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HT813 With FreePBX - can't make outbound calls

siptrunk
configuration
freepbx
Tags: #<Tag:0x00007fafce8fc7c8> #<Tag:0x00007fafce90ff80> #<Tag:0x00007fafce90f558>

#1

First post here, semi-frequent user of FreePBX (never used SIP trunking before, had always used Google Voice and kinda stopped when that became no-longer-a-thing).

I have successfully setup my HT813 to use the FXO port to receive incoming calls. I primarily followed the youtube video www(dot)youtube(dot)com(slash)watch?v=J6oJSMDJzEI. This config didn’t work perfectly, so I did have to make a few tweaks, but I was able to get inbound calls working. However, I can only make outbound calls if I first dial the trunk I setup for the HT813, which gives me a dialtone, at which I can then dial out to the PSTN. Directly dialing the PSTN number results in an “All circuits are busy” message. Outbound routes are setup to use this trunk for the proper dialing patterns (US). Any time I try to make an outbound call by directly dialing the 10 digit number it results in the circuits busy message, and this appearing in the asterisk -rvvvvv output:

Note: I have Anonymized the log: 1234567890 represents number I’m trying to call. 9876543210 represents my outbound CID. 1234567 is the trunk name in FreePBX that is used for the HT813. 10.0.1.113 and 10.5.1.40 are the IP addresses of the FreePBX server. 10.5.1.41 is my Cisco SPA303. 3011 is the extension for the SPA303. 10.0.1.222 is the IP of the HT813.

