Retrasmission timeout - Originate between conferences

Asterisk Asterisk Support
1

Hi everyone,
I have two Asterisk boxes connected through a trunk,

[ul]Asterisk 11.4.0 ( ip = 10.9.1.8 )
Asterisk 1.8.15.0 ( ip = 10.9.40.251 )[/ul]

on both ones there are extensions that add to a conference room: 31 ( on 10.9.40.251 ) and 0000995 ( on 10.9.1.8 ).
At startup of 10.9.40.251 I send an originate to merge the two conferences

channel originate SIP/0000995@0000-main extension 31@from-internal

After around 15 minutes the connection falls with the following error

[2013-11-15 18:21:08] WARNING[3279]: chan_sip.c:4169 retrans_pkt: Retransmission timeout reached on transmission 28f4197e538068f27189dc9c402454e2@10.9.40.251:5060 for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6400ms with no response [2013-11-15 18:21:08] WARNING[3279]: chan_sip.c:4198 retrans_pkt: Hanging up call 28f4197e538068f27189dc9c402454e2@10.9.40.251:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). == Spawn extension (from-internal, 0000995, 7) exited non-zero on 'SIP/0301-tar-00000000' == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/0301-tar-00000000'

I googled the error and it seems it’s related to the nat setting, but there’s no nat in my network so the peers used for the trunk are configured like this.

[0301-tar] username=0301-tar authname=0301-tar type=peer secret=voiptrunk qualify=yes nat=no language=it insecure=port,invite host=dynamic disallow=all allow=alaw context=from-0301-tar canreinvite=no

[0000-main] username=0000-main authname=0000-main type=peer secret=voiptrunk qualify=yes nat=no language=it insecure=port,invite host=dynamic disallow=all allow=alaw ;allow=ulaw context=from-0000-main canreinvite=no

I dunno what I am doing wrong.

Thanks for your help

P.s. The problem here is an update of this one that I solved with a workaround (now i check the channel locally and the conference partecipants remotely)

2

At 15 minutes, it is most likely due to a problem with session timers.

Generally with retransmission timeouts, you need to enable at least sip history, although normally one would do a full sip trace with sip set debug on, and see exactly which packet is not getting through.

Not related to your current problem, canreinvite is deprecated, and may even be non-functional in the latest versions, and, more importantly, you have an invalid combination of options from a security point of view. insecure=invite is incompatible with host=ip and the configuring of secret on the peer; the secret will never be used. The chances are that insecure=port is not benefiting you, either.

Unfortunately, because making things secure can make misconfigurations more obvious, most cookbook configurations seem to include “insecure=port, invite”, or the, obsolete, equivalent, “very”. However, the name, “insecure”, was supposed to warn you that this option reduces security and should only be used if you understand what it does and why you need it.

3

Hi david55, first of all thank you VERY much for answering

[quote=“david55”]At 15 minutes, it is most likely due to a problem with session timers.

Generally with retransmission timeouts, you need to enable at least sip history, although normally one would do a full sip trace with sip set debug on, and see exactly which packet is not getting through.
[/quote]

I enabled debugging and this is what i got

This is from box 10.9.1.8 that has the 0301-tar peer

[code]<— SIP read from UDP:10.9.40.251:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.9.1.8:5060;branch=z9hG4bK424657eb;received=10.9.1.8;rport=5060
From: “Unknown” sip:Unknown@10.9.1.8;tag=as5628d370
To: sip:s@10.9.40.251:5060;tag=as4dcef6ba
Call-ID: 4a336f1c7c20767a6cd0c96702916bcb@10.9.1.8:5060
CSeq: 102 OPTIONS
Server: Asterisk PBX 1.8.15.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:10.9.40.251:5060
Accept: application/sdp
Content-Length: 0

<------------->
— (12 headers 0 lines) —
Really destroying SIP dialog ‘4a336f1c7c20767a6cd0c96702916bcb@10.9.1.8:5060’ Method: OPTIONS
Really destroying SIP dialog ‘59481a593e4b3919139f5e983161f0cb@10.9.40.251:5060’ Method: OPTIONS
[2013-11-18 09:44:32] NOTICE[3279]: chan_sip.c:14948 sip_reregister: – Re-registration for 0000-main@10.9.40.251
REGISTER 11 headers, 0 lines
Reliably Transmitting (no NAT) to 10.9.40.251:5060:
REGISTER sip:10.9.40.251 SIP/2.0
Via: SIP/2.0/UDP 10.9.1.8:5060;branch=z9hG4bK5345b143
Max-Forwards: 70
From: sip:0000-main@10.9.40.251;tag=as39d7bcc3
To: sip:0000-main@10.9.40.251
Call-ID: 577d13a42b4ec2233e99f3bf151b4789@10.138.21.118
CSeq: 4466 REGISTER
User-Agent: FPBX-2.11.0(11.4.0)
Authorization: Digest username=“0000-main”, realm=“asterisk”, algorithm=MD5, uri=“sip:10.9.40.251”, nonce=“3bb49214”, response="c8c8e4ec9b4f3948d4aede65871f741b"
Expires: 120
Contact: sip:s@10.9.1.8:5060
Content-Length: 0

<— SIP read from UDP:10.9.40.251:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.9.1.8:5060;branch=z9hG4bK5345b143;received=10.9.1.8
From: sip:0000-main@10.9.40.251;tag=as39d7bcc3
To: sip:0000-main@10.9.40.251;tag=as273de12f
Call-ID: 577d13a42b4ec2233e99f3bf151b4789@10.138.21.118
CSeq: 4466 REGISTER
Server: Asterisk PBX 1.8.15.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="36890416"
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Responding to challenge, registration to domain/host name 10.9.40.251
REGISTER 11 headers, 0 lines
Reliably Transmitting (no NAT) to 10.9.40.251:5060:
REGISTER sip:10.9.40.251 SIP/2.0
Via: SIP/2.0/UDP 10.9.1.8:5060;branch=z9hG4bK3ac3ecbb
Max-Forwards: 70
From: sip:0000-main@10.9.40.251;tag=as3c63a265
To: sip:0000-main@10.9.40.251
Call-ID: 577d13a42b4ec2233e99f3bf151b4789@10.138.21.118
CSeq: 4467 REGISTER
User-Agent: FPBX-2.11.0(11.4.0)
Authorization: Digest username=“0000-main”, realm=“asterisk”, algorithm=MD5, uri=“sip:10.9.40.251”, nonce=“36890416”, response="96c53bddcaef0412afa389ce1750469b"
Expires: 120
Contact: sip:s@10.9.1.8:5060
Content-Length: 0

