Hi Muhammad



From: "Muhammad Talha" <[EMAIL PROTECTED]>

To: "Asterisk Users Mailing List - Non-Commercial Discussion"

<[email protected]>

Sent: Wednesday, December 22, 2004 2:07 PM

Subject: Re: [Asterisk-Users] Zaphfc/BRI Configuration help



Thanks for lan for your reply can you share your extention.conf

setting .



this is my extention.conf lines

[globals]

OUTGOING => Zap/1

exten => _9XXXXXXXXXXXX,1(${OUTGOING}/${EXTEN:1})



i am trying to dial pstn through firefly using 9-55212323 ( Suppose

this

my Pstn number ) i get these error :



SIP Status: 484 Address Incomplete

19.973650 -> 192.168192.68 SIP/SDP Request: INVITE

sip:[EMAIL PROTECTED], with session description 19.974283

192.168192.68 -> 192.168192.33 SIP Status: 407 Proxy

Authentication

Required

19.983174 192.168192.33 -> 192.168192.68 SIP Request: ACK

sip:[EMAIL PROTECTED] 19.984367 192.168192.33 ->

192.168192.68 SIP/SDP Request: INVITE sip:[EMAIL PROTECTED],

with session description 19.984765 192.168192.68 -> 192.168192.33 SIP

Status: 484 Address

Incomplete

19.988290 192.168192.33 -> 192.168192.68 SIP Request: ACK

sip:[EMAIL PROTECTED]

I am quite new to this myself and I am not quite sure what is wrong there.

There are a number of ways to set up sip.conf but my entries look like:-

[jackie.clough]
context=from_SIP_extension
type=friend
regexten=1105
username=jackie.clough
secret=xxxxxxxx
callerid="Jackie Clough" <1105>
host=dynamic
nat=no
canreinvite=no
disallow=all
allow=gsm
allow=ulaw
allow=alaw



Your client (firefly) will register with asterisk and you can see the message on the asterisk console. The command sip show users will show you a list of registered users



Asterisk Ready.
*CLI> -- Registered SIP 'jackie.clough' at 192.168.1.13 port 5060 expires 180
-- Saved useragent "X-Lite release 1103m" for peer jackie.clough



*CLI> sip show users Username Secret Accountcode Def.Context ACL NAT rons.desk.at.in xxxxxxxx from_SIP_extens No RFC35 ians.desk.at.in xxxxxxxx from_SIP_extens No RFC35 jackie.clough xxxxxxxx from_SIP_extens No RFC35 ian.clough xxxxxxxx from_SIP_extens No RFC35 fwd-outgoing xxxxxxxx from-sip No No -- Registered SIP 'jackie.clough' at 192.168.1.10 port 5060 expires 180

In extensions.conf the context from_SIP_extensions could be something like:-

[from_SIP_extension]
include => external
include => sipcalls
include => SIPextensions
include => voicemail

and the context external would be:-

[external]
; Dial 9 for an outside line. Allow any call for now
exten => _9.,1,Dial(Zap/g1/${EXTEN:1})
exten => _9.,2,Playback(invalid)
exten => _9.,3,Hangup

Where g1 is the group 1 I defined in zapata.conf

Its not very pretty but it works for me

Ian




_______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

Reply via email to