Sunr1ses / google-voice-sipsorcery-dialplans

Automatically exported from code.google.com/p/google-voice-sipsorcery-dialplans
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Dialing out on ATA Phone Rings Rings Rings but never switches #53

Closed GoogleCodeExporter closed 9 years ago

GoogleCodeExporter commented 9 years ago
I'm using an RTP300 and everything was working fine up until 2 days ago
(not sure what switched).  I get a dial tone and I ring out to a phone and
it will ring and ring and ring and after about 6 rings, it turns to a busy
tone; however, the phone I'm dialing just STARTS to ring as soon as I get
the busy tone on my ATA phone.

What's going on?  I made sure that sip1.sipsorcery.com is the proxy address
for the RTP300.  My ATA phone shows up under the bindings at sipsorcery and
I just updated the dial plan as well.

I know calls are going through as well b/c my sipgate shows all these
voicemails of when the other person picks up or the ringing of the other phone.

Any help?

Original issue reported on code.google.com by UCIR...@gmail.com on 19 May 2010 at 4:30

GoogleCodeExporter commented 9 years ago
Read Sipsorcery blog. It may be related to IP address change. If so, the things 
should 
come back to normal in about a day or so. Or, you can try replacing symbolic 
names like 
sip1.sipsorcery.com with IP address.

Original comment by mte...@gmail.com on 19 May 2010 at 5:08

GoogleCodeExporter commented 9 years ago
Still having the same issue even after I replaced the name.  Real frustrating 
as I'm
unsure why this suddenly changed.  Maybe it has to do w/ the dial plan?

Original comment by UCIR...@gmail.com on 19 May 2010 at 7:40

GoogleCodeExporter commented 9 years ago
Could you post the Console log here? I'd like to see what's going on there...

Original comment by mte...@gmail.com on 19 May 2010 at 7:45

GoogleCodeExporter commented 9 years ago
Sure just made a call to 411:

DialPlan 19:48:30:182 sip1: New call from udp:76.169.6.213:5060 successfully
authenticated by digest.
DialPlan 19:48:30:338 sip1: Using dialplan Unified for Out call to
sip:411@sip1.sipsorcery.com.
NewCall 19:48:30:400 sip1: Executing script dial plan for call to 411.
DialPlan 19:48:30:432 sip1: ** Call from
<sip:ryanstomel@sip1.sipsorcery.com>;tag=a32889193a90dd16o0 to 411 **
DialPlan 19:48:30:432 sip1: Calling 18004664411 via Google Voice
DialPlan 19:48:31:557 sip1: Google Voice home page loaded successfully.
DialPlan 19:48:31:650 sip1: Call key Hici94rMXc68rUREwW2mvAuRkgo= successfully
retrieved for uciryan@gmail.com, proceeding with callback.
DialPlan 19:48:31:666 sip1: SIP Proxy setting application server for next call 
to
user ryanstomel as udp:10.249.142.143:5070.
DialPlan 19:48:31:822 sip1: Google Voice Call to 18004664411 forwarding to
19496251175 successfully initiated, callback timeout=30s.
DialPlan 19:48:31:650 sip2: SIP Proxy setting application server for next call 
to
user ryanstomel as udp:10.249.142.143:5070.
ContactRegisterInProgress 19:48:54:136 sip2: Checking SIP Provider registration 
for
sipgate line 1.
ContactRegisterInProgress 19:48:54:136 sip2: Sending initial register for 
ryanstomel
and sipgate line 1 to udp:204.155.28.10:5060.
ContactRegisterInProgress 19:48:54:245 sip2: Initiating registration for 
ryanstomel
on sip:sipgate.com.
ContactRegistered 19:48:54:558 sip2: Contact successfully registered for 
ryanstomel
on sip:sipgate.com, expiry 60s.
DialPlan 19:49:01:824 sip1: Google Voice Call timed out waiting for callback.
DialPlan 19:49:01:964 sip1: Dial plan execution completed without answering and 
with
no last failure status.
DialPlan 19:49:01:964 sip1: UAS call failed with a response status of 480.

Original comment by UCIR...@gmail.com on 19 May 2010 at 7:49

GoogleCodeExporter commented 9 years ago
> ... sip2: Contact successfully registered for ryanstomel
on sip:sipgate.com, expiry 60s.

