Calls being dropped when selecting option two from IVR

Hey, whatup! I am new here so if i am providing the wrong information or asking it wrong i apologize. So ive been able to work though a good amount of issues and feel comfortable setting up a basic setup. (no paid modules, CLI update, manually setting up trunks in/outbound routes and extensions, Nat and Firewall issues, remote client, CLI debugging…)

So i bring it up at work the other day and my boss asks me to fix a clients setup. I would appreciate some wisdom from someone more experienced before i make any decisions. The tiny differences in versions combined with the flood of ‘answers’ online point to the only solution… Join the FreePBX forums and get some help from someone knowledgeable instead of trying to plug in answers from other peoples questions … no matter how similar they may look.

Their setup! So the client runs an ‘organic’ pizza place. They have two locations with two PBX systems (sadly different versions!) Both are using pfSense as a firewall. The system was running fine until one location switched ISP’s. Now when calling the 1800 number it gives you two options. Location A and Location B. They also have their own local area code numbers configured too. Calling the 1800 number and selecting Option TWO will drop the call.

I think it has something to do with the caller ID being passed between the two PBX systems.

Verbosity was 0 and is now 6
  == CDR updated on SIP/dt-[COUCHTOMWC]-0000000e
    -- Executing [[email protected]:1] Goto("SIP/dt-[COUCHTOMWC]-0000000e", "ext-miscdests,6,1") in new stack
    -- Goto (ext-miscdests,6,1)
    -- Executing [[email protected]:1] NoOp("SIP/dt-[COUCHTOMWC]-0000000e", "MiscDest: Mobile-Web-Number-2-West-Chester") in new stack
    -- Executing [[email protected]:2] Goto("SIP/dt-[COUCHTOMWC]-0000000e", "from-internal,8001,1") in new stack
    -- Goto (from-internal,8001,1)
    -- Executing [[email protected]:1] Macro("SIP/dt-[COUCHTOMWC]-0000000e", "user-callerid,LIMIT,EXTERNAL,") in new stack
    -- Executing [[email protected]:1] Set("SIP/dt-[COUCHTOMWC]-0000000e", "TOUCH_MONITOR=1534913595.123") in new stack
    -- Executing [[email protected]:2] Set("SIP/dt-[COUCHTOMWC]-0000000e", "AMPUSER=[MYCELLPHON]") in new stack
    -- Executing [[email protected]:3] GotoIf("SIP/dt-[COUCHTOMWC]-0000000e", "0?report") in new stack
    -- Executing [[email protected]:4] ExecIf("SIP/dt-[COUCHTOMWC]-0000000e", "1?Set(REALCALLERIDNUM=[MYCELLPHON])") in new stack
    -- Executing [[email protected]:5] Set("SIP/dt-[COUCHTOMWC]-0000000e", "AMPUSER=") in new stack
    -- Executing [[email protected]:6] GotoIf("SIP/dt-[COUCHTOMWC]-0000000e", "0?limit") in new stack
    -- Executing [[email protected]:7] Set("SIP/dt-[COUCHTOMWC]-0000000e", "AMPUSERCIDNAME=") in new stack
    -- Executing [[email protected]:8] GotoIf("SIP/dt-[COUCHTOMWC]-0000000e", "1?report") in new stack
    -- Goto (macro-user-callerid,s,16)
