[Asterisk-Users] outbound PSTN numbers over SIP failing

[asterisk]

Hi, 

 

I am currently trying out the asterisk at 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 at 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 at budgetphone.nl 

May 8 10:47:11 DEBUG[1563]: (Provisional) Stopping retransmission (but
retaining packet) on '1f6df4381299d2161fc94ea9202acb42 at 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 at 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 at 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 at 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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