Outgoing call stop working!

[akoei]

Hi freephoneline support, I am your SIP customer more than 3 years. Yesterday I realized outgoing calls stopped working, I am not sure when it started since I didn't make any outgoing 2-3 days, while incoming works well. From my online account's call log, my last good outgoing was Aug 22.

My environment is asterisk 1.4, from the console, I see the error is:
"Failed to authenticate on INVITE to xxxxxxx"

I tried software phone and ATA, it works well

Below is from my sip debug output, would you please take a look and point me out?

Thanks

Shawn
--------------------------------------------------------------------------------------------------------------------------------
-- Executing [9059515000@default:1] NoOp("SIP/99-00610df8", "") in new stack
-- Executing [9059515000@default:2] Dial("SIP/99-00610df8", "SIP/19059515000@fpl1|120|tT") in new stack
Audio is at 174.138.206.153 port 10074
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 208.65.240.165:5060:
INVITE sip:19059515000@voip.freephoneline.ca SIP/2.0
Via: SIP/2.0/UDP 174.138.206.153:6631;branch=z9hG4bK26111f17;rport
From: "14164778887" <sip:14164778887@174.138.206.153:6631>;tag=as7a392dfc
To: <sip:19059515000@voip.freephoneline.ca>
Contact: <sip:14164778887@174.138.206.153:6631>
Call-ID: 68af4e2e1dc4c9353a8406d309a8963d@174.138.206.153
CSeq: 102 INVITE
User-Agent: Linksys/RT31P2-3.1.6(LI)
Max-Forwards: 70
Date: Sat, 25 Aug 2012 16:24:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 244

v=0
o=root 8400 8400 IN IP4 174.138.206.153
s=session
c=IN IP4 174.138.206.153
t=0 0
m=audio 10074 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
-- Called 19059515000@fpl1
OpenWrt*CLI>
<--- SIP read from 208.65.240.165:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 174.138.206.153:6631;branch=z9hG4bK26111f17;rport=6631
To: <sip:19059515000@voip.freephoneline.ca>
From: "14164778887"<sip:14164778887@174.138.206.153:6631>;tag=as7a392dfc
Call-ID: 68af4e2e1dc4c9353a8406d309a8963d@174.138.206.153
CSeq: 102 INVITE
Content-Length: 0


<------------->
--- (7 headers 0 lines) ---

<--- SIP read from 208.65.240.165:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 174.138.206.153:6631;branch=z9hG4bK26111f17;rport=6631
Record-Route: <sip:208.65.240.165:5060;transport=udp;lr>
To: <sip:19059515000@voip.freephoneline.ca>
From: "14164778887"<sip:14164778887@174.138.206.153:6631>;tag=as7a392dfc
Call-ID: 68af4e2e1dc4c9353a8406d309a8963d@174.138.206.153
CSeq: 102 INVITE
Server: Sippy
WWW-Authenticate: Digest realm="voip.freephoneline.ca",nonce="13610bd079376096859b4db3c50eb925742b"
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
Transmitting (NAT) to 208.65.240.165:5060:
ACK sip:19059515000@voip.freephoneline.ca SIP/2.0
Via: SIP/2.0/UDP 174.138.206.153:6631;branch=z9hG4bK26111f17;rport
From: "14164778887" <sip:14164778887@174.138.206.153:6631>;tag=as7a392dfc
To: <sip:19059515000@voip.freephoneline.ca>
Contact: <sip:14164778887@174.138.206.153:6631>
Call-ID: 68af4e2e1dc4c9353a8406d309a8963d@174.138.206.153
CSeq: 102 ACK
User-Agent: Linksys/RT31P2-3.1.6(LI)
Max-Forwards: 70
Content-Length: 0


---
[Aug 25 12:24:37] NOTICE[574]: chan_sip.c:12377 handle_response_invite: Failed to authenticate on INVITE to '"14164778887" <sip:14164778887@174.138.206.153:6631>;tag=as7a392dfc'
-- SIP/fpl1-00614d88 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)

[Bloodsong]

Support for asterisk is going to be somewhat limited, but I'll help if I can.

Is this a vanilla Asterisk install or is it attached to a GUI extension like FreePBX or Trixbox?

[akoei]

Thanks. It is a optware version asterisk, on a router, no GUI shell.

[Bloodsong]

Oh, if I had read a little further I would have seen "OpenWrt"
Try the user-agent setting in the trunk registration.

Seeing as you're on OpenWRT, updating the asterisk version isn't likely going to happen. (Just trying to think what might have changed to invalidate your settings.)

[akoei]

Oh, if I had read a little further I would have seen "OpenWrt"
Try the user-agent setting in the trunk registration.

Seeing as you're on OpenWRT, updating the asterisk version isn't likely going to happen. (Just trying to think what might have changed to invalidate your settings.)

I've already had useragent setting 3 yrs ago, this time is different. I will try a regular linux box and see if it helps

[akoei]

Tried on asterisk 1.8.11.0, same error. I have been googling for whole day but no clue.
From previous post, FPL is declared they won't denied PBX like Asterisk, so:
Freephoneline support, please help me.

[Jake]

Tried on asterisk 1.8.11.0, same error. I have been googling for whole day but no clue.
From previous post, FPL is declared they won't denied PBX like Asterisk, so:
Freephoneline support, please help me.

I use Asterisk and don't have a problem with it, as do many here - so I don't think it has anything to do with that.

