SIP Configuration with DID (no in or outbound)

Asterisk Asterisk Support
1

Hello;

I like to figure these things out on my own, but this is really throwing me a curve ball.

I have installed *now beta 6 and I cannot get the gui based service providers to work with my DID provider (vbuzzer). I even edited the providers.conf file and it seems to register, but I cannot dial inbound and I cannot dial outbound.

I resorted to adding the entries into my sip.conf and extensions.conf file myself. Again, it seems to register although cannot dial in or outbound. I have been searching high and low all over the net with no avail.

Config:
*now is behind a firewall.
DID provider is vbuzzer.com
two ata adapters and softphones to test

For the below entries, I followed the guide in the wiki entry for vbuzzer. Other stuff in extensions.conf exists for the default install of *now.

sip.conf

[b][size=75] [authentication]

[general]
register=username@vbuzzer/416273xxxx
[vbuzzer]
type=friend
user=username
host=vbuzzer.com
port=80
fromdomain=vbuzzer.com
fromuser=username
secret=secret
username=username
insecure=very
context=incoming
authname=username
dtmf=rfc2833
dtmfmode=rfc2833
canreinvite=no
nat=yes
useragent=vbuzzer/1.0
disallow=all
allow=gsm
allow=alaw
allow=ulaw
qualify=200[/size][/b]

extensions.conf
[size=75][incoming]
exten=NXXNXXNXXX,1,Answer()
[outgoing]
exten=_1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@vbuzzer|999|TW)
[globals]
CONSOLE=Console/dsp
IAXINFO=guest
TRUNK=Zap/g2
TRUNKMSD=1
trunksip=SIP/vbuzzer
[numberplan-custom-1]
plancomment=DialPlan1
include=default
exten=_91NXXNXXXXXX,1,Dial(SIP/${EXTEN}@vbuzzer|999|TW)
exten=_91NXXNXXXXXX,1,Macro(SIP/vbuzzer,${EXTEN}@vbuzzer)
exten=_91XXXXXXXXXX.,1,Macro(trunkdial,${}/${EXTEN:1})
comment=_91XXXXXXXXXX.,1,Longdistance,standard
exten=_91700XXXXXXX.,1,Macro(trunkdial,${}/${EXTEN:1})
comment=_91700XXXXXXX.,1,IAXTEL,standard
exten=_9256XXXXXXX.,1,Macro(trunkdial,${}/${EXTEN:4})
comment=_9256XXXXXXX.,1,Local,standard
exten=_9011XXXXXXX.,1,Macro(trunkdial,${}/${EXTEN:1})
comment=_9011XXXXXXX.,1,International,standard
exten=_9XXXXXXX.,1,Macro(trunkdial,${}/${EXTEN:1})
comment=_9XXXXXXX.,1,Local,standard
exten=_911.,1,Macro(trunkdial,${}/${EXTEN:0})
comment=_911.,1,911,standard[/size]

When I call inbound, I hear a message saying that my line is not assigned. But if i connect with my ata (sipura 2002) it works fine (in and out).

When I call outbound, the console shows me this message below:

[size=75]
voip*CLI> sip show registry
Host Username Refresh State Reg.Time
vbuzzer.com:80 username 105 Registered Fri, 07 Sep 2007 23:31:33
– Executing [91416nxxxxxx@numberplan-custom-1:1] Dial(“SIP/600-08222ea0”, “SIP/91416nxxxxxx@vbuzzer|999|TW”) in new stack
[Sep 7 23:32:12] WARNING[17049]: app_dial.c:1106 dial_exec_full: Unable to create channel of type ‘SIP’ (cause 3 - No route to destination)
== Everyone is busy/congested at this time (1:0/0/1)
== Auto fallthrough, channel ‘SIP/600-08222ea0’ status is ‘CHANUNAVAIL’[/size]

In there is also my registration status.
thank you in advance, any help would be appreciated.

Jon S :confused:

2

Try this

Exten => 1234,1,Dial(SIP/somenumber@vbuzzer)
and see what happens. Also all your lines in your configuration should have:

Exten =>

and NOT
Exten = (with no >)

3

Hi Dovid;

I put in the > however, the *now gui removes them when I save the file out. I added them all through SSH just now and it did not help.

Where do I put the line you suggested, in extensions or sip.conf?

10x

4

Try doing this:

Exten => 1234,1,Dial(SIP/somenumber@vbuzzer)

replace somenumber with say your cell phone number and post the CLI here .

Also why are you using port 80 ? SIP’s default port is 5060.

5

where do I put this line?

'Exten => 1234,1,Dial(SIP/somenumber@vbuzzer) "

I am using port 80 because my service provider uses port 80.

sorry for the seemingly simple questions, this has got me.

6

where do i put this line?
Exten => 1234,1,Dial(SIP/somenumber@vbuzzer)

My service provider uses port 80, so I have to.

Sorry for the seemingly simple questions, but this has just got me :confused:

7

here is the outpu

[size=75]-- Executing [1234@numberplan-custom-1:1] Dial(“SIP/6000-0839b818”, “SIP/1416nxxnxxxxx@vbuzzer”) in new stack
[Sep 10 23:32:34] WARNING[3545]: chan_sip.c:2786 create_addr: No such host: vbuzzer
[Sep 10 23:32:34] WARNING[3545]: app_dial.c:1106 dial_exec_full: Unable to create channel of type ‘SIP’ (cause 3 - No route to destination)
== Everyone is busy/congested at this time (1:0/0/1)
== Auto fallthrough, channel ‘SIP/6000-0839b818’ status is ‘CHANUNAVAIL’[/size]

8

I would double check and make sure they are using port 80 for voip. When telneting in to vbuzzer.com on port 80 I get an apache conenction. Switch port=80 port=5060 and see what happens.

9

Also post this in your extensions.conf

Exten => 1234,1,Dial(SIP/18005558355@sipbroker.com)

See if the call goes through. If it does than you know it is an issue with your vbuzzer configs.

10

Ok, the vbuzzer sip port is 5060 but you need port 80 for their proxy. I followed the wiki entry for vbuzzer. The entry specifies that vbuzzer users use port 80.

http://www.voip-info.org/wiki/view/Asterisk+settings+vBuzzer

Now, I can call inbound. However, i get disconnected after about 20 seconds. I cannot call outbound, i get an error on CLI. The outbound error is attached here below.

[size=75]-- Executing [91416xxxxxxx@numberplan-custom-1:1] Macro(“SIP/6000-08383a38”, “trunkdial|SIP/1416xxxxxxx@vbuzzer.com|999|TW”) in new stack
– Executing [s@macro-trunkdial:1] Dial(“SIP/6000-08383a38”, “SIP/1416xxxxxxx@vbuzzer.com”) in new stack
– Called 14165251761@vbuzzer.com
[Sep 13 18:21:59] NOTICE[2373]: chan_sip.c:11906 handle_response_invite: Failed to authenticate on INVITE to ‘“Jonathan” sip:6000@xxx.xxx.xxx.xxx;tag=as784466ae’
– SIP/vbuzzer.com-08388ba8 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
– Executing [s@macro-trunkdial:2] Goto(“SIP/6000-08383a38”, “s-CONGESTION|1”) in new stack
– Goto (macro-trunkdial,s-CONGESTION,1)
– Executing [s-CONGESTION@macro-trunkdial:1] NoOp(“SIP/6000-08383a38”, “”) in new stack
== Auto fallthrough, channel ‘SIP/6000-08383a38’ status is ‘CONGESTION’[/size]

Thank you.