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Asterisk 1.6 on vmware

Asterisk Asterisk Support

kgoedert December 21, 2012, 12:27pm 1

Hi,

I installed a asterisk 1.6 on a vmware box. But I can not make outgoing calls using my voip provider.
This is my sip.conf

[troncodovono]
type=peer
username=
secret=
domain=vono.net.br
fromuser=
fromdomain=vono.net.br
host=vono.net.br
insecure=invite,port; (no asterisk > 1.4 utilize “invite,port”)
qualify=no
port=5060
nat=no
disallow=all
allow=ulaw
dtmfmode=rfc2833
context=recebe_vono
reinvite=no
canreinvite=no
externip= <ip_addr>
localnet= 10.1.1.0/255.0.0.0

Setting the debug on I get the following output:

originate SIP/troncodovono/<numero_telefone> extension
== Using SIP RTP CoS mark 5
Audio is at 10.1.1.185 port 17802
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 201.86.87.35:5060:
INVITE sip:<numero_telefone>@vono.net.br SIP/2.0
Via: SIP/2.0/UDP 10.1.1.185:5060;branch=z9hG4bK37ac5281;rport
Max-Forwards: 70
From: “asterisk” <sip:@vono.net.br>;tag=as0db490a4
To: <sip:<numero_telefone>@vono.net.br>
Contact: <sip:@10.1.1.185>
Call-ID: 22e5e71411de93e808ee469c7ee6b704@vono.net.br
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0.1
Date: Fri, 21 Dec 2012 11:32:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 259

v=0
o=root 1719013047 1719013047 IN IP4 10.1.1.185
s=Asterisk PBX 1.6.0.1
c=IN IP4 10.1.1.185
t=0 0
m=audio 17802 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

Retransmitting #1 (no NAT) to 201.86.87.35:5060:
INVITE sip:<numero_telefone>@vono.net.br SIP/2.0
Via: SIP/2.0/UDP 10.1.1.185:5060;branch=z9hG4bK37ac5281;rport
Max-Forwards: 70
From: “asterisk” <sip:@vono.net.br>;tag=as0db490a4
To: <sip:<numero_telefone>@vono.net.br>
Contact: <sip:@10.1.1.185>
Call-ID: 22e5e71411de93e808ee469c7ee6b704@vono.net.br
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0.1
Date: Fri, 21 Dec 2012 11:32:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 259

v=0
o=root 1719013047 1719013047 IN IP4 10.1.1.185
s=Asterisk PBX 1.6.0.1
c=IN IP4 10.1.1.185
t=0 0
m=audio 17802 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

Retransmitting #2 (no NAT) to 201.86.87.35:5060:
INVITE sip:<numero_telefone>@vono.net.br SIP/2.0
Via: SIP/2.0/UDP 10.1.1.185:5060;branch=z9hG4bK37ac5281;rport
Max-Forwards: 70
From: “asterisk” <sip:@vono.net.br>;tag=as0db490a4
To: <sip:<numero_telefone>@vono.net.br>
Contact: <sip:@10.1.1.185>
Call-ID: 22e5e71411de93e808ee469c7ee6b704@vono.net.br
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0.1
Date: Fri, 21 Dec 2012 11:32:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 259

v=0
o=root 1719013047 1719013047 IN IP4 10.1.1.185
s=Asterisk PBX 1.6.0.1
c=IN IP4 10.1.1.185
t=0 0
m=audio 17802 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

Retransmitting #3 (no NAT) to 201.86.87.35:5060:
INVITE sip:<numero_telefone>@vono.net.br SIP/2.0
Via: SIP/2.0/UDP 10.1.1.185:5060;branch=z9hG4bK37ac5281;rport
Max-Forwards: 70
From: “asterisk” <sip:@vono.net.br>;tag=as0db490a4
To: <sip:<numero_telefone>@vono.net.br>
Contact: <sip:@10.1.1.185>
Call-ID: 22e5e71411de93e808ee469c7ee6b704@vono.net.br
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0.1
Date: Fri, 21 Dec 2012 11:32:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 259

v=0
o=root 1719013047 1719013047 IN IP4 10.1.1.185
s=Asterisk PBX 1.6.0.1
c=IN IP4 10.1.1.185
t=0 0
m=audio 17802 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

