HI
I have asterisk 1.6.13 running. My ATA are connected via vpn (openvpn)
Out going calls from asterisk to the ata works fine
Incoming calls from the ata to asterisk cuts off with the error msg
Maximum retries exceeded on transmission [email protected] for
seqno 102 (Critical Response) -- See doc/sip-retransmit.txt.
[Dec 6 13:52:43] WARNING[3921]: chan_sip.c:3858 retrans_pkt:
Hanging up call [email protected] - no reply to our critical
packet (see doc/sip-retransmit.txt).
I been googling this error and it was mentioned to use
t1min= 500 however its only delaying the problem.
any ideas on what is the cause of this problem.
Only 2-3 atas are having this problem the rest are fine.
Here is the sip debug
the sip invites are not being received
and in one of the message a busy response was sent back.
Retransmitting #4 (no NAT) to 10.168.7.103:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.168.7.103:5060;branch=z9hG4bK-60ef482f;received=10.168.7.103
From: Ridge <sip:[email protected]>;tag=f314fd35733eba9bo0
To: <sip:[email protected]>;tag=as4593172b
Call-ID: [email protected]
CSeq: 102 INVITE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:[email protected]>
Content-Type: application/sdp
Content-Length: 337
v=0
o=root 777980638 777980638 IN IP4 41.146.208.131
s=Asterisk PBX 1.6.2.13
c=IN IP4 41.146.208.131
t=0 0
m=audio 19726 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 - - - -
a=ptime:20
a=sendrecv
---
Retransmitting #5 (no NAT) to 10.168.7.103:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.168.7.103:5060;branch=z9hG4bK-60ef482f;received=10.168.7.103
From: Ridge <sip:[email protected]>;tag=f314fd35733eba9bo0
To: <sip:[email protected]>;tag=as4593172b
Call-ID: [email protected]
CSeq: 102 INVITE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:[email protected]>
Content-Type: application/sdp
Content-Length: 337
v=0
o=root 777980638 777980638 IN IP4 41.146.208.131
s=Asterisk PBX 1.6.2.13
c=IN IP4 41.146.208.131
t=0 0
m=audio 19726 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 - - - -
a=ptime:20
a=sendrecv
---
Reliably Transmitting (no NAT) to 10.168.7.103:5060:
OPTIONS sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 41.146.208.131:5060;branch=z9hG4bK226c4b89;rport
Max-Forwards: 70
From: "asterisk" <sip:[email protected]>;tag=as21bdce7e
To: <sip:[email protected]:5060>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.2.13
Date: Mon, 06 Dec 2010 12:47:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
---
<--- SIP read from UDP:10.168.7.103:5060 --->
SIP/2.0 486 Busy Here
To: <sip:[email protected]:5060>;tag=18c8b9ab85ca5068i0
From: "asterisk" <sip:[email protected]>;tag=as21bdce7e
Call-ID: [email protected]
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 41.146.208.131:5060;branch=z9hG4bK226c4b89
Server: Linksys/SPA3102-3.3.6(GW)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '[email protected]'
Method: OPTIONS
Retransmitting #6 (no NAT) to 10.168.7.103:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.168.7.103:5060;branch=z9hG4bK-60ef482f;received=10.168.7.103
From: Ridge <sip:[email protected]>;tag=f314fd35733eba9bo0
To: <sip:[email protected]>;tag=as4593172b
Call-ID: [email protected]
CSeq: 102 INVITE
Server: Asterisk PBX 1.6.2.13
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:[email protected]>
Content-Type: application/sdp
Content-Length: 337
v=0
o=root 777980638 777980638 IN IP4 41.146.208.131
s=Asterisk PBX 1.6.2.13
c=IN IP4 41.146.208.131
t=0 0
m=audio 19726 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 - - - -
a=ptime:20
a=sendrecv
zakir
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