Closed GoogleCodeExporter closed 9 years ago
Oh, and I am a stay at home mom of twins so $$$ is a huge issue right now! But
I would literally pay someone to look at this for me. I've been struggling
with it for months and I'm amazed I got this far but just would like a regular
working phone. Thank you times a million to whoever can help me fix this.
Original comment by mirandaj...@gmail.com
on 30 Sep 2010 at 9:56
On Sipsorcery website, go to the "Console" and click on "Connect". The console
will display:
Monitor 03:12:04:084: basetype=console, ipaddress=*, user=mtelis, event=*,
request=*, serveripaddress=*, server=*, regex=.*.
which means you're connected. Call your Google Voice number from some other
telephone, select the messages that appear in Console (it's called Console
trace), copy them by pressing Ctrl-C and post them here. You may want to
replace any private data (like the phone number you called from) with 'x'.
Original comment by mte...@gmail.com
on 1 Oct 2010 at 3:14
Well I must be way off because I didn't even see the phone number I called
from. I used another cell and called through and got my GV voicemail. Sorry
if I copied and pasted too much info.
Monitor 03:17:25:109: basetype=console, ipaddress=*, user=m****, event=*,
request=*, serveripaddress=*, server=*, regex=.*.
Registrar 03:17:30:468 sip1: Authentication required for m****@sipsorcery.com
from udp:76.122.77.20:5060.
Registrar 03:17:30:843 sip1: Binding update request for m****@sipsorcery.com
from udp:76.122.77.20:5060, expiry requested 3600s granted 3600s.
RegisterSuccess 03:17:30:905 sip1: Registration successful for
m****@sipsorcery.com from udp:76.122.77.20:5060 (proxy=udp:69.59.142.213:5060),
expiry 3600s.
NATKeepAlive 03:17:33:296 sip1: Requesting NAT keep-alive from proxy socket
udp:69.59.142.213:5060 to udp:76.122.77.20:5060.
NATKeepAlive 03:17:43:608 sip1: Requesting NAT keep-alive from proxy socket
udp:69.59.142.213:5060 to udp:76.122.77.20:5060.
NATKeepAlive 03:17:53:905 sip1: Requesting NAT keep-alive from proxy socket
udp:69.59.142.213:5060 to udp:76.122.77.20:5060.
NATKeepAlive 03:18:04:186 sip1: Requesting NAT keep-alive from proxy socket
udp:69.59.142.213:5060 to udp:76.122.77.20:5060.
NATKeepAlive 03:18:14:373 sip1: Requesting NAT keep-alive from proxy socket
udp:69.59.142.213:5060 to udp:76.122.77.20:5060.
NATKeepAlive 03:18:24:576 sip1: Requesting NAT keep-alive from proxy socket
udp:69.59.142.213:5060 to udp:76.122.77.20:5060.
NATKeepAlive 03:18:34:842 sip1: Requesting NAT keep-alive from proxy socket
udp:69.59.142.213:5060 to udp:76.122.77.20:5060.
NATKeepAlive 03:18:45:029 sip1: Requesting NAT keep-alive from proxy socket
udp:69.59.142.213:5060 to udp:76.122.77.20:5060.
NATKeepAlive 03:18:55:201 sip1: Requesting NAT keep-alive from proxy socket
udp:69.59.142.213:5060 to udp:76.122.77.20:5060.
DialPlan 03:19:04:935 sip1: No dialplan specified for incoming call to
m****@sipsorcery.com, registered bindings will be used.
DialPlan 03:19:04:951 sip1: Forwarding incoming call for m****@sipsorcery.com
to 1 bindings.
NewCall 03:19:04:951 sip1: Executing script dial plan for call to m****.
DialPlan 03:19:04:982 sip1: Commencing Dial with: m****@sipsorcery.com.
DialPlan 03:19:04:997 sip1: Call leg is for local domain looking up bindings
for m****@sipsorcery.com for call leg m****@sipsorcery.com.
DialPlan 03:19:05:013 sip1: 1 found for m****@sipsorcery.com.
DialPlan 03:19:05:013 sip1: ForkCall commencing call leg to
sip:m****@76.122.77.20:5060.
DialPlan 03:19:05:013 sip1: SIPClientUserAgent Call using alternate outbound
proxy of udp:69.59.142.213:5060.
DialPlan 03:19:05:013 sip1: Switching to sip:m****@76.122.77.20:5060 via
udp:69.59.142.213:5060.
DialPlan 03:19:05:013 sip1: SDP on UAC call had public IP not mangled, RTP
socket 204.155.29.58:19724.
NATKeepAlive 03:19:05:435 sip1: Requesting NAT keep-alive from proxy socket
udp:69.59.142.213:5060 to udp:76.122.77.20:5060.
