Help with Dialplan

Getting started with the SIP Sorcery
MikeTelis
Posts: 1582
Joined: Wed Jul 30, 2008 6:48 am

Re: Help with Dialplan

Post by MikeTelis » Fri Dec 03, 2010 7:35 am

Go to the Console, click on "Connect" and check if you see incoming call-related messages when you call your GV number. If you don't see any, probably you forgot to "tick" forwarding number in GV / Settings / Voice Settings / Phones / Forwards to, put GV in Do Not Disturb mode or Ring schedule in the Phone / Edit / Advanced prohibits your phone to ring.

If you do see activity in the console but the call doesn't reach your ATA most certainly it's related to your router and/or ATA settings. Please show me what you get in the Console and I'll try to guess what it might be.

stav
Posts: 36
Joined: Fri Aug 14, 2009 4:20 pm

Re: Help with Dialplan

Post by stav » Fri Dec 03, 2010 2:01 pm

Mike,
Here is my Console report.
"Monitor 13:29:06:066: basetype=console, ipaddress=*, user=MYSSACOUNTID, event=*, request=*, serveripaddress=*, server=*, regex=.*.
NATKeepAlive 13:29:08:034 sip1(3756): Requesting NAT keep-alive from proxy socket udp:69.59.142.213:5060 to udp:65.2.58.242:51809.
DialPlan 13:29:08:956 sip1(7620): Using dialplan default for In call to sip:MYSSACOUNTID@sipsorcery.com;rinstance=918993.
NewCall 13:29:08:988 sip1(7620): Executing script dial plan for call to MYSSACOUNTID.
DialPlan 13:29:09:081 sip1(7620): ** Call from "MYSIPGATEPHONENUMBER" <sip:1MYGVNUMBER@sipgate.com>;tag=as31a30603 to MYSSACOUNTID **
DialPlan 13:29:09:113 sip1(7620): Caller's number: '1MYGVNUMBER'
DialPlan 13:29:09:113 sip1(7620): Commencing Dial with: MYSSACOUNTID@local[fu=1MYGVNUMBER].
DialPlan 13:29:09:128 sip1(7620): No sip account could be found for local call leg MYSSACOUNTID@local[fu=1MYGVNUMBER].
DialPlan 13:29:09:128 sip1(7620): The dial string did not result in any call legs.
DialPlan 13:29:09:128 sip1(7620): Dialplan cleanup for MYSSACOUNTID.
DialPlan 13:29:09:331 sip1(7620): Dial plan execution completed without answering and with no last failure status.
DialPlan 13:29:09:331 sip1(7620): UAS call failed with a response status of 480.
NATKeepAlive 13:29:18:253 sip1(3756): Requesting NAT keep-alive from proxy socket udp:69.59.142.213:5060 to udp:65.2.58.242:51809.
NATKeepAlive 13:29:28:581 sip1(3756): Requesting NAT keep-alive from proxy socket udp:69.59.142.213:5060 to udp:65.2.58.242:51809.
NATKeepAlive 13:29:38:831 sip1(3756): Requesting NAT keep-alive from proxy socket udp:69.59.142.213:5060 to udp:65.2.58.242:51809."

I double checked my GV Voice settings and I do forward to my sipgate phone number and no, I dont have "Do not disturb" checked. The ring schedule is set as per default to ring all weekdays and weekends.

Any help welcome,
Thanks

stav
Posts: 36
Joined: Fri Aug 14, 2009 4:20 pm

Re: Help with Dialplan

Post by stav » Fri Dec 03, 2010 5:48 pm

Strange,
I have no SIP Bindings anymore.! Something wrong with my username?
Admin,
can you hellp here please?