    -- Executing [[email protected]:16] GotoIf("SIP/dt-[COUCHTOMWC]-0000000e", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,29)
    -- Executing [[email protected]:29] Set("SIP/dt-[COUCHTOMWC]-0000000e", "CALLERID(number)=[MYCELLPHON]") in new stack
    -- Executing [[email protected]:30] Set("SIP/dt-[COUCHTOMWC]-0000000e", "CALLERID(name)=[MYCELLPHON]") in new stack
    -- Executing [[email protected]:31] Set("SIP/dt-[COUCHTOMWC]-0000000e", "CDR(cnum)=[MYCELLPHON]") in new stack
    -- Executing [[email protected]:32] Set("SIP/dt-[COUCHTOMWC]-0000000e", "CDR(cnam)=[MYCELLPHON]") in new stack
    -- Executing [[email protected]:33] Set("SIP/dt-[COUCHTOMWC]-0000000e", "CHANNEL(language)=en") in new stack
    -- Executing [[email protected]:2] Set("SIP/dt-[COUCHTOMWC]-0000000e", "INTRACOMPANYROUTE=YES") in new stack
    -- Executing [[email protected]:3] Set("SIP/dt-[COUCHTOMWC]-0000000e", "MOHCLASS=default") in new stack
    -- Executing [[email protected]:4] Set("SIP/dt-[COUCHTOMWC]-0000000e", "_NODEST=") in new stack
    -- Executing [[email protected]:5] Gosub("SIP/dt-[COUCHTOMWC]-0000000e", "sub-record-check,s,1(out,8001,)") in new stack
    -- Executing [[email protected]:1] Set("SIP/dt-[COUCHTOMWC]-0000000e", "REC_POLICY_MODE_SAVE=") in new stack
    -- Executing [[email protected]:2] GotoIf("SIP/dt-[COUCHTOMWC]-0000000e", "1?check") in new stack
    -- Goto (sub-record-check,s,7)
    -- Executing [[email protected]:7] Set("SIP/dt-[COUCHTOMWC]-0000000e", "__MON_FMT=wav") in new stack
    -- Executing [[email protected]:8] GotoIf("SIP/dt-[COUCHTOMWC]-0000000e", "1?next") in new stack
    -- Goto (sub-record-check,s,11)
    -- Executing [[email protected]:11] ExecIf("SIP/dt-[COUCHTOMWC]-0000000e", "0?Return()") in new stack
    -- Executing [[email protected]:12] ExecIf("SIP/dt-[COUCHTOMWC]-0000000e", "0?Set(__REC_POLICY_MODE=)") in new stack
    -- Executing [[email protected]:13] GotoIf("SIP/dt-[COUCHTOMWC]-0000000e", "0?out,1") in new stack
    -- Executing [[email protected]:14] Set("SIP/dt-[COUCHTOMWC]-0000000e", "__REC_STATUS=INITIALIZED") in new stack
    -- Executing [[email protected]:15] Set("SIP/dt-[COUCHTOMWC]-0000000e", "NOW=1534913610") in new stack
    -- Executing [[email protected]:16] Set("SIP/dt-[COUCHTOMWC]-0000000e", "__DAY=22") in new stack
    -- Executing [[email protected]:17] Set("SIP/dt-[COUCHTOMWC]-0000000e", "__MONTH=08") in new stack
    -- Executing [[email protected]:18] Set("SIP/dt-[COUCHTOMWC]-0000000e", "__YEAR=2018") in new stack
    -- Executing [[email protected]:19] Set("SIP/dt-[COUCHTOMWC]-0000000e", "__TIMESTR=20180822-005330") in new stack
    -- Executing [[email protected]:20] Set("SIP/dt-[COUCHTOMWC]-0000000e", "__FROMEXTEN=[MYCELLPHON]") in new stack
    -- Executing [[email protected]:21] Set("SIP/dt-[COUCHTOMWC]-0000000e", "__CALLFILENAME=out-8001-[MYCELLPHON]-20180822-005330-1534913595.123") in new stack
    -- Executing [[email protected]:22] Goto("SIP/dt-[COUCHTOMWC]-0000000e", "out,1") in new stack
