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Could you clarify, what is that SIPBroker test number? May I see the Console
output when you dial this number?
I'm also wondering if the situation improves when you revert the router's port
forwarding to their previous settings.
Original comment by mte...@gmail.com
on 19 Feb 2011 at 5:32
Sure.
The SIPBroker test number (*011188888@sipbroker.com or '*011188888#' on my
phone) is for testing whether or not the SIP account is set up properly.
I'd love to provide you the console logs of before I reset the forwarding
settings, but there is nothing logged at all. Well, nothing but keep-alive
messages. But no logs from calling the number show up anywhere, after waiting a
minute or two.
I haven't yet had a chance to try whether or not it improves if I change those
settings back. I'll try that out in a second.
Original comment by XANAVi...@gmail.com
on 19 Feb 2011 at 5:53
Oh, by the way, just a moment before I start testing whether or not reversing
the port forwarding settings, I checked the router logs for incoming dropped
data.
There was nothing listed, so either the router is not dropping packets itself
or the logging feature is broken (of course, the second one I doubt since it
logs outgoing connections okay, but maybe always a possibility).
(A few minutes later)
Okay, after changing the port forwarding settings back to what they were
originally..
Unfortunately, nothing has changed for me. SIPSorcery's console log still only
shows keep-alive messages.
I am, of course, logged in correctly to SIPSorcery on my Linksys PAP2T-NA.
Otherwise, no Google Voice calling (which by the still does work A-OK).
I don't even hear a ringing sound that I believe should come through to show
that it is dialing the SIPBroker test number (but I am not sure if SIPSorcery
or my ATA generates it). I hear two when doing Google Voice calls, one is the
'better sounding' one and the other Google Voice's normal ringing sound.
Original comment by XANAVi...@gmail.com
on 19 Feb 2011 at 6:06
Here is the output of my Linksys PAP2T-NA's status page during a call to the
SIPBroker number.
If you'd like a status page log from me during a Google Voice call to compare
(you might), just ask. I cam't make calls at this time since it would rude to
do that at 1:30AM, so I'll do it in the morning.
It's in .doc format for Word 97/2000/XP versions. If it doesn't work I can
provide it in ODF format.
It's fsr too long to fit in this text box (Google says I'm sending an invalid
request when I just to try post it in this text box).
Original comment by XANAVi...@gmail.com
on 19 Feb 2011 at 6:28
Attachments:
> Call Mapped RTP Port:
> 8796 >> 0
Apparently you're not forwarding RTP port and it may be the cause of your
problem.
> Last Called Number:
> *011188888@sipbroker.com
Apparently you're communicating with sipbroker.com directly, that's why the
call doesn't appear in SS console. I believe you're programmed this number in
your ATA rather than used SS to connect it for you (by adding the number to
your Speeddial table, for example).
What I'm saying, this issue has nothing to do with Sipsorcery and its dialplan.
Now let's think what might be wrong. When you're dealing with Sipsorcery, it
acts like a STUN server, replacing local IPs with your external IP. That's why
the thing works even if STUN is not enabled in your ATA. Sipbroker is a
different story!
Please check STUN settings in your ATA, you may need to enable STUN mapping and
enter STUN server name/address (e.g. stun01.sipphone.com).
Original comment by mte...@gmail.com
on 19 Feb 2011 at 10:12
I have STUN enabled in my ATA.
The following server is configured:
stun.call4-free.com
I do see 'Stunning' as a status message when calling numbers, too. So it is
contacting that server.
Anyway, I just thought I'd tell you that.
I haven't contacted VOIPLink yet.
Yes, my dialplan in my ATA is:
(*xxx[x*].
<:@sipbroker.com>|*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.)
Because, otherwise (see Issue #116) I will not be able to dial those numbers,
since I'll just get a busy tone immediately for some reason.
I can try to remove the SIPBroker portion out my ATA dialplan and see if it
still does it, but I'm fairly certain I won't be able to dial those numbers
again.
I do have SIPBroker configured in my SIPSorcery dialplan (not that you're
doubting that) since I can dial their numbers directly using any SIP client on
my cellphone (i.e. Linphone) or on my computer (ExpressTalk).