<— SIP read from UDP:10.9.40.251:5060 —>
OPTIONS sip:s@10.9.1.8:5060 SIP/2.0
Via: SIP/2.0/UDP 10.9.40.251:5060;branch=z9hG4bK40ec9bc1
Max-Forwards: 70
From: “asterisk” <sip:0301000 S1@10.9.40.251>;tag=as09a7e11a
To: sip:s@10.9.1.8:5060
Contact: <sip:0301000 S1@10.9.40.251:5060>
Call-ID: 4297596d59a5da273aa080a872a2f2c7@10.9.40.251:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.15.0
Date: Mon, 18 Nov 2013 08:44:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
— (13 headers 0 lines) —
Sending to 10.9.40.251:5060 (no NAT)
Looking for s in from-sip-external (domain 10.9.1.8)

<— Transmitting (no NAT) to 10.9.40.251:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.9.40.251:5060;branch=z9hG4bK40ec9bc1;received=10.9.40.251
From: “asterisk” <sip:0301000 S1@10.9.40.251>;tag=as09a7e11a
To: sip:s@10.9.1.8:5060;tag=as12954c03
Call-ID: 4297596d59a5da273aa080a872a2f2c7@10.9.40.251:5060
CSeq: 102 OPTIONS
Server: FPBX-2.11.0(11.4.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:10.9.1.8:5060
Accept: application/sdp
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘4297596d59a5da273aa080a872a2f2c7@10.9.40.251:5060’ in 32000 ms (Method: OPTIONS)

<— SIP read from UDP:10.9.40.251:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.9.1.8:5060;branch=z9hG4bK3ac3ecbb;received=10.9.1.8
From: sip:0000-main@10.9.40.251;tag=as3c63a265
To: sip:0000-main@10.9.40.251;tag=as273de12f
Call-ID: 577d13a42b4ec2233e99f3bf151b4789@10.138.21.118
CSeq: 4467 REGISTER
Server: Asterisk PBX 1.8.15.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 120
Contact: sip:s@10.9.1.8:5060;expires=120
Date: Mon, 18 Nov 2013 08:44:32 GMT
Content-Length: 0

<------------->
— (13 headers 0 lines) —
Scheduling destruction of SIP dialog ‘577d13a42b4ec2233e99f3bf151b4789@10.138.21.118’ in 32000 ms (Method: REGISTER)
[2013-11-18 09:44:32] NOTICE[3279]: chan_sip.c:23314 handle_response_register: Outbound Registration: Expiry for 10.9.40.251 is 120 sec (Scheduling reregistration in 105 s)

<— SIP read from UDP:10.9.40.251:5060 —>
REGISTER sip:10.9.1.8 SIP/2.0
Via: SIP/2.0/UDP 10.9.40.251:5060;branch=z9hG4bK77750b11;rport
Max-Forwards: 70
From: sip:0301-tar@10.9.1.8;tag=as2da51302
To: sip:0301-tar@10.9.1.8
Call-ID: 5e2d1c8649bc49827a13da0030b55f8a@127.0.1.1
CSeq: 4480 REGISTER
User-Agent: Asterisk PBX 1.8.15.0
Authorization: Digest username=“0301-tar”, realm=“asterisk”, algorithm=MD5, uri=“sip:10.9.1.8”, nonce=“467b8231”, response="74abc08d8025b7bc66e8a35ec0a03ef0"
Expires: 120
Contact: sip:s@10.9.40.251:5060
Content-Length: 0

<------------->
— (12 headers 0 lines) —
Sending to 10.9.40.251:5060 (no NAT)
Sending to 10.9.40.251:5060 (no NAT)

<— Transmitting (no NAT) to 10.9.40.251:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.9.40.251:5060;branch=z9hG4bK77750b11;received=10.9.40.251;rport=5060
From: sip:0301-tar@10.9.1.8;tag=as2da51302
To: sip:0301-tar@10.9.1.8;tag=as236b6452
Call-ID: 5e2d1c8649bc49827a13da0030b55f8a@127.0.1.1
CSeq: 4480 REGISTER
Server: FPBX-2.11.0(11.4.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="74aa7fb3"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘5e2d1c8649bc49827a13da0030b55f8a@127.0.1.1’ in 32000 ms (Method: REGISTER)

<— SIP read from UDP:10.9.40.251:5060 —>
REGISTER sip:10.9.1.8 SIP/2.0
Via: SIP/2.0/UDP 10.9.40.251:5060;branch=z9hG4bK435cfb16;rport
Max-Forwards: 70
From: sip:0301-tar@10.9.1.8;tag=as673f7d1d
To: sip:0301-tar@10.9.1.8
Call-ID: 5e2d1c8649bc49827a13da0030b55f8a@127.0.1.1
CSeq: 4481 REGISTER
User-Agent: Asterisk PBX 1.8.15.0
Authorization: Digest username=“0301-tar”, realm=“asterisk”, algorithm=MD5, uri=“sip:10.9.1.8”, nonce=“74aa7fb3”, response="fa4aca72a742fcf41c38f6c8d12167e4"
Expires: 120
Contact: sip:s@10.9.40.251:5060
Content-Length: 0

<------------->
— (12 headers 0 lines) —
Sending to 10.9.40.251:5060 (no NAT)
Reliably Transmitting (no NAT) to 10.9.40.251:5060:
OPTIONS sip:s@10.9.40.251:5060 SIP/2.0
Via: SIP/2.0/UDP 10.9.1.8:5060;branch=z9hG4bK3c2b17bc
Max-Forwards: 70
From: “Unknown” sip:Unknown@10.9.1.8;tag=as6af300c5
To: sip:s@10.9.40.251:5060
Contact: sip:Unknown@10.9.1.8:5060
Call-ID: 25cbdd250292b06033ee832424782856@10.9.1.8:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.11.0(11.4.0)
Date: Mon, 18 Nov 2013 08:44:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<— Transmitting (no NAT) to 10.9.40.251:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.9.40.251:5060;branch=z9hG4bK435cfb16;received=10.9.40.251;rport=5060
From: sip:0301-tar@10.9.1.8;tag=as673f7d1d
To: sip:0301-tar@10.9.1.8;tag=as236b6452
Call-ID: 5e2d1c8649bc49827a13da0030b55f8a@127.0.1.1
CSeq: 4481 REGISTER
Server: FPBX-2.11.0(11.4.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 120
Contact: sip:s@10.9.40.251:5060;expires=120
Date: Mon, 18 Nov 2013 08:44:37 GMT
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘5e2d1c8649bc49827a13da0030b55f8a@127.0.1.1’ in 32000 ms (Method: REGISTER)

