Re: [asterisk-users] Rookie / sip and extensions
On Sat, Jul 7, 2012 at 1:48 PM, Thomas Perron wrote:
> extensions.conf
> [globals]
>
> ;
> ;
> [incoming]
> ;
> ;exten=> s,1,Goto(125010155_incoming)
> ;
> ;[125010155_incoming]
> exten => s,1,Answer
> exten => s,n,Dial(SIP/16175551212)
>
>
> sip.conf
> [general]
> ;register => 125010155:funnyti...@sip3.voipvoip.com/125010155
> register => 125010...@sip3.voipvoip.com:funnytiger@69.90.209.11
> ;
> [incoming]
> username=125010155
> type=peer
> secret=funnytiger
> nat=auto
> insecure=invite,port
> host=69.90.209.11
> fromdomain=69.90.209.11
> dtmfmode=rfc2833
> context=incoming
> allow=g729
> allow=ulaw
> allow=alaw
> allow=ilbc
> srvlookup=yes
>
If these are actual copy / pastes from your extensions.conf and sip.conf
files, with just passwords changed, your issue probably comes from your
over abundant use of semi-colons (";") at the start of several lines. The
semi-colon indicates a comment line to the asterisk parser, and thus isn't
parsed. Your only exten => line in your [incoming] context is commented
out, as is the name of your [125010155_incoming] context, and your first
register statement.
Set the CLI verbosity to 6 (core set verbose 6) and then try to dial in
again, and paste the failed output as a response to this email, and we can
diagnose from there.
--
Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com
--
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[asterisk-users] Rookie / sip and extensions
Sorry for blasting another desperate note but I am trying! I have changed the username and password and IP to protect my system. But, the logic is unchanged. It is does not work! I simply want to dial the telephone number provided to me for my DID which corresponds with my SIP info. And, then it should connect and hit the "incoming" context and simply dial the 617 number. I am close but no cigar. Now I get a fast busy tone only. What is missing or what is needed please? extensions.conf [globals] ; ; [incoming] ; ;exten=> s,1,Goto(125010155_incoming) ; ;[125010155_incoming] exten => s,1,Answer exten => s,n,Dial(SIP/16175551212) sip.conf [general] ;register => 125010155:funnyti...@sip3.voipvoip.com/125010155 register => 125010...@sip3.voipvoip.com:funnytiger@69.90.209.11 ; [incoming] username=125010155 type=peer secret=funnytiger nat=auto insecure=invite,port host=69.90.209.11 fromdomain=69.90.209.11 dtmfmode=rfc2833 context=incoming allow=g729 allow=ulaw allow=alaw allow=ilbc srvlookup=yes -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trixbox or FreePBX or Elastix or PBX In a Flash
Thanks Tim. One of the problem that I am facing is the complicated generated configuration for the FreePBX, is it the same thing in the Elastix? To understand this complicated generated commands, is there a documentation to explain this for FreePBX or Elastix? One of my friend told me that he installed (as I remember) FreePBX and there were already existed the TFTP files that the Cisco IP Phones is requesting (for sip or skiny) and already there were a TFTP server. Which module to do this? Regards Bilal -- > > Hi All; > > > > Based on what I have to use Trixbox or FreePBX? > > > > Can someone advise? > > Trixbox includes FreePBX as it's GUI. However, keep in mind > it is a bastardized, forked version of FreePBX that has seen > nary any new development or innovation in some time. At this > point, for a standard PBX installation, my recommendations > would be (in this order): > > 1. Elastix > 3. AsteriskNOW > 2. PBX In a Flash > > --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