NATKeepAlive 03:19:15:669 sip1: Requesting NAT keep-alive from proxy socket
udp:69.59.142.213:5060 to udp:76.122.77.20:5060.
NATKeepAlive 03:19:25:841 sip1: Requesting NAT keep-alive from proxy socket
udp:69.59.142.213:5060 to udp:76.122.77.20:5060.
DialPlan 03:19:35:950 sip1: Client call cancelled halting dial plan.
DialPlan 03:19:35:950 sip1: Dialplan call was terminated by client side due to
ClientCancelled.
DialPlan 03:19:35:950 sip1: Cancelling all call legs for ForkCall app.
DialPlan 03:19:35:950 sip1: Cancelling forwarded call leg, sending CANCEL to
sip:m****@76.122.77.20:5060.
DialPlan 03:19:35:950 sip1: Dial command was halted by cancellation of client
call after 30.94s.
DialPlan 03:19:35:966 sip1: Dialplan cleanup for m****.
NATKeepAlive 03:19:36:106 sip1: Requesting NAT keep-alive from proxy socket
udp:69.59.142.213:5060 to udp:76.122.77.20:5060.
DialPlan 03:19:36:184 sip1: Dial plan execution completed with normal clearing.
NATKeepAlive 03:19:56:621 sip1: Requesting NAT keep-alive from proxy socket
udp:69.59.142.213:5060 to udp:76.122.77.20:5060.
Original comment by mirandaj...@gmail.com
on 1 Oct 2010 at 3:27
It seems that incoming calls can not reach your PAP2T and most certainly it's
because of your router. Put the PAP2T to DMZ of your router and try again. If
it works, remove DMZ and forward TCP/UDP port 5060 to the PAP2T.
Original comment by mte...@gmail.com
on 1 Oct 2010 at 7:49
[deleted comment]
[deleted comment]
Wow, worked perfectly! I can't thank you enough!
Original comment by mirandaj...@gmail.com
on 1 Oct 2010 at 4:50
Ok nevermind, after two incoming calls working fine, my internet cut out and I
reset the wireless router by pushing the button on the back. Over the past few
months I have found with just the outgoing calls that after a few uses my
internet will cut out and I usually reset the router and after a few minutes
everything is up and running again.
Now I can still do outgoing calls but this is what I'm getting for incoming.
NATKeepAlive 17:26:11:322 sip1: Requesting NAT keep-alive from proxy socket
udp:69.59.142.213:5060 to udp:76.122.77.20:5060.
DialPlan 17:26:15:729 sip1: No dialplan specified for incoming call to
miranda@sipsorcery.com, registered bindings will be used.
DialPlan 17:26:15:744 sip1: Forwarding incoming call for miranda@sipsorcery.com
to 2 bindings.
NewCall 17:26:15:744 sip1: Executing script dial plan for call to miranda.
DialPlan 17:26:15:791 sip1: Commencing Dial with: miranda@sipsorcery.com.
DialPlan 17:26:15:822 sip1: Call leg is for local domain looking up bindings
for miranda@sipsorcery.com for call leg miranda@sipsorcery.com.
DialPlan 17:26:15:822 sip1: 2 found for miranda@sipsorcery.com.
DialPlan 17:26:15:822 sip1: Call leg sip:miranda@76.122.77.20:5060 already
added duplicate ignored.
DialPlan 17:26:15:822 sip1: ForkCall commencing call leg to
sip:miranda@76.122.77.20:5060.
DialPlan 17:26:15:822 sip1: SIPClientUserAgent Call using alternate outbound
proxy of udp:69.59.142.213:5060.
DialPlan 17:26:15:822 sip1: Switching to sip:miranda@76.122.77.20:5060 via
udp:69.59.142.213:5060.
DialPlan 17:26:15:822 sip1: SDP on UAC call had public IP not mangled, RTP
socket 204.155.29.57:16964.
NATKeepAlive 17:26:21:510 sip1: Requesting NAT keep-alive from proxy socket
udp:69.59.142.213:5060 to udp:76.122.77.20:5060.
NATKeepAlive 17:26:31:760 sip1: Requesting NAT keep-alive from proxy socket
udp:69.59.142.213:5060 to udp:76.122.77.20:5060.
NATKeepAlive 17:26:42:025 sip1: Requesting NAT keep-alive from proxy socket
udp:69.59.142.213:5060 to udp:76.122.77.20:5060.
DialPlan 17:26:47:853 sip1: Dialplan cleanup for miranda.
DialPlan 17:26:48:165 sip1: Dial plan execution completed without answering and
a last failure status of TemporarilyUnavailable Timeout, no response from
server.
DialPlan 17:26:48:165 sip1: UAS call failed with a response status of 480 and
Timeout, no response from server.