MikeTelis
Posts: 1582
Joined: Wed Jul 30, 2008 6:48 am

Re: Help with Dialplan

Post by MikeTelis » Fri Dec 03, 2010 6:36 pm

Code: Select all

DialPlan 13:29:09:113 sip1(7620): Commencing Dial with: MYSSACOUNTID@local[fu=1MYGVNUMBER].
DialPlan 13:29:09:128 sip1(7620): No sip account could be found for local call leg MYSSACOUNTID@local[fu=1MYGVNUMBER].
Apparently something is wrong with your the SIP account. You need to PM to Aaron with your SS login name and SS SIP account name. Sorry, I can't help you because I do not have access to the account database.

DaveSin
Posts: 20
Joined: Wed Sep 23, 2009 12:59 pm

Re: Help with Dialplan

Post by DaveSin » Sat Dec 04, 2010 4:07 pm

stav wrote:I did exactly what Mike suggested but still deosnt work. Here is what I want to do and here is where I'm stuck:
I want to use GV number for out and in calls and I want people to call me to my GV number and have my phone connected in Line 1 (Sipsorcery) of the ATA to ring. Now, SS seems OK, I'm registered with sipgate and it appears in the bindings. I can dial out and it works great, showing my GV number. But, when I try to call my GV number from another phone not defined in my GV account settings, it rings 4 times and it goes to the GV voice mail. I should add that my Sipgate number is verified and checked in GV account.
I copied the dial plan as per Mike's recommendations in this thread and did (at least I believe so) all necessary changes and editing. Can someone have a look at it and tell me what's wrong with the syntax or anything else?
Thx,
Stavros
Dial Plan
REA_CODE = '561' # my area code
GV_USER = 'username@gmail.com' # my GV e-mail address (user@gmail.com)
GV_PASS = 'GV password' # my GV password
CB_NUMBER = '1aaaxxxyyyy' # my 11-digit sipgate number
TIME_OUT = 15 # callback time-out (in seconds)

# Uncomment 3 lines below and insert your primary GV phone number, your secondary
# GV account name and password, respectively. The second account will be used to call
# your primary GV number so that you could check voicemail from your ATA
# Note that your 2nd account must be configured with the same callback number(s) as primary

GV_NUMBER = '1561.......' # My primary GV number
VM_USER = 'username@gmail.com' # my secondary GV e-mail address (user@gmail.com)
VM_PASS = 'GV password' # my secondary GV password

SPEED_DIAL = { # my speed dial numbers
'1' => '19879879876', # Mom
'123' => '12345678901', # Work
'45' => '17479876543', # Gizmo BFF
'411' => '8004664411', # Google 411
'266' => '4153767253@podlinez.net', # CNN Headlines
}

# CNAM table: number in ENUM format => caller's name

CNAM = {
'12125551212' => 'Dear mom',
'12153332211' => 'Bratty kid',
}

# Uncomment next line and insert your White Pages API key, if you have it
# WP_key = 'Your_White_Pages_API_key_here' # White Pages API key

begin
sys.Log "** Call from #{req.Header.From} to #{req.URI.User} **"
sys.ExtendScriptTimeout(15) # preventing long running dialscript time-out

if sys.Out # if outbound call
num = req.URI.User.to_s # Get a string copy of the number to dial

num = SPEED_DIAL[num] || num # Substitute with speed dial entry, if any

case num
when /@/ then sys.Dial num # URI dialing
when /^[2-9]\d{6}$/ # Local call, 7-digit number
num = '1'+ AREA_CODE + num # prefix it with country and area code
when /^[01]?([2-9]\d{9})/ # US number with or without country code
num = '1' + $1 # add country code and truncate number to 10-digit
when /^(011|00|\+)(\d{9,})/ # international number
num = '+' + $2 # GoogleVoiceCall works with '+' prefix only
else sys.Respond 603, 'Wrong number, check & dial again'
end

user, pass = GV_USER, GV_PASS # assume it's not VM call
user, pass = VM_USER, VM_PASS if defined?(GV_NUMBER) && num == GV_NUMBER
The behavior you are experiencing seems to be O.K, since if the calls are not answered after 4-rings, you are suppose to get the GV voicemail (after 4-rings). I might be wrong here, but I do not see a problem. Are you saying that if the number the person is calling from is "defined in my GV account settings", it will ring more than 4 times and/or not go in the GV Voice mail system after 4-rings? I'm confused.....