Asterisk 13.19.1, Copyright (C) 1999 - 2014, Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
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 13.19.1 currently running on freepbx (pid = 2086)
  == Setting global variable 'SIPDOMAIN' to '10.5.1.40'
  == Using SIP RTP Audio TOS bits 184
  == Using SIP RTP Audio TOS bits 184 in TCLASS field.
  == Using SIP RTP Audio CoS mark 5
    -- Executing [1234567890@from-internal:1] Macro("PJSIP/3011-00000046", "user-callerid,LIMIT,EXTERNAL,") in new stack
    -- Executing [s@macro-user-callerid:1] Set("PJSIP/3011-00000046", "TOUCH_MONITOR=1560276701.124") in new stack
    -- Executing [s@macro-user-callerid:2] Set("PJSIP/3011-00000046", "AMPUSER=3011") in new stack
    -- Executing [s@macro-user-callerid:3] GotoIf("PJSIP/3011-00000046", "0?report") in new stack
    -- Executing [s@macro-user-callerid:4] ExecIf("PJSIP/3011-00000046", "1?Set(REALCALLERIDNUM=3011)") in new stack
    -- Executing [s@macro-user-callerid:5] Set("PJSIP/3011-00000046", "AMPUSER=3011") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("PJSIP/3011-00000046", "0?limit") in new stack
    -- Executing [s@macro-user-callerid:7] Set("PJSIP/3011-00000046", "AMPUSERCIDNAME=Office Phone") in new stack
    -- Executing [s@macro-user-callerid:8] ExecIf("PJSIP/3011-00000046", "0?Set(__CIDMASQUERADING=TRUE)") in new stack
    -- Executing [s@macro-user-callerid:9] GotoIf("PJSIP/3011-00000046", "0?report") in new stack
    -- Executing [s@macro-user-callerid:10] Set("PJSIP/3011-00000046", "AMPUSERCID=3011") in new stack
    -- Executing [s@macro-user-callerid:11] Set("PJSIP/3011-00000046", "__DIAL_OPTIONS=HhTtr") in new stack
    -- Executing [s@macro-user-callerid:12] Set("PJSIP/3011-00000046", "CALLERID(all)="Office Phone" <3011>") in new stack
    -- Executing [s@macro-user-callerid:13] GotoIf("PJSIP/3011-00000046", "0?limit") in new stack
    -- Executing [s@macro-user-callerid:14] ExecIf("PJSIP/3011-00000046", "1?Set(GROUP(concurrency_limit)=3011)") in new stack
    -- Executing [s@macro-user-callerid:15] ExecIf("PJSIP/3011-00000046", "0?Set(CHANNEL(language)=)") in new stack
    -- Executing [s@macro-user-callerid:16] NoOp("PJSIP/3011-00000046", "Macro Depth is 1") in new stack
    -- Executing [s@macro-user-callerid:17] GotoIf("PJSIP/3011-00000046", "1?report2:macroerror") in new stack
    -- Goto (macro-user-callerid,s,18)
    -- Executing [s@macro-user-callerid:18] GotoIf("PJSIP/3011-00000046", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,37)
    -- Executing [s@macro-user-callerid:37] Set("PJSIP/3011-00000046", "CALLERID(number)=3011") in new stack
    -- Executing [s@macro-user-callerid:38] Set("PJSIP/3011-00000046", "CALLERID(name)=Office Phone") in new stack
    -- Executing [s@macro-user-callerid:39] GotoIf("PJSIP/3011-00000046", "0?cnum") in new stack
    -- Executing [s@macro-user-callerid:40] Set("PJSIP/3011-00000046", "CDR(cnam)=Office Phone") in new stack
    -- Executing [s@macro-user-callerid:41] Set("PJSIP/3011-00000046", "CDR(cnum)=3011") in new stack
    -- Executing [s@macro-user-callerid:42] Set("PJSIP/3011-00000046", "CHANNEL(language)=en") in new stack
    -- Executing [1234567890@from-internal:2] Gosub("PJSIP/3011-00000046", "sub-record-check,s,1(out,1234567890,dontcare)") in new stack
    -- Executing [s@sub-record-check:1] GotoIf("PJSIP/3011-00000046", "0?initialized") in new stack
    -- Executing [s@sub-record-check:2] Set("PJSIP/3011-00000046", "__REC_STATUS=INITIALIZED") in new stack
    -- Executing [s@sub-record-check:3] Set("PJSIP/3011-00000046", "NOW=1560276701") in new stack
