Asterisk do not recognize that the called party picked up


My Asterisk works fine so far exept one case.
Out- and incomming calls from CAPI (chan_capi) and incomming
calls from are working fine.
But if i want to make an outgoing call via it takes about 5-10 sec. at the beginning of the call until Asterisk prompts

even though the called party already picked up. Within this time the called party and i can´t hear anything.

Because my Asterisk is behind a NAT i put the Machine into DMZ and disabled all Firewall option of my Linksys WRT54G (2.2). This did not help (nothing changed). Because of this it seems not to be a NAT problem!?

Do someone know what the reason for this can be?

Thanks, Mike.

Like you said, looks like a NAT problem. The RTP traffic is probably not getting to your box. tcpdump and/or ethereal can give you an idea of what is going on.

I´ve set RTP ports in rtp.conf (10000-10005) and activeted port-forwarding (5060 UDP/TCP and 10000-10005 UDP/TCP), but i can’t find RTP port-porwarding on my router!?
Sorry, but im not that firm with routing. But if i put the machine into DMZ is there still port forwarding active?

Do you give it a unique IP when you put it in the DMZ?

What do you mean with unique IP?

The IP of the * Machine is With this IP NAT ist still active because of the non offical IP, shure?

I´ve downloaded etherreal now, but i cant see the Traffic of the * Machine. My Router seems not to bring the packages to the port my WIN-Notebook is attached to. I only can see traffic to and from the Notebook.
I will try to setup etherreal directly on the * machine.

Now i´ve put a hub into my network to hear all the traffic. here are the results.

Start Calling -> - SIP/SDP - INVITE ... m=audio 10000 -> - SIP - 407 Proxy Authentication Required -> - SIP/SDP - ACK -> - SIP/SDP - INVITE ... m=audio 10000 -> - SIP - 100 trying -- your call is important to us ... -> - SIP - 183 Session Progress

then i recive a few packages like -> - RTP - Payload type =ITU-T G.729 ....

in between are some packets like -> - RTP - Payload type =ITU-T G.729 ....

But neither me nore the called party can hear anything.

at about that time the called party pick up there are three packes -> - SIP/SDP - 200 OK ...Record Route: <sip:xyz@>Record Route: <sip:xyz@> -> - SIP - ACK sip:xyz@ -> - MDNS - Standrad Query A sipgate.local

Seem that the problems is here, but i don’t know why the * Machine is looking for sipgate.local.
After that a few hundred RTP packages are received from to and the again -> - MDNS - Standrad Query A sipgate.local

later on there is a package -> - SIP - Request: OPTIONS

and about 50 packages later -> - SIP - Status: 482 Loop Detected ... received= "asterisk" <sip: asterisk@> ...

What is this?
About 170 Packages later the called party can hear me an vice versa and asterisk shows

SIP/... answered SIP/...
Attempting native bridge of SIP/... and SIP/...

Any idea?

What does your extensions.conf look like for calls going out through sipgate?

Here is my sip.conf

register => [SIPID]:[PASSWORD]

callerid="Mike Gahn" <2762720>


and here my extensions.conf

MIKE => SIP/2762720

exten => 2762720,1,SetLanguage(de)
exten => 2762720,2,Dial(${MIKE},20)
exten => 2762720,3,VoiceMail,u2762720
exten => 2702720,4,Congestion
exten => 2762720,102,VoiceMail,b2762720

include => capi-in

include => mike-capi-out
include => mike-voicemail
include => mike-sipgate-out

exten => 1000,1,Ringing
exten => 1000,2,Wait(2)
exten => 1000,3,VoicemailMain,s2762720

exten => _9.,1,SetLanguage(de)
exten => _9.,2,Ringing
exten => _9.,3,Dial(CAPI/2762720:${EXTEN:1})
exten => _9.,4,Congestion
exten => _9.,5,Wait(5)
exten => _9.,6,Hangup

exten => _8.,1,Dial(SIP/${EXTEN:1}@sipgate-mike,60,tT)
exten => _8.,2,Congestion
exten => _8.,3,Busy
exten => _8.,4,Hangup


In between i was thinking about -> - MDNS - Standrad Query A sipgate.local 

My Machine-Name is linuxhome.local and in “DNS and Hostname”-Settings
i´ve put my routers-ip as nameserver and as domain-search value “local”.
Now i´ve changes domain-search value to “de” and everything works fine. i have´nt checked with etherreal now, but be shure that the package is now -> - MDNS - Standrad Query A

But this must be a hack only, or? what if i would like to make phonecalles with to put the tld into domain-search value cant be the right thing.

what do i have to put into domain-search value?

I can solve the problem too, if i change sip.conf

[sipgate-mike] -> []

i thought the value HOST will specify the target!?