I would like an example configuration of sip.conf.
I have a carrier that offers a sip connection without registration, I assume I can use a sip peer definition as I have tried and it works. the issue is that they also offer a soft client that if used they sendd the registration to the asterisk from the same peer address ad this seems to confuse asterisk so to get around it I was able to get a temporary setup on a different source ip address.
does anyone have a configuration that would work?
frank
I am using Asterisk 1.4.11
Im not sure i understand what you are telling me exactly, but it is unlikely that you could use a softphone as well as asterisk on the same sip connection. I would expect them to boot each other off.
Usually you would have Asterisk using the sip connection then connect your softphone to asterisk.
Are you set up for incoming and outgoing calls through this sip provider or just one or the other?
I have two way service.
let me try to explain it a little better.
the provider hs an SBC platform, from that plat form they can setup a trunk profile for the two way service to the Asterisk and they handle the NAT traversal if the asterisk is behind a NAT. They also have a complimentary service for soft client to use their SBC to handle the NAT traversalif the clients are behind a NAT. when the client sends the registration they 302 the registration to the Asterisk. that being said the Asterisk wuld then receive the registration from the same SBC interface that it uses for the two way service.
Your explanation is more clear now, but I am unfamiliar with this configuration.
Im guessing you do not have a static ip address? If you did you could, of course, forward the sip port to your asterisk box from your router and have Asterisk register directly. Short of that i dont have any ideas. Sorry.
the issue is not with asterisk registering, it is with the soft client sitting externally being able to register with asterisk. the registration would arrive with the same IP address as is defined in sip.conf as a peer.
The only advice i can provide is to remove that funky software from the equation.
I have not had(too much) trouble connecting asterisk to various sip providers and there was never a need for this extra software.
Although… now that i think of it I had the whole server out on DMZ rather than forwarding ports.
example config I am running
general]
defaultexpirey=3600
bindport=5060
bindaddr=0.0.0.0
disallow=all
allow=ulaw
allow=alaw
allow=g729
allow=g723.1
allow=iLBC
srvlookup=yes
dtmfmode=auto
context=inbound
maxexpirey=500
tos_sip=cs3
tos_audio=ef
tos_video=af41
rtptimeout=1800
rtpholdtimeout=86400
rtpkeepalive=1
realm=SipService
callerid=SipService
useragent=SipService
sendrpid=yes
[SIPGW]
insecure=very
type=friend
host=sip.domain.com
[+12342342345SIPGW]
host=sip.domain.com
insecure=very
secret=67f910ad23abdfc99224
type=friend
username=+12342342345
[6000]
mailbox=1
callerid=6000
canreinvite=no
context=custom
dtmfmode=auto
host=dynamic
secret=6000
type=friend
username=6000
[6001]
mailbox=1
callerid=6001
canreinvite=no
context=custom
dtmfmode=auto
host=dynamic
secret=aoc7yV3G
type=friend
username=6001
thanks for looking.
the only issue I have is with the soft clients registering with asterisk and the registration is from the same ip address as the sip gateway defined in the sip.conf. other that that everything is working fine both inbound and out bound call work just fine.