Both local incoming and outgoing calls failed

[jchen]

Just regestered and got local number 1-613-689-xxxx (Belleville, ON). However I can't make any outgoing call to any other Belleville local numbers, I always got "603 Declined". When I called my FPL number from other phone, I always heard "This call is not covered by your tel's plan". However I can make an outgoing call to Toronto number (area code 416). I'm using FreePhoneLine soft phone downloaded from your website when I registered.

[FONGO_kris]

To start, if this regards Asterisk, please refer here, there is some assemblance of a "fix" there : http://forum.freephoneline.ca/viewtopic.php?f=10&t=678

Secondly: If NOT Asterisk, please refer to my first post here: http://forum.freephoneline.ca/viewtopic.php?f=13&t=687

[jchen]

As I said in the first message, I'm using FreePhoneLine softphone, nothing to do with Asterisk. And your first post (viewtopic.php?f=13&t=687) is not my case. My case is not the RTP (in the other words, audio) issue, it is the call signaling issue. Your SIP server voip.freephoneline.ca (208.65.240.142) always decline calls. Please refer to follow wireshark traces for an outgoing call (for privacy reason, I repalced my number with 1613689XXXX. From the traces you can see the port is ok (the soft phone initiated the call at port 6060), it offerred g711u, g723 and gsm at RTP port 13000. However, after 100 Trying and 180 Session In Progress, voip.freephoneline.ca never replied 200 OK SDP to negotiate RTP, it declined the call with 603 instead. Thus it is not the port or audio issue since it never started to negotiate RTP.

For an incoming calls, there wasn't any SIP packet arrived at the PC, thus the call from local PSTN was blocked at the server side.

As a VOIP software engineer, I summarized that the problem is on the server side. However I need your help to fix my account, perhaps the server problem because I do not have any view inside your server.


INVITE sip:16139675000@voip.freephoneline.ca SIP/2.0
Via: SIP/2.0/UDP 192.168.1.104:6060;rport;branch=z9hG4bK02934
Max-Forwards: 70
To: <sip:16139675000@voip.freephoneline.ca>
From: <sip:1613689XXXX@208.65.240.142>;tag=z9hG4bK30536638
Call-ID: 029214996392@208.65.240.142
CSeq: 1 INVITE
Contact: <sip:1613689XXXX@192.168.1.104:6060>
Expires: 3600
User-Agent: FreePhoneLine 2.2.0.0
Content-Type: application/sdp
Content-Length: 208

v=0
o=sip:1613689XXXX@208.65.240.142 0 0 IN IP4 192.168.1.104
s=Session SIP/SDP
c=IN IP4 192.168.1.104
t=0 0
m=audio 13000 RTP/AVP 0 4 3
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=rtpmap:3 GSM/8000


SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 192.168.1.104:6060;rport=6060;branch=z9hG4bK02934;received=216.99.56.195
To: <sip:16139675000@voip.freephoneline.ca>
From: <sip:1613689XXXX@208.65.240.142>;tag=z9hG4bK30536638
Call-ID: 029214996392@208.65.240.142
CSeq: 1 INVITE
Server: Sip EXpress router (0.9.6 (i386/freebsd))
Content-Length: 0


SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.104:6060;received=216.99.56.195;rport=6060;branch=z9hG4bK02934
Record-Route: <sip:208.65.240.142;ftag=z9hG4bK30536638;lr>
From: <sip:1613689XXXX@208.65.240.142>;tag=z9hG4bK30536638
To: <sip:16139675000@voip.freephoneline.ca>
Call-ID: 029214996392@208.65.240.142
CSeq: 1 INVITE
Server: Sippy
WWW-Authenticate: Digest realm="208.65.240.142",nonce="2b8f08dfbbde755c5b4a91ae9b0dc003de4a"


ACK sip:16139675000@voip.freephoneline.ca SIP/2.0
Via: SIP/2.0/UDP 192.168.1.104:6060;rport;branch=z9hG4bK02934
Max-Forwards: 70
To: <sip:16139675000@voip.freephoneline.ca>
From: <sip:1613689XXXX@208.65.240.142>;tag=z9hG4bK30536638
Call-ID: 029214996392@208.65.240.142
CSeq: 1 ACK
User-Agent: FreePhoneLine 2.2.0.0
Content-Length: 0


