[asterisk-users] NAT-troubles with RTP
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi Because it seems my mail from 30th august didn't make it to the list i send it again. If the mail _did_ get to the list and i didn't see it please excuse the duplicate post Below is the mail from the 30th: I have a setup like this: An asterisk-server with SIP-phones on the outside of a NAT. For example: asterisk with local IP-address (on the network interface) 1.2.3.4 and the extern IP-address should be 2.3.4.5 (yes, i'm aware these are no correct IP-addresses. It's only for describing my setup). SIP communication works perfekt. The phones are able to register on asterisk and to place calls on other SIP-phones. But when it comes to speaking, nothing works. A little bit of tcpdumping told me that the phones are trying to send the RTP traffic to 1.2.3.4 instead of 2.3.4.5 because asterisk sends 1.2.3.4 as contact-address in the SIP-header. I thought i enabled everything necessary but something must be still missing. Here is my sip.conf (with changed IP-addresses of course): [general] port = 5060 bindaddr = 0.0.0.0 context = others ;nat specific stuff below externip = 2.3.4.5 canreinvite = no nat = yes qualify = yes ;qos stuff tos_sip = cs3 tos_audio = ef ;;; ;template for voip-testing [testphones](!) context = voiptest type = friend host = dynamic nat = yes disallow = all ;disable all codecs except the ones defined below allow = ulaw allow = g723.1 allow = g729 ;;; [2000](testphones) secret = foobar [2001](testphones) secret = foobar [2002](testphones) secret = foobar Any hints on what i did wrong or forgot? Ah, and asterisk is version 1.2 on a debian etch with kernel 2.6.18 regards Florian - -- Florian Arthofer Technik Web- und Mailservices/Administrator Web- and Mailservices lagis Internet Serviceprovider GmbH Wiener Straße 151, 4021 Linz, Austria Phone +43(0)732/3400-5636 Fax +43(0)732/3400-5644 E-Mail [EMAIL PROTECTED] URL http://www.lagis.at FN 270805 w des Landesgerichtes Linz -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFG3TPXNAbU6R7INwcRAlRTAJ9RyzmD8FOL2cLBC9Fp1LSE/BUgmQCaA2xt YUci7H9wuFGgDPBirAR3FHc= =xmtz -END PGP SIGNATURE- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Strange ISDN Troubles
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Ahoy
Jared Smith wrote:
On Mon, 2007-07-30 at 16:46 +0200, Florian Arthofer wrote:
Shouldn't i see _something_ on the console, even if
the DID which is dialed isn't configured yet?
Unfortunately, I don't think so. You might want to add a pattern match
to your dialplan that would match any DID, and see if that helps.
Ok, i added this to my extensions.conf :
[isdn-in]
exten = _.,1,Dial(SIP/${EXTEN},5)
exten = t,1,Hangup
This should match any number dialed from the outside and hangup after 5
seconds.
In my sip.conf i configured this one:
[123]
context = isdn-in
type = friend
host = dynamic
disallow = all ;disable all codecs except the ones defined below
allow = ulaw
allow = g723.1
allow = g729
secret = X
So, if my ISDN-number is for example 1234567, then i should, if i dial
1234567-123, see _something_ on the console and at least i should hear
it ringing on the phone i place the call with. Am i right?
regards
Florian Arthofer
- --
Florian Arthofer
Technik Web- und Mailservices/Administrator Web- and Mailservices
lagis Internet Serviceprovider GmbH
Wiener Straße 151, 4021 Linz, Austria
Phone +43(0)732/3400-5636
Fax +43(0)732/3400-5644
E-Mail [EMAIL PROTECTED]
URL http://www.lagis.at
FN 270805 w des Landesgerichtes Linz
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[asterisk-users] Strange ISDN Troubles
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Ahoy
I'm trying to setup Asterisk on debian etch (with the debian packages)
with a Fritz!Card PCI ISDN card and chan_capi.
