date:20060412

In an effort to bring Asterisk Users from across the state of Texas together, the Texas Asterisk Users Group has been formed. The goal is to help Asterisk users meet other is their area and to help spread the word about the Asterisk community. I anticipate regional meetings of members and look forward to all of our members being able to attend the Astricon Event in Dallas in October.
 I invite all of the Asterisk users in Texas to visit our web new website, still under development, and join our group and our mailing list. I am currently taking suggestions on the format and backend of the website and look forward to helping build a stronger asterisk community in Texas.
-- BruceNortex Networks

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[Asterisk-Users] Problem with Voice Quality

2006-04-12 Thread mkumar

Hi All,

We are making a VOIP application for Mobiles (PDA's) and we are using 
Asterisk
for it. We have a setup consisting of both SER and Asterisk. SER acts as a SIP
router and routes everything to Asterisk. We also have rtpproxy for SER. Our
packet delivery from clients (Mobiles, PDA's) is inconsistent and ranges
between 10 to 60 ms delay but the average is near to 20 ms. We use SIP. How can
we solve this problem, is there any setting at the server end to handle this,
as clients have very limited resources we have to manage this at the server
end, please tell me how can I do this?

Thanks,
Manoj.

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RE: [Asterisk-Users] freepbx dialing prefix

2006-04-12 Thread Kerry Garrison




Submit a bug report to the FreePBX 
team?

  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Sean 
  GarlandSent: Wednesday, April 12, 2006 8:46 PMTo: 
  asterisk-users@lists.digium.comSubject: [Asterisk-Users] freepbx 
  dialing prefix
  
  
  I need to put a ‘w’ 
  in the dialing prefix, but it says it isn’t valid.  If I manually modify 
  the extension file, it then affects all calls made over any trunk.  Any 
  ideas?
   
  Sean
  --No virus found in this outgoing message.Checked by 
  AVG Free Edition.Version: 7.1.385 / Virus Database: 268.4.1/309 - Release 
  Date: 4/11/2006
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Re: [Asterisk-Users] How to terminate ringing call before it is answered

2006-04-12 Thread Darren Wiebe

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Dial

Checkout options "H" and "h".

Darren Wiebe
[EMAIL PROTECTED]


Obelix wrote:

>
> Is there a way to terminate a ringing call before it is answered?
>
> I am speaking of prepaid card application in which you want to make
> another call, because you current number it is not being answered,
> and you don't want to hangup before dialling again.
>
> /Obelix ___ --Bandwidth
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> http://lists.digium.com/mailman/listinfo/asterisk-users


-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.1 (GNU/Linux)
Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org

iD8DBQFEPcn24DADnh+tnOQRAjzdAJ9ZOQQZ2OHXtZCT1kDiT67YxmqewACZAXdg
vfg1ND0chkk7tFc5q3iPYrM=
=zFFk
-END PGP SIGNATURE-

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[Asterisk-Users] freepbx dialing prefix

2006-04-12 Thread Sean Garland









I need to put a ‘w’ in the
dialing prefix, but it says it isn’t valid.  If I manually modify the
extension file, it then affects all calls made over any trunk.  Any ideas?

 

Sean








--
No virus found in this outgoing message.
Checked by AVG Free Edition.
Version: 7.1.385 / Virus Database: 268.4.1/309 - Release Date: 4/11/2006
 
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[Asterisk-Users] How to terminate ringing call before it is answered

2006-04-12 Thread Obelix



Is there a way to terminate a ringing call before it is answered?

I am speaking of prepaid card application in which you want to make another
call, because you current number it is not being answered, and you don't want
to hangup before dialling again.

/Obelix
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Re: [Asterisk-Users] Re: update - 512 Simultaneous Callswith DigitalRecording

2006-04-12 Thread asterisk


On Wed, 12 Apr 2006, Leo Ann Boon wrote:
I'm not sure tmpfs is the right solution for the OP's problem - disk access 
slowing down the system. My understanding of tmpfs is that it will swap pages 
in and out to/from disk. Wouldn't that be as bad as directly writing to disk? 
I can see tmpfs will have some advantage over direct disk IO when the files 
are small and short-lived, i.e. less likely to be swapped.


One way around this is to not have swap at all. Then there is no disk i/o 
to worry about. Everything will be in ram.


This is what I do for embedded asterisk servers. tmpfs and no swap.

Ram is cheap.

Problem solved.

-Dan
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[Asterisk-Users] need help

2006-04-12 Thread Dirgan Putra

  hi All      I need your help , for used Digium Card TE405P, for setting this Board AS E1 ISDN PRI.     1 .Current for make sure my config its rights or no I inform my configurations in Board Jumper T1/E1 is Closed is that rights or no ? for E1 i closed the
 Jumper.        2. I Want To seeting E1 in ASterisk/PC  Back To Back    To Cisco E1 AS5300 Use ISDN Signaling,      my configutration :     softphone --- > ASterisk TE405P  E1 --->  E1 AS5300     I need your help me to rights configurations Zapata.conf and zaptel.conf     FYI iam already load :  modpobe zaptel  modprobe wct4xxp t1e1override=15 debug=1     and current /etc/zaptel.conf  span=1,0,0,ccs,hdb3,crc4,yellow  span=2,0,0,ccs,hdb3,crc4,yellow  span=3,0,0,ccs,hdb3,crc4,yellow  span=4,0,0,ccs,hdb3,crc4,yellow     current i have problem error message in cisco as5300 after make a call, if used debug isdn q931     in cisco AS5300 config :     isdn siwtch-type primary-net5     controller e1 0  framming crc4  linecode hdb3  pri-group 1-31        interface serial0:15  isdn switch-type primary-net5  isdn incomming-voice modem  isdn T310 6        dialpeer,.blah,...blahh        thanks for ur help     Dirgan   
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Re: [Asterisk-Users] SPA-941/942 Bulk provisioning

2006-04-12 Thread Ronald Wiplinger


[EMAIL PROTECTED] wrote:

On Tue, 11 Apr 2006, tracinet wrote:

Unfortunately, Linksys is reserving the provisioning tools/info to their
official resellers.  The idea is that you pay your Linksys reseller to
provision your phones (does not make ANY sense to me all).  As a service
provider, we should be able to buy these phones and have access to 
the bulk
provisioning tools.  You may want to ask your Linksys vendor and see 
if they
can provide these tools for you, but understand that they are under 
strict

orders to not disclose the tools.


No need. It can be done easily enough with xml. Linksys likes to hide 
the info, but you can dig around a bit and put the pieces together.


I'm in the process of writing an autoprovisioner which can handle 
fresh out-of-the-box linksys, snom, and grandstream with 0-config 
(other than entering the mac into a textfile). You never have to touch 
the phone. Just plug it in.


It's all done via http and php (yes, even grandstream).


Dan,

is there a way to configure softphones that way too, like x-lite?


bye

Ronald Wiplinger
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Re: [Asterisk-Users] SPA-941/942 Bulk provisioning

2006-04-12 Thread asterisk


On Tue, 11 Apr 2006, tracinet wrote:

Unfortunately, Linksys is reserving the provisioning tools/info to their
official resellers.  The idea is that you pay your Linksys reseller to
provision your phones (does not make ANY sense to me all).  As a service
provider, we should be able to buy these phones and have access to the bulk
provisioning tools.  You may want to ask your Linksys vendor and see if they
can provide these tools for you, but understand that they are under strict
orders to not disclose the tools.


No need. It can be done easily enough with xml. Linksys likes to hide the 
info, but you can dig around a bit and put the pieces together.


I'm in the process of writing an autoprovisioner which can handle fresh 
out-of-the-box linksys, snom, and grandstream with 0-config (other than 
entering the mac into a textfile). You never have to touch the phone. Just 
plug it in.


It's all done via http and php (yes, even grandstream).

-Dan
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[Asterisk-Users] RE: Asterisk-Users Digest, Vol 21, Issue 70

2006-04-12 Thread chan \(Alpha Trilogies Networks\)


Ok, 
I did check it before and nothings related to this "#" key, if it's then
system will announce that "Please key in the extensions", but not in this
case. By default, the blind transfer is #1.
Some one can help?



>Check your features.conf file for conflicting key set. # is the default 
>key for blind transfer feature.

>[]'s
>MM


chan (Alpha Trilogies Networks) wrote:
> Hi,
> Did someone experience that Asterisk OS 1.2.5 voicemail issues?
> Problem description:
> Some one call to the extensions 200,
> After 10 sec ring then go to voicemail [EMAIL PROTECTED]
> Announcement "Please leave me a messages.blar blar.."
> When I completed to leave a message...
> IF :
> I press the pound "#"key ...
> Then it says "Transfer"
> IF :
> I Press the zero "0"key
> Then it say "Please confirm your recording "
> IF :
> I hangup after leaving a message...then things get normal.
> 
> What is this 
> Funny.
> Pls some one reply.


