Errore when using include

Asterisk Asterisk Support
1

Dear all,

i’m experiencing a really strange thing, i have moved some “code” from [employees] section to [branches] section.
Then i have included [branches] into [employees].
Now, the execution is very very slow and oftern appears:

chan_sip.c:4066 retrans_pkt: Retransmission timeout reached on transmission 0_1750665163@172.16.18.50 for seqno 2 (Critical Response) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions

I tried to move back the code and everything is ok.

Why this happens?

Thanks and regards,
Davide

2

We would need to see the configuration and full console output to understand what is going on. The limited information you’ve provided is not enough.

3

I have written this code in [employees]:

;Call to test
exten => _661XX,1,NoOp(Call to test)
exten => _661XX,n,Dial(SIP/PSTN/${EXTEN})
exten => _661XX,n,Hangup()

Then i created

[branches]
;Call to test
exten => _661XX,1,NoOp(Call to test)
exten => _661XX,n,Dial(SIP/PSTN/${EXTEN})
exten => _661XX,n,Hangup()

and in employees i wrote include => branches

4

== Using SIP VIDEO CoS mark 6
== Using SIP RTP CoS mark 5
– Executing [66101@employees:1] NoOp(“SIP/0339-0000001e”, “Call to test”) in new stack
[Nov 8 15:35:32] WARNING[2390]: chan_sip.c:4066 retrans_pkt: Retransmission timeout reached on transmission 0_3924166936@172.20.22.32 for seqno 2 (Critical Response) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6400ms with no response

5

That dialplan change itself would not cause such a problem. What output do you get if you do “sip set debug on” as well?

6

SIP Debugging Enabled for IP: 192.168.0.1

<— SIP read from UDP:192.168.0.1:5060 —>
INVITE sip:66101@192.168.0.254:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK1614261585
From: “0339 - Test” sip:0339@192.168.0.254:5060;tag=4079818826
To: sip:66101@192.168.0.254:5060
Call-ID: 0_4179623970@192.168.0.1
CSeq: 1 INVITE
Contact: sip:0339@192.168.0.1:5060
Content-Type: application/sdp
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: Yealink SIP-T27P 45.81.0.15
Allow-Events: talk,hold,conference,refer,check-sync
Supported: replaces
Content-Length: 306

v=0
o=- 20021 20021 IN IP4 192.168.0.1
s=SDP data
c=IN IP4 192.168.0.1
t=0 0
m=audio 12056 RTP/AVP 9 0 8 18 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=ptime:20
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
— (14 headers 15 lines) —
Sending to 192.168.0.1:5060 (no NAT)
Sending to 192.168.0.1:5060 (no NAT)
Using INVITE request as basis request - 0_4179623970@192.168.0.1
Found peer ‘0339’ for ‘0339’ from 192.168.0.1:5060

<— Reliably Transmitting (no NAT) to 192.168.0.1:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK1614261585;received=192.168.0.1
From: “0339 - Test” sip:0339@192.168.0.254:5060;tag=4079818826
To: sip:66101@192.168.0.254:5060;tag=as34ded3dd
Call-ID: 0_4179623970@192.168.0.1
CSeq: 1 INVITE
Server: Asterisk PBX 14.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“debortoli.lan”, nonce="138d28b7"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘0_4179623970@192.168.0.1’ in 6400 ms (Method: INVITE)

<— SIP read from UDP:192.168.0.1:5060 —>
ACK sip:66101@192.168.0.254:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK1614261585
From: “0339 - Test” sip:0339@192.168.0.254:5060;tag=4079818826
To: sip:66101@192.168.0.254:5060;tag=as34ded3dd
Call-ID: 0_4179623970@192.168.0.1
CSeq: 1 ACK
Content-Length: 0

