Premiers pas avec Asterisk, besoin d'aide ! :-)

**Miskia** · 26/09/2013, 11h34

Bonjour,

Pour commencer, je voudrais pouvoir émettre un appel depuis un softphone, en passant par un provider SIP.
J'ai donc choisi Nomado, j'ai créé mon trunk SIP et une extension. J'ai créé également une outbound route.

Je parviens à me connecter depuis le softphone.

En faisant un "show sip registry" je vois que c'est Registred et en faisant "show sip peers" je vois également que c'est Registered.

Quand j'essaie d'émettre mon appel, celui-ci est "Forbidden", je vois bien cette erreur dans le log cependant je ne sais pas d'où elle provient.
De plus, dans le log, je vois bien que mon appel tente de se faire, par contre je comprends mal, j'ai l'impression que l'appel ne trouve pas la route et donc n'arrive pas sur le sip trunk.

Pouvez-vous m'aider?

Voici le log complet (avec le show sip registry et le show sip peers), ensuite j'essaie de passer 1 ou 2 coups de fil.

Cordialement,
Miskia.

Code:

bureau*CLI> sip set debug on
SIP Debugging re-enabled
Really destroying SIP dialog '5f0837b45d4217eb3c5dc9ba5d519cc1@127.0.0.1' Method: REGISTER
bureau*CLI> sip show registry
Host                                    dnsmgr Username       Refresh State                Reg.Time                 
sip1.nomado.eu:5060                     N      0899367            105 Registered           Thu, 26 Sep 2013 11:20:36
1 SIP registrations.

<--- SIP read from UDP:192.168.1.2:51262 --->

<------------->
Really destroying SIP dialog 'pTOrlBOc4L8UkuZKxUSnBF954FF1ZhIt' Method: REGISTER
bureau*CLI> sip show peers
Name/username             Host                                    Dyn Forcerport ACL Port     Status      Description                      
10/10                     192.168.1.2                              D                 51262    OK (11 ms)                                   
Nomado/0899367            178.32.41.64                                 N             5060     Unmonitored                                  
2 sip peers [Monitored: 1 online, 0 offline Unmonitored: 1 online, 0 offline]

<--- SIP read from UDP:192.168.1.2:51262 --->

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

<--- SIP read from UDP:192.168.1.2:51262 --->
INVITE sip:+32499411198@192.168.1.251 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:51262;rport;branch=z9hG4bKPjmbTxLPZgS01Hkhnco1Fv0f9K0Z0lQIn3
Max-Forwards: 70
From: "Arnaud CRESP" <sip:10@192.168.1.251>;tag=zwWWgMiUNk6bC.VXlxXODS5JSVFzY.gO
To: <sip:+32499411198@192.168.1.251>
Contact: "Arnaud CRESP" <sip:10@192.168.1.2:51262;ob>
Call-ID: I.FrcyZahOCWNvXfgDP89EFLqNfsaEuT
CSeq: 17215 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: Telephone 1.0.4
Content-Type: application/sdp
Content-Length: 456

v=0
o=- 3589176084 3589176084 IN IP4 192.168.1.2
s=pjmedia
c=IN IP4 192.168.1.2
t=0 0
a=X-nat:0
m=audio 4002 RTP/AVP 103 102 104 109 3 0 8 9 101
a=rtcp:4003 IN IP4 192.168.1.2
a=rtpmap:103 speex/16000
a=rtpmap:102 speex/8000
a=rtpmap:104 speex/32000
a=rtpmap:109 iLBC/8000
a=fmtp:109 mode=30
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (15 headers 20 lines) ---
Sending to 192.168.1.2:51262 (no NAT)
Sending to 192.168.1.2:51262 (no NAT)
Using INVITE request as basis request - I.FrcyZahOCWNvXfgDP89EFLqNfsaEuT
Found peer '10' for '10' from 192.168.1.2:51262

