Elastix + ippi

**olisnow** · 28/01/2015, 13h22

v=0
o=root 2083461824 2083461824 IN IP4 192.168.1.37
s=Asterisk PBX 11.14.1
c=IN IP4 192.168.1.37
t=0 0
m=audio 11672 RTP/AVP 8 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:213.215.45.230:5060 --->
SIP/2.0 403 GW call denied
Via: SIP/2.0/UDP 192.168.1.37:5060;received=89.2.111.99;rport=55024 ;branch=z9hG4bK1f8ddc17
From: "bsmnimes" <sip:bsmnimes@192.168.1.37>;tag=as5756b27b
To: <sip:0616251476@ippi.fr>;tag=a910c8153188470b28416 23c513a131f.f481
Call-ID: 551dadc303ea8819446183030592179a@192.168.1.37:5060
CSeq: 102 INVITE
Server: OpenSIPS (1.8.2-tls (i386/linux))
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Transmitting (no NAT) to 213.215.45.230:5060:
ACK sip:0616251476@ippi.fr SIP/2.0
Via: SIP/2.0/UDP 192.168.1.37:5060;branch=z9hG4bK1f8ddc17
Max-Forwards: 70
From: "bsmnimes" <sip:bsmnimes@192.168.1.37>;tag=as5756b27b
To: <sip:0616251476@ippi.fr>;tag=a910c8153188470b28416 23c513a131f.f481
Contact: <sip:bsmnimes@192.168.1.37:5060>
Call-ID: 551dadc303ea8819446183030592179a@192.168.1.37:5060
CSeq: 102 ACK
User-Agent: FPBX-2.11.0(11.14.1)
Content-Length: 0

---
[2015-01-28 12:14:14] WARNING[2828][C-00000007]: chan_sip.c:23027 handle_response_invite: Received response: "Forbidden" from '"bsmnimes" <sip:bsmnimes@192.168.1.37>;tag=as5756b27b'
Scheduling destruction of SIP dialog '551dadc303ea8819446183030592179a@192.168.1.37:506 0' in 32000 ms (Method: INVITE)
Audio is at 11830
Adding codec 100004 (alaw) to SDP
Adding codec 100008 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 213.215.45.230:5060:
INVITE sip:ippi.fr SIP/2.0
Via: SIP/2.0/UDP 192.168.1.37:5060;branch=z9hG4bK0a8ef638
Max-Forwards: 70
From: "TECHNIQUE" <sip:bsmnimes@192.168.1.37>;tag=as68744ae8
To: <sip:ippi.fr>
Contact: <sip:bsmnimes@192.168.1.37:5060>
Call-ID: 4c3eb6796ccd8ac108ea1b810630e3dd@192.168.1.37:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(11.14.1)
Date: Wed, 28 Jan 2015 11:14:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 281

v=0
o=root 662821603 662821603 IN IP4 192.168.1.37
s=Asterisk PBX 11.14.1
c=IN IP4 192.168.1.37
t=0 0
m=audio 11830 RTP/AVP 8 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:213.215.45.230:5060 --->
SIP/2.0 404 unknown service
Via: SIP/2.0/UDP 192.168.1.37:5060;received=89.2.111.99;rport=55024 ;branch=z9hG4bK0a8ef638
From: "TECHNIQUE" <sip:bsmnimes@192.168.1.37>;tag=as68744ae8
To: <sip:ippi.fr>;tag=a910c8153188470b2841623c513a131f .3cd1
Call-ID: 4c3eb6796ccd8ac108ea1b810630e3dd@192.168.1.37:5060
CSeq: 102 INVITE
Server: OpenSIPS (1.8.2-tls (i386/linux))
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Transmitting (no NAT) to 213.215.45.230:5060:
ACK sip:ippi.fr SIP/2.0
Via: SIP/2.0/UDP 192.168.1.37:5060;branch=z9hG4bK0a8ef638
Max-Forwards: 70
From: "TECHNIQUE" <sip:bsmnimes@192.168.1.37>;tag=as68744ae8
To: <sip:ippi.fr>;tag=a910c8153188470b2841623c513a131f .3cd1
Contact: <sip:bsmnimes@192.168.1.37:5060>
Call-ID: 4c3eb6796ccd8ac108ea1b810630e3dd@192.168.1.37:5060
CSeq: 102 ACK
User-Agent: FPBX-2.11.0(11.14.1)
Content-Length: 0

---
Scheduling destruction of SIP dialog '4c3eb6796ccd8ac108ea1b810630e3dd@192.168.1.37:506 0' in 32000 ms (Method: INVITE)
Scheduling destruction of SIP dialog 'f356d01b17c88310' in 6400 ms (Method: INVITE)

<--- Reliably Transmitting (no NAT) to 192.168.1.121:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.1.121;branch=z9hG4bKf92fdabc0a6cc4a76.ad42 85990873d9d57;received=192.168.1.121
From: "1002" <sip:1002@192.168.1.37:5060>;tag=d9fef05245
To: <sip:0616251476@192.168.1.37:5060;user=phone>;tag= as6417ee0f
Call-ID: f356d01b17c88310
CSeq: 7359 INVITE
Server: FPBX-2.11.0(11.14.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

