0

I'm trying to find out why this call failed, with:

<--- SIP read from UDP:63.247.69.226:5060 --->
SIP/2.0 100 Trying

and then

Everyone is busy/congested at this time (1:0/0/1)

full debug session:

mordor*CLI> 
mordor*CLI> 
mordor*CLI> sip set debug off
SIP Debugging Disabled
mordor*CLI> sip set debug on
SIP Debugging enabled
mordor*CLI> channel originate SIP/piter extension 18003569377@outbound
  == Using SIP RTP CoS mark 5
Audio is at 11012
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 (NAT) to 192.168.1.5:5063:
INVITE sip:piter@192.168.1.5:5063 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.8:5060;branch=z9hG4bK263def10;rport
Max-Forwards: 70
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as3cc65ffd
To: <sip:piter@192.168.1.5:5063>
Contact: <sip:anonymous@192.168.1.8:5060>
Call-ID: 4640418d0afa30671ce2edee71956234@192.168.1.8:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 13.1.0~dfsg-1.1ubuntu3
Date: Mon, 04 Jul 2016 09:47:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Type: application/sdp
Content-Length: 312

v=0
o=root 2081787154 2081787154 IN IP4 192.168.1.8
s=Asterisk PBX 13.1.0~dfsg-1.1ubuntu3
c=IN IP4 192.168.1.8
t=0 0
m=audio 11012 RTP/AVP 0 8 3 101
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 piter

<--- SIP read from UDP:192.168.1.5:5063 --->
SIP/2.0 100 Trying
To: <sip:piter@192.168.1.5:5063>
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as3cc65ffd
Call-ID: 4640418d0afa30671ce2edee71956234@192.168.1.8:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.1.8:5060;branch=z9hG4bK263def10
Server: Linksys/SPA942-6.1.5(a)
Content-Length: 0

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

<--- SIP read from UDP:192.168.1.5:5063 --->
SIP/2.0 180 Ringing
To: <sip:piter@192.168.1.5:5063>;tag=1702b369152c1657i3
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as3cc65ffd
Call-ID: 4640418d0afa30671ce2edee71956234@192.168.1.8:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.1.8:5060;branch=z9hG4bK263def10
Contact: "piter" <sip:piter@192.168.1.5:5063>
Server: Linksys/SPA942-6.1.5(a)
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
sip_route_dump: route/path hop: <sip:piter@192.168.1.5:5063>
    -- SIP/piter-0000000e is ringing

<--- SIP read from UDP:192.168.1.5:5063 --->
SIP/2.0 200 OK
To: <sip:piter@192.168.1.5:5063>;tag=1702b369152c1657i3
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as3cc65ffd
Call-ID: 4640418d0afa30671ce2edee71956234@192.168.1.8:5060
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.1.8:5060;branch=z9hG4bK263def10
Contact: "piter" <sip:piter@192.168.1.5:5063>
Server: Linksys/SPA942-6.1.5(a)
Content-Length: 206
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces
Content-Type: application/sdp

v=0
o=- 2140205 2140205 IN IP4 192.168.1.5
s=-
c=IN IP4 192.168.1.5
t=0 0
m=audio 16416 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
<------------->
--- (12 headers 11 lines) ---
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|alaw|gsm), 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 192.168.1.5:16416
sip_route_dump: route/path hop: <sip:piter@192.168.1.5:5063>
Transmitting (NAT) to 192.168.1.5:5063:
ACK sip:piter@192.168.1.5:5063 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.8:5060;branch=z9hG4bK6a3a6746;rport
Max-Forwards: 70
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as3cc65ffd
To: <sip:piter@192.168.1.5:5063>;tag=1702b369152c1657i3
Contact: <sip:anonymous@192.168.1.8:5060>
Call-ID: 4640418d0afa30671ce2edee71956234@192.168.1.8:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 13.1.0~dfsg-1.1ubuntu3
Content-Length: 0