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

<— SIP read from UDP:192.168.0.1:5060 —>
INVITE sip:66101@192.168.0.254:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK1813797491
From: “0339 - Test” sip:0339@192.168.0.254:5060;tag=4079818826
To: sip:66101@192.168.0.254:5060
Call-ID: 0_4179623970@192.168.0.1
CSeq: 2 INVITE
Contact: sip:0339@192.168.0.1:5060
Authorization: Digest username=“0339”, realm=“debortoli.lan”, nonce=“138d28b7”, uri=“sip:66101@192.168.0.254:5060”, response=“adf4877fa94acd0d8343d52ad3749573”, algorithm=MD5
Content-Type: application/sdp
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: Yealink SIP-T27P 45.81.0.15
Allow-Events: talk,hold,conference,refer,check-sync
Supported: replaces
Content-Length: 306

v=0
o=- 20021 20021 IN IP4 192.168.0.1
s=SDP data
c=IN IP4 192.168.0.1
t=0 0
m=audio 12056 RTP/AVP 9 0 8 18 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=ptime:20
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
— (15 headers 15 lines) —
Sending to 192.168.0.1:5060 (no NAT)
Using INVITE request as basis request - 0_4179623970@192.168.0.1
Found peer ‘0339’ for ‘0339’ from 192.168.0.1:5060
== Using SIP VIDEO CoS mark 6
== Using SIP RTP CoS mark 5
Found RTP audio format 9
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Found audio description format G722 for ID 9
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 - (gsm|alaw|ulaw|h264|h263p|h263|g723|g726|g726aal2|adpcm|slin|slin|slin|slin|slin|slin|slin|slin|slin|lpc10|g729|speex|speex|speex|ilbc|g722|siren7|siren14|testlaw|g719|opus|jpeg|png|h261|mpeg4|vp8|red|t140|silk|silk|silk|silk), peer - audio=(ulaw|alaw|g722|g729)/video=(nothing)/text=(nothing), combined - (alaw|ulaw|g729|g722)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.0.1:12056
Peer doesn’t provide video
Peer doesn’t provide T.140
Looking for 66101 in employees-mb (domain 192.168.0.254)
sip_route_dump: route/path hop: sip:0339@192.168.0.1:5060

<— Transmitting (no NAT) to 192.168.0.1:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK1813797491;received=192.168.0.1
From: “0339 - Test” sip:0339@192.168.0.254:5060;tag=4079818826
To: sip:66101@192.168.0.254:5060
Call-ID: 0_4179623970@192.168.0.1
CSeq: 2 INVITE
Server: Asterisk PBX 14.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:66101@192.168.0.254:5060
Content-Length: 0

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

<— SIP read from UDP:192.168.0.1:5060 —>
INVITE sip:66101@192.168.0.254:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK1813797491
From: “0339 - Test” sip:0339@192.168.0.254:5060;tag=4079818826
To: sip:66101@192.168.0.254:5060
Call-ID: 0_4179623970@192.168.0.1
CSeq: 2 INVITE
Contact: sip:0339@192.168.0.1:5060
Authorization: Digest username=“0339”, realm=“debortoli.lan”, nonce=“138d28b7”, uri=“sip:66101@192.168.0.254:5060”, response=“adf4877fa94acd0d8343d52ad3749573”, algorithm=MD5
Content-Type: application/sdp
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: Yealink SIP-T27P 45.81.0.15
Allow-Events: talk,hold,conference,refer,check-sync
Supported: replaces
Content-Length: 306

v=0
o=- 20021 20021 IN IP4 192.168.0.1
s=SDP data
c=IN IP4 192.168.0.1
t=0 0
m=audio 12056 RTP/AVP 9 0 8 18 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=ptime:20
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
— (15 headers 15 lines) —
Ignoring this INVITE request

<— Transmitting (no NAT) to 192.168.0.1:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK1813797491;received=192.168.0.1
From: “0339 - Test” sip:0339@192.168.0.254:5060;tag=4079818826
To: sip:66101@192.168.0.254:5060
Call-ID: 0_4179623970@192.168.0.1
CSeq: 2 INVITE
Server: Asterisk PBX 14.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:66101@192.168.0.254:5060
Content-Length: 0

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

<— SIP read from UDP:192.168.0.1:5060 —>
INVITE sip:66101@192.168.0.254:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK1813797491
From: “0339 - Test” sip:0339@192.168.0.254:5060;tag=4079818826
To: sip:66101@192.168.0.254:5060
Call-ID: 0_4179623970@192.168.0.1
CSeq: 2 INVITE
Contact: sip:0339@192.168.0.1:5060
Authorization: Digest username=“0339”, realm=“debortoli.lan”, nonce=“138d28b7”, uri=“sip:66101@192.168.0.254:5060”, response=“adf4877fa94acd0d8343d52ad3749573”, algorithm=MD5
Content-Type: application/sdp
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: Yealink SIP-T27P 45.81.0.15
Allow-Events: talk,hold,conference,refer,check-sync
Supported: replaces
Content-Length: 306

v=0
o=- 20021 20021 IN IP4 192.168.0.1
s=SDP data
c=IN IP4 192.168.0.1
t=0 0
m=audio 12056 RTP/AVP 9 0 8 18 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=ptime:20
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
— (15 headers 15 lines) —
Ignoring this INVITE request

