Hi, I have installed Asterisk and it seemed to go well except that i can
not dial out nor in.
This scenario should be plain and simple, but there has to be a small
detail i am missing.
I am trying to call with softphones via Asterisk. Softphone and Asterisk
are behind same firewall. Where SIP/RTP ports are opened.
Dialing begins and i get tone on phone but get strange message back from
my SIP provider.
Both softphone and my account at my local SIP provider are registered on
Asterisk and i do not get any error messages within start of Asterisk.
Message i get in Asterisk in verbose is:
Executing Dial(SIP/2000-cd1a", "SIP/XXXXXXXX@sipprovider|60|r")
in new
stack
Called XXXXXXXX@sipprovider
Got SIP response 404 "Not Found" back from 62.97.243.50
SIP/sipprovider-775a is circuit-busy
Everyone is busy/congested at this time
NOTICE[111335136]: rtp.c:420 ast_rtp_read: RTP: Received packet with bad
UDP checksum
WARNING[111335136]:pbx.c:1933 ast_pbx_run: Timeout, butno rule 't' in
context 'default'
from sip.conf
[general]
context=default
port=5060
bindaddr=0.0.0.0
nat=yes
disallow=all
allow=alaw
allow=ulaw
allow=gsm
register => mylogin:mypass@62.97.243.50/21674999
:21674999 my number, not sure if it should be there
externip = 81.0.162.32
localnet= 192.168.10.0/255.255.0.0
[sip_proxy]
type=friend
context=default
[sipprovdider] :same info as on register
type=peer
:username=21674999 :my nymber from SIP provider, but i assume its not
needed here
fromuser=mylogin
secret=mypass
host=62.97.243.50
dtmfmode=inband
nat=yes
[2000]
type=friend
username=2000
secret=2000
host=dynamic
[2001]
type=friend
username=2001
secret=2001
host=dynamic
from extensions.conf
[general]
static=yes
writeprotect=no
[globals]
CONSOLE=SIP/2000
CONSOLE=SIP/2001
[out]
exten => _XXXXXXX.,1,Dial(SIP/${EXTEN}@sipprovdider,60,r)
[default]
exten => 21674999,1,Dial(SIP/${2000},10,Ttm)
exten => 1,1,Dial(SIP/2000,20,tr)
exten => 2,1,Dial(SIP/2001,20,tr)
include => out
Anykind of help is appreciated
Cin
--
Cinoss
cinosss@f-m.fm
Well i have now sorted dialing out. Only needed to add fromdomain= to my [sipprovider]. Still got small problem with it. The call gets automaticly hang-up after 10secs. I tried both canreinvite=no and yes and my sip.conf but it doesn't seem to do any difference, well other than fail code. -- Cinoss cinosss@f-m.fm
Thanks for reply. Yes i am getting audio. It hangs-up automaticly after 10 secs, or the line goes down. Softphone has the line still open though. I dont get this 404 anymore, it was just before the missing canreinvite -----Original Message----- Cinoss, Are you getting audio during the call? Or are you just seeing the call setup? Is it really 10 seconds? Or just seems like it? I have seen the SIP 404 when the codec matches were incorrect. With Debug and Versobose, -dvvvvvgc, on in Asterisk look for messages about codec matching. Also turn on sip debugging. "*CLI> sip debug" Found description format CN Capabilities: us - 0x10e(GSM|ULAW|ALAW|G729A), peer - audio=0x105(G723|ULAW|G729A)/video=0x0(EMPTY), combined - 0x104(ULAW|G729A) Non-codec capabilities: us - 0x1(G723), peer - 0x3(G723|GSM), combined - 0x1(G723) Urgent handler If "combined" equals empty then you need to adjust the codecs. -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Cinoss Sent: 19 October 2004 13:04 To: asterisk-users at lists.digium.com Subject: [Asterisk-Users] SIP 404 - circuit busy when dialing out Well i have now sorted dialing out. Only needed to add fromdomain= to my [sipprovider]. Still got small problem with it. The call gets automaticly hang-up after 10secs. I tried both canreinvite=no and yes and my sip.conf but it doesn't seem to do any difference, well other than fail code. -- Cinoss cinosss at f-m.fm _______________________________________________ Asterisk-Users mailing list Asterisk-Users at lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Cinoss cinosss@f-m.fm