<--- Reliably Transmitting (no NAT) to 192.168.1.2:51262 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.2:51262;branch=z9hG4bKPjmbTxLPZgS01Hkhnco1Fv0f9K0Z0lQIn3;received=192.168.1.2;rport=51262
From: "Arnaud CRESP" <sip:10@192.168.1.251>;tag=zwWWgMiUNk6bC.VXlxXODS5JSVFzY.gO
To: <sip:+32499411198@192.168.1.251>;tag=as434ab967
Call-ID: I.FrcyZahOCWNvXfgDP89EFLqNfsaEuT
CSeq: 17215 INVITE
Server: FPBX-2.8.1(11.5.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="53cc13fe"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'I.FrcyZahOCWNvXfgDP89EFLqNfsaEuT' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:192.168.1.2:51262 --->
ACK sip:+32499411198@192.168.1.251 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:51262;rport;branch=z9hG4bKPjmbTxLPZgS01Hkhnco1Fv0f9K0Z0lQIn3
Max-Forwards: 70
From: "Arnaud CRESP" <sip:10@192.168.1.251>;tag=zwWWgMiUNk6bC.VXlxXODS5JSVFzY.gO
To: <sip:+32499411198@192.168.1.251>;tag=as434ab967
Call-ID: I.FrcyZahOCWNvXfgDP89EFLqNfsaEuT
CSeq: 17215 ACK
Content-Length: 0

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

<--- SIP read from UDP:192.168.1.2:51262 --->
INVITE sip:+32499411198@192.168.1.251 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:51262;rport;branch=z9hG4bKPjZHDpy5-PDgc7GyaQpu.BMnd0ugTptKg9
Max-Forwards: 70
From: "Arnaud CRESP" <sip:10@192.168.1.251>;tag=zwWWgMiUNk6bC.VXlxXODS5JSVFzY.gO
To: <sip:+32499411198@192.168.1.251>
Contact: "Arnaud CRESP" <sip:10@192.168.1.2:51262;ob>
Call-ID: I.FrcyZahOCWNvXfgDP89EFLqNfsaEuT
CSeq: 17216 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: Telephone 1.0.4
Authorization: Digest username="10", realm="asterisk", nonce="53cc13fe", uri="sip:+32499411198@192.168.1.251", response="6490a733de77a33f23de1cc571aa18c9", algorithm=MD5
Content-Type: application/sdp
Content-Length: 456

v=0
o=- 3589176084 3589176084 IN IP4 192.168.1.2
s=pjmedia
c=IN IP4 192.168.1.2
t=0 0
a=X-nat:0
m=audio 4002 RTP/AVP 103 102 104 109 3 0 8 9 101
a=rtcp:4003 IN IP4 192.168.1.2
a=rtpmap:103 speex/16000
a=rtpmap:102 speex/8000
a=rtpmap:104 speex/32000
a=rtpmap:109 iLBC/8000
a=fmtp:109 mode=30
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (16 headers 20 lines) ---
Sending to 192.168.1.2:51262 (no NAT)
Using INVITE request as basis request - I.FrcyZahOCWNvXfgDP89EFLqNfsaEuT
Found peer '10' for '10' from 192.168.1.2:51262

<--- Reliably Transmitting (no NAT) to 192.168.1.2:51262 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.1.2:51262;branch=z9hG4bKPjZHDpy5-PDgc7GyaQpu.BMnd0ugTptKg9;received=192.168.1.2;rport=51262
From: "Arnaud CRESP" <sip:10@192.168.1.251>;tag=zwWWgMiUNk6bC.VXlxXODS5JSVFzY.gO
To: <sip:+32499411198@192.168.1.251>;tag=as434ab967
Call-ID: I.FrcyZahOCWNvXfgDP89EFLqNfsaEuT
CSeq: 17216 INVITE
Server: FPBX-2.8.1(11.5.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'I.FrcyZahOCWNvXfgDP89EFLqNfsaEuT' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:192.168.1.2:51262 --->
ACK sip:+32499411198@192.168.1.251 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:51262;rport;branch=z9hG4bKPjZHDpy5-PDgc7GyaQpu.BMnd0ugTptKg9
Max-Forwards: 70
From: "Arnaud CRESP" <sip:10@192.168.1.251>;tag=zwWWgMiUNk6bC.VXlxXODS5JSVFzY.gO
To: <sip:+32499411198@192.168.1.251>;tag=as434ab967
Call-ID: I.FrcyZahOCWNvXfgDP89EFLqNfsaEuT
CSeq: 17216 ACK
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog 'I.FrcyZahOCWNvXfgDP89EFLqNfsaEuT' Method: ACK