    -- Goto (sub-record-check,out,1)
    -- Executing [[email protected]:1] ExecIf("SIP/dt-[COUCHTOMWC]-0000000e", "1?Set(__REC_POLICY_MODE=)") in new stack
    -- Executing [[email protected]:2] GosubIf("SIP/dt-[COUCHTOMWC]-0000000e", "0?record,1(exten,8001,[MYCELLPHON])") in new stack
    -- Executing [[email protected]:3] Return("SIP/dt-[COUCHTOMWC]-0000000e", "") in new stack
    -- Executing [[email protected]:6] Macro("SIP/dt-[COUCHTOMWC]-0000000e", "dialout-trunk,4,8001,,off") in new stack
    -- Executing [[email protected]:1] Set("SIP/dt-[COUCHTOMWC]-0000000e", "DIAL_TRUNK=4") in new stack
    -- Executing [[email protected]:2] GosubIf("SIP/dt-[COUCHTOMWC]-0000000e", "0?sub-pincheck,s,1()") in new stack
    -- Executing [[email protected]:3] GotoIf("SIP/dt-[COUCHTOMWC]-0000000e", "0?disabletrunk,1") in new stack
    -- Executing [[email protected]:4] Set("SIP/dt-[COUCHTOMWC]-0000000e", "DIAL_NUMBER=8001") in new stack
    -- Executing [[email protected]:5] Set("SIP/dt-[COUCHTOMWC]-0000000e", "DIAL_TRUNK_OPTIONS=Ttr") in new stack
    -- Executing [[email protected]:6] Set("SIP/dt-[COUCHTOMWC]-0000000e", "OUTBOUND_GROUP=OUT_4") in new stack
    -- Executing [[email protected]:7] GotoIf("SIP/dt-[COUCHTOMWC]-0000000e", "0?nomax") in new stack
    -- Executing [[email protected]:8] GotoIf("SIP/dt-[COUCHTOMWC]-0000000e", "0?chanfull") in new stack
    -- Executing [[email protected]:9] GotoIf("SIP/dt-[COUCHTOMWC]-0000000e", "1?skipoutcid") in new stack
    -- Goto (macro-dialout-trunk,s,12)
    -- Executing [[email protected]:12] GosubIf("SIP/dt-[COUCHTOMWC]-0000000e", "0?sub-flp-4,s,1()") in new stack
    -- Executing [[email protected]:13] Set("SIP/dt-[COUCHTOMWC]-0000000e", "OUTNUM=8001") in new stack
    -- Executing [[email protected]:14] Set("SIP/dt-[COUCHTOMWC]-0000000e", "custom=IAX2/inter-company-iax") in new stack
    -- Executing [[email protected]:15] ExecIf("SIP/dt-[COUCHTOMWC]-0000000e", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default)Ttr)") in new stack
    -- Executing [[email protected]:16] ExecIf("SIP/dt-[COUCHTOMWC]-0000000e", "0?Set(DIAL_TRUNK_OPTIONS=TtrM(confirm))") in new stack
    -- Executing [[email protected]:17] Macro("SIP/dt-[COUCHTOMWC]-0000000e", "dialout-trunk-predial-hook,") in new stack
    -- Executing [[email protected]:1] MacroExit("SIP/dt-[COUCHTOMWC]-0000000e", "") in new stack
    -- Executing [[email protected]:18] GotoIf("SIP/dt-[COUCHTOMWC]-0000000e", "0?bypass,1") in new stack
    -- Executing [[email protected]:19] ExecIf("SIP/dt-[COUCHTOMWC]-0000000e", "0?Set(CONNECTEDLINE(num,i)=8001)") in new stack
    -- Executing [[email protected]:20] ExecIf("SIP/dt-[COUCHTOMWC]-0000000e", "0?Set(CONNECTEDLINE(name,i)=CID:[MYCELLPHON])") in new stack
    -- Executing [[email protected]:21] GotoIf("SIP/dt-[COUCHTOMWC]-0000000e", "0?customtrunk") in new stack
    -- Executing [[email protected]:22] Dial("SIP/dt-[COUCHTOMWC]-0000000e", "IAX2/inter-company-iax/8001,300,Ttr") in new stack

