Asterisk configuration

[akoei]

I have purchased fpl configuration and configured to asterisk. It has been working for a long time, but just started from today:
When I make a outgoing calls, I got a voice prompt "this account number is not valid", but incoming are all OK.

I then tried using a softphone(sjphone), no any problems.

I am wondering if FreePhoneLine change some server settings? Or port to asterisk is NOT allowed any more?

Please help.

Thanks

Shawn

[nicolas_dh]

Hello, we've been happily using our FPL line
through an Asterisk PBX,
for a little more than a month now
with spurious service downs...

But tonight (well a few hours back now)
every out call attempts are responded
by a voice message telling three times:
"This account number is not valid"
and the call gets hanged up with
the following "asterisk logged" message:
-- Got SIP response 603 "Declined" back from xxx.yy.zzz.www
where "xxx.yy.zzz.www" is the ip number
of the Inbound/Outbound Proxy given in the SIP settings

BUT, no change was made to our SIP config for a while
and we WERE able to make outgoing calls with these same
settings few hours before and all the days before...

Our SIP settings still seems to be
in accordance with the "SIP settigs" tab, from our
http://www.freephoneline.ca
account web page... (doesn't seem to have changed either)
That is, ACCOUNT (phone number), SECRET (pass)
seems to be the same as before according to the web page...

Has something changed with SIP connections ?
Or are there some "temporary" issues being taken care of ?

Thanks,

[aegyssus]

Please, anyone can post a valid configuration for Asterisk?
My config worked fine for a while but it's not working anymore since yesterday... I keep getting this account number is not valid.
BTW, more and more devices support asterisk, so it's very important to clarify this.
Thanks!

[FONGO_kris]

That's odd. Steve (Moderator) was having issue with it yesterday as well and when I took an old machine and loaded AsteriskNOW! onto it, it did not work either virtually or with my PAP2T!

Steve did get it working however and I'll see if he wants to post his settings up here. I know he changed quite a lot regarding registration so we'll see.

Keep in mind I do not know if he will post this since Asterisk is a SoftPBX and FPL is only for Residential, but I do know where you are coming from.

[FONGO_steve]

I encountered the same problem on my system last night too, even though I'm told nothing has changed.
Can you post your PEER and USER details here from your trunk? I won't have access to mine for a day or or two as I'm really busy, but I could try and decipher yours and see if I can catch anything off hand.

[akoei]

register => 1416xxxxxxx:xxxxxxxx@voip.freephoneline.ca/1416xxxxxxx

type=friend
context=fpl_in
host=voip.freephoneline.ca
username=1416xxxxxxx
secret=xxxxxxxxxxxx
port=5060
disallow=all
allow=ulaw

[teamer]

I have the same problem, i get a 603 returned. So I called the fpl support, and i got it confirmed that they have made channges in there sip servers, but the person i talked to could not say what changes they made. He just said that they dont want ppl to use asterisk since it is alot of companys that use it. I wish they told me before i signed up for a number.
I asked him if they are puting effort into solving the problem, and the response i got was that they dont see it as a major problem and that they are not prioritizing it.

[aegyssus]

it's still not working...
my initial trixbox config (which worked for a year) was:

Code: Select all

username=1514xxxyyyy
type=friend
secret=xxxxxx
qualify=no
insecure=very
host=voip.freephoneline.ca
fromdomain=voip.freephoneline.ca
disallow=all
canreinvite=yes
allow=ulaw&alaw&gsm

however, the things work fine in x-lite. I will keep on sniffing to notice the important differences between these configurations but filtering out extensions' SIP conversations is not an easy job... at one point I remember seeing the 603 message, but I'm not sure i get it consistently.
Many thanks,
aegyssus

[aegyssus]

He just said that they dont want ppl to use asterisk since it is alot of companys that use it. I wish they told me before i signed up for a number.

I agree 100%. On one way they're promoting their business, on other way they are shooting customers from behind and also they treat registered users as testers.

[FONGO_steve]

I have the same problem, i get a 603 returned. So I called the fpl support, and i got it confirmed that they have made channges in there sip servers, but the person i talked to could not say what changes they made. He just said that they dont want ppl to use asterisk since it is alot of companys that use it. I wish they told me before i signed up for a number.
I asked him if they are puting effort into solving the problem, and the response i got was that they dont see it as a major problem and that they are not prioritizing it.

Can you tell me who you spoke with?