<--- SIP read from UDP:192.168.1.2:51262 --->

<------------->
Reliably Transmitting (no NAT) to 192.168.1.2:51262:
OPTIONS sip:10@192.168.1.2:51262;ob SIP/2.0
Via: SIP/2.0/UDP 192.168.1.251:5060;branch=z9hG4bK06773e61
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.1.251>;tag=as6b38c50f
To: <sip:10@192.168.1.2:51262;ob>
Contact: <sip:Unknown@192.168.1.251:5060>
Call-ID: 408e9a2a6b45973e70aecfe128dfec14@192.168.1.251:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.8.1(11.5.0)
Date: Thu, 26 Sep 2013 09:21:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:192.168.1.2:51262 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.251:5060;received=192.168.1.251;branch=z9hG4bK06773e61
Call-ID: 408e9a2a6b45973e70aecfe128dfec14@192.168.1.251:5060
From: "Unknown" <sip:Unknown@192.168.1.251>;tag=as6b38c50f
To: <sip:10@192.168.1.2;ob>;tag=z9hG4bK06773e61
CSeq: 102 OPTIONS
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain
Supported: replaces, 100rel, timer, norefersub
Allow-Events: presence, message-summary, refer
User-Agent: Telephone 1.0.4
Content-Type: application/sdp
Content-Length: 445

v=0
o=- 3589176099 3589176099 IN IP4 192.168.1.2
s=pjmedia
c=IN IP4 192.168.1.2
t=0 0
m=audio 4000 RTP/AVP 103 102 104 109 3 0 8 9 101
a=rtcp:4001 IN IP4 192.168.1.2
a=rtpmap:103 speex/16000
a=rtpmap:102 speex/8000
a=rtpmap:104 speex/32000
a=rtpmap:109 iLBC/8000
a=fmtp:109 mode=30
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (13 headers 19 lines) ---
Really destroying SIP dialog '408e9a2a6b45973e70aecfe128dfec14@192.168.1.251:5060' Method: OPTIONS

<--- SIP read from UDP:192.168.1.2:51262 --->

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

<--- SIP rea

**yold** · 26/09/2013, 15h14

Comment as-tu déclaré ton trunk dans le sip.conf ?

**jean** · 26/09/2013, 15h39

je pense que la numérotation présentée n'est pas bonne. tu envoies +32499xxxxxxxx - je pense que nomado accepte soit au format e164 (snas le +) ou numérotation nationale (ajouter 00)

**Miskia** · 26/09/2013, 15h43

Je viens d'essayer avec 32499411198, pareil.
Par contre je n'ai rien configuré directement dans les fichiers de config, j'utilise l'interface web de Elastix pour se faire.
Je viens de vérifier le sip.conf, visiblement il n'y a rien d'autre que des choses commentées (des include)

**jean** · 26/09/2013, 16h27

essaie avec 0032
et essaie 444 c'est le no de test

**Miskia** · 26/09/2013, 16h40

Justement, j'allais éditer mon post, j'ai essayé également le 444 et le 0032.

Cordialement

**jean** · 26/09/2013, 16h55

peux tu faire un ngrep port 5060 and host <ip de nomado>

**Miskia** · 26/09/2013, 21h49

Voilà ce que donne le ngrep

Code:

