Simple newbie problems

Asterisk Asterisk Support
1

Hi,

I am a newbie and I ahve ordered a SIP Trunk from a SP to get trained on SIP.

I just cannot make any outgoing call from my asterisk. The rest seems OK and I have no clue why it is not working here are the config files and the logs.

I have tested the SIP Trunk and it is OK with SIP Phones…

[Apr 19 15:32:46] WARNING[9486]: chan_sip.c:3641 retrans_pkt: Retransmission timeout reached on transmission 6516c18e51a8084c7baae19438369239@91.121.152.100:5060 for seqno 102 (Critical Request) – See wiki.asterisk.org/wiki/display/ … nsmissions
Packet timed out after 6399ms with no response
[Apr 19 15:32:46] WARNING[9486]: chan_sip.c:3670 retrans_pkt: Hanging up call 6516c18e51a8084c7baae19438369239@91.121.152.100:5060 - no reply to our critical packet (see wiki.asterisk.org/wiki/display/ … nsmissions).
== Everyone is busy/congested at this time (1:0/0/1)
– Auto fallthrough, channel ‘SIP/210-00000004’ status is ‘CHANUNAVAIL’
== End MixMonitor Recording SIP/210-00000004
– Unregistered SIP ‘210’
– Registered SIP ‘210’ at 109.25.254.163:5060
– Unregistered SIP ‘210’
– Registered SIP ‘210’ at 109.25.254.163:5060
– Unregistered SIP ‘210’
– Registered SIP ‘210’ at 109.25.254.163:5060
== Using SIP RTP CoS mark 5
– Executing [0972409194@appel-sortant:1] Set(“SIP/210-00000006”, “DIRNAME=2013-04-19”) in new stack
– Executing [0972409194@appel-sortant:2] Set(“SIP/210-00000006”, “FILENAME=2013-04-19/EXT_fred@fredbovycom_2013-04-19_15:34:09”) in new stack
– Executing [0972409194@appel-sortant:3] Set(“SIP/210-00000006”, “OUTPUT=/var/spool/asterisk/monitor/2013-04-19/EXT_fred@fredbovycom_2013-04-19_15:34:09”) in new stack
– Executing [0972409194@appel-sortant:4] System(“SIP/210-00000006”, “/bin/mkdir -p /var/spool/asterisk/monitor/2013-04-19”) in new stack
– Executing [0972409194@appel-sortant:5] MixMonitor(“SIP/210-00000006”, “2013-04-19/EXT_fred@fredbovycom_2013-04-19_15:34:09.wav,b”) in new stack
– Executing [0972409194@appel-sortant:6] Dial(“SIP/210-00000006”, “SIP/0972409194@anonymous”) in new stack
== Using SIP RTP CoS mark 5
– Called SIP/0972409194@anonymous
== Begin MixMonitor Recording SIP/210-00000006
[Apr 19 15:34:16] WARNING[9486]: chan_sip.c:3641 retrans_pkt: Retransmission timeout reached on transmission 7267080808f16cb30b632aa6241fbe77@91.121.152.100:5060 for seqno 102 (Critical Request) – See wiki.asterisk.org/wiki/display/ … nsmissions
Packet timed out after 6399ms with no response
[Apr 19 15:34:16] WARNING[9486]: chan_sip.c:3670 retrans_pkt: Hanging up call 7267080808f16cb30b632aa6241fbe77@91.121.152.100:5060 - no reply to our critical packet (see wiki.asterisk.org/wiki/display/ … nsmissions).
== Everyone is busy/congested at this time (1:0/0/1)
– Auto fallthrough, channel ‘SIP/210-00000006’ status is ‘CHANUNAVAIL’
== End MixMonitor Recording SIP/210-00000006

/etc/asterisk# cat sip.conf
[general]
context=anonymous
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
register => 003397241YYYY:XXXXXXXX@siptrunk.ovh.net
qualify=yes

[210]
type=friend
auth=md5
username=210
md5=5d0e01873d786XXXXXXX52781ff07720b
callerid=“210” fred@fredbovy.com
host=dynamic
context=appel-sortant
language=fr
insecure=port
nat=yes
canreinvite=no
dtmfmode=auto
video=no
restrictcid=no
amaflags=default
mailbox=210@ipv6forlife.com
qualify=yes

[220]
type=friend
username=220
secret=XXXX
callerid=“220” fredbovy@fredbovy.com
host=dynamic
context=appel-sortant
language=fr
insecure=port
nat=yes
canreinvite=no
dtmfmode=auto
video=no
restrictcid=no
amaflags=default
mailbox=220@ipv6forlife.com
qualify=yes