INVITE sip:16139675000@voip.freephoneline.ca SIP/2.0
Via: SIP/2.0/UDP 192.168.1.104:6060;rport;branch=z9hG4bK02934
Max-Forwards: 70
To: <sip:16139675000@voip.freephoneline.ca>
From: <sip:1613689XXXX@208.65.240.142>;tag=z9hG4bK30536638
Call-ID: 029214996392@208.65.240.142
CSeq: 2 INVITE
Contact: <sip:1613689XXXX@192.168.1.104:6060>
Expires: 3600
User-Agent: FreePhoneLine 2.2.0.0
Authorization: Digest username="1613689XXXX", realm="208.65.240.142", nonce="2b8f08dfbbde755c5b4a91ae9b0dc003de4a", uri="sip:16139675000@voip.freephoneline.ca", response="93bb82b65781a413fd1ea0f60f847b72"
Content-Type: application/sdp
Content-Length: 208

v=0
o=sip:1613689XXXX@208.65.240.142 0 0 IN IP4 192.168.1.104
s=Session SIP/SDP
c=IN IP4 192.168.1.104
t=0 0
m=audio 13000 RTP/AVP 0 4 3
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=rtpmap:3 GSM/8000


SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 192.168.1.104:6060;rport=6060;branch=z9hG4bK02934;received=216.99.56.195
To: <sip:16139675000@voip.freephoneline.ca>
From: <sip:1613689XXXX@208.65.240.142>;tag=z9hG4bK30536638
Call-ID: 029214996392@208.65.240.142
CSeq: 2 INVITE
Server: Sip EXpress router (0.9.6 (i386/freebsd))
Content-Length: 0

SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.104:6060;received=216.99.56.195;rport=6060;branch=z9hG4bK02934
Record-Route: <sip:208.65.240.142;ftag=z9hG4bK30536638;lr>
From: <sip:1613689XXXX@208.65.240.142>;tag=z9hG4bK30536638
To: <sip:16139675000@voip.freephoneline.ca>;tag=ee2d8c70c1dd33d69fd8a6ffe06afb95i
Call-ID: 029214996392@208.65.240.142
CSeq: 2 INVITE
Server: Sippy
Content-Length: 212
Content-Type: application/sdp

v=0
o=Sippy 157718444 1 IN IP4 208.65.240.142
s=session
t=0 0
m=audio 35860 RTP/AVP 0 3
c=IN IP4 208.65.240.142
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv


SIP/2.0 603 Declined
Via: SIP/2.0/UDP 192.168.1.104:6060;received=216.99.56.195;rport=6060;branch=z9hG4bK02934
Record-Route: <sip:208.65.240.142;ftag=z9hG4bK30536638;lr>
From: <sip:1613689XXXX@208.65.240.142>;tag=z9hG4bK30536638
To: <sip:16139675000@voip.freephoneline.ca>;tag=ee2d8c70c1dd33d69fd8a6ffe06afb95i
Call-ID: 029214996392@208.65.240.142
CSeq: 2 INVITE
Server: Sippy


SIP/2.0 603 Declined
Via: SIP/2.0/UDP 192.168.1.104:6060;received=216.99.56.195;rport=6060;branch=z9hG4bK02934
Record-Route: <sip:208.65.240.142;ftag=z9hG4bK30536638;lr>
From: <sip:1613689XXXX@208.65.240.142>;tag=z9hG4bK30536638
To: <sip:16139675000@voip.freephoneline.ca>;tag=ee2d8c70c1dd33d69fd8a6ffe06afb95i
Call-ID: 029214996392@208.65.240.142
CSeq: 2 INVITE
Server: Sippy


SIP/2.0 603 Declined
Via: SIP/2.0/UDP 192.168.1.104:6060;received=216.99.56.195;rport=6060;branch=z9hG4bK02934
Record-Route: <sip:208.65.240.142;ftag=z9hG4bK30536638;lr>
From: <sip:1613689XXXX@208.65.240.142>;tag=z9hG4bK30536638
To: <sip:16139675000@voip.freephoneline.ca>;tag=ee2d8c70c1dd33d69fd8a6ffe06afb95i
Call-ID: 029214996392@208.65.240.142
CSeq: 2 INVITE
Server: Sippy


SIP/2.0 603 Declined
Via: SIP/2.0/UDP 192.168.1.104:6060;received=216.99.56.195;rport=6060;branch=z9hG4bK02934
Record-Route: <sip:208.65.240.142;ftag=z9hG4bK30536638;lr>
From: <sip:1613689XXXX@208.65.240.142>;tag=z9hG4bK30536638
To: <sip:16139675000@voip.freephoneline.ca>;tag=ee2d8c70c1dd33d69fd8a6ffe06afb95i
Call-ID: 029214996392@208.65.240.142
CSeq: 2 INVITE
Server: Sippy

[FONGO_kris]

I assume from your tracelogs and what you've said here I can indeed verify that you've forwarded the appropriate ports. UDP - 13000 & 13001.