---
    -- SIP/piter-0000000e answered
    -- Executing [18003569377@outbound:1] NoOp("SIP/piter-0000000e", "") in new stack
    -- Executing [18003569377@outbound:2] Log("SIP/piter-0000000e", "NOTICE, Dialing out from "piter" <> to 8003569377 through SIP/12345678GW1") in new stack
[Jul  4 02:47:12] NOTICE[13652][C-00000007]: Ext. 18003569377:2 @ outbound:  Dialing out from "piter" <> to 8003569377 through SIP/12345678GW1
    -- Executing [18003569377@outbound:3] Dial("SIP/piter-0000000e", "SIP/12345678GW1/8003569377,60") in new stack
  == Using SIP RTP CoS mark 5
Audio is at 16374
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 (NAT) to 63.247.69.226:5060:
INVITE sip:8003569377@gw1.siptrunk.com:5060 SIP/2.0
Via: SIP/2.0/UDP <my.external.ip.address>:5060;branch=z9hG4bK40fff6ea;rport
Max-Forwards: 70
From: "piter" <sip:asterisk@gw1.siptrunk.com>;tag=as79460c11
To: <sip:8003569377@gw1.siptrunk.com:5060>
Contact: <sip:asterisk@<my.external.ip.address>:5060>
Call-ID: 134102a92a4f311667b70b5473076c16@gw1.siptrunk.com
CSeq: 102 INVITE
User-Agent: Asterisk PBX 13.1.0~dfsg-1.1ubuntu3
Date: Mon, 04 Jul 2016 09:47:12 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Type: application/sdp
Content-Length: 318

v=0
o=root 1092551220 1092551220 IN IP4 <my.external.ip.address>
s=Asterisk PBX 13.1.0~dfsg-1.1ubuntu3
c=IN IP4 <my.external.ip.address>
t=0 0
m=audio 16374 RTP/AVP 0 8 3 101
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/12345678GW1/8003569377
       > 0x7fd8bc003860 -- Probation passed - setting RTP source address to 192.168.1.5:16416

<--- SIP read from UDP:63.247.69.226:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP <my.external.ip.address>:5060;branch=z9hG4bK40fff6ea;rport=49192;received=<my.external.ip.address>
From: "piter" <sip:asterisk@gw1.siptrunk.com>;tag=as79460c11
To: <sip:8003569377@gw1.siptrunk.com:5060>
Call-ID: 134102a92a4f311667b70b5473076c16@gw1.siptrunk.com
CSeq: 102 INVITE
Content-Length: 0

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

<--- SIP read from UDP:63.247.69.226:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP <my.external.ip.address>:5060;branch=z9hG4bK40fff6ea;rport=49192;received=<my.external.ip.address>
From: "piter" <sip:asterisk@gw1.siptrunk.com>;tag=as79460c11
To: <sip:8003569377@gw1.siptrunk.com:5060>;tag=14575c43173f48ab35024af4364ef5fa.cbfb
Call-ID: 134102a92a4f311667b70b5473076c16@gw1.siptrunk.com
CSeq: 102 INVITE
Proxy-Authenticate: Digest realm="gw1.siptrunk.com", nonce="V3oy+Fd6MKB3huy6iBHjbvTIK9i+xRjZYUpDfJ2/BcrVMlyQNNwwhRmWoIjB", qop="auth"
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Transmitting (NAT) to 63.247.69.226:5060:
ACK sip:8003569377@gw1.siptrunk.com:5060 SIP/2.0
Via: SIP/2.0/UDP <my.external.ip.address>:5060;branch=z9hG4bK40fff6ea;rport
Max-Forwards: 70
From: "piter" <sip:asterisk@gw1.siptrunk.com>;tag=as79460c11
To: <sip:8003569377@gw1.siptrunk.com:5060>;tag=14575c43173f48ab35024af4364ef5fa.cbfb
Contact: <sip:asterisk@<my.external.ip.address>:5060>
Call-ID: 134102a92a4f311667b70b5473076c16@gw1.siptrunk.com
CSeq: 102 ACK
User-Agent: Asterisk PBX 13.1.0~dfsg-1.1ubuntu3
Content-Length: 0


