Outgoing call

Asterisk Asterisk Support
1

Hi,

i have configured the zapata and extensions as follows

[erro]

exten => _NXXNXXXXXX,1,Dial(Zap/g1/${EXTEN})
exten => s,1,Wait(1)
exten => s,2,Answer()
exten => s,3,Playback(your-temp-greeting)
exten => s,4,Hangup()

exten => 218,1,Dial(SIP/218,30,r)
exten => 218,2,VoiceMail(218@default)
exten => 218,3,PlayBack(vm-goodbye)
exten => 218,4,Hangup()

and zapata.conf

[channels]
language=en
context=erro
rxwink=300 ; Atlas seems to use long (250ms) winks
usecallerid=yes
callwaiting=yes
;restrictcid=no
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
rxgain=0.0
txgain=0.0
callgroup=1
pickupgroup=1
immediate=no
echocancel=yes ; You can set this to 32, 64, or 128, tweak to your needs.
echocancelwhenbridged=yes
;echotraining=yes ; Asterisk trains to the beginning of the call, number is in milliseconds
callerid=asreceived
group=1
signalling=fxs_ks
channel=> 1-16 ; Again if you only have one FXO module remove the ‘-4’

and when i make the call some pstn number… my soft phone is saying connected and not able to hear the anything… and cli prompt says

Executing Dial(“SIP/294-11bf”, “Zap/g1/xxxxxxxxxx”) in new stack
– Called g1/xxxxxxxxxx
– Zap/1-1 answered SIP/294-11bf

Any help may be appreciated to fix the same.

2

any help may be aprreciated?

3

can you dial into a local extension on the asterisk server (the Voicemail app, for instance) and hear sound?

this may not be an asterisk problem, but rather a softphone problem.

just a thought, anyway.

4

Hi,
What is in your sip.conf file. And also add ‘r’ to this line.
exten => _NXXNXXXXXX,1,Dial(Zap/g1/${EXTEN}).
ie,
exten => _NXXNXXXXXX,1,Dial(Zap/g1/${EXTEN},20,r)

Im not sure this will help u. But check this also.

Regards,
ijb

5

hi.

i did this … and not able to hear anything… and morever i am able to talk with in extensions and able to hear the voice mail applications.

6

Hi when i do sip debug

– (9 headers 0 lines)—
Transmitting (NAT) to 192.168.1.137:5060:
SIP/2.0 503 Server error
Via: SIP/2.0/UDP 192.168.1.137:5060;branch=z9hG4bK3187B63FAE194C009B6F3BE5D4CCE4FD;received=192.168.1.137;rport=5060
From: Rizwan sip:294@192.168.111.141;tag=3269903163
To: sip:xxxxxxxxxx@192.168.111.141;tag=as73ff13c7
Call-ID: D461C985-FFEF-4DEA-88DB-E598970A31C7@192.168.1.137
CSeq: 53412 ACK
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE,

Any help may be appreciated

7

Hi,
Have you checked without “s” extension lines .
ie,
;exten => s,1,Wait(1)
;exten => s,2,Answer()
;exten => s,3,Playback(your-temp-greeting)
;exten => s,4,Hangup()
commented.
Check this also. I think it first go to the s extension.
Regards,
ijb