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[Asterisk-Users] Multiple phones in same call

2006-04-12 Thread Rudolf Ladyzhenskii

Hi, all

This is what I would like to do:

Someone is on the phone and nother person ant to join in. Like in
house wheer all phones are connected to same line.
I can do it with MeetMe, but my understanding is that all parties have
to call "meeting room" number. What I want instead is to have some
"magic extension" or a "*" or "#" service that will allow people to
join in.
I am running Asterisk at my home and normally have 3 or 4 phones only,
so ideally I would number all phones 1-4 and pressing say "*1" on
phones 2-4 will join to the call on phone 1.

Is there an easy way of doing that?

Thanks,
Rudolf
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Re: [Asterisk-Users] help -- voicemail

2006-04-12 Thread El Flynn


chan (Alpha Trilogies Networks) wrote:

Hi,
Did someone experience that Asterisk OS 1.2.5 voicemail issues?
Problem description:
Some one call to the extensions 200,
After 10 sec ring then go to voicemail [EMAIL PROTECTED]
Announcement "Please leave me a messages.blar blar.."
When I completed to leave a message...
IF :
I press the pound "#"key ...
Then it says "Transfer"
IF :
I Press the zero "0"key
Then it say "Please confirm your recording "
IF :
I hangup after leaving a message...then things get normal.



check your Dial command, looks like you've enabled the CALLER to transfer -- 
which is why you get the "Transfer" when you hit the # key.


Flynn


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[Asterisk-Users] Text labels on incoming call appearances?

2006-04-12 Thread William P.N. Smith


Hi All,

I'm looking at an Asterix system for a small home office, 3 incoming 
lines and eight phones scattered about the house, probably with a GUI 
wrapper like SwitchVox to help with the administration.  I'm still 
looking at phones, and am getting pretty confused.  It looks like the 
phones have all the features _and_ the PBX has all the features, and 
I'm not sure if I'm buying too much phone for what I want to do.


I'm upgrading from a Venture 3-line KSU-less phone system with 8 
sets, which has the following features that we like:


3 lines
shared directory
paging (all or specific phone)
programmable keys (one for each phone, I can tell which is in use and 
can do individual pages)


I'm thinking that I need four-line phones, but with call appearances 
I'm no longer sure.  Can I do the following with 1-line or 2-line phones:


When line 1 rings, all the phones would display "Home" and the 
CID.  Someone would pick up their handset and be connected to that line.


Then line 2 rings, all phones display "Work" and the CID.  Someone 
picks up and gets that call.


Line 3 rings, all phones display "Fax" and the CID. More of the same.

It's easy to figure out how this works with four-line phones, but 
with 1 or 2-line phones, how can the person select which line they 
want to pick up?  Obviously it's easy to just "give me the line 
that's ringing", but if I want to put one on hold and have someone 
else pick up from somewhere else, or I want to make an outgoing call 
on the "Work" line, how do I control my line selection?


Shared directories is very important to us, I can enter a name and 
number at my desk and use it at any phone.  I don't seem to see this 
as a feature on the phones I'm looking at.  Can this be done by (say) 
reading all the phone's private directories, combining them and 
writing them back to all the phones?


In an ideal world, I could look at _any_ phone and see my voicemail 
indicator, is there a way to put up to four voicemail indicators on 
every phone in parallel?


I'm looking at Linksys SPA-941 or 942 phones, and maybe the Polycom 
IP-501, are there any better ones I'm missing?


Many thanks in advance!

--
William Smith [EMAIL PROTECTED]
ComputerSmiths Consulting, Inc. www.compusmiths.com

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[Asterisk-Users] ASterisk Back2back

2006-04-12 Thread Dirgan Putra

hi All      I need your help , for used Digium Card TE405P, for setting this Board AS E1 ISDN PRI.     1 .Current for make sure my config its rights or no I inform my configurations in Board Jumper T1/E1 is Closed is that rights or no ? for E1 i closed the Jumper.        2. I Want To seeting E1 in ASterisk/PC  Back To Back    To Cisco E1 AS5300 Use ISDN Signaling,      my configutration :     softphone --- > ASterisk ---> AS5300     I need your help me to rights
 configurations Zapata.conf and zaptel.conf     FYI iam already load :  modpobe zaptel  modprobe wct4xxp t1e1override=15 debug=1     and current /etc/zaptel.conf  span=1,0,0,ccs,hdb3,crc4,yellow  span=2,0,0,ccs,hdb3,crc4,yellow  span=3,0,0,ccs,hdb3,crc4,yellow  span=4,0,0,ccs,hdb3,crc4,yellow     current i have problem error message in cisco as5300 after make a call, if used debug isdn q931     in cisco AS5300 config :     isdn siwtch-type primary-net5     controller e1 0  framming crc4  linecode hdb3  pri-group 1-31        interface serial0:15  isdn switch-type primary-net5  isdn incomming-voice modem  isdn T310 6        dialpeer,.blah,...blahh        thanks for ur help     Dirgan   Send instant messages to your online friends http://asia.messenger.yahoo.com ___
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Re: [Asterisk-Users] Asterisk BRI in the USA

2006-04-12 Thread Leo Ann Boon






Next, I got a Eicon Diva board and tried to get the hisax kernel 
driver working. It's ni-1 implementation, the only one I could find, 
isn't very complete. It was written by a guy in Australia using only 
an isdn simulator, a significant accomplishment. It appears that it's 
intent was to just place outgoing data calls.  At best, it would 
signal my POTS line, but give up during call setup. Unfortunately, our 
layer 3 protocol is secret and the specs have to be purchased from 
Telcordia. The last time I checked, assuming I chose the right 
publication, it was about $600.


Why don't you try a Diva Server board and chan_capi?

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[Asterisk-Users] Asterisk 1.2.7 Released!

2006-04-12 Thread Asterisk Development Team

The Asterisk Development Team is pleased to announce the release of
Asterisk 1.2.7. This release include a number of important bug fixes,
including SIP presence (subscriptions) handling and MixMonitor call
recording, and users are encouraged to upgrade their systems when
possible. See the included ChangeLog file for more details on what has
been fixed.

The release is available on the Digium FTP servers as PGP signed
tarballs and also as PGP signed patch files, to ease upgrading from the
previous versions. The keys used to sign these files can be verified by
using the keyserver at pgp.mit.edu.

Thanks for your support of Asterisk!

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RE: [Asterisk-Users] Call Forward and AGI

2006-04-12 Thread Josh McAllister

I'm sure there is more than 1 way to do this, but the first thing that
comes to my mind is to set a channel variable with the exten # at the
top of your extensions macro. Then use that channel var instead of CLID.

Josh McAllister


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jon Farmer
Sent: Wednesday, April 12, 2006 1:32 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Call Forward and AGI

Hi

i have a agi script that gets called when a user wants to dialout
externally. it gets passed in the exten number and the number dialled
and looks up in a db to see if they are allowed to dial the number. the
problem is if someone forwards their phone to a external number the
CALLERIDNUM is the CLID of the calling party not the extension forwarded
thus the call is denied. Can anyone think of a way around this?
-- 
Jon Farmer
Telford, Shropshire
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Re: [Asterisk-Users] OT: Polycom IP501 Config file error - Error is 0x4020 (during autoboot...)

2006-04-12 Thread Jim Rice

On Wed, 2006-04-12 at 17:18 -0400, Andrew Kohlsmith wrote:
> How are we to know what you tried and didn't try?

I didn't think it necessary at the time.

Had I have documented the process and included config files and log
files and tcpdump traces, wouldn't I have received the TMI lecture
instead?

I did not ask for what you have tried.  Only if you had seen the error.

> When I used FTP provisioning for my ip501s this past weekend I ran into 
> several errors very similar to that one.  Pretty much every one of them was 
> due to screwups in the [macaddress].cfg file.
> 
> First, I tried 
...
> Next, I'd tried to be smart and 
...
> Next, I'd tried to be smart and
...
> After that, I couldn't figure out why 
...
> My point is that 
...
> No need to apologize, but 
...
> -A.

OK, now I throw up my hands.

Your answer then, is no.

Thank you.

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Re: [Asterisk-Users] Texas User Group

2006-04-12 Thread Linda Robertson

I will be out of the office until Wednesday, 09/21/2005. If you need
immediate assistance, please contact Phill Miller at 412.262.8503 or
[EMAIL PROTECTED] Thank you.

>>> asterisk-users 04/12/06 18:25 >>>

I setup www.txaug.net with a temporary page and have a working mailing
list
at list.txaug.net. Look forward to comments and suggestions.

