Hi and thanks in advance for you help, I am going to be crazy with an italian voip provider Clouditalia. This is the problem:
I have a pbx with Asterisk 11.5 and Freepbx 2.11 and 2 trunk voip: voipcheap, messagenet and clouditalia.
- If I call outside by clouditalia there is any audio and after 31 sec the call drop
- If I call other clouditalia number the audio is OK
- If I call every number by messagenet or voipcheap all is perfect
- I I receive a call on any trunk all is perfect
The problem is the first step. This is the trunk setting and sip debug:
[0123456789]
host=voip.eutelia.it
username=0123456789
secret=123456
type=friend
insecure=port,invite
fromuser=0123456789
context=from-pstn
SIP_GENERAL
vmexten=*97
faxdetect=yes
context=from-sip-external
callerid=Unknown
notifyringing=yes
notifyhold=yes
tos_sip=cs3
tos_audio=ef
tos_video=af41
alwaysauthreject=yes
useragent=FPBX-2.11.0(11.4.0)
disallow=all
allow=alaw
allow=g729
allow=ulaw
languageprefix=yes
t38pt_udptl=yes,redundancy,maxdatagram=400
qualify=yes
faxdetect=yes
directmedia=no
callevents=no
language=it
jbenable=no
defaultexpiry=120
minexpiry=60
notifyhold=yes
registerattempts=0
registertimeout=20
allowguest=no
srvlookup=no
maxexpiry=3600
notifyringing=yes
checkmwi=10
t38pt_udptl=yes,redundancy,maxdatagram=400
g726nonstandard=no
videosupport=no
maxcallbitrate=384
canreinvite=no
rtptimeout=30
rtpholdtimeout=300
rtpkeepalive=0
nat=yes
externip=x.y.z.k
localnet=192.168.8.0/255.255.255.0
DEBUG (08.80.80.80 is PBX IP)
<— SIP read from UDP:83.211.227.21:5060 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 80.80.80.80:5060;received=80.80.80.80;branch=z9hG4bK2af1fb2f;rport=5060
From: “500” sip:[email protected];tag=as1078c9ce
To: sip:[email protected];tag=382C01EC-12BE
Date: Mon, 07 Oct 2013 16:59:10 GMT
Call-ID: [email protected]:5060
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 103 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
Allow-Events: telephone-event
Contact: sip:[email protected]:5060
Record-Route: sip:83.211.227.21;lr=on;ftag=as1078c9ce
Content-Disposition: session;handling=required
Content-Type: application/sdp
Content-Length: 270
v=0
o=CiscoSystemsSIP-GW-UserAgent 4634 4017 IN IP4 83.211.2.220
s=SIP Call
c=IN IP4 62.94.199.40
t=0 0
m=audio 51858 RTP/AVP 18 101
c=IN IP4 62.94.199.40
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
<------------->
— (15 headers 12 lines) —
list_route: hop: sip:83.211.227.21;lr=on;ftag=as1078c9ce
Found RTP audio format 18
Found RTP audio format 101
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|g729), peer - audio=(g729)/video=(nothing)/text=(nothing), combined - (g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 62.94.199.40:51858
– SIP/0802141855-00000313 is making progress passing it to SIP/500-00000312
[2013-10-07 18:59:14] DEBUG[26295][C-0000013d]: sip/sdp_crypto.c:310 sdp_crypto_offer: Crypto line: a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:JIdjqQWFtm5eq88tLyir8t46giFh44vKHRu7jqYr
<— SIP read from UDP:83.211.227.21:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 80.80.80.80:5060;received=80.80.80.80;branch=z9hG4bK2af1fb2f;rport=5060
From: “500” sip:[email protected];tag=as1078c9ce
To: sip:[email protected];tag=382C01EC-12BE
Date: Mon, 07 Oct 2013 16:59:10 GMT
Call-ID: [email protected]:5060
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 103 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
Supported: replaces
Allow-Events: telephone-event
Contact: sip:[email protected]:5060
Record-Route: sip:83.211.227.21;lr=on;ftag=as1078c9ce
Content-Type: application/sdp
Content-Length: 270
v=0
o=CiscoSystemsSIP-GW-UserAgent 4634 4017 IN IP4 83.211.2.220
s=SIP Call
c=IN IP4 62.94.199.40
t=0 0
m=audio 51858 RTP/AVP 18 101
c=IN IP4 62.94.199.40
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
<------------->
— (15 headers 12 lines) —
list_route: hop: sip:83.211.227.21;lr=on;ftag=as1078c9ce
set_destination: Parsing sip:83.211.227.21;lr=on;ftag=as1078c9ce for address/port to send to
set_destination: set destination to 83.211.227.21:5060
Transmitting (NAT) to 83.211.227.21:5060:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 80.80.80.80:5060;branch=z9hG4bK007df846;rport
Route: sip:83.211.227.21;lr=on;ftag=as1078c9ce
Max-Forwards: 70
From: “500” sip:[email protected];tag=as1078c9ce
To: sip:[email protected];tag=382C01EC-12BE
Contact: sip:[email protected]:5060
Call-ID: [email protected]:5060
CSeq: 103 ACK
User-Agent: FPBX-2.11.0(11.4.0)
Content-Length: 0