[fthorsby]

I hope someone could point me in the right direction. I have successfully, (I think) installed Asterisk PBX, upgraded to the latest CVS version, have subscribed to Voicepulse regular acccount with a Sipura SPA-2000, purchase a SAP-3000 and a Voicepulse Open Acess Account.
I followed the Voxilla How-To, by Dorian Grey, for the Asterisk, Sipura 2000 and Voicepulse Connect, attempting to amend the configs for my Openaccess

Now for my problem, I can get dial tone to each of my extension phones, one off line 2 of SPA-2000 and one off Line 1 of SPA3000. I can dial out from FWD, But when I try to dial out, using 6 as in the example, I just get a busy signal. My Asterisk log states that it can't find the host voicepulse@8102230327 and that it can't create the Sip channel.
Can anyone help with my configs and get this straighented out.

[muppetmaster]

Please post your sip.conf (sans passwds), section of extensions.conf you are dialing out with and the output from the Asterisk CLI when you make a dial attempt. We might be able to spot something with that.

[fthorsby]

Okay, here is my sip.conf, extensions.conf and tail of /var/log/asterisk/messages. It is not very complicated. My FWD and Voicepulse acounts are registering as well as my Sipura SPA-2000 and SPA3000. I basically cut and pasted Dorians How-To

Thanks
Fred Thorsby

[fthorsby]

I don't beleive my sip.conf and extensions.conf were attached, I will just paste them in:
(extensions.conf)

[general]
static=yes
writeprotect=no

; In the the globals section, we define some constants which we can use later
; to make things neater and more efficient.
[globals]
;
;
; First, our two Sipura lines, which we will call extensions 2201 and 2202 in this
; example. Any extension number could be used as long as we are consistent across
; all .conf files as well as in the SPA-2000 configuration screens.
; SIP/ tells asterisk that we are referring to a SIP device or channel.
;
; line 1
;
PHONES1=SIP/2201
PHONES1VM=2201
;
; line 2
;
PHONES2=SIP/2202
PHONES2VM=2202
;
; Second, we need to enter our login for VoicePulse Connect! In your order
; confirmation email from VoicePulse, you should have received information
; that looks something like this:
; Login: 1234567890
;
; So we want to put that login information here:
VOICEPULSEID=vplogin
;
;
; Next, we'll enter our number for Free World Dialup
;
FWDUSERID=488071
;
;
; Now we need to provide our outgoing caller ID information, which can be
; set to whatever we like. VoicePulse Connect! will send it exactly as we specify
; here. So make sure it looks right to you!
;
MYNAME=Fred thorsby
MYPHONE=8102230327
;
;
;
; We'll include a simple macro that takes an extension as its argument,
; connects the caller to the voice mailbox of that extension, and then hangs up after
; playing a couple short messages.
;
[macro-vmessage]
exten => s,1,VoiceMail2(u${ARG1})
exten => s,2,Playback(groovy)
exten => s,3,Playback(goodbye)
exten => s,4,Hangup
;
;
; And also a fairly simple macro for dialing out using VoicePulse Connect!
; (note how we re-use the globally defined constants. slick, eh?).
;
[macro-dialvpconnect]
;
; Here we can set caller ID number and name, if we like
;
exten => s,1,SetCallerID(${MYPHONE})
exten => s,2,SetCIDName(${MYNAME})
;
; Here is where we dial out through VoicePulse Connect! and use a couple
; arguments that must be passed to the macro: ARG1 will be the number we're
; trying to dial (e.g. 12125551212) and ARG2 will be how many seconds to try
; before giving up, e.g. 60
;
exten => s,3,Dial(SIP/${VOICEPULSEID}@access1.voicepulse.com/${ARG1},${ARG2},Tr)
exten => s,4,Hangup
;
;
;
; The dialout context can be included in contexts which should have access
; to an outside line. Normally we would include many different outgoing contexts,
; but for simplicity, we mention only "vpconnect-forced" and "fwd-out" in this case.
;
[dialout]
;
; if someone dials a "6" in front of their number, send out via VoicePulse Connect!
;
include => vpconnect-forced
;
; If someone dials a "7" in front of their number, send to Free World Dialup
;
include => fwd-forced
;
;
; It's "forced" because we require a "6" to be dialed to match this context.
; In fact, it would certainly be possible to set up our dialplan without the
; "forced" leading "6" or "7" so that numbers of a certain length
; (e.g. 5 or 6 digits) dialed out to FWD, and numbers starting with a "1" or
; even specific area codes dialed out to VoicePulse or another provider.
;
[vpconnect-forced]
;
; Dial out on VoicePulse Connect! and wait for 70 seconds for a connect.
; If no connection is made in 70 seconds, jump to the "fastbusy" macro.
; Note that ${EXTEN:1} will be passed as ARG1 of our macro, i.e.
; strip the leading "6" and pass the rest of the number. "70" will
; then be ARG2 of the macro, the dial timeout in #seconds.
;
exten => _61XXXXXXXXXX,1,Macro(dialvpconnect,${EXTEN:1},70)
;
;
[fwd-forced]
; Check to see if the called number starts with a "7" and
; if so, set the call parameters and bounce the call to the
; Free World Dialup SIP server.
;
; NOTE: Calls to unknown users will result in "invalid extension"
; message being played.
;
exten => _7.,1,SetCallerID(${FWDUSERID})
exten => _7.,2,SetCIDName(${MYNAME})
exten => _7.,3,Dial(SIP/${EXTEN:1}@fwd)
exten => _7.,4,Playback(invalid)
exten => _7.,5,Hangup
;
;
; This is the home context. Any phone or device that has access to this
; context will be able to make outgoing calls.
;
[home]
;
; First, we definitely want to include the dialout context,
; so we'll be able to dial out!
;
include => dialout
;
; Next, add an extension for voicemail .
; now if we dial 8, we can check voicemail.
;
exten => 8,1,VoiceMailMain2
exten => 8,2,Hangup
;
;
; Add some more extensions for the two Sipura lines . now
; we'll be able to call one line from the other.
; And if no one answers, it will go to the mailbox for that line.
;
; Sipura line 1
;
exten => 2201,1,Dial(${PHONES1},20,Ttm)
exten => 2201,2,Macro(vmessage,${PHONES1VM})
exten => 2201,3,Hangup
;
;
; Sipura line 2
;
exten => 2202,1,Dial(${PHONES2},20,Ttm)
exten => 2202,2,Macro(vmessage,${PHONES2VM})
exten => 2202,3,Hangup
;
;
; NOTE: it will be important to remember the name of the context
; "from-sip" . later, we will need to use it in sip.conf
;
[from-sip]
; To receive calls inbound from FWD, we set the extension
; to our FWD user ID, in this case 94896
;
; As currently written, incoming calls from FWD will ring
; only line 1 of the SPA-2000. However, changing the "Dial"
; directive to something like this:
; Dial(${PHONES1}&${PHONES2},15,Ttm)
; would cause both lines of the Sipura device to ring
;
exten => 488071,1,Dial(${PHONES1},15,Ttm)
exten => 488071,2,Voicemail2(u${PHONES1VM})
exten => 488071,3,Hangup