    -- Called IAX2/inter-company-iax/8001
    -- Call accepted by 10.0.0.2 (format ulaw)
    -- Format for call is ulaw
    -- IAX2/inter-company-iax-19368 is making progress passing it to SIP/dt-[COUCHTOMWC]-0000000e
    -- IAX2/inter-company-iax-19368 is making progress passing it to SIP/dt-[COUCHTOMWC]-0000000e
    -- IAX2/inter-company-iax-19368 is circuit-busy
    -- Hungup 'IAX2/inter-company-iax-19368'
  
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Executing [[email protected]:23] NoOp("SIP/dt-[COUCHTOMWC]-0000000e", "Dial failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 34") in new stack
    -- Executing [[email protected]:24] GotoIf("SIP/dt-[COUCHTOMWC]-0000000e", "0?continue,1:s-CONGESTION,1") in new stack
    -- Goto (macro-dialout-trunk,s-CONGESTION,1)
    -- Executing [[email protected]:1] Set("SIP/dt-[COUCHTOMWC]-0000000e", "RC=34") in new stack
    -- Executing [[email protected]:2] Goto("SIP/dt-[COUCHTOMWC]-0000000e", "34,1") in new stack
    -- Goto (macro-dialout-trunk,34,1)
    -- Executing [[email protected]:1] Goto("SIP/dt-[COUCHTOMWC]-0000000e", "continue,1") in new stack
    -- Goto (macro-dialout-trunk,continue,1)
    -- Executing [[email protected]:1] NoOp("SIP/dt-[COUCHTOMWC]-0000000e", "TRUNK Dial failed due to CONGESTiON HANGUPCAUSE: 34 - failing through to other trunks") in new stack
    -- Executing [[email protected]:2] Set("SIP/dt-[COUCHTOMWC]-0000000e", "CALLERID(number)=") in new stack
    -- Executing [[email protected]:7] Macro("SIP/dt-[COUCHTOMWC]-0000000e", "outisbusy,") in new stack
    -- Executing [[email protected]:1] Progress("SIP/dt-[COUCHTOMWC]-0000000e", "") in new stack
    -- Executing [[email protected]:2] GotoIf("SIP/dt-[COUCHTOMWC]-0000000e", "0?emergency,1") in new stack
    -- Executing [[email protected]:3] GotoIf("SIP/dt-[COUCHTOMWC]-0000000e", "1?intracompany,1") in new stack
    -- Goto (macro-outisbusy,intracompany,1)
    -- Executing [[email protected]:1] Playback("SIP/dt-[COUCHTOMWC]-0000000e", "all-circuits-busy-now&pls-tRy-call-later, noanswer") in new stack
    -- <SIP/dt-[COUCHTOMWC]-0000000e> Playing 'all-circuits-busy-now.ulaw' (language 'en')
    -- <SIP/dt-[COUCHTOMWC]-0000000e> Playing 'pls-try-call-later.ulaw' (language 'en')
    -- Executing [[email protected]:2] Congestion("SIP/dt-[COUCHTOMWC]-0000000e", "20") in new stack

localhost*CLI> core set verbose 0
localhost*CLI> sip set debug peer inter-company
SIP Debugging Enabled for IP: 192.168.5.2
<--- SIP read from UDP:192.168.5.2:5060 --->
OPTIONS sip:10.0.0.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.2:5060;branch=[brancsomething]
Max-Forwards: 70
From: "Unknown" <sip:[email protected]>;tag=[ShortTagCode]
To: <sip:10.0.0.2>
Contact: <sip:[email protected]:5060>
Call-ID: [LongRandomNumbers]@192.168.5.2:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.11.0(1.8.26.1)
Date: Sat, 25 Aug 2018 04:35:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Sending to 192.168.5.2:5060 (NAT)
Looking for s in from-sip-external (domain 10.0.0.2)

<--- Transmitting (NAT) to 192.168.5.2:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.5.2:5060;branch=[brancsomething];received=192.168.5.2;rport=5060
From: "Unknown" <sip:[email protected]>;tag=[ShortTagCode]
To: <sip:10.0.0.2>;tag=[ShortTagCode 2]
Call-ID: [LongRandomNumbers]@192.168.5.2:5060
CSeq: 102 OPTIONS
Server: FPBX-12.0.19(11.14.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:10.0.0.2:5060>
Accept: application/sdp
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '[LongRandomNumbers]@192.168.5.2:5060' in 32000 ms (Method: OPTIONS)

<--- SIP read from UDP:192.168.5.2:5060 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.5.2:5060;branch=[LongRandomNumbers 3]
Max-Forwards: 70
From: "[MY PHONE]" <sip:[MY PHONE]@192.168.5.2>;tag=[ShortTagCode 3]
To: <sip:[email protected]>
Contact: <sip:[MY PHONE]@192.168.5.2:5060>
Call-ID: [LongRandomNumbers 2]@192.168.5.2
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(1.8.26.1)
Date: Sat, 25 Aug 2018 04:35:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 235

v=0
o=root [UnknownNumber - New?] [UnknownNumber - New?] IN IP4 192.168.5.2
s=Asterisk PBX 1.8.26.1
c=IN IP4 192.168.5.2
t=0 0
m=audio 13510 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 11 lines) ---
Sending to 192.168.5.2:5060 (NAT)
Sending to 192.168.5.2:5060 (NAT)
Using INVITE request as basis request - [LongRandomNumbers 2]@192.168.5.2
Found peer 'inter-company' for '[MY PHONE]' from 192.168.5.2:5060
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.5.2:13510
Looking for 8001 in from-internal (domain 10.0.0.2)
list_route: hop: <sip:[MY PHONE]@192.168.5.2:5060>