<— Transmitting (no NAT) to 192.168.0.1:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK1813797491;received=192.168.0.1
From: “0339 - Test” sip:0339@192.168.0.254:5060;tag=4079818826
To: sip:66101@192.168.0.254:5060
Call-ID: 0_4179623970@192.168.0.1
CSeq: 2 INVITE
Server: Asterisk PBX 14.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:66101@192.168.0.254:5060
Content-Length: 0

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

<— SIP read from UDP:192.168.0.1:5060 —>
SUBSCRIBE sip:asterisk@192.168.0.254:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK2031535651
From: “0339 - Test” sip:0339@192.168.0.254:5060;tag=2537575104
To: sip:129@192.168.0.254:5060;tag=as1886e6cf
Call-ID: 0_3424179002@192.168.0.1
CSeq: 3 SUBSCRIBE
Contact: sip:0339@192.168.0.1:5060
Authorization: Digest username=“0339”, realm=“debortoli.lan”, nonce=“33eb327a”, uri=“sip:asterisk@192.168.0.254:5060”, response=“67964b2a7c656c65ecccef3899ab85d3”, algorithm=MD5
Accept: application/simple-message-summary
Max-Forwards: 70
User-Agent: Yealink SIP-T27P 45.81.0.15
Expires: 0
Event: message-summary
Content-Length: 0

<------------->
— (14 headers 0 lines) —
Found peer ‘0339’ for ‘0339’ from 192.168.0.1:5060
[Nov 8 15:38:53] NOTICE[2390]: chan_sip.c:17216 check_auth: Correct auth, but based on stale nonce received from ‘“0339 - Test” sip:0339@192.168.0.254:5060;tag=2537575104’

<— Transmitting (no NAT) to 192.168.0.1:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK2031535651;received=192.168.0.1
From: “0339 - Test” sip:0339@192.168.0.254:5060;tag=2537575104
To: sip:129@192.168.0.254:5060;tag=as1886e6cf
Call-ID: 0_3424179002@192.168.0.1
CSeq: 3 SUBSCRIBE
Server: Asterisk PBX 14.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“debortoli.lan”, nonce=“32585781”, stale=true
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘0_3424179002@192.168.0.1’ in 6400 ms (Method: SUBSCRIBE)

<— SIP read from UDP:192.168.0.1:5060 —>
CANCEL sip:66101@192.168.0.254:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK1813797491
From: “0339 - Test” sip:0339@192.168.0.254:5060;tag=4079818826
To: sip:66101@192.168.0.254:5060
Call-ID: 0_4179623970@192.168.0.1
CSeq: 2 CANCEL
Max-Forwards: 70
User-Agent: Yealink SIP-T27P 45.81.0.15
Content-Length: 0

<------------->
— (9 headers 0 lines) —
Sending to 192.168.0.1:5060 (no NAT)

<— Reliably Transmitting (no NAT) to 192.168.0.1:5060 —>
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK1813797491;received=192.168.0.1
From: “0339 - Test” sip:0339@192.168.0.254:5060;tag=4079818826
To: sip:66101@192.168.0.254:5060;tag=as404235a5
Call-ID: 0_4179623970@192.168.0.1
CSeq: 2 INVITE
Server: Asterisk PBX 14.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

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

<— Transmitting (no NAT) to 192.168.0.1:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK1813797491;received=192.168.0.1
From: “0339 - Test” sip:0339@192.168.0.254:5060;tag=4079818826
To: sip:66101@192.168.0.254:5060;tag=as404235a5
Call-ID: 0_4179623970@192.168.0.1
CSeq: 2 CANCEL
Server: Asterisk PBX 14.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