<— SIP read from UDP:10.9.40.251:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.9.1.8:5060;branch=z9hG4bK3c2b17bc;received=10.9.1.8;rport=5060
From: “Unknown” sip:Unknown@10.9.1.8;tag=as6af300c5
To: sip:s@10.9.40.251:5060;tag=as2589ae4b
Call-ID: 25cbdd250292b06033ee832424782856@10.9.1.8:5060
CSeq: 102 OPTIONS
Server: Asterisk PBX 1.8.15.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:10.9.40.251:5060
Accept: application/sdp
Content-Length: 0

<------------->
— (12 headers 0 lines) —
Really destroying SIP dialog ‘25cbdd250292b06033ee832424782856@10.9.1.8:5060’ Method: OPTIONS
Really destroying SIP dialog ‘4297596d59a5da273aa080a872a2f2c7@10.9.40.251:5060’ Method: OPTIONS
Really destroying SIP dialog ‘577d13a42b4ec2233e99f3bf151b4789@10.138.21.118’ Method: REGISTER
Really destroying SIP dialog ‘5e2d1c8649bc49827a13da0030b55f8a@127.0.1.1’ Method: REGISTER
ETIPBX-lab*CLI> meetme list 1000
User #: 01 0301000 S1 Ata+ Channel: SIP/0301-tar-000000ee (unmonitored) 00:12:13
1 users in that conference.

<— SIP read from UDP:10.9.40.251:5060 —>
OPTIONS sip:s@10.9.1.8:5060 SIP/2.0
Via: SIP/2.0/UDP 10.9.40.251:5060;branch=z9hG4bK09cb1a87
Max-Forwards: 70
From: “asterisk” <sip:0301000 S1@10.9.40.251>;tag=as10a4753b
To: sip:s@10.9.1.8:5060
Contact: <sip:0301000 S1@10.9.40.251:5060>
Call-ID: 58aa2a042e3cbf3157c53ed263803535@10.9.40.251:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.15.0
Date: Mon, 18 Nov 2013 08:45:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
— (13 headers 0 lines) —
Sending to 10.9.40.251:5060 (no NAT)
Looking for s in from-sip-external (domain 10.9.1.8)

<— Transmitting (no NAT) to 10.9.40.251:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.9.40.251:5060;branch=z9hG4bK09cb1a87;received=10.9.40.251
From: “asterisk” <sip:0301000 S1@10.9.40.251>;tag=as10a4753b
To: sip:s@10.9.1.8:5060;tag=as3878eb34
Call-ID: 58aa2a042e3cbf3157c53ed263803535@10.9.40.251:5060
CSeq: 102 OPTIONS
Server: FPBX-2.11.0(11.4.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:10.9.1.8:5060
Accept: application/sdp
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘58aa2a042e3cbf3157c53ed263803535@10.9.40.251:5060’ in 32000 ms (Method: OPTIONS)
Reliably Transmitting (no NAT) to 10.9.40.251:5060:
OPTIONS sip:s@10.9.40.251:5060 SIP/2.0
Via: SIP/2.0/UDP 10.9.1.8:5060;branch=z9hG4bK50f977f5
Max-Forwards: 70
From: “Unknown” sip:Unknown@10.9.1.8;tag=as45506dc1
To: sip:s@10.9.40.251:5060
Contact: sip:Unknown@10.9.1.8:5060
Call-ID: 63fe76044aae46e60da26d6d3f5e59fc@10.9.1.8:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.11.0(11.4.0)
Date: Mon, 18 Nov 2013 08:45:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:10.9.40.251:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.9.1.8:5060;branch=z9hG4bK50f977f5;received=10.9.1.8;rport=5060
From: “Unknown” sip:Unknown@10.9.1.8;tag=as45506dc1
To: sip:s@10.9.40.251:5060;tag=as20499ced
Call-ID: 63fe76044aae46e60da26d6d3f5e59fc@10.9.1.8:5060
CSeq: 102 OPTIONS
Server: Asterisk PBX 1.8.15.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:10.9.40.251:5060
Accept: application/sdp
Content-Length: 0

<------------->
— (12 headers 0 lines) —
Really destroying SIP dialog ‘63fe76044aae46e60da26d6d3f5e59fc@10.9.1.8:5060’ Method: OPTIONS
Really destroying SIP dialog ‘58aa2a042e3cbf3157c53ed263803535@10.9.40.251:5060’ Method: OPTIONS
[2013-11-18 09:46:17] NOTICE[3279]: chan_sip.c:14948 sip_reregister: – Re-registration for 0000-main@10.9.40.251
REGISTER 11 headers, 0 lines
Reliably Transmitting (no NAT) to 10.9.40.251:5060:
REGISTER sip:10.9.40.251 SIP/2.0
Via: SIP/2.0/UDP 10.9.1.8:5060;branch=z9hG4bK41b60924
Max-Forwards: 70
From: sip:0000-main@10.9.40.251;tag=as04c63f3b
To: sip:0000-main@10.9.40.251
Call-ID: 577d13a42b4ec2233e99f3bf151b4789@10.138.21.118
CSeq: 4468 REGISTER
User-Agent: FPBX-2.11.0(11.4.0)
Authorization: Digest username=“0000-main”, realm=“asterisk”, algorithm=MD5, uri=“sip:10.9.40.251”, nonce=“36890416”, response="96c53bddcaef0412afa389ce1750469b"
Expires: 120
Contact: sip:s@10.9.1.8:5060
Content-Length: 0

