[Asterisk-Users] Zaptel compile errors on x86_64

Wed Mar 15 07:21:50 MST 2006

Yep,

kernel-devel-2.6.9-22.EL
kernel-devel-2.6.9-34.EL
kernel-2.6.9-34.EL
kernel-utils-2.4-13.1.69
kernel-smp-devel-2.6.9-34.EL
kernel-2.6.9-22.EL
glibc-kernheaders-2.4-9.1.98.EL

all installed...

> ------------------------------
>
> Message: 18
> Date: Wed, 15 Mar 2006 11:18:36 +0100
> From: Dave Cotton <dcotton at linuxautrement.com>
> Subject: Re: [Asterisk-Users] Zaptel compile errors on x86_64
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <1142417917.22407.0.camel at RobinHood.LinuxAutrement.local>
> Content-Type: text/plain
>
> On Wed, 2006-03-15 at 17:49 +0800, Walter Klomp wrote:
>> Hi,
>>
>> Just downloaded the latest cvs from zaptel on my sparking new Athlon64
>> Centos4.2 system, but hitting a stumbling block... (sorry for the long 
>> post)
>
> Kernel source installed?
> -- 
> Dave Cotton <dcotton at linuxautrement.com>
>
>
>
> ------------------------------
>
> Message: 19
> Date: Wed, 15 Mar 2006 15:19:32 +0500
> From: "Mazhar Hussain" <mmazhar at gmail.com>
> Subject: [Asterisk-Users] There is lacking behind in recorded calls
> via sox
> To: asterisk-users at lists.digium.com,
> asterisk-users-request at lists.digium.com
> Message-ID:
> <e9dcec110603150219w72605510wadf5712945acda35 at mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Hi ,
>
>
>
> I have been using, Monitor (wav, ${CALLFILENAME}) to records calls from
> extensions.conf and soxmix software to compiles calls. The problem I am
> facing that for long calls more that 2 minutes there is disturbance in
> sequence of calls, calls from both ends are not in sequence and there is
> always lack behind from one end, I have tried a lot by checking different
> version of Sox  but still I am facing same issue .Can any one of you will
> let me know the reason of this lacking in calls form end after compilation
> while calls conversation goes fine in live calls .
>
>
>
>
>
>
>
> Thanks,
>
> Mazhar
>
> Nettechltd.com
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> ------------------------------
>
> Message: 20
> Date: Wed, 15 Mar 2006 15:23:08 +0500
> From: "Mazhar Hussain" <mmazhar at gmail.com>
> Subject: [Asterisk-Users] there is lack behind in recoded calls via
> sox
> To: asterisk-users-request at lists.digium.com,
> asterisk-users at lists.digium.com
> Message-ID:
> <e9dcec110603150223q5fb6d90en4ee5d2957a483274 at mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Hi ,
>
>
>
> I have been using, Monitor (wav, ${CALLFILENAME}) to records calls from
> extensions.conf and soxmix software to compiles calls. The problem I am
> facing that for long calls more that 2 minutes there is disturbance in
> sequence of calls, calls from both ends are not in sequence and there is
> always lack behind from one end, I have tried a lot by checking different
> version of Sox  but still I am facing same issue .Can any one of you will
> let me know the reason of this lacking in calls form end after compilation
> while calls conversation goes fine in live calls .Also I am using Asterisk
> 1.2.5 version
>
>
>
>
>
>
>
> Thanks,
>
> Mazhar
>
> Nettechltd.com
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> ------------------------------
>
> Message: 21
> Date: Wed, 15 Mar 2006 11:23:29 +0100
> From: "Alejandro Vargas" <alejandro.anv at gmail.com>
> Subject: [Asterisk-Users] spa 3000/2100 noise
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>
> Message-ID: <acb80a700603150223u459ece7o at mail.gmail.com>
> Content-Type: text/plain; charset=ISO-8859-1
>
> I've a problem. I've some spa3000 and spa2100. Asterisk 1.2.4.
> Prefered codec g711u in both. Calleng from a fxs of spa2100 to the fxo
> of spa3000, all works ok. Then I call from a sip phone configured for
> using g729, to the fxo of spa3000, it also works ok.
>
> The problem is that after this, when, making again a new call from
> spa2100 to spa3000, spa2100 receives only white noise. I suspect a
> codec mismatch. The problem disappears by powering off and on the
> spa3000.
>
> ¿Any ideas on how to check?
>
> --
> Alejandro Vargas
>
>
> ------------------------------
>
> Message: 22
> Date: Wed, 15 Mar 2006 11:29:37 +0100
> From: "Alejandro Vargas" <alejandro.anv at gmail.com>
> Subject: Re: [Asterisk-Users] Asterisk to receive fax
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>