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

<--- SIP read from UDP:192.168.1.121:1036 --->
ACK sip:0616251476@192.168.1.37:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.121;branch=z9hG4bKf92fdabc0a6cc4a76.ad42 85990873d9d57
Route: <sip:192.168.1.37:5060;lr>
Max-Forwards: 70
From: "1002" <sip:1002@192.168.1.37:5060>;tag=d9fef05245
To: <sip:0616251476@192.168.1.37:5060;user=phone>;tag= as6417ee0f
Call-ID: f356d01b17c88310
CSeq: 7359 ACK
User-Agent: Aastra 6731i/3.2.2.3077
Content-Length: 0

**olisnow** · 28/01/2015, 13h25

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog 'faa82903030cd663' Method: REGISTER

<--- SIP read from UDP:192.168.1.120:3225 --->
REGISTER sip:192.168.1.37:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.120:5060;branch=z9hG4bKc0fa309f5c440d9a5
Route: <sip:192.168.1.37:5060;lr>
Max-Forwards: 70
From: "1001" <sip:1001@192.168.1.37:5060>;tag=0086b14fa3
To: "1001" <sip:1001@192.168.1.37:5060>
Call-ID: 82c6e967c0f82f06
CSeq: 43973 REGISTER
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Allow-Events: talk, hold, conference, LocalModeStatus
Authorization: Digest username="1001",realm="asterisk",nonce="19d20289", uri="sip:192.168.1.37:5060",response="1f1d761876bb 8acddb17e13f7f53824f",algorithm=MD5
Contact: "1001" <sip:1001@192.168.1.120:5060;transport=udp>;+sip.i nstance="<urn:uuid:00000000-0000-1000-8000-00085D3F11A7>"
Supported: path, gruu
User-Agent: Aastra 6731i/3.2.2.3077
Content-Length: 0

<------------->
--- (15 headers 0 lines) ---
Sending to 192.168.1.120:5060 (no NAT)
Sending to 192.168.1.120:5060 (no NAT)

<--- Transmitting (no NAT) to 192.168.1.120:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.120:5060;branch=z9hG4bKc0fa309f5c440d9a5 ;received=192.168.1.120
From: "1001" <sip:1001@192.168.1.37:5060>;tag=0086b14fa3
To: "1001" <sip:1001@192.168.1.37:5060>;tag=as67cad833
Call-ID: 82c6e967c0f82f06
CSeq: 43973 REGISTER
Server: FPBX-2.11.0(11.14.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1f695d73"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog '82c6e967c0f82f06' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:192.168.1.120:3225 --->
REGISTER sip:192.168.1.37:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.120:5060;branch=z9hG4bK8babe24be6dbfaec0
Route: <sip:192.168.1.37:5060;lr>
Max-Forwards: 70
From: "1001" <sip:1001@192.168.1.37:5060>;tag=0086b14fa3
To: "1001" <sip:1001@192.168.1.37:5060>
Call-ID: 82c6e967c0f82f06
CSeq: 43974 REGISTER
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Allow-Events: talk, hold, conference, LocalModeStatus
Authorization: Digest username="1001",realm="asterisk",nonce="1f695d73", uri="sip:192.168.1.37:5060",response="80cf13c3c40e 1f8295d81b0d79b76b2b",algorithm=MD5
Contact: "1001" <sip:1001@192.168.1.120:5060;transport=udp>;+sip.i nstance="<urn:uuid:00000000-0000-1000-8000-00085D3F11A7>"
Supported: path, gruu
User-Agent: Aastra 6731i/3.2.2.3077
Content-Length: 0

<------------->
--- (15 headers 0 lines) ---
Sending to 192.168.1.120:5060 (no NAT)
Reliably Transmitting (no NAT) to 192.168.1.120:5060:
OPTIONS sip:1001@192.168.1.120:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.37:5060;branch=z9hG4bK4c39b038
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.1.37>;tag=as43139ce5
To: <sip:1001@192.168.1.120:5060;transport=udp>
Contact: <sip:Unknown@192.168.1.37:5060>
Call-ID: 394609a3490804d8623a39245408239b@192.168.1.37:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.11.0(11.14.1)
Date: Wed, 28 Jan 2015 11:14:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

---

<--- Transmitting (no NAT) to 192.168.1.120:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.120:5060;branch=z9hG4bK8babe24be6dbfaec0 ;received=192.168.1.120
From: "1001" <sip:1001@192.168.1.37:5060>;tag=0086b14fa3
To: "1001" <sip:1001@192.168.1.37:5060>;tag=as67cad833
Call-ID: 82c6e967c0f82f06
CSeq: 43974 REGISTER
Server: FPBX-2.11.0(11.14.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 120
Contact: <sip:1001@192.168.1.120:5060;transport=udp>;expire s=120
Date: Wed, 28 Jan 2015 11:14:20 GMT
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog '4f5417b739d97bdb12893c8a72311750@192.168.1.37:506 0' in 6400 ms (Method: NOTIFY)
Reliably Transmitting (no NAT) to 192.168.1.120:5060:
NOTIFY sip:1001@192.168.1.120:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.37:5060;branch=z9hG4bK4d50b6d6
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.1.37>;tag=as2a002cf2
To: <sip:1001@192.168.1.120:5060;transport=udp>
Contact: <sip:Unknown@192.168.1.37:5060>
Call-ID: 4f5417b739d97bdb12893c8a72311750@192.168.1.37:5060
CSeq: 102 NOTIFY
User-Agent: FPBX-2.11.0(11.14.1)
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 87

