SIP and Asterisk Configuration

[D]

We are going thru Phase 5 of Scratch Install. In Section 3 of Phase 5 shows how to configure the SIP phone with Asterisk. I would like to know which IP should be included where it says Host = (Our server static public IP or the IP of our Telecom provider)?

Also, we are using codec G729. Do we need to mention that somewhere here? Do we need to enter the same info again for [gs 102] also?

Our Telecom provider does not use username or Secret. Can we delete those line or should we leave those blank? Here is what our Telecom provider has given us (please let us know if this good enough to configure SIP):

disallow=all
allow=alaw
allow=ulaw
type=peer
host=61.255.198.82
context=from-pstn
dtmfmode=rfc2833
insecure=verycanreinvite = no
qualify=yes

Below is what is written in Scratch Install:
; setup account for SIP trunking:
; [SIPtrunk]
; disallow=all
; allow=ulaw
; allow=alaw
; type=friend
; username=testSIPtrunk
; secret=test
; host=10.10.10.16
; dtmfmode=inband
; qualify=1000

[gs102]
disallow=all
allow=ulaw
allow=alaw
type=friend
username=gs102
secret=test
host=dynamic
dtmfmode=inband
defaultip=10.10.10.16
qualify=1000
mailbox=102
[spa2000]
disallow=all
allow=ulaw
allow=alaw
type=friend
username=spa2000
secret=test
host=dynamic
dtmfmode=inband
defaultip=10.10.10.17
qualify=1000
mailbox=2000
[spa2001]
disallow=all
allow=ulaw
allow=alaw
type=friend
username=spa2001
secret=test
host=dynamic
dtmfmode=inband
defaultip=10.10.10.17
qualify=1000
mailbox=2001

[mflorell]

You can always use "dynamic" for host unless you want to use the IP address of the phone.

As for codec, as long as Asterisk has the G729 licenses installed it will work if you have allow=all in your sip.conf.

As for other settings, you just have to play around with them to see what works for your phones.

[D]

Thanks for your prompt response. Do we have to copy the same IP and other info that we enetered in SIP.conf into [gs102]?

We are using the softphone Express Talk (since our telecom provider only supports SIP and H323). In the Express Talk configuration, we do not get an IP for the phone. We just enter our telecom's IP in the phone as the Host Server IP. We do have static public IP on the machine that the softphone is installed on but as far as I know - Express Talk did not assign or ask for our static public IP during configuration. Would this create a problem? How would Asterisk communicate with this phone? Are there any other good SIP phone that you can recommend? I looked at Firefly but it seems that they want the traffic to go thru them. We have our on Telecom provider.

[mflorell]

Are you trying to setup a SIP trunk through Asterisk to your provider or are you trying to setup a SIP phone to connect to asterisk?

you can use dynamic for the host because the SIP phone initiates the connection to the server not the other way around.

We recommend IDEfisk for softphones.

[D]

Can we setup a SIP trunk directly to our provider from Asterisk or do we have to have a separate phone (SIP - soft or hard phone)?

If we cannot connect directly from Asterisk to our provider - how do we configure Asterisk to connect to our SIP phone?

From what I have seen of IDEFisk - its an IAX phone and our provider only supports SIP and H323. Apart from Firefly do you know of any good SIP softphone that allows users to use their own telecom providers? We currently have Express Talk installed but don't know how it will communicate with Asterisk since they have to be on separate machines. Any suggestions?

Thanks

[espencer]

sjphone from sjlabs.com is a free sip softphone that i like. there is also a pocketpc version that works well. i have the pc and pocketpc versions connected to my home asterisk server.

[mflorell]

You can setup SIP trunks to work with Asterisk and setup whatever kind of phone you want that will be able to dial out through that SIP trunk(IAX/SIP/Skinny/Zap/MGCP/H323). However some providers like Vonage often don't allow you to hookup non-provided equipment to their services.

To setup a SIP trunk you can take a look at the example in SCRATCH_INSTALL or you could just do a search on voip-info.org which will provide you with a lot more information than I could give you.

[D]

Thanks Matt. Our telecom provider allows us to use any equipment. The have us connect from our IP to theirs and they provide PSTN and temination for us.

Does the Softphone intiate the call call and connection thru Asterisk meaning does the call go from the softphone to the Asterisk server and then Asterisk connects to our telecom provider? Or is it the other way around - Asterisk initiates the call to the softphone and the softphone connects to our telecom provider?

[mflorell]

The call routing does go through Asterisk, your phone sends the numbers for the call to Asterisk, then Asterisk connects with your provider to place the call.

[D]

Thanks. To clarify - The softphone gets the number to dial from the database and then the phone routes it to Asterisk?

If that's the case, how do we send the numbers to be called from the database to the softphone?

[mflorell]

OK, now I'm confused. Are you talking about VICIDIAL now, or just basic outbound dialing from a softphone?

