Digium PRI Gateway and FindMe/FollowMe

Trying to use a manual FindMe within my dialplan which will have the person’s desk phone and at least their cell phone. When I use Dial(SIP/desk-phone-exten&SIP/gateway-ip/cell-phone-number) or Dial(SIP/desk-phone-exten&SIP/cell-phone-number@gateway-ip), it rings their desk phone briefly, but as soon as the call is picked up by the gateway, it stops ringing the desk phone. I assume that the gateway is Answering the call before it routes it out the PRI. Once the call is answered, it stops ringing other phones.

Is there a way to stop the gateway from answering the sip call so that both phones could ring simultaneously? My next step is to backup the config from the gateway, and modify the dialplan (if that is indeed the problem), then restore the modified config to the gateway.

Thanks in advance.
Mark

Possibly, but you haven’t even said what the gateway is. You will have to do it on the gateway.

The 2, 4, and 8 ports are all the same, but it’s the following model…

http://www.voipsupply.com/digium-1g400f

I can’t find an option for “Don’t answer first” in the web config. I should be able to modify the dialplan and upload it back so long as I don’t factory reset the gateway.

If anyone has other suggestions, I’m open to them. Thanks.

Well, I looked at the dialplan on the Digium Gateway, and there is no Answer(). I ran a packet trace and found that the gateway sends a SIP/SDP message, and immediately the server sends and ACK to the gateway and a CANCEL to the internal ringing phone. I’m just using a ringgroup for testing this…

SIP/11000&SIP/{external number}@{the gateway}

As soon as the external phone starts to ring, the internal (11000) phone stops ringing. Maybe there’s a sip setting I’m missing.

Here’s the whole sip debug…I replaced the outside number with “{outside number replaced}”.
<------------>
– Executing [3270@from-internal-sip:1] Gosub(“SIP/1000-0000001e”, “subInternalDialer,s,1(3270)”) in new stack
– Executing [s@subInternalDialer:1] NoOp(“SIP/1000-0000001e”, “CALLERID = 1000 ARG1 = 3270”) in new stack
– Executing [s@subInternalDialer:2] Set(“SIP/1000-0000001e”, “caller=1000”) in new stack
– Executing [s@subInternalDialer:3] Set(“SIP/1000-0000001e”, “dialed=3270”) in new stack
– Executing [s@subInternalDialer:4] Set(“SIP/1000-0000001e”, “ARRAY(cluster,ext,devicetype,devopt,vmbox,findfollow,sectodial,forwardflag,forwardnum)=asterisk-nuc,3270,ringgroup,0,20,0,”) in new stack
> Found no rows [select ext_lockout from extensions where ext_dialed=‘1000’ or ext_did=‘1000’]
– Executing [s@subInternalDialer:5] Set(“SIP/1000-0000001e”, “lockout=”) in new stack
– Executing [s@subInternalDialer:6] Set(“SIP/1000-0000001e”, “canbepickedup=0”) in new stack
– Executing [s@subInternalDialer:7] GotoIf(“SIP/1000-0000001e”, “0?callforward,1”) in new stack
– Executing [s@subInternalDialer:8] GotoIf(“SIP/1000-0000001e”, “1?skippickupmark”) in new stack
– Goto (subInternalDialer,s,10)
– Executing [s@subInternalDialer:10] GotoIf(“SIP/1000-0000001e”, “0?invalidext”) in new stack
– Executing [s@subInternalDialer:11] GotoIf(“SIP/1000-0000001e”, “0?lockout”) in new stack
– Executing [s@subInternalDialer:12] GotoIf(“SIP/1000-0000001e”, “0?findme”) in new stack
– Executing [s@subInternalDialer:13] GotoIf(“SIP/1000-0000001e”, “0?followme”) in new stack
– Executing [s@subInternalDialer:14] GotoIf(“SIP/1000-0000001e”, “0?remoteSystem”) in new stack
– Executing [s@subInternalDialer:15] GotoIf(“SIP/1000-0000001e”, “0?phonedial”) in new stack
– Executing [s@subInternalDialer:16] Goto(“SIP/1000-0000001e”, “ringgroup,s,1”) in new stack
– Goto (ringgroup,s,1)
– Executing [s@ringgroup:1] Set(“SIP/1000-0000001e”, “rg_data=SIP/11000&SIP/{outside number replaced}@gateway-wh01”) in new stack
– Executing [s@ringgroup:2] GotoIf(“SIP/1000-0000001e”, “0?hangup”) in new stack
– Executing [s@ringgroup:3] Set(“SIP/1000-0000001e”, “DEVICE_STATE(Custom:3270@hints)=RINGING”) in new stack
– Executing [s@ringgroup:4] Dial(“SIP/1000-0000001e”, “SIP/11000&SIP/{outside number replaced}@gateway-wh01”) in new stack
== Using SIP RTP CoS mark 5
== Using SIP RTP CoS mark 5
Audio is at 15540
Adding codec g722 to SDP
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 172.17.106.1:5061:
INVITE sip:11000@172.17.106.1:5061 SIP/2.0
Via: SIP/2.0/UDP 172.17.250.1:5060;branch=z9hG4bK58bbdaaa
Max-Forwards: 70
From: sip:1000@172.17.250.1;tag=as7c2f3c50
To: sip:11000@172.17.106.1:5061
Contact: sip:1000@172.17.250.1:5060
Call-ID: 43d4e4544d45045c1891ef2822488cf9@172.17.250.1:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 13.5.0
Date: Mon, 04 Apr 2016 21:56:16 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 322

