* You are viewing the archive for May 10th, 2012

Can run from shell but not from Asterisk System command

Hi All,

I have this strange problem on a newly installed PBX. 1.8.12.0. I have other installs of 1.8.12.0 that does not exhibit this problem.

I can run from the console
/usr/sbin/fax2mail –cid-name “Console Execuation” –cid-number 61123123 –dest-exten 8960 -f /var/spool/asterisk/fax/1336702728.59

… and an email will be emailed to me.

The following does not produce an email.
exten => 1122,1,Answer
exten => 1122,n,System(/usr/sbin/fax2mail –cid-name “Console Execuation” –cid-number 61123123 –dest-exten 8960 -f /var/spool/asterisk/fax/1336702728.59)
exten => 1122,n,HangUp

result from CLI
== Using SIP RTP CoS mark 5

Event response (AMI)

When I execute the Action commands set then the Event would response back.
How would I know which Action are they belong/reference to?

For example:

ACTION: Originate
Channel: SIP/test
Exten: 215
Timeout: 30000
Context: test
Priority: 1
ActionID: 111112222333

Response: Success
ActionID: 111112222333
Message: Originate successfully queued

Event response when I hang up the call:

Event: Hangup
Privilege: call,all
Channel: SIP/test-0000007f
Uniqueid: 1336690030.189
CallerIDNum:
CallerIDName:

Cause: 16
Cause-txt: Normal Clearing

As you can see, how would I know which which ACTION was that belong to?

If I were coding in PHP (AMI) to Originate the calls then I want to detect
which call hanged up.

Thanks

SLA – Shared Line Appearance – Polycom

Oh yeah, damn small things.

Asterisk 1.8.7.1

Polycom IP650

Thanks,

enabling dialing by sip uri

On 05/10/2012 09:39 AM, Arif Hossain wrote:
> I have following sip account :
>
> Name/username Host Dyn
> Forcerport ACL Port Status Description
> demo-alice/demo-alice 192.168.7.47 D
> N 1080 Unmonitored
> demo-bob/demo-bob 192.168.7.47 D
> N 5060 Unmonitored
>
> and i have set up the following extensions for them:
>
> ASTERISK_IP=192.168.7.39
>
> [users]
> exten=>6001,1,Dial(SIP/demo-alice,20)
> exten=>6002,1,Dial(SIP/demo-bob,20)
>
> exten => _.,n,GotoIf($[${SIPDOMAIN} = ${ASTERISK_IP}]?unhandled)
> exten => _.,n,GotoIf($[${SIPDOMAIN} = ${ASTERISK_IP}:5060]?unhandled)
> exten => _.,n,Macro(uri-dial,${EXTEN}@${SIPDOMAIN})
> exten => _.,n,HangUp()u
>
> [macro-uri-dial]
> exten=>s,n,NoOp(Calling as SIP address: ${ARG1})
> exten=>s,n,Dial(SIP/${ARG1},60)
>
>
> But if i dial sip uri the call does not happen. asterisk cli shows
> extension is rejected.

Asterisk is not a SIP proxy. If you are entering a SIP URI into your
phone, and that URI does not resolve to the Asterisk server as its
target, then the INVITE request sent by the phone should not even be
sent to Asterisk at all (it should go to wherever the URI resolves to).

Email-to-Fax

Dear,
I am using Fax-to-Email feature of FreePBX, now i am looking Email-To-Fax
option with freePBX, kindly update is it possible to have this feature with
FreePBX?
kindly contact me on my email mianasif@msn.com if anybody have this type of
solution. thanks.

Open Source Realtime Dinner in Barcelona – June 13th

Hello!

I will be running an Asterisk SIP Masterclass – the last one – in Barcelona in June. During this week, I will organize a dinner for everyone working with or interested in Asterisk, Kamailio and other Open Source platforms for realtime communication. It’s June 13th somewhere in Barcelona – location will be announced later. You pay our own dinner (unless we can find sponsors) and enjoy the geeky company for free!

To join the event, use this Facebook event https://www.facebook.com/events/307548349321608/

See you in Barcelona!

/O