<--- Transmitting (no NAT) to 192.168.5.2:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.5.2:5060;branch=[LongRandomNumbers 3];received=192.168.5.2
From: "[MY PHONE]" <sip:[MY PHONE]@192.168.5.2>;tag=[ShortTagCode 3]
To: <sip:[email protected]>
Call-ID: [LongRandomNumbers 2]@192.168.5.2
CSeq: 102 INVITE
Server: FPBX-12.0.19(11.14.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:[email protected]:5060>
Content-Length: 0


<------------>
Audio is at 11192
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Transmitting (no NAT) to 192.168.5.2:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.5.2:5060;branch=[LongRandomNumbers 3];received=192.168.5.2
From: "[MY PHONE]" <sip:[MY PHONE]@192.168.5.2>;tag=[ShortTagCode 3]
To: <sip:[email protected]>;tag=[ShortTagCode 4]
Call-ID: [LongRandomNumbers 2]@192.168.5.2
CSeq: 102 INVITE
Server: FPBX-12.0.19(11.14.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:[email protected]:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 226

v=0
o=root 883207303 883207303 IN IP4 10.0.0.2
s=Asterisk PBX 11.14.1
c=IN IP4 10.0.0.2
t=0 0
m=audio 11192 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
Reliably Transmitting (no NAT) to 192.168.5.2:5060:
OPTIONS sip:192.168.5.2 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.2:5060;branch=[brancsomething2]
Max-Forwards: 70
From: "Unknown" <sip:[email protected]>;tag=[ShortTagCode 5]
To: <sip:192.168.5.2>
Contact: <sip:[email protected]:5060>
Call-ID: [CALLER ID]@10.0.0.2:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-12.0.19(11.14.1)
Date: Sat, 25 Aug 2018 04:35:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:192.168.5.2:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.2:5060;branch=[brancsomething2];received=10.0.0.2;rport=5060
From: "Unknown" <sip:[email protected]>;tag=[ShortTagCode 5]
To: <sip:192.168.5.2>;tag=[ShortTagCode 6]
Call-ID: [CALLER ID]@10.0.0.2:5060
CSeq: 102 OPTIONS
Server: FPBX-2.11.0(1.8.26.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:192.168.5.2:5060>
Accept: application/sdp
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Really destroying SIP dialog '[CALLER ID]@10.0.0.2:5060' Method: OPTIONS

<--- Reliably Transmitting (no NAT) to 192.168.5.2:5060 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.5.2:5060;branch=[LongRandomNumbers 3];received=192.168.5.2
From: "[MY PHONE]" <sip:[MY PHONE]@192.168.5.2>;tag=[ShortTagCode 3]
To: <sip:[email protected]>;tag=[ShortTagCode 4]
Call-ID: [LongRandomNumbers 2]@192.168.5.2
CSeq: 102 INVITE
Server: FPBX-12.0.19(11.14.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Content-Length: 0


<------------>
[2018-08-25 00:35:39] WARNING[22067][C-000000b5]: channel.c:4860 ast_prod: Prodding channel 'SIP/inter-company-00000240' failed

<--- SIP read from UDP:192.168.5.2:5060 --->
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.2:5060;branch=[LongRandomNumbers 3]
Max-Forwards: 70
From: "[MY PHONE]" <sip:[MY PHONE]@192.168.5.2>;tag=[ShortTagCode 3]
To: <sip:[email protected]>;tag=[ShortTagCode 4]
Contact: <sip:[MY PHONE]@192.168.5.2:5060>
Call-ID: [LongRandomNumbers 2]@192.168.5.2
CSeq: 102 ACK
User-Agent: FPBX-2.11.0(1.8.26.1)
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '[LongRandomNumbers 2]@192.168.5.2' Method: ACK
localhost*CLI> sip set debug off
SIP Debugging Disabled
localhost*CLI>

Have you checked the IAX trunk between those FreePBXs? What happens if you try to call from one extension on A to an extension on B ?

Honestly im not sure, ive been given the issue: “When we press two it says ‘all circuits busy’ fix it but dont break it more”

:slight_smile:

Honestly this is just some learning expedition and i would gladly do any testing i can do (with some clear instructions or steps if possible.)

Right now i do not have have a way do call from extension A to extension B because i do not have a phone to do that.

I can try getting a softphone setup on the remote PC i use to access the PBX web GUI but i rather not do that just yet.