v=0
o=root 2021301186 2021301186 IN IP4 172.17.250.1
s=Asterisk PBX 13.5.0
c=IN IP4 172.17.250.1
t=0 0
m=audio 15540 RTP/AVP 9 0 8 3 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv


-- Called SIP/11000

set_destination: Parsing sip:11000@172.17.106.1:5061 for address/port to send to
set_destination: set destination to 172.17.106.1:5061
Audio is at 13578
Adding codec g722 to SDP
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 172.17.106.1:5061:
NOTIFY sip:11000@172.17.106.1:5061 SIP/2.0
Via: SIP/2.0/UDP 172.17.250.1:5060;branch=z9hG4bK788f862b
Max-Forwards: 70
From: sip:11000@172.17.250.1;tag=as32f31f6a
To: “11000” sip:11000@172.17.250.1;tag=299345b82257f737
Contact: sip:11000@172.17.250.1:5060
Call-ID: e4ef9ed6-5a5fc4fd@172.17.106.1
CSeq: 252 NOTIFY
User-Agent: Asterisk PBX 13.5.0
Subscription-State: active
Event: dialog
Content-Type: application/dialog-info+xml
Content-Length: 226

<?xml version="1.0"?> early

== Extension Changed 11000[hints] new state Ringing for Notify User 11000
Reliably Transmitting (no NAT) to 172.30.79.204:5060:
INVITE sip:{outside number replaced}@172.30.79.204:5060 SIP/2.0
Via: SIP/2.0/UDP 172.17.250.1:5060;branch=z9hG4bK5973b88e
Max-Forwards: 70
From: sip:1000@172.17.250.1;tag=as293e7775
To: <sip:{outside number replaced}@172.30.79.204:5060>
Contact: sip:1000@172.17.250.1:5060
Call-ID: 4958e88c38f418cc4e3619d64049f505@172.17.250.1:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 13.5.0
Date: Mon, 04 Apr 2016 21:56:16 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 322

v=0
o=root 1538211407 1538211407 IN IP4 172.17.250.1
s=Asterisk PBX 13.5.0
c=IN IP4 172.17.250.1
t=0 0
m=audio 13578 RTP/AVP 9 0 8 3 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv


-- Called SIP/{outside number replaced}@gateway-wh01
-- SIP/gateway-wh01-00000020 connected line has changed. Saving it until answer for SIP/1000-0000001e
-- SIP/11000-0000001f connected line has changed. Saving it until answer for SIP/1000-0000001e

<— SIP read from UDP:172.30.79.204:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.17.250.1:5060;branch=z9hG4bK5973b88e;received=172.17.250.1
From: sip:1000@172.17.250.1;tag=as293e7775
To: <sip:{outside number replaced}@172.30.79.204:5060>
Call-ID: 4958e88c38f418cc4e3619d64049f505@172.17.250.1:5060
CSeq: 102 INVITE
Server: Digium Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:{outside number replaced}@172.30.79.204:5060>
Content-Length: 0