<— SIP read from UDP:10.9.40.251:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.9.1.8:5060;branch=z9hG4bK41b60924;received=10.9.1.8
From: sip:0000-main@10.9.40.251;tag=as04c63f3b
To: sip:0000-main@10.9.40.251;tag=as7dd84c7a
Call-ID: 577d13a42b4ec2233e99f3bf151b4789@10.138.21.118
CSeq: 4468 REGISTER
Server: Asterisk PBX 1.8.15.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="62390ddd"
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Responding to challenge, registration to domain/host name 10.9.40.251
REGISTER 11 headers, 0 lines
Reliably Transmitting (no NAT) to 10.9.40.251:5060:
REGISTER sip:10.9.40.251 SIP/2.0
Via: SIP/2.0/UDP 10.9.1.8:5060;branch=z9hG4bK343b7f1c
Max-Forwards: 70
From: sip:0000-main@10.9.40.251;tag=as7329a9dd
To: sip:0000-main@10.9.40.251
Call-ID: 577d13a42b4ec2233e99f3bf151b4789@10.138.21.118
CSeq: 4469 REGISTER
User-Agent: FPBX-2.11.0(11.4.0)
Authorization: Digest username=“0000-main”, realm=“asterisk”, algorithm=MD5, uri=“sip:10.9.40.251”, nonce=“62390ddd”, response="8c82233a06874e5e325f1e587db52763"
Expires: 120
Contact: sip:s@10.9.1.8:5060
Content-Length: 0

<— SIP read from UDP:10.9.40.251:5060 —>
OPTIONS sip:s@10.9.1.8:5060 SIP/2.0
Via: SIP/2.0/UDP 10.9.40.251:5060;branch=z9hG4bK5fb93015
Max-Forwards: 70
From: “asterisk” <sip:0301000 S1@10.9.40.251>;tag=as25094f0e
To: sip:s@10.9.1.8:5060
Contact: <sip:0301000 S1@10.9.40.251:5060>
Call-ID: 005b98e31708949a0644588a75710baa@10.9.40.251:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.15.0
Date: Mon, 18 Nov 2013 08:46:17 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
— (13 headers 0 lines) —
Sending to 10.9.40.251:5060 (no NAT)
Looking for s in from-sip-external (domain 10.9.1.8)

<— Transmitting (no NAT) to 10.9.40.251:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.9.40.251:5060;branch=z9hG4bK5fb93015;received=10.9.40.251
From: “asterisk” <sip:0301000 S1@10.9.40.251>;tag=as25094f0e
To: sip:s@10.9.1.8:5060;tag=as7d27734b
Call-ID: 005b98e31708949a0644588a75710baa@10.9.40.251:5060
CSeq: 102 OPTIONS
Server: FPBX-2.11.0(11.4.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:10.9.1.8:5060
Accept: application/sdp
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘005b98e31708949a0644588a75710baa@10.9.40.251:5060’ in 32000 ms (Method: OPTIONS)

<— SIP read from UDP:10.9.40.251:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.9.1.8:5060;branch=z9hG4bK343b7f1c;received=10.9.1.8
From: sip:0000-main@10.9.40.251;tag=as7329a9dd
To: sip:0000-main@10.9.40.251;tag=as7dd84c7a
Call-ID: 577d13a42b4ec2233e99f3bf151b4789@10.138.21.118
CSeq: 4469 REGISTER
Server: Asterisk PBX 1.8.15.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 120
Contact: sip:s@10.9.1.8:5060;expires=120
Date: Mon, 18 Nov 2013 08:46:17 GMT
Content-Length: 0

<------------->
— (13 headers 0 lines) —
Scheduling destruction of SIP dialog ‘577d13a42b4ec2233e99f3bf151b4789@10.138.21.118’ in 32000 ms (Method: REGISTER)
[2013-11-18 09:46:17] NOTICE[3279]: chan_sip.c:23314 handle_response_register: Outbound Registration: Expiry for 10.9.40.251 is 120 sec (Scheduling reregistration in 105 s)

<— SIP read from UDP:10.9.40.251:5060 —>
REGISTER sip:10.9.1.8 SIP/2.0
Via: SIP/2.0/UDP 10.9.40.251:5060;branch=z9hG4bK0d9128ee;rport
Max-Forwards: 70
From: sip:0301-tar@10.9.1.8;tag=as210b2a2e
To: sip:0301-tar@10.9.1.8
Call-ID: 5e2d1c8649bc49827a13da0030b55f8a@127.0.1.1
CSeq: 4482 REGISTER
User-Agent: Asterisk PBX 1.8.15.0
Authorization: Digest username=“0301-tar”, realm=“asterisk”, algorithm=MD5, uri=“sip:10.9.1.8”, nonce=“74aa7fb3”, response="fa4aca72a742fcf41c38f6c8d12167e4"
Expires: 120
Contact: sip:s@10.9.40.251:5060
Content-Length: 0

<------------->
— (12 headers 0 lines) —
Sending to 10.9.40.251:5060 (no NAT)
Sending to 10.9.40.251:5060 (no NAT)

<— Transmitting (no NAT) to 10.9.40.251:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.9.40.251:5060;branch=z9hG4bK0d9128ee;received=10.9.40.251;rport=5060
From: sip:0301-tar@10.9.1.8;tag=as210b2a2e
To: sip:0301-tar@10.9.1.8;tag=as14036749
Call-ID: 5e2d1c8649bc49827a13da0030b55f8a@127.0.1.1
CSeq: 4482 REGISTER
Server: FPBX-2.11.0(11.4.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="23291bfd"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘5e2d1c8649bc49827a13da0030b55f8a@127.0.1.1’ in 32000 ms (Method: REGISTER)

<— SIP read from UDP:10.9.40.251:5060 —>
REGISTER sip:10.9.1.8 SIP/2.0
Via: SIP/2.0/UDP 10.9.40.251:5060;branch=z9hG4bK35c4e985;rport
Max-Forwards: 70
From: sip:0301-tar@10.9.1.8;tag=as684106da
To: sip:0301-tar@10.9.1.8
Call-ID: 5e2d1c8649bc49827a13da0030b55f8a@127.0.1.1
CSeq: 4483 REGISTER
User-Agent: Asterisk PBX 1.8.15.0
Authorization: Digest username=“0301-tar”, realm=“asterisk”, algorithm=MD5, uri=“sip:10.9.1.8”, nonce=“23291bfd”, response="95f0952ef929890bb594c614718405ed"
Expires: 120
Contact: sip:s@10.9.40.251:5060
Content-Length: 0