    -- Executing [s@sub-record-check:4] Set("PJSIP/3011-00000046", "__DAY=11") in new stack
    -- Executing [s@sub-record-check:5] Set("PJSIP/3011-00000046", "__MONTH=06") in new stack
    -- Executing [s@sub-record-check:6] Set("PJSIP/3011-00000046", "__YEAR=2019") in new stack
    -- Executing [s@sub-record-check:7] Set("PJSIP/3011-00000046", "__TIMESTR=20190611-181141") in new stack
    -- Executing [s@sub-record-check:8] Set("PJSIP/3011-00000046", "__FROMEXTEN=3011") in new stack
    -- Executing [s@sub-record-check:9] Set("PJSIP/3011-00000046", "__MON_FMT=wav") in new stack
    -- Executing [s@sub-record-check:10] NoOp("PJSIP/3011-00000046", "Recordings initialized") in new stack
    -- Executing [s@sub-record-check:11] ExecIf("PJSIP/3011-00000046", "0?Set(ARG3=dontcare)") in new stack
    -- Executing [s@sub-record-check:12] Set("PJSIP/3011-00000046", "REC_POLICY_MODE_SAVE=") in new stack
    -- Executing [s@sub-record-check:13] ExecIf("PJSIP/3011-00000046", "0?Set(REC_STATUS=NO)") in new stack
    -- Executing [s@sub-record-check:14] GotoIf("PJSIP/3011-00000046", "3?checkaction") in new stack
    -- Goto (sub-record-check,s,17)
    -- Executing [s@sub-record-check:17] GotoIf("PJSIP/3011-00000046", "1?sub-record-check,out,1") in new stack
    -- Goto (sub-record-check,out,1)
    -- Executing [out@sub-record-check:1] NoOp("PJSIP/3011-00000046", "Outbound Recording Check from 3011 to 1234567890") in new stack
    -- Executing [out@sub-record-check:2] Set("PJSIP/3011-00000046", "RECMODE=dontcare") in new stack
    -- Executing [out@sub-record-check:3] ExecIf("PJSIP/3011-00000046", "1?Goto(routewins)") in new stack
    -- Goto (sub-record-check,out,7)
    -- Executing [out@sub-record-check:7] Gosub("PJSIP/3011-00000046", "recordcheck,1(dontcare,out,1234567890)") in new stack
    -- Executing [recordcheck@sub-record-check:1] NoOp("PJSIP/3011-00000046", "Starting recording check against dontcare") in new stack
    -- Executing [recordcheck@sub-record-check:2] Goto("PJSIP/3011-00000046", "dontcare") in new stack
    -- Goto (sub-record-check,recordcheck,3)
    -- Executing [recordcheck@sub-record-check:3] Return("PJSIP/3011-00000046", "") in new stack
    -- Executing [out@sub-record-check:8] Return("PJSIP/3011-00000046", "") in new stack
    -- Executing [1234567890@from-internal:3] ExecIf("PJSIP/3011-00000046", "0 ?Set(CDR(accountcode)=)") in new stack
    -- Executing [1234567890@from-internal:4] Set("PJSIP/3011-00000046", "MOHCLASS=default") in new stack
    -- Executing [1234567890@from-internal:5] ExecIf("PJSIP/3011-00000046", "0?Set(TRUNKCIDOVERRIDE=9876543210)") in new stack
    -- Executing [1234567890@from-internal:6] Set("PJSIP/3011-00000046", "_NODEST=") in new stack
    -- Executing [1234567890@from-internal:7] Macro("PJSIP/3011-00000046", "dialout-trunk,1,1234567890,,off") in new stack
    -- Executing [s@macro-dialout-trunk:1] Set("PJSIP/3011-00000046", "DIAL_TRUNK=1") in new stack
    -- Executing [s@macro-dialout-trunk:2] GosubIf("PJSIP/3011-00000046", "0?sub-pincheck,s,1()") in new stack
    -- Executing [s@macro-dialout-trunk:3] ExecIf("PJSIP/3011-00000046", "0?Set(CALLERID(num)=3011)") in new stack
    -- Executing [s@macro-dialout-trunk:4] GotoIf("PJSIP/3011-00000046", "0?disabletrunk,1") in new stack
    -- Executing [s@macro-dialout-trunk:5] Set("PJSIP/3011-00000046", "DIAL_NUMBER=1234567890") in new stack
    -- Executing [s@macro-dialout-trunk:6] Set("PJSIP/3011-00000046", "DIAL_TRUNK_OPTIONS=HhTtr") in new stack
