Maximum retries exceeded on transmission

Asterisk Asterisk Support
1

Hi

I get this message on my asterisk debug. The Asterisk debug is as follows:

I don’t know if this error is relevant to my problem I am having. since the beginning of the week I am getting alot of dropped calls. The call holds for 5 to 30 sec and it takes along time to connect to the other side of the call sometimes the call just cuts out without getting to the other end.

Your help will be appreciated, thanks you in advance.

<------------->
— (14 headers 0 lines) —
Really destroying SIP dialog ‘15c097a240ef640464cd4fec10846107@192.168.95.10’ Method: OPTIONS
Really destroying SIP dialog ‘3c2670548302-h64qvu0qmf4w’ Method: REGISTER
MyPBX*CLI>
<— SIP read from UDP:192.168.6.123:2048 —>
SUBSCRIBE sip:Unknown@192.168.95.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.6.123:2048;branch=z9hG4bK-kmj3xj5r83a9;rport
From: sip:2689@192.168.95.10;tag=ec1mag209i
To: sip:*2@192.168.95.10;user=phone;tag=as3c0c5704
Call-ID: 3c2670225d60-jj2r57r0yw6t
CSeq: 7005 SUBSCRIBE
Max-Forwards: 70
Contact: sip:2689@192.168.6.123:2048;line=3v410926;reg-id=1
Event: message-summary
Accept: application/simple-message-summary
User-Agent: snom300/8.4.32
Expires: 3600
Content-Length: 0

<------------->
— (14 headers 0 lines) —
Really destroying SIP dialog ‘15c097a240ef640464cd4fec10846107@192.168.95.10’ Method: OPTIONS
Really destroying SIP dialog ‘3c2670548302-h64qvu0qmf4w’ Method: REGISTER
MyPBX*CLI>
<— SIP read from UDP:192.168.6.123:2048 —>
SUBSCRIBE sip:Unknown@192.168.95.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.6.123:2048;branch=z9hG4bK-kmj3xj5r83a9;rport
From: sip:2689@192.168.95.10;tag=ec1mag209i
To: sip:*2@192.168.95.10;user=phone;tag=as3c0c5704
Call-ID: 3c2670225d60-jj2r57r0yw6t
CSeq: 7005 SUBSCRIBE
Max-Forwards: 70
Contact: sip:2689@192.168.6.123:2048;line=3v410926;reg-id=1
Event: message-summary
Accept: application/simple-message-summary
User-Agent: snom300/8.4.32
Expires: 3600
Content-Length: 0

<------------->
— (13 headers 0 lines) —
Found peer ‘2689’ for ‘2689’ from 192.168.6.123:2048
MyPBX*CLI>
<— Transmitting (NAT) to 192.168.6.123:2048 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.6.123:2048;branch=z9hG4bK-kmj3xj5r83a9;received=192.168.6.123;rport=2048
From: sip:2689@192.168.95.10;tag=ec1mag209i
To: sip:*2@192.168.95.10;user=phone;tag=as3c0c5704
Call-ID: 3c2670225d60-jj2r57r0yw6t
CSeq: 7005 SUBSCRIBE
Server: MyPBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="48bf2c69"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘3c2670225d60-jj2r57r0yw6t’ in 6400 ms (Method: SUBSCRIBE)
[2015-01-22 13:06:48] WARNING[589]: chan_sip.c:3882 retrans_pkt: Maximum retries exceeded on transmission f5dbe0-c0a801f9-13c4-ce0-324e3a-5a549db9@192.168.1.249 for seqno 1 (Critical Response) – See doc/sip-retransmit.txt.
[2015-01-22 13:06:48] WARNING[589]: chan_sip.c:3909 retrans_pkt: Hanging up call f5dbe0-c0a801f9-13c4-ce0-324e3a-5a549db9@192.168.1.249 - no reply to our critical packet (see doc/sip-retransmit.txt).
– Executing [h@macro-trunkdial-failover-0.3:1] NoOp(“SIP/trunk-sps-samsungSipPeer-00002e28”, “no thing to do”) in new stack
MyPBX*CLI>
<— SIP read from UDP:192.168.6.123:2048 —>
SUBSCRIBE sip:Unknown@192.168.95.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.6.123:2048;branch=z9hG4bK-hrkhtmc0fn4v;rport
From: sip:2689@192.168.95.10;tag=ec1mag209i
To: sip:*2@192.168.95.10;user=phone;tag=as3c0c5704
Call-ID: 3c2670225d60-jj2r57r0yw6t
CSeq: 7006 SUBSCRIBE
Max-Forwards: 70
Contact: sip:2689@192.168.6.123:2048;line=3v410926;reg-id=1
Event: message-summary
Accept: application/simple-message-summary
User-Agent: snom300/8.4.32
Authorization: Digest username=“2689”,realm=“asterisk”,nonce=“48bf2c69”,uri="sip:Unknown@192.168.95.10",response=“6563593e9865c032550b710745197c87”,algorithm=MD5
Expires: 3600
Content-Length: 0

<------------->
— (14 headers 0 lines) —
Found peer ‘2689’ for ‘2689’ from 192.168.6.123:2048
Scheduling destruction of SIP dialog ‘3c2670225d60-jj2r57r0yw6t’ in 250000 ms (Method: SUBSCRIBE)