JR, you definately have a great setup for user group meetings.
--
Bruce
Nortex Networks


On 4/12/06, JR Richardson <[EMAIL PROTECTED]> wrote:
>
> Hi,
> I'm in the Dallas area, my office is in Irving.  I would be willing to
> host a user group.  I have a decent size conference room, big
whiteboard and
> on-site lab with 5 asterisk servers dedicated for testing/development.
 I
> can host web space and e-mail list also.  I do like the wiki idea
though,
> everyone gets so much e-mail these days, it's hard to keep up. 
Currently
> I'm working on large scale asterisk clustering and in review to speak
at
> Astricon in Oct.
>
> Bruce, I'm really interested in your deployment.
>
> JR
>
>
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Re: [Asterisk-Users] Company List

That is a perfect name since it is a customer to this company.On 4/12/06, C F <[EMAIL PROTECTED]> wrote:
I know Halliburton is implementing Asterisk, their techincalrequirement for the proposal alone is 38 pages in French.
On 4/12/06, Aaron Daniel <[EMAIL PROTECTED]> wrote:> I know Digium has a few case studies on their website.>> 
http://www.digium.com/en/asteriskbusinesses/casestudies/>> Aaron>> On Wed, 12 Apr 2006, Bruce Reeves wrote:>> > That's exactly right, I had hoped that Digium had something like this, and
> > may at their booth at conferences. I just need something to prove> > credibility. If you haven't seen is the Forbes article> > 
http://www.forbes.com/free_forbes/2006/0410/063.html does some of that with> > the mention of 3 customers. I don't want any one's customer list, just> > references :)> >> > Bruce
> >> > On 4/12/06, Curt Shaffer <[EMAIL PROTECTED]> wrote:> >>> >> I disagree a bit. A lot of companies publish their "customer list" for
> >> reasons of advertisement. If I have a client that is joe blow fortune 500> >> company, I'm gonna publish that for my credibility. I think that is what> >> we> >> are looking for (I think I can safely speak for both of us on this).
> >>> >> Curt> >>> >>> >>> --> Aaron Daniel> Computer Systems Technician> Sam Houston State University> 
[EMAIL PROTECTED]> (936) 294-4198> ___> --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>___
--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:   
http://lists.digium.com/mailman/listinfo/asterisk-users-- BruceNortex Networks
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Re: [Asterisk-Users] Texas User Group

I setup www.txaug.net with a temporary page and have a working mailing list at list.txaug.net. Look forward to comments and suggestions.JR, you definately have a great setup for user group meetings. 
-- 
Bruce
Nortex NetworksOn 4/12/06, JR Richardson <[EMAIL PROTECTED]> wrote:
Hi,
I'm in the Dallas area, my office is in Irving.  I would be willing to host a user group.  I have a decent size conference room, big whiteboard and on-site lab with 5 asterisk servers dedicated for testing/development.  I can host web space and e-mail list also.  I do like the wiki idea though, everyone gets so much e-mail these days, it's hard to keep up.  Currently I'm working on large scale asterisk clustering and in review to speak at Astricon in Oct.

 
Bruce, I'm really interested in your deployment.
JR
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Re: [Asterisk-Users] Cisco 7960 won't dial (sccp)

2006-04-12 Thread Lacy Moore - Aspendora

Shawn,
 
What Sergio meant was you misspelled internal under [lines].  Not sure if it is that way in your file, of if it was mistyped here.
 
context     = from-sccp-intenal
 
That's listed under the lines, note the missing 'r'.
 
On 4/12/06, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
On Wed, Apr 12, 2006 at 09:32:12PM +0200, Sergio Chersovani wrote:> 
[EMAIL PROTECTED] ha scritto:> >context = from-sccp-intenal> >> I guess "intenal" is not the righe context :-)>> SergioThe from-sccp-internal is almost an exact copy of my from-sip-internal context,
which works fine[from-sccp-internal]include => local-extensionsinclude => always-out-potsinclude => local-calls-potsinclude => ld-callsexten => h,1,Hangupexten => i,1,Congestion
exten => i,2,Hangup[from-sip-internal]include => local-extensionsinclude => always-out-potsinclude => local-calls-potsinclude => ld-callsexten => h,1,Hangupexten => i,1,Congestion
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Re: [Asterisk-Users] OT: Polycom IP501 Config file error - Error is 0x4020 (during autoboot...)

2006-04-12 Thread Avi Miller


Jim Rice wrote:

I only asked this list as a last resort, having already exhausted many
other avenues.  I even mentioned that it was OT, but have seen numerous
postings for phones of all kinds.


A thought: I had similar problems with one phone of mine after I 
power-cycled it during the provisioning process. Its a known issue with 
the Polycoms that they can become.. confused.. if power-cycled while 
they're booting.


Have you tried booting the phone offline and formatting its filesystem 
via the Advanced menu?


cYa,
Avi

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Re: [Asterisk-Users] Company List

2006-04-12 Thread C F

I know Halliburton is implementing Asterisk, their techincal
requirement for the proposal alone is 38 pages in French.

On 4/12/06, Aaron Daniel <[EMAIL PROTECTED]> wrote:
> I know Digium has a few case studies on their website.
>
> http://www.digium.com/en/asteriskbusinesses/casestudies/
>
> Aaron
>
> On Wed, 12 Apr 2006, Bruce Reeves wrote:
>
> > That's exactly right, I had hoped that Digium had something like this, and
> > may at their booth at conferences. I just need something to prove
> > credibility. If you haven't seen is the Forbes article
> > http://www.forbes.com/free_forbes/2006/0410/063.html does some of that with
> > the mention of 3 customers. I don't want any one's customer list, just
> > references :)
> >
> > Bruce
> >
> > On 4/12/06, Curt Shaffer <[EMAIL PROTECTED]> wrote:
> >>
> >> I disagree a bit. A lot of companies publish their "customer list" for
> >> reasons of advertisement. If I have a client that is joe blow fortune 500
> >> company, I'm gonna publish that for my credibility. I think that is what
> >> we
> >> are looking for (I think I can safely speak for both of us on this).
> >>
> >> Curt
> >>
> >>
> >
>
> --
> Aaron Daniel
> Computer Systems Technician
> Sam Houston State University
> [EMAIL PROTECTED]
> (936) 294-4198
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Re: [Asterisk-Users] OT: Polycom IP501 Config file error - Error is 0x4020 (during autoboot...)

2006-04-12 Thread Andrew Kohlsmith

On Wednesday 12 April 2006 17:03, Jim Rice wrote:
> What, beyond RTFM, Google, list archives, Polycom support (yeah, right),
> and literally hours of trial and error?

How are we to know what you tried and didn't try?

> My intent was that if someone had seen this specific error, they would
> reply with a solution, or might then ask for more details.

When I used FTP provisioning for my ip501s this past weekend I ran into 
several errors very similar to that one.  Pretty much every one of them was 
due to screwups in the [macaddress].cfg file.

First, I tried just not having a bootrom.ld or sip.ld in the directory.  That 
didn't get me very far at all, as I'd assumed the ip501 would just continue 
with what it had if it couldn't find the code there.

Next, I'd tried to be smart and provide a directory for MISC_FILES, but that's 
for listing misc files, not a directory for them.

Next, I'd tried to be smart and include an ipmid.cfg in the list of 
CONFIG_FILES, but I didn't have one, and then it didn't like an empty one.

After that, I couldn't figure out why my settings weren't being seen.  Then I 
read the docs a little more carefully and noted that they say the FIRST file 
giving a config wins out, not the last one.  Oops.

My point is that while the Polycoms aren't known for their user-friendliness, 
they do write out pretty decent logs in the directory given by 
LOG_FILE_DIRECTORY, and their errors, while terse, do try to guide you in the 
right direction.

> My apologies for asking twice.  But thank you for your observation.

No need to apologize, but those of use who do like to help are often 
frustrated when those asking don't give enough (or any!) information to allow 
us to help.  Even now, I don't see any paste of the .cfg nor a 
[specific_mac_addy].cfg, nor any mention of a tcpdump trace to show us what 
the phone's actually requesting (and not getting).

We are here to help, but please... help us help you!

-A.
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Re: [Asterisk-Users] OT: Polycom IP501 Config file error - Error is 0x4020 (during autoboot...)

2006-04-12 Thread Jim Rice

On Wed, 2006-04-12 at 16:35 -0400, Andrew Kohlsmith wrote:
> On Wednesday 12 April 2006 16:06, Jim Rice wrote:
> > > Then it displays:
> > > Config file error
> > > Error is 0x4020
> 
> > I cannot be the only one with Polycom 501s to have seen this error?!
> 
> Surely when presented with a problem you don't throw your hands up in the air 
> and expect others to fix it.

Dear -A.,

I do not.  I was merely asking if anyone had seen this particular error.

> The phone is saying "config file error" -- do you not think it would be 
> prudent to examine your configuration files and ensure you don't have 
> something buggered up in them?  Perhaps your [macid].cfg file is specifying 
> an invalid file, or one of the files it is referring to is corrupt somehow?  
> Did you try pulling any changes out (reverting to the default fileset that 
> comes with the firmware update) and using strictly default config files?
> 
> Honestly, where are your troubleshooting skills?  If you aren't technically 
> proficient you may want to leave the configuration of technical devices to 
> someone else.

What, beyond RTFM, Google, list archives, Polycom support (yeah, right),
and literally hours of trial and error?

