[2012-04-13 01:10:17] VERBOSE[5379] pbx.c: -- Executing [[email protected]:1] Progress("SIP/100-00000001", "") in new stack
[2012-04-13 01:10:17] VERBOSE[5379] pbx.c: -- Executing [[email protected]:2] GotoIf("SIP/100-00000001", "0?emergency,1") in new stack
[2012-04-13 01:10:17] VERBOSE[5379] pbx.c: -- Executing [[email protected]:3] GotoIf("SIP/100-00000001", "0?intracompany,1") in new stack
[2012-04-13 01:10:17] VERBOSE[5379] pbx.c: -- Executing [[email protected]:4] Playback("SIP/100-00000001", "all-circuits-busy-now&pls-try-call-later, noanswer") in new stack
[2012-04-13 01:10:17] VERBOSE[5379] file.c: -- <SIP/100-00000001> Playing 'all-circuits-busy-now.ulaw' (language 'en')
[2012-04-13 01:10:18] VERBOSE[5379] app_macro.c: == Spawn extension (macro-outisbusy, s, 4) exited non-zero on 'SIP/100-00000001' in macro 'outisbusy'
[2012-04-13 01:10:18] VERBOSE[5379] pbx.c: == Spawn extension (from-internal, xxxxxxxxxx, 7) exited non-zero on 'SIP/100-00000001'
[2012-04-13 01:10:18] VERBOSE[5379] pbx.c: -- Executing [[email protected]:1] Hangup("SIP/100-00000001", "") in new stack
[2012-04-13 01:10:18] VERBOSE[5379] pbx.c: == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/100-00000001'

FreePBX is a registered trademark of Bandwidth.com
FreePBX 2.10.0.4 is licensed under GPL
Sponsored by:Bandwidth.com
The FreePBX project is sponsored in part by:
Schmooze Com., Inc.
Proud sponsors, contributors,
and providers of Professional Support & Services

Bump?

The DAHDI channel is unavailable. What is the output of
’dahdi show channels’ from Asterisk?

-bash: dahdi: command not found

Some how I can now get incoming calls but no outbound. Outbounds says all circuits are busy.

It’s an Asterisk command not a Linux program.

That would make more sense. Here it is:
Chan Extension Context Language MOH Interpret Blocked State
pseudo default default In Service
1 from-zaptel en default In Service
2 from-zaptel en default In Service
3 from-internal en default In Service
4 from-internal en default In Service

You need the lines about on the log you sent. You started at can’t create channel. That is last thing I need to see. Need dial command above.

Basically make sure that only port 1 and 2 are in group 0 if your dial command is G0

or you could make two trunks for channel 1 and 2 and put them in the order you want them hunted in the outbound route.

Here is the whole log from the pickup to hangup.

