Date: Mon, 5 Nov 2007 15:30:49 +0500
From: "Rizwan Hisham" <rizwanhasham@gmail.com>
Subject: Re: [asterisk-users] asterisk-users Digest, Vol 40, Issue 5
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users@lists.digium.com>
Message-ID:
<4809880c0711050230i7131d31bo394b5350e978334b@mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1
i dont know how to remove these errors. But i think you should try
asterisk-addons package available from asterisk download site. it
contains the h323 channel also. You only need to compile it. remove
the asterisk-oh323 package from your system and install the
asterisk-addons package. I hope this solves your problem.
On Nov 5, 2007 8:42 AM, Bincy K. Philip <bincy.philip@nestgroup.net> wrote:
> Hello
>
>
> Thanks for the reply..
>
> I could use Asterisk as SIP server and establish call using two SIP phones.
>
> But I need H323 support also.
>
> For that I have compiled the files in asterisk/channel/h323 and installed without problem.
> But even after i have started Asterisk,it is not supporting h323 commands like h323 debug,h323 show codecs.
>
> So i tried to install compile asterisk-oh323. i got an error that channel_pvt.h is missing..when i downloaded and put the same file i got double declaration error.
> I have excluded channel_pvt.h from chan_oh323.c include file list, but got errors.
> Anyone please help!!!!!
>
>
> Thanks & Regards
> Bincy K Philip
>
>
>
>
>
------------------------------
Message: 8
Date: Mon, 05 Nov 2007 01:52:24 +0200
From: Michael Davidson <michael@bbd.co.za>
Subject: [asterisk-users] Need Reference sites
To: asterisk-users@lists.digium.com
Message-ID: <472E5B38.2080609@bbd.co.za>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Hi,
I'am comparative newbie to the world of Asterisk. I'd like to
introduce an Asterisk based PBX into my company but need to convince my
executive of it's worthiness. I need some reference sites to quote in my
discussion, preferably well known companies of course. I have surfed the
net but not come up with anything of note, if anyone can help it would
be greatly appreciated.
Thanks, Mike D.
------------------------------
Message: 9
Date: Mon, 05 Nov 2007 11:17:39 +1100
From: Paul Hales <pdhales@optusnet.com.au>
Subject: Re: [asterisk-users] 7960 Queue Issue
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Message-ID: <1194221859.3696.2.camel@localhost.localdomain>
Content-Type: text/plain
My memory tells me that there is a flag (something like 'ringinuse')
which can make sure this sort of thing does not happen.
PaulH
On Mon, 2007-11-05 at 10:26 +1100, Nick Brown wrote:
> Morning All,
>
> Quick question that has me stumped. Have a queue with several members
> (Statically defined in queues.conf at this stage for testing) who use Cisco
> 7960's.
>
> The queue is configured to use rrmemory and generally this works correctly.
> However if a member is already on a call their phone will still ring (The
> 7960 can show multiple incoming calls for one line). I really don't want
> members who are on calls to get more calls. Especially when we start logging
> out members who don't answer.
>
> Asterisk shows;
> -- Called 1014
> -- SIP/1014-08f2e4d0 is ringing
> -- Local/1014@queuestations-e3e2;1 is ringing
> -- Nobody picked up in 15000 ms
>
> Short of disabling the feature to show multiple incoming calls on the 7960's
> (Which I don't know if it can be done anyway), has anyone got any
> suggestions?
>
> Thanks in advance!
>
> Nick.
>
>
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>
http://lists.digium.com/mailman/listinfo/asterisk-users
------------------------------
Message: 10
Date: Mon, 5 Nov 2007 00:51:10 +0000
From: "Frank Church" <voipfc@googlemail.com>
Subject: [asterisk-users] Are the ATAs which can allow multiple
extensions from one network connection?
To: asterisk-users@lists.digium.com
Message-ID:
<84b7c6460711041651g1d597f95rb093cbdcc17a142d@mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1
Are there ATAs that allow different phone numbers from one network connection?
Such as supporting multiple IP addresses so that each RJ11 has a
different extension or some other way?
------------------------------
Message: 11
Date: Sun, 4 Nov 2007 19:57:07 -0500
From: "Eric Merkel" <ejmerkel@gmail.com>
Subject: Re: [asterisk-users] 7960 Queue Issue
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users@lists.digium.com>
Message-ID:
<4ae05cce0711041657g26e4d4ban3a746a64ed30fd0d@mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1
On 11/4/07, Nick Brown <Nick@ipera.com.au> wrote:
> Morning All,
>
> Quick question that has me stumped. Have a queue with several members
> (Statically defined in queues.conf at this stage for testing) who use Cisco
> 7960's.
>
> The queue is configured to use rrmemory and generally this works correctly.
> However if a member is already on a call their phone will still ring (The
> 7960 can show multiple incoming calls for one line). I really don't want
> members who are on calls to get more calls. Especially when we start logging
> out members who don't answer.