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

<— SIP read from UDP:172.17.106.1:5061 —>
SIP/2.0 100 Trying
To: sip:11000@172.17.106.1:5061
From: sip:1000@172.17.250.1;tag=as7c2f3c50
Call-ID: 43d4e4544d45045c1891ef2822488cf9@172.17.250.1:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 172.17.250.1:5060;branch=z9hG4bK58bbdaaa
Server: Cisco/SPA508G-7.5.5
Content-Length: 0

<------------->
— (8 headers 0 lines) —

<— SIP read from UDP:172.17.106.1:5061 —>
SIP/2.0 200 OK
To: “11000” sip:11000@172.17.250.1;tag=299345b82257f737
From: sip:11000@172.17.250.1;tag=as32f31f6a
Call-ID: e4ef9ed6-5a5fc4fd@172.17.106.1
CSeq: 252 NOTIFY
Via: SIP/2.0/UDP 172.17.250.1:5060;branch=z9hG4bK788f862b
Server: Cisco/SPA508G-7.5.5
Content-Length: 0

<------------->
— (8 headers 0 lines) —

<— SIP read from UDP:172.17.106.1:5061 —>
SIP/2.0 180 Ringing
To: sip:11000@172.17.106.1:5061;tag=c500c680a5a765cfi0
From: sip:1000@172.17.250.1;tag=as7c2f3c50
Call-ID: 43d4e4544d45045c1891ef2822488cf9@172.17.250.1:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 172.17.250.1:5060;branch=z9hG4bK58bbdaaa
Contact: “11000” sip:11000@172.17.106.1:5061
Server: Cisco/SPA508G-7.5.5
Content-Length: 0

<------------->
— (9 headers 0 lines) —
sip_route_dump: route/path hop: sip:11000@172.17.106.1:5061
– SIP/11000-0000001f is ringing

<— Transmitting (no NAT) to 192.168.247.247:9000 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.247.247:9000;branch=z9hG4bK1044022597;received=192.168.247.247;rport=9000
From: sip:1000@172.17.250.1;tag=262586417
To: sip:3270@172.17.250.1;tag=as38531fbd
Call-ID: 1438751804
CSeq: 21 INVITE
Server: Asterisk PBX 13.5.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:3270@172.17.250.1:5060
Content-Length: 0

<------------>

<— SIP read from UDP:172.30.79.204:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.17.250.1:5060;branch=z9hG4bK5973b88e;received=172.17.250.1
From: sip:1000@172.17.250.1;tag=as293e7775
To: <sip:{outside number replaced}@172.30.79.204:5060>
Call-ID: 4958e88c38f418cc4e3619d64049f505@172.17.250.1:5060
CSeq: 102 INVITE
Server: Digium Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:{outside number replaced}@172.30.79.204:5060>
Content-Length: 0

<------------->
— (8 headers 0 lines) —

<— SIP read from UDP:172.17.250.201:5060 —>
OPTIONS sip:172.17.250.1 SIP/2.0
Via: SIP/2.0/UDP 172.17.250.201:5060;branch=z9hG4bK58a8ff65
Max-Forwards: 70
From: “asterisk” sip:asterisk@172.17.250.201;tag=as190ff350
To: sip:172.17.250.1
Contact: sip:asterisk@172.17.250.201:5060
Call-ID: 73e047607290ed461c14aa406b6029f0@172.17.250.201:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.5.0
Date: Mon, 04 Apr 2016 21:54:17 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------->
— (13 headers 0 lines) —
Sending to 172.17.250.201:5060 (no NAT)
Looking for s in default (domain 172.17.250.1)

<— Transmitting (no NAT) to 172.17.250.201:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.17.250.201:5060;branch=z9hG4bK58a8ff65;received=172.17.250.201
From: “asterisk” sip:asterisk@172.17.250.201;tag=as190ff350
To: sip:172.17.250.1;tag=as1bdc0291
Call-ID: 73e047607290ed461c14aa406b6029f0@172.17.250.201:5060
CSeq: 102 OPTIONS
Server: Asterisk PBX 13.5.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:172.17.250.1:5060
Accept: application/sdp
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘73e047607290ed461c14aa406b6029f0@172.17.250.201:5060’ in 32000 ms (Method: OPTIONS)

