Sponsor: VoiceMeUp - Corporate & Wholesale VoIP Services

VoIP Mailing List Archives
Mailing list archives for the VoIP community
 SearchSearch 

[asterisk-users] ${HASH(SIP_CAUSE,<channel-name>)}


 
Post new topic   Reply to topic    VoIP Mailing List Archives Forum Index -> Asterisk Users
View previous topic :: View next topic  
Author Message
jonas.kellens at telen...
Guest





PostPosted: Thu Oct 30, 2014 8:52 am    Post subject: [asterisk-users] ${HASH(SIP_CAUSE,<channel-name>)} Reply with quote

Hello,

I read on the wiki :

Asterisk 1.8 will allow to read SIP response codes in the dialplan via ${HASH(SIP_CAUSE,<channel-name>)}. Additionally make sure you're using the destination channel, not the source channel.

But when I use this in my dialplan, this 'variable' is empty.

Dialplan :

exten => h,n,NoOp(sip cause = ${HASH(SIP_CAUSE,${CHANNEL})})
exten => h,n,NoOp(sip cause = ${HASH(SIP_CAUSE,CHANNEL)})

CLI :

[Oct 30 14:48:03] -- Executing [h@pbx-routing:5] NoOp("SIP/SipAT01-00000015", "sip cause = ") in new stack
[Oct 30 14:48:03] -- Executing [h@pbx-routing:6] NoOp("SIP/SipAT01-00000015", "sip cause = ") in new stack


Can anyone tell me how this should be used ?


Kind regards,

Jonas.
Back to top
mailinglist+asterisk a...
Guest





PostPosted: Thu Oct 30, 2014 11:03 am    Post subject: [asterisk-users] ${HASH(SIP_CAUSE,<channel-name>)} Reply with quote

On 30/10/14 13:52, Jonas Kellens wrote:

Quote:
Hello,

I read on the wiki :

Asterisk 1.8 will allow to read SIP response codes in the dialplan via ${HASH(SIP_CAUSE,<channel-name>)}. Additionally make sure you're using the destination channel, not the source channel.

But when I use this in my dialplan, this 'variable' is empty.

Dialplan :

exten => h,n,NoOp(sip cause = ${HASH(SIP_CAUSE,${CHANNEL})})
exten => h,n,NoOp(sip cause = ${HASH(SIP_CAUSE,CHANNEL)})

CLI :

[Oct 30 14:48:03]     -- Executing [h@pbx-routing:5] NoOp("SIP/SipAT01-00000015", "sip cause = ") in new stack
[Oct 30 14:48:03]     -- Executing [h@pbx-routing:6] NoOp("SIP/SipAT01-00000015", "sip cause = ") in new stack



Take a look at my blog entry about it :-

http://gblades.blogspot.co.uk/2013/07/how-to-get-sip-response-code-in.html
Back to top
paul.belanger at polyb...
Guest





PostPosted: Thu Oct 30, 2014 3:08 pm    Post subject: [asterisk-users] ${HASH(SIP_CAUSE,<channel-name>)} Reply with quote

On Thu, Oct 30, 2014 at 9:52 AM, Jonas Kellens <jonas.kellens@telenet.be> wrote:
Quote:
Hello,

I read on the wiki :

Asterisk 1.8 will allow to read SIP response codes in the dialplan via
${HASH(SIP_CAUSE,<channel-name>)}. Additionally make sure you're using the
destination channel, not the source channel.

But when I use this in my dialplan, this 'variable' is empty.

Dialplan :

exten => h,n,NoOp(sip cause = ${HASH(SIP_CAUSE,${CHANNEL})})
exten => h,n,NoOp(sip cause = ${HASH(SIP_CAUSE,CHANNEL)})

CLI :

[Oct 30 14:48:03] -- Executing [h@pbx-routing:5]
NoOp("SIP/SipAT01-00000015", "sip cause = ") in new stack
[Oct 30 14:48:03] -- Executing [h@pbx-routing:6]
NoOp("SIP/SipAT01-00000015", "sip cause = ") in new stack


Can anyone tell me how this should be used ?

sip.conf: storesipcause=yes


--
Paul Belanger | PolyBeacon, Inc.
Jabber: paul.belanger@polybeacon.com | IRC: pabelanger (Freenode)
Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger

--
_____________________________________________________________________
-- 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
Back to top
Display posts from previous:   
Post new topic   Reply to topic    VoIP Mailing List Archives Forum Index -> Asterisk Users All times are GMT - 5 Hours
Page 1 of 1

 
Jump to:  
You cannot post new topics in this forum
You cannot reply to topics in this forum
You cannot edit your posts in this forum
You cannot delete your posts in this forum
You cannot vote in polls in this forum


Powered by phpBB © 2001, 2005 phpBB Group

VoiceMeUp - Corporate & Wholesale VoIP Services