>
> Asterisk shows;
> -- Called 1014
> -- SIP/1014-08f2e4d0 is ringing
> -- Local/1014@queuestations-e3e2;1 is ringing
> -- Nobody picked up in 15000 ms
>
> Short of disabling the feature to show multiple incoming calls on the 7960's
> (Which I don't know if it can be done anyway), has anyone got any
> suggestions?
>
Yes, you can turn off this in the phone. Go into call preferences on
the phone and turn off call waiting. Not optimal but can be done.
-Eric
------------------------------
Message: 12
Date: Mon, 05 Nov 2007 12:09:48 +1100
From: Nick Brown <Nick@ipera.com.au>
Subject: Re: [asterisk-users] 7960 Queue Issue
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Message-ID: <C354B88C.803%Nick@ipera.com.au>
Content-Type: text/plain; charset="US-ASCII"
Thanks Eric, this is the case. A bit of a shame that it removes the
functionality for the member to see calls that have not come from a queue
however there is not much choice in the matter.
FWIW to get this option a firmware upgrade was required (Now running
POS3-08-8-00).
Cheers.
On 5/11/07 11:57 AM, "Eric Merkel" <ejmerkel@gmail.com> wrote:
> On 11/4/07, Nick Brown <Nick@ipera.com.au> wrote:
>> Morning All,
>>
>> Quick question that has me stumped. Have a queue with several members
>> (Statically defined in queues.conf at this stage for testing) who use Cisco
>> 7960's.
>>
>> The queue is configured to use rrmemory and generally this works correctly.
>> However if a member is already on a call their phone will still ring (The
>> 7960 can show multiple incoming calls for one line). I really don't want
>> members who are on calls to get more calls. Especially when we start logging
>> out members who don't answer.
>>
>> Asterisk shows;
>> -- Called 1014
>> -- SIP/1014-08f2e4d0 is ringing
>> -- Local/1014@queuestations-e3e2;1 is ringing
>> -- Nobody picked up in 15000 ms
>>
>> Short of disabling the feature to show multiple incoming calls on the 7960's
>> (Which I don't know if it can be done anyway), has anyone got any
>> suggestions?
>>
>
> Yes, you can turn off this in the phone. Go into call preferences on
> the phone and turn off call waiting. Not optimal but can be done.
>
> -Eric
>
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>
http://lists.digium.com/mailman/listinfo/asterisk-users
------------------------------
Message: 13
Date: Sun, 4 Nov 2007 20:20:21 -0500
From: Dave Bour <dcbour@desktopsolutioncenter.ca>
Subject: Re: [asterisk-users] Are the ATAs which can allow multiple
extensions from one network connection?
To: "voipfc@gmail.com" <voipfc@gmail.com>, Asterisk Users Mailing List
- Non-Commercial Discussion <asterisk-users@lists.digium.com>
Message-ID:
<40EAA9DE9015244685020EE950693D021AF17BE90B@VMBX102.ihostexchange.net>
Content-Type: text/plain; charset="us-ascii"
Your question seems to be two I think so I've covered both options here.
Mediatrix does two different series of boxes - 4 port version ...
4 port to extensions - a 1104 (also in 2 port and higher numbers too)...each is an analog line to a phone, ie extensions for house, small office, etc
4 port to analog standard Bell lines (phone numbers) - a 1204 (also in higher port numbers too) - ie, back to the telco for various incoming lines for business or multiple line home.
A number of other vendors also do boxes to do this. Both boxes each run from a single IP address (dhcp or static) per device, not per port. The box handles the extension/phone numbers in conjunction with what you tell it out of Asterisk.
Dave Bour
Desktop Solution Center
905.381.0077 X501
dcbour@desktopsolutioncenter.ca
http://www.desktopsolutioncenter.ca
> -----Original Message-----
> From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-
> bounces@lists.digium.com] On Behalf Of Frank Church
> Sent: Sunday, November 04, 2007 7:51 PM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Are the ATAs which can allow multiple
> extensions from one network connection?
>
> Are there ATAs that allow different phone numbers from one network
> connection?
>
> Such as supporting multiple IP addresses so that each RJ11 has a
> different extension or some other way?
>
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>
http://lists.digium.com/mailman/listinfo/asterisk-users
------------------------------
Message: 14
Date: Mon, 5 Nov 2007 02:08:45 +0000
From: jadams@clearcasetechnology.com
Subject: Re: [asterisk-users] 7960 Queue Issue
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users@lists.digium.com>
Message-ID:
<1769171547-1194228463-cardhu_decombobulator_blackberry.rim.net-1483053805-@bxe015.bisx.prod.on.blackberry>
Content-Type: text/plain
Have you tried the ringinuse option? This will not ring phones if they are busy.
Sent from my Verizon Wireless BlackBerry
-----Original Message-----
From: Nick Brown <Nick@ipera.com.au>
Date: Mon, 05 Nov 2007 10:26:19
To:"asterisk-users@lists.digium.com" <asterisk-users@lists.digium.com>
Subject: [asterisk-users] 7960 Queue Issue
Morning All,
Quick question that has me stumped. Have a queue with several members
(Statically defined in queues.conf at this stage for testing) who use Cisco
7960's.
