VOIP PSTN inbound and outbound

Asterisk Asterisk Support
1

Can someone give me some sample code for extensions.conf that would allow inbound/outbound PSTN (by pressing 9) and inbound/outbound VOIP (by pressing 0 or some other digit).

Thanks!!!

2

Hi,

Here is some a small sample; need some for infor from you.

  1. Do you have a VOIP povider setup and working?
  2. How are you conecting to the PSTN (hardware)?

[outgoing]
;VOIP-Out
exten => _9.,1,Dial(SIP/engin-out/${EXTEN:1})
exten => _9.,2,Hangup()

;PSTN-Out
exten => _0.,1,Dial(Zap/1/${EXTEN:1})
exten => _0.,2,Hangup()

3

Here’s what I have:

sip.conf=======================

[general]
context=default
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes

register =>spa400@192.168.1.151
register =>spa921:mytest@192.168.1.142
register =>6000@192.168.1.140/6000
register =>5435377:fuente1@sip.varphonex.com/5435377

[spa400]
type=peer
user=sip400
;secret=Admin
host=192.168.1.151
dtmfmode=rfc2833
canreinvite=no
nat=yes
context=from-pstn
disallow=all
allow=ulaw
allow=alaw
allow=g729
insecure=very

[spa921]
username=spa921
secret=mytest
type=friend
host=dynamic
context=to-varphonex ;used to be “to-pstn”

[6000]
type=friend
context=default
host=dynamic

[varphonex]
type=peer
host=sip.varphonex.com
fromuser=5435377
fromdomain=sip.varphonex.com
username=5435377
secret=fuente1
context=to-varphonex
canreinvite=no

[sip.varphonex.com]
username=5435377
user=5435377
type=friend
insecure=very
host=sip.varphonex.com
fromdomain=sip.varphonex.com
context=from-pstn

extensions.conf=====================
[[general]
static = yes
writeprotect = yes
clearglobalvars = yes

[global]
;DIALOUT = 9

[default]

[from-pstn]
exten => spa400,1,Dial(SIP/spa921)
exten => spa400,n,Hangup()

[to-pstn]
exten => _9xxxxxxxx,1,Dial(SIP/${EXTEN:1}@spa400,20)
exten => _9xxxxxxxx.,n,Hangup()

[to-varphonex]
exten => _0xxxxxxxxxx,1,Dial(SIP/${EXTEN:1}@varphonex)
exten => _0.,n,Hangup()

[from-varphonex]
exten => varphonex,1,Dial(SIP/spa921)

=================================

Here’s the result of an outbound call starting with “0” to dial-out:

== Using SIP RTP CoS mark 5
– Executing [07734904492@to-varphonex:1] Dial(“SIP/spa921-c0008a60”, “SIP/7734904492@varphonex”) in new stack
== Using SIP RTP CoS mark 5
– Called 7734904492@varphonex
– SIP/varphonex-02514450 is making progress passing it to SIP/spa921-c0008a60
– SIP/varphonex-02514450 is making progress passing it to SIP/spa921-c0008a60
– SIP/varphonex-02514450 is making progress passing it to SIP/spa921-c0008a60
– SIP/varphonex-02514450 answered SIP/spa921-c0008a60
– Packet2Packet bridging SIP/spa921-c0008a60 and SIP/varphonex-02514450
== Spawn extension (to-varphonex, 07734904492, 1) exited non-zero on ‘SIP/spa921-c0008a60’

======================================

It looks like I’m getting there and then something pukes at the tend… get the busy tone.

Other problems:

PSTN in and out. can’t hear any sound. Just rings and answers and nothing else. Using ulaw codec.

I have one test phone for a demo. How do I get it to reference two different “contexts” 9 for PSTN and 0 for VOIP without recoding the section [spa921]???

Thanks again for your help!!!

4

Why would SIP/varphonex pass the call to SIP/SPA921 which is my phone. Shouldn’t it be the other way around? Perhaps something is buggy in the code?

5

It’s passing the “call is making progress” indication, not the call.

Nothing is wrong in that respect.

6

After a lot of troubleshooting… I got frustrated and just shut down my firewall. Low and behold… that did it. Calls go through perfect VOIP and PSTN. Now, I just need to figure out which ports it is trying to use.