Much like any ISP and their modems, we only provide support for our hardware and software. We have no problem if you want to use your own, but it's not our responsibility to know every single SIP client and ATA out there to assist. After all, we are a free service, and the fact we offer phone/email support in the first place does put us ahead of most.

Updates were made to our SIP server by the vendor PortaOne. At present time, we are unaware what they have updated or if they realize it has caused an "asterisk outage".

Asterisk does still work, but configuration must be setup differently than before. I spoke with one of our customers last night who was using Asterisk just like normal after a few peer/user details needed to be tweaked. So it does work, we're just waiting on someone to share the settings with us. I'll gladly try when I'm at home and have access to an asterisk system, but right now I do not.

Additionally, you might have mis-understood the agent you spoke with in regards to Asterisk. We catch a lot of businesses abusing our system and harvesting our numbers to use as business numbers on Asterisk systems. We are very precautions about this and take heavy steps to monitor any accounts with unusual looking activity. But like I said, asterisk is classed as third-party hardware, it will work, but we don't sit in our office trying to figure out how.

[akoei]

That's odd. Steve (Moderator) was having issue with it yesterday as well and when I took an old machine and loaded AsteriskNOW! onto it, it did not work either virtually or with my PAP2T!

Steve did get it working however and I'll see if he wants to post his settings up here. I know he changed quite a lot regarding registration so we'll see.

Keep in mind I do not know if he will post this since Asterisk is a SoftPBX and FPL is only for Residential, but I do know where you are coming from.

Distinguish business and resident should NOT from SIP agent. I am resident for sure, I use my Belkin router with Asterisk on OpenWRT, do you think a business user will use a cheap router host their PBX? The way to tell what kind of user is to see call volume, call time(if most of calls are in night time, can you say it is business user?), or some else .

[FONGO_steve]

That's odd. Steve (Moderator) was having issue with it yesterday as well and when I took an old machine and loaded AsteriskNOW! onto it, it did not work either virtually or with my PAP2T!

Steve did get it working however and I'll see if he wants to post his settings up here. I know he changed quite a lot regarding registration so we'll see.

Keep in mind I do not know if he will post this since Asterisk is a SoftPBX and FPL is only for Residential, but I do know where you are coming from.

Distinguish business and resident should NOT from SIP agent. I am resident for sure, I use my Belkin router with Asterisk on OpenWRT, do you think a business user will use a cheap router host their PBX? The way to tell what kind of user is to see call volume, call time(if most of calls are in night time, can you say it is business user?), or some else .

You are correct. For the most part we do not, of course I can't say all the methods we use to find business users, but I can say that they are almost always correct.

We have absolutely no issue with home users who want to use Asterisk. We allow third-party user agents and do not block any user agents either. Although, support is limited, as I've said. VOIP is a growing market and it's definitely a lot of fun to have an asterisk system at home as it opens up a world of possibilities for calling.

[teamer]

Much like any ISP and their modems, we only provide support for our hardware and software. We have no problem if you want to use your own, but it's not our responsibility to know every single SIP client and ATA out there to assist. After all, we are a free service, and the fact we offer phone/email support in the first place does put us ahead of most.

Updates were made to our SIP server by the vendor PortaOne. At present time, we are unaware what they have updated or if they realize it has caused an "asterisk outage".

Asterisk is widely used, and the fact that a vendor make changes in your SIP servers and you dont know what they did is for me strange. You should be able to get the info from PortaOne and they should also be able to tell you what changes the asterisk users need to make in the peer/user details. The message we get is a 603 Declined and with that info and the info about the change they should be able to solve the problem or give us some usable info.

[FONGO_steve]

Much like any ISP and their modems, we only provide support for our hardware and software. We have no problem if you want to use your own, but it's not our responsibility to know every single SIP client and ATA out there to assist. After all, we are a free service, and the fact we offer phone/email support in the first place does put us ahead of most.

Updates were made to our SIP server by the vendor PortaOne. At present time, we are unaware what they have updated or if they realize it has caused an "asterisk outage".

Asterisk is widely used, and the fact that a vendor make changes in your SIP servers and you dont know what they did is for me strange. You should be able to get the info from PortaOne and they should also be able to tell you what changes the asterisk users need to make in the peer/user details. The message we get is a 603 Declined and with that info and the info about the change they should be able to solve the problem or give us some usable info.

We are still exploring where the problem lies, as soon as I have info I will share it here first. There's a few other paths taken along the way that might cause this problem too, so I'm just waiting back on a couple emails.

[aegyssus]

We allow third-party user agents and do not block any user agents either.

