[SR-Users] TM Module - Unfortunately error on sending to next hop occurred (477/TM)

Ilie Soltanici iliusha.md at gmail.com
Mon Jan 28 19:16:17 CET 2019


Hello,

Can someone help me to understand this error given by TM Module?

I have a Kamailio SIP Proxy running in front of a few Asterisk boxes. On
average, he is processing 70-80 requests per second. The problem is that
sometimes (90-100 times/day) - I see such errors in the logs:

500 I'm terribly sorry, server error occurred (1/SL)
477 Unfortunately error on sending to next hop occurred (477/TM)

Trying to investigate this error - I found that this is happening randomly
and for INVITES coming from the Asterisk Box. For ex, the INVITE below:

192.168.180.10 - Kamailio Server
192.168.180.36 - Asterisk Server

INVITE sip:1001 at 192.168.180.10:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.180.36:5060;branch=z9hG4bK4624ab4e;rport
Max-Forwards: 70
From: "Anonymous" <sip:anonymous at 192.168.180.10>;tag=as11bed8a6
To: <sip:1001 at 192.168.180.10:5060>
Contact: <sip:anonymous at 192.168.180.36:5060>
Call-ID: 0ea8535d3a5ecaa6432912a2566c807f at 192.168.180.10
CSeq: 102 INVITE
User-Agent: MYCOMPANY
Date: Mon, 28 Jan 2019 16:01:56 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 251

v=0
o=SIP 2139556641 2139556641 IN IP4 192.168.180.36
s=MYCOMPANY PBX
c=IN IP4 192.168.180.36
t=0 0
m=audio 14674 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

Kamailio is replying back with 100 Trying and then with 500/477 errors:

SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.180.36:5060
;branch=z9hG4bK4624ab4e;rport=5060;received=192.168.180.36
From: "Anonymous" <sip:anonymous at 192.168.180.10>;tag=as11bed8a6
To: <sip:1001 at 192.168.180.10:5060>
Call-ID: 0ea8535d3a5ecaa6432912a2566c807f at 192.168.180.10
CSeq: 102 INVITE
Server: MYCOMPANY SBC
Content-Length: 0

SIP/2.0 500 I'm terribly sorry, server error occurred (1/SL)
Via: SIP/2.0/UDP 192.168.180.36:5060
;branch=z9hG4bK4624ab4e;rport=5060;received=192.168.180.36
From: "Anonymous" <sip:anonymous at 192.168.180.10>;tag=as11bed8a6
To: <sip:1001 at 192.168.180.10:5060>;tag=9dd61ff61e802d8e2bef5f14621ef3c2.2e93
Call-ID: 0ea8535d3a5ecaa6432912a2566c807f at 192.168.180.10
CSeq: 102 INVITE
Server: MYCOMPANY SBC
Content-Length: 0

SIP/2.0 477 Unfortunately error on sending to next hop occurred (477/TM)
Via: SIP/2.0/UDP 192.168.180.36:5060
;branch=z9hG4bK4624ab4e;rport=5060;received=192.168.180.36
From: "Anonymous" <sip:anonymous at 192.168.180.10>;tag=as11bed8a6
To: <sip:1001 at 192.168.180.10:5060>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-2e93
Call-ID: 0ea8535d3a5ecaa6432912a2566c807f at 192.168.180.10
CSeq: 102 INVITE
Server: MYCOMPANY SBC
Content-Length: 0

And I cannot actually get what is wrong with this INVITE - and why Kamailio
cannot process it? Other calls for the same extension are working fine,
this is happening randomly and with different extensions.
Load on the server is very low:

[root at kamailio root]# nproc
8

[root at kamailio root]# uptime
 17:49:15 up 8 days, 21:17,  6 users,  load average: 0.66, 0.56, 0.52

[root at kamailio root]# free -m
              total        used        free      shared  buff/cache
 available
Mem:           7981        4283         142         818        3556
2581
Swap:          6141         137        6004


[root at kamailio root]# ss -4 -n -l | grep 5060
udp    UNCONN     0      0       192.168.180.10  :5060                  *:*
udp    UNCONN     0      0      127.0.0.1:5060                  *:*
tcp    LISTEN     0      128     192.168.180.10  :5060                  *:*
tcp    LISTEN     0      128    127.0.0.1:5060                  *:*

In the kamailio logs I found that kamailio was able to get the contact
address:
kamailio[18168]: INFO: {INVITE (1) Contacts loaded for 1001}
kamailio[18168]: INFO: {INVITE (1)  t_next_contacts - Only one contact
found for 1001, calling}
kamailio[18168]: INFO: {INVITE (1)  Next Hop: <192.168.180.211:3126>}

I don't know how to reproduce this - I tried to disconnect the phone from
the power source - and made a call to that extension, and it is giving
timeouts - like it is supposed to be.

What could be the problem and how I can fix it?
Thank You.
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