Calls drop after 30 seconds - Retransmission timeout reached

vincent_ · October 15, 2011, 9:26am

Hi,

My Asterisk server 1.8 is on a public IP adress (91.x.x.x)

I am using X-lite and Blink as softphones. After 30 secondes the calls hang up and I have this message :

[Oct 15 07:38:18] WARNING[20945]: chan_sip.c:3386 retrans_pkt: Retransmission timeout reached on transmission YjM4NDAxNGNjM2IwN2U1ZmViZGRjNmU5YWJiN2IyMDB. for seqno 2 (Critical Response) – See doc/sip-retransmit.txt.
Packet timed out after 32001ms with no response

I had a look a this post : http://forums.digium.com/viewtopic.php?t=15112
and I tried to set in my sip.conf the following :

[general]
rtcachefriends=yes
realm = real.com
bindport=5060
srvlookup=yes
language=en
trustrpid = yes
sendrpid = yes
progressinband = never
session-timers=refuse
nat=yes
qualify=yes
externip=91.X.X.X (my public server address)
localnet=192.168.0.0/255.255.0.0
localnet=10.0.0.0/255.0.0.0
localnet=172.16.0.0/12
localnet=169.254.0.0/255.255.0.0

I am not really sure for the last 2 ones externip and localnet
It’s still not working

When I try it with my neighbors wifi networks (another Internet provider) , I don’t have any problem. When I try to use my VoIP provider account directly in my softphones, and my own connection I have no problem.

Is there anyone who could save me from the craziness that is slowly taking me over ?

Thank you very much.

Regards

Vincent

david55 · October 15, 2011, 9:41am

(At least some versions of) X-Lite have broken re-invite handling and will do this after the call is established, if you have directmedia (used to be called canreinvite) set to yes. I would advise using a real IP phone.

As with all SIP problems, you really need to provide the SIP debug output, so that one can see exactly what is going wrong, but my two thoughts are:

your service provider bars SIP (normally only a problem with mobile phone operators);
you have direct media enabled on the service provider side and they have an implementation that is broken as described above (it ignores re-invites, rather than rejecting them).

vincent_ · October 16, 2011, 11:05am

Hi,

Thank you very much for your expertise. Here is the SIP trace :

Really destroying SIP dialog ‘ZWU0MjBhMWZlMDc0Yjc1N2FiYzRkMjhmNzlhMGIwYmU.’ Method: ACK
Retransmitting #9 (NAT) to 222.255.198.86:6880:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 222.255.198.86;branch=z9hG4bK-d8754z-ea7fd7fcc20f3dbe-1—d8754z-;received=222.255.198.86;rport=6880
From: "8888"sip:8888@mydomain.com;tag=16c2c997
To: sip:111@mydomain.com;tag=as29260d2d
Call-ID: MjE2MjBmYjNiZmRiZDU0NzAzODk2ZDk5NDU3MDM1Y2M.
CSeq: 2 INVITE
Server: Asterisk PBX 1.8.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces
Contact: sip:111@91.121.72.132:5060
Content-Type: application/sdp
Content-Length: 287

v=0
o=root 1285846866 1285846866 IN IP4 91.121.72.132
s=Asterisk PBX 1.8.1
c=IN IP4 91.121.72.132
t=0 0
m=audio 14776 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

[Oct 16 12:54:50] WARNING[20945]: chan_sip.c:3386 retrans_pkt: Retransmission timeout reached on transmission MjE2MjBmYjNiZmRiZDU0NzAzODk2ZDk5NDU3MDM1Y2M. for seqno 2 (Critical Response) – See doc/sip-retransmit.txt.
Packet timed out after 29249ms with no response
[Oct 16 12:54:50] WARNING[20945]: chan_sip.c:3415 retrans_pkt: Hanging up call MjE2MjBmYjNiZmRiZDU0NzAzODk2ZDk5NDU3MDM1Y2M. - no reply to our critical packet (see doc/sip-retransmit.txt).
== Spawn extension (test_ing, 111, 2) exited non-zero on 'SIP/8888-00000bb4’
Scheduling destruction of SIP dialog ‘MjE2MjBmYjNiZmRiZDU0NzAzODk2ZDk5NDU3MDM1Y2M.’ in 29248 ms (Method: INVITE)
set_destination: Parsing sip:8888@222.255.198.86 for address/port to send to
set_destination: set destination to 222.255.198.86:5060
Reliably Transmitting (NAT) to 222.255.198.86:6880:
BYE sip:8888@222.255.198.86 SIP/2.0
Via: SIP/2.0/UDP 91.121.72.132:5060;branch=z9hG4bK4ec26204;rport
Max-Forwards: 70
From: sip:111@mydomain.com;tag=as29260d2d
To: “8888"sip:8888@mydomain.com;tag=16c2c997
Call-ID: MjE2MjBmYjNiZmRiZDU0NzAzODk2ZDk5NDU3MDM1Y2M.
CSeq: 102 BYE
User-Agent: Asterisk PBX 1.8.1
Proxy-Authorization: Digest username=“AMARIXsia”, realm=“real.com”, algorithm=MD5, uri=“mydomain.com”, nonce=”", response="f763e96bdb5f8dc755c4a05d72b92bec"
X-Asterisk-HangupCause: Protocol error, unspecified
X-Asterisk-HangupCauseCode: 111
Content-Length: 0

