On 13/09/12 11:16, Olivier wrote:
2012/9/13 Benedikt Schöffmann <benedikt.schoeffm...@gmail.com
<mailto:benedikt.schoeffm...@gmail.com>>
Hi there,
I'm setting up a Asterisk network and I ran into some problems ...
as you might have guessed :)
The set up is like this:
Internal Communication in the company should be handled through
softphones over an asterisk server (works).
Outbound Communication should be handled through a HUAWEI E169
stick, accessed by the chan_dongle project.
http://code.google.com/p/asterisk-chan-dongle/
When I call internal numbers, everything works fine, but when I try
to access outside, I get the following error:
== Using SIP RTP CoS mark 5
-- Executing [06766770031@internal:1]
Answer("SIP/1001-00000023", "") in new stack
-- Executing [06766770031@internal:2] Dial("SIP/1001-00000023",
"dongle0/r1/06766770031,20,r") in new stack
[Sep 13 11:33:31] WARNING[9835]: channel.c:5603 ast_request: No
channel type registered for 'dongle0'
[Sep 13 11:33:31] WARNING[9835]: app_dial.c:2218 dial_exec_full:
Unable to create channel of type 'dongle0' (cause 66 - Channel not
implemented)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [06766770031@internal:3]
Hangup("SIP/1001-00000023", "") in new stack
== Spawn extension (internal, 06766770031, 3) exited non-zero on
'SIP/1001-00000023'
From googling my way around, I know this type of error normally
relates to a module not being loaded, but chan_dongle.so shows up
when I type a "module show". I've been fiddling around with this for
days and frankly I don't really know where the problem could lie.
Below are excerpts from sip.conf and extensions.conf
SIP.conf
<code>
[general]
bindport = 5060
bindaddr = 192.168.61.25
tcpbindaddr = 192.168.61.25
tcpenable = yes
context = internal
transport = udp
disallow = all
allow = gsm
allow = ulaw
allow = alaw
[dongle0]
type=friend
context=internal
audio=/dev/ttyUSB1
data=/dev/ttyUSB2
imei=359638011610601
imsi=232018830482446
transport=udp
disallow = all
allow = gsm
allow = ulaw
allow = alaw
[1000]
type=friend
callerid = "Benny" <1000>
secret=1000
host=dynamic
canreinvite=no
dtmfmode=rfc2833
mailbox=1000
disallow=all
allow=gsm
allow=ulaw
allow=alaw
transport=udp
context=internal
[1001]
type=friend
callerid = "Timme" <1001>
secret=1001
host=dynamic
canreinvite=no
dtmfmode=rfc2833
mailbox=1001
disallow=all
allow=gsm
allow=ulaw
allow=alaw
</code>
Extensions.conf
<code>
[internal]
; for 4-digit numbers, assume it's a SIP number in our own context
; call it
exten => _XXXX,1,Answer()
exten => _XXXX,n,Dial(SIP/${EXTEN},20,r)
exten => _XXXX,n,Hangup
; else
; for a number starting with zero try to call via Dongle
exten => _0X.,1,Answer()
exten => _0X.,n,Dial(dongle0/r1/${EXTEN},20,r)
exten => _0x.,n,Hangup
</code>
Please shed some light on this .....
Kind regards,
Benedikt
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I've never tried chan_dongle, but to me, the Dial statement is incorrect.
Maybe the following would be better:
exten => _0X.,n,Dial(dongle/dongle0/r1/${EXTEN},20,r)
Looking at the chan_dongle documentation, it looks like you need to have
a dongle.conf in /etc/asterisk. Do you have it and does it contain the
right stuff? It looks like some of the stuff you've added to the
sip.conf should really go in dongle.conf, at least according to this page:
http://wiki.e1550.mobi/doku.php?id=configuration
Actually, I'm not sure you should have any settings connected with the
dongle in sip.conf - as SIP and dongle are different channel types and
use different configuration files.
According to the examples on the same page, your Dial string should not
include the name of the device, but the channel type, more like:
exten => _0X.,n,Dial(Dongle/r1/${EXTEN},20,r)
Also, what do you get when you run in Asterisk CLI:
dongle show devices
That should give you idea if the dongle is setup correctly in dongle.conf.
Hope the above helps,
Sebastian
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