"remote end hungup" after switching from GV proc

Found something wrong ?
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sipsorjoef
Posts: 20
Joined: Sat Mar 05, 2011 4:17 pm

"remote end hungup" after switching from GV proc

Post by sipsorjoef » Mon Mar 24, 2014 5:35 pm

Hi!
About a month ago, I switched my GV processing from the sys.GoogleVoiceCall style of using GV to the
GV sip provider : "sys.Dial(MyGV)" and all was working until I tried to make a call today.

Now I get the calling phone goes dead just as the called phone starts ringing and the caller is disconnected.

In the call logs I get: "remote end hungup"

Since it WAS working I can't see what may have changed to make it stop working.

Any clues? thanks!

Joe F
Here's the trace: (with the guilty changed to protect) :-)

NATKeepAlive 17:27:31:148 sip1(19928): Requesting NAT keep-alive from proxy socket udp:67.222.131.147:5060 to udp:50.139.96.10:5080.
DialPlan 17:27:33:742 sip1(22560): New call from udp:50.139.96.10:5080 successfully authenticated by digest.
DialPlan 17:27:33:789 sip1(22560): Using dialplan default for Out call to sip:15031234567@sipsorcery.com:5060.
NewCall 17:27:33:835 sip1(22560): Executing script dial plan for call to 15031234567.
DialPlan 17:27:33:914 sip1(22560): INside starting DEFAULT dialplan
DialPlan 17:27:33:992 sip1(22560): New incoming call received to 15031234567 from <sip:mysipclient75633@sipsorcery.com>;tag=SP16a55f7727bd8d53.
DialPlan 17:27:33:992 sip1(22560): Invoking GV
DialPlan 17:27:33:992 sip1(22560): Commencing Dial with: My215GV.
DialPlan 17:27:34:007 sip1(22560): Attempting to locate a provider for call leg: sip:15031234567@My215GV.
DialPlan 17:27:34:007 sip1(22560): ForkCall commencing call leg to 15031234567@My215GV.
DialPlan 17:27:34:007 sip1(22560): Creating Google Voice user agent for 15031234567@My215GV.
DialPlan 17:27:34:007 sip1(22560): SDP on Google Voice call could not be mangled, using original RTP socket of 50.139.96.10:16610.
DialPlan 17:27:34:007 sip1(22560): UAS call progressing with Ringing.
DialPlan 17:27:34:023 sip1(22560): Logging into google.com for joeeff@gmail.com.
DialPlan 17:27:36:367 sip1(22560): Google Voice pre-login page loaded successfully.
DialPlan 17:27:36:382 sip1(22560): GALX key wMw5UwJBqb0 successfully retrieved.
DialPlan 17:27:37:460 sip1(22560): Google Voice home page loaded successfully.
DialPlan 17:27:37:460 sip1(22560): Call key P/KXEq+DgsyjyAElHtuq+DI/5I4= successfully retrieved for joeeff@gmail.com, proceeding with callback.
DialPlan 17:27:37:476 sip1(17756): SIP Proxy setting application server for next call to user sipsorjoef as udp:67.222.131.147:5070.
DialPlan 17:27:37:898 sip1(22560): Google Voice Call to 15031234567 initiated, callback #206anIPKall#, phone type 3, timeout 60s.
NATKeepAlive 17:27:41:320 sip1(19928): Requesting NAT keep-alive from proxy socket udp:67.222.131.147:5060 to udp:50.139.96.10:5080.
DialPlan 17:27:49:383 sip1(17756): SIP Proxy directing incoming call for user sipsorjoef to application server udp:67.222.131.147:5070.
DialPlan 17:27:49:445 sip1(22560): Google Voice Call callback received.
DialPlan 17:27:49:461 sip1(22560): Call leg for Google Voice call to 15031234567@My215GV answered.
DialPlan 17:27:49:461 sip1(22560): Cancelling all call legs for ForkCall app.
DialPlan 17:27:49:461 sip1(22560): Answering client call with a response status of 200.
DialPlan 17:27:49:508 sip1(22560): Dial command was successfully answered in 15.50s.
DialPlan 17:27:49:508 sip1(22560): Dialplan cleanup for sipsorjoef.
DialPlan 17:27:49:711 sip1(22560): Dial plan execution completed with normal clearing.
Registrar 17:27:57:961 sip1(19928): Binding update request for mysipclient75633@sipsorcery.com from udp:50.139.96.10:5080, expiry requested 120s granted 120s.
NATKeepAlive 17:28:01:570 sip1(19928): Requesting NAT keep-alive from proxy socket udp:67.222.131.147:5060 to udp:50.139.96.10:5080.

User avatar
Aaron
Site Admin
Posts: 4554
Joined: Thu Jul 12, 2007 12:13 am

Re: "remote end hungup" after switching from GV proc

Post by Aaron » Tue Mar 25, 2014 9:45 am

There doesn't seem to be anything wrong according to your dialplan trace. My guess would be something was up with GV or the callback provider. Has the call subsequently started working?

sipsorjoef
Posts: 20
Joined: Sat Mar 05, 2011 4:17 pm

Re: "remote end hungup" after switching from GV proc

Post by sipsorjoef » Tue Mar 25, 2014 4:06 pm

Tried it again and paid more attention to what's happening.

It's not just the ringing that stops. It's the sound entirely for the caller that
stops when the ringing for the callee begins.

Make sense? I've seen this explained before it has to do with NAT maybe?
thanks
Joe F

User avatar
Aaron
Site Admin
Posts: 4554
Joined: Thu Jul 12, 2007 12:13 am

Re: "remote end hungup" after switching from GV proc

Post by Aaron » Wed Mar 26, 2014 10:50 am

NAT is the number one culprit for audio issues on SIP calls. The very first thing to try is to reboot your router to get it to clear its NAT mappings and see if that helps.

sipsorjoef
Posts: 20
Joined: Sat Mar 05, 2011 4:17 pm

Re: "remote end hungup" after switching from GV proc

Post by sipsorjoef » Thu Mar 27, 2014 3:01 am

Aaron,

thanks and thanks for your patience and help.

I did more testing today. The suspect router was a Netgear 6200 (ac-wireless) but it's not the problem
I tried two other routers: my previous Trendnet 639 and the routing capability in a spa3102 with no
change. I also plugged the OBIhai directly into my comcast modem and that didn't help.

I have also used a sip client on my Nexus 5 and that ALWAYS works whether I use the above network wifi
or switch to AT&T LTE.

So I'm thinking it is the ObiHai 110 device.. I writing this before I research known problems between
ObiHai / Sipsorcery and/or GV. It does work when I go back to the XMPP way that we're losing in May.
I'm thinking that I didn't have it working , as I said at the start, and that's always been the problem.

thanks again.
Let me know if you have any knowledge about OBI issue.
cheers
Joe F

sipsorjoef
Posts: 20
Joined: Sat Mar 05, 2011 4:17 pm

Re: "remote end hungup" after switching from GV proc

Post by sipsorjoef » Fri Mar 28, 2014 12:18 am

I wrote to Obihai and got this missive :

For your OBi

We changed the DNS servers that your OBi uses.
We also changed SP2 's RTP
Please try to make phone calls out from phone handset connected to OBi's
PHONE port.
Can the remote end hear you ok ?

-Obihai Support Team

and .... problem solved!

thanks again

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