<— SIP read from UDP:172.30.79.204:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.17.250.1:5060;branch=z9hG4bK5973b88e;received=172.17.250.1
From: sip:1000@172.17.250.1;tag=as293e7775
To: <sip:{outside number replaced}@172.30.79.204:5060>;tag=as2e0958f6
Call-ID: 4958e88c38f418cc4e3619d64049f505@172.17.250.1:5060
CSeq: 102 INVITE
Server: Digium Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:{outside number replaced}@172.30.79.204:5060>
Content-Type: application/sdp
Content-Length: 330

v=0
o=root 1864686640 1864686640 IN IP4 172.30.79.204
s=Digium Gateway
c=IN IP4 172.30.79.204
t=0 0
m=audio 10410 RTP/AVP 0 8 9 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------->
— (13 headers 15 lines) —
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 9
Found RTP audio format 3
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 G722 for ID 9
Found audio description format GSM for ID 3
Found audio description format telephone-event for ID 101
Capabilities: us - (g722|ulaw|alaw|gsm), peer - audio=(ulaw|gsm|alaw|g722)/video=(nothing)/text=(nothing), combined - (g722|ulaw|alaw|gsm)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.30.79.204:10410
sip_route_dump: route/path hop: <sip:{outside number replaced}@172.30.79.204:5060>
set_destination: Parsing <sip:{outside number replaced}@172.30.79.204:5060> for address/port to send to
set_destination: set destination to 172.30.79.204:5060
Transmitting (no NAT) to 172.30.79.204:5060:
ACK sip:{outside number replaced}@172.30.79.204:5060 SIP/2.0
Via: SIP/2.0/UDP 172.17.250.1:5060;branch=z9hG4bK05b9d46b
Max-Forwards: 70
From: sip:1000@172.17.250.1;tag=as293e7775
To: <sip:{outside number replaced}@172.30.79.204:5060>;tag=as2e0958f6
Contact: sip:1000@172.17.250.1:5060
Call-ID: 4958e88c38f418cc4e3619d64049f505@172.17.250.1:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 13.5.0
Content-Length: 0

HERE’S THE ANSWER/CANCEL


-- SIP/gateway-wh01-00000020 connected line has changed. Saving it until answer for SIP/1000-0000001e
-- SIP/gateway-wh01-00000020 answered SIP/1000-0000001e

Scheduling destruction of SIP dialog ‘43d4e4544d45045c1891ef2822488cf9@172.17.250.1:5060’ in 32000 ms (Method: INVITE)
Reliably Transmitting (no NAT) to 172.17.106.1:5061:
CANCEL sip:11000@172.17.106.1:5061 SIP/2.0
Via: SIP/2.0/UDP 172.17.250.1:5060;branch=z9hG4bK58bbdaaa
Max-Forwards: 70
From: sip:1000@172.17.250.1;tag=as7c2f3c50
To: sip:11000@172.17.106.1:5061
Call-ID: 43d4e4544d45045c1891ef2822488cf9@172.17.250.1:5060
CSeq: 102 CANCEL
User-Agent: Asterisk PBX 13.5.0
Reason: SIP;cause=200;text="Call completed elsewhere"
Content-Length: 0


set_destination: Parsing sip:11000@172.17.106.1:5061 for address/port to send to
Scheduling destruction of SIP dialog ‘43d4e4544d45045c1891ef2822488cf9@172.17.250.1:5060’ in 32000 ms (Method: INVITE)
set_destination: set destination to 172.17.106.1:5061
Reliably Transmitting (no NAT) to 172.17.106.1:5061:
NOTIFY sip:11000@172.17.106.1:5061 SIP/2.0
Via: SIP/2.0/UDP 172.17.250.1:5060;branch=z9hG4bK00ae9001
Max-Forwards: 70
From: sip:11000@172.17.250.1;tag=as32f31f6a
To: “11000” sip:11000@172.17.250.1;tag=299345b82257f737
Contact: sip:11000@172.17.250.1:5060
Call-ID: e4ef9ed6-5a5fc4fd@172.17.106.1
CSeq: 253 NOTIFY
User-Agent: Asterisk PBX 13.5.0
Subscription-State: active
Event: dialog
Content-Type: application/dialog-info+xml
Content-Length: 209

