Moving over from PRI to SIP

[Nortelguy]

I recently picked up a SIP gateway provider and would like to configure my system to now point the GW instead of the PRI interface. I was reading through the configuration files on what I need to change when moving from a PRI connection to a SIP. Here is what I gather, zaptel.conf needs to have the lines REM'd that reflect my PRI here is what I have there now:
span=1,1,0,esf,b8zs
bchan=1-23
dchan=24

--
I am guessing no SIP statements go in that file. Just put # next to all those lines.

Next I would go into sip.conf and adjust the following:
[general]
port = 5060
bindaddr = 0.0.0.0
context = default
; register SIP account on remote machine if using SIP trunks
; register => testSIPtrunk:test@10.10.10.16:5060
;
; 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

Is there anything else in this file that needs modification ?

After that I would go to ADMIN - SERVERS - (Server Name in List) - Adjust "Max VICIDIAL Trunks" beyond 23 (for PRI) so that I can place more then 23 calls obviously. I am guessing this is dependant on the SIP provider and hardware/resources in the computer. Whats a good target number for a dual core machine ?

Also I have a SANGOMA A101D for the PRI currently. Its being used as clock source instead of using ZTDUMMY ? What would I do now for a clock source and where would I make those changes ? Would be still be possible to use the SANGOMA as a clock source instead of using a software based one ?

Also has anyone tried out the free G729 from technology ready folks ? Its probably hack but is it worth even attempting to use ? I want all my calls to be placed in this codec to save on bandwidth.

TIA

[Nortelguy]

Anybody

[mflorell]

what kind of gateway are you using?

As for the free G729 codec, we do not recommend usng it.

[avi]

Dear NortelGuy,

I too have a 2 E1 SIP gateway that i have been using along with the Digium cards.
I did not do any extra setting for that .

All I did is that in my extensions file i have explicitly mentioned to route the calls via that gateway.

Code: Select all

exten=>0X.,1,Dial(SIP/${EXTEN}@ip_sip_gateway)

See if it helps you

[Nortelguy]

Dear NortelGuy,

I too have a 2 E1 SIP gateway that i have been using along with the Digium cards.
I did not do any extra setting for that .

All I did is that in my extensions file i have explicitly mentioned to route the calls via that gateway.

Code: Select all

exten=>0X.,1,Dial(SIP/${EXTEN}@ip_sip_gateway)

See if it helps you

I guess my quesiton is in order to use a SIP gateway what other things need to be changed. I believe the dialer needs to login to the gateway in some fashion. I believe that is done in the SIP.CONF IIRC.

[Nortelguy]

what kind of gateway are you using?

As for the free G729 codec, we do not recommend usng it.

Its a SIP Proxy/Billing system .. Whats wrong with the free codec other then not paying for the license ?

[Nortelguy]

Ok since nobody wanted to reply on how to do this I figured it out on my own and I'll post here incase any other n00bs cant figure out how either.

In extensions.conf under global, instead of changing the TRUNK variable for outbound calls just rem the statement you have now for the PRI

Example:
TRUNK=Zap/g1

to TRUNK=SIP/IP-ADDRESS-WHATEVER-HERE

Next configure your SIP.conf for your carriers SIP info pretty straight forward