[asterisk-users] [SOLVED] asterisk call completion issue

Danny Nicholas danny at debsinc.com
Mon May 2 13:00:38 CDT 2011


If I recall correctly, callcounter supercedes call-limit in 1.8.

 

  _____  

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of satish patel
Sent: Monday, May 02, 2011 12:57 PM
To: asterisk-users
Subject: Re: [asterisk-users] [SOLVED] asterisk call completion issue

 


After adding callcounter=yes at sip.conf it works! 

Cheers!



  _____  

From: satish_lx at hotmail.com
To: asterisk-users at lists.digium.com
Date: Mon, 2 May 2011 17:34:32 +0000
Subject: Re: [asterisk-users] asterisk call completion issue


I have call-limit=1 at sip.conf



  _____  

From: danny at debsinc.com
To: asterisk-users at lists.digium.com
Date: Mon, 2 May 2011 12:20:40 -0500
Subject: Re: [asterisk-users] asterisk call completion issue

  _____  

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of satish patel
Sent: Monday, May 02, 2011 12:19 PM
To: asterisk-users
Subject: [asterisk-users] asterisk call completion issue

 

Hi All,

I am testing CC feature with asterisk 1.8 but i am having some issue. We
have polycom 501 SIP phone and those are configured with two line with same
extensions. When i am requesting for CC i am not getting call back from
asterisk but it works if i reboot my polycom phone ( In short when phone get
register ) 

Is this because of two line configured ? or some configuration issue ? 

[Danny Nicholas] 

I would check call-limit and see what reducing that would do for you.


-- _____________________________________________________________________ --
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 
-- _____________________________________________________________________ --
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 

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110502/76cd57d8/attachment.htm>


More information about the asterisk-users mailing list