I have an Openvox board connected to a handset.
I then have a VOIP connection for the outbound.
I do this so that my handset for the office (connected as line 2) will ring throughout the house.
I ran a yum update on my machine and have now (mostly) hosed system.
I can receive calls from external sources.
I can call from a softphone extension to the analog phone.
I can call out from neither the softphone nor the analog handset.
Here is the (edited) cli from a call from the analog handset:
Verbosity is at least 26
– Starting simple switch on ‘DAHDI/2-1’
– Executing [[email protected]:1] NoOp(“DAHDI/2-1”, “Catch-All DID Match - Found 91NNNYYYZZZZ - You probably want a DID for this.”) in new stack
– Executing [[email protected]:2] Goto(“DAHDI/2-1”, “ext-did,s,1”) in new stack
– Goto (ext-did,s,1)
– Executing [[email protected]:1] Set(“DAHDI/2-1”, “__FROM_DID=s”) in new stack
– Executing [[email protected]:2] Gosub(“DAHDI/2-1”, “app-blacklist-check,s,1()”) in new stack
– Executing [[email protected]:1] GotoIf(“DAHDI/2-1”, “0?blacklisted”) in new stack
– Executing [[email protected]:2] Set(“DAHDI/2-1”, “CALLED_BLACKLIST=1”) in new stack
– Executing [[email protected]:3] Return(“DAHDI/2-1”, “”) in new stack
– Executing [[email protected]:3] Set(“DAHDI/2-1”, “CDR(did)=s”) in new stack
…
It seems to be blowing up on the second line. The extension is defined as ‘from-internal’, by the way.
The softphone cli output reads:
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using UDPTL TOS bits 184
== Using UDPTL CoS mark 5
– Executing [[email protected]:1] ResetCDR(“SIP/200-0000000f”, “”) in new stack
– Executing [[email protected]:2] NoCDR(“SIP/200-0000000f”, “”) in new stack
– Executing [[email protected]:3] Progress(“SIP/200-0000000f”, “”) in new stack
– Executing [[email protected]:4] Wait(“SIP/200-0000000f”, “1”) in new stack
– Executing [[email protected]:5] Progress(“SIP/200-0000000f”, “”) in new stack
– Executing [[email protected]:6] Playback(“SIP/200-0000000f”, “silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer”) in new stack
…
Any thoughts on this would be appreciated; clearly a file somewhere has a bad parameter since incoming calls work just fine.
looks like your chan_dadhi.conf got screwed up and if it’s not properly including your chan_dahdi_additional.conf file that FreePBX generates then none of your configuration will be pulled in.
I checked the chan_dahdi.conf file and it has the appropriate lines per the Openvox installation guide.
dahdi show channels shows:
PhoneSystem*CLI> dahdi show channels
Chan Extension Context Language MOH Interpret Blocked State
pseudo default default In Service
1 default default In Service
2 from-pstn en default In Service
which appears to be correct (although shouldin’t that say from-internal?).
Is there a ‘next’ I should check? I have run dahdi_genconf and checked the files a few times.
If you would prefer to believe Openvox you may want to try their forum for help.
Since you came to the FreePBX forum I think it’s generally understood you are looking for FreePBX help… If Openvox disagrees with what I said, then they don’t understand what FreePBX needs…
;--------------------------------------------------------------------------------;
; Do NOT edit this file as it is auto-generated by FreePBX. All modifications to ;
; this file must be done via the web gui. There are alternative files to make ;
; custom modifications, details at: http://freepbx.org/configuration_files ;
;--------------------------------------------------------------------------------;
;
Since the Openvox forum won’t even allow me to correctly register, going in that direction isn’t even an option for me.
I appreciate the help, I’m just a little on the slow side and far from being considered even a novice in understanding how to interpret what might be incorrect in these files. A google search shed no light on the matter.