(sip.conf)

[general]
;
;
; here we set the context to "from-sip" exactly as in extensions.conf,
; so that incoming calls from FWD can be sent to the Sipura device.
;
context = from-sip
;
; As in iax.conf, specify what codecs we will allow
disallow=all ; Disallow all codecs
allow=gsm
allow=ilbc
;allow=ima-adpcm
allow=ulaw
allow=alaw
;
;
; Here we register our FWD phone number so that when someone calls it,
; we'll be able to receive that incoming call over SIP.
;
register=488071asswd@fwd.pulver.com:5060
register=vploginasswd@access1.voicepulse.com:506 0
;
;
; Next we set up some more info for FWD . this part is what will
; allow us to make outgoing calls over SIP using FWD.
;
[fwd]
type=friend
secret=passwd
username=488071
host=fwd.pulver.com
dtmfmode=inband
;
[vpaccess]
type=friend
secret=passwd
username=vplogin
host=access1.voicepulse.com
;
;
; Here is where we define those two extensions that were mentioned earlier,
; and attach them to the two lines on the SPA-2000
;
; line 1
;
[2201]
type=friend
;
; Although the SPA-2000 can be set to a static IP address, its registration will
; fail unless we set host as dynamic.
host=dynamic
;
; Here, the context is very important! We want to allow access
; to "home", which is where all outgoing calls are made in
; our dialplan.
context=home
;
; This password must match the one we later set in the Sipura device
secret=passwd
;
; This is the caller id that will show up if we call line2 from line1.
callerid="SPA1" <2201>
;
; If the voice mailbox specified here has new messages,
; this line will have a stuttered dialtone when we pick up the phone.
mailbox=2201
;
; Note: dtmfmode=inband will only work with g711, not gsm!
; On the SPA-2000 configuration screen, rfc2833 is called "AVT."
; This does not need to be changed unless Asterisk is having
; trouble recognizing keypad input from our telephone.
dtmfmode=rfc2833
;
; Since our SPA-2000 is only talking locally to our asterisk machine,
; special consideration for NAT (Network Address Translation) is not needed.
nat=0
;
; The configuration of the second line is very similar to the first.
; line 2
[2202]
type=friend
host=dynamic
context=home
secret=passwd
callerid="SPA2" <2202>
mailbox=2202
dtmfmode=rfc2833
nat=0


Fred Thorsby

[sk3-483]

Making SIP work through NAT is a pain. I use VoicePulse Connect via IAX2. IAX2 is much easier to pass through NAT with only one open port.

Moreover, with VPC you pay a low per-minute fee rather than a flat monthly rate. If you're not using 1000+ minutes a month then VPC is cheaper.

Michael

[FamilyGuy]

Originally Posted by sk3-483

Making SIP work through NAT is a pain. I use VoicePulse Connect via IAX2. IAX2 is much easier to pass through NAT with only one open port.

Moreover, with VPC you pay a low per-minute fee rather than a flat monthly rate. If you're not using 1000+ minutes a month then VPC is cheaper.

Michael

Plus they utilize six second billing. I have a Local Unlimited Account +200 with VP and regualrly log just over 300 "full minutes" and have yet to be charged for any overage thanks to the six second billing.