<------------->
— (12 headers 0 lines) —
Sending to 10.9.40.251:5060 (no NAT)
Reliably Transmitting (no NAT) to 10.9.40.251:5060:
OPTIONS sip:s@10.9.40.251:5060 SIP/2.0
Via: SIP/2.0/UDP 10.9.1.8:5060;branch=z9hG4bK1dc816b6
Max-Forwards: 70
From: “Unknown” sip:Unknown@10.9.1.8;tag=as04e8d8be
To: sip:s@10.9.40.251:5060
Contact: sip:Unknown@10.9.1.8:5060
Call-ID: 30217e6d548033905de5f81224286494@10.9.1.8:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.11.0(11.4.0)
Date: Mon, 18 Nov 2013 08:46:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<— Transmitting (no NAT) to 10.9.40.251:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.9.40.251:5060;branch=z9hG4bK35c4e985;received=10.9.40.251;rport=5060
From: sip:0301-tar@10.9.1.8;tag=as684106da
To: sip:0301-tar@10.9.1.8;tag=as14036749
Call-ID: 5e2d1c8649bc49827a13da0030b55f8a@127.0.1.1
CSeq: 4483 REGISTER
Server: FPBX-2.11.0(11.4.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 120
Contact: sip:s@10.9.40.251:5060;expires=120
Date: Mon, 18 Nov 2013 08:46:22 GMT
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘5e2d1c8649bc49827a13da0030b55f8a@127.0.1.1’ in 32000 ms (Method: REGISTER)

<— SIP read from UDP:10.9.40.251:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.9.1.8:5060;branch=z9hG4bK1dc816b6;received=10.9.1.8;rport=5060
From: “Unknown” sip:Unknown@10.9.1.8;tag=as04e8d8be
To: sip:s@10.9.40.251:5060;tag=as4784ff59
Call-ID: 30217e6d548033905de5f81224286494@10.9.1.8:5060
CSeq: 102 OPTIONS
Server: Asterisk PBX 1.8.15.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:10.9.40.251:5060
Accept: application/sdp
Content-Length: 0

<------------->
— (12 headers 0 lines) —
Really destroying SIP dialog ‘30217e6d548033905de5f81224286494@10.9.1.8:5060’ Method: OPTIONS
Really destroying SIP dialog ‘005b98e31708949a0644588a75710baa@10.9.40.251:5060’ Method: OPTIONS
Really destroying SIP dialog ‘577d13a42b4ec2233e99f3bf151b4789@10.138.21.118’ Method: REGISTER
Really destroying SIP dialog ‘5e2d1c8649bc49827a13da0030b55f8a@127.0.1.1’ Method: REGISTER
ETIPBX-lab*CLI> meetme list 1000
User #: 01 0301000 S1 Ata+ Channel: SIP/0301-tar-000000ee (unmonitored) 00:13:53
1 users in that conference.

<— SIP read from UDP:10.9.40.251:5060 —>
OPTIONS sip:s@10.9.1.8:5060 SIP/2.0
Via: SIP/2.0/UDP 10.9.40.251:5060;branch=z9hG4bK0a0069ef
Max-Forwards: 70
From: “asterisk” <sip:0301000 S1@10.9.40.251>;tag=as7d11d235
To: sip:s@10.9.1.8:5060
Contact: <sip:0301000 S1@10.9.40.251:5060>
Call-ID: 22873e097cf9e6e443f394ba03f6f916@10.9.40.251:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.15.0
Date: Mon, 18 Nov 2013 08:47:17 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
— (13 headers 0 lines) —
Sending to 10.9.40.251:5060 (no NAT)
Looking for s in from-sip-external (domain 10.9.1.8)

<— Transmitting (no NAT) to 10.9.40.251:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.9.40.251:5060;branch=z9hG4bK0a0069ef;received=10.9.40.251
From: “asterisk” <sip:0301000 S1@10.9.40.251>;tag=as7d11d235
To: sip:s@10.9.1.8:5060;tag=as0ce97fae
Call-ID: 22873e097cf9e6e443f394ba03f6f916@10.9.40.251:5060
CSeq: 102 OPTIONS
Server: FPBX-2.11.0(11.4.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:10.9.1.8:5060
Accept: application/sdp
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘22873e097cf9e6e443f394ba03f6f916@10.9.40.251:5060’ in 32000 ms (Method: OPTIONS)
Reliably Transmitting (no NAT) to 10.9.40.251:5060:
OPTIONS sip:s@10.9.40.251:5060 SIP/2.0
Via: SIP/2.0/UDP 10.9.1.8:5060;branch=z9hG4bK3f91faf9
Max-Forwards: 70
From: “Unknown” sip:Unknown@10.9.1.8;tag=as6446b0d6
To: sip:s@10.9.40.251:5060
Contact: sip:Unknown@10.9.1.8:5060
Call-ID: 27b8d98d2d355dcc4fb4386e45e196c5@10.9.1.8:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.11.0(11.4.0)
Date: Mon, 18 Nov 2013 08:47:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<— SIP read from UDP:10.9.40.251:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.9.1.8:5060;branch=z9hG4bK3f91faf9;received=10.9.1.8;rport=5060
From: “Unknown” sip:Unknown@10.9.1.8;tag=as6446b0d6
To: sip:s@10.9.40.251:5060;tag=as27cfd4fb
Call-ID: 27b8d98d2d355dcc4fb4386e45e196c5@10.9.1.8:5060
CSeq: 102 OPTIONS
Server: Asterisk PBX 1.8.15.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:10.9.40.251:5060
Accept: application/sdp
Content-Length: 0

<------------->
— (12 headers 0 lines) —
Really destroying SIP dialog ‘27b8d98d2d355dcc4fb4386e45e196c5@10.9.1.8:5060’ Method: OPTIONS
Really destroying SIP dialog ‘22873e097cf9e6e443f394ba03f6f916@10.9.40.251:5060’ Method: OPTIONS
[2013-11-18 09:48:02] NOTICE[3279]: chan_sip.c:14948 sip_reregister: – Re-registration for 0000-main@10.9.40.251
REGISTER 11 headers, 0 lines
Reliably Transmitting (no NAT) to 10.9.40.251:5060:
REGISTER sip:10.9.40.251 SIP/2.0
Via: SIP/2.0/UDP 10.9.1.8:5060;branch=z9hG4bK599e8dea
Max-Forwards: 70
From: sip:0000-main@10.9.40.251;tag=as1f0dec41
To: sip:0000-main@10.9.40.251
Call-ID: 577d13a42b4ec2233e99f3bf151b4789@10.138.21.118
CSeq: 4470 REGISTER
User-Agent: FPBX-2.11.0(11.4.0)
Authorization: Digest username=“0000-main”, realm=“asterisk”, algorithm=MD5, uri=“sip:10.9.40.251”, nonce=“62390ddd”, response="8c82233a06874e5e325f1e587db52763"
Expires: 120
Contact: sip:s@10.9.1.8:5060
Content-Length: 0