The queue is configured to use rrmemory and generally this works correctly.
However if a member is already on a call their phone will still ring (The
7960 can show multiple incoming calls for one line). I really don't want
members who are on calls to get more calls. Especially when we start logging
out members who don't answer.
Asterisk shows;
-- Called 1014
-- SIP/1014-08f2e4d0 is ringing
-- Local/1014@queuestations-e3e2;1 is ringing
-- Nobody picked up in 15000 ms
Short of disabling the feature to show multiple incoming calls on the 7960's
(Which I don't know if it can be done anyway), has anyone got any
suggestions?
Thanks in advance!
Nick.
_______________________________________________
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
------------------------------
Message: 15
Date: Sun, 4 Nov 2007 21:09:17 -0500
From: "Joseph Begumisa" <joe@cfi.co.ug>
Subject: Re: [asterisk-users] Compatibility Issues with dell poweredge
1950 and TE110P card - Update
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
<asterisk-users@lists.digium.com>
Message-ID: <00bd01c81f50$df7cc540$9e764fc0$@co.ug>
Content-Type: text/plain; charset="US-ASCII"
> -----Original Message-----
> From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-
> bounces@lists.digium.com] On Behalf Of Paul Hales
> Sent: Wednesday, October 24, 2007 9:50 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Compatibility Issues with dell poweredge 195
and
> TE110P card
>
>
> We had issues with TE110p cards in Dell 860's, but TE120p's fixed the
> problem.
>
> PaulH
It is now 1 week since I replaced the TE110P with the TE120P in the Dell
Poweredge 1950 and I have not had any problems. The TE120P seems to have
resolved the earlier problem I had.
Joseph
------------------------------
Message: 16
Date: Mon, 05 Nov 2007 15:15:00 +1300
From: Duncan Turnbull <duncan@e-simple.co.nz>
Subject: Re: [asterisk-users] 7960 Queue Issue
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users@lists.digium.com>
Message-ID: <472E7CA4.4030503@e-simple.co.nz>
Content-Type: text/plain; charset=windows-1252; format=flowed
------------------------------
Message: 28
Date: Mon, 5 Nov 2007 15:30:49 +0500
From: "Rizwan Hisham" <rizwanhasham@gmail.com>
Subject: Re: [asterisk-users] asterisk-users Digest, Vol 40, Issue 5
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users@lists.digium.com>
Message-ID:
<4809880c0711050230i7131d31bo394b5350e978334b@mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1
i dont know how to remove these errors. But i think you should try
asterisk-addons package available from asterisk download site. it
contains the h323 channel also. You only need to compile it. remove
the asterisk-oh323 package from your system and install the
asterisk-addons package. I hope this solves your problem.
On Nov 5, 2007 8:42 AM, Bincy K. Philip <bincy.philip@nestgroup.net> wrote:
> Hello
>
>
> Thanks for the reply..
>
> I could use Asterisk as SIP server and establish call using two SIP phones.
>
> But I need H323 support also.
>
> For that I have compiled the files in asterisk/channel/h323 and installed without problem.
> But even after i have started Asterisk,it is not supporting h323 commands like h323 debug,h323 show codecs.
>
> So i tried to install compile asterisk-oh323. i got an error that channel_pvt.h is missing..when i downloaded and put the same file i got double declaration error.
> I have excluded channel_pvt.h from chan_oh323.c include file list, but got errors.
> Anyone please help!!!!!
>
>
> Thanks & Regards
> Bincy K Philip
>
>
>
>
>
> Date: Fri, 2 Nov 2007 17:50:57 +0500
> From: "Rizwan Hisham" <rizwanhasham@gmail.com>
> Subject: Re: [asterisk-users] asterisk as a gateway
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users@lists.digium.com>
> Message-ID:
> <4809880c0711020550k15b55b71q5b7590b669e4e0fb@mail.gmail.com>
> Content-Type: text/plain; charset=ISO-8859-1
>
> Hi,
> You should visit the following websites for help
> www.voip-info.org
> www.asteriskguru.com
> www.nerdvittles.com
>
> But the best step for beginners is to read the "Asterisk, The Future
> of Telephony" book which is available freely on asterisk website. It
> will help you great deal in understanding basics of asterisk.
>
> Im not sure about h323 but the book will help you to add some contents
> in extensions.conf. You can start with sip.conf instead coz its help
> is provided in the book.
>
> On Nov 2, 2007 2:26 PM, Bincy K. Philip <bincy.philip@nestgroup.net> wrote:
> >
> >
> >
> > Hello,
> >
> > Could anyone please give some information on configuring asterisk as a
> > gateway.
> > What contents have to add in h.323 .conf and extensions.conf files ?
> >
> > Thanks & Regards
> > Bincy K Philip
> >
> >
>
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>
http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
Best Regards
Rizwan Hisham
Software Engineer
Axvoice Inc.
www.axvoice.com
------------------------------
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End of asterisk-users Digest, Vol 40, Issue 11
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