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 ? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110502/f2ce0eb7/attachment.htm>
_____ 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. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110502/c1a3af5f/attachment.htm>
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.
--
_____________________________________________________________________
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satish patel
2011-May-02 17:57 UTC
[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
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Danny Nicholas
2011-May-02 18:00 UTC
[asterisk-users] [SOLVED] asterisk call completion issue
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>