Déclarer des bornes DECT IP Aastra

[Kriss]

Tu peux essayer un "sip set debug IP.DE.TA.BORNEMAITRE"
Avant de rebooter la dite borne maître, et regardé si ça s'agite un peu dans ta CLI (rechercher des réponse Forbiden, au bad Auth, on sait jamais ...
Tu peux aussi faire tourner un tcpdump sur ton rasp pour voir ce que tu reçois de la part de ta borne.

./Kriss

[Kavoli]

Donc j'ai essayé "sip set debug ip 192.168.65.50" qui est l'ip de ma borne maîtresse. Alors j'ai un peu regardé j'ai un forbidden:

<--- Transmitting (no NAT) to 192.168.65.50:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.65.50;branch=z9hG4bK-1fbd01558fec;received=192.168.65.50
From: "ABO 201" <sip:201@192.168.65.1>;tag=f-ea604ffd045a2
To: "ABO 201" <sip:201@192.168.65.1>;tag=as533c2c94
Call-ID: 62474b6dea60384313942@DSIP
CSeq: 2 REGISTER
Server: FPBX2.11.0beta3(11.3.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

Alors à ce niveau là à part comprendre qu'il y ait une interdiction à un niveau je ne sait pas trop où ça bloque .

je vais tester tcpdumb comme tu m'as conseil

[Kavoli]

Alors j'ai réussi à ne plus avoir de forbidden, les appels se passent dans un sens (d'un poste sip à un dect) mais sans voix et le dect décroche de suite.
Dans l'autre sens (dect à poste sip), rien et j'ai un 415 Unsupported Media Type avec sip set debug:

<--- SIP read from UDP:192.168.65.50:5060 --->
MESSAGE sip:system@192.168.65.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.65.50;branch=z9hG4bK-753003f6e729f
From: "ABO 201" <sip:201@192.168.65.1>;tag=f-753003d48b61a26
To: <sip:system@192.168.65.1>
Call-ID: 17a4d05c75300384315d5e@DSIP
CSeq: 373638 MESSAGE
Max-Forwards: 70
Supported: replaces, 100rel
User-Agent: DSIP
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, REFER, NOTIFY, INFO, MESSAGE, UPDATE, PRACK
Content-Type: application/x-amt-dect-stimuli
Accept: application/sdp,application/x-amt-dect-stimuli
Content-Length: 6

2C0160
<------------->
--- (13 headers 1 lines) ---
Receiving message!

<--- Transmitting (no NAT) to 192.168.65.50:5060 --->
SIP/2.0 415 Unsupported Media Type
Via: SIP/2.0/UDP 192.168.65.50;branch=z9hG4bK-753003f6e729f;received=192.168.65.50
From: "ABO 201" <sip:201@192.168.65.1>;tag=f-753003d48b61a26
To: <sip:system@192.168.65.1>;tag=as0ac1ee61
Call-ID: 17a4d05c75300384315d5e@DSIP
CSeq: 373638 MESSAGE
Server: FPBX2.11.0beta3(11.3.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

Je vais essayer de chercher quel est le problème.

[Kriss]

Pour le forbiden, tu devais avoir un souci de mot de passe de ton extension.
Sip tu fais un "sip show peers", tes extensions correspondant a tes DECT sont bien OK ?

Pour le nouveau problème que tu rencontres, c'est visiblement un problème de codecs non suportés. Regarde au niveau des allow / disallow que tu as configuré sur tes extensions.
Comme c'est du LAN, je te conseil d'utilise du alaw des deux cotés (Borne + Asterisk).

./Kriss

[Kavoli]

Pour le forbidden c'est bon j'ai réussi à régler le soucis. Maintenant avec un "sip show peers" mes DECT apparaissent et sont bien enregistrés.