I do have RTP forwarding enabled. I'm forwarding the ports [5060] and
[8000-8999] to 172.16.1.249 (the inside IP for my ATA).
Original comment by XANAVi...@gmail.com
on 20 Feb 2011 at 4:40
Yeah, I removed the 'Dial SIPBroker directly' portion and I can't dial those
numbers anymore.
My ATA reports it as 'Invalid' if I try to dial star codes (like the SIPBroker
test number available at *011188888). My ATA supports those special codes like
from the phone company has, as it turns out.
Like *82 (unblock) and *67 (block). I wonder if that could be the cause of not
being able to dial SIPBroker star codes or something.
I'm still working at it, I'm going to continue and maybe post some more if I
figure something out.
Original comment by XANAVi...@gmail.com
on 20 Feb 2011 at 4:48
Sorry for updating so much. I hope you don't have this as starred...
I'd hate to fill up your inbox with just this one issue.
So, I can't find any settings in my Linksys PAP2T-NA to disable feature service
codes.
That's bad.
So, I did a Google Voice call just to see something and it turns out that info
status you referenced in your post is always 0 on the other end.
I am forwarding in my router settings ports [8000-8999] and the ports my ATA's
selecting are ones that are in that range.
From yesterday's SIPBroker test call:
> Call Mapped RTP Port:
> 8796 >> 0
From my recent Google Voice call:
> Call Mapped RTP Port:
> 8140 >> 0
Original comment by XANAVi...@gmail.com
on 20 Feb 2011 at 5:05
For my final update of today (since it's Saturday), I'm going to provide you
with an status list from my Linksys's call info webpage.
I didn't get any closer to solving why I can't dial SIPBroker star codes
directly even though I can with everything else linked to my SIPSorcery account.
But here's the info for during a Google Voice call. I removed the area code
number from the phone number I was dialing.
Original comment by XANAVi...@gmail.com
on 20 Feb 2011 at 5:19
Attachments:
Look, your ATA is forwarding test calls directly to Sipbroker and therefore,
the issue is not related to Sipsorcery and your dialplan. Probably it makes
this trouble ticket invalid here.
As to service codes, I'm sure it has nothing to do with the issue. If your ATA
treated dialed number as the service code, it wouldn't connect to Sipbroker at
all.
If you want to be able to dial star-prefixed numbers, you need a corresponding
entry in your ATA's dialplan (by default, only digits are accepted). This entry
can forward star-prefixed numbers to Sipbroker, Sipsorcery or any other
provider of your choice. I have to mention that your test only validates
connection to Sipbroker and independent from SS status: test number may work
even though SS server is down, your dialplan is incorrect etc. Reverse is also
true: if SS works, test connection may not work (your case).
Original comment by mte...@gmail.com
on 20 Feb 2011 at 6:56
Okay, very well then.
You sound angry at me for being a novice and posting here for some reason.
I'm not angry back but it does make me sad to see that my asking for help here
is not an accepted solution for being stuck.
In any case,
I modified the '@sipbroker.com' part of my before mentioned dialplan (the one
from before I changed it back) into '@sipsorcery.com'.
It works now, so therefore thank you Mike T.
Original comment by XANAVi...@gmail.com
on 20 Feb 2011 at 7:24
I'm not angry, I'm just trying to stay on the topic (which is Sipsorcery, Ruby
dialplans and Google Voice). There are many other VoIP-related forums (such as
Voxilla).
Re: Sipbroker test number works when you connect via Sipsorcery. This is yet
another argument that the issue is STUN-related. The difference between "direct
to Sipbroker" and "Sipbroker via Sipsorcery" is that in latter case SS server
mangles IP addresses. Probably you need to try a different STUN server. I don't
have experience with PAP2T but SPA3102 has separate "NAT mapping enabled"
settings for each and every gateway (LINE1, PSTN, GWn). It might be the case
with PAP2T, too!
Original comment by mte...@gmail.com
on 20 Feb 2011 at 9:21
Original comment by easter...@gmail.com
on 20 Feb 2011 at 2:55
Original issue reported on code.google.com by
XANAVi...@gmail.com
on 18 Feb 2011 at 3:34