SOLVED: Weird Nextiva Inbound DID Issue

kwriley87 · July 8, 2015, 2:23am

Hello,

I have previous experience in using other carriers but I am having a weird issue with Nextiva and wanted to see if anyone else has heard of this.

I am running FreePBX Distro 12.0.70

I purchased 3 DIDs and 3 trunks from Nextiva… Typically when you add an inbound route into FreePBX, you just add the DID via the inbound routes module then point it to a destination… however, when I do that for my three DIDs… inbound and outbound calls to those DIDs to not work… Here is an output from the logs when I attempt to dial one of these DIDs:
– Executing [861167117@from-trunk:1] Set(“SIP/NEXTIVA-00000068”, “__FROM_DID=861167117”) in new stack – Executing [861167117@from-trunk:2] NoOp(“SIP/NEXTIVA-00000068”, “Received an unknown call with DID set to 861167117”) in new stack – Executing [861167117@from-trunk:3] Goto(“SIP/NEXTIVA-00000068”, “s,a2”) in new stack – Goto (from-trunk,s,2) – Executing [s@from-trunk:2] Answer(“SIP/NEXTIVA-00000068”, “”) in new stack > 0xb7338360 – Probation passed - setting RTP source address to 208.73.146.93:19704 – Executing [s@from-trunk:3] Log(“SIP/NEXTIVA-00000068”, “WARNING,Friendly Scanner from 208.73.146.93”) in new stack [2015-07-07 21:20:33] WARNING[32088][C-00000043]: Ext. s:3 @ from-trunk: Friendly Scanner from 208.73.146.93 – Executing [s@from-trunk:4] Wait(“SIP/NEXTIVA-00000068”, “2”) in new stack – Executing [s@from-trunk:5] Playback(“SIP/NEXTIVA-00000068”, “ss-noservice”) in new stack – <SIP/NEXTIVA-00000068> Playing ‘ss-noservice.ulaw’ (language ‘en’)
I get a “The number you have dialed is not in service” message.

However, when I add a DID and use my Nextiva account number as the DID and point it somewhere, all of a sudden my 3 numbers are working… but they all go whatever destination I specified for the DID that is my Nextiva account number…

What is the deal with that? I’ve never seen that before… What if I don’t want all of my Nextiva DIDs to point to the same destination??? Nextiva is next to no help, hopefully someone has an idea what is going on here.

Thanks,
Kenny

dicko · July 8, 2015, 5:05am

Generally if you have multiple DID from the same provider, then you should arrange with the provider that all calls arrive on the same trunk, if they don’t then you might need to parse out the SIP headers for the individual calls to separate them by the unnecessary/spurious trunks which are all registered/sent to the same IP and that is likely going to be a problem.

kwriley87 · July 8, 2015, 1:41pm

Hi dicko

Let me correct myself, I may have not worded things correctly… I have one trunk with Nextiva, with 3 channels… so not exactly three trunks.

Do I need to call Nextiva again? Honestly their tech support isn’t of much help. If the DID hits my PBX, they are pretty much not responsible on their end.

dicko · July 8, 2015, 2:43pm

Sorry I have no expereience with them. I would turn sip debugging on and investigate the headers.

kwriley87 · July 8, 2015, 3:48pm

How would I manage to do that? Sorry… I have experience with FreePBX, but not the point that I consider myself an expert, especially at the CLI

dicko · July 8, 2015, 3:50pm

From the cli
sip set debug on

kwriley87 · July 8, 2015, 4:03pm

Ok here is the output…

<— SIP read from UDP:208.73.146.93:5060 —>
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 208.73.146.93:5060;branch=z9hG4bK7kif4s00704g95tj30t1.1
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed
Call-Id: [email protected]
Contact: sip:[email protected]:5060;transport=udp
Content-Disposition: session; handling=required
Content-Length: 306
Content-Type: application/sdp
CSeq: 1 INVITE
From: “BUSINESS SOLUTI” sip:[email protected]:5060;tag=gK046daa24
In-Reply-To: [email protected]
Supported: timer
To: sip:[email protected]:5060
Max-Forwards: 70