    -- Executing [s@macro-dialout-trunk:7] Set("PJSIP/3011-00000046", "OUTBOUND_GROUP=OUT_1") in new stack
    -- Executing [s@macro-dialout-trunk:8] Set("PJSIP/3011-00000046", "DIAL_TRUNK_OPTIONS=T") in new stack
    -- Executing [s@macro-dialout-trunk:9] GotoIf("PJSIP/3011-00000046", "0?nomax") in new stack
    -- Executing [s@macro-dialout-trunk:10] GotoIf("PJSIP/3011-00000046", "0?chanfull") in new stack
    -- Executing [s@macro-dialout-trunk:11] GotoIf("PJSIP/3011-00000046", "0?skipoutcid") in new stack
    -- Executing [s@macro-dialout-trunk:12] Macro("PJSIP/3011-00000046", "outbound-callerid,1") in new stack
    -- Executing [s@macro-outbound-callerid:1] NoOp("PJSIP/3011-00000046", "3011") in new stack
    -- Executing [s@macro-outbound-callerid:2] NoOp("PJSIP/3011-00000046", "") in new stack
    -- Executing [s@macro-outbound-callerid:3] NoOp("PJSIP/3011-00000046", "off") in new stack
    -- Executing [s@macro-outbound-callerid:4] ExecIf("PJSIP/3011-00000046", "0?Set(CALLERPRES(name-pres)=)") in new stack
    -- Executing [s@macro-outbound-callerid:5] ExecIf("PJSIP/3011-00000046", "0?Set(CALLERPRES(num-pres)=)") in new stack
    -- Executing [s@macro-outbound-callerid:6] ExecIf("PJSIP/3011-00000046", "0?Set(REALCALLERIDNUM=3011)") in new stack
    -- Executing [s@macro-outbound-callerid:7] GotoIf("PJSIP/3011-00000046", "1?normcid") in new stack
    -- Goto (macro-outbound-callerid,s,11)
    -- Executing [s@macro-outbound-callerid:11] Set("PJSIP/3011-00000046", "USEROUTCID=3011") in new stack
    -- Executing [s@macro-outbound-callerid:12] Set("PJSIP/3011-00000046", "EMERGENCYCID=") in new stack
    -- Executing [s@macro-outbound-callerid:13] Set("PJSIP/3011-00000046", "TRUNKOUTCID=9876543210") in new stack
    -- Executing [s@macro-outbound-callerid:14] GotoIf("PJSIP/3011-00000046", "1?trunkcid") in new stack
    -- Goto (macro-outbound-callerid,s,19)
    -- Executing [s@macro-outbound-callerid:19] ExecIf("PJSIP/3011-00000046", "1?Set(CALLERID(all)=9876543210)") in new stack
    -- Executing [s@macro-outbound-callerid:20] ExecIf("PJSIP/3011-00000046", "1?Set(CALLERID(all)=3011)") in new stack
    -- Executing [s@macro-outbound-callerid:21] ExecIf("PJSIP/3011-00000046", "0?Set(CALLERID(all)=)") in new stack
    -- Executing [s@macro-outbound-callerid:22] ExecIf("PJSIP/3011-00000046", "0?Set(CALLERPRES(name-pres)=prohib_passed_screen)") in new stack
    -- Executing [s@macro-outbound-callerid:23] ExecIf("PJSIP/3011-00000046", "0?Set(CALLERPRES(num-pres)=prohib_passed_screen)") in new stack
    -- Executing [s@macro-outbound-callerid:24] Set("PJSIP/3011-00000046", "CDR(outbound_cnum)=3011") in new stack
    -- Executing [s@macro-outbound-callerid:25] Set("PJSIP/3011-00000046", "CDR(outbound_cnam)=") in new stack
    -- Executing [s@macro-dialout-trunk:13] GosubIf("PJSIP/3011-00000046", "0?sub-flp-1,s,1()") in new stack
    -- Executing [s@macro-dialout-trunk:14] Set("PJSIP/3011-00000046", "OUTNUM=1234567890") in new stack
    -- Executing [s@macro-dialout-trunk:15] Set("PJSIP/3011-00000046", "custom=SIP/1234567") in new stack
    -- Executing [s@macro-dialout-trunk:16] ExecIf("PJSIP/3011-00000046", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default)T)") in new stack
    -- Executing [s@macro-dialout-trunk:17] ExecIf("PJSIP/3011-00000046", "0?Set(DIAL_TRUNK_OPTIONS=TM(confirm))") in new stack
    -- Executing [s@macro-dialout-trunk:18] Macro("PJSIP/3011-00000046", "dialout-trunk-predial-hook,") in new stack
    -- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("PJSIP/3011-00000046", "") in new stack
