[RESOLU] Neuftalk congestion

**Reaper** · 21/03/2011, 16h43

Active la trace sip sur l'ip de votre fournisseur effectue un test d'appel et colle les ici, pour le faire:
sip set debug IP ou IP est celle de votre fournisseur dans l'asterisk CLI

**ylan** · 21/03/2011, 17h51

Merci pour ton aide,

Voici ce que tu as demandé, j'avoue que c'est un peu long
SIP SHOW PEERS

Code:

telecom*CLI> sip show peers
Name/username              Host            Dyn Nat ACL Port     Status
maison/maison            93.12.91.104     D   N      16654    OK (711 ms)
sfr/014635xxxx             213.91.9.206         N      5060     OK (26 ms)
2 sip peers [Monitored: 2 online, 0 offline Unmonitored: 0 online, 0 offline]

SIP SHOW REGISTRY

Code:

telecom*CLI> sip show registry
Host                           dnsmgr Username       Refresh State                Reg.Time
voip.wengo.fr:5060             N      kalaza            1785 Registered           Mon, 21 Mar 2011 16:23:52
1 SIP registrations.

SIP SET DEBUG IP

Code:

telecom*CLI> sip set debug ip 213.91.9.206
SIP Debugging Enabled for IP: 213.91.9.206

Emission d'appel

Code:

 == Using SIP RTP CoS mark 5
    -- Executing [061669xxxx@outgoing:1] Dial("SIP/maison-00000000", "SIP/sfr/061669xxxx") in new stack
  == Using SIP RTP CoS mark 5
Audio is at MON_IP port 21442
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x10 (g726aal2) to SDP
Adding codec 0x20 (adpcm) to SDP
Adding codec 0x40 (slin) to SDP
Adding codec 0x80 (lpc10) to SDP
Adding codec 0x800 (g726) to SDP
Adding codec 0x1000 (g722) to SDP
Adding codec 0x8000 (slin16) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 213.91.9.206:5060:
INVITE sip:061669xxxx@voip.wengo.fr SIP/2.0
Via: SIP/2.0/UDP MON_IP:5060;branch=z9hG4bK3a4c5572;rport
Max-Forwards: 70
From: "maison" <sip:014635xxxx@MON_IP>;tag=as05e0254f
To: <sip:061669xxxx@voip.wengo.fr>
Contact: <sip:014635xxxx@MON_IP>
Call-ID: 7840143a029de92c23d9af5347a7597d@MON_IP
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.17
Date: Mon, 21 Mar 2011 15:47:12 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 451

v=0
o=root 1665343704 1665343704 IN IP4 MON_IP
s=Asterisk PBX 1.6.2.17
c=IN IP4 MON_IP
t=0 0
m=audio 21442 RTP/AVP 0 8 3 112 5 10 7 111 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
    -- Called sfr/061669xxxx

<--- SIP read from UDP:213.91.9.206:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP MON_IP:5060;branch=z9hG4bK3a4c5572;rport=5060
From: "maison" <sip:014635xxxx@MON_IP>;tag=as05e0254f
To: <sip:061669xxxx@voip.wengo.fr>;tag=b6a81649b79ec740d23d81d957711b9e.090a
Call-ID: 7840143a029de92c23d9af5347a7597d@MON_IP
CSeq: 102 INVITE
Server: OpenSer (1.1.0-notls (i386/linux))
Content-Length: 0
Warning: 392 192.168.70.88:5062 "Noisy feedback tells:  pid=7323 req_src_ip=192.168.70.249 req_src_port=5060 in_uri=sip:061669xxxx@voip.wengo.fr out_uri=sip:061669xxxx@voip.wengo.fr via_cnt==2"


<------------->
--- (9 headers 0 lines) ---
Transmitting (NAT) to 213.91.9.206:5060:
ACK sip:061669xxxx@voip.wengo.fr SIP/2.0
Via: SIP/2.0/UDP MON_IP:5060;branch=z9hG4bK3a4c5572;rport
Max-Forwards: 70
From: "maison" <sip:014635xxxx@MON_IP>;tag=as05e0254f
To: <sip:061669xxxx@voip.wengo.fr>;tag=b6a81649b79ec740d23d81d957711b9e.090a
Contact: <sip:014635xxxx@MON_IP>
Call-ID: 7840143a029de92c23d9af5347a7597d@MON_IP
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.17
Content-Length: 0