> Message-ID: <acb80a700603150229l16381e71u at mail.gmail.com>
> Content-Type: text/plain; charset=ISO-8859-1
>
> 2006/3/15, Gidean Chan <gidean at navitele.com>:
>> Can anyone tell me how to configure my system so that fax can be received
>> and forward to email account?
>
> You can install iaxfax. It acts as a software modem that connects to
> asterisk as a iax phone. It creates a device that can be accesed as a
> faxmodem. Then, you can use hylafax that is very powerfull and can be
> configured to forward faxes to email, convert it to pdf, etc. etc
> (read the documentation).
>
> --
> Alejandro Vargas
>
>
> ------------------------------
>
> Message: 23
> Date: Wed, 15 Mar 2006 05:34:08 -0500
> From: <brett at websmyths.com>
> Subject: Re: [Asterisk-Users] Stuck. Extenions.conf? Realtime? MySQL?
> Grrrrr!
> To: asterisk-users at lists.digium.com
> Message-ID: <HSJVVZ2o.1142418843.2730200.brett at websmyths.com>
> Content-Type: text/plain; charset=ISO-8859-1
>
> On 3/15/2006, "Douglas Garstang" <dgarstang at oneeighty.com> wrote:
>>Boy, am I stuck...
> [snip]
>>My brain hurts.
>
> Doug,
>
> Whenever I have gotten to this point in a project, I use two rules for
> handling the situation.
>
> Rule 1. Booze
> Rule 2. Throw money at it.
>
> Rule 1 makes me feel better.
> Rule 2 takes care of the problem but...
> If the boss isn't happy - fall back to Rule 1.
>
> The hardest target to hit in the programming shooting gallery is the
> moving one.  Unless you 'sold' the powers that be that Asterisk is the
> answer to all questions... then you made your bed... but as I remember,
> I think you got 'stuck' with this one.
>
> You can probably (but I doubt it) buy a system that will do all this
> for you. Probably not out of the box tho and probably not without a
> large 'programmers' bill to boot. And several third-party packages.
>
> So grab your favorite alcoholic beverage, nail down what they want, and
> start solving the problems.  Even if it takes a year - it will be better
> and cheaper than anything they can purchase.
>
> Brett
>
>
> ------------------------------
>
> Message: 24
> Date: Wed, 15 Mar 2006 21:40:41 +1100
> From: "James Harper" <james.harper at bendigoit.com.au>
> Subject: RE: [Asterisk-Users] Re: Cisco phones and Linksys SRW224P
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>
> Message-ID: <AEC6C66638C05B468B556EA548C1A77DAF0CA0 at trantor>
> Content-Type: text/plain; charset="us-ascii"
>
>>
>> One more thing. Cisco 7905 phone that is working is 74-3092-04 Rev.F0.
>> Cisco 7905 phone that is not working is 74-3092-08 Rev.A0.
>>
>> Anybody know about any hardware issue with this revisions?
>>
>
> Nothing for sure, and you may already know this, but some early Cisco
> phones only knew how to speak Cisco PoE, not the 802 standard which was
> defined a bit later. The Cisco web site should tell you which phone
> talks which protocol though.
>
> James
>
>
>
> ------------------------------
>
> Message: 25
> Date: Wed, 15 Mar 2006 11:44:40 +0100
> From: Simone Cittadini <mymailforlists at gmail.com>
> Subject: [Asterisk-Users] (unexplicable) peaks of machine load
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <4417F018.2030403 at gmail.com>
> Content-Type: text/plain; charset=ISO-8859-15; format=flowed
>
> I have strange peaks of machine load on my asterisk servers, looking at
> top the load is very high even if cpu usage is low and no swap memory is
> used.
>
> This happens on all the machines, some of them have asterisk, mysql, agi
> and digium cards on them, so I thought I was only asking too much, but
> yesterday I noticed the same behaviour on an asterisk machine with only
> two digium in it, no other service and a two line extension.
> I thought it can be a problem with digium cards but the interrupts
> aren't shared, and I have the same problem on a pure-voip server.
>
> Asterisk version varies from 1.2.1 to 1.2.5, the kernels are 2.4 or 2.6
> (right ones for the installed cpu, not generic 386)
> The only things in common are :
>
> Linux debian, iax channels are used, with jitterbuffer
>
> When this "ghost load" becomes too high (> 3) asterisk starts losing
> packets, and the users starts losing patience ...
>
> Anyone experiencing a similar problem ?
>
>
>
> ------------------------------
>
> Message: 26
> Date: Wed, 15 Mar 2006 05:56:20 -0500
> From: "Matt Florell" <astmattf at gmail.com>