    -- Executing [s@macro-dialout-trunk:19] GotoIf("PJSIP/3011-00000046", "0?skipcrm") in new stack
    -- Executing [s@macro-dialout-trunk:20] Set("PJSIP/3011-00000046", "__CRM_DIRECTION=OUTBOUND") in new stack
    -- Executing [s@macro-dialout-trunk:21] Set("PJSIP/3011-00000046", "__CRM_DESTINATION=1234567890") in new stack
    -- Executing [s@macro-dialout-trunk:22] Set("PJSIP/3011-00000046", "__CRM_SOURCE=3011") in new stack
    -- Executing [s@macro-dialout-trunk:23] AGI("PJSIP/3011-00000046", "sangomacrm.agi") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/sangomacrm.agi
    -- <PJSIP/3011-00000046>AGI Script sangomacrm.agi completed, returning 0
    -- Executing [s@macro-dialout-trunk:24] Set("PJSIP/3011-00000046", "CHANNEL(hangup_handler_push)=crm-hangup,s,1") in new stack
    -- Executing [s@macro-dialout-trunk:25] NoOp("PJSIP/3011-00000046", "CRM Finished") in new stack
    -- Executing [s@macro-dialout-trunk:26] GotoIf("PJSIP/3011-00000046", "0?bypass,1") in new stack
    -- Executing [s@macro-dialout-trunk:27] ExecIf("PJSIP/3011-00000046", "1?Set(CONNECTEDLINE(num,i)=1234567890)") in new stack
    -- Executing [s@macro-dialout-trunk:28] ExecIf("PJSIP/3011-00000046", "1?Set(CONNECTEDLINE(name,i)=CID:3011)") in new stack
    -- Executing [s@macro-dialout-trunk:29] ExecIf("PJSIP/3011-00000046", "0?Set(CONNECTEDLINE(name,i)=CID:(Hidden)3011)") in new stack
    -- Executing [s@macro-dialout-trunk:30] GotoIf("PJSIP/3011-00000046", "0?customtrunk") in new stack
    -- Executing [s@macro-dialout-trunk:31] Dial("PJSIP/3011-00000046", "SIP/1234567/1234567890,300,Tb(func-apply-sipheaders^s^1)") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
    -- SIP/1234567-00000036 Internal Gosub(func-apply-sipheaders,s,1) start
    -- Executing [s@func-apply-sipheaders:1] ExecIf("SIP/1234567-00000036", "0?Set(CHANNEL(hangup_handler_push)=crm-hangup,s,1)") in new stack
    -- Executing [s@func-apply-sipheaders:2] NoOp("SIP/1234567-00000036", "Applying SIP Headers to channel") in new stack
    -- Executing [s@func-apply-sipheaders:3] Set("SIP/1234567-00000036", "SIPHEADERKEYS=") in new stack
    -- Executing [s@func-apply-sipheaders:4] While("SIP/1234567-00000036", "0") in new stack
    -- Jumping to priority 8
    -- Executing [s@func-apply-sipheaders:9] Return("SIP/1234567-00000036", "") in new stack
  == Spawn extension (from-trunk-sip-1234567, 1234567890, 1) exited non-zero on 'SIP/1234567-00000036'
    -- SIP/1234567-00000036 Internal Gosub(func-apply-sipheaders,s,1) complete GOSUB_RETVAL=
    -- Called SIP/1234567/1234567890
[2019-06-11 18:11:42] WARNING[2159][C-00000044]: chan_sip.c:24015 handle_response_invite: Received response: "Forbidden" from '<sip:3011@10.0.1.113:5160>;tag=as3acba838'
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [s@macro-dialout-trunk:32] NoOp("PJSIP/3011-00000046", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 21") in new stack
    -- Executing [s@macro-dialout-trunk:33] GotoIf("PJSIP/3011-00000046", "0?continue,1:s-CHANUNAVAIL,1") in new stack
    -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
    -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] Set("PJSIP/3011-00000046", "RC=21") in new stack
    -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:2] Goto("PJSIP/3011-00000046", "21,1") in new stack
    -- Goto (macro-dialout-trunk,21,1)
    -- Executing [21@macro-dialout-trunk:1] Goto("PJSIP/3011-00000046", "continue,1") in new stack
    -- Goto (macro-dialout-trunk,continue,1)