Yes.  I thought it prudent to check and recheck the config files.
Numerous times.  Different versions, default sets, again and again.
Trying new phones right out of the box.  Using known working files...

> That isn't a personal slam, it's an observation.  I don't know the first 
> thing 
> about fixing transmissions, but I don't try to fix it anyway and then throw 
> my hands up when I can't get the damn thing out of 1st gear anymore...  I 
> take it apart and try and verify that I'd done the repair correctly... I try 
> this several times, and I present my case to my transmission-specialist 
> friend, who after laughing at me, helps me fix it right.

I have been working with a specialist.  He is as stumped as I am.
That's why the shotgun approach.  Get the word out.  Just maybe...

> How is this any different?  Your post didn't present any information, didn't 
> say whether you're using default configs, default configs you've "barely 
> adjusted" or completely custom configs.  It doesn't say what you've tried and 
> what the results were.
> 
> Honestly, how are we to help?
> 
> -A.

I only asked this list as a last resort, having already exhausted many
other avenues.  I even mentioned that it was OT, but have seen numerous
postings for phones of all kinds.

My intent was that if someone had seen this specific error, they would
reply with a solution, or might then ask for more details.

This list is usually quick to respond, and with useful info.

My apologies for asking twice.  But thank you for your observation.

Jim

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RE: [Asterisk-Users] Asterisk BRI in the USA

I was reading that Junghanns was palnning on supporting National (Q.931)
if they do, all you should need is a NT1 to turn the U insterface in an
S/T.

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Walt Reed
> Sent: Wednesday, April 12, 2006 4:39 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Asterisk BRI in the USA
> 
> On Wed, Apr 12, 2006 at 09:10:09AM -1000, Mark Coccimiglio said:
> > I guess what I need to find out first if there is anyone out there 
> > using Asterisk & BRI in the USA?  If so what hardware have 
> they been 
> > able to use.  I no longer want to hack around with analog 
> circuits.  
> > BRI has the potential of PRI with only 2 B channels.  A 
> great idea for 
> > a small office such as my own.  VoIP may be an option, but I would 
> > need a ITSP that would allow calls to transfer from my 
> asterisk box to 
> > the remote phone set.  My link to the internet is fast, but its 
> > pointless to route a call into the office just to stream it 
> back out.  
> > More work more work more work.
> 
> I'm in a similar situation. Being on the end of a long loop, 
> POTS sucks
> - echo / static / crappy calling features.
> 
> Paying around $2K-3K for BRI solution is a non-starter 
> though. It needs to get down to the $200-400 / port level 
> (more ports = cheaper per
> port) to be viable. Soho / Very small business (under 12 
> people) is definately a 1-2 port market which my guess would 
> be the bulk of sales for BRI.
> 
> It would be awesome to see a Sangoma BRI card. It's hard to 
> say what the market would be since the US telco companies 
> have really tried to kill BRI service.
> 
> Considering what a full PRI costs, there is also a point 
> where too many BRI ports no longer makes sense, but that 
> number is probably >4-6 BRI's. I was in a situation where I 
> really only wanted 4 BRI's, but had to look at a PRI instead 
> which ended up wasting a lot of money in the long run. POTS 
> was a non-option.
> 
> 
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Re: [Asterisk-Users] Company List

2006-04-12 Thread Aaron Daniel


I know Digium has a few case studies on their website.

http://www.digium.com/en/asteriskbusinesses/casestudies/

Aaron

On Wed, 12 Apr 2006, Bruce Reeves wrote:


That's exactly right, I had hoped that Digium had something like this, and
may at their booth at conferences. I just need something to prove
credibility. If you haven't seen is the Forbes article
http://www.forbes.com/free_forbes/2006/0410/063.html does some of that with
the mention of 3 customers. I don't want any one's customer list, just
references :)

Bruce

On 4/12/06, Curt Shaffer <[EMAIL PROTECTED]> wrote:


I disagree a bit. A lot of companies publish their "customer list" for
reasons of advertisement. If I have a client that is joe blow fortune 500
company, I'm gonna publish that for my credibility. I think that is what
we
are looking for (I think I can safely speak for both of us on this).

Curt






--
Aaron Daniel
Computer Systems Technician
Sam Houston State University
[EMAIL PROTECTED]
(936) 294-4198
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Re: [Asterisk-Users] Asterisk BRI in the USA

2006-04-12 Thread Walt Reed

On Wed, Apr 12, 2006 at 09:10:09AM -1000, Mark Coccimiglio said:
> I guess what I need to find out first if there is anyone out there using
> Asterisk & BRI in the USA?  If so what hardware have they been able to
> use.  I no longer want to hack around with analog circuits.  BRI has the
> potential of PRI with only 2 B channels.  A great idea for a small
> office such as my own.  VoIP may be an option, but I would need a ITSP
> that would allow calls to transfer from my asterisk box to the remote
> phone set.  My link to the internet is fast, but its pointless to route
> a call into the office just to stream it back out.  More work more work
> more work.

I'm in a similar situation. Being on the end of a long loop, POTS sucks
- echo / static / crappy calling features.

Paying around $2K-3K for BRI solution is a non-starter though. It needs
to get down to the $200-400 / port level (more ports = cheaper per
port) to be viable. Soho / Very small business (under 12 people) is
definately a 1-2 port market which my guess would be the bulk of sales
for BRI.

It would be awesome to see a Sangoma BRI card. It's hard to say what the
market would be since the US telco companies have really tried to kill
BRI service.

Considering what a full PRI costs, there is also a point where too
many BRI ports no longer makes sense, but that number is probably >4-6
BRI's. I was in a situation where I really only wanted 4 BRI's, but had
to look at a PRI instead which ended up wasting a lot of money in the
long run. POTS was a non-option.


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Re: [Asterisk-Users] CAPI Installation Eicon Diva Server

2006-04-12 Thread Avi Miller


[EMAIL PROTECTED] wrote:
Asterisk says it has 30 capi channels available, but my mistake may be 
in configuring the trunks...


When I was debugging my Eicon Diva 4-BRI board, I found it useful to 
play with extensions_custom.conf (in AMP) just to ensure I got the 
Custom Dial String absolutely correct. According to the latest 
chan_capi-cm, the Dial String should be:


CAPI///

Where:

 = Contr1 or g1 (Controller or Group ID)
 = Phone number
 = Things like B or b for Early B3 and other things. I have 'b' 
in my options, but I do admit that I have no idea what early B3 is. :)


Hope that helps in some way,
Avi

P.S. I wrote a quick config page for the 4-BRI for freePBX here: 
http://aussievoip.com/wiki/index.php?page=freePBX-EiconDiva


It might have a few things to consider as well.

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Re: [Asterisk-Users] OT: Polycom IP501 Config file error - Error is 0x4020 (during autoboot...)

2006-04-12 Thread Andrew Kohlsmith

On Wednesday 12 April 2006 16:06, Jim Rice wrote:
> > Then it displays:
> > Config file error
> > Error is 0x4020

> I cannot be the only one with Polycom 501s to have seen this error?!

Surely when presented with a problem you don't throw your hands up in the air 
and expect others to fix it.

The phone is saying "config file error" -- do you not think it would be 
prudent to examine your configuration files and ensure you don't have 
something buggered up in them?  Perhaps your [macid].cfg file is specifying 
an invalid file, or one of the files it is referring to is corrupt somehow?  
Did you try pulling any changes out (reverting to the default fileset that 
comes with the firmware update) and using strictly default config files?

Honestly, where are your troubleshooting skills?  If you aren't technically 
proficient you may want to leave the configuration of technical devices to 
someone else.

That isn't a personal slam, it's an observation.  I don't know the first thing 
about fixing transmissions, but I don't try to fix it anyway and then throw 
my hands up when I can't get the damn thing out of 1st gear anymore...  I 
take it apart and try and verify that I'd done the repair correctly... I try 
this several times, and I present my case to my transmission-specialist 
friend, who after laughing at me, helps me fix it right.

How is this any different?  Your post didn't present any information, didn't 
say whether you're using default configs, default configs you've "barely 
adjusted" or completely custom configs.  It doesn't say what you've tried and 
what the results were.

Honestly, how are we to help?

-A.
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Re: [Asterisk-Users] Company List

That's exactly right, I had hoped that Digium had something like this, and may at their booth at conferences. I just need something to prove credibility. If you haven't seen is the Forbes article 
http://www.forbes.com/free_forbes/2006/0410/063.html does some of that with the mention of 3 customers. I don't want any one's customer list, just references :)BruceOn 4/12/06, 
Curt Shaffer <[EMAIL PROTECTED]> wrote:
I disagree a bit. A lot of companies publish their "customer list" forreasons of advertisement. If I have a client that is joe blow fortune 500company, I'm gonna publish that for my credibility. I think that is what we
are looking for (I think I can safely speak for both of us on this).Curt
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RE: [Asterisk-Users] Company List

2006-04-12 Thread Curt Shaffer

I disagree a bit. A lot of companies publish their "customer list" for
reasons of advertisement. If I have a client that is joe blow fortune 500
company, I'm gonna publish that for my credibility. I think that is what we
are looking for (I think I can safely speak for both of us on this).