<— SIP read from UDP:10.9.40.251:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.9.1.8:5060;branch=z9hG4bK599e8dea;received=10.9.1.8
From: sip:0000-main@10.9.40.251;tag=as1f0dec41
To: sip:0000-main@10.9.40.251;tag=as47ae7732
Call-ID: 577d13a42b4ec2233e99f3bf151b4789@10.138.21.118
CSeq: 4470 REGISTER
Server: Asterisk PBX 1.8.15.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="329d6308"
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Responding to challenge, registration to domain/host name 10.9.40.251
REGISTER 11 headers, 0 lines
Reliably Transmitting (no NAT) to 10.9.40.251:5060:
REGISTER sip:10.9.40.251 SIP/2.0
Via: SIP/2.0/UDP 10.9.1.8:5060;branch=z9hG4bK5c10d840
Max-Forwards: 70
From: sip:0000-main@10.9.40.251;tag=as37e2cbef
To: sip:0000-main@10.9.40.251
Call-ID: 577d13a42b4ec2233e99f3bf151b4789@10.138.21.118
CSeq: 4471 REGISTER
User-Agent: FPBX-2.11.0(11.4.0)
Authorization: Digest username=“0000-main”, realm=“asterisk”, algorithm=MD5, uri=“sip:10.9.40.251”, nonce=“329d6308”, response="bace834c1c163589f57e40e6b51691bd"
Expires: 120
Contact: sip:s@10.9.1.8:5060
Content-Length: 0

<— SIP read from UDP:10.9.40.251:5060 —>
OPTIONS sip:s@10.9.1.8:5060 SIP/2.0
Via: SIP/2.0/UDP 10.9.40.251:5060;branch=z9hG4bK0f31a696
Max-Forwards: 70
From: “asterisk” <sip:0301000 S1@10.9.40.251>;tag=as7c801173
To: sip:s@10.9.1.8:5060
Contact: <sip:0301000 S1@10.9.40.251:5060>
Call-ID: 360cad9c04ca719b56e2a6454013ca0c@10.9.40.251:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.15.0
Date: Mon, 18 Nov 2013 08:48:02 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
— (13 headers 0 lines) —
Sending to 10.9.40.251:5060 (no NAT)
Looking for s in from-sip-external (domain 10.9.1.8)

<— Transmitting (no NAT) to 10.9.40.251:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.9.40.251:5060;branch=z9hG4bK0f31a696;received=10.9.40.251
From: “asterisk” <sip:0301000 S1@10.9.40.251>;tag=as7c801173
To: sip:s@10.9.1.8:5060;tag=as4ecc8417
Call-ID: 360cad9c04ca719b56e2a6454013ca0c@10.9.40.251:5060
CSeq: 102 OPTIONS
Server: FPBX-2.11.0(11.4.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:10.9.1.8:5060
Accept: application/sdp
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘360cad9c04ca719b56e2a6454013ca0c@10.9.40.251:5060’ in 32000 ms (Method: OPTIONS)

<— SIP read from UDP:10.9.40.251:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.9.1.8:5060;branch=z9hG4bK5c10d840;received=10.9.1.8
From: sip:0000-main@10.9.40.251;tag=as37e2cbef
To: sip:0000-main@10.9.40.251;tag=as47ae7732
Call-ID: 577d13a42b4ec2233e99f3bf151b4789@10.138.21.118
CSeq: 4471 REGISTER
Server: Asterisk PBX 1.8.15.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 120
Contact: sip:s@10.9.1.8:5060;expires=120
Date: Mon, 18 Nov 2013 08:48:02 GMT
Content-Length: 0

<------------->
— (13 headers 0 lines) —
Scheduling destruction of SIP dialog ‘577d13a42b4ec2233e99f3bf151b4789@10.138.21.118’ in 32000 ms (Method: REGISTER)
[2013-11-18 09:48:02] NOTICE[3279]: chan_sip.c:23314 handle_response_register: Outbound Registration: Expiry for 10.9.40.251 is 120 sec (Scheduling reregistration in 105 s)
ETIPBX-lab*CLI> meetme list 1000
User #: 01 0301000 S1 Ata+ Channel: SIP/0301-tar-000000ee (unmonitored) 00:14:53
1 users in that conference.

<— SIP read from UDP:10.9.40.251:5060 —>
REGISTER sip:10.9.1.8 SIP/2.0
Via: SIP/2.0/UDP 10.9.40.251:5060;branch=z9hG4bK20d51f0a;rport
Max-Forwards: 70
From: sip:0301-tar@10.9.1.8;tag=as62788d00
To: sip:0301-tar@10.9.1.8
Call-ID: 5e2d1c8649bc49827a13da0030b55f8a@127.0.1.1
CSeq: 4484 REGISTER
User-Agent: Asterisk PBX 1.8.15.0
Authorization: Digest username=“0301-tar”, realm=“asterisk”, algorithm=MD5, uri=“sip:10.9.1.8”, nonce=“23291bfd”, response="95f0952ef929890bb594c614718405ed"
Expires: 120
Contact: sip:s@10.9.40.251:5060
Content-Length: 0

<------------->
— (12 headers 0 lines) —
Sending to 10.9.40.251:5060 (no NAT)
Sending to 10.9.40.251:5060 (no NAT)

<— Transmitting (no NAT) to 10.9.40.251:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.9.40.251:5060;branch=z9hG4bK20d51f0a;received=10.9.40.251;rport=5060
From: sip:0301-tar@10.9.1.8;tag=as62788d00
To: sip:0301-tar@10.9.1.8;tag=as52c30229
Call-ID: 5e2d1c8649bc49827a13da0030b55f8a@127.0.1.1
CSeq: 4484 REGISTER
Server: FPBX-2.11.0(11.4.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="3c96aa63"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘5e2d1c8649bc49827a13da0030b55f8a@127.0.1.1’ in 32000 ms (Method: REGISTER)