[2012-04-18 14:02:45] VERBOSE[12183] netsock2.c: == Using SIP RTP TOS bits 184
[2012-04-18 14:02:45] VERBOSE[12183] netsock2.c: == Using SIP RTP CoS mark 5
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:1] Macro("SIP/100-00000007", "user-callerid,LIMIT,") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:1] Set("SIP/100-00000007", "AMPUSER=100") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:2] GotoIf("SIP/100-00000007", "0?report") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:3] ExecIf("SIP/100-00000007", "1?Set(REALCALLERIDNUM=100)") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:4] Set("SIP/100-00000007", "AMPUSER=100") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:5] Set("SIP/100-00000007", "AMPUSERCIDNAME=100") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:6] GotoIf("SIP/100-00000007", "0?report") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:7] Set("SIP/100-00000007", "AMPUSERCID=100") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:8] Set("SIP/100-00000007", "CALLERID(all)="100" <100>") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:9] GotoIf("SIP/100-00000007", "0?limit") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:10] ExecIf("SIP/100-00000007", "1?Set(GROUP(concurrency_limit)=100)") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:11] GosubIf("SIP/100-00000007", "0?sub-ccss,s,1(from-internal,xxxxxxxxxx)") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:12] GotoIf("SIP/100-00000007", "1?continue") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Goto (macro-user-callerid,s,25)
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:25] Set("SIP/100-00000007", "CALLERID(number)=100") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:26] Set("SIP/100-00000007", "CALLERID(name)=100") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:27] Set("SIP/100-00000007", "CHANNEL(language)=en") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:2] Set("SIP/100-00000007", "MOHCLASS=default") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:3] Set("SIP/100-00000007", "_NODEST=") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:4] Gosub("SIP/100-00000007", "sub-record-check,s,1(out,xxxxxxxxxx,)") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:1] GotoIf("SIP/100-00000007", "1?check") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Goto (sub-record-check,s,3)
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:3] Set("SIP/100-00000007", "MON_FMT=wav") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:4] GotoIf("SIP/100-00000007", "1?next") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Goto (sub-record-check,s,7)
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:7] ExecIf("SIP/100-00000007", "0?Return()") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:8] GotoIf("SIP/100-00000007", "0?out,1") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:9] Set("SIP/100-00000007", "__REC_STATUS=INITIALIZED") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:10] ExecIf("SIP/100-00000007", "0?Set(__REC_POLICY_MODE=)") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:11] Set("SIP/100-00000007", "NOW=1334772165") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:12] Set("SIP/100-00000007", "__DAY=18") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:13] Set("SIP/100-00000007", "__MONTH=04") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:14] Set("SIP/100-00000007", "__YEAR=2012") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:15] Set("SIP/100-00000007", "__TIMESTR=20120418-140245") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]record-check:16] Set("SIP/100-00000007", "__FROMEXTEN=100") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:17] Set("SIP/100-00000007", "__CALLFILENAME=out-xxxxxxxxxx-100-20120418-140245-1334772165.9") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:18] Goto("SIP/100-00000007", "out,1") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Goto (sub-record-check,out,1)
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:1] ExecIf("SIP/100-00000007", "1?Set(__REC_POLICY_MODE=dontcare)") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:2] GosubIf("SIP/100-00000007", "0?record,1(exten,xxxxxxxxxx,100)") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:3] Return("SIP/100-00000007", "") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:5] Macro("SIP/100-00000007", "dialout-trunk,1,xxxxxxxxxx,") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:1] Set("SIP/100-00000007", "DIAL_TRUNK=1") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:2] GosubIf("SIP/100-00000007", "0?sub-pincheck,s,1()") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:3] GotoIf("SIP/100-00000007", "0?disabletrunk,1") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:4] Set("SIP/100-00000007", "DIAL_NUMBER=xxxxxxxxxx") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:5] Set("SIP/100-00000007", "DIAL_TRUNK_OPTIONS=tr") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:6] Set("SIP/100-00000007", "OUTBOUND_GROUP=OUT_1") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:7] GotoIf("SIP/100-00000007", "0?nomax") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:8] GotoIf("SIP/100-00000007", "0?chanfull") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:9] GotoIf("SIP/100-00000007", "0?skipoutcid") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:10] Set("SIP/100-00000007", "DIAL_TRUNK_OPTIONS=") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:11] Macro("SIP/100-00000007", "outbound-callerid,1") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:1] ExecIf("SIP/100-00000007", "0?Set(CALLERPRES()=)") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:2] ExecIf("SIP/100-00000007", "0?Set(REALCALLERIDNUM=100)") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:3] GotoIf("SIP/100-00000007", "1?normcid") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Goto (macro-outbound-callerid,s,6)