Curt

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson
Sent: Wednesday, April 12, 2006 4:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Company List

I would doubt that anyone is going to share their "customer list" for
obvious reasons. I'd have to guess that in access of 80% of the 
production implementations are sold by resellers (of various sizes), and 
maybe 20% are actual in-house implementations by those that frequent 
this list. The 80% is probably what you'd be interested in, but not 
likely to be published anywhere.

Curt Shaffer wrote:
> I have not but if you find one, please pass it on because I have the 
> same requirement.
> 
>  
> 
> Curt
> 
>  
> 
> 
> 
> *From:* [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] *On Behalf Of *Bruce
Reeves
> *Sent:* Wednesday, April 12, 2006 3:51 PM
> *To:* asterisk-users@lists.digium.com
> *Subject:* [Asterisk-Users] Company List
> 
>  
> 
> The question was raised by a CFO who is looking at Asterisk if there is 
> a list of companies using Asterisk. I have not found one yet, has anyone 
> seen anything like this I can give him.
> 
> -- 
> Bruce
> Nortex Networks
> 
> 
> 
> 
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Re: [Asterisk-Users] Cisco 7960 won't dial (sccp)

2006-04-12 Thread shawnl

On Wed, Apr 12, 2006 at 09:32:12PM +0200, Sergio Chersovani wrote:
> [EMAIL PROTECTED] ha scritto:
> >context = from-sccp-intenal
> >  
> I guess "intenal" is not the righe context :-)
> 
> Sergio

The from-sccp-internal is almost an exact copy of my from-sip-internal context,
which works fine



[from-sccp-internal]
include => local-extensions
include => always-out-pots
include => local-calls-pots
include => ld-calls
exten => h,1,Hangup
exten => i,1,Congestion
exten => i,2,Hangup

[from-sip-internal]
include => local-extensions
include => always-out-pots
include => local-calls-pots
include => ld-calls
exten => h,1,Hangup
exten => i,1,Congestion
exten => i,2,Hangup



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Re: [Asterisk-Users] Company List

2006-04-12 Thread Rich Adamson


I would doubt that anyone is going to share their "customer list" for
obvious reasons. I'd have to guess that in access of 80% of the 
production implementations are sold by resellers (of various sizes), and 
maybe 20% are actual in-house implementations by those that frequent 
this list. The 80% is probably what you'd be interested in, but not 
likely to be published anywhere.



Curt Shaffer wrote:
I have not but if you find one, please pass it on because I have the 
same requirement.


 


Curt

 




*From:* [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] *On Behalf Of *Bruce Reeves

*Sent:* Wednesday, April 12, 2006 3:51 PM
*To:* asterisk-users@lists.digium.com
*Subject:* [Asterisk-Users] Company List

 

The question was raised by a CFO who is looking at Asterisk if there is 
a list of companies using Asterisk. I have not found one yet, has anyone 
seen anything like this I can give him.


--
Bruce
Nortex Networks




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Re: [Asterisk-Users] OT: Polycom IP501 Config file error - Error is 0x4020 (during autoboot...)

2006-04-12 Thread Jim Rice

On Tue, 2006-04-11 at 04:56 -0700, Jim Rice wrote:
> Using FTP to configure 501.
> Gets past "Running...  App = sip.ld"
> and:  Welcome!  Processing configuration...
> "This may take a few seconds."
> 
> Then it displays:
> 
> Config file error
> 
> Error is 0x4020
> 
> and reboots continuously, repeating the above.
> 
> Anyone seen this before?
> 
> Is this a corrupt *.ld file?
> An FTP error?  (PASV)?
> A missing SETUP setting?
> 
> Thanks!
> 

Anybody?

I cannot be the only one with Polycom 501s to have seen this error?!

-- 
Jim Rice
by Design Publishing
11626 N. Tracey Road
Hayden, Idaho  83835

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Re: [Asterisk-Users] Company List

2006-04-12 Thread Matt Roth


>> Bruce wrote:
>>
>>
>> The question was raised by a CFO who is looking at Asterisk if there 
is a list of
>> companies using Asterisk. I have not found one yet, has anyone seen 
anything like

>> this I can give him.
>
> Curt Shaffer wrote:
>
> I have not but if you find one, please pass it on because I have the 
same requirement.


If you don't get a good response here, you might have better luck on the 
biz list.


Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer
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[Asterisk-Users] Texas User Group

2006-04-12 Thread JR Richardson

Hi,
I'm in the Dallas area, my office is in Irving.  I would be willing to host a user group.  I have a decent size conference room, big whiteboard and on-site lab with 5 asterisk servers dedicated for testing/development.  I can host web space and e-mail list also.  I do like the wiki idea though, everyone gets so much e-mail these days, it's hard to keep up.  Currently I'm working on large scale asterisk clustering and in review to speak at Astricon in Oct.

 
Bruce, I'm really interested in your deployment.
JR
> > Message: 16> Date: Wed, 12 Apr 2006 10:33:37 -0500 (CDT)> From: Aaron Daniel <[EMAIL PROTECTED]>> Subject: Re: [Asterisk-Users] Texas User Group
> To: Asterisk Users Mailing List - Non-Commercial Discussion>  > Message-ID: <
[EMAIL PROTECTED]>> Content-Type: TEXT/PLAIN; charset=US-ASCII; format=flowed> > That may be the best idea.  Unfortunately we're such a huge state that > it's going to be pretty hard to get everyone in the same room unless 
> there's some big event going on.  Astricon may be a good time to get > together in person though.> > As for the site, a simple wiki may be best, and if everyone wants a forum > (personally prefer mailing lists, easier to filter through, but that's 
> just me) that'd be nifty as well.  Perhaps when this gets started, we may > find more users in the state and do mini-sessions in different parts of > the state.> > Aaron> > On Wed, 12 Apr 2006, Bruce Reeves wrote:
> > > It sounds like what might be best is a Texas User group, since most of us> > are spread out across our great state. With Astircon 2006 coming to Dallas> > this year, we could all probably get together at that time. Mainly I would
> > like to see a user group in Texas because I am deploying a wide spread> > asterisk setup in several cities across the state and Oklahoma and> > Louisiana. It would be nice to know some possible local asterisk contacts.
> >> > I am willing to setup some space on my website or a new domain for a user> > group if a state wide group sounds good. Any suggestions as to features for> > the site? Wiki? Forums? ??
> >> > Bruce> > Nortex Networks-- JR RichardsonEngineering for the Masses 
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RE: [Asterisk-Users] Company List

2006-04-12 Thread Curt Shaffer









I have not but if you find one, please
pass it on because I have the same requirement.

 

Curt

 









From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bruce Reeves
Sent: Wednesday, April 12, 2006
3:51 PM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Company
List



 

The question was raised by a CFO who is looking at Asterisk if there is
a list of companies using Asterisk. I have not found one yet, has anyone seen
anything like this I can give him.

-- 
Bruce
Nortex Networks 






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[Asterisk-Users] Playback sound file while on-line

2006-04-12 Thread Andre Courchesne - Consultant


Hi,

  I am looking for a way to play a sound file (wav, gsm or whatever) 
while a SIP client (extension) is on-line with a Zap channel. Ideally 
both ends would hear the sound file.


  Any hints or pointers appreciated.

Andre
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[Asterisk-Users] Company List

The question was raised by a CFO who is looking at Asterisk if there is a list of companies using Asterisk. I have not found one yet, has anyone seen anything like this I can give him.-- BruceNortex Networks
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Re: [Asterisk-Users] SIP MWI

2006-04-12 Thread Olle E Johansson



12 apr 2006 kl. 21.30 skrev David Gomillion:

If it's already been covered, please forgive the repetition.  I  
searched

Mantis, but couldn't come up with anything.

We upgraded to Asterisk 1.2.6, and suddenly the Polycom MWI stopped
working on SP IP 300s and 600s.  All of them.

I tried splitting the friend entries in sip.conf into user and  
peer.  I
made sure the context of the voicemail box was on the end of the  
mailbox

option in the sip.conf file.  I checked and rechecked the config files
for the phones.

Nothing worked to restore the MWI's until I reverted to 1.2.5.  Then
everything just worked like it should.

Has anyone else seen this?  Is there an open bug, or a fix already
merged into svn?


I haven't seen this. My polycom chirps now and then.

Try getting a SIP debug of the messaging and see if the polycom
returns an error message when Asterisk sends a NOTIFY.

/O
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Re: [Asterisk-Users] Recording queue transfers

2006-04-12 Thread lenz

Hello,
you can use the veryu same technique to turn recording on in any context -
what matters is that you bring along the uniqueid of the original call, so
you know how to match the different recordings you may find on your hard
disk!

