Hello.
Sorry to repost this message but, I don't have the original message in my
inbox nor in my sent box.
Well, last week I posted a problem I am having trying to use an asterisk server
use a voip provider and a pstn. Pstn works fine but, I cant even connect to my
provider's server. I don't know what I'm doing wrong.
I tried using a soft phone and I'm able to register and make calls with it
but, when it comes to rerouting the call through asterisk I not able to
establish a call.
This is my setup:
modem ------ router/firewall -------- LAN
The asterisk server is on the lan side. I have the modem in bridge mode which
assings my router/firewall the external ip address. I have FORWARD to ACCEPT in
the router and I still cant establish a connection.
My sip.conf file looks like this:
[general]
externhost=optimumwireless.com
localnet=172.16.0.0/16
register => username:secret at my.service_provider.tld
language=es
;allow=gsm
allow=all
[voipprovider]
type=friend
host=208.78.163.3
username=username
fromuser=username
secret=password
port=5060
dtmfmode=rfc2833
nat=yes
insucure=port,invite
allow=all
careinvite=yes
I don't know what else to try. When I try to call I get this at the cli:
== Using SIP RTP CoS mark 5
-- Executing [91xxx763xxxx at default:1] Dial("SIP/102-b6a06a40",
"SIP/1xxx763xxxx at voipprovider") in new stack
== Using SIP RTP CoS mark 5
-- Called 1xxx763xxxx at voipprovider
Please help me with this I'm running out of options.
Thanks in advanced for your help.
On Mon, Nov 16, 2009 at 2:40 PM, Landy Landy <landysaccount at yahoo.com>wrote: <snip>> I don't know what else to try. When I try to call I get this at the cli: > > == Using SIP RTP CoS mark 5 > -- Executing [91xxx763xxxx at default:1] Dial("SIP/102-b6a06a40", > "SIP/1xxx763xxxx at voipprovider") in new stack > == Using SIP RTP CoS mark 5 > -- Called 1xxx763xxxx at voipprovider ><snip> We could really use a little more of the CLI output of a failed call. Maybe increase your verbosity to at least 10. Also, what does the SIP debug of a call to the VOIP provider look like (from the cli, type "sip set debug peer voipprovider")? -- Thanks, --Warren Selby http://www.selbytech.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091116/2f0001a9/attachment.htm