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Asterisk, DID incoming call trouble

We are working with Asterisk and we have some problems to receive call using DID numbers. When we are calling the did number Asterisk do not see the incoming calls. Nothing happens. We have checked on the voip server and we get the calls but Asterisk is not taking the calls.

Please find below the conf files:

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,Dial(SIP/201,30) 
e    xten => s,4,Hangup(16) 

[outgoing_calls]
exten => _X.,1,Dial(SIP/${EXTEN:1}@forfait-ovh)

SIP.conf
[general]
context=forfait-ovh
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
register => login:password@sip5.5voip.be
registerattempts=0
registertimeout=3600

[201]
type=friend
username=201
callerid="201" <3223315331>
secret=201
host=dynamic
context=appel_sortant
language=fr
insecure=port
nat=yes
canreinvite=no
dtmfmode=auto
video=no
restrictcid=no
amaflags=default
mailbox=201@mondomaine.com

[202]
type=friend
username=202
callerid="202" <3223315331>
secret=202
host=dynamic
context=appel_sortant
language=fr
insecure=port
nat=yes
canreinvite=no
dtmfmode=auto
video=no
restrictcid=no
amaflags=default
mailbox=202@mondomaine.com

[forfait-ovh]
type=peer
host=sip5.5voip.be
context=ovh_sip
language=fr
insecure=port,invite
username=3223315331
secret=5telecom
nat=yes
canreinvite=no
dtmfmode=auto
video=no
restrictcid=no
amaflags=default 

The sip account is not registered all the time , when we type "show sip registry" it is always empty.

Thank you in advance for your answer.

Try adding "qualify=yes" to your SIP trunk options for [forfait-ovh] above, and (as payne says) turn on SIP debugging.

In addition, just to be sure: you say you're familiar with VoIP so perhaps this is a dumb suggestion, but ...

I guess your Asterisk server is behind a firewall right? So I imagine that you've done all the sensible and necessary steps to allow an externally originated call to be delivered to your Asterisk box, by setting up the necessary port forwarding rules on your router/firewall?

There is an overview here: http://www.voip-info.org/wiki/view/port+forwarding and also here: http://forums.whirlpool.net.au/archive/679361

In fact there are loads of such things ...

If Asterisk is not registered with your VOIP provider, then your VOIP provider won't send any calls to you. That's the whole point of registration; it's in effect saying, "I'm here to take calls!"

I'd figure out why you are not registering. Double check your login and password information. Make sure you don't have a firewall or router in the middle that's preventing registration. Ask your VOIP provider if they have any debugging information they can share with you on registration attempts. Turn on Asterisk verbose logging ('core set verbose 100') and see if you are getting any useful messages regarding registration failures.

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