v=0
o=Sonus_UAC 10065 58 IN IP4 208.73.146.93
s=SIP Media Capabilities
c=IN IP4 208.73.146.93
t=0 0
m=audio 24946 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=maxptime:20
<------------->
— (14 headers 14 lines) —
Sending to 208.73.146.93:5060 (NAT)
Sending to 208.73.146.93:5060 (NAT)
Using INVITE request as basis request - [email protected]
Found peer ‘NEXTIVA’ for ‘+19728147186’ from 208.73.146.93:5060
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw), peer - audio=(ulaw|alaw|g729)/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 208.73.146.93:24946
Looking for 861167117 in from-trunk (domain 107.155.85.31)
list_route: hop: sip:[email protected]:5060;transport=udp

<— Transmitting (no NAT) to 208.73.146.93:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 208.73.146.93:5060;branch=z9hG4bK7kif4s00704g95tj30t1.1;received=208.73.146.93
From: “BUSINESS SOLUTI” sip:[email protected]:5060;tag=gK046daa24
To: sip:[email protected]:5060
Call-ID: [email protected]
CSeq: 1 INVITE
Server: FPBX-12.0.70(11.15.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Contact: sip:[email protected]:5060
Content-Length: 0

<------------>
– Executing [861167117@from-trunk:1] Set(“SIP/NEXTIVA-00000081”, “__FROM_DID=861167117”) in new stack
– Executing [861167117@from-trunk:2] NoOp(“SIP/NEXTIVA-00000081”, “Received an unknown call with DID set to 861167117”) in new stack
– Executing [861167117@from-trunk:3] Goto(“SIP/NEXTIVA-00000081”, “s,a2”) in new stack
– Goto (from-trunk,s,2)
– Executing [s@from-trunk:2] Answer(“SIP/NEXTIVA-00000081”, “”) in new stack
Audio is at 15720
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Reliably Transmitting (no NAT) to 208.73.146.93:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 208.73.146.93:5060;branch=z9hG4bK7kif4s00704g95tj30t1.1;received=208.73.146.93
From: “BUSINESS SOLUTI” sip:[email protected]:5060;tag=gK046daa24
To: sip:[email protected]:5060;tag=as4878edd1
Call-ID: [email protected]
CSeq: 1 INVITE
Server: FPBX-12.0.70(11.15.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Content-Length: 236

v=0
o=root 207480661 207480661 IN IP4 107.155.85.31
s=Asterisk PBX 11.15.0
c=IN IP4 107.155.85.31
t=0 0
m=audio 15720 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

<------------>
> 0xb7345060 – Probation passed - setting RTP source address to 208.73.146.93:24946
– Executing [s@from-trunk:3] Log(“SIP/NEXTIVA-00000081”, “WARNING,Friendly Scanner from 208.73.146.93”) in new stack
[2015-07-08 11:00:42] WARNING[23146][C-0000004f]: Ext. s:3 @ from-trunk: Friendly Scanner from 208.73.146.93
– Executing [s@from-trunk:4] Wait(“SIP/NEXTIVA-00000081”, “2”) in new stack

<— SIP read from UDP:208.73.146.93:5060 —>
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 208.73.146.93:5060;branch=z9hG4bKdsj39m308otgfucse0u1.1
Call-Id: [email protected]
Content-Length: 0
CSeq: 1 ACK
From: “BUSINESS SOLUTI” sip:[email protected]:5060;tag=gK046daa24
To: sip:[email protected]:5060;tag=as4878edd1
Max-Forwards: 70

<------------->
— (8 headers 0 lines) —
– Executing [s@from-trunk:5] Playback(“SIP/NEXTIVA-00000081”, “ss-noservice”) in new stack
– <SIP/NEXTIVA-00000081> Playing ‘ss-noservice.ulaw’ (language ‘en’)

