Multiple lines

[Shaggy]

Heyall, i'm fairly new with asterisk and i got a few questions....
So we have 4 lines coming into the building and we are wondering about how we can get asterisk to distiguish each line separately.
so right now we have 1 SIP line that is being used by all the phones and we are wondering if we will be making multiple SIP trunks for each line? if so how do you do this?

thanks in advance!

[J. Oquendo]

Don't be fooled by numbers, think of them as names. You have a number coming to your PBX. What do you want to do when your PBX sees those numbers?

Assuming they're sip trunks...

# In sip.conf
[200]
type=friend
username=200
secret=somepasswordgoeshere
host=dynamic
context=THISCONTEXT
mailbox=200@your-pbx
; Only you can make these call below
; dtmfmode=rfc2833
;nat=yes
;canreinvite=yes
callerid="something"<12125551212>
disallow=all
; I don't know what codecs you use...
allow=g729
qualify=yes

# In extensions.conf

; So what they're numbers, if you had an insane dialing set up for names you could do:
; exten => dave,1,Goto(THISCONTEXT|200|1)
; So forget about whether or not you can or can't. You are able to send as many
; numbers as you can handle... Kept the extensions context down to a minimum...
; too lazy to type

; So we have three distinct numbers coming in...

; This will be the DID for extension 200. When a call comes in go to THISCONTEXT line 200
exten => 2125551212,1,Goto(THISCONTEXT|200|1)

; This will be the DID for extension 201. When a call comes in go to THISCONTEXT line 201
exten => 2125551213,1,Goto(THISCONTEXT|201|1)

; This will be the DID for extension 202. When a call comes in go to THISCONTEXT line 202
exten => 2125551214,1,Goto(THISCONTEXT|202|1)

[THISCONTEXT]
; Remember, as shown above you told Asterisk that anything coming for 2125551212
; should go here first. What does this line do first... Ring a sip device with username(extension) 200
exten => 200,1,Dial(SIP/your-sip-device-200|20|tr)
exten => 200,2,Voicemail(u200@your-pbx)

As shown above you told Asterisk that anything coming for 2125551213
; should go here first. What does this line do first... Ring a sip device with username(extension) 201
exten => 201,1,Dial(SIP/your-sip-device-201|20|tr)
exten => 201,2,Voicemail(u201@your-pbx)

; As shown above you told Asterisk that anything coming for 2125551214
; should go here first. What does this line do first... Ring a sip device with username(extension) 202
exten => 202,1,Dial(SIP/your-sip-device-202|20|tr)
exten => 202,2,Voicemail(u200@your-pbx)

[Lance Koster]

Your SIP provider can allow as many simultaneous calls as you need. Make sure under the extension you allow call waiting, this will allow the second, third... inbound call. The simultaneous call path will also allow you to make the second, third... outbound call.

As far as your land lines. You can purchase an inexpensive analog adapter that will bring your existing telephone lines in and of course they can easily be distinguished many different ways. If you need details on "how to" just give me an e-mail [[e-mail]].

But unless you are using those lines for faxing, you should port them into your SIP provider, and they will simply send the calls via the trunk.

Lance - Tampa, FL.

[Charlie Gill]

"4 lines coming into the building" infers 4 land lines. I'm a newbie also and spent several days working on this same problem. There are several posts and help tips covering this but they all seemed to be out of date or apply to a different distro. Anyway...you have to edit zapata.conf and change ALL OCCURANCES of context=from-pstn to context=from-zaptel INCLUDING any files that are #included. That last part is what none of the other post mentioned. In addition, some of the included files are auto generated and specifically say you are not to edit them (but you must).
After this, you can set up your incoming trunks by Zaptel channel rather that CID/DID. The Zaptel channels are just 1, 2, etc, (not Zap/1-1) something else the documents don't specify.
Good luck,
Charlie

[Shaggy]

ok sweet thanks for the info guys...
i was just talking to the IT here of the place and he said he also found a PDF on the server with everything we need... so i will add your info onto it and relate, thanks guys!!!

[Frank Rabbito]

I am searching for a person that has the ability to make some specialized programming to my Evolution VOIP Box. The box is currently fully functional, however, it requires some special features for a law enforcement client that is beyond my limited knowledge.
Of course, the programmer will be paid for his/her time.
I may be reaced at the following number: 800-794-0159 extension #4445

Thank You,
Frank Rabbito