Hi folks,
I’m trying to get inbound faxing working on my install of FreePBX (converted from Trixbox!). I have gotten T.38 to cooperate by ensuring that it was turned on at my SIP provider, and now I’m stuck.
I have a DID routed directly to an extension set to receive faxes. The line picks up and I hear fax tones, and if I watch in the asterisk CLI I see everything appear to move smoothly, then it throws a repeated error that doesn’t mean anything to me at the moment. Here’s an excerpt:
== Using SIP RTP CoS mark 5
-- Executing [xxx@from-pstn:1] Set("SIP/xxx", "__FROM_DID=xxx") in new stack
-- Executing [xxx@from-pstn:2] Gosub("SIP/xxx", "app-blacklist-check,s,1()") in new stack
-- Executing [s@app-blacklist-check:1] GotoIf("SIP/xxx", "0?blacklisted") in new stack
-- Executing [s@app-blacklist-check:2] Set("SIP/xxx", "CALLED_BLACKLIST=1") in new stack
-- Executing [s@app-blacklist-check:3] Return("xxx", "") in new stack
-- Executing [xxx@from-pstn:3] Gosub("xxx", "cidlookup,cidlookup_1,1()") in new stack
-- Executing [cidlookup_1@cidlookup:1] Set("xxx", "CURLOPT(httptimeout)=7") in new stack
-- Executing [cidlookup_1@cidlookup:2] Set("SIP/xxx", "CALLERID(name)=") in new stack
-- Executing [cidlookup_1@cidlookup:3] Return("SIP/xxx", "") in new stack
-- Executing [xxx@from-pstn:4] Set("SIP/xxx", "CDR(did)=xxx") in new stack
-- Executing [xxx@from-pstn:5] ExecIf("SIP/xxx", "1 ?Set(CALLERID(name)=xxx)") in new stack
[2012-10-24 13:34:35] WARNING[5923]: func_callerid.c:817 callerpres_read: CALLERPRES is deprecated. Use CALLERID(name-pres) or CALLERID(num-pres) instead.
-- Executing [xxx@from-pstn:6] Set("SIP/xxx", "__CALLINGPRES_SV=allowed_not_screened") in new stack
-- Executing [xxx@from-pstn:7] Set("SIP/xxx", "CALLERPRES()=allowed_not_screened") in new stack
-- Executing [xxx@from-pstn:8] Goto("SIP/xxx", "ext-fax,100,1") in new stack
-- Goto (ext-fax,100,1)
-- Executing [100@ext-fax:1] Set("SIP/xxx", "FAX_FOR=General Mailbox (100)") in new stack
-- Executing [100@ext-fax:2] NoOp("SIP/xxx", "Receiving Fax for: General Mailbox (100), From: "xxx" <xxx>") in new stack
-- Executing [100@ext-fax:3] Set("SIP/xxx", "[email protected]") in new stack
-- Executing [100@ext-fax:4] Goto("SIP/xxx", "s,receivefax") in new stack
-- Goto (ext-fax,s,3)
-- Executing [s@ext-fax:3] StopPlayTones("SIP/xxx", "") in new stack
-- Executing [s@ext-fax:4] ReceiveFAX("SIP/xxx", "/var/spool/asterisk/fax/1351100068.0.tif,f") in new stack
-- Channel 'SIP/xxx' receiving FAX '/var/spool/asterisk/fax/1351100068.0.tif'
== Using UDPTL TOS bits 184
== Using UDPTL CoS mark 5
[2012-10-24 13:34:39] NOTICE[5923]: udptl.c:1092 ast_udptl_write: UDPTL (SIP/xxx): Transmission error to 255.255.255.255:4684: Permission denied
{{repeat previous line about 100x}}
-- Executing [s@ext-fax:5] ExecIf("SIP/xxx", "1?Set(FAXSTATUS="FAILED: error: Disconnected after permitted retries statusstr: Disconnected after permitted retries")") in new stack
-- Executing [s@ext-fax:6] Hangup("SIP/xxx", "") in new stack
== Spawn extension (ext-fax, s, 6) exited non-zero on 'xxx'
-- Executing [h@ext-fax:1] GotoIf("SIP/xxx", "1?failed") in new stack
-- Goto (ext-fax,h,103)
-- Executing [h@ext-fax:103] NoOp("SIP/xxx", "FAX "FAILED: error: Disconnected after permitted retries statusstr: Disconnected after permitted retries" for: [email protected] , From: "xxx" <xxx>") in new stack
-- Executing [h@ext-fax:104] Macro("SIP/xxx", "hangupcall,") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/xxx", "1?theend") in new stack
-- Goto (macro-hangupcall,s,3)
-- Executing [s@macro-hangupcall:3] ExecIf("SIP/xxx", "0?Set(CDR(recordingfile)=)") in new stack
-- Executing [s@macro-hangupcall:4] Hangup("SIP/xxx", "") in new stack
== Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'xxx' in macro 'hangupcall'
== Spawn extension (ext-fax, h, 104) exited non-zero on 'SIP/xxx'
So from what I can gather here, there is some sort of permission error, but I have no idea where this 255.255.255.255 is coming from - in other people’s posted logs, this seems to point to an IP for their box, but this is all happening on the same PBX.
If anyone has any thoughts or advice to offer, it would be much appreciated!