<— SIP read from UDP:71.96.207.234:30910 —>

<------------->
– Executing [s@from-trunk:6] SayAlpha(“SIP/NEXTIVA-00000081”, “861167117”) in new stack
– <SIP/NEXTIVA-00000081> Playing ‘digits/8.ulaw’ (language ‘en’)
– <SIP/NEXTIVA-00000081> Playing ‘digits/6.ulaw’ (language ‘en’)
– <SIP/NEXTIVA-00000081> Playing ‘digits/1.ulaw’ (language ‘en’)
– <SIP/NEXTIVA-00000081> Playing ‘digits/1.ulaw’ (language ‘en’)
– <SIP/NEXTIVA-00000081> Playing ‘digits/6.ulaw’ (language ‘en’)
– <SIP/NEXTIVA-00000081> Playing ‘digits/7.ulaw’ (language ‘en’)
– <SIP/NEXTIVA-00000081> Playing ‘digits/1.ulaw’ (language ‘en’)
– <SIP/NEXTIVA-00000081> Playing ‘digits/1.ulaw’ (language ‘en’)
– <SIP/NEXTIVA-00000081> Playing ‘digits/7.ulaw’ (language ‘en’)
– Executing [s@from-trunk:7] Hangup(“SIP/NEXTIVA-00000081”, “”) in new stack
== Spawn extension (from-trunk, s, 7) exited non-zero on ‘SIP/NEXTIVA-00000081’
– Executing [h@from-trunk:1] Macro(“SIP/NEXTIVA-00000081”, “hangupcall,”) in new stack
– Executing [s@macro-hangupcall:1] ExecIf(“SIP/NEXTIVA-00000081”, “0?Set(CDR(recordingfile)=.)”) in new stack
– Executing [s@macro-hangupcall:2] GotoIf(“SIP/NEXTIVA-00000081”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,4)
– Executing [s@macro-hangupcall:4] Hangup(“SIP/NEXTIVA-00000081”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 4) exited non-zero on ‘SIP/NEXTIVA-00000081’ in macro ‘hangupcall’
== Spawn extension (from-trunk, h, 1) exited non-zero on 'SIP/NEXTIVA-00000081’
Scheduling destruction of SIP dialog ‘[email protected]’ in 6400 ms (Method: ACK)
set_destination: Parsing sip:[email protected]:5060;transport=udp for address/port to send to
set_destination: set destination to 208.73.146.93:5060
Reliably Transmitting (no NAT) to 208.73.146.93:5060:
BYE sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 107.155.85.31:5060;branch=z9hG4bK1ba150a6
Max-Forwards: 70
From: sip:[email protected]:5060;tag=as4878edd1
To: “BUSINESS SOLUTI” sip:[email protected]:5060;tag=gK046daa24
Call-ID: [email protected]
CSeq: 102 BYE
User-Agent: FPBX-12.0.70(11.15.0)
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

<— SIP read from UDP:208.73.146.93:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 107.155.85.31:5060;branch=z9hG4bK1ba150a6
From: sip:[email protected]:5060;tag=as4878edd1
To: “BUSINESS SOLUTI” sip:[email protected]:5060;tag=gK046daa24
Call-ID: [email protected]
CSeq: 102 BYE
Content-Length: 0

<------------->
— (7 headers 0 lines) —
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog ‘[email protected]’ Method: ACK
pbx*CLI>
Disconnected from Asterisk server
Asterisk cleanly ending (0).
Executing last minute cleanups
[root@pbx ~]#

I just don’t understand this… I have my three 3 DIDs in my inbound routes pointed to different destinations. Here is one of my DIDs that I’m calling: 972-295-9324, but when it hits my PBX, it’s trying to dial my Nextiva account number: 861167117, not the DID that I’m actually dialing. Does this make sense?

dicko · July 8, 2015, 4:12pm

Presmably in

Found peer ‘NEXTIVA’ for ‘+19728147186’ from 208.73.146.93:5060

Use from-pstn-toheader in extensions.conf modified if necessary as your trunk context

kwriley87 · July 8, 2015, 4:23pm

My trunk name is Nextiva, the 9728147186 is the number I’m calling from, and hostname for my trunk is trunking.voipdnsserers.com which resolves to 208.73.146.93