    -- Executing [continue@macro-dialout-trunk:1] NoOp("PJSIP/3011-00000046", "TRUNK Dial failed due to CHANUNAVAIL HANGUPCAUSE: 21 - failing through to other trunks") in new stack
    -- Executing [continue@macro-dialout-trunk:2] ExecIf("PJSIP/3011-00000046", "1?Set(CALLERID(number)=3011)") in new stack
    -- Executing [1234567890@from-internal:8] Macro("PJSIP/3011-00000046", "outisbusy,") in new stack
    -- Executing [s@macro-outisbusy:1] Progress("PJSIP/3011-00000046", "") in new stack
    -- Executing [s@macro-outisbusy:2] GotoIf("PJSIP/3011-00000046", "0?emergency,1") in new stack
    -- Executing [s@macro-outisbusy:3] GotoIf("PJSIP/3011-00000046", "0?intracompany,1") in new stack
    -- Executing [s@macro-outisbusy:4] Playback("PJSIP/3011-00000046", "all-circuits-busy-now&please-try-call-later, noanswer") in new stack
    -- <PJSIP/3011-00000046> Playing 'all-circuits-busy-now.g722' (language 'en')
       > 0x7fc26400da10 -- Strict RTP learning after remote address set to: 10.5.1.41:16494
       > 0x7fc26400da10 -- Strict RTP switching to RTP target address 10.5.1.41:16494 as source
    -- <PJSIP/3011-00000046> Playing 'please-try-call-later.ulaw' (language 'en')
    -- Executing [s@macro-outisbusy:5] Congestion("PJSIP/3011-00000046", "20") in new stack
  == Spawn extension (macro-outisbusy, s, 5) exited non-zero on 'PJSIP/3011-00000046' in macro 'outisbusy'
  == Spawn extension (from-internal, 1234567890, 8) exited non-zero on 'PJSIP/3011-00000046'
    -- Executing [h@from-internal:1] Macro("PJSIP/3011-00000046", "hangupcall") in new stack
    -- Executing [s@macro-hangupcall:1] GotoIf("PJSIP/3011-00000046", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,3)
    -- Executing [s@macro-hangupcall:3] ExecIf("PJSIP/3011-00000046", "0?Set(CDR(recordingfile)=)") in new stack
    -- Executing [s@macro-hangupcall:4] NoOp("PJSIP/3011-00000046", " monior file= ") in new stack
    -- Executing [s@macro-hangupcall:5] AGI("PJSIP/3011-00000046", "attendedtransfer-rec-restart.php,,") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/attendedtransfer-rec-restart.php
    -- <PJSIP/3011-00000046>AGI Script attendedtransfer-rec-restart.php completed, returning 0
    -- Executing [s@macro-hangupcall:6] Hangup("PJSIP/3011-00000046", "") in new stack
  == Spawn extension (macro-hangupcall, s, 6) exited non-zero on 'PJSIP/3011-00000046' in macro 'hangupcall'
  == Spawn extension (from-internal, h, 1) exited non-zero on 'PJSIP/3011-00000046'
    -- PJSIP/3011-00000046 Internal Gosub(crm-hangup,s,1) start
    -- Executing [s@crm-hangup:1] NoOp("PJSIP/3011-00000046", "Sending Hangup to CRM") in new stack
    -- Executing [s@crm-hangup:2] NoOp("PJSIP/3011-00000046", "HANGUP CAUSE: 34") in new stack
    -- Executing [s@crm-hangup:3] ExecIf("PJSIP/3011-00000046", "0?Set(__CRM_VOICEMAIL=)") in new stack
    -- Executing [s@crm-hangup:4] NoOp("PJSIP/3011-00000046", "MASTER CHANNEL: 1560276701.124 = 1560276701.124") in new stack
    -- Executing [s@crm-hangup:5] GotoIf("PJSIP/3011-00000046", "0?return") in new stack
    -- Executing [s@crm-hangup:6] Set("PJSIP/3011-00000046", "__CRM_HANGUP=1") in new stack
    -- Executing [s@crm-hangup:7] AGI("PJSIP/3011-00000046", "sangomacrm.agi") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/sangomacrm.agi
    -- <PJSIP/3011-00000046>AGI Script sangomacrm.agi completed, returning 0
    -- Executing [s@crm-hangup:8] Return("PJSIP/3011-00000046", "") in new stack
  == Spawn extension (from-internal, h, 1) exited non-zero on 'PJSIP/3011-00000046'
    -- PJSIP/3011-00000046 Internal Gosub(crm-hangup,s,1) complete GOSUB_RETVAL=