<— SIP read from UDP:222.255.198.86:6880 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 222.255.198.86:6880;branch=z9hG4bK4ec26204;rport=5060
Contact: sip:8888@222.255.198.86:6880
To: "8888"sip:8888@mydomain.com;tag=16c2c997
From: sip:111@mydomain.com;tag=as29260d2d
Call-ID: MjE2MjBmYjNiZmRiZDU0NzAzODk2ZDk5NDU3MDM1Y2M.
CSeq: 102 BYE
User-Agent: X-Lite 4 release 4.1 stamp 63214
Content-Length: 0

<------------->
— (9 headers 0 lines) —
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog ‘MjE2MjBmYjNiZmRiZDU0NzAzODk2ZDk5NDU3MDM1Y2M.’ Method: INVITE

vincent_ · October 16, 2011, 11:19am

I tried with X-lite, Blink and Express talk, and I always have the same problem

david55 · October 16, 2011, 12:25pm

The trace is incomplete. I cant’ tell if the ACK is missing or corrupt. You really need the trace from the first INVITE to the point were it is clear that the packets aren’t getting through.

I would suggest that you either have no valid route from Asterisk to 222.255.198.86, port 6880 or the remote device has no valid route to 91.121.72.132 port 5060. Given that 5060 is standard and 6880 non-standard, I’d suggest you have a problem in the outward direction.

vincent_ · October 17, 2011, 4:45am

Sorry for the last SIP trace, this is the full one. Thank you very much for your help :

s351866*CLI>
e[0K
<— SIP read from UDP:222.255.200.220:61292 —>
INVITE sip:111@mydomain.com SIP/2.0
Via: SIP/2.0/UDP 222.255.200.220:61292;branch=z9hG4bK-d8754z-2e30e0c12b29dbb9-1—d8754z-;rport
Max-Forwards: 70
Contact: sip:8888@222.255.200.220:61292
To: sip:111@mydomain.com
From: "8888"sip:8888@mydomain.com;tag=8e997321
Call-ID: OWY2MjhjZmI0NTU0MzEyODM0OTg2NjYwZjgzN2MxZjk.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: X-Lite 4 release 4.1 stamp 63214
Content-Length: 236

v=0
o=- 12963299534920448 1 IN IP4 222.255.200.220
s=CounterPath X-Lite 4.1
c=IN IP4 222.255.200.220
t=0 0
m=audio 58864 RTP/AVP 107 0 8 101
a=rtpmap:107 BV32/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
— (13 headers 10 lines) —

e[Kns351866*CLI>
e[0KSending to 222.255.200.220:61292 (NAT)
Using INVITE request as basis request - OWY2MjhjZmI0NTU0MzEyODM0OTg2NjYwZjgzN2MxZjk.

e[Kns351866*CLI>
e[0KFound peer ‘8888’ for ‘8888’ from 222.255.200.220:61292

e[Kns351866*CLI>
e[0K
<— Reliably Transmitting (NAT) to 222.255.200.220:61292 —>
SIP/2.0 401 Unauthorized

Via: SIP/2.0/UDP 222.255.200.220:61292;branch=z9hG4bK-d8754z-2e30e0c12b29dbb9-1—d8754z-;received=222.255.200.220;rport=61292

From: "8888"sip:8888@mydomain.com;tag=8e997321

To: sip:111@mydomain.com;tag=as0eba9ab5

Call-ID: OWY2MjhjZmI0NTU0MzEyODM0OTg2NjYwZjgzN2MxZjk.