<?xml version="1.0"?> terminated

== Extension Changed 11000[hints] new state Idle for Notify User 11000
Audio is at 13504
Adding codec g722 to SDP
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Reliably Transmitting (no NAT) to 192.168.247.247:9000 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.247.247:9000;branch=z9hG4bK1044022597;received=192.168.247.247;rport=9000
From: sip:1000@172.17.250.1;tag=262586417
To: sip:3270@172.17.250.1;tag=as38531fbd
Call-ID: 1438751804
CSeq: 21 INVITE
Server: Asterisk PBX 13.5.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:3270@172.17.250.1:5060
Content-Type: application/sdp
Content-Length: 351

v=0
o=root 1660615913 1660615913 IN IP4 172.17.250.1
s=Asterisk PBX 13.5.0
c=IN IP4 172.17.250.1
t=0 0
m=audio 13504 RTP/AVP 9 0 8 3 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
m=video 0 RTP/AVP 103 99 98

<------------>
– Channel SIP/gateway-wh01-00000020 joined ‘simple_bridge’ basic-bridge <293ad79c-255c-4019-8128-39d890379f38>
– Channel SIP/1000-0000001e joined ‘simple_bridge’ basic-bridge <293ad79c-255c-4019-8128-39d890379f38>

<— SIP read from UDP:172.17.106.1:5061 —>
SIP/2.0 487 Request Terminated
To: sip:11000@172.17.106.1:5061;tag=c500c680a5a765cfi0
From: sip:1000@172.17.250.1;tag=as7c2f3c50
Call-ID: 43d4e4544d45045c1891ef2822488cf9@172.17.250.1:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 172.17.250.1:5060;branch=z9hG4bK58bbdaaa
Server: Cisco/SPA508G-7.5.5
Content-Length: 0

<------------->
— (8 headers 0 lines) —
Transmitting (no NAT) to 172.17.106.1:5061:
ACK sip:11000@172.17.106.1:5061 SIP/2.0
Via: SIP/2.0/UDP 172.17.250.1:5060;branch=z9hG4bK58bbdaaa
Max-Forwards: 70
From: sip:1000@172.17.250.1;tag=as7c2f3c50
To: sip:11000@172.17.106.1:5061;tag=c500c680a5a765cfi0
Contact: sip:1000@172.17.250.1:5060
Call-ID: 43d4e4544d45045c1891ef2822488cf9@172.17.250.1:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 13.5.0
Content-Length: 0


Scheduling destruction of SIP dialog ‘43d4e4544d45045c1891ef2822488cf9@172.17.250.1:5060’ in 32000 ms (Method: INVITE)

<— SIP read from UDP:172.17.106.1:5061 —>
SIP/2.0 200 OK
To: sip:11000@172.17.106.1:5061;tag=c500c680a5a765cfi0
From: sip:1000@172.17.250.1;tag=as7c2f3c50
Call-ID: 43d4e4544d45045c1891ef2822488cf9@172.17.250.1:5060
CSeq: 102 CANCEL
Via: SIP/2.0/UDP 172.17.250.1:5060;branch=z9hG4bK58bbdaaa
Server: Cisco/SPA508G-7.5.5
Content-Length: 0

<------------->
— (8 headers 0 lines) —
> 0x7f1048158640 – Probation passed - setting RTP source address to 172.30.79.204:10410

<— SIP read from UDP:172.17.106.1:5061 —>
SIP/2.0 200 OK
To: “11000” sip:11000@172.17.250.1;tag=299345b82257f737
From: sip:11000@172.17.250.1;tag=as32f31f6a
Call-ID: e4ef9ed6-5a5fc4fd@172.17.106.1
CSeq: 253 NOTIFY
Via: SIP/2.0/UDP 172.17.250.1:5060;branch=z9hG4bK00ae9001
Server: Cisco/SPA508G-7.5.5
Content-Length: 0

<------------->
— (8 headers 0 lines) —

<— SIP read from UDP:192.168.247.247:9000 —>
ACK sip:3270@172.17.250.1:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.247.247:9000;rport;branch=z9hG4bK118045155
From: sip:1000@172.17.250.1;tag=262586417
To: sip:3270@172.17.250.1;tag=as38531fbd
Call-ID: 1438751804
CSeq: 21 ACK
Contact: sip:1000@192.168.247.247:9000
Max-Forwards: 70
User-Agent: Linphone/3.6.1 (eXosip2/4.0.0)
Content-Length: 0

<------------->
— (10 headers 0 lines) —
> 0x1d838c0 – Probation passed - setting RTP source address to 192.168.247.247:16384