Idk if it matters but the two sites are running a VPN so i can say remote access Location A and still have access to Location B’s subnet. (10.0.0.2 and 192.168.5.2 private networks across towns.)

localhost*CLI> sip set debug peer inter-company
SIP Debugging Enabled for IP: 10.0.0.2
[2018-08-25 00:35:22] NOTICE[32115]: chan_sip.c:15094 check_auth: Correct auth, but based on stale nonce received from '"101" <sip:[email protected]>;tag=1731694264'
Reliably Transmitting (no NAT) to 10.0.0.2:5060:
OPTIONS sip:10.0.0.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.2:5060;branch=[Branch 3]
Max-Forwards: 70
From: "Unknown" <sip:[email protected]>;tag=[Tag SIP 4]
To: <sip:10.0.0.2>
Contact: <sip:[email protected]:5060>
Call-ID: [CID Code 0]@192.168.5.2:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.11.0(1.8.26.1)
Date: Sat, 25 Aug 2018 04:35:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:10.0.0.2:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.5.2:5060;branch=[Branch 3];received=192.168.5.2;rport=5060
From: "Unknown" <sip:[email protected]>;tag=[Tag SIP 4]
To: <sip:10.0.0.2>;tag=[Tag SIP 5]
Call-ID: [CID Code 0]@192.168.5.2:5060
CSeq: 102 OPTIONS
Server: FPBX-12.0.19(11.14.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:10.0.0.2:5060>
Accept: application/sdp
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Really destroying SIP dialog '[CID Code 0]@192.168.5.2:5060' Method: OPTIONS
[2018-08-25 00:35:27] NOTICE[32115]: chan_sip.c:15094 check_auth: Correct auth, but based on stale nonce received from '"105" <sip:[email protected]>;tag=[Tag 0]'
[2018-08-25 00:35:30] NOTICE[32115]: chan_sip.c:15094 check_auth: Correct auth, but based on stale nonce received from '"101" <sip:[email protected]>;tag=[Tag 1]'
Audio is at 13510
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.0.0.2:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.5.2:5060;branch=[Branc 0]
Max-Forwards: 70
From: "[MYCELL]" <sip:[MYCELL]@192.168.5.2>;tag=[Tag 0]
To: <sip:[email protected]>
Contact: <sip:[MYCELL]@192.168.5.2:5060>
Call-ID: [CID Code 1]@192.168.5.2
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(1.8.26.1)
Date: Sat, 25 Aug 2018 04:35:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 235

v=0
o=root 1242873871 1242873871 IN IP4 192.168.5.2
s=Asterisk PBX 1.8.26.1
c=IN IP4 192.168.5.2
t=0 0
m=audio 13510 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:10.0.0.2:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.5.2:5060;branch=[Branc 0];received=192.168.5.2
From: "[MYCELL]" <sip:[MYCELL]@192.168.5.2>;tag=[Tag 0]
To: <sip:[email protected]>
Call-ID: [CID Code 1]@192.168.5.2
CSeq: 102 INVITE
Server: FPBX-12.0.19(11.14.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:[email protected]:5060>
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---

<--- SIP read from UDP:10.0.0.2:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.5.2:5060;branch=[Branc 0];received=192.168.5.2
From: "[MYCELL]" <sip:[MYCELL]@192.168.5.2>;tag=[Tag 0]
To: <sip:[email protected]>;tag=[Tag SIP1]
Call-ID: [CID Code 1]@192.168.5.2
CSeq: 102 INVITE
Server: FPBX-12.0.19(11.14.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:[email protected]:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 226

v=0
o=root 883207303 883207303 IN IP4 10.0.0.2
s=Asterisk PBX 11.14.1
c=IN IP4 10.0.0.2
t=0 0
m=audio 11192 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 11 lines) ---
list_route: hop: <sip:[email protected]:5060>
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.0.0.2:11192
[2018-08-25 00:35:31] NOTICE[32115]: chan_sip.c:15094 check_auth: Correct auth, but based on stale nonce received from '"101" <sip:[email protected]>;tag=[Tag 2]'
[2018-08-25 00:35:31] NOTICE[32115]: chan_sip.c:15094 check_auth: Correct auth, but based on stale nonce received from '"101" <sip:[email protected]>;tag=[Tag 3]'
[2018-08-25 00:35:31] NOTICE[32115]: chan_sip.c:15094 check_auth: Correct auth, but based on stale nonce received from '"101" <sip:[email protected]>;tag=[Tag 4]'