CSeq: 1 INVITE

Server: Asterisk PBX 1.8.1

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces

WWW-Authenticate: Digest algorithm=MD5, realm=“real.com”, nonce=“4e18197f”

Content-Length: 0

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

e[Kns351866*CLI>
e[0KScheduling destruction of SIP dialog ‘OWY2MjhjZmI0NTU0MzEyODM0OTg2NjYwZjgzN2MxZjk.’ in 22848 ms (Method: INVITE)

e[Kns351866*CLI>
e[0KReally destroying SIP dialog ‘MmZiYmY3NGU0NzZhM2M2ZjdhNzI4MjM1ZmMyMGYzZjQ.’ Method: ACK

e[Kns351866*CLI>
e[0K
<— SIP read from UDP:222.255.200.220:61292 —>
ACK sip:111@mydomain.com SIP/2.0
Via: SIP/2.0/UDP 222.255.200.220:61292;branch=z9hG4bK-d8754z-2e30e0c12b29dbb9-1—d8754z-;rport
Max-Forwards: 70
To: sip:111@mydomain.com;tag=as0eba9ab5
From: "8888"sip:8888@mydomain.com;tag=8e997321
Call-ID: OWY2MjhjZmI0NTU0MzEyODM0OTg2NjYwZjgzN2MxZjk.
CSeq: 1 ACK
Content-Length: 0

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

e[Kns351866*CLI>
e[0K
<— SIP read from UDP:222.255.200.220:61292 —>
INVITE sip:111@mydomain.com SIP/2.0
Via: SIP/2.0/UDP 222.255.200.220:61292;branch=z9hG4bK-d8754z-cebe658596639ce3-1—d8754z-;rport
Max-Forwards: 70
Contact: sip:8888@222.255.200.220:61292
To: sip:111@mydomain.com
From: “8888"sip:8888@mydomain.com;tag=8e997321
Call-ID: OWY2MjhjZmI0NTU0MzEyODM0OTg2NjYwZjgzN2MxZjk.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: X-Lite 4 release 4.1 stamp 63214
Authorization: Digest username=“8888”,realm=“real.com”,nonce=“4e18197f”,uri="sip:111@mydomain.com”,response=“861d27753d6329089f1c323c179c56de”,algorithm=MD5
Content-Length: 236

v=0
o=- 12963299534920448 1 IN IP4 222.255.200.220
s=CounterPath X-Lite 4.1
c=IN IP4 222.255.200.220
t=0 0
m=audio 58864 RTP/AVP 107 0 8 101
a=rtpmap:107 BV32/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
— (14 headers 10 lines) —

e[Kns351866*CLI>
e[0KSending to 222.255.200.220:61292 (NAT)
Using INVITE request as basis request - OWY2MjhjZmI0NTU0MzEyODM0OTg2NjYwZjgzN2MxZjk.

e[Kns351866*CLI>
e[0KFound peer ‘8888’ for ‘8888’ from 222.255.200.220:61292

e[Kns351866*CLI>
e[0K == Using SIP RTP CoS mark 5

e[Kns351866*CLI>
e[0KFound RTP audio format 107
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101

e[Kns351866*CLI>
e[0KFound audio description format BV32 for ID 107

e[Kns351866*CLI>
e[0KFound audio description format telephone-event for ID 101

e[Kns351866*CLI>
e[0KNon-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)

e[Kns351866*CLI>
e[0KPeer audio RTP is at port 222.255.200.220:58864

e[Kns351866*CLI>
e[0KLooking for 111 in test_ing (domain mydomain.com)

e[Kns351866*CLI>
e[0Klist_route: hop: sip:8888@222.255.200.220:61292

e[Kns351866*CLI>
e[0K
<— Transmitting (NAT) to 222.255.200.220:61292 —>
SIP/2.0 100 Trying

Via: SIP/2.0/UDP 222.255.200.220:61292;branch=z9hG4bK-d8754z-cebe658596639ce3-1—d8754z-;received=222.255.200.220;rport=61292

From: "8888"sip:8888@mydomain.com;tag=8e997321

To: sip:111@mydomain.com

Call-ID: OWY2MjhjZmI0NTU0MzEyODM0OTg2NjYwZjgzN2MxZjk.

CSeq: 2 INVITE

Server: Asterisk PBX 1.8.1

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces

Contact: sip:111@91.121.72.132:5060

Content-Length: 0

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

e[Kns351866*CLI>
e[0K – Executing [111@test_ing:1] e[1;36mAnswere[0m(“e[1;35mSIP/8888-00000bf1e[0m”, “e[1;35me[0m”) in new stack

e[Kns351866*CLI>
e[0KAudio is at 5060

e[Kns351866*CLI>
e[0KAdding codec 0x4 (ulaw) to SDP

e[Kns351866*CLI>
e[0KAdding codec 0x8 (alaw) to SDP

e[Kns351866*CLI>
e[0KAdding non-codec 0x1 (telephone-event) to SDP

e[Kns351866*CLI>
e[0K
<— Reliably Transmitting (NAT) to 222.255.200.220:61292 —>
SIP/2.0 200 OK

Via: SIP/2.0/UDP 222.255.200.220:61292;branch=z9hG4bK-d8754z-cebe658596639ce3-1—d8754z-;received=222.255.200.220;rport=61292

From: "8888"sip:8888@mydomain.com;tag=8e997321

To: sip:111@mydomain.com;tag=as2161406c

Call-ID: OWY2MjhjZmI0NTU0MzEyODM0OTg2NjYwZjgzN2MxZjk.