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

<— SIP read from UDP:192.168.0.1:5060 —>
SUBSCRIBE sip:asterisk@192.168.0.254:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK2031535651
From: “0339 - Test” sip:0339@192.168.0.254:5060;tag=2537575104
To: sip:129@192.168.0.254:5060;tag=as1886e6cf
Call-ID: 0_3424179002@192.168.0.1
CSeq: 3 SUBSCRIBE
Contact: sip:0339@192.168.0.1:5060
Authorization: Digest username=“0339”, realm=“debortoli.lan”, nonce=“33eb327a”, uri=“sip:asterisk@192.168.0.254:5060”, response=“67964b2a7c656c65ecccef3899ab85d3”, algorithm=MD5
Accept: application/simple-message-summary
Max-Forwards: 70
User-Agent: Yealink SIP-T27P 45.81.0.15
Expires: 0
Event: message-summary
Content-Length: 0

<------------->
— (14 headers 0 lines) —
Ignoring this SUBSCRIBE request

<— Transmitting (no NAT) to 192.168.0.1:5060 —>
SIP/2.0 404 Not found
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK2031535651;received=192.168.0.1
From: “0339 - Test” sip:0339@192.168.0.254:5060;tag=2537575104
To: sip:129@192.168.0.254:5060;tag=as1886e6cf
Call-ID: 0_3424179002@192.168.0.1
CSeq: 3 SUBSCRIBE
Server: Asterisk PBX 14.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

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

<— SIP read from UDP:192.168.0.1:5060 —>
CANCEL sip:66101@192.168.0.254:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK1813797491
From: “0339 - Test” sip:0339@192.168.0.254:5060;tag=4079818826
To: sip:66101@192.168.0.254:5060
Call-ID: 0_4179623970@192.168.0.1
CSeq: 2 CANCEL
Max-Forwards: 70
User-Agent: Yealink SIP-T27P 45.81.0.15
Content-Length: 0

<------------->
— (9 headers 0 lines) —
Sending to 192.168.0.1:5060 (no NAT)

<— Reliably Transmitting (no NAT) to 192.168.0.1:5060 —>
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK1813797491;received=192.168.0.1
From: “0339 - Test” sip:0339@192.168.0.254:5060;tag=4079818826
To: sip:66101@192.168.0.254:5060;tag=as404235a5
Call-ID: 0_4179623970@192.168.0.1
CSeq: 2 INVITE
Server: Asterisk PBX 14.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

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

<— Transmitting (no NAT) to 192.168.0.1:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK1813797491;received=192.168.0.1
From: “0339 - Test” sip:0339@192.168.0.254:5060;tag=4079818826
To: sip:66101@192.168.0.254:5060;tag=as404235a5
Call-ID: 0_4179623970@192.168.0.1
CSeq: 2 CANCEL
Server: Asterisk PBX 14.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

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

-- Executing [66101@employees-mb:1] NoOp("SIP/0339-0000001f", "Call to test") in new stack

<— SIP read from UDP:192.168.0.1:5060 —>
SUBSCRIBE sip:asterisk@192.168.0.254:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK2409661889
From: “0339 - Test” sip:0339@192.168.0.254:5060;tag=2537575104
To: sip:129@192.168.0.254:5060;tag=as1886e6cf
Call-ID: 0_3424179002@192.168.0.1
CSeq: 4 SUBSCRIBE
Contact: sip:0339@192.168.0.1:5060
Authorization: Digest username=“0339”, realm=“debortoli.lan”, nonce=“32585781”, uri=“sip:asterisk@192.168.0.254:5060”, response=“11dc84181d43d0dfd8a510a370af6841”, algorithm=MD5
Accept: application/simple-message-summary
Max-Forwards: 70
User-Agent: Yealink SIP-T27P 45.81.0.15
Expires: 0
Event: message-summary
Content-Length: 0

<------------->
— (14 headers 0 lines) —
Sending to 192.168.0.1:5060 (no NAT)

<— Transmitting (no NAT) to 192.168.0.1:5060 —>
SIP/2.0 481 Call/Transaction Does Not Exist
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK2409661889;received=192.168.0.1
From: “0339 - Test” sip:0339@192.168.0.254:5060;tag=2537575104
To: sip:129@192.168.0.254:5060;tag=as1886e6cf
Call-ID: 0_3424179002@192.168.0.1
CSeq: 4 SUBSCRIBE
Server: Asterisk PBX 14.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘0_3424179002@192.168.0.1’ in 32000 ms (Method: SUBSCRIBE)
Retransmitting #1 (no NAT) to 192.168.0.1:5060:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK1813797491;received=192.168.0.1
From: “0339 - Test” sip:0339@192.168.0.254:5060;tag=4079818826
To: sip:66101@192.168.0.254:5060;tag=as404235a5
Call-ID: 0_4179623970@192.168.0.1
CSeq: 2 INVITE
Server: Asterisk PBX 14.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