stav
Posts: 36
Joined: Fri Aug 14, 2009 4:20 pm

Re: Help with Dialplan

Post by stav » Sat Dec 04, 2010 4:44 pm

Apparently there is a problem with my SS account and Admin is looking into it.
Thanks,
Stav

stav
Posts: 36
Joined: Fri Aug 14, 2009 4:20 pm

Re: Help with Dialplan

Post by stav » Sun Dec 05, 2010 3:45 pm

Now I can register my SIP again, I can see it under SIP Bindings, No idea what happened. Anyways, problem persists. When I try to call my GVPHONENUMBER (from a phone that's not declared in the GV settings), then my phone that's connected in line 1 of my ATA never rings and after 4 rings it goes in my voice mail.
For any useful purposes, here is my console log.
Thanks for helping,
Stav
Monitor 15:01:38:178: basetype=console, ipaddress=*, user=MYSSLOGGINNAME, event=*, request=*, serveripaddress=*, server=*,

regex=.*.
NATKeepAlive 15:01:46:771 sip1(3756): Requesting NAT keep-alive from proxy socket udp:69.59.xxx.xxx:5060 to

udp:65.11.xxx.xxx:5060.
DialPlan 15:02:18:052 sip1(1220): New call from udp:65.11.xxx.xxx:5060 successfully authenticated by digest.
DialPlan 15:02:18:067 sip1(1220): Using dialplan default for Out call to sip:1561xxxxxxx@sip1.sipsorcery.com.
NewCall 15:02:18:083 sip1(1220): Executing script dial plan for call to 1561xxxxxxx.
DialPlan 15:02:18:161 sip1(1220): ** Call from "sipsorcery" <sip:MYSIPUSERNAME@sip1.sipsorcery.com>;tag=86fb3caf7ab226fao0 to 1561XXXXXXX.
DialPlan 15:02:18:161 sip1(1220): Calling 1561xxxxxxx via Google Voice
DialPlan 15:02:18:161 sip1(1220): SDP on GoogleVoiceCall call had RTP socket mangled from 192.XXX.X.XXX:16474 to
65.xx.xxx.xxx:16474.
DialPlan 15:02:18:161 sip1(1220): UAS call progressing with Ringing.
DialPlan 15:02:18:161 sip1(1220): Logging into google.com for MYGMAIL@gmail.com.
DialPlan 15:02:18:208 sip1(1220): Google Voice pre-login page loaded successfully.
DialPlan 15:02:18:224 sip1(1220): GALX key XXXXXXXXXXX successfully retrieved.
DialPlan 15:02:19:958 sip1(1220): Google Voice home page loaded successfully.
DialPlan 15:02:19:989 sip1(1220): Call key XXXXXXXX/XXXXXXXXXXXXXXXXXX= successfully retrieved for MYGMAIL@gmail.com,
proceeding with callback.
DialPlan 15:02:19:989 sip1(1704): SIP Proxy setting application server for next call to user MYSSLOGINNAME as
udp:69.59.XXX.XXX:5070.
DialPlan 15:02:20:083 sip1(1220): Google Voice Call to 1561XXXXXXX forwarding to MYSIPGATEPHONENUMBER successfully initiated,
callback timeout=30s.
DialPlan 15:02:23:645 sip1(1220): Google Voice Call callback received.
DialPlan 15:02:23:645 sip1(1220): Answering client call with a response status of 200.
DialPlan 15:02:23:739 sip1(1220): Google Voice Call was successfully answered in 5.58s.
DialPlan 15:02:23:739 sip1(1220): Dialplan cleanup for MYSSLOGINNAME.
DialPlan 15:02:24:114 sip1(1220): Dial plan execution completed with normal clearing.
NATKeepAlive 15:02:27:833 sip1(3756): Requesting NAT keep-alive from proxy socket udp:69.59.XXX.XXX:5060 to
udp:65.11.XXX.XXX:5060.
DialPlan 15:02:31:473 sip1(1220): Matching dialogue found for BYE to sip:69.59.XXX.XXX:5060 from udp:69.59.XXX.XXX:5060.
NATKeepAlive 15:02:38:051 sip1(3756): Requesting NAT keep-alive from proxy socket udp:69.59.XXX.XXX:5060 to
udp:65.11.XXX.XXX:5060.
DialPlan 15:02:47:036 sip1(1220): Using dialplan default for In call to sip:MYSIPUSERNAME@sipsorcery.com;rinstance=580221.
NewCall 15:02:47:051 sip1(1220): Executing script dial plan for call to MYSIPUSERNAME.
DialPlan 15:02:47:145 sip1(1220): ** Call from "MYSIPGATEPHONENUMBER" <sip:561MYGOOLEPHONENUMBER@sipgate.com>;tag=as654713a7 to MYSIPUSERNAME **
DialPlan 15:02:47:145 sip1(1220): Commencing Dial with: MYSSLOGINNAME@local.
DialPlan 15:02:47:208 sip1(1220): No sip account could be found for local call leg MYSSLOGINNAME@local.
DialPlan 15:02:47:208 sip1(1220): The dial string did not result in any call legs.
DialPlan 15:02:47:208 sip1(1220): Dialplan cleanup for MYSSLOGINNAME.
DialPlan 15:02:47:645 sip1(1220): Dial plan execution completed without answering and with no last failure status.
DialPlan 15:02:47:645 sip1(1220): UAS call failed with a response status of 480.
NATKeepAlive 15:02:48:348 sip1(3756): Requesting NAT keep-alive from proxy socket udp:69.59.XXX.XXX:5060 to