CSeq: 2 INVITE

Server: Asterisk PBX 1.8.1

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces

Contact: sip:111@91.121.72.132:5060

Content-Type: application/sdp

Content-Length: 285

v=0

o=root 936706724 936706724 IN IP4 91.121.72.132

s=Asterisk PBX 1.8.1

c=IN IP4 91.121.72.132

t=0 0

m=audio 13072 RTP/AVP 0 8 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -

a=ptime:20

a=sendrecv

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

e[Kns351866*CLI>
e[0KRetransmitting #1 (NAT) to 222.255.200.220:61292:
SIP/2.0 200 OK

Via: SIP/2.0/UDP 222.255.200.220:61292;branch=z9hG4bK-d8754z-cebe658596639ce3-1—d8754z-;received=222.255.200.220;rport=61292

From: "8888"sip:8888@mydomain.com;tag=8e997321

To: sip:111@mydomain.com;tag=as2161406c

Call-ID: OWY2MjhjZmI0NTU0MzEyODM0OTg2NjYwZjgzN2MxZjk.

CSeq: 2 INVITE

Server: Asterisk PBX 1.8.1

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces

Contact: sip:111@91.121.72.132:5060

Content-Type: application/sdp

Content-Length: 285

v=0

o=root 936706724 936706724 IN IP4 91.121.72.132

s=Asterisk PBX 1.8.1

c=IN IP4 91.121.72.132

t=0 0

m=audio 13072 RTP/AVP 0 8 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -

a=ptime:20

a=sendrecv

e[Kns351866*CLI>
e[0K – Executing [111@test_ing:2] e[1;36mDiale[0m(“e[1;35mSIP/8888-00000bf1e[0m”, “e[1;35mSIP/841203785443@193.105.217.3e[0m”) in new stack

e[Kns351866*CLI>
e[0K == Using SIP RTP CoS mark 5

e[Kns351866*CLI>
e[0K – Called 841203785443@193.105.217.3

e[Kns351866*CLI>
e[0KRetransmitting #2 (NAT) to 222.255.200.220:61292:
SIP/2.0 200 OK

Via: SIP/2.0/UDP 222.255.200.220:61292;branch=z9hG4bK-d8754z-cebe658596639ce3-1—d8754z-;received=222.255.200.220;rport=61292

From: "8888"sip:8888@mydomain.com;tag=8e997321

To: sip:111@mydomain.com;tag=as2161406c

Call-ID: OWY2MjhjZmI0NTU0MzEyODM0OTg2NjYwZjgzN2MxZjk.

CSeq: 2 INVITE

Server: Asterisk PBX 1.8.1

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces

Contact: sip:111@91.121.72.132:5060

Content-Type: application/sdp

Content-Length: 285

v=0

o=root 936706724 936706724 IN IP4 91.121.72.132

s=Asterisk PBX 1.8.1

c=IN IP4 91.121.72.132

t=0 0

m=audio 13072 RTP/AVP 0 8 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -

a=ptime:20

a=sendrecv

e[Kns351866*CLI>
e[0K – SIP/193.105.217.3-00000bf2 is making progress passing it to SIP/8888-00000bf1

e[Kns351866*CLI>
e[0KRetransmitting #3 (NAT) to 222.255.200.220:61292:
SIP/2.0 200 OK

Via: SIP/2.0/UDP 222.255.200.220:61292;branch=z9hG4bK-d8754z-cebe658596639ce3-1—d8754z-;received=222.255.200.220;rport=61292

From: "8888"sip:8888@mydomain.com;tag=8e997321

To: sip:111@mydomain.com;tag=as2161406c

Call-ID: OWY2MjhjZmI0NTU0MzEyODM0OTg2NjYwZjgzN2MxZjk.

CSeq: 2 INVITE

Server: Asterisk PBX 1.8.1

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces

Contact: sip:111@91.121.72.132:5060

Content-Type: application/sdp

Content-Length: 285

v=0

o=root 936706724 936706724 IN IP4 91.121.72.132

s=Asterisk PBX 1.8.1

c=IN IP4 91.121.72.132

t=0 0

m=audio 13072 RTP/AVP 0 8 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -

a=ptime:20

a=sendrecv

e[Kns351866*CLI>
e[0KRetransmitting #4 (NAT) to 222.255.200.220:61292:
SIP/2.0 200 OK

Via: SIP/2.0/UDP 222.255.200.220:61292;branch=z9hG4bK-d8754z-cebe658596639ce3-1—d8754z-;received=222.255.200.220;rport=61292

From: "8888"sip:8888@mydomain.com;tag=8e997321

To: sip:111@mydomain.com;tag=as2161406c

Call-ID: OWY2MjhjZmI0NTU0MzEyODM0OTg2NjYwZjgzN2MxZjk.