Messages-Waiting: no
Message-Account: sip:*97@192.168.1.37
Voice-Message: 0/0 (0/0)

---
Scheduling destruction of SIP dialog '82c6e967c0f82f06' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:192.168.1.120:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.37:5060;branch=z9hG4bK4c39b038
From: "Unknown" <sip:Unknown@192.168.1.37>;tag=as43139ce5
To: <sip:1001@192.168.1.120:5060;transport=udp>;tag=65 5704210
Call-ID: 394609a3490804d8623a39245408239b@192.168.1.37:5060
CSeq: 102 OPTIONS
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Server: Aastra 6731i/3.2.2.3077
Supported: path
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '394609a3490804d8623a39245408239b@192.168.1.37:506 0' Method: OPTIONS

<--- SIP read from UDP:192.168.1.120:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.37:5060;branch=z9hG4bK4d50b6d6
From: "Unknown" <sip:Unknown@192.168.1.37>;tag=as2a002cf2
To: <sip:1001@192.168.1.120:5060;transport=udp>;tag=33 65219826
Call-ID: 4f5417b739d97bdb12893c8a72311750@192.168.1.37:5060
CSeq: 102 NOTIFY
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Allow-Events: talk, hold, conference, LocalModeStatus
Contact: "1001" <sip:1001@192.168.1.120:5060;transport=udp>;+sip.i nstance="<urn:uuid:00000000-0000-1000-8000-00085D3F11A7>"
Server: Aastra 6731i/3.2.2.3077
Supported: path
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Really destroying SIP dialog '4f5417b739d97bdb12893c8a72311750@192.168.1.37:506 0' Method: NOTIFY
Really destroying SIP dialog 'f356d01b17c88310' Method: ACK

**tanguyd** · 28/01/2015, 15h27

Route sortante:
j ai mis juste dans thunk sequence for matched routes: 0--> mon trunk sip
et optional destination on congestion --> trunks --> mon trunk sip

OK

Route sortante:
N] sda => 334.... mon tel
prefixe au cid -->sda1
choix destination --> trunks --> mon sip

C'est pas plutôt route entrante ? Si oui pourquoi as tu mis le trunk comme destination, la destination devrait plutôt être l'extension interne.

Je ne comprends pas ce Received response: "Forbidden" from '"bsmnimes" <sip:bsmnimes@192.168.1.37>. Cette IP, c'est bien l'IP locale du serveur ?

Sinon, même si un trace SIP est utile pour une analyse fine, les logs de l'asterisk en lui même en sont général plus faciles a lire, je pense qu'il faut mieux commencer par regarder ces logs, avant de se plonger dans les traces SIP.

**olisnow** · 28/01/2015, 15h33

alors j ai mis en route entrante
vers l extension de mon tel 1002

tel 1002 ip 1.121
serveur ip 1.37

**olisnow** · 28/01/2015, 15h46

quand j appelle de mon portable, vers mon sip
j ai trouver une trace

[2015-01-28 14:44:48] NOTICE[2005][C-00000002]: chan_sip.c:25545 handle_request_invite: Call from 'ippi-in' (213.215.45.230:5060) to extension 's' rejected because extension not found in context 'from_ippi'

pour moi extension from_ippi, c'est ici, et la configuration viens du site ippi
[ippi_incoming]
; configuration des appels entrants depuis ippi
type=peer
host=ippi.fr
context=from_ippi
nat=yes
canreinvite=no

**olisnow** · 28/01/2015, 16h04

quand j'appelle

voici les logs,

[2015-01-28 15:03:55] WARNING[4474]: res_pjsip_pubsub.c:2858 pubsub_on_rx_publish_request: No registered publish handler for event presence
[2015-01-28 15:03:55] WARNING[4474]: res_pjsip_pubsub.c:608 subscription_get_handler_from_rdata: No registered subscribe handler for event presence.winfo
[2015-01-28 15:04:03] WARNING[4910][C-00000007]: channel.c:4816 ast_prod: Prodding channel 'PJSIP/1000-00000002' failed
[2015-01-28 15:04:03] WARNING[1871]: res_pjsip_pubsub.c:2858 pubsub_on_rx_publish_request: No registered publish handler for event presence
[2015-01-28 15:04:03] WARNING[1871]: res_pjsip_pubsub.c:608 subscription_get_handler_from_rdata: No registered subscribe handler for event presence.winfo

**tanguyd** · 28/01/2015, 16h17

Ok, as tu édité cette conf avec l'interface web ou en dur dans les fichiers de conf comme c'est dit sur le site de ippi ?

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