You say that incoming calls are fine, but not outgoing calls, right? If FPL was blocking you somehow, then you wouldn't receive calls either.

What does it say in your FPL online admin when you go to the sip settings page? Does it say you are connected with your user agent, or does it say disconnected?

The only reason I can think of why you are getting incoming calls but not outgoing is down to your settings. I use Elastix for the GUI side of things and I have quite a simple PEER details that works for me. I have tried others but this one seems to be fairly stable.

context=from-trunk
host=voip.freephoneline.ca
qualify=yes
username=1613366XXXX
secret={MY_SECRET}
type=friend
insecure=invite

and of course a registration string

Code: Select all

1613366XXXX:{MY_SECRET}@voip.freephoneline.ca/

If your FPL admin is saying you are connected, it should work - bar any funny settings Asterisk side. As a last Hail Mary attempt to fix it, you might try changing your user-agent to something else. I know you have changed it already reading through your log, but you never know.

[akoei]

Thanks, Jake, I have tried every thing you mentioned, cuz those also are everything I can try.

The reason I am asking Fongo support is: it suddenly stopped working sometime last week, before it was fine. I know I didn't change anything; and, this happened this June, around 3 days, and fixed itself. From other forums like dslreport, I saw someone mentioned nothing can be done from our side, it is VSP side.

Since it sounds like only me, I am wondering where else could be matter?
ISP: I am using distributel DSL, was CIA and 3web
Router: I am using Tomato v1.28

If the ISP and router do nothing with this fail, then the only point is Fongo......

[Bing Kol]

...
From my online account's call log, my last good outgoing was Aug 22.

My environment is asterisk 1.4, from the console, I see the error is:
"Failed to authenticate on INVITE to xxxxxxx"

I tried software phone and ATA, it works well

...
Shawn

...

Thanks, Jake, I have tried every thing you mentioned, cuz those also are everything I can try.

The reason I am asking Fongo support is: it suddenly stopped working sometime last week, before it was fine. I know I didn't change anything; and, this happened this June, around 3 days, and fixed itself. From other forums like dslreport, I saw someone mentioned nothing can be done from our side, it is VSP side.

Since it sounds like only me, I am wondering where else could be matter?
ISP: I am using distributel DSL, was CIA and 3web
Router: I am using Tomato v1.28

If the ISP and router do nothing with this fail, then the only point is Fongo......

The statements in bold confuse me.

So, which comes first? The egg or the chicken?

[akoei]

...
From my online account's call log, my last good outgoing was Aug 22.

My environment is asterisk 1.4, from the console, I see the error is:
"Failed to authenticate on INVITE to xxxxxxx"

I tried software phone and ATA, it works well

...
Shawn

...

Thanks, Jake, I have tried every thing you mentioned, cuz those also are everything I can try.

The reason I am asking Fongo support is: it suddenly stopped working sometime last week, before it was fine. I know I didn't change anything; and, this happened this June, around 3 days, and fixed itself. From other forums like dslreport, I saw someone mentioned nothing can be done from our side, it is VSP side.

Since it sounds like only me, I am wondering where else could be matter?
ISP: I am using distributel DSL, was CIA and 3web
Router: I am using Tomato v1.28

If the ISP and router do nothing with this fail, then the only point is Fongo......

The statements in bold confuse me.

So, which comes first? The egg or the chicken?

Sorry confused you what you bold above, just my troubleshooting:
first FPL in my asterisk stopped working for outgoing;
then I tried soft phone (actually didn't try ATA but I think same as soft phone) works;
then I tried a lot conf changes but none of work
the very last step will be looking into my ISP and router since looks only me got this issue, if there is no any word from Fongo.

But, I'd like believe Fongo did something since it just stopped working suddenly, and it makes no sense of my ISP did the trick

[Jake]

But, I'd like believe Fongo did something since it just stopped working suddenly, and it makes no sense of my ISP did the trick

If Fongo had done something to stop your outgoing calls via asterisk, there would be a lot of other people on here complaining - so I think it has something to do with your setup.

If it shows that you are registered on both your * , and in the fongo admin - AND you can receive incoming calls; I don't believe anything is being blocked by fongo. Your account must be OK if you can use the softphone.

I think you are better off looking into your settings (trunks, call routes, and codecs), or maybe even something in your router.

I know it is probably a pain, but you could eliminate your ISP and router as the problem by taking your * box round to a friends house and trying it there.
Another test could be to set up another trunk with someone else and see if you can get incoming calls from that.

I will say, you are very unlikely to get support from fongo. It's hard enough getting any help with an ATA you didn't buy through them. They will only support hardware that they sell - but then this is where the forum comes in to try and help. Have you tried on the * forums? They will probably be able to help you more.

[akoei]

Hi Jake, I have to say what you said if quite right, I am considering take my * to friend's house but that a lot more work, and time.... before that, I still can do something like install an asteriskNow into PC, etc.

I will post my investigation, thanks!

[akoei]

Good sign:
Installed AasteriskNow 2.0, which is asterisk 1.8.11 into vmware, then it.......WORKS! Based a report from DSLReport, I even removed useragent, now my account shows: "Asterisk PBX 1.8.11-cert1"!

I will continue work on finding out why my asterisk 1.4 stopped working.

[Jake]

Well that's good to hear!

It's always a pain when things just stop without changing anything. Good luck.