<— Transmitting (NAT) to 192.168.6.123:2048 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.6.123:2048;branch=z9hG4bK-hrkhtmc0fn4v;received=192.168.6.123;rport=2048
From: sip:2689@192.168.95.10;tag=ec1mag209i
To: sip:*2@192.168.95.10;user=phone;tag=as3c0c5704
Call-ID: 3c2670225d60-jj2r57r0yw6t
CSeq: 7006 SUBSCRIBE
Server: MyPBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Expires: 240
Contact: sip:Unknown@192.168.95.10;expires=240
Content-Length: 0

2

Here is another call:

<— Reliably Transmitting (NAT) to 192.168.16.188:2048 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.16.188:2048;branch=z9hG4bK-c1oj991ve98q;received=192.168.16.188;rport=2048
From: “2661” sip:2661@192.168.95.10;tag=1bt7wj30ay
To: sip:2404@192.168.95.10;user=phone;tag=as5686e5b5
Call-ID: 3c272f0d6f46-sb38fvymnki2
CSeq: 2 INVITE
Server: MyPBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: sip:2404@192.168.95.10
Content-Type: application/sdp
Content-Length: 283

<------------>
MyPBX*CLI>
<— SIP read from UDP:192.168.16.188:2048 —>
ACK sip:2404@192.168.95.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.16.188:2048;branch=z9hG4bK-yfhvesguolw8;rport
From: “2661” sip:2661@192.168.95.10;tag=1bt7wj30ay
To: sip:2404@192.168.95.10;user=phone;tag=as5686e5b5
Call-ID: 3c272f0d6f46-sb38fvymnki2
CSeq: 2 ACK
Max-Forwards: 70
Contact: sip:2661@192.168.16.188:2048;line=p9yna13e;reg-id=1
Content-Length: 0

<------------->
— (14 headers 0 lines) —
Really destroying SIP dialog ‘1fbb8b801e2f6ae017e9dfb92768601c@192.168.95.10’ Method: OPTIONS
Really destroying SIP dialog ‘3c2670210a67-kmdn1crgqmg8’ Method: REGISTER
– Executing [9-CHANUNAVAIL@macro-trunkdial-failover-0.3:2] Hangup(“SIP/2661-00002e1a”, “”) in new stack
== Spawn extension (macro-trunkdial-failover-0.3, 9-CHANUNAVAIL, 2) exited non-zero on ‘SIP/2661-00002e1a’ in macro ‘trunkdial-failover-0.3’
== Spawn extension (DLPN_DialPlan2661, 2404, 4) exited non-zero on ‘SIP/2661-00002e1a’
– Executing [h@DLPN_DialPlan2661:1] NoOp(“SIP/2661-00002e1a”, “no thing to do”) in new stack
– Executing [h@DLPN_DialPlan2661:2] Hangup(“SIP/2661-00002e1a”, “”) in new stack
== Spawn extension (DLPN_DialPlan2661, h, 2) exited non-zero on 'SIP/2661-00002e1a’
Scheduling destruction of SIP dialog ‘3c272f0d6f46-sb38fvymnki2’ in 6400 ms (Method: ACK)
set_destination: Parsing sip:2661@192.168.16.188:2048;line=p9yna13e for address/port to send to
set_destination: set destination to 192.168.16.188, port 2048
Reliably Transmitting (NAT) to 192.168.16.188:2048:
BYE sip:2661@192.168.16.188:2048;line=p9yna13e SIP/2.0
Via: SIP/2.0/UDP 192.168.95.10:5060;branch=z9hG4bK4b22778f;rport
Max-Forwards: 70
From: sip:2404@192.168.95.10;user=phone;tag=as5686e5b5
To: “2661” sip:2661@192.168.95.10;tag=1bt7wj30ay
Call-ID: 3c272f0d6f46-sb38fvymnki2
CSeq: 102 BYE
User-Agent: MyPBX
X-Asterisk-HangupCause: No user responding
X-Asterisk-HangupCauseCode: 18
Content-Length: 0

MyPBX*CLI>
<— SIP read from UDP:192.168.16.188:2048 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.95.10:5060;branch=z9hG4bK4b22778f;rport=5060
From: sip:2404@192.168.95.10;user=phone;tag=as5686e5b5
To: “2661” sip:2661@192.168.95.10;tag=1bt7wj30ay
Call-ID: 3c272f0d6f46-sb38fvymnki2
CSeq: 102 BYE
Contact: sip:2661@192.168.16.188:2048;line=p9yna13e;reg-id=1
User-Agent: snom300/8.4.32
RTP-RxStat: Total_Rx_Pkts=92,Rx_Pkts=0,Rx_Pkts_Lost=1,Remote_Rx_Pkts_Lost=0
RTP-TxStat: Total_Tx_Pkts=783,Tx_Pkts=787,Remote_Tx_Pkts=0
Content-Length: 0

3

Did you follow the instructions in the message?

In any case, you have not included the packet that was being retransmitted.

Based on the timing, you are probably not getting an ACK back. The most common cause for that is bad NAT settings.