<— SIP read from UDP:172.30.79.204:5060 —>
OPTIONS sip:172.17.250.1 SIP/2.0
Via: SIP/2.0/UDP 172.30.79.204:5060;branch=z9hG4bK4aab76e8
Max-Forwards: 70
From: “asterisk” sip:asterisk@172.30.79.204;tag=as7d190263
To: sip:172.17.250.1
Contact: sip:asterisk@172.30.79.204:5060
Call-ID: 3315fcb15fff310206e445f37ca2fec2@172.30.79.204:5060
CSeq: 102 OPTIONS
User-Agent: Digium Gateway
Date: Mon, 04 Apr 2016 21:54:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
— (13 headers 0 lines) —
Sending to 172.30.79.204:5060 (no NAT)
Looking for s in default (domain 172.17.250.1)

<— Transmitting (no NAT) to 172.30.79.204:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.79.204:5060;branch=z9hG4bK4aab76e8;received=172.30.79.204
From: “asterisk” sip:asterisk@172.30.79.204;tag=as7d190263
To: sip:172.17.250.1;tag=as528a2b72
Call-ID: 3315fcb15fff310206e445f37ca2fec2@172.30.79.204:5060
CSeq: 102 OPTIONS
Server: Asterisk PBX 13.5.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:172.17.250.1:5060
Accept: application/sdp
Content-Length: 0

<------------->
— (8 headers 0 lines) —
Reliably Transmitting (no NAT) to 172.30.79.204:5060:
OPTIONS sip:172.30.79.204 SIP/2.0
Via: SIP/2.0/UDP 172.17.250.1:5060;branch=z9hG4bK4d0c9b41
Max-Forwards: 70
From: “asterisk” sip:asterisk@172.17.250.1;tag=as2b087d14
To: sip:172.30.79.204
Contact: sip:asterisk@172.17.250.1:5060
Call-ID: 5df0b4ca255d7e52545cba1f5daa0573@172.17.250.1:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.5.0
Date: Mon, 04 Apr 2016 21:56:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<— SIP read from UDP:172.30.79.204:5060 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 172.17.250.1:5060;branch=z9hG4bK4d0c9b41;received=172.17.250.1;rport=5060
From: “asterisk” sip:asterisk@172.17.250.1;tag=as2b087d14
To: sip:172.30.79.204;tag=as628e3929
Call-ID: 5df0b4ca255d7e52545cba1f5daa0573@172.17.250.1:5060
CSeq: 102 OPTIONS
Server: Digium Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Really destroying SIP dialog ‘5df0b4ca255d7e52545cba1f5daa0573@172.17.250.1:5060’ Method: OPTIONS

<— SIP read from UDP:192.168.247.247:9000 —>
jaK
<------------->

<— SIP read from UDP:192.168.247.247:9000 —>
BYE sip:3270@172.17.250.1:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.247.247:9000;rport;branch=z9hG4bK1044696395
From: sip:1000@172.17.250.1;tag=262586417
To: sip:3270@172.17.250.1;tag=as38531fbd
Call-ID: 1438751804
CSeq: 22 BYE
Contact: sip:1000@192.168.247.247:9000
Proxy-Authorization: Digest username=“1000”, realm=“asterisk-nuc01”, nonce=“15ee7a44”, uri=“sip:3270@172.17.250.1:5060”, response=“ccc2896bbcb1427244889873845d0537”, algorithm=MD5
Max-Forwards: 70
User-Agent: Linphone/3.6.1 (eXosip2/4.0.0)
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Sending to 192.168.247.247:9000 (no NAT)
Scheduling destruction of SIP dialog ‘1438751804’ in 32000 ms (Method: BYE)

