* You are viewing the archive for February 9th, 2011

Unable to make outgoing calls with Internode

Surely there must be someone here who can help me with this problem.

I have spent weeks trying to get this damned service to work with no
luck. I have incoming calls working, but no outgoing. If get outgoing
working then incoming don’t work.

I have sent this problem to this list a couple of times with little or
no response, and I _really_ need some help to sort it out.

I have an asterisk 1.8 server running on FreeBSD 8.1, and another
FreeBSD 8.1 running as a firewall/gateway with PF.

I have a nodephone service with Internode (who have been absolutely
useless in helping me- they point blank refuse, or they say to open
everything right up to their server; which didn’t wok anyway btw).

I have been running endless tests on settings changes, tcpdumps on both
the firewall and asterisk, and hours poring over SIP rfc’s. I’ve only
managed to get a headache…

I have tried following best practices, worst practices, and still
nothing works.

My sip.conf looks like this:

[general]
context = default
bindport = 5060
bindaddr = 0.0.0.0
srvlookup = yes
allow = all
;allow = t140red
textsupport = yes
videosupport = yes
;allow = h263
maxcallbitrate = 384
register => sip-in? number>:@sip.internode.on.net/ externip =
localnet =
canreinvite = no
hasvoicemail = no
qualify = yes
nat = no
;rtptimeout = 120
rtpkeepalive = 5
;ignoresdpversion = yes
;directmediapermit =

[sip-in]
type = peer
host = sip.internode.on.net
context = internode-incoming
;externip =
;domain = internode.on.net,internode-incoming
;fromdomain = sip.internode.on.net
;fromuser = ;username = ;secret =
;auth = :@BroadWorks
;insecure = invite,port
;register => :@sip.internode.on.net
;nat = never
qualify = yes
canreinvite = no
;expire = 240

[sip-out]
type = peer
host = sip.internode.on.net
context = internode-outgoing
externip =
;username = fromuser = ;fromdomain = internode.on.net
;secret =
;qualify = yes
canreinvite = no
;auth = :@BroadWorks
;nat = never
;pedantic = yes
;insecure = invite,port
;ignoresdpversion = yes
;compactheaders = yes

As you can see I’ve tried lots of settings. It registers and peers with
the provider, but no outgoing. The provider can call me though.

In extensions.conf:

[internode-outgoing]
exten => _X.,1,Dial(SIP/${EXTEN}@sip-out)
exten => _X.,n,Answer(2)
exten => _X.,n,Playback(ss-noservice)

With debugging enabled, verbose 9, debug 9:

SIP Debugging enabled

<--- SIP read from UDP::5060 —>
INVITE sip:0871271201@ SIP/2.0
Via: SIP/2.0/UDP :5060;branch=z9hG4bK-78cdde11;rport
From: @ server>>;tag=600053496208a4a8o1
To: >
Call-ID: e2895c9d-55b90b64@
CSeq: 101 INVITE
Max-Forwards: 70
Contact: @:5060>
Expires: 240
User-Agent: Linksys/PAP2T-3.1.15(LS)
Content-Length: 446
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp

v=0
o=- 5330142 5330142 IN IP4
s=-
c=IN IP4
t=0 0
m=audio 16436 RTP/AVP 0 2 4 8 18 96 97 98 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/— (14 headers 20 lines) —
Sending to :5060 (no NAT)
Using INVITE request as basis request – e2895c9d-55b90b64@
Found peer ‘‘ for ‘‘ from :5060

<--- Reliably Transmitting (no NAT) to :5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP :5060;branch=z9hG4bK-78cdde11;received= ata ip>;rport=5060
From: Skinner’s Home @ server>>;tag=600053496208a4a8o1
To: >;tag=as6957dfb9
Call-ID: e2895c9d-55b90b64@192.168.0.196
CSeq: 101 INVITE
Server: Asterisk PBX 1.8.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=”asterisk”, nonce=”12eb6973″
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘e2895c9d-55b90b64@‘ in
6400 ms (Method: INVITE)

<--- SIP read from UDP::5060 —>
ACK sip:0871271201@ SIP/2.0
Via: SIP/2.0/UDP :5060;branch=z9hG4bK-78cdde11;rport
From: @ server>>;tag=600053496208a4a8o1
To: >;tag=as6957dfb9
Call-ID: e2895c9d-55b90b64@
CSeq: 101 ACK
Max-Forwards: 70
Contact: @:5060>
User-Agent: Linksys/PAP2T-3.1.15(LS)
Content-Length: 0

<------------->

AEL Eswitches

On Wednesday 09 February 2011 13:31:36 Thiago Maluf wrote:
> Hi List,
>
> Would someone can to explain me the main difference in SWITCHES or
> ESWITCHES in AEL.
>
> context default {
> switches {
> DUNDi/e164;
> IAX2/box5;
> };
> eswitches {
> IAX2/context@${CURSERVER};
> };
> };

A switch evaluates variables at load time. An eswitch evaluates variables
locally at query time. An lswitch sends the variables intact through to
the switch backend.

Error loading module ��Է�Vi.so

Just recently I noticed that my Asterisk 1.8 server is giving the
following error at startup:

[Feb 9 17:48:56] WARNING[7968]: loader.c:387 load_dynamic_module: Error
loading module ‘��Է�Vi’: /usr/lib/asterisk/modules/��Է�Vi.so: cannot
open shared object file: No such file or directory

I have checked the modules directory and there are no files with
strange characters that I can see. The only extra modules I have are
the G729 codec and Lumenvox speech recognition. Both are loaded and
working. All expected functionality for Asterisk is working so I really
do not know what that module may be. Any ideas?

Defining what an extension should do after the Dial() command returns busy.

We have a customer who wants to forward an extension to their cell phone,
if and only if that extension is “unavailable”, or when the Dial() command
times out. However, should the Dial() command return “busy” it should go
to voicemail instead.

As far as I know, the dialplan doesn’t support this. Certainly not
natively or in any particularly easy or obvious way, and I can’t find
anything on voip-info.org to suggest that there is.

queue called by agi doesn’t re-enter the script

http://www.voip-info.org/wiki/view/Asterisk+cmd+DeadAGI

read the part at the bottom about ignoring sighup, if you’re using a
later version i think there is a option like agisighup that you can
use in the dialplan

SIP MESSAGE outside calls – state of the art?

I have a mobile phone (UTStarCom GF-210) that uses SIP MESSAGE to send
“SMS” messages over VoIP. My Asterisk 1.4 installation drops these
messages and returns a failure condition to the phone:

[Feb 9 10:17:22] WARNING[11960]: chan_sip.c:9859 receive_message: Received message to from “Display Name” ;tag=87739132, dropped it…
Content-Type:text/plain; charset=UTF-8
Message: test message

(Packet trace shows a SIP MESSAGE, answered by a 405.)

…and apparently is unable to originate them either; SendText, which
looks as though it ought to be the right way to send them, produces (in
the context of a call, since I can’t send the message outside one):