<— SIP read from UDP:10.9.40.251:5060 —>
REGISTER sip:10.9.1.8 SIP/2.0
Via: SIP/2.0/UDP 10.9.40.251:5060;branch=z9hG4bK4edbee74;rport
Max-Forwards: 70
From: sip:0301-tar@10.9.1.8;tag=as5b342369
To: sip:0301-tar@10.9.1.8
Call-ID: 5e2d1c8649bc49827a13da0030b55f8a@127.0.1.1
CSeq: 4485 REGISTER
User-Agent: Asterisk PBX 1.8.15.0
Authorization: Digest username=“0301-tar”, realm=“asterisk”, algorithm=MD5, uri=“sip:10.9.1.8”, nonce=“3c96aa63”, response="dabddb693249c8a494b78307c9b699fb"
Expires: 120
Contact: sip:s@10.9.40.251:5060
Content-Length: 0

<------------->
— (12 headers 0 lines) —
Sending to 10.9.40.251:5060 (no NAT)
Reliably Transmitting (no NAT) to 10.9.40.251:5060:
OPTIONS sip:s@10.9.40.251:5060 SIP/2.0
Via: SIP/2.0/UDP 10.9.1.8:5060;branch=z9hG4bK38228855
Max-Forwards: 70
From: “Unknown” sip:Unknown@10.9.1.8;tag=as21b22a0d
To: sip:s@10.9.40.251:5060
Contact: sip:Unknown@10.9.1.8:5060
Call-ID: 67907f6b40227d5a7becda501f7f066b@10.9.1.8:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.11.0(11.4.0)
Date: Mon, 18 Nov 2013 08:48:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<— Transmitting (no NAT) to 10.9.40.251:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.9.40.251:5060;branch=z9hG4bK4edbee74;received=10.9.40.251;rport=5060
From: sip:0301-tar@10.9.1.8;tag=as5b342369
To: sip:0301-tar@10.9.1.8;tag=as52c30229
Call-ID: 5e2d1c8649bc49827a13da0030b55f8a@127.0.1.1
CSeq: 4485 REGISTER
Server: FPBX-2.11.0(11.4.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 120
Contact: sip:s@10.9.40.251:5060;expires=120
Date: Mon, 18 Nov 2013 08:48:07 GMT
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘5e2d1c8649bc49827a13da0030b55f8a@127.0.1.1’ in 32000 ms (Method: REGISTER)

<— SIP read from UDP:10.9.40.251:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.9.1.8:5060;branch=z9hG4bK38228855;received=10.9.1.8;rport=5060
From: “Unknown” sip:Unknown@10.9.1.8;tag=as21b22a0d
To: sip:s@10.9.40.251:5060;tag=as66bc3107
Call-ID: 67907f6b40227d5a7becda501f7f066b@10.9.1.8:5060
CSeq: 102 OPTIONS
Server: Asterisk PBX 1.8.15.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:10.9.40.251:5060
Accept: application/sdp
Content-Length: 0

<------------->
— (12 headers 0 lines) —
Really destroying SIP dialog ‘67907f6b40227d5a7becda501f7f066b@10.9.1.8:5060’ Method: OPTIONS
set_destination: Parsing <sip:0301000 S1@10.9.40.251:5060> for address/port to send to
set_destination: set destination to 10.9.40.251:5060
Audio is at 17422
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.9.40.251:5060:
INVITE sip:0301000 S1@10.9.40.251:5060 SIP/2.0
Via: SIP/2.0/UDP 10.9.1.8:5060;branch=z9hG4bK487b4b9b
Max-Forwards: 70
From: sip:0000995@10.9.1.8:5060;tag=as447458bb
To: “Anonymous” <sip:0301000 S1@anonymous.invalid>;tag=as485d5ddc
Contact: sip:0000995@10.9.1.8:5060
Call-ID: 591fc15e42e1514c3fe612f956707b21@10.9.40.251:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(11.4.0)
Session-Expires: 1800;refresher=uac
Min-SE: 90
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (Session-Timers)
Content-Type: application/sdp
Content-Length: 225

v=0
o=root 752480471 752480471 IN IP4 10.9.1.8
s=Asterisk PBX 11.4.0
c=IN IP4 10.9.1.8
t=0 0
m=audio 17422 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

Retransmitting #1 (no NAT) to 10.9.40.251:5060:
INVITE sip:0301000 S1@10.9.40.251:5060 SIP/2.0
Via: SIP/2.0/UDP 10.9.1.8:5060;branch=z9hG4bK487b4b9b
Max-Forwards: 70
From: sip:0000995@10.9.1.8:5060;tag=as447458bb
To: “Anonymous” <sip:0301000 S1@anonymous.invalid>;tag=as485d5ddc
Contact: sip:0000995@10.9.1.8:5060
Call-ID: 591fc15e42e1514c3fe612f956707b21@10.9.40.251:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(11.4.0)
Session-Expires: 1800;refresher=uac
Min-SE: 90
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (Session-Timers)
Content-Type: application/sdp
Content-Length: 225

v=0
o=root 752480471 752480471 IN IP4 10.9.1.8
s=Asterisk PBX 11.4.0
c=IN IP4 10.9.1.8
t=0 0
m=audio 17422 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

Retransmitting #2 (no NAT) to 10.9.40.251:5060:
INVITE sip:0301000 S1@10.9.40.251:5060 SIP/2.0
Via: SIP/2.0/UDP 10.9.1.8:5060;branch=z9hG4bK487b4b9b
Max-Forwards: 70
From: sip:0000995@10.9.1.8:5060;tag=as447458bb
To: “Anonymous” <sip:0301000 S1@anonymous.invalid>;tag=as485d5ddc
Contact: sip:0000995@10.9.1.8:5060
Call-ID: 591fc15e42e1514c3fe612f956707b21@10.9.40.251:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(11.4.0)
Session-Expires: 1800;refresher=uac
Min-SE: 90
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (Session-Timers)
Content-Type: application/sdp
Content-Length: 225

v=0
o=root 752480471 752480471 IN IP4 10.9.1.8
s=Asterisk PBX 11.4.0
c=IN IP4 10.9.1.8
t=0 0
m=audio 17422 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

Retransmitting #3 (no NAT) to 10.9.40.251:5060:
INVITE sip:0301000 S1@10.9.40.251:5060 SIP/2.0
Via: SIP/2.0/UDP 10.9.1.8:5060;branch=z9hG4bK487b4b9b
Max-Forwards: 70
From: sip:0000995@10.9.1.8:5060;tag=as447458bb
To: “Anonymous” <sip:0301000 S1@anonymous.invalid>;tag=as485d5ddc
Contact: sip:0000995@10.9.1.8:5060
Call-ID: 591fc15e42e1514c3fe612f956707b21@10.9.40.251:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(11.4.0)
Session-Expires: 1800;refresher=uac
Min-SE: 90
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (Session-Timers)
Content-Type: application/sdp
Content-Length: 225

v=0
o=root 752480471 752480471 IN IP4 10.9.1.8
s=Asterisk PBX 11.4.0
c=IN IP4 10.9.1.8
t=0 0
m=audio 17422 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