The PEER details I’m using are (again, anonymized using the same description as above):

username=1234567
type=peer
secret=correcthorsebatterystaple
qualify=60000
port=5062
insecure=port,invite
host=10.0.1.222
dtmfmode=rfc2833
disallow=all
canreivite=no
allow=ulaw

I have also tried it with type=friend as shown in the video.

Any help is greatly, greatly appreciated. Thanks!


#2

The ‘forbidden’ error seems bizarre. At the Asterisk command prompt, type
sip set debug on
pjsip set logger on
which will cause the SIP traffic to be included in the log.

Make another call attempt and post the sanitized log.
Normally, what you see in Reports->Asterisk Logfiles has timestamps on each line. Did you get this log from another source? The timestamps would be useful to see how long it takes the HT to respond with whatever error it’s sending.


#3

If you receive dial tone, then you haven’t correctly configured one stage dialing on the ht813


#4

@arielgrin Looks promising… Any idea how one might configure it? The only section I see in the manual is short and doesn’t help a lot.


#5

@Stewart1 I’ll try when I get home tonight, and post results.


#6

Let me take a look at its manual and I’ll get back to you. I guess it should be similar to the ht503, but I might be wrong.


#7

@arielgrin Thanks! From what I’ve heard too they’re very similar.


#8

Here ya are, @Stewart1. Same sanitization protocol as my first one. Apologies for the delay as a power outage killed my internet for most of the day yesterday.

The other IP you may see in this one: 10.0.1.118 is my iPhone that has a SIP client on it. No idea why it would show up as I didn’t even touch it, but I saw it come up so I figured I’d list that. It’s registered to extension 3058. It’s too big to fit here fit here, so posted it at gist(dot)github(dot)com/lightbord/795414fb11537b7f184b55a1d9348a0f


#9

@arielgrin I just found the setting for one stage dialing, and indeed it was set to 2. Changed it to 1 and now I can make outbound calls! Only thing is now whenever I call a number, there seems to be a 5-6 second delay before I can hear the audio of the call. This is fine when calling a human, just ask them to repeat. But when an IVR picks up, that may cause you to miss the first few seconds of the recording, which may not be so easy to convince it to repeat. So it would be great if anyone might know why there’s such a delay.

@Stewart1 I’d still be curious about those “Forbidden” errors, and whether they are something I should be concerned about or not.


#10

The 403 Forbidden was related to the 2 stage dialing – confirm that you are no longer seeing it.

It’s normal to have traffic to inactive devices such as your iPhone; there are both registration updates and OPTIONS sent to confirm that the device is still reachable (qualify).

The only FXO setting I see that might be relevant is Enable Polarity Reversal, which should be No.

What is the FXO port connected to (copper pair from central office, cable MTA, fiber ONT, etc.)?

If you call a number that is not quickly answered, do you start hearing ringback tone after (setup time + 6 seconds)? When the call is then answered, do you have audio immediately?

Make a test call to 1 800 437 7950. Note the time when you first hear audio (the clock on your mobile is probably accurate to within one second, good enough for this purpose). Also, note the first thing you hear (the initial “Thank you for calling …” starts about 1.5 seconds after answer). Then, post that info and the log. The best way to post a log here is to paste it as pastebin.freepbx.org and post the link here. If the forum doesn’t let you post links, replace the last dot with %2E, for example
pastebin.freepbx%2Eorg/view/fa3c7524
which will open as a proper link in most browsers.


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