Trunk between, Call manager <-> Asterisk

Asterisk Asterisk Support
1

Hello guys,

I have an asterisk connect to my SIP provider, all the calls work fine, outbound and inbound
I made a trunk between my Cisco Call manager and my asterisk
for made outbound call with my Call manager.
the calls inbound works fine asterisk well redirect the calls to my call manager
but I can’t make a call outbound with my call manager the log said :

== Using SIP RTP CoS mark 5 -- Executing [0688XXXXX@appel-sortant:1] Dial("SIP/callman-00000004", "SIP/0688XXXXX@forfait-ovh,60") in new stack == Using SIP RTP CoS mark 5 -- Called SIP/0688XXXXX@forfait-ovh [Jun 19 16:10:49] WARNING[5196]: chan_sip.c:20366 handle_response_invite: Received response: "Forbidden" from '"8006" <sip:8006@172.30.X.XX>;tag=as2a71e178' == Everyone is busy/congested at this time (1:0/1/0) -- Auto fallthrough, channel 'SIP/callman-00000004' status is 'CONGESTION'

there is my configuration :

sip.conf :

[quote][general]
context=forfait-ovh
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
register => 033972xxxx:xxxxx@siptrunk.ovh.net

[9008]
type=friend
username=9008
callerid=“9008” <033972xxxx>
secret=azerty
host=dynamic
context=appel-sortant
language=fr
insecure=port
nat=no
canreinvite=no

[forfait-ovh]
type=peer
host=siptrunk.ovh.net
context=ovh-sip
language=fr
insecure=port,invite
username=033972xxxx
secret=xxxxx
nat=yes
canreinvite=no
externip=

[callman]
type=friend
context=appel-sortant
host=172.30.X.XX
transport=udp
insecure=port,invite
nat=no
canreinvite=no
;qualify=yes[/quote]

extensions.conf

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

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

[ovh-sip] ; nom du plan
exten => s,1,Dial(SIP/8007@callman,10,Tt)

[appel-sortant]
exten => _8XXX,1,Dial(SIP/${EXTEN}@callman,10,Tt)
exten => _9XXX,1,Dial(SIP/${EXTEN},10,Tt)
exten => 700,1,VoiceMailMain(${CALLERID(name)}@voicemail)
exten => 9000,1,Goto(ivr,s,1)
exten => _0.,1,Dial(SIP/${EXTEN}@forfait-ovh,60)

exten => _9XXX,2,VoiceMail(${EXTEN}@voicemail) [/quote]

2

This will be a call manager configuration problem.

Note you have the normal sorts of sub-optimal settings/obsolete names for insecure, type and probably nat, but none of these are likely to cause the problem.

3

what type of problem, can you be more specific ?

4

type=peer usually causes less problems.

canreinvite is now called directmedia

Most people don’t need insecure=port, and remotesecret is a cleaner way of doing it in the latest versions.

nat= is for working round poor NAT implementation, particularly when the NAT is on the far end. In many cases it is not needed for simple NAT cases.

These make the system less secure than it need be, more fragile, and canrinvite, will eventually stop being recognized.

5

I mean on the call manager ?
I know I have obsolete parameters

6

I don’t deal with configuring CUCM, but the diagnostic clearly indiates that it is CUCM that is unhappy.

7

Yeah but when I replace the CUCM with an Asterisk (just for try)
(So Asterisk1 —trunk—> asterisk2 ----trunk----> sip provider)
when I try to make an outbound call from asterisk1 I have the same error on asterisk2

8

Please do “sip set debug on”. Also please provide the configuration on the second Asterisk.