---
Audio is at 16374
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 (NAT) to 63.247.69.226:5060:
INVITE sip:8003569377@gw1.siptrunk.com:5060 SIP/2.0
Via: SIP/2.0/UDP <my.external.ip.address>:5060;branch=z9hG4bK3e38a92a;rport
Max-Forwards: 70
From: "piter" <sip:asterisk@gw1.siptrunk.com>;tag=as79460c11
To: <sip:8003569377@gw1.siptrunk.com:5060>
Contact: <sip:asterisk@<my.external.ip.address>:5060>
Call-ID: 134102a92a4f311667b70b5473076c16@gw1.siptrunk.com
CSeq: 103 INVITE
User-Agent: Asterisk PBX 13.1.0~dfsg-1.1ubuntu3
Proxy-Authorization: Digest username="12345678", realm="gw1.siptrunk.com", algorithm=MD5, uri="sip:8003569377@gw1.siptrunk.com:5060", nonce="V3oy+Fd6MKB3huy6iBHjbvTIK9i+xRjZYUpDfJ2/BcrVMlyQNNwwhRmWoIjB", response="294576e37f5430f6d5191445bd4429d1", qop=auth, cnonce="32702902", nc=00000001
Date: Mon, 04 Jul 2016 09:47:12 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Type: application/sdp
Content-Length: 318

v=0
o=root 1092551220 1092551221 IN IP4 <my.external.ip.address>
s=Asterisk PBX 13.1.0~dfsg-1.1ubuntu3
c=IN IP4 <my.external.ip.address>
t=0 0
m=audio 16374 RTP/AVP 0 8 3 101
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

---

<--- SIP read from UDP:63.247.69.226:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP <my.external.ip.address>:5060;branch=z9hG4bK3e38a92a;rport=49192;received=<my.external.ip.address>
From: "piter" <sip:asterisk@gw1.siptrunk.com>;tag=as79460c11
To: <sip:8003569377@gw1.siptrunk.com:5060>
Call-ID: 134102a92a4f311667b70b5473076c16@gw1.siptrunk.com
CSeq: 103 INVITE
Content-Length: 0

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

<--- SIP read from UDP:63.247.69.226:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP <my.external.ip.address>:5060;branch=z9hG4bK3e38a92a;rport=49192;received=<my.external.ip.address>
From: "piter" <sip:asterisk@gw1.siptrunk.com>;tag=as79460c11
To: <sip:8003569377@gw1.siptrunk.com:5060>;tag=14575c43173f48ab35024af4364ef5fa.bf26
Call-ID: 134102a92a4f311667b70b5473076c16@gw1.siptrunk.com
CSeq: 103 INVITE
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Transmitting (NAT) to 63.247.69.226:5060:
ACK sip:8003569377@gw1.siptrunk.com:5060 SIP/2.0
Via: SIP/2.0/UDP <my.external.ip.address>:5060;branch=z9hG4bK3e38a92a;rport
Max-Forwards: 70
From: "piter" <sip:asterisk@gw1.siptrunk.com>;tag=as79460c11
To: <sip:8003569377@gw1.siptrunk.com:5060>;tag=14575c43173f48ab35024af4364ef5fa.bf26
Contact: <sip:asterisk@<my.external.ip.address>:5060>
Call-ID: 134102a92a4f311667b70b5473076c16@gw1.siptrunk.com
CSeq: 103 ACK
User-Agent: Asterisk PBX 13.1.0~dfsg-1.1ubuntu3
Content-Length: 0


---
Scheduling destruction of SIP dialog '134102a92a4f311667b70b5473076c16@gw1.siptrunk.com' in 6400 ms (Method: INVITE)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [18003569377@outbound:4] PlayTones("SIP/piter-0000000e", "congestion") in new stack
    -- Executing [18003569377@outbound:5] Hangup("SIP/piter-0000000e", "") in new stack
  == Spawn extension (outbound, 18003569377, 5) exited non-zero on 'SIP/piter-0000000e'
Scheduling destruction of SIP dialog '4640418d0afa30671ce2edee71956234@192.168.1.8:5060' in 6400 ms (Method: INVITE)
Reliably Transmitting (NAT) to 192.168.1.5:5063:
BYE sip:piter@192.168.1.5:5063 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.8:5060;branch=z9hG4bK487f69fb;rport
Max-Forwards: 70
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as3cc65ffd
To: <sip:piter@192.168.1.5:5063>;tag=1702b369152c1657i3
Call-ID: 4640418d0afa30671ce2edee71956234@192.168.1.8:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX 13.1.0~dfsg-1.1ubuntu3
X-Asterisk-HangupCause: Unallocated (unassigned) number
X-Asterisk-HangupCauseCode: 1
Content-Length: 0


