Hadar Pedhazur
2005-Mar-03 15:17 UTC
[Asterisk-Users] FWD and SIPPHONE problems after upgrading to CVS HEAD
I have been successfully connected (incoming and outgoing)
to FWD for a very long time. A few months ago, I changed
from SIP-based FWD service to IAX2-based, and that went fine
as well, both incoming and outgoing.
At the time, I was running Asterisk 1.0.3 Stable.
I rarely use the service, so other than noticing that I was
always successfully registered to FWD, I didn't make or
receive calls recently.
2 weeks ago, I upgraded to CVS HEAD:
Asterisk CVS-HEAD-02/21/05-09:07:50
Still didn't make or receive calls to FWD since the upgrade,
but everything else has worked flawlessly (including sixTel,
NuFone, etc.). All my softphones (SIP and IAX2) and
Sipura-2000's work perfectly too.
On to the problem... A few days ago, I signed up for an
account with SIPPhone. When I did a "sip reload", which had
the register statement, I immediately got a call "welcoming"
me, so I thought everything was fine. It wasn't.
I have been unable to make any calls to sipphone, and even
though the registration appears to work (and my.sipphone.com
shows me as "online"), all calls to my number actually claim
that I am unavailable, and go directly to voicemail.
Before I show my configs and CLI output, a few more
background data points:
I can successfully connect to sipphone with their own
download of X-Lite (pre-configured), and I can set a profile
in SJPhone by hand and it works too, both incoming and
outgoing, so I have the correct password, etc.
Today, I tested outgoing calls on FWD (actually to use the
peering to test incoming on sipphone), and my calls to FWD
are failing now as well. Incoming from FWD (via IAX2) still
works correctly. Worse, I also tried to go back to SIP-based
outgoing to FWD, and I get the same error as I do for
sipphone, so now I am starting to suspect that it's Asterisk
CVS HEAD that's possibly the problem...
Finally, the machine that is connected to both FWD and
SIPPHONE is on a public static IP address, so there are no
NAT issues involved here, and no STUN services needed
either.
OK, here is the sip.conf entry:
register=1747XXXXXXX:YYYYYYY@proxy01.sipphone.com/4321
[proxy01.sipphone.com]
type=peer
;auth=md5
secret=YYYYYYY
username=1747XXXXXXX
fromuser=1747XXXXXXX
fromdomain=proxy01.sipphone.com
host=proxy01.sipphone.com
nat=no
qualify=no
canreinvite=no
disallow=all
allow=ulaw
;context=default
;callerid="Hadar Pedhazur" <1747XXXXXXX>
(The above has been variously named sipphone, sipphone-out
and now proxy01.sipphone.com, all with the same exact
result! Also, the above has been tried with auth=md5
uncommented as well, and also no password, and
insecure=vary, etc.)
Now extensions.conf:
; Dial SIPPhone with a prefix of 76
exten => _76.,1,SetCallerID(${SIPPHONENUM})
exten => _76.,2,SetCIDName("Hadar Pedhazur")
exten => _76.,3,Dial(SIP/${EXTEN:2}@proxy01.sipphone.com)
OK, here's the output from a call:
-- Called 411@proxy01.sipphone.com
-- Got SIP response 500 "I'm terribly sorry, server error occured
(1/SL)" back from 198.65.166.131
-- SIP/proxy01.sipphone.com-78d5 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
Notice that at the end of the "Got SIP response" line, is
the correct IP address of their server, so it's finding the
correct server. As mentioned above, if I switch FWD to call
via SIP, I get the same _exact_ error message, but from
FWD's correct IP address rather than SIPPhone. This seems
very suspicious to me...
Finally, just for completeness, here is the CLI output for
attempting to call FWD via IAX2. This used to work, though I
can't say when it started failing:
-- Called fwd-gw/612
-- Call accepted by 65.39.205.121 (format ulaw)
-- Format for call is ulaw
-- IAX2/fwd-gw-4 is busy
I have called _many_ times, and every time I get an instant
"is busy" in the CLI, and I can receive calls without a
problem, so I don't think it's that they really are busy.
For now, I'm more interested in fixing the SIPPhone problem,
and if that ends up working, and doesn't shed light on the
FWD problem, I'll move on to that. Of course, PITA that it
would be, my next move if no one here can help will be to
restore my settings from a few weeks back (yes, I back up
religiously :-), and see if 1.0.3 will "just work".
Thanks in advance to any kind soul who has some insight!
Hadar Pedhazur
2005-Mar-05 08:00 UTC
[Asterisk-Users] FWD and SIPPHONE problems after upgrading to CVSHEAD - VERIFIED
Replying to my own post :-(
Yes, I'm top-posting, because no one ever seems to reply to my posts
anyway, I don't want to make you re-read my old post just to find out
what I'm adding.
I have _not_ solved the problem, but I reverted briefly to 1.0.3, and
I can indeed call to FWD without any problems. This is with _no
changes_ to the iax.conf between the two, so something in the recent
CVS HEAD has caused me to be able to receive calls from FWD (via
IAX2), but no longer call FWD.
I can't believe this is only happening to me, but apparently, it must
be...
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Hadar
Pedhazur
Sent: Thursday, March 03, 2005 5:18 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] FWD and SIPPHONE problems after upgrading to
CVSHEAD
I have been successfully connected (incoming and outgoing)
to FWD for a very long time. A few months ago, I changed
from SIP-based FWD service to IAX2-based, and that went fine
as well, both incoming and outgoing.