CSeq: 2 INVITE

Server: Asterisk PBX 1.8.1

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces

Contact: sip:111@91.121.72.132:5060

Content-Type: application/sdp

Content-Length: 285

v=0

o=root 936706724 936706724 IN IP4 91.121.72.132

s=Asterisk PBX 1.8.1

c=IN IP4 91.121.72.132

t=0 0

m=audio 13072 RTP/AVP 0 8 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -

a=ptime:20

a=sendrecv

e[Kns351866*CLI>
e[0KRetransmitting #5 (NAT) to 222.255.200.220:61292:
SIP/2.0 200 OK

Via: SIP/2.0/UDP 222.255.200.220:61292;branch=z9hG4bK-d8754z-cebe658596639ce3-1—d8754z-;received=222.255.200.220;rport=61292

From: "8888"sip:8888@mydomain.com;tag=8e997321

To: sip:111@mydomain.com;tag=as2161406c

Call-ID: OWY2MjhjZmI0NTU0MzEyODM0OTg2NjYwZjgzN2MxZjk.

CSeq: 2 INVITE

Server: Asterisk PBX 1.8.1

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces

Contact: sip:111@91.121.72.132:5060

Content-Type: application/sdp

Content-Length: 285

v=0

o=root 936706724 936706724 IN IP4 91.121.72.132

s=Asterisk PBX 1.8.1

c=IN IP4 91.121.72.132

t=0 0

m=audio 13072 RTP/AVP 0 8 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -

a=ptime:20

a=sendrecv

e[Kns351866*CLI>
e[0KRetransmitting #6 (NAT) to 222.255.200.220:61292:
SIP/2.0 200 OK

Via: SIP/2.0/UDP 222.255.200.220:61292;branch=z9hG4bK-d8754z-cebe658596639ce3-1—d8754z-;received=222.255.200.220;rport=61292

From: "8888"sip:8888@mydomain.com;tag=8e997321

To: sip:111@mydomain.com;tag=as2161406c

Call-ID: OWY2MjhjZmI0NTU0MzEyODM0OTg2NjYwZjgzN2MxZjk.

CSeq: 2 INVITE

Server: Asterisk PBX 1.8.1

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces

Contact: sip:111@91.121.72.132:5060

Content-Type: application/sdp

Content-Length: 285

v=0

o=root 936706724 936706724 IN IP4 91.121.72.132

s=Asterisk PBX 1.8.1

c=IN IP4 91.121.72.132

t=0 0

m=audio 13072 RTP/AVP 0 8 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -

a=ptime:20

a=sendrecv

e[Kns351866*CLI>
e[0K – SIP/193.105.217.3-00000bf2 answered SIP/8888-00000bf1

e[Kns351866*CLI>
e[0K – Locally bridging SIP/8888-00000bf1 and SIP/193.105.217.3-00000bf2

e[Kns351866*CLI>
e[0KRetransmitting #7 (NAT) to 222.255.200.220:61292:
SIP/2.0 200 OK

Via: SIP/2.0/UDP 222.255.200.220:61292;branch=z9hG4bK-d8754z-cebe658596639ce3-1—d8754z-;received=222.255.200.220;rport=61292

From: "8888"sip:8888@mydomain.com;tag=8e997321

To: sip:111@mydomain.com;tag=as2161406c

Call-ID: OWY2MjhjZmI0NTU0MzEyODM0OTg2NjYwZjgzN2MxZjk.

CSeq: 2 INVITE

Server: Asterisk PBX 1.8.1

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces

Contact: sip:111@91.121.72.132:5060

Content-Type: application/sdp

Content-Length: 285

v=0

o=root 936706724 936706724 IN IP4 91.121.72.132

s=Asterisk PBX 1.8.1

c=IN IP4 91.121.72.132

t=0 0

m=audio 13072 RTP/AVP 0 8 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -

a=ptime:20

a=sendrecv

e[Kns351866*CLI>
e[0KRetransmitting #8 (NAT) to 222.255.200.220:61292:
SIP/2.0 200 OK

Via: SIP/2.0/UDP 222.255.200.220:61292;branch=z9hG4bK-d8754z-cebe658596639ce3-1—d8754z-;received=222.255.200.220;rport=61292

From: "8888"sip:8888@mydomain.com;tag=8e997321

To: sip:111@mydomain.com;tag=as2161406c

Call-ID: OWY2MjhjZmI0NTU0MzEyODM0OTg2NjYwZjgzN2MxZjk.