What anti-virus are you currently running? Norton and McAfee seem to be the big culprits in the "No Connection to Server" and one way audio issues. Norton I believe has the ability to Port Scan all incoming traffic, so you may want to turn it off for at least a test-trace.

[jchen]

Thanks for the reply. As per your suggestion, I disabled the Symentec Endpoint Protection, and retestd outgoing call. I got the exactly same result.

I would argue again, this is not the time to discussion audio because the FPL server had not replied 200 OK SDP to negotiate RTP yet. This is a signaling issue that call can not be established because the server decline the call. Perhaps permission issue on my account?

For the incoming call, the sip message never arrived my PC, and was rejected at the server. Thus we should focus on the server side, instead of the my PC.

[FONGO_kris]

Can you PM me your FPL number and I'll pull up the server generated SIP logs.

[Jake]

Maybe I am looking at this the wrong way, but when I try and call a Belleville number (6139686481 - The city of Belleville inquiry line) I get the 'This call is not covered by your tariff plan' also.

I'm going to suggest that although you can get a Belleville number, the Belleville numbers (613 968 - 613 967) are not in the FPL system as 'local' to be called. The same as has happened in this thread with other 'local' numbers http://forum.freephoneline.ca/viewtopic.php?f=8&t=347

Could this be the case Admin?

Can you call Ottawa numbers? try 613 239 5000 (Tourism inquiries)

When you call in, it could be that the line you are calling from (cell?) does not know a 613 689 XXXX number as being local rather it is long distance, so it is them that are not connecting you. What happens if you try and dial 1 613 689 XXXX ?

[Bloodsong]

My cousin whom I just had sign up in the Barrie area, and warned him no Barrie numbers existed yet has signed up with a Barrie local number. Upon looking at the FPL front page I noticed a few Barrie area cities have been made available.

This suggests to me that new number blocks have been purchased, perhaps some new partnerships/cities have been explored. It is possible that Belleville is one of these new blocks. If that's the case then server side does not yet seem to be setup to properly handle calls to these locations, as Barrie/Midland/Wasaga/Penetang are still unreachable as well despite now having locally available FPL numbers.

I've been keeping my eye on the Barrie area for numbers coming up as I have much family there. Thus I know this is a few days, maybe a week tops these numbers have been available. So just a theory, but Belleville might be new as well.

If so, I expect some growing pains. However I also appreciate the lengths FPL is going to in order to bring Canadians an amazing service.

[FONGO_steve]

You are correct about the new cities not being in the tariff plan. I sent an email off to see how that happened and how soon it can be done
That would solve our problem we're having here!

[Bloodsong]

LOL While I'm PMing steve, he's posting on my comment.

You rock Steve!

[jchen]

For outgoing call issue to local Belleville numbers (613-967-xxxx carrier Bell, 613-827-xxxx carrier Petro Canada Mobile), it could be the tariff plan issue like people said. Probably FPL server doesn't reconize them as local number with my FPL number 613-689-xxxx.

However for incoming call problem, there may be more issue on FPL server than that. Since the incoming call is unlimited, and any cost should charge to the caller if the call comes from a non-local phone. So theoretically an incoming call should get into my softphone. My test indicates none of incoming reached my softphone, for either call from local phone, or from other cities like Toronto, Ottawa. Since caller heard "This call is not covered by your tariff's plan" which was played by FPL server, the problem should be in FPL server, instead of other carriers. Basically any restriction should be removed for any incoming call to FPL users.

[FONGO_kris]

Incoming calls are not restricted in any way, even international incoming works as it would regularly, and for the other incoming issues I'll look into it tomorrow when I get in the office at 7am, since our meta is IP restricted for obvious reasons. For the outgoing tariff issues those can be easily added to our meta-switch in a matter of 20 minutes.