That's very good to know.
I think FPL provides exactly what's needed for personal use whether we use ATA, softphone or a free pbx. With all due respect, for business purposes one might need a bit more... But that's why Worldline and Fibernetics exist and it's ok.
Hopefully such changes will not occur very often in the future.
Many thanks.

[lipeng8577]

HI Steve,

We as paid SIP user all wish your business is growing and is well. I just boasted your service with my Asterisk/openwrt/linksyswrt54GS yesterday or a day before at TAUG.ca (toronto asterisk user group) and posted my working config for other guys to reference. Suddenly, yesterday or day before, my wife said that my damn home voip system is not working and we all receive that dreadful "account invalid" voice message; but we can receive our call.

sip registration is also ok. I just learned that you're an Asterisk user as well, I am willing donate one linksys router with openwrt/asterisk configured to you so you guys can have a easy/simple test machine for now and for the future.

We're just a few tech guys, same as you and like to explore voip in home and don't like pay much to Bell. Hope your team fix this issue sooner.

the good part though, is that you are frankly with us and acknowlege the issue and trying to fix it. And, you're open minded and welcoming and working on new exciting technology. We like this. I am also planning to try out Freeswitch.org on openwrt/linksyswrt54gs and openSER as well,once I get some time. let's all work together build a healthy and warm community around this.

B.regards,
Peng Li

[FONGO_steve]

We allow third-party user agents and do not block any user agents either.

That's very good to know.
I think FPL provides exactly what's needed for a personal use whether we use ATA, softphone or a free pbx. With all due respect, for business purposes one might need a bit more... But that's why Worldline and Fibernetics exist and it's ok.
Hopefully such changes will not occur very often in the future.
Many thanks.

You seem to know us very well

We're still actively investigating this issue, as I was now told that we were not notified of any changes being made to the system by PortaOne themselves. We're still investigating other probable causes to this since no major changes appear to have been made - it's got a lot of us puzzled.

When I'm home tonight on my own time I will try and get a working config posted here for everyone.

[FONGO_steve]

Here's what I have working for outgoing and incoming. On a call with it right now that I placed outgoing. I'm no asterisk pro, so I could have unnecessary stuff in here.

Outbound Caller ID: Freephoneline number

Trunk Name: Freephoneline number

PEER Details:
allow=ulaw&g729
bindport=5060
canredirect=no
defaultexpiry=3600
disallow=all
dtmfmode=rfc2833
fromuser=1519804####
host=voip.freephoneline.ca
insecure=very
maxexpirey=3600
nat=no
qualify=no
secret=########
type=friend
username=1519804####

USER Details:
allow=ulaw&g729
canredirect=no
context=from-trunk
fromdomain=voip.freephoneline.ca
fromuser=1519804####
secret=########
type=user
username=1519804####

[i15]

You might try faking useragent to PAP2T to see if you can defeat FPL server

Many providers do not want Asterisk / Trixbos PBX'es connected to their system and they will screen them using the "useragent" field. The default for any trixbox or Asterisk installations is set to "Asterisk PBX", thats why registration fails.

To get around this, on Trixbox I simply added the following line to sip_custom.conf

useragent=LinksysPAP(2T)

After that registration succeded and I can receive incoming/outgoing calls, even 2 or more simultaneous calls over that trunk. You can use anything you want just as long as it doesnt contain the words Asterisk, PBX or Trixbox in it. This also explains why your ATA works and your PBX doesnt.

HTH

http://www.dslreports.com/forum/r223423 ... honelineca

[lipeng8577]

Here's what has worked two days ago on my asterisk@openwrt on Linksys wrt54gs. *************

*****************************************************************
Sip.config

register => 12891234567:password@fpl/12891234567
[fpl]
host=voip.freephoneline.ca

type=peer
context=fpl_in

;auth = 12891234567:password@voip.freephoneline.ca ***doesn't work****

username=12891234567
secret=password
fromuser=12891234567
fromdomain=voip.freephoneline.ca
insecure=port,invite
port=5060
canreinvite=no
;useragent=LinksysPAP(2T)

Extentions.conf

[fpl_in]
exten => _12891234567/_800.,1,Hangup
exten => _12891234567/_866.,1,Hangup
exten => _12891234567/_877.,1,Hangup
exten => _12891234567/_888.,1,Hangup
exten => _12891234567/_1800.,1,Hangup
exten => _12891234567/_1866.,1,Hangup
exten => _12891234567/_1877.,1,Hangup
exten => _12891234567/_1888.,1,Hangup

exten => _12891234567,1,Dial(SIP/2001&SIP/2002&SIP/2003&SIP/2004,${R
T})
exten => _12891234567,n,Goto(in_fpl-${DIALSTATUS},1)
exten => _12891234567,n,Hangup

exten => _in_fpl-.,1,Hangup(16)