Retransmitting #2 (no NAT) to 192.168.0.1:5060:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK1813797491;received=192.168.0.1
From: “0339 - Test” sip:0339@192.168.0.254:5060;tag=4079818826
To: sip:66101@192.168.0.254:5060;tag=as404235a5
Call-ID: 0_4179623970@192.168.0.1
CSeq: 2 INVITE
Server: Asterisk PBX 14.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

Retransmitting #3 (no NAT) to 192.168.0.1:5060:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK1813797491;received=192.168.0.1
From: “0339 - Test” sip:0339@192.168.0.254:5060;tag=4079818826
To: sip:66101@192.168.0.254:5060;tag=as404235a5
Call-ID: 0_4179623970@192.168.0.1
CSeq: 2 INVITE
Server: Asterisk PBX 14.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

Retransmitting #4 (no NAT) to 192.168.0.1:5060:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK1813797491;received=192.168.0.1
From: “0339 - Test” sip:0339@192.168.0.254:5060;tag=4079818826
To: sip:66101@192.168.0.254:5060;tag=as404235a5
Call-ID: 0_4179623970@192.168.0.1
CSeq: 2 INVITE
Server: Asterisk PBX 14.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

Retransmitting #5 (no NAT) to 192.168.0.1:5060:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK1813797491;received=192.168.0.1
From: “0339 - Test” sip:0339@192.168.0.254:5060;tag=4079818826
To: sip:66101@192.168.0.254:5060;tag=as404235a5
Call-ID: 0_4179623970@192.168.0.1
CSeq: 2 INVITE
Server: Asterisk PBX 14.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

Retransmitting #6 (no NAT) to 192.168.0.1:5060:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK1813797491;received=192.168.0.1
From: “0339 - Test” sip:0339@192.168.0.254:5060;tag=4079818826
To: sip:66101@192.168.0.254:5060;tag=as404235a5
Call-ID: 0_4179623970@192.168.0.1
CSeq: 2 INVITE
Server: Asterisk PBX 14.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

[Nov 8 15:39:00] WARNING[2390]: chan_sip.c:4066 retrans_pkt: Retransmission timeout reached on transmission 0_4179623970@192.168.0.1 for seqno 2 (Critical Response) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6399ms with no response
Really destroying SIP dialog ‘0_4179623970@192.168.0.1’ Method: CANCEL
Reliably Transmitting (no NAT) to 192.168.0.1:5060:
OPTIONS sip:0339@192.168.0.1:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.254:5060;branch=z9hG4bK7ffaa09d
Max-Forwards: 70
From: “asterisk” sip:asterisk@192.168.0.254;tag=as682706c7
To: sip:0339@192.168.0.1:5060
Contact: sip:asterisk@192.168.0.254:5060
Call-ID: 161d2b9477186415252dca9106e59258@192.168.0.254:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 14.1.1
Date: Tue, 08 Nov 2016 14:39:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:192.168.0.1:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.254:5060;branch=z9hG4bK7ffaa09d
From: “asterisk” sip:asterisk@192.168.0.254;tag=as682706c7
To: sip:0339@192.168.0.1:5060;tag=601529505
Call-ID: 161d2b9477186415252dca9106e59258@192.168.0.254:5060
CSeq: 102 OPTIONS
User-Agent: Yealink SIP-T27P 45.81.0.15
Content-Length: 0

<------------->
— (8 headers 0 lines) —
Really destroying SIP dialog ‘161d2b9477186415252dca9106e59258@192.168.0.254:5060’ Method: OPTIONS
Really destroying SIP dialog ‘0_3424179002@192.168.0.1’ Method: SUBSCRIBE

<— SIP read from UDP:192.168.0.1:5060 —>
REGISTER sip:192.168.0.254:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK3199882911
From: “0339 - Test” sip:0339@192.168.0.254:5060;tag=3936992920
To: “0339 - Test” sip:0339@192.168.0.254:5060
Call-ID: 0_4134003658@192.168.0.1
CSeq: 1 REGISTER
Contact: sip:0339@192.168.0.1:5060
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: Yealink SIP-T27P 45.81.0.15
Expires: 900
Allow-Events: talk,hold,conference,refer,check-sync
Content-Length: 0