I made the edit to extensions.conf that you suggested, this is what it looks like now…

; from-trunk:
;
; Context is really just an aliax of from-pstn
;
[from-trunk]
include => from-pstn-toheader

Here is the output of the debugging logs when I attempted to call 972-295-9324 and got the same the number you have called is not in service message:

<— SIP read from UDP:208.73.146.93:5060 —>
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 208.73.146.93:5060;branch=z9hG4bK1rf72v305gug5s80m5f1.1
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-relay, multipart/mixed
Call-Id: [email protected]
Contact: sip:[email protected]:5060;transport=udp
Content-Disposition: session; handling=required
Content-Length: 308
Content-Type: application/sdp
CSeq: 1 INVITE
From: “BUSINESS SOLUTI” sip:[email protected]:5060;tag=gK0d39c9c9
In-Reply-To: [email protected]
Supported: timer
To: sip:[email protected]:5060
Max-Forwards: 70

v=0
o=Sonus_UAC 8746 24811 IN IP4 208.73.146.93
s=SIP Media Capabilities
c=IN IP4 208.73.146.93
t=0 0
m=audio 31270 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=maxptime:20
<------------->
— (14 headers 14 lines) —
Sending to 208.73.146.93:5060 (NAT)
Sending to 208.73.146.93:5060 (NAT)
Using INVITE request as basis request - [email protected]
Found peer ‘NEXTIVA’ for ‘+19728147186’ from 208.73.146.93:5060
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw), peer - audio=(ulaw|alaw|g729)/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 208.73.146.93:31270
Looking for 861167117 in from-pstn (domain 107.155.85.31)
list_route: hop: sip:[email protected]:5060;transport=udp

<— Transmitting (no NAT) to 208.73.146.93:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 208.73.146.93:5060;branch=z9hG4bK1rf72v305gug5s80m5f1.1;received=208.73.146.93
From: “BUSINESS SOLUTI” sip:[email protected]:5060;tag=gK0d39c9c9
To: sip:[email protected]:5060
Call-ID: [email protected]
CSeq: 1 INVITE
Server: FPBX-12.0.70(11.15.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Contact: sip:[email protected]:5060
Content-Length: 0

<------------>
– Executing [861167117@from-pstn:1] Set(“SIP/NEXTIVA-00000001”, “__FROM_DID=861167117”) in new stack
– Executing [861167117@from-pstn:2] NoOp(“SIP/NEXTIVA-00000001”, “Received an unknown call with DID set to 861167117”) in new stack
– Executing [861167117@from-pstn:3] Goto(“SIP/NEXTIVA-00000001”, “s,a2”) in new stack
– Goto (from-pstn,s,2)
– Executing [s@from-pstn:2] Answer(“SIP/NEXTIVA-00000001”, “”) in new stack
Audio is at 14606
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Reliably Transmitting (no NAT) to 208.73.146.93:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 208.73.146.93:5060;branch=z9hG4bK1rf72v305gug5s80m5f1.1;received=208.73.146.93
From: “BUSINESS SOLUTI” sip:[email protected]:5060;tag=gK0d39c9c9
To: sip:[email protected]:5060;tag=as30ed81f6
Call-ID: [email protected]
CSeq: 1 INVITE
Server: FPBX-12.0.70(11.15.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Contact: sip:[email protected]:5060
Content-Type: application/sdp
Content-Length: 236

v=0
o=root 992107094 992107094 IN IP4 107.155.85.31
s=Asterisk PBX 11.15.0
c=IN IP4 107.155.85.31
t=0 0
m=audio 14606 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:208.73.146.93:5060 —>
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 208.73.146.93:5060;branch=z9hG4bK8jnu82306gd1sqojp471.1
Call-Id: [email protected]
Content-Length: 0
CSeq: 1 ACK
From: “BUSINESS SOLUTI” sip:[email protected]:5060;tag=gK0d39c9c9
To: sip:[email protected]:5060;tag=as30ed81f6
Max-Forwards: 70