Retransmitting #4 (no NAT) to 201.86.87.35:5060:
INVITE sip:<numero_telefone>@vono.net.br SIP/2.0
Via: SIP/2.0/UDP 10.1.1.185:5060;branch=z9hG4bK37ac5281;rport
Max-Forwards: 70
From: “asterisk” <sip:@vono.net.br>;tag=as0db490a4
To: <sip:<numero_telefone>@vono.net.br>
Contact: <sip:@10.1.1.185>
Call-ID: 22e5e71411de93e808ee469c7ee6b704@vono.net.br
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0.1
Date: Fri, 21 Dec 2012 11:32:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 259

v=0
o=root 1719013047 1719013047 IN IP4 10.1.1.185
s=Asterisk PBX 1.6.0.1
c=IN IP4 10.1.1.185
t=0 0
m=audio 17802 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

Retransmitting #5 (no NAT) to 201.86.87.35:5060:
INVITE sip:<numero_telefone>@vono.net.br SIP/2.0
Via: SIP/2.0/UDP 10.1.1.185:5060;branch=z9hG4bK37ac5281;rport
Max-Forwards: 70
From: “asterisk” <sip:@vono.net.br>;tag=as0db490a4
To: <sip:<numero_telefone>@vono.net.br>
Contact: <sip:@10.1.1.185>
Call-ID: 22e5e71411de93e808ee469c7ee6b704@vono.net.br
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0.1
Date: Fri, 21 Dec 2012 11:32:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 259

v=0
o=root 1719013047 1719013047 IN IP4 10.1.1.185
s=Asterisk PBX 1.6.0.1
c=IN IP4 10.1.1.185
t=0 0
m=audio 17802 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

Reliably Transmitting (no NAT) to 201.86.87.35:5060:
CANCEL sip:<numero_telefone>@vono.net.br SIP/2.0
Via: SIP/2.0/UDP 10.1.1.185:5060;branch=z9hG4bK37ac5281;rport
Max-Forwards: 70
From: “asterisk” <sip:@vono.net.br>;tag=as0db490a4
To: <sip:<numero_telefone>@vono.net.br>
Call-ID: 22e5e71411de93e808ee469c7ee6b704@vono.net.br
CSeq: 102 CANCEL
User-Agent: Asterisk PBX 1.6.0.1
Content-Length: 0

Scheduling destruction of SIP dialog '22e5e71411de93e808ee469c7ee6b704@vono.net.br’ in 32000 ms (Method: INVITE)
Retransmitting #1 (no NAT) to 201.86.87.35:5060:
CANCEL sip:<numero_telefone>@vono.net.br SIP/2.0
Via: SIP/2.0/UDP 10.1.1.185:5060;branch=z9hG4bK37ac5281;rport
Max-Forwards: 70
From: “asterisk” <sip:@vono.net.br>;tag=as0db490a4
To: <sip:<numero_telefone>@vono.net.br>
Call-ID: 22e5e71411de93e808ee469c7ee6b704@vono.net.br
CSeq: 102 CANCEL
User-Agent: Asterisk PBX 1.6.0.1
Content-Length: 0

Retransmitting #2 (no NAT) to 201.86.87.35:5060:
CANCEL sip:<numero_telefone>@vono.net.br SIP/2.0
Via: SIP/2.0/UDP 10.1.1.185:5060;branch=z9hG4bK37ac5281;rport
Max-Forwards: 70
From: “asterisk” <sip:@vono.net.br>;tag=as0db490a4
To: <sip:<numero_telefone>@vono.net.br>
Call-ID: 22e5e71411de93e808ee469c7ee6b704@vono.net.br
CSeq: 102 CANCEL
User-Agent: Asterisk PBX 1.6.0.1
Content-Length: 0

