AW: [Asterisk-Users] Re: asterisk with german SIPGATE ?

[Thorsten Gehrig]

Hi experts (hope so),

I´ve behind a firewall (Linux FLI4L) - but i have configured all possible
Forwardings.
Two Problems at this time:
a) after many tries I have registered on siptel
*CLI> sip show registry
Host                  Username     Refresh State
217.10.79.9:5060      8003440          120 Registered

I can make calls - but I can´t hear anything!

Asterisk shows:
-- Executing Dial("SIP/thorstengehrig-2641", "SIP/10000 at sipgate1|30") in new
stack
    -- Called 10000 at sipgate1
    -- SIP/sipgate1-dd8a answered SIP/thorstengehrig-2641
    -- Attempting native bridge of SIP/thorstengehrig-2641 and
SIP/sipgate1-dd8a

I think the problem is that the RDP is not coming to the Asterisk?
SIPGate-Website shows me as online!

b) the second Problem is that Phonecalls to Sipgate are not forwarded to my
Asterisk (I cant see anything on the console). (but i´m in  Registered state
and Website shows me online).

I´ve configurated the sip-parts with "qualify=yes" and I recive this
information:
*CLI> sip show peers
Name/username    Host                 Mask             Port     Status
thorstengehrig/  192.168.0.100   (D)  255.255.255.255  5060     OK (2 ms)
sipgate1/800344  217.10.79.9          255.255.255.255  5060     OK (118 ms)


My Router is 0n 192.168.0.1, my Asterisk Server is 192.168.0.105
Here are my portforwardings:
PORTFW_10='5004 192.168.0.105 UDP'	 # Port für SIP (RTP) zum Debian
PORTFW_11='5060 192.168.0.105 tcp'	 # Port für SIP TCP zum Debian
PORTFW_12='5060 192.168.0.105 udp'	 # Port für SIP UDP zum Debian
PORTFW_13='5060 192.168.0.105 tcp'	 # Port für SIP UDP zum Debian
PORTFW_14='5070-5080 192.168.0.105 udp'	 # für das RTP vom SIP-Phone
PORTFW_24='8000-8012 192.168.0.105 udp'  # SIPGATE ?!?!
PORTFW_25='10000-20000 192.168.0.105 udp'# SIP ?!?!

here is my SIP.conf
[general]
port = 5060                     ; Port to bind to
bindaddr = 0.0.0.0              ; Address to bind to
context = default               ; Default for incoming calls
srvlookup = no                  ; Enable SRV lookups on outbound calls
;pedantic = yes                 ; Enable slow, pedantic checking for Pingtel
tos=lowdelay                    ; Type of Service kiwdekaym throughput, ...
;tos=184
;maxexpirey=3600                ; Max length of incoming registration we
allow
registratio
;notifymimetype=text/plain      ; Allow overriding of mime type in NOTIFY
videosupport=yes                ; Turn on support for SIP video
;disallow=all                   ; Disallow all codecs
allow=ulaw                      ; Allow codecs in order of preference
allow=ilbc
allow=gsm
nat=no
localnet = 192.168.0.0
localmask = 255.255.255.0

register => 8003440:passwd at sipgate.de/8003440

[sipgate1]
type=friend
username=8003440
secret=passwd
host=sipgate.de
nat=no
canreinvite=no
fromuser=8003440
fromdomain=sipgate.net
qualify=yes
dtmfmode=rfc2833

im open for any hints!

regards
thorsten



-----Ursprüngliche Nachricht-----
Von: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com] Im Auftrag von nicolas
Gesendet: Sonntag, 9. Mai 2004 12:43
An: asterisk-users at lists.digium.com
Betreff: [Asterisk-Users] Re: asterisk with german SIPGATE ?

Hi,

whats is your problem ?
For me it works, but had problems too.

nicolas