<------------->
— (8 headers 0 lines) —
> 0xb7507050 – Probation passed - setting RTP source address to 208.73.146.93:31270
– Executing [s@from-pstn:3] Log(“SIP/NEXTIVA-00000001”, “WARNING,Friendly Scanner from 208.73.146.93”) in new stack
[2015-07-08 11:19:58] WARNING[23929][C-00000001]: Ext. s:3 @ from-pstn: Friendly Scanner from 208.73.146.93
– Executing [s@from-pstn:4] Wait(“SIP/NEXTIVA-00000001”, “2”) in new stack
– Executing [s@from-pstn:5] Playback(“SIP/NEXTIVA-00000001”, “ss-noservice”) in new stack
– <SIP/NEXTIVA-00000001> Playing ‘ss-noservice.ulaw’ (language ‘en’)
– Executing [s@from-pstn:6] SayAlpha(“SIP/NEXTIVA-00000001”, “861167117”) in new stack
– <SIP/NEXTIVA-00000001> Playing ‘digits/8.ulaw’ (language ‘en’)
– <SIP/NEXTIVA-00000001> Playing ‘digits/6.ulaw’ (language ‘en’)
Reliably Transmitting (NAT) to 71.96.207.234:30910:
OPTIONS sip:[email protected]:30910 SIP/2.0
Via: SIP/2.0/UDP 107.155.85.31:5060;branch=z9hG4bK20a9c92b;rport
Max-Forwards: 70
From: “Unknown” sip:[email protected];tag=as135e6ea3
To: sip:[email protected]:30910
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-12.0.70(11.15.0)
Date: Wed, 08 Jul 2015 16:20:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Length: 0

<— SIP read from UDP:71.96.207.234:30910 —>
SIP/2.0 405 Method Not Allowed
Via: SIP/2.0/UDP 107.155.85.31:5060;branch=z9hG4bK20a9c92b;rport=5060
To: sip:[email protected]:30910;tag=325e204d
From: “Unknown” sip:[email protected];tag=as135e6ea3
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, MESSAGE
Content-Length: 0

<------------->
— (8 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]:5060’ Method: OPTIONS
– <SIP/NEXTIVA-00000001> Playing ‘digits/1.ulaw’ (language ‘en’)
Reliably Transmitting (no NAT) to 208.73.146.93:5060:
OPTIONS sip:trunking.voipdnsservers.com SIP/2.0
Via: SIP/2.0/UDP 107.155.85.31:5060;branch=z9hG4bK549ce56f
Max-Forwards: 70
From: “Unknown” sip:[email protected];tag=as65e745fb
To: sip:trunking.voipdnsservers.com
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-12.0.70(11.15.0)
Date: Wed, 08 Jul 2015 16:20:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Length: 0

<— SIP read from UDP:208.73.146.93:5060 —>
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 107.155.85.31:5060;branch=z9hG4bK549ce56f
From: “Unknown” sip:[email protected];tag=as65e745fb
To: sip:trunking.voipdnsservers.com;tag=aprqngfrt-kq4c0i30000c6
Call-ID: [email protected]:5060
CSeq: 102 OPTIONS