CSeq: 2 INVITE

Server: Asterisk PBX 1.8.1

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces

Contact: sip:111@91.121.72.132:5060

Content-Type: application/sdp

Content-Length: 285

v=0

o=root 936706724 936706724 IN IP4 91.121.72.132

s=Asterisk PBX 1.8.1

c=IN IP4 91.121.72.132

t=0 0

m=audio 13072 RTP/AVP 0 8 101

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -

a=ptime:20

a=sendrecv

e[Kns351866*CLI>
e[0K[Oct 17 06:32:46] e[1;31mWARNINGe[0m[20945]: e[1;37mchan_sip.ce[0m:e[1;37m3386e[0m e[1;37mretrans_pkte[0m: Retransmission timeout reached on transmission OWY2MjhjZmI0NTU0MzEyODM0OTg2NjYwZjgzN2MxZjk. for seqno 2 (Critical Response) – See doc/sip-retransmit.txt.
Packet timed out after 22849ms with no response
[Oct 17 06:32:46] e[1;31mWARNINGe[0m[20945]: e[1;37mchan_sip.ce[0m:e[1;37m3415e[0m e[1;37mretrans_pkte[0m: Hanging up call OWY2MjhjZmI0NTU0MzEyODM0OTg2NjYwZjgzN2MxZjk. - no reply to our critical packet (see doc/sip-retransmit.txt).

e[Kns351866*CLI>
e[0K
<— SIP read from UDP:222.255.200.220:61292 —>

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

e[Kns351866*CLI>
e[0K == Spawn extension (test_ing, 111, 2) exited non-zero on ‘SIP/8888-00000bf1’

e[Kns351866*CLI>
e[0KScheduling destruction of SIP dialog ‘OWY2MjhjZmI0NTU0MzEyODM0OTg2NjYwZjgzN2MxZjk.’ in 22848 ms (Method: INVITE)

e[Kns351866*CLI>
e[0Kset_destination: Parsing sip:8888@222.255.200.220:61292 for address/port to send to

e[Kns351866*CLI>
e[0Kset_destination: set destination to 222.255.200.220:61292

e[Kns351866*CLI>
e[0KReliably Transmitting (NAT) to 222.255.200.220:61292:
BYE sip:8888@222.255.200.220:61292 SIP/2.0

Via: SIP/2.0/UDP 91.121.72.132:5060;branch=z9hG4bK15807f83;rport

Max-Forwards: 70

From: sip:111@mydomain.com;tag=as2161406c

To: "8888"sip:8888@mydomain.com;tag=8e997321

Call-ID: OWY2MjhjZmI0NTU0MzEyODM0OTg2NjYwZjgzN2MxZjk.

CSeq: 102 BYE

User-Agent: Asterisk PBX 1.8.1

Proxy-Authorization: Digest username=“AMARIXsia”, realm=“real.com”, algorithm=MD5, uri=“mydomain.com”, nonce="", response=“f763e96bdb5f8dc755c4a05d72b92bec”

X-Asterisk-HangupCause: Protocol error, unspecified

X-Asterisk-HangupCauseCode: 111

Content-Length: 0

e[Kns351866*CLI>
e[0KRetransmitting #1 (NAT) to 222.255.200.220:61292:
BYE sip:8888@222.255.200.220:61292 SIP/2.0

Via: SIP/2.0/UDP 91.121.72.132:5060;branch=z9hG4bK15807f83;rport

Max-Forwards: 70

From: sip:111@mydomain.com;tag=as2161406c

To: "8888"sip:8888@mydomain.com;tag=8e997321

Call-ID: OWY2MjhjZmI0NTU0MzEyODM0OTg2NjYwZjgzN2MxZjk.

CSeq: 102 BYE

User-Agent: Asterisk PBX 1.8.1

Proxy-Authorization: Digest username=“AMARIXsia”, realm=“real.com”, algorithm=MD5, uri=“mydomain.com”, nonce="", response=“f763e96bdb5f8dc755c4a05d72b92bec”

X-Asterisk-HangupCause: Protocol error, unspecified

X-Asterisk-HangupCauseCode: 111

Content-Length: 0

e[Kns351866*CLI>
e[0K
<— SIP read from UDP:222.255.200.220:61292 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 222.255.200.220:61292;branch=z9hG4bK15807f83;rport=5060
Contact: sip:8888@222.255.200.220:61292
To: "8888"sip:8888@mydomain.com;tag=8e997321
From: sip:111@mydomain.com;tag=as2161406c
Call-ID: OWY2MjhjZmI0NTU0MzEyODM0OTg2NjYwZjgzN2MxZjk.
CSeq: 102 BYE
User-Agent: X-Lite 4 release 4.1 stamp 63214
Content-Length: 0