<— Transmitting (no NAT) to 192.168.247.247:9000 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.247.247:9000;branch=z9hG4bK1044696395;received=192.168.247.247;rport=9000
From: sip:1000@172.17.250.1;tag=262586417
To: sip:3270@172.17.250.1;tag=as38531fbd
Call-ID: 1438751804
CSeq: 22 BYE
Server: Asterisk PBX 13.5.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>
– Channel SIP/1000-0000001e left ‘simple_bridge’ basic-bridge <293ad79c-255c-4019-8128-39d890379f38>
– Channel SIP/gateway-wh01-00000020 left ‘simple_bridge’ basic-bridge <293ad79c-255c-4019-8128-39d890379f38>
Scheduling destruction of SIP dialog ‘4958e88c38f418cc4e3619d64049f505@172.17.250.1:5060’ in 6400 ms (Method: INVITE)
== Spawn extension (ringgroup, s, 4) exited non-zero on ‘SIP/1000-0000001e’
– Executing [h@ringgroup:1] Set(“SIP/1000-0000001e”, “DEVICE_STATE(Custom:3270@hints)=NOT_INUSE”) in new stack
set_destination: Parsing <sip:{outside number replaced}@172.30.79.204:5060> for address/port to send to
set_destination: set destination to 172.30.79.204:5060
Reliably Transmitting (no NAT) to 172.30.79.204:5060:
BYE sip:{outside number replaced}@172.30.79.204:5060 SIP/2.0
Via: SIP/2.0/UDP 172.17.250.1:5060;branch=z9hG4bK278f616c
Max-Forwards: 70
From: sip:1000@172.17.250.1;tag=as293e7775
To: <sip:{outside number replaced}@172.30.79.204:5060>;tag=as2e0958f6
Call-ID: 4958e88c38f418cc4e3619d64049f505@172.17.250.1:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX 13.5.0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


-- Executing [h@ringgroup:2] Hangup("SIP/1000-0000001e", "") in new stack

== Spawn extension (ringgroup, h, 2) exited non-zero on ‘SIP/1000-0000001e’

<— SIP read from UDP:172.30.79.204:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.17.250.1:5060;branch=z9hG4bK278f616c;received=172.17.250.1
From: sip:1000@172.17.250.1;tag=as293e7775
To: <sip:{outside number replaced}@172.30.79.204:5060>;tag=as2e0958f6
Call-ID: 4958e88c38f418cc4e3619d64049f505@172.17.250.1:5060
CSeq: 103 BYE
Server: Digium Gateway
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
— (8 headers 0 lines) —

<— SIP read from UDP:172.17.106.1:5061 —>
SUBSCRIBE sip:11000@172.17.250.1:5060 SIP/2.0
Via: SIP/2.0/UDP 172.17.106.1:5061;branch=z9hG4bK-dec17e36
From: “11000” sip:11000@172.17.250.1;tag=299345b82257f737
To: sip:11000@172.17.250.1;tag=as32f31f6a
Call-ID: e4ef9ed6-5a5fc4fd@172.17.106.1
CSeq: 1265 SUBSCRIBE
Max-Forwards: 70
Authorization: Digest username=“11000”,realm=“asterisk-nuc01”,nonce=“034a742d”,uri=“sip:11000@172.17.250.1:5060”,algorithm=MD5,response="7217ec9f91ce6ede1dea5ec92a5a63bd"
Contact: “11000” sip:11000@172.17.106.1:5061
Accept: application/dialog-info+xml
Expires: 60
Event: dialog
User-Agent: Cisco/SPA508G-7.5.5
Content-Length: 0

<------------->
— (14 headers 0 lines) —
Found peer ‘11000’ for ‘11000’ from 172.17.106.1:5061
[Apr 4 17:56:38] NOTICE[12913]: chan_sip.c:16696 check_auth: Correct auth, but based on stale nonce received from ‘“11000” sip:11000@172.17.250.1;tag=299345b82257f737’

<— Transmitting (no NAT) to 172.17.106.1:5061 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.17.106.1:5061;branch=z9hG4bK-dec17e36;received=172.17.106.1
From: “11000” sip:11000@172.17.250.1;tag=299345b82257f737
To: sip:11000@172.17.250.1;tag=as32f31f6a
Call-ID: e4ef9ed6-5a5fc4fd@172.17.106.1
CSeq: 1265 SUBSCRIBE
Server: Asterisk PBX 13.5.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk-nuc01”, nonce=“58433fc6”, stale=true
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘e4ef9ed6-5a5fc4fd@172.17.106.1’ in 32000 ms (Method: SUBSCRIBE)

<— SIP read from UDP:172.17.106.1:5061 —>
SUBSCRIBE sip:11000@172.17.250.1:5060 SIP/2.0
Via: SIP/2.0/UDP 172.17.106.1:5061;branch=z9hG4bK-6bf477f2
From: “11000” sip:11000@172.17.250.1;tag=299345b82257f737
To: sip:11000@172.17.250.1;tag=as32f31f6a
Call-ID: e4ef9ed6-5a5fc4fd@172.17.106.1
CSeq: 1266 SUBSCRIBE
Max-Forwards: 70
Authorization: Digest username=“11000”,realm=“asterisk-nuc01”,nonce=“58433fc6”,uri=“sip:11000@172.17.250.1:5060”,algorithm=MD5,response="b8af085a6412eb3a5c9690741e2e74f4"
Contact: “11000” sip:11000@172.17.106.1:5061
Accept: application/dialog-info+xml
Expires: 60
Event: dialog
User-Agent: Cisco/SPA508G-7.5.5
Content-Length: 0