NATKeepAlive 17:26:52:368 sip1: Requesting NAT keep-alive from proxy socket
udp:69.59.142.213:5060 to udp:76.122.77.20:5060.
DialPlan 17:26:57:072 sip1: No dialplan specified for incoming call to
miranda@sipsorcery.com, registered bindings will be used.
DialPlan 17:26:57:103 sip1: Forwarding incoming call for miranda@sipsorcery.com
to 2 bindings.
NewCall 17:26:57:103 sip1: Executing script dial plan for call to miranda.
DialPlan 17:26:57:134 sip1: Commencing Dial with: miranda@sipsorcery.com.
DialPlan 17:26:57:150 sip1: Call leg is for local domain looking up bindings
for miranda@sipsorcery.com for call leg miranda@sipsorcery.com.
DialPlan 17:26:57:165 sip1: 2 found for miranda@sipsorcery.com.
DialPlan 17:26:57:165 sip1: Call leg sip:miranda@76.122.77.20:5060 already
added duplicate ignored.
DialPlan 17:26:57:165 sip1: ForkCall commencing call leg to
sip:miranda@76.122.77.20:5060.
DialPlan 17:26:57:165 sip1: SIPClientUserAgent Call using alternate outbound
proxy of udp:69.59.142.213:5060.
DialPlan 17:26:57:165 sip1: Switching to sip:miranda@76.122.77.20:5060 via
udp:69.59.142.213:5060.
DialPlan 17:26:57:165 sip1: SDP on UAC call had public IP not mangled, RTP
socket 204.155.29.54:19454.
Original comment by mirandaj...@gmail.com
on 1 Oct 2010 at 5:31
Once again, most certainly the problem is in your router. In order to make
sure, try installing a proven softphone like X-Lite on your PC and check
whether it can receive incoming calls or fails miserably just like the PAP2T.
To make the experiment clean, turn off the PAP2T and reboot the router before
you start the softphone.
There also is a good article by Sipsorcery principal developer:
http://sipsorcery.wordpress.com/2009/12/15/the-bullet-proof-solution-to-one-way-
audio-buy-a-new-router/
Original comment by mte...@gmail.com
on 1 Oct 2010 at 7:21
Two more things you can try to overcome the problem with *existing* router:
1. Enable keep-alive in the PAP2T. It should be on the Line page, in the NAT
settings (NAT Mapping Enable -> yes, NAT Keep Alive Enable -> yes). Check if it
helps, let the things running for a few minutes and try receiving a test call.
2. If the above didn't help, try enabling keep-alives on Sipsorcery side. It's
in the "SIP accounts" page, open your SIP account settings and check "Keep
alives" box.
Original comment by mte...@gmail.com
on 2 Oct 2010 at 3:09
Ok, I haven't tried X-Lite yet but I did want to let you know that my Sipgate
softphone has always worked perfectly fine for incoming and outgoing calls.
I tried both Enabling the keep-alive in the PAP2T and in Sipsorcery, and every
combination inbetween (I actually saw that the keep-alives option was checked
to begin with when I logged into Sipsorcery).
I tried having the PAP2T in the DMZ again and it just works like a charm. Why
does this work so well? Wish I understood more about what I was doing here...
Original comment by mirandaj...@gmail.com
on 2 Oct 2010 at 3:38
Your PAP2T is behind a NAT. The router has just one external IP address, all
local nodes (including your PC and the PAP2T) work thru this IP. With outbound
calls, the things are more or less simple, but what should the router do when
Sipsorcery contacts it about incoming call? Should it pass this "unsolicited"
request to your PC or the PAP2T?
DMZ is about unsolicited incoming connections. When the router gets unsolicited
incoming connection request, it forwards it to DMZ node on the local network.
Since we know that the PAP2T is listening only on the port 5060, it's wise to
forward only this port instead of all ports (that's what DMZ does).
Hope it's more or less clear now.
Original comment by mte...@gmail.com
on 2 Oct 2010 at 4:28
One more thing: if everything works when you put the PAP2T into DMZ and you do
not have other servers (like FTP or Web server) on your home network, just
leave it there. DMZ is not acceptable only if you have several servers running
on different nodes (computers) within the same LAN. If you have only one
server, it's perfectly okay to put it into DMZ.
Original comment by mte...@gmail.com
on 2 Oct 2010 at 4:43
Any updates Miranda? I'm trying to clean up the ticket queue.
Original comment by easter...@gmail.com
on 8 Feb 2011 at 1:11
Closing due to inactivity.
Original comment by easter...@gmail.com
on 9 Feb 2011 at 10:45
Original issue reported on code.google.com by
mirandaj...@gmail.com
on 30 Sep 2010 at 9:37