<------------->
— (9 headers 0 lines) —

e[Kns351866*CLI>
e[0KSIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog ‘OWY2MjhjZmI0NTU0MzEyODM0OTg2NjYwZjgzN2MxZjk.’ Method: INVITE

e[Kns351866*CLI>
e[0K
<— SIP read from UDP:222.255.200.220:61292 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 222.255.200.220:61292;branch=z9hG4bK15807f83;rport=5060
Contact: sip:8888@222.255.200.220:61292
To: "8888"sip:8888@mydomain.com;tag=8e997321
From: sip:111@mydomain.com;tag=as2161406c
Call-ID: OWY2MjhjZmI0NTU0MzEyODM0OTg2NjYwZjgzN2MxZjk.
CSeq: 102 BYE
User-Agent: X-Lite 4 release 4.1 stamp 63214
Content-Length: 0

<------------->
— (9 headers 0 lines) —

e[Kns351866*CLI>
e[0KReliably Transmitting (NAT) to 222.255.200.220:61292:
OPTIONS sip:8888@222.255.200.220:61292;rinstance=ae1900069a92d085 SIP/2.0

Via: SIP/2.0/UDP 91.121.72.132:5060;branch=z9hG4bK47cc0631;rport

Max-Forwards: 70

From: “asterisk” sip:asterisk@91.121.72.132;tag=as64e2aac4

To: sip:8888@222.255.200.220:61292;rinstance=ae1900069a92d085

Contact: sip:asterisk@91.121.72.132:5060

Call-ID: 4f27e50a2e7d0aa835613df128fe2fd6@91.121.72.132:5060

CSeq: 102 OPTIONS

User-Agent: Asterisk PBX 1.8.1

Date: Mon, 17 Oct 2011 04:32:48 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH

Supported: replaces

Content-Length: 0

e[Kns351866*CLI>
e[0K
<— SIP read from UDP:222.255.200.220:61292 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 91.121.72.132:5060;branch=z9hG4bK47cc0631;rport=5060
Contact: sip:192.168.1.3:61292
To: sip:8888@222.255.200.220:61292;rinstance=ae1900069a92d085;tag=bf115017
From: "asterisk"sip:asterisk@91.121.72.132;tag=as64e2aac4
Call-ID: 4f27e50a2e7d0aa835613df128fe2fd6@91.121.72.132:5060
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Supported: replaces
User-Agent: X-Lite 4 release 4.1 stamp 63214
Content-Length: 0

<------------->
— (13 headers 0 lines) —

e[Kns351866*CLI>
e[0KReally destroying SIP dialog ‘4f27e50a2e7d0aa835613df128fe2fd6@91.121.72.132:5060’ Method: OPTIONS

e[Kns351866*CLI>

david55 · October 17, 2011, 7:34am

That’s not straightforward. The caller is seeing some of the responses and is able to send ACK after the attempt without authorisation, but is neither retransmitting the INVITE, nor returning an ACK on the authorised attempt.

Either a proxy or the remote system is really messed up. Maybe the proxy opened up temporarily.

Your incoming SIP requests are coming from random ports, which strongly indicates that a proxy is messing with them.

vincent_ · October 18, 2011, 11:22am

Thank you very much. So do you think I have to configure something in Asterisk ?

Do you think it is my provider who does something strange ?

As I told you, It is working well with my neighbor connection… So… I don’t really know what to do …

david55 · October 18, 2011, 11:26am

I think your provider or router has done something strange.

redmapleleaf · July 17, 2013, 5:44pm

vincent

have you been able to get your asterisk to work yet? I have similar problem where an outgoing call got drop within the first minute. It is ok if my sip phone is part of my local network. However if my sip phone was logging in to my asterisk server remotely then the outgoing calls just get dropped in the first minute of call. My asterisk server is behind a nat router.

david55 · July 17, 2013, 8:08pm

You need to provide a SIP trace.

dsultan · July 18, 2013, 6:55pm

same problem here
asreisk it public, client is behind adsl nat
SIP ALG is switched off

here is asterisk side:

[code]Retransmitting #9 (NAT) to 109.184.159.49:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 109.184.159.49:5060;branch=z9hG4bK544348511;received=109.184.159.49;rport=5060
From: sip:2003@asterisk;tag=310558963
To: sip:84959398000@asterisk;tag=as12a0f774
Call-ID: 1413745589
CSeq: 21 INVITE
Server: Asterisk PBX 1.8.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:84959398000@asterisk:5060
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 108756213 108756214 IN IP4 asterisk ip
s=Asterisk PBX 1.8.7.0
c=IN IP4 asterisk ip
t=0 0
m=audio 15006 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=video 0 RTP/AVP 103 99