<------------->
— (14 headers 0 lines) —
Found peer ‘11000’ for ‘11000’ from 172.17.106.1:5061
Looking for 11000 in hints (domain 172.17.250.1)
Scheduling destruction of SIP dialog ‘e4ef9ed6-5a5fc4fd@172.17.106.1’ in 70000 ms (Method: SUBSCRIBE)

<— Transmitting (no NAT) to 172.17.106.1:5061 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.17.106.1:5061;branch=z9hG4bK-6bf477f2;received=172.17.106.1
From: “11000” sip:11000@172.17.250.1;tag=299345b82257f737
To: sip:11000@172.17.250.1;tag=as32f31f6a
Call-ID: e4ef9ed6-5a5fc4fd@172.17.106.1
CSeq: 1266 SUBSCRIBE
Server: Asterisk PBX 13.5.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 60
Contact: sip:11000@172.17.250.1:5060;expires=60
Content-Length: 0

<------------>
set_destination: Parsing sip:11000@172.17.106.1:5061 for address/port to send to
set_destination: set destination to 172.17.106.1:5061
Reliably Transmitting (no NAT) to 172.17.106.1:5061:
NOTIFY sip:11000@172.17.106.1:5061 SIP/2.0
Via: SIP/2.0/UDP 172.17.250.1:5060;branch=z9hG4bK2ef22ddd
Max-Forwards: 70
From: sip:11000@172.17.250.1;tag=as32f31f6a
To: “11000” sip:11000@172.17.250.1;tag=299345b82257f737
Contact: sip:11000@172.17.250.1:5060
Call-ID: e4ef9ed6-5a5fc4fd@172.17.106.1
CSeq: 254 NOTIFY
User-Agent: Asterisk PBX 13.5.0
Subscription-State: active
Event: dialog
Content-Type: application/dialog-info+xml
Content-Length: 209

<?xml version="1.0"?> terminated

<— SIP read from UDP:172.17.106.1:5061 —>
SIP/2.0 200 OK
To: “11000” sip:11000@172.17.250.1;tag=299345b82257f737
From: sip:11000@172.17.250.1;tag=as32f31f6a
Call-ID: e4ef9ed6-5a5fc4fd@172.17.106.1
CSeq: 254 NOTIFY
Via: SIP/2.0/UDP 172.17.250.1:5060;branch=z9hG4bK2ef22ddd
Server: Cisco/SPA508G-7.5.5
Content-Length: 0

The gateway has answered. That’s the 200 OK.

I see that, but why is the gateway answering, and the server sending a CANCEL to the internal phone?

– SIP/gateway-wh01-00000020 connected line has changed. Saving it until answer for SIP/1000-0000001e
– SIP/gateway-wh01-00000020 answered SIP/1000-0000001e

Scheduling destruction of SIP dialog ‘43d4e4544d45045c1891ef2822488cf9@172.17.250.1:5060’ in 32000 ms (Method: INVITE)
Reliably Transmitting (no NAT) to 172.17.106.1:5061:
CANCEL sip:11000@172.17.106.1:5061 SIP/2.0
Via: SIP/2.0/UDP 172.17.250.1:5060;branch=z9hG4bK58bbdaaa
Max-Forwards: 70
From: ;tag=as7c2f3c50
To:
Call-ID: 43d4e4544d45045c1891ef2822488cf9@172.17.250.1:5060
CSeq: 102 CANCEL
User-Agent: Asterisk PBX 13.5.0
Reason: SIP;cause=200;text="Call completed elsewhere"
Content-Length: 0

The server is sending CANCEL because the gateway has answered.

I get the impression that the Gateway is a commercial product, so you should use the vendor’s commercial support channels for why it is answering. In particular, the dialplan may be copyright, even if it is an Asterisk based machine, so you shouldn’t post the substantial extracts that would be needed to understand what it was doing.

I would note that Playback can answer, as well as Answer and Dial, and it may be that the circuit switched network provider is doing early media in a bad way.