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

nicolas albers at na-computer.de
Sun May 9 09:24:25 MST 2004


Hi,

so i would do following:

1. you type your external ip into the sip.conf externip=x.x.x.x
2. nat=yes

Your forwarding are ok, think you need udp only : 5060 and 10000-20000 see
it in your rtp.conf.
And you need open your ports to.

should work.
nico


Thorsten Gehrig wrote:

> 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
> 
> 
> 
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