Sipura analog gateway and asterisk pbx

Asterisk Asterisk Support
1

Hi, I have an pure SIP asterisk.(I don’t have any analog pci card). Then I borrowed a sipura 3000. How can I receive and make call using this setup.
What would be the config in asterisk. I wanted to happen is this, if someone dials my home number from a pstn, They will here some prompts.After they hear the voice prompts it will be transferred to my sip fone. And when am going to make a outside/pstn call using my sip phone I would like it to dial 9 first then dial the desired number
See exmple below.
Example.

Sipura 3000 Ip would be 192.16.1.2
asterisk server 192.168.1.3
sip phones - ip 192.168.1.4 ---- sipnumber 101
My home phone number - 7896664

Would my extensions.conf would be like this?
[incoming]
exten=>7896664,1,Playback(welcome to mywolrd)
exten=>7896664,2,Playback(please wait)
exten=>7896664,3,Goto(sip-phne,${EXTEN},1)

[sip-phne]
exten=>101,1,Dial(SIP/${EXTEN},20,ro)
exten=>101,2,Hangup

[outbound]
exten=>_9X.,1,Dial(SIP/${EXTEN:1}@192.168.1.2,20,ro) ;-> SIPURA 3000
exten=>_9X.,2,Hangup

Is this correct?Please advised. Thank you very much

2

If you are using pure voip then you should run ztdummy if you want to have confrencing and or music on hold.

As far as the SPA-300 you will need to point the SIP account to send all incoming calls to the incoming context and to go to exten7896664. Also have you edited your sip.conf ? As far as evrything else your configs look OK. The way to know for sure is to test it.

3

hi, thanks for the immediate reply. I have set-up the sip.conf file corectly.I guess. How about the outgoing calls?Is that all I need?

I’ll be doing the test tomorrow at our office. please advise. Thank you very much.

regards,
newbie_aste

4

Seems to be. Test and see what happens.