Newbie: Help with Carrier setup (Gradwell)

[cpsuk]

Hi there we are running a VPS - Kernel Version - 2.6.35.4-cloudPBX (SMP) Distro Name GoAutoDial CE 3.0
dual core Intel(R) Xeon(R) CPU E5606 @ 2.13GHz

we are trying to setup our Gradwell SIP outbound trunk and are having problems with a working dialplan, does anyone know what will work

*tried posting config information but we are unable to post because of url links looking like spam as we are a new user

[cpsuk]

Hi people, please be kind this is the first time configuring a dialer system!
And we are having problems connecting to calls from our dialer, our SIP trunk is authenticated via IP and has been configured in the carrier settings

on the dialer and we added a line of code to our /etc/asterisk/extensions.conf = "SIPtrunk=SIP/ sip_trunk_gradwell_(om" ; SIP trunk also added the

following = "exten => _0.,1,SetCallerID(OUR CLI)
exten => _0.,2,Dial(SIP/${EXTEN}@sip_trunk_gradwell_(om)" we also updated the Server IP. We also added "externip = 88.208.xxx.xxx" to our

/etc/asterisk/sip.conf

[cpsuk]

When we input "sip show peers" we get the following

Connected to Asterisk 1.4.39.1-vici.go RPM by demian@goautodial currently running on dialer (pid = 1632)
Verbosity is at least 21
dialer*CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
8001/8001 86.26.xxx.xxx D N 4168 OK (136 ms)
Gradwell 109.224.232.216 N 5060 OK (1 ms)

[cpsuk]

when we input "sip show registry" we get =

Host Username Refresh State Reg.Time

our current dial plan is
exten => _9XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _9XXXXXXXXXX,2,Dial(sip.trunk_gradwell_(om/${EXTEN:1},,To)
exten => _9XXXXXXXXXX,3,Hangup

This is from the "sip set debug ip 109.224.232.216:5060" output

<--- SIP read from 109.224.232.216:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 88.208.XXX.XXX:5060;received=88.208.XXX.XXX;branch=z9hG4bK7bddb9c7;rport=5060
From: "asterisk" <sip:asterisk@88.208.XXX.XXX>;tag=as1c7388e6
To: <sip:sip.trunk.gradwell_(om;cpd=on>;tag=6461e2e65644146a10275ef1ab60a3a2.fc07
Call-ID: 4cccacab2337bb5a0a9562f36e4c357c@88.208.XXX.XXX
CSeq: 102 OPTIONS
Accept: */*
Accept-Encoding:
Accept-Language: en
Supported:
Server: OpenSIPS (1.6.4-2-notls (x86_64/linux))
Content-Length: 0

Would appreciate if someone would kindly let me know what I have done wrong, first time configuring a server and I think I have got the carrier dial

plan wrong as it looks like it does connect to my SIP trunk provider?

[williamconley]

1) Welcome to the Party!

2) As you are obviously new here, I have some suggestions to help us all help you:

When you post, please post your entire configuration including (but not limited to) your installation method and vicidial version with build.

This IS a requirement for posting along with reading the stickies (at the top of each forum) and the manager's manual (available on EFLO.net, both free and paid versions)

You should also post: Asterisk version, telephony hardware (model number is helpful here), cluster information if you have one, and whether any other software is installed in the box. If your installation method is "from scratch" you must post your operating system and should also post the .iso version from which you installed your original operating system. If your installation is "Hosted" list the site name of the host.

If this is a "Cloud" or "Virtual" server, please note the technology involved along with the version of that techology (ie: VMware Server Version 2.0.2). If it is not, merely stating the Motherboard model # and CPU would be helpful.

Similar to This:

Vicibox X.X from .iso | Vicidial X.X.X-XXX Build XXXXXX-XXXX | Asterisk X.X.X | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation | Intel DG35EC | Core2Quad Q6600

3) Setting up a carrier has been covered in the forum so many times it's silly. LOL I note that your "dial(" line has some unusual inclusions in it.

4) Change your dialplan to match an existing standard one and you may be on your way. Note that you should NOT be including "gradwell" anything directly in the dial command. The carrier to which you'll send the call and the extension that will be dialed should both be addressed as Variables. While the EXTEN variable is automatically going to be the number being dialed, the carrier will be identified by a properly created Globals String.

5) _9XXXXXXXXXX is NOT a good US extension pattern. _91NXXNXXXXXX is the proper us extension pattern. Note the "1" and those "N"s. While the "N"s being replaced by "X"s will not kill your dialplan, that missing "1" definitely will.

6) We generally only charge $25 to configure a standard carrier. Seriously. And there are LOTS of samples of properly configured carriers on this forum.

Happy Hunting!

[gardo]

You can use the add new carrier wizard to easily setup your carrier. I highly recommend going over our GOautodial Getting Started Guide here: http://goautodial.org/projects/goautodialce/wiki/Goautodial_Getting_Started_Guidev3.