Par contre j'ai toujours ce problème de "Unsupported Media Type" malgré être passé en "alaw" partout. Et en plus je me retrouve avec un "404 not found" lors d'un "sip set debug ip <ip-borne>:
<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '68b6799a09ec529d22aba8516018e018@192.168.65.1:506 0' Method: OPTIONS
Reliably Transmitting (no NAT) to 192.168.65.50:5060:
OPTIONS sip:192.168.65.50 SIP/2.0
Via: SIP/2.0/UDP 192.168.65.1:5060;branch=z9hG4bK73e6d2af
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.65.1>;tag=as0bfb7787
To: <sip:192.168.65.50>
Contact: <sip:Unknown@192.168.65.1:5060>
Call-ID: 42847ccb2fd24e426d2e17597eb46a5e@192.168.65.1:5060
CSeq: 102 OPTIONS
User-Agent: FPBX2.11.0beta3(11.3.0)
Date: Mon, 27 May 2013 13:33:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:192.168.65.50:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.65.1:5060;branch=z9hG4bK73e6d2af
From: "Unknown" <sip:Unknown@192.168.65.1>;tag=as0bfb7787
To: <sip:192.168.65.50>;tag=t-1bec701796083eaa
Call-ID: 42847ccb2fd24e426d2e17597eb46a5e@192.168.65.1:5060
CSeq: 102 OPTIONS
Server: DSIP
Supported: replaces, 100rel
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, REFER, NOTIFY, INFO, MESSAGE, UPDATE, PRACK
Accept: application/sdp, application/sdp,application/x-amt-dect-stimuli
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '42847ccb2fd24e426d2e17597eb46a5e@192.168.65.1:506 0' Method: OPTIONS


En revanche nouveau soucis, mais cette fois ci au niveau des postes SIP. Ils sont maintenant en mode "no service" et leurs extensions ne sont pas enregistrés. J'ai reseté les postes, cassé recréé les extensions mais ça reste toujours pareil. J'ai un "Unauthorized" quand je fais un "sip set debug ip <ip-d'un-poste>".
EDIT: problème "no service" des postes SIP résolu. Mes dect et postes sip apparaissent dans sip show peers mais toujours pas d'appel possible au niveau dect dans aucun sens.

[Kriss]

Peux-tu montrer ton sip.conf ?

[Kavoli]

Oui mais je configure essentiellement les poste avec FreePBX.
voilà mon sip.conf:

[general]

; These files will all be included in the [general] context
;
#include sip_general_additional.conf

;sip_general_custom.conf is the proper file location for placing any sip general
;options that you might need set. For example: enable and force the sip jitterbuffer.
;If these settings are desired they should be set the sip_general_custom.conf file.
;
; jbenable=yes
; jbforce=yes
;
;It is also the proper place to add the lines needed for sip nat'ing when going
;through a firewall. For nat'ing you'd need to add the following lines:
; nat=yes , externip= , localhost= , and optionally fromdomain= .
;
#include sip_general_custom.conf

;sip_nat.conf is here for legacy support reasons and for those that upgrade
;from previous versions. If you have this file with lines in it please make
;sure they are not duplicated in sip_general_custom.conf, if so remove them
;from sip_nat.conf as sip_general_custom.conf will have precedence.
#include sip_nat.conf

;sip_registrations_custom.conf is for any customizations you might need to do to
;the automatically generated registrations that FreePBX makes.
;
#include sip_registrations_custom.conf
#include sip_registrations.conf

; These files should all be expected to come after the [general] context
;


;sip_general_custom.conf is the proper file location for placing any sip general
;options that you might need set. For example: enable and force the sip jitterbuffer.
;If these settings are desired they should be set the sip_general_custom.conf file.
;
; jbenable=yes
; jbforce=yes
;
;It is also the proper place to add the lines needed for sip nat'ing when going
;through a firewall. For nat'ing you'd need to add the following lines:
; nat=yes , externip= , localhost= , and optionally fromdomain= .
;
#include sip_general_custom.conf

;sip_nat.conf is here for legacy support reasons and for those that upgrade
;from previous versions. If you have this file with lines in it please make
;sure they are not duplicated in sip_general_custom.conf, if so remove them
;from sip_nat.conf as sip_general_custom.conf will have precedence.
#include sip_nat.conf

;sip_registrations_custom.conf is for any customizations you might need to do to
;the automatically generated registrations that FreePBX makes.
;
#include sip_registrations_custom.conf
#include sip_registrations.conf

; These files should all be expected to come after the [general] context
;
#include sip_custom.conf
#include sip_additional.conf

;sip_custom_post.conf If you have extra parameters that are needed for a
;extension to work to for example, those go here. So you have extension
;1000 defined in your system you start by creating a line [1000](+) in this
;file. Then on the next line add the extra parameter that is needed.
;When the sip.conf is loaded it will append your additions to the end of
;that extension.
;
#include sip_custom_post.conf