<------------->
— (13 headers 0 lines) —
Sending to 192.168.0.1:5060 (no NAT)
Sending to 192.168.0.1:5060 (no NAT)

<— Transmitting (no NAT) to 192.168.0.1:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK3199882911;received=192.168.0.1
From: “0339 - Test” sip:0339@192.168.0.254:5060;tag=3936992920
To: “0339 - Test” sip:0339@192.168.0.254:5060;tag=as1ec2f5c1
Call-ID: 0_4134003658@192.168.0.1
CSeq: 1 REGISTER
Server: Asterisk PBX 14.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“debortoli.lan”, nonce="2f39b3cd"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘0_4134003658@192.168.0.1’ in 32000 ms (Method: REGISTER)

<— SIP read from UDP:192.168.0.1:5060 —>
REGISTER sip:192.168.0.254:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK2434476758
From: “0339 - Test” sip:0339@192.168.0.254:5060;tag=3936992920
To: “0339 - Test” sip:0339@192.168.0.254:5060
Call-ID: 0_4134003658@192.168.0.1
CSeq: 2 REGISTER
Contact: sip:0339@192.168.0.1:5060
Authorization: Digest username=“0339”, realm=“debortoli.lan”, nonce=“2f39b3cd”, uri=“sip:192.168.0.254:5060”, response=“859ada044307c75afc6be81993464eda”, algorithm=MD5
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: Yealink SIP-T27P 45.81.0.15
Expires: 900
Allow-Events: talk,hold,conference,refer,check-sync
Content-Length: 0

<------------->
— (14 headers 0 lines) —
Sending to 192.168.0.1:5060 (no NAT)
Reliably Transmitting (no NAT) to 192.168.0.1:5060:
OPTIONS sip:0339@192.168.0.1:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.254:5060;branch=z9hG4bK65b03588
Max-Forwards: 70
From: “asterisk” sip:asterisk@192.168.0.254;tag=as645ab26d
To: sip:0339@192.168.0.1:5060
Contact: sip:asterisk@192.168.0.254:5060
Call-ID: 3268ce95084e02047bf996fc57224473@192.168.0.254:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 14.1.1
Date: Tue, 08 Nov 2016 14:39:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<— Transmitting (no NAT) to 192.168.0.1:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK2434476758;received=192.168.0.1
From: “0339 - Test” sip:0339@192.168.0.254:5060;tag=3936992920
To: “0339 - Test” sip:0339@192.168.0.254:5060;tag=as1ec2f5c1
Call-ID: 0_4134003658@192.168.0.1
CSeq: 2 REGISTER
Server: Asterisk PBX 14.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 900
Contact: sip:0339@192.168.0.1:5060;expires=900
Date: Tue, 08 Nov 2016 14:39:32 GMT
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘0_4134003658@192.168.0.1’ in 32000 ms (Method: REGISTER)

<— SIP read from UDP:192.168.0.1:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.254:5060;branch=z9hG4bK65b03588
From: “asterisk” sip:asterisk@192.168.0.254;tag=as645ab26d
To: sip:0339@192.168.0.1:5060;tag=2970804908
Call-ID: 3268ce95084e02047bf996fc57224473@192.168.0.254:5060
CSeq: 102 OPTIONS
User-Agent: Yealink SIP-T27P 45.81.0.15
Content-Length: 0

<------------->
— (8 headers 0 lines) —
Really destroying SIP dialog ‘3268ce95084e02047bf996fc57224473@192.168.0.254:5060’ Method: OPTIONS

<— SIP read from UDP:192.168.0.1:5060 —>
SUBSCRIBE sip:129@192.168.0.254:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK230367053
From: “0339 - Test” sip:0339@192.168.0.254:5060;tag=3222809128
To: sip:129@192.168.0.254:5060
Call-ID: 0_2832182223@192.168.0.1
CSeq: 1 SUBSCRIBE
Contact: sip:0339@192.168.0.1:5060
Accept: application/simple-message-summary
Max-Forwards: 70
User-Agent: Yealink SIP-T27P 45.81.0.15
Expires: 3600
Event: message-summary
Content-Length: 0

