Hi,
I am currently trying out the asterisk@home (version 1) release of
Asterisk, and I want to configure it as follows:
Calls from regular telephony network (PSTN) come in through my VoIP
provider over SIP and outgoing calls to the PSTN should be routed
through the ViOP provider onto the PSTN network. I thus have no direct
PSTN connection, but only a SIP connection.
Incomming calls work fine. No problems with this.
I have found numerous documents (on this list & on the voip-info.org
website) that desribe how to dial out to PSTN numbers via the VoIP
provider (even the one I'm using), but none of them work for me, or
better yet, I haven't been able to get them up and running, most likely
as I don't really know where to start.
In the log file, it states the following:
May 8 10:47:11 VERBOSE[1563]: -- Executing
[1;36;40mSetCallerID[0;37;40m("[1;35;40mSIP/200-8f7f[0;37;40m",
"[1;35;40m31437110323[0;37;40m") in new stack
May 8 10:47:11 VERBOSE[1563]: -- Executing
[1;36;40mSetCIDName[0;37;40m("[1;35;40mSIP/200-8f7f[0;37;40m",
"[1;35;40m31437110323[0;37;40m") in new stack
May 8 10:47:11 VERBOSE[1563]: -- Executing
[1;36;40mSetCIDNum[0;37;40m("[1;35;40mSIP/200-8f7f[0;37;40m",
"[1;35;40m31437110323[0;37;40m") in new stack
May 8 10:47:11 VERBOSE[1563]: -- Executing
[1;36;40mDial[0;37;40m("[1;35;40mSIP/200-8f7f[0;37;40m",
"[1;35;40mSIP/XXXXXXXXXX@budgetphone.nl|30|r[0;37;40m") in new stack
May 8 10:47:11 DEBUG[1563]: SIMPLE DIAL (NO URL)
May 8 10:47:11 DEBUG[1563]: Outgoing Call for XXXXXXXXXX
May 8 10:47:11 DEBUG[1563]: XXXXXXXXXX is not a local user
May 8 10:47:11 VERBOSE[1563]: -- Called XXXXXXXXXX@budgetphone.nl
May 8 10:47:11 DEBUG[1563]: (Provisional) Stopping retransmission (but
retaining packet) on '1f6df4381299d2161fc94ea9202acb42@192.168.123.151'
Request 102: Found
May 8 10:47:11 DEBUG[1563]: Acked pending invite 102
May 8 10:47:11 DEBUG[1563]: Stopping retransmission on
'1f6df4381299d2161fc94ea9202acb42@192.168.123.151' of Request 102: Found
May 8 10:47:11 WARNING[1563]: Forbidden - wrong password on
authentication for INVITE to '"31437110323" ;tag=as01c07be8'
May 8 10:47:11 VERBOSE[1563]: -- SIP/budgetphone.nl-25eb is circuit-busy
May 8 10:47:11 DEBUG[1563]: update_user_counter(XXXXXXXXXX) - decrement
outUse counter
May 8 10:47:11 DEBUG[1563]: XXXXXXXXXX is not a local user
May 8 10:47:11 VERBOSE[1563]: == Everyone is busy/congested at this time
May 8 10:47:11 DEBUG[1563]: Exiting with DIALSTATUS=CONGESTION.
XXXXXXXXXX is the number I've dialed, deleted for security reasons.
So I get a bad password, even though registering with my SIP provider
using that password does not fail. I think that, when dialing out, no
authentication is sent to my SIP Provider, but how do I integrate this
in my call. Above all, I have found several articles on the internet
stating this WARNING[1563], but they all have more information after the
INVITE than I do.
Below you can find part of my extensions.conf file:
[outrt-001-9_outside]
exten => _XXXXXXXXXX,1,SetCallerID(31437110323)
exten => _XXXXXXXXXX,2,SetCIDName(31437110323)
exten => _XXXXXXXXXX,3,SetCIDNum(31437110323)
exten => _XXXXXXXXXX,4,Dial(SIP/0${EXTEN:1}@budgetphone.nl)
;exten => _XXXXXXXXXX,5,Playback(invalid)
exten => _XXXXXXXXXX,5,Hangup()
[from-sip-t2y]
;exten => 31437110323,1,Dial(SIP/200,20)
exten => 31437110323,1,Macro(exten-vm,200@default,200)
And of course my sip.conf
[general]
port = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on
machine)
context = from-sip-external ; Send unknown SIP callers to this context
;context = from-budgetphone ; Send unknown SIP callers to this context
disallow=all
allow=ulaw
allow=alaw
allow=g723.1
allow=g726
allow=g729
callerid = Unknown
srvlookup=yes
proxy_register = 1
dtmfmode=inband
register => 31437110323:mypassword@budgetphone.nl/31437110323
#include sip_nat.conf
#include sip_additional.conf
[31437110323]
type=friend
context=from-sip-t2y
;context=from-talkin2ya
host=budgetphone.nl
fromuser=31437110323
fromdomain=budgetphone.nl
username=31437110323
insecure=very
nat=yes
secret=mypassword
qualify=no
port=5060
Question is how to get outbound calling working. If you need more info,
then please let me know.
Thanks for the help
Cheers
Guy
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