[Jul 18 22:03:04] WARNING[13188]: chan_sip.c:3622 retrans_pkt: Retransmission timeout reached on transmission 1413745589 for seqno 21 (Critical Response) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32000ms with no response
[Jul 18 22:03:04] WARNING[13188]: chan_sip.c:3651 retrans_pkt: Hanging up call 1413745589 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
== Spawn extension (users, 84959398000, 2) exited non-zero on 'SIP/2003-00000d0a’
Scheduling destruction of SIP dialog ‘1413745589’ in 32000 ms (Method: INVITE)
set_destination: Parsing sip:dsultan@109.184.159.49:5060 for address/port to send to
set_destination: set destination to 109.184.159.49:5060
Reliably Transmitting (NAT) to 109.184.159.49:5060:
BYE sip:dsultan@109.184.159.49:5060 SIP/2.0
Via: SIP/2.0/UDP asterisk:5060;branch=z9hG4bK5ff440aa;rport
Max-Forwards: 70
From: sip:84959398000@asetrisk;tag=as12a0f774
To: sip:2003@asterisk;tag=310558963
Call-ID: 1413745589
CSeq: 102 BYE
User-Agent: Asterisk PBX 1.8.7.0
X-Asterisk-HangupCause: Protocol error, unspecified
X-Asterisk-HangupCauseCode: 111
Content-Length: 0 [/code]

and here is what going on on the other leg

[code]SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.3:5060;branch=z9hG4bK544348511;received=109.184.159.49;rport=5060
From: sip:2003@asterisk;tag=310558963
To: sip:84959398000@asterisk;tag=as12a0f774
Call-ID: 1413745589
CSeq: 21 INVITE
Server: Asterisk PBX 1.8.7.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:84959398000@192.168.1.3:5060
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 108756213 108756214 IN IP4 asterisk ip
s=Asterisk PBX 1.8.7.0
c=IN IP4 asterisk ip
t=0 0
m=audio 15006 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
m=video 0 RTP/AVP 103 99

message: MESSAGE REC. CALLID:1413745589
message: Message received from: asterisk ip:5060
message: This is a request
message: 2xx restransmission receveid.
message: DNS resolution with 192.168.1.3:5060
message: getaddrinfo returned the following addresses:
message: 192.168.1.3 port 5060
message: Message sent: (to dest=192.168.1.3:5060)
ACK sip:84959398000@192.168.1.3:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.3:5060;rport;branch=z9hG4bK1886009751
From: sip:2003@asterisk;tag=310558963
To: sip:84959398000@asterisk;tag=as12a0f774
Call-ID: 1413745589
CSeq: 21 ACK
Contact: sip:dsultan@109.184.159.49
Max-Forwards: 70
User-Agent: Linphone/3.6.1 (eXosip2/3.6.0)
Content-Length: 0[/code]

somehow local side send ACK to himself (192.168.1.3) when it gets invite form Asterisk

i have tried many soft phones and hard one - spa-3000 gateway

same problem everywhere, 30 seconds and outgoing call is dropped

incoming are fine

david55 · July 18, 2013, 9:19pm

This is unlikely to be right:

Contact: sip:84959398000@asterisk:5060

It needs to contain a domain name or IP address which will allow the remote party to reatch the sender.

Setting this to Asterisk suggests that you have explicitly specified an invalid externhost.

My guess is the media is only working because comedia is overriding the specified addreses.

dsultan · July 19, 2013, 6:15am

sorry for misanterstooding

I have been replaced real asterisk ip with “asterisk” word.

everything ok in contact field

any suggestions?

david55 · July 19, 2013, 7:10am

Almost certainly, you haven’t configured Asterisk for NAT. nat= doesn’t do that. You need to use externip, externaddr, or stunaddr.

dsultan · July 19, 2013, 8:29am

my aster is not behind nat

should I anyway configure externip externaddr?

my local softphone is behind nat, nat=yes is in sip.conf for it

tried to switch nat=no, just media broken, but retransmits are still here

david55 · July 19, 2013, 8:59am

The posting shows a mixture of routeable and non-routeable addresses. That looks very much like a NAT situation to me!!!

Topic		Replies	Views
All outbound calls drop after softphone timeout Asterisk Support	9	1038	October 25, 2013
Retransmission timeout reached on transmission Asterisk Support	5	482	September 21, 2013
SIP Retransmission warning in Asterisk 1.8.2 Asterisk Support	3	369	February 21, 2011
Asterisk call starts to fail few mins after calling starts with retransmission timeout error Asterisk SIP	6	1778	May 12, 2020
F1 Me this Bug - call disconnected after 30 seconds Asterisk Support	6	429	March 22, 2012

Calls drop after 30 seconds - Retransmission timeout reached

Related topics