> Subject: Re: [Asterisk-Users] (unexplicable) peaks of machine load
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>
> Message-ID:
> <61575c810603150256l776ce256le3c691bdbea2f557 at mail.gmail.com>
> Content-Type: text/plain; charset=ISO-8859-1
>
> I've noticed this as well from pre 1.0 versions through to 1.2.5
> across 12 separate Asterisk servers. The severity seems to be random
> mostly. I still haven't figured out what is causing it.
>
> MATT---
>
> On 3/15/06, Simone Cittadini <mymailforlists at gmail.com> wrote:
>> I have strange peaks of machine load on my asterisk servers, looking at
>> top the load is very high even if cpu usage is low and no swap memory is
>> used.
>>
>> This happens on all the machines, some of them have asterisk, mysql, agi
>> and digium cards on them, so I thought I was only asking too much, but
>> yesterday I noticed the same behaviour on an asterisk machine with only
>> two digium in it, no other service and a two line extension.
>> I thought it can be a problem with digium cards but the interrupts
>> aren't shared, and I have the same problem on a pure-voip server.
>>
>> Asterisk version varies from 1.2.1 to 1.2.5, the kernels are 2.4 or 2.6
>> (right ones for the installed cpu, not generic 386)
>> The only things in common are :
>>
>> Linux debian, iax channels are used, with jitterbuffer
>>
>> When this "ghost load" becomes too high (> 3) asterisk starts losing
>> packets, and the users starts losing patience ...
>>
>> Anyone experiencing a similar problem ?
>>
>> _______________________________________________
>> --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
>>
>
>
> ------------------------------
>
> Message: 27
> Date: Wed, 15 Mar 2006 22:13:23 +1100
> From: James Gardiner <asterisk at crafted.com.au>
> Subject: [Asterisk-Users] CALL FOR COMMENTS - Dialplan
> To: asterisk-users at lists.digium.com, asterisk-dev at lists.digium.com
> Message-ID: <4417F6D3.6050507 at crafted.com.au>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
>
> Hello Asterisk community,
> I have written a document that covers an Asterisk implementation I am
> building.
> I want to place it on the lists so USERS can view and make comments on,
> the ideas contained within.
>
> I think it is an important issue to develop a standardised Dialplan for
> applications, not just for Asterisk, but for pbx systems in general.  As
> they become cheaper and more common place, each install has its own
> ideas of how to implement features.
>
> This makes it very hard for users to move from one system to another.
>
> In any case,
> If you have time, please do review the document and make comments to the
> list or to me directly.
>
> The document can be found at http://www.crafted.com.au/comments
> I cannot post it directly to the list as its TOO BIG.
>
> Thanks,
> James
>
>
>
>
> ------------------------------
>
> Message: 28
> Date: Wed, 15 Mar 2006 22:13:23 +1100
> From: James Gardiner <asterisk at crafted.com.au>
> Subject: [Asterisk-Users] [SPAM] [asterisk-dev] CALL FOR COMMENTS -
> Dialplan
> To: asterisk-users at lists.digium.com, asterisk-dev at lists.digium.com
> Message-ID: <4417F6D3.6050507 at crafted.com.au>
> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
>
>
> Hello Asterisk community,
> I have written a document that covers an Asterisk implementation I am
> building.
> I want to place it on the lists so USERS can view and make comments on,
> the ideas contained within.
>
> I think it is an important issue to develop a standardised Dialplan for
> applications, not just for Asterisk, but for pbx systems in general.  As
> they become cheaper and more common place, each install has its own
> ideas of how to implement features.
>
> This makes it very hard for users to move from one system to another.
>
> In any case,
> If you have time, please do review the document and make comments to the
> list or to me directly.
>
> The document can be found at http://www.crafted.com.au/comments
> I cannot post it directly to the list as its TOO BIG.
>
> Thanks,
> James
>
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-dev mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-dev
>
>
> ------------------------------
>
> _______________________________________________
> --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
>
>
> End of Asterisk-Users Digest, Vol 20, Issue 98
> **********************************************
> 