Anyone here can PM me a list of NPA-NXX's (Thats the area code & following 3 digits) that they get "This call is not covered by your tariff plan" for and I'll see what I can do. Obviously I can't guarantee anything but I can work some magic in meta and usually get it working.

[jchen]

Any progress?

Beside the Belleville local NPA-NXX's that I PMed you this moring (613-961, 613-962, 613-965, 613-966, 613-967, 163-968), here is more. All of them are Belleville local numbers. I personally tested them and got failure results ("This call is not covered by your tariff plan")

613-210, 613-392, 613-393, 613-394, 613-395, 613-396, 613-397, 613-398, 613-471, 613-472, 613-473, 613-475, 613-476, 613-477, 613-478, 613-779, 613-963, 613-969

[FONGO_steve]

A ticket is opened for them to be added to our tariffs. It seems some confusion came up when those numbers were added and we were left with an issue in the tariff plans. Should be fixed soon though!

[jchen]

Adding local numbers case by case may not be a good solution. You may get endless tickets when people from other cities or municipalities
join. A table that covers all ares should be imported, like a carrier does to determine a call is long distance or local. If this database is not available to import, you may use a simple rule instead: calls within the same area code are taken as local call (except a few municipalities which have multiple area codes, like Toronto), like 7-eleven mobility does.

[FONGO_steve]

Adding local numbers case by case may not be a good solution. You may get endless tickets when people from other cities or municipalities
join. A table that covers all ares should be imported, like a carrier does to determine a call is long distance or local. If this database is not available to import, you may use a simple rule instead: calls within the same area code are taken as local call (except a few municipalities which have multiple area codes, like Toronto), like 7-eleven mobility does.

That's what we're doing

[Bloodsong]

Update on this issue:
Calls from FPL to Barrie area FPL are working.
Calls from Landline to Barrie area FPL are working.
Calls from FPL to Barrie area landlines/cells still not working.

So hopefully we get some updated tariffs for calling to POTS/PSTN soon, but a great start.

[FONGO_kris]

I'll look up all cell NPA-NXX's in that area and get them added to our tariff's.

[FONGO_steve]

Update on this issue:
Calls from FPL to Barrie area FPL are working.
Calls from Landline to Barrie area FPL are working.
Calls from FPL to Barrie area landlines/cells still not working.

So hopefully we get some updated tariffs for calling to POTS/PSTN soon, but a great start.

It's happening today, try again tomorrow morning at the latest

[Bloodsong]

Well I just tested 705-526 (Oldest Midland land line circuit I know.)
Now works, so I don't know if there are any that got skipped, but this definitely means FPL is once again a phone solution for me You guys are awesome.

As to the last post, I just want to further emphasize how absolutely GREAT you guys have been (Kris&Steve + FPL backroom crew we never see.) Realizing as I did not capitalize any of the previously stated greatness.

Now all I have to do is set up my ATA... which means finding out which box I packed it in... and I'll be able to convince my Wife to use FPL more

[FONGO_steve]

Well I just tested 705-526 (Oldest Midland land line circuit I know.)
Now works, so I don't know if there are any that got skipped, but this definitely means FPL is once again a phone solution for me You guys are awesome.

As to the last post, I just want to further emphasize how absolutely GREAT you guys have been (Kris&Steve + FPL backroom crew we never see.) Realizing as I did not capitalize any of the previously stated greatness.

Now all I have to do is set up my ATA... which means finding out which box I packed it in... and I'll be able to convince my Wife to use FPL more

Many thanks for the kind words

and we do apologize for those new numbers sliding on the website prematurely before the network team had a chance to setup the tariffs for them. Making sure it won't happen again!

[Bloodsong]

Heh heh, I'm happy either way, though it was a bit of a shocker when it had trouble communicating even within FPL's own network

*pokes at Belleville from the first message* You're up and running now to I assume?

[jchen]

Heh heh, I'm happy either way, though it was a bit of a shocker when it had trouble communicating even within FPL's own network

*pokes at Belleville from the first message* You're up and running now to I assume?

Thanks for the great job the admin and Steve have done. I'm the Belleville guy. Now the call with local PSTN numbers is working, and the FollowMe works with local numbers as well. Thanks again.

For outgoing call, I found a security hole. I will report to the admin or Steve by PM.