[fpl_out]
exten => _911,1,SetCallerPres(allowed)
exten => _911,n,Set(CALLERID(all)=P LI <12891234567>)
exten => _911,n,Dial(SIP/${EXTEN}@fpl,${RT})
exten => _911,n,Goto(out_911-${DIALSTATUS},1)
exten => _911,n,Hangup
exten => _out_911-.,1,Hangup(16)

exten => _NXXNXXXXXX,1,SetCallerPres(allowed)
exten => _NXXNXXXXXX,n,Set(CALLERID(all)=P LI <12891234567>)
exten => _NXXNXXXXXX,n,Dial(SIP/${EXTEN}@fpl,${RT})
exten => _NXXNXXXXXX,n,Goto(out_lc-${DIALSTATUS},1)
exten => _NXXNXXXXXX,n,Hangup
exten => _out_lc-.,1,Hangup(16)

exten => _1900NXXXXXX,1,Congestion
exten => _1976NXXXXXX,1,Congestion

************************************************************

I tried compare this with what Steve and Chris's new asterisk config and could not see any difference. I tried tcpdump on my router's WAN interface and here's what I get.

1. Registration is good, which means user/pass is fine and I can receive call.
2. when calling out, I first shoot out one Invite, then got 100 Trying back and got 183 Sesstion Progress back and followd by receiving a 603 Declined. those message seem being generated by FPL's SER 0.96 or Sippy server. Then we heard this female voice "this account number is not valid"

So, if there's problem in authentication part which seems not uses something different than "registration authentication" process. I am willing to spend some time with you guys. Can you guys do some trace in your Sippy and see whether that process goes? What triggered this femail voice?

you can call my cell 416-648-4498 and i'd be happy to make a few calls so you can do a trace.

thanks
peng

[nicolas_dh]

*** see later post, the option useragent=LinksysPAP(2T) is not at the right place in this post***(nicolas_dh)
As I was writing this post, I saw peng's, showing I am not the only one
still receiving the "This account number is not valid" when attempting
outbound calls...

I also Tried, Steve's, Kris's, and even some adaptations of my own previous configuration with theirs
But I am still getting "This account number is not valid".
Tried "forcing" caller'ID to my FPL number:

Code: Select all

;;; extract from: /etc/asterisk/extensions.conf
;;; my phone devices keep adding "65" in front of dialed numbers, hence I remove them with EXTEN:2
;;; now trying to force the CALLER-ID to get oubound calls through to FREEPHONELINE
exten => _65418XXXXXXX,1,Set(CALLERID(all)="1418xxxxxxx" <1418xxxxxxx>)
exten => _65418XXXXXXX,n,Dial(SIP/${EXTEN:2}@freephoneline)

And also tried adding the useragent feature, with Kris's like settings:

Code: Select all

## extract from: /etc/asterisk/sip.conf
[freephoneline]
type=friend
context=from-freephoneline
username=1418xxxxxxx
secret=********
fromuser=1418xxxxxxx
host=voip.freephoneline.ca
qualify=yes
insecure=very
nat=yes
canreinvite=no
fromdomain=voip.freephoneline.ca
sendrpid=yes
disallow=all
allow=ulaw
allow=g729
useragent=LinksysPAP(2T) ;;;;; THIS OPTION IS AT THE WRONG PLACE ;;; see later post

But I still get: "This account number is not valid" message three times
then the busy line tones and I can read on the "asterisk -rvvv" output lines:

Code: Select all

    -- Executing [65418yyyyyyy@from_1000:1] Set("SIP/1000-081f0448", "CALLERID(all)="1418xxxxxxx" <1418xxxxxxx>") in new stack
    -- Executing [65418yyyyyyy@from_1000:2] Dial("SIP/1000-081f0448", "SIP/418yyyyyyy@freephoneline") in new stack
    -- Called 418yyyyyyy@freephoneline
    -- SIP/freephoneline-081cc478 is making progress passing it to SIP/1000-081f0448
    -- Got SIP response 603 "Declined" back from 208.65.240.142
    -- SIP/freephoneline-081cc478 is busy
  == Everyone is busy/congested at this time (1:1/0/0)
  == Auto fallthrough, channel 'SIP/1000-081f0448' status is 'BUSY'

Note: replaced the actual FPL number with "x"s and called number with "y"s....
Note: 208.65.240.142 is the current public ip address of: voip.freephoneline.ca
Note: I am using "Asterisk 1.4.17"

[magicray]

Wow you guys are seriously reading into this too much. There might have been a change in the back end for the useragent verification.