interface: eth0 (192.168.1.0/255.255.255.0)
filter: (ip or ip6) and ( port 5060 )
#
U 192.168.1.2:57693 -> 192.168.1.250:5060
  ..................                                                                                                      
#
U 192.168.1.2:57693 -> 192.168.1.250:5060
  ..................                                                                                                          
#
U 192.168.1.250:5060 -> 192.168.1.2:57693
  OPTIONS sip:10@192.168.1.2:57693;ob SIP/2.0..Via: SIP/2.0/UDP 192.168.1.250:5060;branch=z9hG4bK42724299..Max-Forwards: 70..F
  rom: "Unknown" <sip:Unknown@192.168.1.250>;tag=as182a97f1..To: <sip:10@192.168.1.2:57693;ob>..Contact: <sip:Unknown@192.168.
  1.250:5060>..Call-ID: 790bf7fc30524e4c213f097e58316404@192.168.1.250:5060..CSeq: 102 OPTIONS..User-Agent: FPBX-2.11.0(11.5.1
  )..Date: Thu, 26 Sep 2013 19:28:07 GMT..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH..S
  upported: replaces, timer..Content-Length: 0....                                                                            
#
U 192.168.1.2:57693 -> 192.168.1.250:5060
  SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.1.250:5060;received=192.168.1.250;branch=z9hG4bK42724299..Call-ID: 790bf7fc30524e4c
  213f097e58316404@192.168.1.250:5060..From: "Unknown" <sip:Unknown@192.168.1.250>;tag=as182a97f1..To: <sip:10@192.168.1.2;ob>
  ;tag=z9hG4bK42724299..CSeq: 102 OPTIONS..Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, 
  OPTIONS..Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/simple-message-summary, message/s
  ipfrag;version=2.0, application/im-iscomposing+xml, text/plain..Supported: replaces, 100rel, timer, norefersub..Allow-Events
  : presence, message-summary, refer..User-Agent: Telephone 1.0.4..Content-Type: application/sdp..Content-Length:   445....v=0
  ..o=- 3589212487 3589212487 IN IP4 192.168.1.2..s=pjmedia..c=IN IP4 192.168.1.2..t=0 0..m=audio 4000 RTP/AVP 103 102 104 109
   3 0 8 9 101..a=rtcp:4001 IN IP4 192.168.1.2..a=rtpmap:103 speex/16000..a=rtpmap:102 speex/8000..a=rtpmap:104 speex/32000..a
  =rtpmap:109 iLBC/8000..a=fmtp:109 mode=30..a=rtpmap:3 GSM/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:9 G722/
  8000..a=sendrecv..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..                                                      
#
U 192.168.1.250:5060 -> 178.32.41.64:5060
  REGISTER sip:sip1.nomado.eu SIP/2.0..Via: SIP/2.0/UDP 81.241.118.168:5060;branch=z9hG4bK448085de;rport..Max-Forwards: 70..Fr
  om: <sip:0899367@sip1.nomado.eu>;tag=as21e6ca92..To: <sip:0899367@sip1.nomado.eu>..Call-ID: 218c615a5365a009392c07dd19b4833a
  @127.0.0.1..CSeq: 110 REGISTER..User-Agent: FPBX-2.11.0(11.5.1)..Authorization: Digest username="0899367", realm="sip1.nomad
  o.eu", algorithm=MD5, uri="sip:sip1.nomado.eu", nonce="5244889000004e90a73af7c44eddc0b715c602c11d02ac5e", response="93648afe
  81f5ef79eaf1edc58c8944d9"..Expires: 120..Contact: <sip:s@81.241.118.168:5060>..Content-Length: 0....                        
#
U 178.32.41.64:5060 -> 192.168.1.250:5060
  SIP/2.0 401 Unauthorized..Via: SIP/2.0/UDP 81.241.118.168:5060;branch=z9hG4bK448085de;rport=5124..From: <sip:0899367@sip1.no
  mado.eu>;tag=as21e6ca92..To: <sip:0899367@sip1.nomado.eu>;tag=490696bd9692aac9a37851d43aaa8e6e.e813..Call-ID: 218c615a5365a0
  09392c07dd19b4833a@127.0.0.1..CSeq: 110 REGISTER..WWW-Authenticate: Digest realm="sip1.nomado.eu", nonce="524488f900005317d5
  18d0a1408bf9b319653bf92f6eb048", stale=true..Server: Enswitch SIP proxy..Content-Length: 0..Warning: 392 178.32.41.64:5060 "
  Noisy feedback tells:  pid=13202 req_src_ip=81.241.118.168 req_src_port=5124 in_uri=sip:sip1.nomado.eu out_uri=sip:sip1.noma
  do.eu via_cnt==1"....

IP de Nomado c'est 178.32.41.64

**Miskia** · 26/09/2013, 21h50

Par contre je viens de changer de "distrib" (je suis sur Raspberry) et j'ai pris RaspPBX. J'ai reconfiguré, de la même manière, et maintenant cela réagit différement, sur mon client softphone je vois "Calling..." pendant qq secondes avant d'avoir un "Server internal failure", le log est également plus parlant:

