Hey All,
I have been having an issue for the last couple of months when making outgoing calls from my asterisk box via VFX. Apon making the call, the call rings, other party picks up, voice passes in both directions, then after 5-ish seconds the call drops. This happens on all outgoing calls. Incoming calls work fine.
Asterisk/VFX have been working fine on this box in the past, and I have SIP trunks with other providers which work fine also so I am not sure where the issue lies. I have included a trace below. Any pointers would be appreciated!
I have tried connecting from different networks (Xnet/xtra/colo), different asterisk versions, and different IP handsets.
Box is on a public IP, hence no NAT settings below. Asterisk version is 1.2.37
mybox*CLI> sip show peer 7985xxxx
* Name : 7985xxxx
Secret : <Set>
MD5Secret : <Not set>
Context : xxxxxxx-incoming
Subscr.Cont. : <Not set>
Language :
AMA flags : Unknown
CallingPres : Presentation Allowed, Not Screened
FromUser : 7985xxxx
FromDomain : pan.wxnz.net
Callgroup :
Pickupgroup :
Mailbox :
VM Extension : asterisk
LastMsgsSent : 32767/65535
Call limit : 0
Dynamic : No
Callerid : "" <>
Expire : -1
Insecure : port,invite
Nat : No
ACL : No
CanReinvite : No
PromiscRedir : No
User=Phone : No
Trust RPID : No
Send RPID : No
DTMFmode : rfc2833
LastMsg : 0
ToHost : pan.wxnz.net
Addr->IP : 58.28.20.150 Port 5060
Defaddr->IP : 0.0.0.0 Port 0
Def. Username: xxxxxxxxxxxxxxxxxxxxxxxxx
SIP Options : 100rel
Codecs : 0x10c (ulaw|alaw|g729)
Codec Order : (alaw,ulaw,g729)
Status : Unmonitored
Useragent :
Reg. Contact :
mybox*CLI> sip debug peer 7985xxxx
SIP Debugging Enabled for IP: 58.28.20.150:5060
We're at 111.222.333.444 port 20012
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 13 lines
Reliably Transmitting (no NAT) to 58.28.20.150:5060:
INVITE sip:07308xxxx@pan.wxnz.net SIP/2.0
Via: SIP/2.0/UDP 111.222.333.444:5060;branch=z9hG4bK034e0c55
From: "Me" <sip:7985xxxx@111.222.333.444>;tag=as258a59bb
To: <sip:07308xxxx@pan.wxnz.net>
Contact: <sip:7985xxxx@111.222.333.444>
Call-ID: 3119eb772977ae2b3f8beff04921dc28@111.222.333.444
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 09 Dec 2009 20:20:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 285
v=0
o=root 4964 4964 IN IP4 111.222.333.444
s=session
c=IN IP4 111.222.333.444
t=0 0
m=audio 20012 RTP/AVP 8 0 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
mybox*CLI>
<-- SIP read from 58.28.20.150:5060:
SIP/2.0 100 Trying
From: "Me" <sip:7985xxxx@111.222.333.444>;tag=as258a59bb
To: <sip:07308xxxx@pan.wxnz.net>
Call-ID: 3119eb772977ae2b3f8beff04921dc28@111.222.333.444
CSeq: 102 INVITE
Via: SIP/2.0/UDP 111.222.333.444:5060;branch=z9hG4bK034e0c55
Content-Length: 0
--- (7 headers 0 lines) ---
mybox*CLI>
<-- SIP read from 58.28.20.150:5060:
SIP/2.0 401 Unauthorized
From: "Me" <sip:7985xxxx@111.222.333.444>;tag=as258a59bb
To: <sip:07308xxxx@pan.wxnz.net>;tag=3201400a-13c4-4b20067a-63fedcc-56cac022
Call-ID: 3119eb772977ae2b3f8beff04921dc28@111.222.333.444
CSeq: 102 INVITE
WWW-Authenticate: DIGEST realm="xport.co.nz", nonce="BroadWorksXg30jo4chTcm196oBW", algorithm=MD5, qop="auth"
Via: SIP/2.0/UDP 111.222.333.444:5060;branch=z9hG4bK034e0c55
Content-Length: 0
--- (8 headers 0 lines) ---
Transmitting (no NAT) to 58.28.20.150:5060:
ACK sip:07308xxxx@pan.wxnz.net SIP/2.0
Via: SIP/2.0/UDP 111.222.333.444:5060;branch=z9hG4bK034e0c55
From: "Me" <sip:7985xxxx@111.222.333.444>;tag=as258a59bb
To: <sip:07308xxxx@pan.wxnz.net>;tag=3201400a-13c4-4b20067a-63fedcc-56cac022
Contact: <sip:7985xxxx@111.222.333.444>