<--- SIP read from UDP:10.0.0.2:5060 --->
OPTIONS sip:192.168.5.2 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.2:5060;branch=[Branch 1]
Max-Forwards: 70
From: "Unknown" <sip:[email protected]>;tag=[Tag SIP1]
To: <sip:192.168.5.2>
Contact: <sip:[email protected]:5060>
Call-ID: [CID Code 1]@10.0.0.2:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-12.0.19(11.14.1)
Date: Sat, 25 Aug 2018 04:35:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Looking for s in from-sip-external (domain 192.168.5.2)

<--- Transmitting (NAT) to 10.0.0.2:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.2:5060;branch=[Branch 1];received=10.0.0.2;rport=5060
From: "Unknown" <sip:[email protected]>;tag=[Tag SIP1]
To: <sip:192.168.5.2>;tag=[Tag SIP1]
Call-ID: [CID Code 1]@10.0.0.2:5060
CSeq: 102 OPTIONS
Server: FPBX-2.11.0(1.8.26.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:192.168.5.2:5060>
Accept: application/sdp
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '[CID Code 1]@10.0.0.2:5060' in 32000 ms (Method: OPTIONS)
[2018-08-25 00:35:32] NOTICE[32115]: chan_sip.c:15094 check_auth: Correct auth, but based on stale nonce received from '"101" <sip:[email protected]>;tag=[Tag 5]'
[2018-08-25 00:35:32] NOTICE[32115]: chan_sip.c:15094 check_auth: Correct auth, but based on stale nonce received from '"101" <sip:[email protected]>;tag=[Tag 6]'

<--- SIP read from UDP:10.0.0.2:5060 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.5.2:5060;branch=[Branc 0];received=192.168.5.2
From: "[MYCELL]" <sip:[MYCELL]@192.168.5.2>;tag=[Tag 0]
To: <sip:[email protected]>;tag=[Tag SIP1]
Call-ID: [CID Code 1]@192.168.5.2
CSeq: 102 INVITE
Server: FPBX-12.0.19(11.14.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
set_destination: Parsing <sip:[email protected]:5060> for address/port to send to
set_destination: set destination to 10.0.0.2:5060
Transmitting (no NAT) to 10.0.0.2:5060:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.2:5060;branch=[Branc 0]
Max-Forwards: 70
From: "[MYCELL]" <sip:[MYCELL]@192.168.5.2>;tag=[Tag 0]
To: <sip:[email protected]>;tag=[Tag SIP1]
Contact: <sip:[MYCELL]@192.168.5.2:5060>
Call-ID: [CID Code 1]@192.168.5.2
CSeq: 102 ACK
User-Agent: FPBX-2.11.0(1.8.26.1)
Content-Length: 0


---
Really destroying SIP dialog '[CID Code 1]@192.168.5.2' Method: INVITE
localhost*CLI> sip set debug off
SIP Debugging Disabled
[2018-08-25 00:36:00] NOTICE[24295]: pbx_spool.c:385 attempt_thread: Call completed to Local/[email protected]
[2018-08-25 00:36:02] NOTICE[24300]: pbx_spool.c:385 attempt_thread: Call completed to Local/[email protected]
localhost*CLI>

Here are some findings i want to note for myself but they might contain something incorrect.

192.168.5.2 - Location A
10.0.0.2 - Location B

192.168.5.2 Misc Destination: Mobile-Web-Number-2
In use by Objec:. Edit IVR: Mobile-Web-Number
In use by object: Route: Mobile-Web-Number

10.0.0.2 No Misc Destination like that. (Just notes not saying this is wrong)

Well i think i fixed it (though ill have to wait until they test it.)

It was

https://192.168.5.2/admin/config.php?display=miscdests&id=6
Misc Destination: Mobile-Web-Number-2--***
Delete Misc Destination Mobile-Web-Number-2--***
Used as Destination by 1 Object: IVR Mobile Web Number / Option 2

Description:Mobile-Web-Number-2--***
Dial:8001 (INCORRECT!)
Dial:5555555555

It was dialing 8001 in this part. I set it to the number of the second store. All seems ok now.

Can anyone help me determine what route this is going out? It might not be using the IAX that is in the setup and just using the SIP route. (If that doesnt make sense my bad i didnt configure this setup and there is Intra-company (also called Intra-company SIP) and intra-company-IAX.)

I might have IAX up and registered but i cant tell if its doing any work for the calls. I think NOT if its routing it out to the 10 digit phone number but i am not really sure. The ‘dialplan’ and how it alters the call still have me a bit confused. (as those are the only setting i just pasted in and they worked)

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