SIP Trunking with Panasonic PBX - Outgoing Call Issue

Hi,

I am using Asterisk 1.8.2

I have created SIP trunk with Panasonic TDE600. I have define the SIP extension of Panasonic PBX in my sip.conf file as trunk as follwoing :

sip.conf

[general]

register => 1261:1261@xxx.xxx.xxx.xxx/1261

[panasonic];the is the sip trunk

type = friend
username = 1261
secret = 1261
fromuser = 1261
host = xxx.xxx.xxx.xxx
dtmfmode = rfc2833
fromdomain = xxx.xxx.xxx.xxx
context = from-pbx
insecure = very
qualify=yes
disallow=all
allow=all
nat=yes
canreinvite=no

[1000] ;this is the extension of the handset
type=friend
context=from-sip
username=1000
callerid=1000 <1000>
host=dynamic
defaultip=0.0.0.0
secret=1000
nat=no
canreinvite=yes
mailbox=1000
disallow=all
allow=all
amaflags=default
accountcode=1000

The dial plan written as below.

extensions.conf

[from-pbx]

exten=> _X.,1,Answer()
exten=> _X.,n,Dial(SIP/1000,40)

[from-sip]

exten=>1215,1,Answer()
exten=>1215,n,Dial(SIP/panasonic/${EXTEN},40)

pansonic sip trunk register properly

Host dnsmgr Username Refresh State Reg.Time
xxx.xxx.xxx.xxx:5060 N 1261 885 Registered Fri, 15 Apr 2011 11:04:57

Note: xxx.xxx.xxx.xxx is the panasonic PBX ip address., 1215 is the Panasonic PBX extension.

Issue : is for making outgonig call through SIP trunk to Pansonic PBX extension. Outgoing call is not working and incoming call is working properly with good sound quality.

As per the above dialplan every incoming call of pansonic sip trunk is working fine and goes to my 1000 extension.

When I am making outgoing call from my 1000 extension is not working. Can any one help me out to sort out this issue.

Thanks is advance, if some one help me out in this issue. its save my hours…

Ketan

Hello Sir,

Good Afternoon. have you solved your problem?? I am also considering a same scenario. I am not good with Panasonic PBX. I have got licence for 16SIP-GW. Registered it. On the trunk slot i have installed V-SIPGW16. Now what settings should i change in Panasonic. Pelase note that My Ext. in Panasonic is 1XXX & My Ext in Asterisk is 2XXX. It will be a pleasure if you guide me what to do.

Arif.

You’ve defined it as an extension, not as a trunk. Typically I would expect a trunk on a local PABX to be defined with static addresses, and to forward dialed digits, not have them forced to a set value. You wouldn’t be setting fromuser on a trunk.

insecure=very is invalid on that version of Asterisk, and I can think of no reason why you would need its replacement. You might need insecure=invite.

I can think of no reason why you neead nat=yes.

canreinvite is deprecated, on that version, in favour of directmedia, although still, I think, recognized.

Although this is more a security issue, you probably want type=peer for both.

Why are you answering your incoming calls before the outgoing side answer? For calls from the PSTN this can result in unnecessary charges to the caller.

In what way are the outgoing calls not working? Please provide console logs, and a description of what you expect to happen and how what actually happens differs.