Call-ID: 3119eb772977ae2b3f8beff04921dc28@111.222.333.444
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
We're at 111.222.333.444 port 20012
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 58.28.20.150:5060:
INVITE sip:07308xxxx@pan.wxnz.net SIP/2.0
Via: SIP/2.0/UDP 111.222.333.444:5060;branch=z9hG4bK6cc544e9
From: "Me" <sip:7985xxxx@111.222.333.444>;tag=as258a59bb
To: <sip:07308xxxx@pan.wxnz.net>
Contact: <sip:7985xxxx@111.222.333.444>
Call-ID: 3119eb772977ae2b3f8beff04921dc28@111.222.333.444
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username="1lHagsbl9uiliw7Hgz", realm="xport.co.nz", algorithm=MD5, uri="sip:07308xxxx@pan.wxnz.net", nonce="BroadWorksXg30jo4chTcm196oBW", response="2a20efb71593c008330bd56b3903ebbf", opaque="", qop=auth, cnonce="09474053", nc=00000001
Date: Wed, 09 Dec 2009 20:20:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 285
v=0
o=root 4964 4965 IN IP4 111.222.333.444
s=session
c=IN IP4 111.222.333.444
t=0 0
m=audio 20012 RTP/AVP 8 0 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
mybox*CLI>
<-- SIP read from 58.28.20.150:5060:
SIP/2.0 100 Trying
From: "Me" <sip:7985xxxx@111.222.333.444>;tag=as258a59bb
To: <sip:07308xxxx@pan.wxnz.net>
Call-ID: 3119eb772977ae2b3f8beff04921dc28@111.222.333.444
CSeq: 103 INVITE
Via: SIP/2.0/UDP 111.222.333.444:5060;branch=z9hG4bK6cc544e9
Content-Length: 0
--- (7 headers 0 lines) ---
mybox*CLI>
<-- SIP read from 58.28.20.150:5060:
SIP/2.0 183 Session Progress
From: "Me" <sip:7985xxxx@111.222.333.444>;tag=as258a59bb
To: <sip:07308xxxx@pan.wxnz.net>;tag=3201400a-13c4-4b20067a-63fedcc-56cac022
Call-ID: 3119eb772977ae2b3f8beff04921dc28@111.222.333.444
CSeq: 103 INVITE
Session: Media
Remote-Party-ID: <sip:07308xxxx@10.251.1.11;user=phone>;screen=yes;party=called;privacy=off;id-type=subscriber
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
Via: SIP/2.0/UDP 111.222.333.444:5060;branch=z9hG4bK6cc544e9
Contact: <sip:07308xxxx@pan.wxnz.net:5060;maddr=58.28.20.150;transport=udp>
Content-Type: application/sdp
Content-Length: 214
v=0
o=BroadWorks 6434586 1 IN IP4 58.28.20.150
s=-
c=IN IP4 58.28.20.150
t=0 0
m=audio 35676 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=bsoft:1 image udptl t38
--- (12 headers 10 lines) ---
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 58.28.20.150:35676
Found description format PCMA
Found description format telephone-event
Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Destroying call '0995ec382c90ae481394b5e25d64f715@111.222.333.444'
mybox*CLI>
<-- SIP read from 58.28.20.150:5060:
BYE sip:7985xxxx@111.222.333.444 SIP/2.0
From: <sip:07308xxxx@pan.wxnz.net>;tag=3201400a-13c4-4b20067a-63fedcc-56cac022
To: "Me" <sip:7985xxxx@111.222.333.444>;tag=as258a59bb
Call-ID: 3119eb772977ae2b3f8beff04921dc28@111.222.333.444
CSeq: 1 BYE
Via: SIP/2.0/UDP 58.28.20.150:5060;branch=z9hG4bK-3c390-4b200686-64019cc-152505d7
Max-Forwards: 69
Content-Length: 0
--- (8 headers 0 lines) ---
Sending to 58.28.20.150 : 5060 (non-NAT)
Transmitting (no NAT) to 58.28.20.150:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 58.28.20.150:5060;branch=z9hG4bK-3c390-4b200686-64019cc-152505d7;received=58.28.20.150
From: <sip:07308xxxx@pan.wxnz.net>;tag=3201400a-13c4-4b20067a-63fedcc-56cac022
To: "Me" <sip:7985xxxx@111.222.333.444>;tag=as258a59bb
Call-ID: 3119eb772977ae2b3f8beff04921dc28@111.222.333.444
CSeq: 1 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:7985xxxx@111.222.333.444>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing
---
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