Ticket #1884 (closed Bugs: fixed)

Opened 5 years ago

Last modified 5 years ago

FreePBX and compatibility with Asterisk 1.4

Reported by: francesco_r Assigned to:
Priority: major Milestone: 2.3
Component: Core Version: 2.3-branch
Keywords: Cc:
Confirmation: SVN Revision (if applicable):
Backend Engine: Asterisk 1.4.x Backend Engine Version:

Description

I did some compatibility tests with freePBX 2.2/asterisk 1.4 and works quite well. Followings, some little modifications to make it works. In sip.conf for example i added this to enable BLF of my phones and jitterbuffer: 

notifyringing=yes
notifyhold=yes
limitonpeers=yes
jbenable = yes
jbforce=yes

and in sip section of core functions.php 

$tmparr['call-limit'] = array('value' => '4', 'level' => 1);

In functions.php of queue module, change all "$CALLERIDNAME" (also obsolete for asterisk 1.2) to "${CALLERID(name)}".

In functions.php of meetme module, change line 18 from: 

$output .= 'conf => '.$meetme."|".$this->_meetmes[$meetme]."\n";

to 

$output .= 'conf => '.$meetme.",".$this->_meetmes[$meetme]."\n";

Change History

04/05/07 07:49:09 changed by p_lindheimer

  • owner deleted.
  • version changed from 2.2.1 to SVN-HEAD.
  • component changed from - choose - to Core.

04/26/07 13:41:03 changed by gregmac

  • engine changed from All to Asterisk 1.4.x.

(follow-up: ↓ 4 ) 05/31/07 11:37:58 changed by p_lindheimer

r3990 and r3991 address the meetme conferencing syntax issue, should be like that on 2.2 also.

Will have a look at the other suggestions - those don't look 1.4 specific (if so, please elaborate).

(in reply to: ↑ 3 ) 05/31/07 21:07:38 changed by francesco_r

Replying to p_lindheimer:

r3990 and r3991 address the meetme conferencing syntax issue, should be like that on 2.2 also. Will have a look at the other suggestions - those don't look 1.4 specific (if so, please elaborate). 

notifyringing=yes
notifyhold=yes
limitonpeers=yes

and also "call-limit" in sip peers are necessary to make hints works with asterisk 1.4. Others essential changes in extensions.conf and other modules like queue: change ${TIMESTAMP} with ${STRFTIME(${EPOCH}%Y%m%d-%H%M%S)}.

06/09/07 05:06:18 changed by p_lindheimer

  • version changed from SVN-HEAD to 2.3-branch.

06/18/07 14:06:33 changed by p_lindheimer

  • status changed from new to closed.
  • resolution set to fixed.

ok - everything but: 

$tmparr['call-limit'] = array('value' => '4', 'level' => 1);

has been added. It is not clear to me what really breaks or not and I'm concerned this may create more problems. However, I'm counting on you guys (as in the general beta community) to help flush this out so please re-open and give me some specifics on this if that is really needed to get the equivalent functionality as we get with Asterisk 1.2.

06/18/07 21:30:44 changed by francesco_r

For more info see this issue in the Digium Issue tracker: http://bugs.digium.com/view.php?id=8800

07/14/07 03:43:31 changed by francesco_r

  • status changed from closed to reopened.
  • resolution deleted.

I have reopened this ticket because i had some trouble with jitterbuffer present in asterisk 1.4. I think that the followig optional parameters now present in sip.conf (that i have suggested some time ago), is better to be removed. If someone want jitterbuffer can enable in sip_custom.conf. 

jbenable = yes
jbforce=yes

I suggest also to increase the default call-limit parameter to 10 (now is 4). Thank you

07/17/07 19:01:24 changed by brettm

I have chased this for ages on 1.4.7.1, and have found that if the peer call-limit is not set, extensions involved in transfers etc, eventually have their device state go haywire. Check out http://bugs.digium.com/view.php?id=8800 which lists out this problem in great detail. I had 5 extensions that had their hint show as on hold forever, which affected BLF's on other phones.

07/20/07 10:22:59 changed by p_lindheimer

  • priority changed from minor to major.

ok - it looks like call-limits will be necessary for 1.4 - what is the implication for 1.2 though??? Anyone play with it on 1.2 to see if it has any effect? I'm thinking of doing a call-limits default of 10 and then it can be set as people see fit per extension. But - need to know what it does to 1.2?

07/26/07 11:42:14 changed by p_lindheimer

  • status changed from reopened to closed.
  • resolution set to fixed.

I think call-limit is all that is left here. #2127 has been logged, closing this a fixed for the others. If I missed something, please open a NEW bug and reference this one for history.