In data Wed, 12 Apr 2006 19:24:56 +0200, Maximiliano J. Goldsmid
<[EMAIL PROTECTED]> ha scritto:

Regarding this article (1) I have one question to make. What can I do to
record the call if the agent makes a transfer using the "flash" button
instead of "transfer button" or using blindxfer or atxfer defined in
features. conf

If the agent makes the transfer with "flash", the comunication between
the
person who is calling and is already in the queue and the target person
who

receive the call doesn't get recorded.

e.g.

Client/ Costumer (P1), contact the Call Center and he is assisted by an
agent (P2), (P2) transfers the call to his supervisor (P3) by pressing
"flash" plus "extension # of his supervisor".

The comunication between P1 and P3 doesn't get recorded.

What can I do to get this recorded?

[1]
http://www.oinko.net/astrecipes/index.php?from=1&q=astrecipes/recording+queue+transfers+to+disk

Maxi
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--
Assum est, versa et manduca.
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Re: [Asterisk-Users] SIP conections, with RTP not going trough Asterisk

2006-04-12 Thread Olle E Johansson



12 apr 2006 kl. 14.58 skrev Ronald Wiplinger:


Tiago Stein D`Agostini wrote:

Hi,

  Ie been looking for some time how to use asterisk  to initiate  
SIP connections between 2 IP phones,  but afetr initiated the  
communication making the RTP go directly from one telephone to the  
other, without passing by asterisk. Unfortunately I found no  
explanations of how to do it.


Does anyone care to give a pointer to any explanation about how to  
do it?



canreinvite=yes
and look at the options for dial()


Thanks in advance

Actually, it's the default mode. Just connect your phones to Asterisk  
on the same LAN, and Asterisk will

get out of the media path.

/Olle


---
* Olle E. Johansson - [EMAIL PROTECTED]
* Asterisk Training http://edvina.net/training/



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[Asterisk-Users] Call Forward and AGI

2006-04-12 Thread Jon Farmer

Hi

i have a agi script that gets called when a user wants to dialout
externally. it gets passed in the exten number and the number dialled
and looks up in a db to see if they are allowed to dial the number. the
problem is if someone forwards their phone to a external number the
CALLERIDNUM is the CLID of the calling party not the extension forwarded
thus the call is denied. Can anyone think of a way around this?
-- 
Jon Farmer
Telford, Shropshire
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Re: [Asterisk-Users] Cisco 7960 won't dial (sccp)

2006-04-12 Thread Sergio Chersovani


[EMAIL PROTECTED] ha scritto:

context = from-sccp-intenal
  

I guess "intenal" is not the righe context :-)

Sergio
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[Asterisk-Users] SIP MWI

2006-04-12 Thread David Gomillion

If it's already been covered, please forgive the repetition.  I searched
Mantis, but couldn't come up with anything.  

We upgraded to Asterisk 1.2.6, and suddenly the Polycom MWI stopped
working on SP IP 300s and 600s.  All of them.

I tried splitting the friend entries in sip.conf into user and peer.  I
made sure the context of the voicemail box was on the end of the mailbox
option in the sip.conf file.  I checked and rechecked the config files
for the phones.

Nothing worked to restore the MWI's until I reverted to 1.2.5.  Then
everything just worked like it should.

Has anyone else seen this?  Is there an open bug, or a fix already
merged into svn?

Thanks,
David Gomillion


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Re: [Asterisk-Users] Polycom VLANs

2006-04-12 Thread Jerry Jones


Use VLANs on Ploys all the time, but manually set also.

Of course switches and routers all need to be setup for the proper  
vlan config also.



On Apr 12, 2006, at 12:18 PM, BJ Weschke wrote:


On 4/12/06, Rob Terhaar <[EMAIL PROTECTED]> wrote:
So has anyone had any experience working with the polycom 501 or  
301 and

vlans?

We run dell managed switches here, so we don't have the luxury of  
running
CDP to force the VOIP vlan. I haven't been able to get the polycom  
phones to
talk on a manually set vlan. I have some junky sipura phones that  
work
fine-(get dhcp, register to asterisk etc) when i manually set them  
to vlan4.


Any advice you guys have would be greatly appreciated!



 Yes. Works fine. You need to make sure there the VLAN ID in the phone
matches the VLAN ID you've got set in your PowerConnect switches and
you should be good to go. Well, that, and the fact that the VLAN for
the phone should be added to that port as a tagged port. :)

--
Bird's The Word Technologies, Inc.
http://www.btwtech.com/
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Re: [Asterisk-Users] Asterisk BRI in the USA

2006-04-12 Thread Mark Coccimiglio


I guess what I need to find out first if there is anyone out there using
Asterisk & BRI in the USA?  If so what hardware have they been able to
use.  I no longer want to hack around with analog circuits.  BRI has the
potential of PRI with only 2 B channels.  A great idea for a small
office such as my own.  VoIP may be an option, but I would need a ITSP
that would allow calls to transfer from my asterisk box to the remote
phone set.  My link to the internet is fast, but its pointless to route
a call into the office just to stream it back out.  More work more work
more work.

Mark Coccimiglio
n3whx @amsat.org
sip:[EMAIL PROTECTED]

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Re: [Asterisk-Users] URL in Queue App / Determining the DID/Queue at Agent's Phone

2006-04-12 Thread Steve Feinstein


Kyle,
That's bloody brilliant

Thanks so much!

-Steve Feinstein
GatherWorks, Inc.

Kyle Sexton wrote:

Have you tried something like:

exten => 2,1,SetCIDName(QUEUENAME: ${CALLERIDNAME})
exten => 2,n,Queue(QUEUENAME)





On 4/12/06, * Steve Feinstein* <[EMAIL PROTECTED] 
> wrote:


Thanks!, I will definitely take a look at that.  We were hoping not to
have to do AGI in the client, but if we have to, we have to.  It'll
probably be useful for other things down the road.

-Steve Feinstein
GatherWorks Inc.

BJ Weschke wrote:
> On 4/12/06, Steve Feinstein <[EMAIL PROTECTED]
> wrote:
>
>> I'd like for our custom soft phone to be able to know what
queue, and/or
>> what DID is calling an Agent's phone before the agent picks
up.  The
>> agent is using the AGENTCALLBACKLOGIN.  One agent can be in
multiple
>> queues so it would be nice if they could get a pop up window
telling
>> them who's on the line before they pick up and hear the
announcement
>> telling them that.  I'd like to lose the announcment all together.
>>
>> It seems like that the phone can easily know what extension was
dialed
>> to make it ring, but at best that's the phone client's
extension (The
>> server dialed it via the Local/ interface), and at worst it's
's'.  Is
>> there anyway I can know the DID of the person who called into
the Queue?
>>
>> I've done ethereal traces and it seems like the DID, that actually
>> called the agent/phone is no where to be found.
>> I've tried also to use the URL string in the Queue()
application, but
>> the server doesn't seem to send it.  (I've also tried having
the client
>> send a URL, and it seems to get sent, yet the server doesn't
seem to
>> forward it.  It seems to just get lost).
>>
>> Has anyone gotten the URL in the Queue application to
work?  And if it
>> does, it it delivered to the phone before, or after the phone
answers?
>>
>> Any hacks,tips,tricks,pointers, would be most appreciated.
>>
>>
>
> http://bugs.digium.com/view.php?id=6843
>
>  Here's code to fire off an AGI to do pretty much anything you
need to
> do on the calling channel after a Queue Member has been assigned to
> it.
>
> --
> Bird's The Word Technologies, Inc.
> http://www.btwtech.com/
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begin:vcard
fn:Steve Feinstein
n:Feinstein;Steve
org:GatherWorks Inc.
adr:;;27 Dupaw Gould Road;Brookline;NH;03033;USA
email;internet:[EMAIL PROTECTED]
tel;work:+1 (603) 672-1472
x-mozilla-html:TRUE
url:http://www.gatherworks.com/
version:2.1
end:vcard

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Re: [Asterisk-Users] URL in Queue App / Determining the DID/Queue at Agent's Phone

2006-04-12 Thread Kyle Sexton

Have you tried something like:exten => 2,1,SetCIDName(QUEUENAME: ${CALLERIDNAME})exten => 2,n,Queue(QUEUENAME)On 4/12/06, 
Steve Feinstein <[EMAIL PROTECTED]> wrote:
Thanks!, I will definitely take a look at that.  We were hoping not tohave to do AGI in the client, but if we have to, we have to.  It'llprobably be useful for other things down the road.-Steve Feinstein
GatherWorks Inc.BJ Weschke wrote:> On 4/12/06, Steve Feinstein <[EMAIL PROTECTED]> wrote:>>> I'd like for our custom soft phone to be able to know what queue, and/or
>> what DID is calling an Agent's phone before the agent picks up.  The>> agent is using the AGENTCALLBACKLOGIN.  One agent can be in multiple>> queues so it would be nice if they could get a pop up window telling
>> them who's on the line before they pick up and hear the announcement>> telling them that.  I'd like to lose the announcment all together. It seems like that the phone can easily know what extension was dialed
>> to make it ring, but at best that's the phone client's extension (The>> server dialed it via the Local/ interface), and at worst it's 's'.  Is>> there anyway I can know the DID of the person who called into the Queue?
 I've done ethereal traces and it seems like the DID, that actually>> called the agent/phone is no where to be found.>> I've tried also to use the URL string in the Queue() application, but
>> the server doesn't seem to send it.  (I've also tried having the client>> send a URL, and it seems to get sent, yet the server doesn't seem to>> forward it.  It seems to just get lost).
 Has anyone gotten the URL in the Queue application to work?  And if it>> does, it it delivered to the phone before, or after the phone answers? Any hacks,tips,tricks,pointers, would be most appreciated.
>> http://bugs.digium.com/view.php?id=6843>>  Here's code to fire off an AGI to do pretty much anything you need to
> do on the calling channel after a Queue Member has been assigned to> it.>> --> Bird's The Word Technologies, Inc.> http://www.btwtech.com/
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[Asterisk-Users] DUNDi with SIP

2006-04-12 Thread Adam Robins

Anyone out there have a functional DUNDi configuration using SIP for the
inter-Asterisk transport?  I've gotten it to work with IAX2, but if I
change it to SIP it does not pass the call over even though it knows
where to send it.  Thanks.