Retransmitting #4 (no NAT) to 10.9.40.251:5060:
INVITE sip:0301000 S1@10.9.40.251:5060 SIP/2.0
Via: SIP/2.0/UDP 10.9.1.8:5060;branch=z9hG4bK487b4b9b
Max-Forwards: 70
From: sip:0000995@10.9.1.8:5060;tag=as447458bb
To: “Anonymous” <sip:0301000 S1@anonymous.invalid>;tag=as485d5ddc
Contact: sip:0000995@10.9.1.8:5060
Call-ID: 591fc15e42e1514c3fe612f956707b21@10.9.40.251:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(11.4.0)
Session-Expires: 1800;refresher=uac
Min-SE: 90
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (Session-Timers)
Content-Type: application/sdp
Content-Length: 225

v=0
o=root 752480471 752480471 IN IP4 10.9.1.8
s=Asterisk PBX 11.4.0
c=IN IP4 10.9.1.8
t=0 0
m=audio 17422 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

Retransmitting #5 (no NAT) to 10.9.40.251:5060:
INVITE sip:0301000 S1@10.9.40.251:5060 SIP/2.0
Via: SIP/2.0/UDP 10.9.1.8:5060;branch=z9hG4bK487b4b9b
Max-Forwards: 70
From: sip:0000995@10.9.1.8:5060;tag=as447458bb
To: “Anonymous” <sip:0301000 S1@anonymous.invalid>;tag=as485d5ddc
Contact: sip:0000995@10.9.1.8:5060
Call-ID: 591fc15e42e1514c3fe612f956707b21@10.9.40.251:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(11.4.0)
Session-Expires: 1800;refresher=uac
Min-SE: 90
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (Session-Timers)
Content-Type: application/sdp
Content-Length: 225

v=0
o=root 752480471 752480471 IN IP4 10.9.1.8
s=Asterisk PBX 11.4.0
c=IN IP4 10.9.1.8
t=0 0
m=audio 17422 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

Retransmitting #6 (no NAT) to 10.9.40.251:5060:
INVITE sip:0301000 S1@10.9.40.251:5060 SIP/2.0
Via: SIP/2.0/UDP 10.9.1.8:5060;branch=z9hG4bK487b4b9b
Max-Forwards: 70
From: sip:0000995@10.9.1.8:5060;tag=as447458bb
To: “Anonymous” <sip:0301000 S1@anonymous.invalid>;tag=as485d5ddc
Contact: sip:0000995@10.9.1.8:5060
Call-ID: 591fc15e42e1514c3fe612f956707b21@10.9.40.251:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(11.4.0)
Session-Expires: 1800;refresher=uac
Min-SE: 90
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (Session-Timers)
Content-Type: application/sdp
Content-Length: 225

v=0
o=root 752480471 752480471 IN IP4 10.9.1.8
s=Asterisk PBX 11.4.0
c=IN IP4 10.9.1.8
t=0 0
m=audio 17422 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

[2013-11-18 09:48:19] WARNING[3279]: chan_sip.c:4169 retrans_pkt: Retransmission timeout reached on transmission 591fc15e42e1514c3fe612f956707b21@10.9.40.251:5060 for seqno 102 (Critical Request) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6399ms with no response
[2013-11-18 09:48:19] WARNING[3279]: chan_sip.c:4198 retrans_pkt: Hanging up call 591fc15e42e1514c3fe612f956707b21@10.9.40.251:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
== Spawn extension (from-internal, 0000995, 7) exited non-zero on ‘SIP/0301-tar-000000ee’
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/0301-tar-000000ee’
Really destroying SIP dialog ‘591fc15e42e1514c3fe612f956707b21@10.9.40.251:5060’ Method: ACK
Really destroying SIP dialog ‘360cad9c04ca719b56e2a6454013ca0c@10.9.40.251:5060’ Method: OPTIONS
Really destroying SIP dialog ‘577d13a42b4ec2233e99f3bf151b4789@10.138.21.118’ Method: REGISTER[/code]

I noticed there’s a periodic disconnection over the trunk that stops when i get the retrasmission timeout

and

but I don’t know which parameters to set; I’ve already tried setting these

session-timers=originate session-expires=1800 session-minse=90 session-refresher=uas

on the peer but what I got was more frequent disconnects

[quote=“david55”]
Not related to your current problem, canreinvite is deprecated, and may even be non-functional in the latest versions, and, more importantly, you have an invalid combination of options from a security point of view. insecure=invite is incompatible with host=ip and the configuring of secret on the peer; the secret will never be used. The chances are that insecure=port is not benefiting you, either.

Unfortunately, because making things secure can make misconfigurations more obvious, most cookbook configurations seem to include “insecure=port, invite”, or the, obsolete, equivalent, “very”. However, the name, “insecure”, was supposed to warn you that this option reduces security and should only be used if you understand what it does and why you need it.[/quote]

I have not understood well what the “insecure” setting might cause, to be honest I have configured it starting from the examples i found online, all I knew before is this.
I don’t get which setting fits better my needs

Thank you again

4

It is a session timer issue. You probably need to disable them.

The re-registers are normal.

Generally reducing security tends to make things more likely to work first time, so most cook book solutions to Asterisk, incorrectly, specify insecure=port, invite. In your case, you probably don’t need it at all, and you may not even need a secret. You certainly shouldn’t need invite.

In this sort of environment, you normally know the static address of both peers (what else do you put in the register line?), so you should not need to use host=dynamic.

5

It is a session timer issue. You probably need to disable them.[/quote]

Thanks david55! I set session-timers=refuse on both sides and now it works :smiley:

[quote=“david55”]Generally reducing security tends to make things more likely to work first time, so most cook book solutions to Asterisk, incorrectly, specify insecure=port, invite. In your case, you probably don’t need it at all, and you may not even need a secret. You certainly shouldn’t need invite.

In this sort of environment, you normally know the static address of both peers (what else do you put in the register line?), so you should not need to use host=dynamic.[/quote]

i understand what you mean now, I’ll try to take out that setting and make some tests!
Thank you very much again!