---

<--- SIP read from UDP:192.168.1.5:5063 --->
SIP/2.0 200 OK
To: <sip:piter@192.168.1.5:5063>;tag=1702b369152c1657i3
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as3cc65ffd
Call-ID: 4640418d0afa30671ce2edee71956234@192.168.1.8:5060
CSeq: 103 BYE
Via: SIP/2.0/UDP 192.168.1.8:5060;branch=z9hG4bK487f69fb
Server: Linksys/SPA942-6.1.5(a)
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '4640418d0afa30671ce2edee71956234@192.168.1.8:5060' Method: INVITE

<--- SIP read from UDP:192.168.1.5:5062 --->
NOTIFY sip:192.168.1.8 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.5:5062;branch=z9hG4bK-aeeaf71e
From: "thufir" <sip:thufir@192.168.1.8>;tag=2dc2ec8b9eeabd55o2
To: <sip:192.168.1.8>
Call-ID: 8361cdb3-79410e4d@192.168.1.5
CSeq: 1427 NOTIFY
Max-Forwards: 70
Contact: "thufir" <sip:thufir@192.168.1.5:5062>
Event: keep-alive
User-Agent: Linksys/SPA942-6.1.5(a)
Content-Length: 0

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

<--- Transmitting (NAT) to 192.168.1.5:5062 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.5:5062;branch=z9hG4bK-aeeaf71e;received=192.168.1.5;rport=5062
From: "thufir" <sip:thufir@192.168.1.8>;tag=2dc2ec8b9eeabd55o2
To: <sip:192.168.1.8>;tag=as210aafe6
Call-ID: 8361cdb3-79410e4d@192.168.1.5
CSeq: 1427 NOTIFY
Server: Asterisk PBX 13.1.0~dfsg-1.1ubuntu3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '8361cdb3-79410e4d@192.168.1.5' in 32000 ms (Method: NOTIFY)

<--- SIP read from UDP:192.168.1.5:5063 --->
NOTIFY sip:192.168.1.8 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.5:5063;branch=z9hG4bK-1f7b1f9d
From: "piter" <sip:piter@192.168.1.8>;tag=426902db72b1b45o3
To: <sip:192.168.1.8>
Call-ID: 8024367b-67307c25@192.168.1.5
CSeq: 1427 NOTIFY
Max-Forwards: 70
Contact: "piter" <sip:piter@192.168.1.5:5063>
Event: keep-alive
User-Agent: Linksys/SPA942-6.1.5(a)
Content-Length: 0

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

<--- Transmitting (NAT) to 192.168.1.5:5063 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.5:5063;branch=z9hG4bK-1f7b1f9d;received=192.168.1.5;rport=5063
From: "piter" <sip:piter@192.168.1.8>;tag=426902db72b1b45o3
To: <sip:192.168.1.8>;tag=as465f40b3
Call-ID: 8024367b-67307c25@192.168.1.5
CSeq: 1427 NOTIFY
Server: Asterisk PBX 13.1.0~dfsg-1.1ubuntu3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '8024367b-67307c25@192.168.1.5' in 32000 ms (Method: NOTIFY)
mordor*CLI> sip set debug off
SIP Debugging Disabled
mordor*CLI> 

and found an article about the exact router I'm using, a NetGear WNR1000v2 - N150 Wireless Router. However, I don't think it's the router at all because

  1. I can dial 'hello@onsip.com` from an Android tablet on the network
  2. I can dial out with the same syntax for telnyx

I have to swap out and reload the sip.conf and extensions.conf, but if outbound dialing through Asterisk works for one SIP trunking provider then it stands to reason that it should work for another and that the router config isn't a concern. Or, does it?

1

This is not a NAT or Firewall problem. It seems that the re-INVITE is not handled correctly. Try to disable re-INVITE and that should fix this problem.

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