Retransmitting #3 (no NAT) to 201.86.87.35:5060:
CANCEL sip:<numero_telefone>@vono.net.br SIP/2.0
Via: SIP/2.0/UDP 10.1.1.185:5060;branch=z9hG4bK37ac5281;rport
Max-Forwards: 70
From: “asterisk” <sip:@vono.net.br>;tag=as0db490a4
To: <sip:<numero_telefone>@vono.net.br>
Call-ID: 22e5e71411de93e808ee469c7ee6b704@vono.net.br
CSeq: 102 CANCEL
User-Agent: Asterisk PBX 1.6.0.1
Content-Length: 0

Retransmitting #4 (no NAT) to 201.86.87.35:5060:
CANCEL sip:<numero_telefone>@vono.net.br SIP/2.0
Via: SIP/2.0/UDP 10.1.1.185:5060;branch=z9hG4bK37ac5281;rport
Max-Forwards: 70
From: “asterisk” <sip:@vono.net.br>;tag=as0db490a4
To: <sip:<numero_telefone>@vono.net.br>
Call-ID: 22e5e71411de93e808ee469c7ee6b704@vono.net.br
CSeq: 102 CANCEL
User-Agent: Asterisk PBX 1.6.0.1
Content-Length: 0

Retransmitting #5 (no NAT) to 201.86.87.35:5060:
CANCEL sip:<numero_telefone>@vono.net.br SIP/2.0
Via: SIP/2.0/UDP 10.1.1.185:5060;branch=z9hG4bK37ac5281;rport
Max-Forwards: 70
From: “asterisk” <sip:@vono.net.br>;tag=as0db490a4
To: <sip:<numero_telefone>@vono.net.br>
Call-ID: 22e5e71411de93e808ee469c7ee6b704@vono.net.br
CSeq: 102 CANCEL
User-Agent: Asterisk PBX 1.6.0.1
Content-Length: 0

Retransmitting #6 (no NAT) to 201.86.87.35:5060:
CANCEL sip:<numero_telefone>@vono.net.br SIP/2.0
Via: SIP/2.0/UDP 10.1.1.185:5060;branch=z9hG4bK37ac5281;rport
Max-Forwards: 70
From: “asterisk” <sip:@vono.net.br>;tag=as0db490a4
To: <sip:<numero_telefone>@vono.net.br>
Call-ID: 22e5e71411de93e808ee469c7ee6b704@vono.net.br
CSeq: 102 CANCEL
User-Agent: Asterisk PBX 1.6.0.1
Content-Length: 0

[Dec 21 09:33:03] WARNING[1847]: chan_sip.c:2787 retrans_pkt: Maximum retries exceeded on transmission 22e5e71411de93e808ee469c7ee6b704@vono.net.br for seqno 102 (Non-critical Request) – See doc/sip-retransmit.txt.
Really destroying SIP dialog '22e5e71411de93e808ee469c7ee6b704@vono.net.br’ Method: INVITE

Can anybody help?

Thanks

david55 December 21, 2012, 12:49pm 2

VMWare is irrelevant; the effect of using it will be on audio quality, not basic routing.

Your localnet parameter is invalid (set bits in address correspond to clear bits in mask).

externip and localnet are only meaningful in the general section. I think this is the specific fault. You are not sending your externip.

insecure=invite,port, in almost all cases, should actually be insecure=invite. Many ITSP’s give bad advice on this.

“reinvite” is not a valid option.

