I'm unable to get HINTS working with the new SVN-Trunk
State never changed when ringing or on the phone.
Below is my configs (Maybe I missed something)
Thanks for any help you could give!!
##sip.conf##
[general]
callerid=unavailable
context=default ; Default context for incoming calls
bindport=5060 ; UDP Port to bind to (SIP standard port
is 5060)
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds
to all)
;allow=all
allow=ulaw
allow=g729
;allow=gsm
;maxexpirey=3600 ; Max length of incoming registration we
allow
;defaultexpirey=120 ; Default length of incoming/outoing
registration
;notifymimetype=text/plain ; Allow overriding of mime type in MWI
NOTIFY
videosupport=yes
allow=h263 ; H.263 is our video codec
allow=h263p ; H.263p is the enhanced video codec
qualify=yes
notifyringing=yes
[101]
type=friend ; "friend" means this device takes and makes
calls
username=101 ; Username on device
callerid=Eric <102>
secret=101 ; Password for device
host=dynamic ; This host is not on the same IP addr every
time
context=default ; Inbound calls from this host go here
mailbox=101@default; Activate the message waiting light if this
canreinvite=no ; Leave this alone for now; see archives for
details
nat=1
qualify=yes
Subscribecontext=default
notifyringing=yes
##extensions.conf##
[general]
static=yes
writeprotect=no
autofallthrough=yes
priorityjumping=yes
[globals]
CONSOLE=Console/dsp ; Console interface for
demo
;CONSOLE=Zap/1
;CONSOLE=Phone/phone0
IAXINFO=guest ; IAXtel
username/password
;IAXINFO=myuser:mypass
TRUNK=Zap/g2
[default]
exten => 101,hint,SIP/101
exten => 102,hint,SIP/102
exten => 101,1,dial(sip/101,20,tw)
exten => 101,n,voicemail(101)
exten => 101,n,hanup()
exten => 102,1,dial(sip/102,20,tw)
exten => 102,n,voicemail(102)
exten => 102,n,hanup()
Commands from the CLI
CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
102/102 206.173.108.30 D N 5060 OK (5
ms)
101/101 206.173.108.25 D N 5060 OK (5
ms)
2 sip peers [Monitored: 2 online, 0 offline Unmonitored: 0 online, 0
offline]
CLI> show hints
-= Registered Asterisk Dial Plan Hints =-
102@default : SIP/102
State:Idle Watchers 1
101@default : SIP/101
State:Idle Watchers 1
----------------
- 2 hints registered
CLI> sip show subscriptions
Peer User Call ID Extension Last state
Type Mailbox
206.173.108.30 102 fb84429adb2 101@default Idle
dialog-info+xml <none>
206.173.108.25 101 499798bcfa4 102@default Idle
dialog-info+xml <none>
2 active SIP subscriptions
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Watkins, Bradley
2006-Sep-20 09:31 UTC
[asterisk-users] HINT problems with SVN-trunk-r43322
You will need to change the type=friend to type=peer and also define call-limit to some value (it can be large if you don't care about the actual limit). That should fix hints for you. Regards, - Brad ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Hall, Eric M. Sent: Wednesday, September 20, 2006 11:39 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] HINT problems with SVN-trunk-r43322 I'm unable to get HINTS working with the new SVN-Trunk State never changed when ringing or on the phone. Below is my configs (Maybe I missed something) Thanks for any help you could give!! ##sip.conf## [general] callerid=unavailable context=default ; Default context for incoming calls bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) ;allow=all allow=ulaw allow=g729 ;allow=gsm ;maxexpirey=3600 ; Max length of incoming registration we allow ;defaultexpirey=120 ; Default length of incoming/outoing registration ;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY videosupport=yes allow=h263 ; H.263 is our video codec allow=h263p ; H.263p is the enhanced video codec qualify=yes notifyringing=yes [101] type=friend ; "friend" means this device takes and makes calls username=101 ; Username on device callerid=Eric <102> secret=101 ; Password for device host=dynamic ; This host is not on the same IP addr every time context=default ; Inbound calls from this host go here mailbox=101@default; Activate the message waiting light if this canreinvite=no ; Leave this alone for now; see archives for details nat=1 qualify=yes