<------------->
— (13 headers 0 lines) —
Sending to 192.168.0.1:5060 (no NAT)
Creating new subscription
Sending to 192.168.0.1:5060 (no NAT)
sip_route_dump: route/path hop: sip:0339@192.168.0.1:5060
Found peer ‘0339’ for ‘0339’ from 192.168.0.1:5060

<— Transmitting (no NAT) to 192.168.0.1:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK230367053;received=192.168.0.1
From: “0339 - Test” sip:0339@192.168.0.254:5060;tag=3222809128
To: sip:129@192.168.0.254:5060;tag=as75fe6b4e
Call-ID: 0_2832182223@192.168.0.1
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX 14.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“debortoli.lan”, nonce="72632168"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘0_2832182223@192.168.0.1’ in 6400 ms (Method: SUBSCRIBE)

<— SIP read from UDP:192.168.0.1:5060 —>
SUBSCRIBE sip:129@192.168.0.254:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK1642300663
From: “0339 - Test” sip:0339@192.168.0.254:5060;tag=3222809128
To: sip:129@192.168.0.254:5060
Call-ID: 0_2832182223@192.168.0.1
CSeq: 2 SUBSCRIBE
Contact: sip:0339@192.168.0.1:5060
Authorization: Digest username=“0339”, realm=“debortoli.lan”, nonce=“72632168”, uri=“sip:129@192.168.0.254:5060”, response=“a8c934edcbadfea0c1e1e9941dd2bd5e”, algorithm=MD5
Accept: application/simple-message-summary
Max-Forwards: 70
User-Agent: Yealink SIP-T27P 45.81.0.15
Expires: 3600
Event: message-summary
Content-Length: 0

<------------->
— (14 headers 0 lines) —
Creating new subscription
Sending to 192.168.0.1:5060 (no NAT)
Found peer ‘0339’ for ‘0339’ from 192.168.0.1:5060
Scheduling destruction of SIP dialog ‘0_2832182223@192.168.0.1’ in 3610000 ms (Method: SUBSCRIBE)

<— Transmitting (no NAT) to 192.168.0.1:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK1642300663;received=192.168.0.1
From: “0339 - Test” sip:0339@192.168.0.254:5060;tag=3222809128
To: sip:129@192.168.0.254:5060;tag=as75fe6b4e
Call-ID: 0_2832182223@192.168.0.1
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX 14.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 3600
Contact: sip:129@192.168.0.254:5060;expires=3600
Content-Length: 0

<------------>
Reliably Transmitting (no NAT) to 192.168.0.1:5060:
NOTIFY sip:0339@192.168.0.1:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.254:5060;branch=z9hG4bK126d7952
Max-Forwards: 70
Route: sip:0339@192.168.0.1:5060
From: “asterisk” sip:asterisk@192.168.0.254;tag=as75fe6b4e
To: sip:0339@192.168.0.1:5060;tag=3222809128
Contact: sip:asterisk@192.168.0.254:5060
Call-ID: 0_2832182223@192.168.0.1
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX 14.1.1
Event: message-summary
Content-Type: application/simple-message-summary
Subscription-State: active
Content-Length: 91

Messages-Waiting: no
Message-Account: sip:asterisk@192.168.0.254
Voice-Message: 0/0 (0/0)

<— SIP read from UDP:192.168.0.1:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.254:5060;branch=z9hG4bK126d7952
From: “asterisk” sip:asterisk@192.168.0.254;tag=as75fe6b4e
To: sip:0339@192.168.0.1:5060;tag=3222809128
Call-ID: 0_2832182223@192.168.0.1
CSeq: 102 NOTIFY
Contact: sip:0339@192.168.0.1:5060
User-Agent: Yealink SIP-T27P 45.81.0.15
Content-Length: 0

<------------->
— (9 headers 0 lines) —
dbgitvoipmb*CLI> sip set debug off
SIP Debugging Disabled

7

It seems that the evaluation of dialplan takes very long time

8

What is the exact dialplan for that case?

9

;Call to test
exten => _661XX,1,NoOp(Call to test)
exten => _661XX,n,Dial(SIP/PSTN/${EXTEN})
exten => _661XX,n,Hangup()

10

That’s the specific extension.

What I’m looking for is the full contents of the contexts as it is in extensions.conf