[D]

I want to use ViciDial as a predictive dialer to dial using a database. When we VICIDIAL - do we also need a Softphone or is VICIDIAL going to alternate as a softphone? We have followed Scratch Install and have installed Asterisk and all the other basic packages.

Our goal is to use the dialer using a database so agents can call and reduce the number of ansering machines, wrong numbers, etc. Agents can be in one place and sometimes be remote. I understand we can load data into MySQL from our web based SQL Database.

What do you suggest in this case. Do we need a separate softphone to do this or would VICIDIAL do the calling thru Asterisk? Thanks for your feedback.

[mflorell]

vicidial can do this. You do need a softphone or hardphone for your agents though.

[espencer]

you can connect SIP softphones and/or hardphones directly to the vicidial server (instructions in the scratch install). i just set up an astguiclient install to connect to a different asterisk server with an IAX trunk that uses the phones connected to the remote computer as well as the dialtone provided by it. to do this, i made sure entries like this:

exten => _99XXX,1,Dial(IAX2/remoteserver/${EXTEN:2},55,Tto)

where i XXX is the extension number local to the remote server were in the dialplan in order to use the extensions on the remote server. when doing this, i changed the phone type in astguiclient admin to EXTERNAL.

[D]

Thanks Matt and ESpencer. The way I understand it is that Asterisk and VICDIAL route and feed the calls to the agent's SIP softphone. Is that correct?

We can assign the same Public Static IP to the SIP softphone and the Windows machine that the SIP softphone is installed on. Would that work - both the sofphone and windows pc having the same Static Public IP? We are currently using Express Talk and its asks for IPs in a couple of places in Settings:

(1) Use Static IP address and mapped ports

(2) Server (SIP Proxy or Virtual PBX) - We are currently putting our telecom provider's IP here. Should we replace it with our Asterisk Server's IP?

How is call routed when we use the Manual Dial Option (Text Box) in VICIDIAL's GUI? Is the call sent to the agent's softphone in this case?

Thanks again for your comments.

[mflorell]

in VICIDIAL an agent is always on the phone in a meetme conference waiting for a call. when a call comes in it is sent to the meetme room. The agent never hangs their phone up.

With VICIDIAL the softphone has no direct connection to your SIP provider.

Will you be using a NAT or internal network at all?

[D]

Is the agent always dialed into the Asterisk server or ViciDial? If yes, does this happen by the agent logging into the ViciDial thru the Browser?

We will be using both internal and external networks. We will have onsite and remote agents.

Our NAT is currently disabled. Do we need to enable it?

Thanks

[mflorell]

Is the agent always dialed into the Asterisk server or ViciDial? If yes, does this happen by the agent logging into the ViciDial thru the Browser?

Yes.

We will be using both internal and external networks. We will have onsite and remote agents.

VICIDIAL can have agents anywhere that the Asterisk server can call and anywhere the web server is accessible with reasonable latency no matter what network they are on.

Our NAT is currently disabled. Do we need to enable it?

How are your agent computers connected to the internet? Do they all have real-world IP addresses?

[D]

Thanks Matt. We will have the agent's softphone always on and have them log in thru the browser.

Currently, we are in the process of testing and therefore the number of agents is small and they all have separate Static Public IP's. We will have remote agents and agents on LAN once we roll this out. In that case, do we enable NAT?

[mflorell]

As for NAT, this really depends on how you have your network setup. If you leave your Asterisk/VICIDIAL server on public IP then you should be fine. If you move it inside the NAT then you will need to change the server_ip settings for everything(conferences, vicidial_conferences, servers, server_updater and the configuration files)

[D]

Thanks, we will keep that in mind.

We just got our X100P (Clone) in the mail. It has 2 ports (both are phone ports - same size). Isn't it supposed to have an ethernet size port in it? This card does not have an ethernet port. Do we only hook up our phone chord from the VOIP Gateway in it?

[mflorell]

You do not need to hook anything up to it. It can exist purely as a timer in a PCI slot.

[D]

Thanks for the info. I was going by what is written in Scratch Install:

11. Now that you have configured Asterisk, it is time to try to start it for the
first time.
- First make sure that your T1 line(or other telco line) is connected to the
digium card.


By the way, do we get some type of dial tone when everything is configured and we are logged in and ready to dial?

[mflorell]

those instructions are based on a T1 install.

When you login to vicidial, your phone will ring and when you pick it up you should hear that you are the only on in your conference.

[D]

Thanks Matt.

How does the call get routed when we try to manually dial Dial using ViciDial by entering a telephone number in the Text box of the GUI (browser)? Does the call go to Asterisk first and after getting connected it is routed to the available agent's softphone?

Thanks again.

[mflorell]

Manual dialing can only be done by an agent that is not on the phone.

It will be sent to their session only.

The call will start directly from the session and the agent will hear the ringing.

[D]

So, instead of manually dialing that number directly from his Softphone, the agent would actaully dial thru the ViciDial browser in this case?

[mflorell]

Yes, and it will have a lead_id(if the number does not already exist) and it will be logged as a normal call in the vicidail logs.