Version 1.6.2 should not have been used for new installs for more than 18 months now, and earlier 1.6 versions should not have been used for even longer. Security fixes on 1.6.2 ceased in Spring this year, and bug fixes a year before that.

kgoedert December 21, 2012, 1:18pm 3

I changed my sip.conf to this:

[general]
externip=x.y.z.a
localnet=10.0.0.0/255.0.0.0
[troncodovono]
type=peer
username=user
secret=pass
domain=vono.net.br
fromuser=user
fromdomain=vono.net.br
host=vono.net.br
insecure=invite
qualify=no
port=5060
nat=yes
disallow=all
allow=ulaw
dtmfmode=rfc2833
context=recebe_vono
reinvite=no
canreinvite=no

but it still does not work. I know I am using an old version. I picked this one because I have a friend that uses it and I hoped that he could help, but he can’t figure this out either. Since I am just installing it for a simple test in a vmware box, I didn’t mind about the version.

david55 December 21, 2012, 1:22pm 4

Can you provide the SIP trace again. The original SIP trace showed a clearly broken Contact header. That, at least, should have changed.

david55 December 21, 2012, 1:26pm 5

Also, nat=no, before, was correct. This parameter is for handling broken NAT configurations at the remote end. ITSPs generally are not behind NAT or at least not behind broken NAT.

If you SIP trace doesn’t show any responses, you may want to try adding back port on the insecure, although I do not know of cases when it is needed. I think you would still see the responses, though.

kgoedert December 21, 2012, 1:26pm 6

asterix*CLI> originate SIP/troncodovono/ extension
== Using SIP RTP CoS mark 5
Audio is at 201.22.86.160 port 16812
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 201.86.87.35:5060:
INVITE sip:@vono.net.br SIP/2.0
Via: SIP/2.0/UDP 201.22.86.160:5060;branch=z9hG4bK5db87c67;rport
Max-Forwards: 70
From: “asterisk” <sip:@vono.net.br>;tag=as705dfa01
To: <sip:@vono.net.br>
Contact: <sip:@201.22.86.160>
Call-ID: 62091d1139055ff548ef6af670a17546@vono.net.br
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0.1
Date: Fri, 21 Dec 2012 13:18:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 1409057316 1409057316 IN IP4 201.22.86.160
s=Asterisk PBX 1.6.0.1
c=IN IP4 201.22.86.160
t=0 0
m=audio 16812 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

Retransmitting #1 (NAT) to 201.86.87.35:5060:
INVITE sip:@vono.net.br SIP/2.0
Via: SIP/2.0/UDP 201.22.86.160:5060;branch=z9hG4bK5db87c67;rport
Max-Forwards: 70
From: “asterisk” <sip:@vono.net.br>;tag=as705dfa01
To: <sip:@vono.net.br>
Contact: <sip:@201.22.86.160>
Call-ID: 62091d1139055ff548ef6af670a17546@vono.net.br
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0.1
Date: Fri, 21 Dec 2012 13:18:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 1409057316 1409057316 IN IP4 201.22.86.160
s=Asterisk PBX 1.6.0.1
c=IN IP4 201.22.86.160
t=0 0
m=audio 16812 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

Retransmitting #2 (NAT) to 201.86.87.35:5060:
INVITE sip:@vono.net.br SIP/2.0
Via: SIP/2.0/UDP 201.22.86.160:5060;branch=z9hG4bK5db87c67;rport
Max-Forwards: 70
From: “asterisk” <sip:@vono.net.br>;tag=as705dfa01
To: <sip:@vono.net.br>
Contact: <sip:@201.22.86.160>
Call-ID: 62091d1139055ff548ef6af670a17546@vono.net.br
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0.1
Date: Fri, 21 Dec 2012 13:18:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 1409057316 1409057316 IN IP4 201.22.86.160
s=Asterisk PBX 1.6.0.1
c=IN IP4 201.22.86.160
t=0 0
m=audio 16812 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

Retransmitting #3 (NAT) to 201.86.87.35:5060:
INVITE sip:@vono.net.br SIP/2.0
Via: SIP/2.0/UDP 201.22.86.160:5060;branch=z9hG4bK5db87c67;rport
Max-Forwards: 70
From: “asterisk” <sip:@vono.net.br>;tag=as705dfa01
To: <sip:@vono.net.br>
Contact: <sip:@201.22.86.160>
Call-ID: 62091d1139055ff548ef6af670a17546@vono.net.br
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0.1
Date: Fri, 21 Dec 2012 13:18:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 1409057316 1409057316 IN IP4 201.22.86.160
s=Asterisk PBX 1.6.0.1
c=IN IP4 201.22.86.160
t=0 0
m=audio 16812 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