---
---
    -- SIP/sfr-00000001 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
    -- Auto fallthrough, channel 'SIP/maison-00000000' status is 'CONGESTION'
Really destroying SIP dialog '7840143a029de92c23d9af5347a7597d@MON_IP' Method: INVITE
Reliably Transmitting (NAT) to 213.91.9.206:5060:
OPTIONS sip:voip.wengo.fr SIP/2.0
Via: SIP/2.0/UDP MON_IP:5060;branch=z9hG4bK23038949;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@MON_IP>;tag=as3940a076
To: <sip:voip.wengo.fr>
Contact: <sip:asterisk@MON_IP>
Call-ID: 049ed9754a43d2315b9c0609240d1dde@MON_IP
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.17
Date: Mon, 21 Mar 2011 15:47:17 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:213.91.9.206:5060 --->
SIP/2.0 405 Method not supported
Via: SIP/2.0/UDP MON_IP:5060;branch=z9hG4bK23038949;rport=5060
From: "asterisk" <sip:asterisk@MON_IP>;tag=as3940a076
To: <sip:voip.wengo.fr>;tag=c563187eab94fcf9eeae89a92b5c0971.c845
Call-ID: 049ed9754a43d2315b9c0609240d1dde@MON_IP
CSeq: 102 OPTIONS
Server: OpenSer (1.1.0-notls (i386/linux))
Content-Length: 0
Warning: 392 192.168.70.249:5060 "Noisy feedback tells:  pid=13724 req_src_ip=MON_IP req_src_port=5060 in_uri=sip:voip.wengo.fr out_uri=sip:voip.wengo.fr via_cnt==1"


<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '049ed9754a43d2315b9c0609240d1dde@MON_IP' Method: OPTIONS
telecom*CLI>

**Reaper** · 21/03/2011, 18h37

Bonjour, il est possible que votre fournisseur vous demande de sortir en E.164 c.a.d au lieu 01xxxxxxxx 331xxxxx
il vous faut modifier votre plan de numérotation en sorte:

Pour national
exten=> _0XXXXXXXXX,1,Dial(SIP/compte/33${EXTEN:1},,Rr)
International
exten=> _00.,1,Dial(SIP/compte/${EXTEN:2},,Rr)

Et pas de answer & hangup devant / après

exten => _0ZXXXXXXXX,1,Answer() >>>>>>Retirer
exten => _0ZXXXXXXXX,n,Dial(SIP/maison/${EXTEN},,Rr)
exten => _0ZXXXXXXXX,n,Hangup() >>>>>>Retirer

**ylan** · 22/03/2011, 00h43

Bonjour,

Merci pour la réponse.

Le résultat est le même en format E.164, circuit-busy

Cependant, pourquoi avec xlite le compte focntionne et en passant par asterisk cela ne fonctionne pas?

**Reaper** · 22/03/2011, 01h15

L'utilisateur maison c'est ton fournisseur je suppose ?
Il te répond 404 Not Found (User not found) tu est certain que tu utilise les bons paramètres de ton fournisseur ?
Met les paramètres de l'exemple:

http://www.voip-info.org/wiki/view/A...ings+for+Wengo

**ylan** · 23/03/2011, 00h54

C'est bon, ça marche:
Pour vous aider voici la conf qui marche pour sfr

Code:

[general]
context=default 
bindport=5060   
register => SIPUSER:SIPPASSWORD@voip.wengo.fr

[wengo]
qualify=yes
type=friend
fromdomain=voip.wengo.fr
fromuser=SIPUSER
username=SIPUSER
secret=SIPPASSWORD
host=voip.wengo.fr
progressinband=yes
realm=voip.wengo.fr
outboundproxy=213.91.9.206
outboundproxyport=5060
disallow=all
allow=gsm
allow=ulaw
allow=alaw
dtmfmode=auto
insecure=port,invite
callerid=nicolas<SIPUSER>
canreinvite=no
autocreatepeer=yes
nat=yes

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