<------------->
— (6 headers 0 lines) —
Really destroying SIP dialog ‘[email protected]:5060’ Method: OPTIONS
– <SIP/NEXTIVA-00000001> Playing ‘digits/1.ulaw’ (language ‘en’)
– <SIP/NEXTIVA-00000001> Playing ‘digits/6.ulaw’ (language ‘en’)
– <SIP/NEXTIVA-00000001> Playing ‘digits/7.ulaw’ (language ‘en’)
– <SIP/NEXTIVA-00000001> Playing ‘digits/1.ulaw’ (language ‘en’)
– <SIP/NEXTIVA-00000001> Playing ‘digits/1.ulaw’ (language ‘en’)
– <SIP/NEXTIVA-00000001> Playing ‘digits/7.ulaw’ (language ‘en’)
– Executing [s@from-pstn:7] Hangup(“SIP/NEXTIVA-00000001”, “”) in new stack
== Spawn extension (from-pstn, s, 7) exited non-zero on ‘SIP/NEXTIVA-00000001’
– Executing [h@from-pstn:1] Macro(“SIP/NEXTIVA-00000001”, “hangupcall,”) in new stack
– Executing [s@macro-hangupcall:1] ExecIf(“SIP/NEXTIVA-00000001”, “0?Set(CDR(recordingfile)=.)”) in new stack
– Executing [s@macro-hangupcall:2] GotoIf(“SIP/NEXTIVA-00000001”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,4)
– Executing [s@macro-hangupcall:4] Hangup(“SIP/NEXTIVA-00000001”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 4) exited non-zero on ‘SIP/NEXTIVA-00000001’ in macro ‘hangupcall’
== Spawn extension (from-pstn, h, 1) exited non-zero on 'SIP/NEXTIVA-00000001’
Scheduling destruction of SIP dialog ‘[email protected]’ in 6400 ms (Method: ACK)
set_destination: Parsing sip:[email protected]:5060;transport=udp for address/port to send to
set_destination: set destination to 208.73.146.93:5060
Reliably Transmitting (no NAT) to 208.73.146.93:5060:
BYE sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 107.155.85.31:5060;branch=z9hG4bK78d7d9e0
Max-Forwards: 70
From: sip:[email protected]:5060;tag=as30ed81f6
To: “BUSINESS SOLUTI” sip:[email protected]:5060;tag=gK0d39c9c9
Call-ID: [email protected]
CSeq: 102 BYE
User-Agent: FPBX-12.0.70(11.15.0)
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

<— SIP read from UDP:208.73.146.93:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 107.155.85.31:5060;branch=z9hG4bK78d7d9e0
From: sip:[email protected]:5060;tag=as30ed81f6
To: “BUSINESS SOLUTI” sip:[email protected]:5060;tag=gK0d39c9c9
Call-ID: [email protected]
CSeq: 102 BYE
Content-Length: 0

<------------->
— (7 headers 0 lines) —
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog ‘[email protected]’ Method: ACK
pbx*CLI>
Disconnected from Asterisk server
Asterisk cleanly ending (0).
Executing last minute cleanups
[root@pbx ~]#

kwriley87 · July 8, 2015, 4:29pm

Wait you fixed it! I changed my peer details of my trunk from context from trunk to pstn while I was troubleshooting a few minutes ago, I changed it back context from trunk after I made your changes and it works. Wow! You are awesome! Thank you so much! Would you mind explaining the fix to me so I can take something away from this experience and understand what happened?

dicko · July 8, 2015, 4:37pm

Never edit extensions.conf read the first few lines but it does contain

; The context is designed for providers who send the DID in the TO: SIP header
; only. The format of this header is:
;
; To: <sip:[email protected]>
;
; So the DID must be extracted between the sip: and the @, which this does
;
[from-pstn-toheader]
exten => _.,1,Goto(from-pstn,${CUT(CUT(SIP_HEADER(To),@,1),:,2)},1)
;-------------------------------------------------------------------------------

kwriley87 · July 8, 2015, 4:45pm

Ok I reverted the changes I made to the extensions.conf file, I just misunderstood what you were asking me to do. As I said, I’m no expert Just learning as I go… As of now, the only change I made was to the context of my trunk… it was context=from-trunk and was changed to context=from-pstn-toheader and that solved my issue.

I’m not sure I understand how that fixed it but I’m glad it’s fixed. Thank you so much for your help.

faisalkhan · July 17, 2018, 9:26pm

Hey dicko,

Man you are great!

i am also having same issue on nextiva.

do you want to add this to “context=from-pstn-toheader” in the peer details of the Trunk Settings.