Retransmitting #4 (NAT) to 201.86.87.35:5060:
INVITE sip:@vono.net.br SIP/2.0
Via: SIP/2.0/UDP 201.22.86.160:5060;branch=z9hG4bK5db87c67;rport
Max-Forwards: 70
From: “asterisk” <sip:@vono.net.br>;tag=as705dfa01
To: <sip:@vono.net.br>
Contact: <sip:@201.22.86.160>
Call-ID: 62091d1139055ff548ef6af670a17546@vono.net.br
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0.1
Date: Fri, 21 Dec 2012 13:18:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 1409057316 1409057316 IN IP4 201.22.86.160
s=Asterisk PBX 1.6.0.1
c=IN IP4 201.22.86.160
t=0 0
m=audio 16812 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

Retransmitting #5 (NAT) to 201.86.87.35:5060:
INVITE sip:@vono.net.br SIP/2.0
Via: SIP/2.0/UDP 201.22.86.160:5060;branch=z9hG4bK5db87c67;rport
Max-Forwards: 70
From: “asterisk” <sip:@vono.net.br>;tag=as705dfa01
To: <sip:@vono.net.br>
Contact: <sip:@201.22.86.160>
Call-ID: 62091d1139055ff548ef6af670a17546@vono.net.br
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0.1
Date: Fri, 21 Dec 2012 13:18:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 1409057316 1409057316 IN IP4 201.22.86.160
s=Asterisk PBX 1.6.0.1
c=IN IP4 201.22.86.160
t=0 0
m=audio 16812 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

Reliably Transmitting (NAT) to 201.86.87.35:5060:
CANCEL sip:@vono.net.br SIP/2.0
Via: SIP/2.0/UDP 201.22.86.160:5060;branch=z9hG4bK5db87c67;rport
Max-Forwards: 70
From: “asterisk” <sip:@vono.net.br>;tag=as705dfa01
To: <sip:@vono.net.br>
Call-ID: 62091d1139055ff548ef6af670a17546@vono.net.br
CSeq: 102 CANCEL
User-Agent: Asterisk PBX 1.6.0.1
Content-Length: 0

Scheduling destruction of SIP dialog '62091d1139055ff548ef6af670a17546@vono.net.br’ in 32000 ms (Method: INVITE)
Retransmitting #1 (NAT) to 201.86.87.35:5060:
CANCEL sip:@vono.net.br SIP/2.0
Via: SIP/2.0/UDP 201.22.86.160:5060;branch=z9hG4bK5db87c67;rport
Max-Forwards: 70
From: “asterisk” <sip:@vono.net.br>;tag=as705dfa01
To: <sip:@vono.net.br>
Call-ID: 62091d1139055ff548ef6af670a17546@vono.net.br
CSeq: 102 CANCEL
User-Agent: Asterisk PBX 1.6.0.1
Content-Length: 0

Retransmitting #2 (NAT) to 201.86.87.35:5060:
CANCEL sip:@vono.net.br SIP/2.0
Via: SIP/2.0/UDP 201.22.86.160:5060;branch=z9hG4bK5db87c67;rport
Max-Forwards: 70
From: “asterisk” <sip:@vono.net.br>;tag=as705dfa01
To: <sip:@vono.net.br>
Call-ID: 62091d1139055ff548ef6af670a17546@vono.net.br
CSeq: 102 CANCEL
User-Agent: Asterisk PBX 1.6.0.1
Content-Length: 0