Code:

raspbx*CLI> sip set debug on
SIP Debugging re-enabled

<--- SIP read from UDP:192.168.1.2:57693 --->
INVITE sip:32499411198@192.168.1.250 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:57693;rport;branch=z9hG4bKPjU7rEcXJHWT-WqCDZscnxC8MYKHUZu2Uq
Max-Forwards: 70
From: "Arnaud CRESP" <sip:10@192.168.1.250>;tag=APF0hcfO8eJaYihQBbOHwG8oCxoRuLyY
To: <sip:32499411198@192.168.1.250>
Contact: "Arnaud CRESP" <sip:10@192.168.1.2:57693;ob>
Call-ID: YyTmjEUeIiijiDpVYirRp2CqrNgeJuOa
CSeq: 5716 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: Telephone 1.0.4
Content-Type: application/sdp
Content-Length: 456

v=0
o=- 3589213431 3589213431 IN IP4 192.168.1.2
s=pjmedia
c=IN IP4 192.168.1.2
t=0 0
a=X-nat:0
m=audio 4008 RTP/AVP 103 102 104 109 3 0 8 9 101
a=rtcp:4009 IN IP4 192.168.1.2
a=rtpmap:103 speex/16000
a=rtpmap:102 speex/8000
a=rtpmap:104 speex/32000
a=rtpmap:109 iLBC/8000
a=fmtp:109 mode=30
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (15 headers 20 lines) ---
Sending to 192.168.1.2:57693 (NAT)
Sending to 192.168.1.2:57693 (NAT)
Using INVITE request as basis request - YyTmjEUeIiijiDpVYirRp2CqrNgeJuOa
Found peer '10' for '10' from 192.168.1.2:57693

<--- Reliably Transmitting (no NAT) to 192.168.1.2:57693 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.2:57693;branch=z9hG4bKPjU7rEcXJHWT-WqCDZscnxC8MYKHUZu2Uq;received=192.168.1.2;rport=57693
From: "Arnaud CRESP" <sip:10@192.168.1.250>;tag=APF0hcfO8eJaYihQBbOHwG8oCxoRuLyY
To: <sip:32499411198@192.168.1.250>;tag=as70387093
Call-ID: YyTmjEUeIiijiDpVYirRp2CqrNgeJuOa
CSeq: 5716 INVITE
Server: FPBX-2.11.0(11.5.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3593fdaf"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'YyTmjEUeIiijiDpVYirRp2CqrNgeJuOa' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:192.168.1.2:57693 --->
ACK sip:32499411198@192.168.1.250 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:57693;rport;branch=z9hG4bKPjU7rEcXJHWT-WqCDZscnxC8MYKHUZu2Uq
Max-Forwards: 70
From: "Arnaud CRESP" <sip:10@192.168.1.250>;tag=APF0hcfO8eJaYihQBbOHwG8oCxoRuLyY
To: <sip:32499411198@192.168.1.250>;tag=as70387093
Call-ID: YyTmjEUeIiijiDpVYirRp2CqrNgeJuOa
CSeq: 5716 ACK
Content-Length: 0

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

<--- SIP read from UDP:192.168.1.2:57693 --->
INVITE sip:32499411198@192.168.1.250 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:57693;rport;branch=z9hG4bKPjn9UF-mxGYJ1do9tU.hoTM7UCw7n3gQ.2
Max-Forwards: 70
From: "Arnaud CRESP" <sip:10@192.168.1.250>;tag=APF0hcfO8eJaYihQBbOHwG8oCxoRuLyY
To: <sip:32499411198@192.168.1.250>
Contact: "Arnaud CRESP" <sip:10@192.168.1.2:57693;ob>
Call-ID: YyTmjEUeIiijiDpVYirRp2CqrNgeJuOa
CSeq: 5717 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: Telephone 1.0.4
Authorization: Digest username="10", realm="asterisk", nonce="3593fdaf", uri="sip:32499411198@192.168.1.250", response="acb03f0fd2589ba0719ed8a80c4afe5b", algorithm=MD5
Content-Type: application/sdp
Content-Length: 456