I had the same problem "this account number is not valid" when wanting to dial out. I use Elastix for my home system and here is what needed to be done to bypass the problem.

1. If you have "useragent=whatever" in your trunk PEER DETAILS - delete that line.
2. Open sip_custom.conf and place "useragent=whatever" in that file and reload.

DONE - if this does not work for people with trixbox or other pbx I'll post my peer details later, I dont have access to it right now.

[choanhemnhe]

magicray, could you please post your configuration here? thanks!

[nicolas_dh]

You might try faking useragent to PAP2T to see if you can defeat FPL server

There might have been a change in the back end for the useragent verification.

We allow third-party user agents and do not block any user agents either.

i15 and magicray, you were right ! WOW... it worked !!!
When I first tested adding line useragent=LinksysPAP(2T), I was putting it in the wrong place,
and had discarded this possibility since some posts were hinting us to believe there were
no "intentions" to "block" asterisk or whatever agents... I then focused elsewhere...
(was now looking at the "nonce" authentication mechanism http://www.trend-watcher.org/post/1/96)

Following "i15" advice and "magicray" re-advice, I recovered my "original" sip.conf file that had been working for a month before yesterday,
then searched for the "useragent" and found that it can only be specified under the [general] section in sip.conf
http://www.voip-info.org/wiki/view/Aste ... +useragent in my case (using Asterisk 1.4.17)
(using sip_custom.conf did not solve the issue for me since it is not a TrixBox, nor an Elastix)

So I did place the line useragent=LinksysPAP(2T) at the right place this time...
and guess what... it works. If I comment it out, guess what... "This account number is invalid".
Yessssssssssss..... Thanks i15 and magicray and to all for working/helping out on this issue.
Hence, the solution (for my family)

Code: Select all

[general]
context=default
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
language=fr
useragent=LinksysPAP(2T)    ;; this is now needed with FPL as of 2009-12-14
context=from-freephoneline
register => 1418xxxxxxx:********@voip.freephoneline.ca/1418xxxxxxx   
;;;  where:   1418xxxxxxx  is my FPL number, and ******** is my FPL password
canreinvite=no


[freephoneline]
context=from-freephoneline
allow=ulaw
disallow=h263
disallow=h263p
dtmfmode=auto
fromdomain=voip.freephoneline.ca
fromuser=1418xxxxxxx                     
;;; where 1418xxxxxxx  is my FPL number
host=voip.freephoneline.ca
insecure=very
language=fr
promiscredir=yes
qualify=no
secret=********
;;; where: ********  would be my FPL "sip" purchased password
type=peer
username=1418xxxxxxx
;;; where: 1418xxxxxxx  is my FPL number
nat=no                               ;;; in my case it is not behind a nat

;; You would also have 
;; specific Asterisk instructions/lines
;;  for your (sip) phone lines
;; 
;; [room01]
;;  - - - - - 
;; [room02] ... or whatever
;; 

Note, "nat=no" is good for my Asterisk setup, but you may need to set it to "nat=yes"
See Steve(mindabscence, nat=no) and Kris(admin, nat=yes "the conf post is gone")

mindabscence, you might want to tell PortaOne
or some of your "inside" people that it really seems like a change
has indeed taken place on 2009-12-14 in the back end for "User-Agent"
verification that seems to block "User-Agent" of type "Asterisk PBX" !
If it is not necessary (nor desired), could this change be reverted ?

Thanks again to all, it is great when it works...

[lipeng8577]

I can confirm that "useragent" stuff under "general" section works!!!!!!!!!!!!!!!!!!

Thanks a lot Nicolas for your post. I had tried that thing under "peer" definition before and "tcpdump" shows it's still "asterisk pbx"; now, under "General" section, it's changed.

It seems that the change from FPL's server denied "Invite" from "useragent=asterisk pbx", but allow registration and incoming call. If this is not intended by FPL, they may need to change setting in Sippy back.

Again, at least, it's a working solution and hope FPL future software upgrade doesn't introduce another way to block us as good faith customers. Hope Steve/Kris FPL team can iron out this issue.

Peng