Your DID(s) must forward incoming calls to the same server. That is, if your 
ATA is 
registered to sip1.sipsorcery.com, incoming calls should be forwarded to sip1 
and not 
to sip2.

Change "Register Contact" in the provider from yourname@sipsorcery.com to 
yourname@sip1.sipsorcery.com.

Original comment by mte...@gmail.com on 19 May 2010 at 7:59

GoogleCodeExporter commented 9 years ago
Hot damn you're fucking great!!  Awesome job on that one!!  You're amazing!!

RESOLVED!!

Original comment by UCIR...@gmail.com on 19 May 2010 at 8:04

GoogleCodeExporter commented 9 years ago
Glad to hear that :-)

Original comment by mte...@gmail.com on 19 May 2010 at 8:12

GoogleCodeExporter commented 9 years ago
mtelis, I'm having the same error.  Not sure why again, but my console isn't 
working
either:

Monitor 16:31:26:652: basetype=console, ipaddress=*, user=ryanstomel, event=*,
request=*, serveripaddress=*, server=*, regex=.*.
Monitor 16:31:27:778: ipaddress=*, user=ryanstomel, event=*, request=*,
serveripaddress=*, server=*, regex=.*.

Original comment by UCIR...@gmail.com on 24 May 2010 at 4:33

GoogleCodeExporter commented 9 years ago
Sipsorcery server has migrated to new hosting. From now on, you can forget 
about sip1 
and sip2, because there is only one server and all DNS entries (sipsorcery.com, 
sip.sipsorcery.com, sip1.sipsorcery.com, sip2.sipsorcery.com) are pointing to 
the 
same IP address.

The migration didn't go smoothly. Right now there is a problem with MS SQL 
Azure db 
connections. Let me give you some links:

http://sipsorcery.wordpress.com/2010/05/23/migration-to-new-hosting-provider/
http://forum.sipsorcery.com/viewtopic.php?f=2&t=2457
http://twitter.com/sipsorcery

My setup works because my callback DIDs are from IPCOMMS and IPKall and I'm 
forwarding calls to myname@sipsorcery.com (as opposed to "register to DID to 
receive 
calls"). Anyway, I guess you'll have to wait (or use forwarding, if supported 
by your 
DID provider).

As to the Console: it doesn't work. You need to SSH to sipsorcery.com. I'm 
using 
putty (SSH client).

Original comment by mte...@gmail.com on 24 May 2010 at 4:44

GoogleCodeExporter commented 9 years ago
mtelis, I am having a similar error, but I am running a local version of 
SipSorcery. 

I haven't worked out the frequency of dropped calls but it goes somethinglike 
this:
Dial out, phone rings, but never gets callback from GV. After a few rings, busy 
tone. However, the number i am trying to call does get a ring and CID indicates 
the 
correct number, but when answered, dead silence. However, what is actually 
happening, is that the person answering is going straight into Sipgate 
voicemail (no 
prompts). So when they say hello, hello and then hang up, I get an email 
notification from sipgate.com of a new voicemail. I am at work, so i can post 
some 
more information tonight if helpful.

Original comment by jfla...@gmail.com on 24 May 2010 at 8:44

GoogleCodeExporter commented 9 years ago
I think I know what's going on. sys.GoogleVoiceCall initiates callback to 
Sipgate but 
for some reason the callback is not forwarded to your local Sipsorcery server. 
If you 
had VM on Sipgate disabled, your call would die at this stage. You have VM on 
Sipgate 
and it answers callback from GV. So, callback is answered and Google starts 
dialing 
the other leg (phone you're trying to call) while VM is playing your greeting. 
By the 
time the person you're calling answers the call, VM greeting is over and they 
hear 
dead silence (VM is trying to take a message).

Your phone keeps ringing until callback time-out, then you get the busy tone.

So, the root of this problem is missing callback. (You also may want to disable 
VM on 
Sipgate to make sure the target phone not called if this happens). Maybe you 
lose the 
callback due to IP address change? Is your local IP static or you're using 
DynDNS or 
alike?

Original comment by mte...@gmail.com on 25 May 2010 at 2:35