Everything seems to be configured the right way (excerpts below),
Asterisk seems to see the ISDN-card but if i try to place a test-call
from the outside i don't see anything on the asterisk-console (set debug
and verbose to 99). Shouldn't i see _something_ on the console, even if
the DID which is dialed isn't configured yet?
I tested the cable and the NTBA both with a ISDN-phone. This worked
properly. But if i plug the cable into the asterisk-server i just get a
number busy-signal and don't see any messages on the asterisk console
or in the log files.
regards
Florian Arthofer
=exerpts=
capiinfo:
Number of Controllers : 1
Controller 1:
Manufacturer: AVM GmbH
CAPI Version: 2.0
Manufacturer Version: 3.11-07 (49.23)
Serial Number: 101
BChannels: 2
Global Options: 0x0039
internal controller supported
DTMF supported
Supplementary Services supported
channel allocation supported (leased lines)
B1 protocols support: 0x411f
64 kbit/s with HDLC framing
64 kbit/s bit-transparent operation
V.110 asynconous operation with start/stop byte framing
V.110 synconous operation with HDLC framing
T.30 modem for fax group 3
Modem asyncronous operation with start/stop byte framing
B2 protocols support: 0x0b1b
ISO 7776 (X.75 SLP)
Transparent
LAPD with Q.921 for D channel X.25 (SAPI 16)
T.30 for fax group 3
ISO 7776 (X.75 SLP) with V.42bis compression
V.120 asyncronous mode
V.120 bit-transparent mode
B3 protocols support: 0x80bf
Transparent
T.90NL, T.70NL, T.90
ISO 8208 (X.25 DTE-DTE)
X.25 DCE
T.30 for fax group 3
T.30 for fax group 3 with extensions
Modem
0100
0200
3900
1f010040
1b0b
bf80
0101 0002
Supplementary services support: 0x03ff
Hold / Retrieve
Terminal Portability
ECT
3PTY
Call Forwarding
Call Deflection
MCID
CCBS
gertrud*CLI capi show channels
CAPI B-channel information:
Line-Name NTmode state i/o bproto isdnstate ton number
-
ISDN1#02 no- - trans 0x00 ''-''
ISDN1#01 no- - trans 0x00 ''-''
capi.conf:
;
; CAPI config
;
;
; general section
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
language=de ;set default language
;ulaw=yes;set this, if you live in u-law world instead of a-law
; interface sections ...
[ISDN1] ;this example interface gets name 'ISDN1' and may be any
;name not starting with 'g' or 'contr'.
;Use one interface section for each isdn port!
;ntmode=yes ;if isdn card operates in nt mode, set this to yes
isdnmode=msn ;'MSN' (point-to-multipoint) or 'DID' (direct inward dial)
;when using NT-mode, 'DID' should be set in any case
incomingmsn=*;allow incoming calls to this list of MSNs/DIDs, * = any
controller=1 ;capi controller number of this interface/port
group=1 ;dialout group
;prefix=0;set a prefix to calling number on incoming calls
softdtmf=on ;enable/disable software dtmf detection, recommended
for AVM cards
relaxdtmf=on ;in addition to softdtmf, you can use relaxed dtmf
detection
accountcode= ;PBX accountcode to use in CDRs
context=isdn-in ;context for incoming calls
echosquelch=1 ;_VERY_PRIMITIVE_ echo suppression
echocancel=yes
;echocancelold=yes;use facility selector 6 instead of correct 8
(necessary for older eicon drivers)
devices=2;number of concurrent calls (b-channels) on this controller
;(2 makes sense for single BRI, 30/23 for PRI/T1)
=excerpts-end
- --
Florian Arthofer
Technik Web- und Mailservices/Administrator Web- and Mailservices
lagis Internet Serviceprovider GmbH
Wiener Straße 151, 4021 Linz, Austria
Phone +43(0)732/3400-5636
Fax +43(0)732/3400-5644
E-Mail [EMAIL PROTECTED]
URL http://www.lagis.at
FN 270805 w des Landesgerichtes Linz
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