Retransmitting #3 (NAT) to 201.86.87.35:5060:
CANCEL sip:@vono.net.br SIP/2.0
Via: SIP/2.0/UDP 201.22.86.160:5060;branch=z9hG4bK5db87c67;rport
Max-Forwards: 70
From: “asterisk” <sip:@vono.net.br>;tag=as705dfa01
To: <sip:@vono.net.br>
Call-ID: 62091d1139055ff548ef6af670a17546@vono.net.br
CSeq: 102 CANCEL
User-Agent: Asterisk PBX 1.6.0.1
Content-Length: 0

Retransmitting #4 (NAT) to 201.86.87.35:5060:
CANCEL sip:@vono.net.br SIP/2.0
Via: SIP/2.0/UDP 201.22.86.160:5060;branch=z9hG4bK5db87c67;rport
Max-Forwards: 70
From: “asterisk” <sip:@vono.net.br>;tag=as705dfa01
To: <sip:@vono.net.br>
Call-ID: 62091d1139055ff548ef6af670a17546@vono.net.br
CSeq: 102 CANCEL
User-Agent: Asterisk PBX 1.6.0.1
Content-Length: 0

Retransmitting #5 (NAT) to 201.86.87.35:5060:
CANCEL sip:@vono.net.br SIP/2.0
Via: SIP/2.0/UDP 201.22.86.160:5060;branch=z9hG4bK5db87c67;rport
Max-Forwards: 70
From: “asterisk” <sip:@vono.net.br>;tag=as705dfa01
To: <sip:@vono.net.br>
Call-ID: 62091d1139055ff548ef6af670a17546@vono.net.br
CSeq: 102 CANCEL
User-Agent: Asterisk PBX 1.6.0.1
Content-Length: 0

Retransmitting #6 (NAT) to 201.86.87.35:5060:
CANCEL sip:@vono.net.br SIP/2.0
Via: SIP/2.0/UDP 201.22.86.160:5060;branch=z9hG4bK5db87c67;rport
Max-Forwards: 70
From: “asterisk” <sip:@vono.net.br>;tag=as705dfa01
To: <sip:@vono.net.br>
Call-ID: 62091d1139055ff548ef6af670a17546@vono.net.br
CSeq: 102 CANCEL
User-Agent: Asterisk PBX 1.6.0.1
Content-Length: 0

[Dec 21 11:19:47] WARNING[1935]: chan_sip.c:2787 retrans_pkt: Maximum retries exceeded on transmission 62091d1139055ff548ef6af670a17546@vono.net.br for seqno 102 (Non-critical Request) – See doc/sip-retransmit.txt.
Really destroying SIP dialog '62091d1139055ff548ef6af670a17546@vono.net.br’ Method: INVITE

david55 December 21, 2012, 1:29pm 7

You probably have a broken router configuration. You should get back a response, even if it is a rejection. Are you sure port 5060 is port forwarded?

kgoedert December 21, 2012, 1:44pm 8

Well, I just checked and I have nothing related to this port on the firewall or the router.

david55 December 21, 2012, 1:49pm 9

The router is unlikely to pass the responses unless explicitly configured to do so.

kgoedert December 21, 2012, 2:00pm 10

I added a rule on the router to forward to the asterisk box, but no luck

kgoedert December 21, 2012, 3:18pm 11

The machine I am using had dhcp enable. I changed it to an static ip address, and now the error is

originate SIP/troncodovono/ extension
== Using SIP RTP CoS mark 5
[Dec 21 13:14:32] WARNING[1331]: chan_sip.c:4181 create_addr: No such host:troncodovono
Really destroying SIP dialog ‘004f125214bf940653b4f9b420e4cd89@127.0.1.1’ Method: INVITE
[Dec 21 13:14:32] NOTICE[1331]: channel.c:3309 __ast_request_and_dial: Unable to request channel SIP/troncodovono/

kgoedert December 21, 2012, 3:35pm 12

Router is set, dns corrected but the error is still the same… I seem to receive no answer to the invite…

david55 December 21, 2012, 4:22pm 13

That error means that the sip.conf section is missging!