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Re: [Asterisk-Users] free video (soft) phone available?

2006-04-12 Thread Mojo with Horan & Company, LLC

We use Neos from neosmt.com to connect to our interoffice jabber server 
and I noticed recently that it can do video and audio via a h.323 
gatekeeper.  Haven't tried it out yet but you might.


Ronald Wiplinger wrote:
I am using eyebeam and I am happy with it. However, it is boring just to 
talk to my son in the other room.
Whenever I try to convince somebody to buy eyebeam, they are scared of 
the price.


Is there a free video soft phone available, that will work with eyebeam 
/ asterisk?



bye

Ronald Wiplinger
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--
Mojo <[EMAIL PROTECTED]>
Office Manger, Horan & Company, LLC
(907) 747- x112
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RE: [Asterisk-Users] playback soundfile stored in mysql database

2006-04-12 Thread Akpome Akpoguma

.want to playback a "raw" binary file without writing into an 
intermediate file which would increase latency




From: "Alexander Lopez" <[EMAIL PROTECTED]>
Reply-To: Asterisk Users Mailing List - Non-Commercial 
Discussion
To: "Asterisk Users Mailing List - Non-Commercial 
Discussion"

Subject: RE: [Asterisk-Users] playback soundfile stored in mysql database
Date: Wed, 12 Apr 2006 13:17:13 -0400

Look at using EAGI.

>
> Hi Guys,
>
> I want to playback a sound file stored in mysql database in
> my perl scriptpls can anyone help with an idea?
> response would be greatly appreciated
>
> Rgds
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_
Express yourself instantly with MSN Messenger! Download today - it's FREE! 
http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/


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[Asterisk-Users] Callback Agents and Dial 'g' option

2006-04-12 Thread Johann

I'm unable to get the Dial option 'g' to work with callback agents.  The plan is 
to use it so that I can redirect a customer to a menu so they can rate the call 
they just had with the agent.  However, when the agent hangs up the call does 
not continue in the dialplan.


I login with the agent.  Call joins the queue.  The agent and call get 
connected.  The agent hangs up and the call should continue to the 
Playback(beep) and the Noop(), however the call is hung up on both sides.


Extensions.conf:
[default]
; Handle login and logout
exten => ,1,Agentcallbacklogin(1,,[EMAIL PROTECTED])
exten => ,1,AgentCallbackLogin(1,s)

; join the queue
exten => ,1,Answer
exten => ,2,Queue(testing)

[queue]
exten => 1,1,Dial(Sip/4000||got)
exten => 1,2,Playback(beep)
exten => 1,3,Noop(Jump to the QA menu now)

Any ideas?


--johann
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Re: [Asterisk-Users] * 1.2.4 & 1.2.6: "Ringing" anamoly

2006-04-12 Thread Chris Shaw


Ronald Lewis wrote:
I was alerted the other day by of all people, my mom, that she wasn't 
hearing a "ring" when she dialed my number. Puzzled, I tried calling 
myself. The call connects, but there's dead silence until voicemail 
picks up. Calling internally, extensions worked perfectly. So, I 
figured, "another damned Broadvoice issue."
 
For kicks, I upgraded to 1.2.6 today, and the end result is the same. 
So, I went to the dialplan playground, and removed a few lines for 
testing. It turns out that if I playback a file before ringing an 
extension, ringing works fine. Without, dead silence.
 
Any ideas?
  
Just out of curiosity did you happen to put an Answer() before playing 
audio or ringing? I use BroadVoice also and I used to have the exact 
same problem but putting Answer() as the first step in the context 
before playing my menu solved the problem.



   -Chris

--
Chris Shaw
IT Manager
Precision Pump, Inc
150 N Main St
Banks, OR 97106

Phone: 503-324-2361
Fax: 503-324-2203
E-Mail: [EMAIL PROTECTED]

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[Asterisk-Users] Cisco 7960 won't dial (sccp)

2006-04-12 Thread shawnl

I'm trying to setup a couple of Cisco 7960's in asterisk.  I have asterisk
working fine for sip clients, and can call the 7960's just fine, but
I can't seem to dial out on them.  

As soon as I enter the first digit, the phone attempts to dial it without
waiting for the rest.  I've changed timeout settings, etc but can't seem to
get it to work.  Any ideas?

Asterisk SVN-trunk-r7498
chan_sccp-20060207

[general]
servername = asterisk
keepalive = 60
debug = 10
context = from-sccp-internal
dateFormat = M/D/YA
port = 2000
disallow=all
allow=alaw
allow=ulaw
allow=g723
firstdigittimeout = 60
digittimeout = 8 
autoanswer_ring_time = 0
autoanswer_tone = 0x32
remotehangup_tone = 0x32
transfer_tone = 0
callwaiting_tone = 0x2d
musicclass=default
language=en
rtptos = 184
echocancel = on
silencesuppression = off
trustphoneip = no
tos = 0x68


[devices]

type = 7960
autologin = 2002
description = phone2002
dtmfmode = inband  
imageversiom = P00307020200
dnd = on
trustphoneip = no
speeddial = 2000
private = on 
device => SEP00036BC3852B


[lines]

id  = 2002  ; future use
pin = 1234  ; future use
label   = 2002  ; button line label 
description = Line 2002 ; top diplay description
context = from-sccp-intenal
incominglimit = 2
transfer = on
mailbox = 1001
vmnum = 2999
cid_name = Phone2002; caller id name
cid_num = 2002
trnsfvm = 1000
secondary_dialtone_digits = 9
secondary_dialtone_tone = 0x22  ; outside dialtone
musicclass=default
language=en
rtptos = 18
echocancel = on
silencesuppression = off
line => 2002


extensions.conf

[from-sccp-internal]
include => local-extensions
include => always-out-pots
include => local-calls-pots
include => ld-calls
exten => h,1,Hangup
exten => i,1,Congestion
exten => i,2,Hangup

[always-out-pots]
exten => _9XXX.,1,Dial(Zap/1/${EXTEN}:1)
exten => _9XXX.,2,Goto(102)
exten => _9XXX.,102,Congestion
exten => _9XXX.,103,Hangup

[local-extensions]
exten => 2002,1,Dial(SCCP/2002)
exten => 2002,2,Voicemail(u2002)
exten => 2002,102,Voicemail(b2002)
exten => 2002,103,Hangup



asterisk*CLI> 
-- SEP00036BC3852B: New call on line 2002
-- SEP00036BC3852B: New call on line 2002
-- SEP00036BC3852B: Cisco Digit: 0009 (9) on line 2002
-- SEP00036BC3852B: Cisco Digit: 0009 (9) on line 2002
-- SEP00036BC3852B: Ending call 1 on line 2002
-- SCCP: Asterisk request to hangup Outbound channel SCCP/2002-0001
-- SEP00036BC3852B: Ending call 1 on line 2002
-- SCCP: Asterisk request to hangup Outbound channel SCCP/2002-0001


1
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RE: [Asterisk-Users] call center running Asterisk-sound quality-critical!

Yes. That's is the one. It is resolved now. 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tamas
Sent: Wednesday, April 12, 2006 1:26 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] call center running Asterisk-sound
quality-critical!


Wai Wu wrote:
> Except that mixmonitor still has a bug in it. 
>   
What kind of bug? Issue number?
FYI: yesterday one issue has been fixed :D
http://bugs.digium.com/view.php?id=6457
Did you mean that type of bug? If something else, please let us know...

T.

> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Kevin P.
> Fleming
> Sent: Wednesday, April 12, 2006 11:45 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] call center running Asterisk -sound 
> quality-critical!
>
> Matt Roth wrote:
>
>   
>> These statements seem contradictory.  I know of no way (short of a 
>> custom patch) to tell Monitor() to mix the in and out legs prior to 
>> writing them to disk.  On the other hand, MixMonitor() does just that

>> and I believe it also buffers the writes in a way that circumvents 
>> the
>> 
>
>   
>> I/O bottleneck associated with Monitor().
>> 
>
> Both of these statements are correct.
> __
>   
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Re: [Asterisk-Users] call center running Asterisk -sound quality-critical!