udp:65.11.XXX.XXX:5060.

MikeTelis
Posts: 1582
Joined: Wed Jul 30, 2008 6:48 am

Re: Help with Dialplan

Post by MikeTelis » Sun Dec 05, 2010 4:46 pm

NewCall 15:02:47:051 sip1(1220): Executing script dial plan for call to MYSIPUSERNAME.
DialPlan 15:02:47:145 sip1(1220): ** Call from "MYSIPGATEPHONENUMBER" <sip:561MYGOOLEPHONENUMBER@sipgate.com>;tag=as654713a7 to MYSIPUSERNAME **
DialPlan 15:02:47:145 sip1(1220): Commencing Dial with: MYSSLOGINNAME@local.
DialPlan 15:02:47:208 sip1(1220): No sip account could be found for local call leg MYSSLOGINNAME@local.
See, your dialplan is forwarding incoming call to MYSSLOGINNAME while your ATA is registered to MYSIPUSERNAME. No wonder the phone doesn't ring! You need to change the dialplan so it would forward incoming calls to MYSIPUSERNAME, something like this:

sys.Dial("#{req.URI.User}@local")

stav
Posts: 36
Joined: Fri Aug 14, 2009 4:20 pm

Re: Help with Dialplan

Post by stav » Sun Dec 05, 2010 5:42 pm

Great,
I added the line exactly as you wrote it just before the
else #sys.out,
then edited the user to my SSUSERID and it works great for in and out.
Thanks very much for you help,
Stavros

MikeTelis
Posts: 1582
Joined: Wed Jul 30, 2008 6:48 am

Re: Help with Dialplan

Post by MikeTelis » Sun Dec 05, 2010 6:37 pm

Actually, you could have left the "In" dial plan field in your SIP account settings blank and it would do the same thing :)
Last edited by MikeTelis on Sun Dec 05, 2010 6:48 pm, edited 1 time in total.

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