At the time, I was running Asterisk 1.0.3 Stable.
I rarely use the service, so other than noticing that I was
always successfully registered to FWD, I didn't make or
receive calls recently.
2 weeks ago, I upgraded to CVS HEAD:
Asterisk CVS-HEAD-02/21/05-09:07:50
Still didn't make or receive calls to FWD since the upgrade,
but everything else has worked flawlessly (including sixTel,
NuFone, etc.). All my softphones (SIP and IAX2) and
Sipura-2000's work perfectly too.
On to the problem... A few days ago, I signed up for an
account with SIPPhone. When I did a "sip reload", which had
the register statement, I immediately got a call "welcoming"
me, so I thought everything was fine. It wasn't.
I have been unable to make any calls to sipphone, and even
though the registration appears to work (and my.sipphone.com
shows me as "online"), all calls to my number actually claim
that I am unavailable, and go directly to voicemail.
Before I show my configs and CLI output, a few more
background data points:
I can successfully connect to sipphone with their own
download of X-Lite (pre-configured), and I can set a profile
in SJPhone by hand and it works too, both incoming and
outgoing, so I have the correct password, etc.
Today, I tested outgoing calls on FWD (actually to use the
peering to test incoming on sipphone), and my calls to FWD
are failing now as well. Incoming from FWD (via IAX2) still
works correctly. Worse, I also tried to go back to SIP-based
outgoing to FWD, and I get the same error as I do for
sipphone, so now I am starting to suspect that it's Asterisk
CVS HEAD that's possibly the problem...
Finally, the machine that is connected to both FWD and
SIPPHONE is on a public static IP address, so there are no
NAT issues involved here, and no STUN services needed
either.
OK, here is the sip.conf entry:
register=1747XXXXXXX:YYYYYYY@proxy01.sipphone.com/4321
[proxy01.sipphone.com]
type=peer
;auth=md5
secret=YYYYYYY
username=1747XXXXXXX
fromuser=1747XXXXXXX
fromdomain=proxy01.sipphone.com
host=proxy01.sipphone.com
nat=no
qualify=no
canreinvite=no
disallow=all
allow=ulaw
;context=default
;callerid="Hadar Pedhazur" <1747XXXXXXX>
(The above has been variously named sipphone, sipphone-out
and now proxy01.sipphone.com, all with the same exact
result! Also, the above has been tried with auth=md5
uncommented as well, and also no password, and
insecure=vary, etc.)
Now extensions.conf:
; Dial SIPPhone with a prefix of 76
exten => _76.,1,SetCallerID(${SIPPHONENUM})
exten => _76.,2,SetCIDName("Hadar Pedhazur")
exten => _76.,3,Dial(SIP/${EXTEN:2}@proxy01.sipphone.com)
OK, here's the output from a call:
-- Called 411@proxy01.sipphone.com
-- Got SIP response 500 "I'm terribly sorry, server error occured
(1/SL)" back from 198.65.166.131
-- SIP/proxy01.sipphone.com-78d5 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
Notice that at the end of the "Got SIP response" line, is
the correct IP address of their server, so it's finding the
correct server. As mentioned above, if I switch FWD to call
via SIP, I get the same _exact_ error message, but from
FWD's correct IP address rather than SIPPhone. This seems
very suspicious to me...
Finally, just for completeness, here is the CLI output for
attempting to call FWD via IAX2. This used to work, though I
can't say when it started failing:
-- Called fwd-gw/612
-- Call accepted by 65.39.205.121 (format ulaw)
-- Format for call is ulaw
-- IAX2/fwd-gw-4 is busy
I have called _many_ times, and every time I get an instant
"is busy" in the CLI, and I can receive calls without a
problem, so I don't think it's that they really are busy.
For now, I'm more interested in fixing the SIPPhone problem,
and if that ends up working, and doesn't shed light on the
FWD problem, I'll move on to that. Of course, PITA that it
would be, my next move if no one here can help will be to
restore my settings from a few weeks back (yes, I back up
religiously :-), and see if 1.0.3 will "just work".
Thanks in advance to any kind soul who has some insight!
Mike Matthews
2005-Mar-06 22:18 UTC
[Asterisk-Users] FWD and SIPPHONE problems after upgrading to CVSHEAD - VERIFIED
This works for me both incoming and outgoing w/Sipphone. Note there is NO username, secret entries in the peer definition. I am using * vers 1.05 register=1747nnnnnnn:pwpw@proxy01.sipphone.com/1747xxxxxxx ; note:extension in extensions.conf matches for incoming [sipphone] type=peer host=proxy01.sipphone.com dtmfmode=rfc2833 canreinvite=no disallow=all allow=ulaw allow=alaw allow=gsm allow=ilbc fromdomain=proxy01.sipphone.com qualify=no reinvite=no insecure=very nat=yes>OK, here is the sip.conf entry: > >register=1747XXXXXXX:YYYYYYY@proxy01.sipphone.com/4321 > >[proxy01.sipphone.com] >type=peer >;auth=md5 >secret=YYYYYYY >username=1747XXXXXXX >fromuser=1747XXXXXXX >fromdomain=proxy01.sipphone.com >host=proxy01.sipphone.com >nat=no >qualify=no >canreinvite=no >disallow=all >allow=ulaw >;context=default >;callerid="Hadar Pedhazur" <1747XXXXXXX> > > >