v=0
o=- 3589213431 3589213431 IN IP4 192.168.1.2
s=pjmedia
c=IN IP4 192.168.1.2
t=0 0
a=X-nat:0
m=audio 4008 RTP/AVP 103 102 104 109 3 0 8 9 101
a=rtcp:4009 IN IP4 192.168.1.2
a=rtpmap:103 speex/16000
a=rtpmap:102 speex/8000
a=rtpmap:104 speex/32000
a=rtpmap:109 iLBC/8000
a=fmtp:109 mode=30
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (16 headers 20 lines) ---
Sending to 192.168.1.2:57693 (no NAT)
Using INVITE request as basis request - YyTmjEUeIiijiDpVYirRp2CqrNgeJuOa
Found peer '10' for '10' from 192.168.1.2:57693
Found RTP audio format 103
Found RTP audio format 102
Found RTP audio format 104
Found RTP audio format 109
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 9
Found RTP audio format 101
Found audio description format speex for ID 103
Found audio description format speex for ID 102
Found audio description format speex for ID 104
Found audio description format iLBC for ID 109
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G722 for ID 9
Found audio description format telephone-event for ID 101
Capabilities: us - (g723|gsm|ulaw|alaw|g726|g729|g722), peer - audio=(gsm|ulaw|alaw|speex|speex16|ilbc|g722|speex32)/video=(nothing)/text=(nothing), combined - (gsm|ulaw|alaw|g722)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.2:4008
Looking for 32499411198 in from-internal (domain 192.168.1.250)
list_route: hop: <sip:10@192.168.1.2:57693;ob>

<--- Transmitting (no NAT) to 192.168.1.2:57693 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.2:57693;branch=z9hG4bKPjn9UF-mxGYJ1do9tU.hoTM7UCw7n3gQ.2;received=192.168.1.2;rport=57693
From: "Arnaud CRESP" <sip:10@192.168.1.250>;tag=APF0hcfO8eJaYihQBbOHwG8oCxoRuLyY
To: <sip:32499411198@192.168.1.250>
Call-ID: YyTmjEUeIiijiDpVYirRp2CqrNgeJuOa
CSeq: 5717 INVITE
Server: FPBX-2.11.0(11.5.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:32499411198@192.168.1.250:5060>
Content-Length: 0


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

<--- SIP read from UDP:192.168.1.2:57693 --->

<------------->
Reliably Transmitting (no NAT) to 192.168.1.2:57693:
OPTIONS sip:10@192.168.1.2:57693;ob SIP/2.0
Via: SIP/2.0/UDP 192.168.1.250:5060;branch=z9hG4bK2b289f5e
Max-Forwards: 70
From: "Unknown" <sip:Unknown@192.168.1.250>;tag=as7c391998
To: <sip:10@192.168.1.2:57693;ob>
Contact: <sip:Unknown@192.168.1.250:5060>
Call-ID: 3267783720f8341c67e0382c0dc75c25@192.168.1.250:5060
CSeq: 102 OPTIONS
User-Agent: FPBX-2.11.0(11.5.1)
Date: Thu, 26 Sep 2013 19:43:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:192.168.1.2:57693 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.250:5060;received=192.168.1.250;branch=z9hG4bK2b289f5e
Call-ID: 3267783720f8341c67e0382c0dc75c25@192.168.1.250:5060
From: "Unknown" <sip:Unknown@192.168.1.250>;tag=as7c391998
To: <sip:10@192.168.1.2;ob>;tag=z9hG4bK2b289f5e
CSeq: 102 OPTIONS
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Accept: application/sdp, application/pidf+xml, application/xpidf+xml, application/simple-message-summary, message/sipfrag;version=2.0, application/im-iscomposing+xml, text/plain
Supported: replaces, 100rel, timer, norefersub
Allow-Events: presence, message-summary, refer
User-Agent: Telephone 1.0.4
Content-Type: application/sdp
Content-Length: 445

v=0
o=- 3589213432 3589213432 IN IP4 192.168.1.2
s=pjmedia
c=IN IP4 192.168.1.2
t=0 0
m=audio 4000 RTP/AVP 103 102 104 109 3 0 8 9 101
a=rtcp:4001 IN IP4 192.168.1.2
a=rtpmap:103 speex/16000
a=rtpmap:102 speex/8000
a=rtpmap:104 speex/32000
a=rtpmap:109 iLBC/8000
a=fmtp:109 mode=30
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (13 headers 19 lines) ---

Merci pour votre aide :-)

**jean** · 26/09/2013, 22h34

c'est bizarre. la 1ere trace ngrep ne montre pas d appel.
dans la 2eme, l'appel est accepte (trying puis ok). a quel momemt cela plante ? quels codecs utilise tu ?

Discussion: Premiers pas avec Asterisk, besoin d'aide ! :-)

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Premiers pas avec Asterisk, besoin d'aide ! :-)

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