Has anybody managed to get a SIP account from sipgate to work with Asterisk, i have set up a SIP trunk but cant get it to register with sipgates server.
Works for me (Asterisk 1.2.4).
Sourced the Internet and followed the tracks of a blogger
Did you use the GUI or command lines ?
I configured directly in the .conf files, so I believe this counts as command line.
If the GUI you are using correctly reflects the configuration options of the relevant .conf files you should be allright. Here are excerpts of my configuration with sipgate.de in /etc/asterisk/sip.conf
[code][general]
context=incoming ; Default context for incoming calls
bindport=5060 ; UDP Port to bind to
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
disallow=all ; First disallow all codecs
allow=ulaw ; Allow codecs in order of preference
language=us ; Default language setting for all users/peers
nat=no ; NAT settings
register => 1234567:ABCDEFG@sipgate.de/1234567
externip = 192.168.1.1 ; Address that we’re going to put in outbound SIP messages
localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks
canreinvite=no
tos=0x 18
[SIPGATE]
type=friend
username=1234567 ; phone number here
fromuser=1234567 ; phone number here
secret=ABCDEFG ; password, from sipgate user panel on website
language=de
qualify=yes
nat=no
host=sipgate.de
fromdomain=sipgate.de
insecure=very
canreinvite=no
context=incoming
disallow=all
allow=alaw ; gsm
tos=0x 18
dtmfmode=info
[/code]
I receive calls through settings in the dialplan /etc/asterisk/extensions.conf
exten => 1234567,1,Set(LANGUAGE()=de) ; Announcements in German for da family
exten => 1234567,n,Macro(stdexten,100,SIP/JP-O&SIP/JP-W)
with JP-O & JP-W internal SIP phones I want to ring simultaniously.
I place calls to Germany as follows (again extensions.conf):
TRUNK_GER=SIP ; Variable: VoIP SIP
exten => _901149.,1,Dial(${TRUNK_GER}/0${EXTEN:6}@SIPGATE,60,tT) ; SIPGATE - Germany only
exten => _901149.,n,Hangup()
On internal SIP phones, I dial 9 to “get a line” and US-style country code prefix 01149 to get to Germany. All that is my decision. The concatinated prefix 901149 could be anyhting else which would not overlap with other dialplan prefixes.
So after punching in settings in the GUI, I assume you should be able to verify your settings on the command line with:
I’ve never work with any of the derivates (Asterisk@home) and GUIs so the location of these configuration files might be somewhere else.
Thanks for the replie, i will have a look at the things you mentioned at the moment iam trying to get a sipgate number to be answered with the IVR. For some reason i cant get sipgate to see the server.
long night ahead
Not sure if it’s relevant to your problem but on 5th Dec I was having probs with incoming calls from Peers. I was told by Sipgate support…
"We are about to fix the connections from our listed peering partners:
sipgate.co.uk/user/tarife.php?show=6
What we don’t accept any more (for months) are direct IP calls by sip-uri."