kgoedert December 21, 2012, 4:28pm 14

I believe it had something to do with dns, once I corrected the address I went back to the first error

david55 December 21, 2012, 4:48pm 15

If the name after the SIP/ is not present in sip.conf, Asterisk will assume that it is a domain name, and try to look it up. You don’t want that to happen here, as it will mean that none of the specific sip.conf settings are being used for outbound calls.

kgoedert December 21, 2012, 4:55pm 16

Yes… it is configured exactly as I posted, with the corrections you suggested. And the error is still Maximum retries exceeded on transmission and apparently as you pointed out I am not receiving and answer to the invite that asterisks sends out.

david55 December 21, 2012, 5:08pm 17

You need to fix the router, or disable any problem firewall rule. If the router allows you to get packet traces or statistics, I would see how far the requests are getting and confirm whether the responses are being received. The ITSP should be able to tell you whether they are seeing your requests.

kgoedert December 21, 2012, 5:35pm 18

I took a dcpdump on my gateway, but I dont think it is much help

tcpdump -i eth0 -s0 'port 5060’
tcpdump: verbose output suppressed, use -v or -vv for full protocol decode
listening on eth0, link-type EN10MB (Ethernet), capture size 65535 bytes
16:54:38.839356 IP 10.1.1.23.sip > 201.86.87.35.static.host.gvt.net.br.sip: SIP, length: 354
16:54:39.838502 IP 10.1.1.23.sip > 201.86.87.35.static.host.gvt.net.br.sip: SIP, length: 354
16:54:40.838927 IP 10.1.1.23.sip > 201.86.87.35.static.host.gvt.net.br.sip: SIP, length: 354
16:54:42.838527 IP 10.1.1.23.sip > 201.86.87.35.static.host.gvt.net.br.sip: SIP, length: 354
16:54:44.066878 IP 10.1.1.23.sip > 201.86.87.35.static.host.gvt.net.br.sip: SIP, length: 827
16:54:45.066304 IP 10.1.1.23.sip > 201.86.87.35.static.host.gvt.net.br.sip: SIP, length: 827
16:54:46.066230 IP 10.1.1.23.sip > 201.86.87.35.static.host.gvt.net.br.sip: SIP, length: 827
16:54:46.838229 IP 10.1.1.23.sip > 201.86.87.35.static.host.gvt.net.br.sip: SIP, length: 354
16:54:48.066330 IP 10.1.1.23.sip > 201.86.87.35.static.host.gvt.net.br.sip: SIP, length: 827
16:54:50.836680 IP 10.1.1.23.sip > 201.86.87.35.static.host.gvt.net.br.sip: SIP, length: 354
16:54:52.065033 IP 10.1.1.23.sip > 201.86.87.35.static.host.gvt.net.br.sip: SIP, length: 827
16:54:54.835882 IP 10.1.1.23.sip > 201.86.87.35.static.host.gvt.net.br.sip: SIP, length: 354
16:55:00.063184 IP 10.1.1.23.sip > 201.86.87.35.static.host.gvt.net.br.sip: SIP, length: 827
16:55:14.087631 IP 10.1.1.23.sip > 201.86.87.35.static.host.gvt.net.br.sip: SIP, length: 354
16:55:15.087056 IP 10.1.1.23.sip > 201.86.87.35.static.host.gvt.net.br.sip: SIP, length: 354
16:55:16.086731 IP 10.1.1.23.sip > 201.86.87.35.static.host.gvt.net.br.sip: SIP, length: 354
16:55:18.086332 IP 10.1.1.23.sip > 201.86.87.35.static.host.gvt.net.br.sip: SIP, length: 354
16:55:22.085785 IP 10.1.1.23.sip > 201.86.87.35.static.host.gvt.net.br.sip: SIP, length: 354
16:55:26.085736 IP 10.1.1.23.sip > 201.86.87.35.static.host.gvt.net.br.sip: SIP, length: 354
16:55:30.084686 IP 10.1.1.23.sip > 201.86.87.35.static.host.gvt.net.br.sip: SIP, length: 354