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:6] Set("SIP/100-00000007", "USEROUTCID=") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:7] Set("SIP/100-00000007", "EMERGENCYCID=") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:8] Set("SIP/100-00000007", "TRUNKOUTCID=") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:9] GotoIf("SIP/100-00000007", "1?trunkcid") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Goto (macro-outbound-callerid,s,12)
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:12] ExecIf("SIP/100-00000007", "0?Set(CALLERID(all)=)") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:13] ExecIf("SIP/100-00000007", "0?Set(CALLERID(all)=)") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:14] ExecIf("SIP/100-00000007", "0?Set(CALLERID(all)=)") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:15] ExecIf("SIP/100-00000007", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:12] GosubIf("SIP/100-00000007", "0?sub-flp-1,s,1()") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:13] Set("SIP/100-00000007", "OUTNUM=xxxxxxxxxx") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:14] Set("SIP/100-00000007", "custom=DAHDI/G1") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:15] ExecIf("SIP/100-00000007", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default))") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:16] ExecIf("SIP/100-00000007", "0?Set(DIAL_TRUNK_OPTIONS=M(confirm))") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:17] Macro("SIP/100-00000007", "dialout-trunk-predial-hook,") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:1] MacroExit("SIP/100-00000007", "") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:18] GotoIf("SIP/100-00000007", "0?bypass,1") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:19] ExecIf("SIP/100-00000007", "1?Set(CONNECTEDLINE(num,i)=xxxxxxxxxx)") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:20] ExecIf("SIP/100-00000007", "1?Set(CONNECTEDLINE(name,i)=CID:100)") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:21] GotoIf("SIP/100-00000007", "0?customtrunk") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:22] Dial("SIP/100-00000007", "DAHDI/G1/xxxxxxxxxx,300,") in new stack
[2012-04-18 14:02:45] WARNING[17387] app_dial.c: Unable to create channel of type 'DAHDI' (cause 0 - Unknown)
[2012-04-18 14:02:45] VERBOSE[17387] app_dial.c: == Everyone is busy/congested at this time (1:0/0/1)
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:23] NoOp("SIP/100-00000007", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 0") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:24] Goto("SIP/100-00000007", "s-CHANUNAVAIL,1") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:1] Set("SIP/100-00000007", "RC=0") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:2] Goto("SIP/100-00000007", "0,1") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Goto (macro-dialout-trunk,0,1)
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:1] Goto("SIP/100-00000007", "continue,1") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Goto (macro-dialout-trunk,continue,1)
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:1] GotoIf("SIP/100-00000007", "1?noreport") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Goto (macro-dialout-trunk,continue,3)
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:3] NoOp("SIP/100-00000007", "TRUNK Dial failed due to CHANUNAVAIL HANGUPCAUSE: 0 - failing through to other trunks") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:4] Set("SIP/100-00000007", "CALLERID(number)=100") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:6] Macro("SIP/100-00000007", "outisbusy,") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:1] Progress("SIP/100-00000007", "") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] pbx.c: -- Executing [[email protected]:2] Playback("SIP/100-00000007", "all-circuits-busy-now,noanswer") in new stack
[2012-04-18 14:02:45] VERBOSE[17387] file.c: -- <SIP/100-00000007> Playing 'all-circuits-busy-now.ulaw' (language 'en')
[2012-04-18 14:02:46] VERBOSE[17387] pbx.c: -- Executing [[email protected]:3] Playback("SIP/100-00000007", "pls-try-call-later,noanswer") in new stack
[2012-04-18 14:02:46] VERBOSE[17387] file.c: -- <SIP/100-00000007> Playing 'pls-try-call-later.ulaw' (language 'en')
[2012-04-18 14:02:49] VERBOSE[17387] pbx.c: -- Executing [[email protected]:4] Macro("SIP/100-00000007", "hangupcall") in new stack
[2012-04-18 14:02:49] VERBOSE[17387] pbx.c: -- Executing [[email protected]:1] GotoIf("SIP/100-00000007", "1?theend") in new stack
[2012-04-18 14:02:49] VERBOSE[17387] pbx.c: -- Goto (macro-hangupcall,s,3)
[2012-04-18 14:02:49] VERBOSE[17387] pbx.c: -- Executing [[email protected]:3] Hangup("SIP/100-00000007", "") in new stack
[2012-04-18 14:02:49] VERBOSE[17387] app_macro.c: == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'SIP/100-00000007' in macro 'hangupcall'
[2012-04-18 14:02:49] VERBOSE[17387] app_macro.c: == Spawn extension (macro-outisbusy, s, 4) exited non-zero on 'SIP/100-00000007' in macro 'outisbusy'
[2012-04-18 14:02:49] VERBOSE[17387] pbx.c: == Spawn extension (from-internal, xxxxxxxxxx, 6) exited non-zero on 'SIP/100-00000007'
[2012-04-18 14:02:49] VERBOSE[17387] pbx.c: -- Executing [[email protected]:1] Hangup("SIP/100-00000007", "") in new stack

