[Asterisk-Users] SIP Registration Timeout, No FW

[Fredrik von Kantzow]

Hi all,

> sip debug

Sep 27 15:41:14 WARNING[98311]: chan_sip.c:681 retrans_pkt: Maximum retries
exceeded on call 6b8b4567327b23c6643c986966334873 at 127.0.0.1 for seqno 102
(Critical Request)
Sep 27 15:41:14 WARNING[98311]: chan_sip.c:681 retrans_pkt: Maximum retries
exceeded on call 3d1b58ba507ed7ab2eb141f241b71efb at 127.0.0.1 for seqno 102
(Critical Request)
Sep 27 15:41:14 WARNING[98311]: chan_sip.c:681 retrans_pkt: Maximum retries
exceeded on call 0216231b1f16e9e81190cde766ef438d at 127.0.0.1 for seqno 102
(Critical Request)
sunset*CLI> sip debug
SIP Debugging Enabled
Sep 27 15:41:28 NOTICE[98311]: chan_sip.c:4035 sip_reg_timeout: Registration
for '8703040 at sipgate.de' timed out, trying again
11 headers, 0 lines
Reliably Transmitting:
REGISTER sip:sipgate.de SIP/2.0
Via: SIP/2.0/UDP 216.110.117.54:5060;branch=z9hG4bK22d54fb3
From: <sip:8703040 at sipgate.de>;tag=as7cf48bcf
To: <sip:8703040 at sipgate.de>
Call-ID: 6b8b4567327b23c6643c986966334873 at 127.0.0.1
CSeq: 103 REGISTER
User-Agent: Asterisk PBX
Expires: 480
Contact: <sip:8703040 at 216.110.117.54>
Event: registration
Content-Length: 0

 (no NAT) to 217.10.79.9:5060
Sep 27 15:41:28 NOTICE[98311]: chan_sip.c:4035 sip_reg_timeout: Registration
for '0317580829 at proxy.digisip.net' timed out, trying again
11 headers, 0 lines
Reliably Transmitting:
REGISTER sip:proxy.digisip.net SIP/2.0
Via: SIP/2.0/UDP 216.110.117.54:5060;branch=z9hG4bK637133fa
From: <sip:0317580829 at proxy.digisip.net>;tag=as45f596d9
To: <sip:0317580829 at proxy.digisip.net>
Call-ID: 3d1b58ba507ed7ab2eb141f241b71efb at 127.0.0.1
CSeq: 103 REGISTER
User-Agent: Asterisk PBX
Expires: 480
Contact: <sip:1011 at 216.110.117.54>
Event: registration
Content-Length: 0

 (no NAT) to 82.209.165.194:5060
Sep 27 15:41:28 NOTICE[98311]: chan_sip.c:4035 sip_reg_timeout: Registration
for '02070196313 at sipauth.deltathree.com' timed out, trying again
11 headers, 0 lines
Reliably Transmitting:
REGISTER sip:sipauth.deltathree.com SIP/2.0
Via: SIP/2.0/UDP 216.110.117.54:5060;branch=z9hG4bK58b9ab09
From: <sip:02070196313 at sipauth.deltathree.com>;tag=as0f1bd1f5
To: <sip:02070196313 at sipauth.deltathree.com>
Call-ID: 0216231b1f16e9e81190cde766ef438d at 127.0.0.1
CSeq: 103 REGISTER
User-Agent: Asterisk PBX
Expires: 480
Contact: <sip:1010 at 216.110.117.54>
Event: registration
Content-Length: 0

> Well posting some configs with user/pass's blacked out might help.

#sip.conf

[general]
context=default                 ; Default context for incoming calls
port=5060                       ; UDP Port to bind to (SIP standard port is
5060)
bindaddr=0.0.0.0                ; IP address to bind to (0.0.0.0 binds to
all)
externip=216.110.117.54
srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
pedantic=no                     ; Enable slow, pedantic checking for Pingtel
tos=lowdelay                   ;
lowdelay,throughput,reliability,mincost,none
maxexpirey=3600                 ; Max length of incoming registration we
allow
defaultexpirey=480              ; Default length of incoming/outoing
registration
videosupport=no                 ; Turn on support for SIP video
musicclass=default              ; Sets the default music on hold class for
all SIP calls
language=en                     ; Default language setting for all
users/peers
nat=always                              ; NAT settings

register => mynr:myuser:mypass at sipauth.deltathree.com/1010
register => mynr:myuser:mypass at proxy.digisip.net/1011
register => mynr:myuser:mypass at sipgate.de/1012

localnet=192.168.0.0/255.255.255.0

[iconnect]
nat=yes
type=friend
secret=mypass
username=myuser
host=sipauth.deltathree.com
disallow=all
allow=gsm
allow=ulaw
allow=alaw
allow=G726
context=incomingiconnect

[digisip]
nat=yes
type=friend
secret=mypass
username=myuser
host=proxy.digisip.net
disallow=all
allow=gsm
allow=ulaw
allow=alaw
allow=G726
context=incomingdigisip

[sipgate]
nat=yes
type=friend
username=myuser
secret=mypass
host=sipgate.de
fromuser=mynr
fromdomain=sipgate.net
;dtmfband=inband
context=incomingsipgate
canreinvite=no

-Fredrik