Subscribecontext=default notifyringing=yes ##extensions.conf## [general] static=yes writeprotect=no autofallthrough=yes priorityjumping=yes [globals] CONSOLE=Console/dsp ; Console interface for demo ;CONSOLE=Zap/1 ;CONSOLE=Phone/phone0 IAXINFO=guest ; IAXtel username/password ;IAXINFO=myuser:mypass TRUNK=Zap/g2 [default] exten => 101,hint,SIP/101 exten => 102,hint,SIP/102 exten => 101,1,dial(sip/101,20,tw) exten => 101,n,voicemail(101) exten => 101,n,hanup() exten => 102,1,dial(sip/102,20,tw) exten => 102,n,voicemail(102) exten => 102,n,hanup() Commands from the CLI CLI> sip show peers Name/username Host Dyn Nat ACL Port Status 102/102 206.173.108.30 D N 5060 OK (5 ms) 101/101 206.173.108.25 D N 5060 OK (5 ms) 2 sip peers [Monitored: 2 online, 0 offline Unmonitored: 0 online, 0 offline] CLI> show hints -= Registered Asterisk Dial Plan Hints =- 102@default : SIP/102 State:Idle Watchers 1 101@default : SIP/101 State:Idle Watchers 1 ---------------- - 2 hints registered CLI> sip show subscriptions Peer User Call ID Extension Last state Type Mailbox 206.173.108.30 102 fb84429adb2 101@default Idle dialog-info+xml <none> 206.173.108.25 101 499798bcfa4 102@default Idle dialog-info+xml <none> 2 active SIP subscriptions The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060920/609ff862/attachment-0001.htm
On Wed, 2006-09-20 at 11:39 -0400, Hall, Eric M. wrote:> I?m unable to get HINTS working with the new SVN-Trunk > > State never changed when ringing or on the phone.Confirmed here, I only noticed because of this message. -- Dave Cotton <dcotton@linuxautrement.com>
Just found out this may only been a sip problem. State work with zap and SCCP when checking status via cli -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Dave Cotton Sent: Wednesday, September 20, 2006 12:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] HINT problems with SVN-trunk-r43322 On Wed, 2006-09-20 at 11:39 -0400, Hall, Eric M. wrote:> I?m unable to get HINTS working with the new SVN-Trunk > > State never changed when ringing or on the phone.Confirmed here, I only noticed because of this message. -- Dave Cotton <dcotton@linuxautrement.com> _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Watkins, Bradley
2006-Sep-21 03:10 UTC
[asterisk-users] HINT problems with SVN-trunk-r43322
The reason is that, at least in the SIP channel in trunk, the structure that keeps track of device state for hinting only gets allocated on peer objects and then only if call-limit is configured to some value. It's been a long time since I've done any development with 1.2 (all my 1.2 systems are waiting for 1.4 to come out so we can add a bunch of features), so I forget how that works there. Rumor has it these restrictions aren't necessary, but I forget. If by '6 months' you mean trunk from that long ago, it's entirely plausible that you got a snapshot during the evolution from where it was in 1.2 to where it is today. Regards, - Brad> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > Hall, Eric M. > Sent: Wednesday, September 20, 2006 10:07 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [asterisk-users] HINT problems with SVN-trunk-r43322 > > Group > Looks like the > > type=peer > call-limit=2 > > Works. Now the question is why? The sample I sent is working > on a system build 6 months ago. > Will do some more checking and will report to the list on > anything I find... > > Thanks Bradley for this bit of info you gave!! > > > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of > Andrew Kohlsmith > Sent: Wednesday, September 20, 2006 1:36 PM > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] HINT problems with SVN-trunk-r43322 > > On Wednesday 20 September 2006 12:31, Watkins, Bradley wrote: > > You will need to change the type=friend to type=peer and > also define > > call-limit to some value (it can be large if you don't care > about the > > actual limit). That should fix hints for you. > > But if you have it set to >1 then busy status won't work, > isn't that the case? > > -A. > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > >The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it.