[anonymous]
type=peer
host=siptrunk.ovh.net
context=ovh-sip
language=fr
insecure=very
username=0033972414xxx
secret=xxxxxxxxxxx
nat=yes
canreinvite=no
dtmfmode=auto
video=no
restrictcid=no
amaflags=default
qualify=yes

etc/asterisk# cat extensions.conf

[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
priorityjumping=no

[globals]
CONSOLE=Console/dsp
IAXINFO=guest
TRUNK=Zap/g2
TRUNKMSD=1

[ovh-sip]
exten => s,1,Ringing(1)
exten => s,2,Answer
exten => s,3,Set(TIMEOUT(digit)=1)
exten => s,4,Background(tt-monkeys)
exten => s,5,Waitexten(15)

exten => 1,1,Answer
exten => 1,2,Set(DIRNAME=${STRFTIME(${EPOCH},%Y-%m-%d)})
exten => 1,3,Set(FILENAME=${DIRNAME}/210_${CALLERID(num)}${STRFTIME(${EPOCH},%Y-%m-%d)}${STRFTIME(${EPOCH},%H:%M:%S)})
exten => 1,4,Set(OUTPUT=/var/spool/asterisk/monitor/${FILENAME})
exten => 1,5,System(/bin/mkdir -p /var/spool/asterisk/monitor/${DIRNAME})
exten => 1,6,MixMonitor(${FILENAME}.wav,b)
exten => 1,7,Dial(SIP/210,30,tm)
exten => 1,8,Voicemail(210,u)
exten => 1,9,Hangup()

exten => 2,1,Answer
exten => 2,2,Set(DIRNAME=${STRFTIME(${EPOCH},%Y-%m-%d)})
exten => 2,3,Set(FILENAME=${DIRNAME}/220_${CALLERID(num)}${STRFTIME(${EPOCH},%Y-%m-%d)}${STRFTIME(${EPOCH},%H:%M:%S)})
exten => 2,4,Set(OUTPUT=/var/spool/asterisk/monitor/${FILENAME})
exten => 2,5,System(/bin/mkdir -p /var/spool/asterisk/monitor/${DIRNAME})
exten => 2,6,MixMonitor(${FILENAME}.wav,b)
exten => 2,7,Dial(SIP/220,30,tm)
exten => 2,8,Voicemail(220,u)
exten => 2,9,Hangup()

exten => 3,1,VoiceMail(300,u)
exten => 3,2,Hangup()

;##################################################################

; Cette partie permet de gerer les appels entre postes 2XX ( exemple 210 et 220 ) connectes sur ce serveur Asterisk
exten => _2XX,1,Wait(1)
exten => _2XX,2,Answer
exten => _2XX,3,Dial(SIP/${EXTEN})
exten => _2XX,4,Hangup()

;##################################################################

[appel-sortant]
; Cette partie gere les appels sortants
exten => X.,1,Set(DIRNAME=${STRFTIME(${EPOCH},%Y-%m-%d)})
exten => X.,2,Set(FILENAME=${DIRNAME}/EXT${CALLERID(num)}${STRFTIME(${EPOCH},%Y-%m-%d)}_${STRFTIME(${EPOCH},%H:%M:%S)})
exten => _X.,3,Set(OUTPUT=/var/spool/asterisk/monitor/${FILENAME})
exten => _X.,4,System(/bin/mkdir -p /var/spool/asterisk/monitor/${DIRNAME})
exten => _X.,5,MixMonitor(${FILENAME}.wav,b)
exten => _X.,6,Dial(SIP/${EXTEN}@anonymous)

2

ks358277CLI> sip show peers
Name/username Host Dyn Forcerport ACL Port Status
210/210 109.25.254.163 D N 5060 OK (49 ms)
220/220 (Unspecified) D N 0 UNKNOWN
anonymous/0033972414892 91.121.129.23 N 5060 OK (4 ms)
3 sip peers [Monitored: 2 online, 1 offline Unmonitored: 0 online, 0 offline]
ks358277CLI>

3

If it is a simple newbie problem, you should be able to find the answer in the documentation.

If you were to ask with a subject that actually summarised the problem, I might point out that you have no NAT settings (nat= is not normally sufficient, and often not necessary).

I might also point out that any version of Asterisk that recognized insecure=very, is obsolete.

Generally, if you are behind NAT, you need one of externip, externaddr, or stunaddr.

If you are using a current version of Asterisk, canreinvite should be directmedia, and the literal equivalent of insecure=very is insecure=port,invite. However, the normal reason for using insecure is better handled by the use of remotesecret, and very people need insecure=port.

In the very latest versions, nat=yes is deprecated.

If insecure=very is not being logged as rejected, you need to install a later version of Asterisk.

4

In order to understand your sip.conf file and know what every single option means and avoid headache you got read and study this document several times. svn.digium.com/svn/asterisk/trun … onf.sample

5

Unfortunately, a lot of people want cook book solutions, but those solutoins are often out of date, and often prepared by an ITSP on the basis of minimising their support costs, which generally means being excessively insecure.