Several items:

Use the [code]tags[/code] when posting code so it is formatted so we can read it faster.

Also try an sanitize the logs.

This is the only relevant info:

You must not have any channels in group 1.

Never mind I figured it out. Thanks for the help

I have a setup which is pretty much identical to yours. I am having the exact same problem. It would be greatly appreciated if you shared the answer to your problem with us. Thanks.

The problem was as stated before. I had not put any channels into group 1. You have to add your channels to the group you are using in your trunks.

I am completely new to AsteriskNow and FreePBX and I apologize for my lack of understanding, but how exactly do I add channels to a group within FreePBX?

extension to extension calls work fine, but I cannot dial out or receive any incoming calls.

I seem to be having the exact issue described in this thread down to the same error messages in the CLI log, but I am still struggling with some basic concepts.

mcsuy - Here is the problem. FreePBX has nothing to do with DAHDI configuration. FreePBX can use a DAHDI trunk but it expects it to be configured and have the proper contexts specified to send the calls into the FreePBX dial plan.

You downloaded a CD produced by Digium that happens to include FreePBX because people wanted it. It just so happens that the FreePBX interface is the first thing people see so they associate FreePBX with the entire system.

To make it even worse Digium doesn’t really know much about FreePBX so they don’t do anything on the AsteriskNow CD to make life easier for FreePBX users. The dahdigenconf scripts are only partially FreePBX aware.

So bottom line is you need to configure your DAHDI drivers, in the Linux command line and the channels need to be in service from Asterisk’s perspective before the dial plan written by FreePBX can use the DAHDI channels.

This explanation probably is going to be quite confusing, however it is what it is.

I had a lot of odd problems and tried several different distros before I just used the freepbx distro. There were a few hiccups but the people here in the forums were very helpful with fixing the problems. I would recommend ditching the other distros for the one on here.

Let me also offer this point of clarification on tigger1197’s kind words.

The FreePBX distro is a separate project from FreePBX. Many of the same folks work on both projects. The distro is just that a distribution of Linux, Asterisk and FreePBX all pre-configured to install from an ISO.

It is not a panacea. The easiest cards to setup in DAHDI are the Sangoma’s. The setup-sangoma script they provide is as close to full proof as you are going to get. It knows about FreePBX and let’s you define the context and group of every port.

From that point it is downhill. Most all DAHDI configs require you to “tweak” the output of the very generic ‘dahdigenconf’ script. You must know how to edit text files, navigate the Linux file system and start and stop services to be successful at configuring DAHDI. If you are not willing to do those things then you should consider engaging a professional such as our FreePBX support team to help you.

What will not happen is someone post ‘step by step’ instructions to setup your xyz config. If the instructions were step by step they would already be published. Each step builds on the previous with various degrees of remediation required depending on the outcome of each step.

The purpose of this is not to be harsh but to ground folks in the reality of rolling their own PBX. Just because it’s free does not mean it is easy. The Free is as in Freedom and with Freedom comes responsibility. Equal opportunity does not generate equal outcome so as always “your mileage may vary”.

One other note, if you are doing work for a client and come asking for help you may notice a decided lack of sympathy. Accepting money to work on a system immediately makes you a professional by the pure definition of the term. Personally I am not here to grow someone else’s business. The harder you try, and the more information your provide the more help you will receive.

I will retreat from my soap box.

Have a great evening…Scott

I greatly appreciate the timely responses. Based on the direction provided from everyone, I did some reading and managed to solve my issue.

In a nutshell, i did the folliwing:

  • Download and install the latest linux kernel + some other libraries (per Digium)
    (gcc ncurses-devel libtermcap-devel kernel-devel gcc-c++ newt-dev zlib-devel unixODBC-devel libtool)
  • Download and install the latest dahdi drivers (http://downloads.digium.com/pub/telephony/dahdi-linux-complete)
  • execute /usr/sbin/dahdi_genconf to create dahdi-channels.conf
  • inserted an include of dahdi-channels.conf into chan_dahdi.conf
  • rebooted and a miracle occurred

For the record, most of this was found in the Digium 800 series user manual

You got it. Glad you got it working.

It is hard for folks to understand that FreePBX is just one part of the puzzle.

Dear All,

For your information. I am in the midst of troubleshooting inbound ./outbound calls from Elastix. The current server has been installed with Sangoma card A200-R. The PSTN connection has been connected into FXO port 4. However, after few setting in outbound/inbound and trunk . I am still facing problem cannot received call in softphones. Unable to to call out to mobile/fix line phones numbers. Eventhou, I already read the elastix without tears and FreepBX 2.5 manual. I could not resolved the issue. Kindly advised. Thanks.