2006-04-12 Thread Matt Roth

>>> Matt Roth wrote:
>>>
>>> These statements seem contradictory.  I know of no way (short of a
>>> custom patch) to tell Monitor() to mix the in and out legs prior to
>>> writing them to disk.  On the other hand, MixMonitor() does just that
>>> and I believe it also buffers the writes in a way that circumvents the
>>> I/O bottleneck associated with Monitor().
>>
>> Kevin P. Fleming wrote:
>>
>> Both of these statements are correct.
>
> Wai Wu wrote:
>
> Except that mixmonitor still has a bug in it.

Wai,

Please explain how "the in and out channels are mixed first before they 
are written to the disk" using "monitor with no mixing onto the scsi 
drive."  I'd love to implement this on our system to cut in half the I/O 
associated with Monitor().

Also, what bug does MixMonitor() have?  It is my understanding that 
MixMonitor() is based on ChanSpy() and we seem to be having an issue 
with ChanSpy() where the legs of a call fall out of synch.  My hunch is 
that it has to do with a caller being muted or placed on hold.  Do these 
issues seem related?

Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer
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[Asterisk-Users] Config with TE210P, Asterisk and Legacy PBX and FreePBX?

2006-04-12 Thread Remco Barende


Hi list!

Has anyone ever tried the following installation :

I want to replace our legacy PBX with Asterisk but... I still need the legacy 
PBX as a 'channel bank' for fax (I need E1 not T1)


I will put a dual port PRI card in the Asterisk box, and for incoming and 
outgoing faxes I want to use native bridging on the TE210P and route fax calls 
(based on DID and prefix when dialling) to / from the legacy PBX.


I guess I do not need to modify anything in the PBX (Alcatel Novo 
Supreme) because I can simply use dialling prefixes to catch outbound 
calls.


Does anyone have example config files how to implement this config?

This would be the setup :

PRI -> Asterisk <--> Legaxy PBX on TE210P
 |-> SIP phones

Would it be possible to use FreePBX to setup such routing (inbound and 
outbound), if anyone could guide me in the basic direction for this I 
would be most grateful.


Thanks!!
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Re: [Asterisk-Users] Macro-hangupcall - has a Wait(5) - [EMAIL PROTECTED] --- why?

2006-04-12 Thread BJ Weschke

On 4/12/06, Marco Mouta <[EMAIL PROTECTED]> wrote:
> [macro-hangupcall]
> exten => s,1,ResetCDR(w)
> exten => s,2,NoCDR()
> exten => s,3,Wait(5)
> exten => s,4,Hangup
>
>
> Hi all, currently i've been getting troubles with SIpphone Sjphone and Xlite
> seems also to get delay but no crash on hanging.
>
> I found that [EMAIL PROTECTED] is executing this Wait(5) and it seems to me 
> that
> Sjphone is giving timeout error because of it...
>
> Why is this 5 seconnds? any one knows?
>


 You may want to pose that question to an Asterisk @ Home forum.

--
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Re: [Asterisk-Users] call center running Asterisk -sound quality-critical!

2006-04-12 Thread Tamas


Wai Wu wrote:
> Except that mixmonitor still has a bug in it. 
>   
What kind of bug? Issue number?
FYI: yesterday one issue has been fixed :D
http://bugs.digium.com/view.php?id=6457
Did you mean that type of bug? If something else, please let us know...

T.

> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Kevin P.
> Fleming
> Sent: Wednesday, April 12, 2006 11:45 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] call center running Asterisk -sound
> quality-critical!
>
> Matt Roth wrote:
>
>   
>> These statements seem contradictory.  I know of no way (short of a 
>> custom patch) to tell Monitor() to mix the in and out legs prior to 
>> writing them to disk.  On the other hand, MixMonitor() does just that 
>> and I believe it also buffers the writes in a way that circumvents the
>> 
>
>   
>> I/O bottleneck associated with Monitor().
>> 
>
> Both of these statements are correct.
> __
>   
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[Asterisk-Users] Recording queue transfers

2006-04-12 Thread Maximiliano J. Goldsmid

Regarding this article (1) I have one question to make. What can I do to
record the call if the agent makes a transfer using the "flash" button
instead of "transfer button" or using blindxfer or atxfer defined in
features. conf

If the agent makes the transfer with "flash", the comunication between the
person who is calling and is already in the queue and the target person who
receive the call doesn't get recorded.

e.g.

Client/ Costumer (P1), contact the Call Center and he is assisted by an
agent (P2), (P2) transfers the call to his supervisor (P3) by pressing
"flash" plus "extension # of his supervisor".

The comunication between P1 and P3 doesn't get recorded.

What can I do to get this recorded?


[1] 
http://www.oinko.net/astrecipes/index.php?from=1&q=astrecipes/recording+queue+transfers+to+disk

Maxi
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Re: [Asterisk-Users] call center running Asterisk -sound quality-critical!

2006-04-12 Thread BJ Weschke

On 4/12/06, Wai Wu <[EMAIL PROTECTED]> wrote:
> Except that mixmonitor still has a bug in it.
>

 Had. Corrected yesterday.

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Re: [Asterisk-Users] Polycom VLANs

2006-04-12 Thread BJ Weschke

On 4/12/06, Rob Terhaar <[EMAIL PROTECTED]> wrote:
> So has anyone had any experience working with the polycom 501 or 301 and
> vlans?
>
> We run dell managed switches here, so we don't have the luxury of running
> CDP to force the VOIP vlan. I haven't been able to get the polycom phones to
> talk on a manually set vlan. I have some junky sipura phones that work
> fine-(get dhcp, register to asterisk etc) when i manually set them to vlan4.
>
> Any advice you guys have would be greatly appreciated!
>

 Yes. Works fine. You need to make sure there the VLAN ID in the phone
matches the VLAN ID you've got set in your PowerConnect switches and
you should be good to go. Well, that, and the fact that the VLAN for
the phone should be added to that port as a tagged port. :)

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RE: [Asterisk-Users] playback soundfile stored in mysql database

Look at using EAGI.
 
> 
> Hi Guys,
> 
> I want to playback a sound file stored in mysql database in 
> my perl scriptpls can anyone help with an idea? 
> response would be greatly appreciated
> 
> Rgds
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RE: [Asterisk-Users] call center running Asterisk -sound quality-critical!

Except that mixmonitor still has a bug in it. 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin P.
Fleming
Sent: Wednesday, April 12, 2006 11:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] call center running Asterisk -sound
quality-critical!

Matt Roth wrote:

> These statements seem contradictory.  I know of no way (short of a 
> custom patch) to tell Monitor() to mix the in and out legs prior to 
> writing them to disk.  On the other hand, MixMonitor() does just that 
> and I believe it also buffers the writes in a way that circumvents the

> I/O bottleneck associated with Monitor().

Both of these statements are correct.
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[Asterisk-Users] playback soundfile stored in mysql database

2006-04-12 Thread Akpome Akpoguma


Hi Guys,

I want to playback a sound file stored in mysql database in my perl 
scriptpls can anyone help with an idea? response would be 
greatly appreciated


Rgds

_
Express yourself instantly with MSN Messenger! Download today it's FREE! 
http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/


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[Asterisk-Users] Polycom VLANs

2006-04-12 Thread Rob Terhaar

So has anyone had any experience working with the polycom 501 or 301 and vlans? We run dell managed switches here, so we don't have the luxury of running CDP to force the VOIP vlan. I haven't been able to get the polycom phones to talk on a manually set vlan. I have some junky sipura phones that work fine-(get dhcp, register to asterisk etc) when i manually set them to vlan4.
Any advice you guys have would be greatly appreciated!
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RE: [Asterisk-Users] Bandwidth Management

Brought over from -users, Please reply to the -dev list.

I agree, lets move the discusstion over to that list as it has to be discussed 
there. After we reach an accord on how it should be done we will open up a 
issue on Mantis.

I see this as being two distinctive parts that would need to be tied together:

First:  We need to make the selection of CODECS technology agnostic, There 
currently exist a facility for CODEC selection (SIP_CODEC) in the sip channel 
but not in others.

Second: Discuss is this sould be an outside application that is called from 
within Asterisk or if it should become a function 
Set(CODEC=${OPTIMALCODEC(quality)})
available options could be:

quality
bandwidth
license 



Any comments.

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of Wai Wu
> Sent: Wednesday, April 12, 2006 10:51 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] Bandwidth Management
> 
> I think this belongs to the development mail-list. 
> 
> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Jean-Michel Hiver
> Sent: Wednesday, April 12, 2006 12:05 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Bandwidth Management
> 
> Andy Tan a écrit :
> 
> >Hi Alex,
> >
> >thanks for the suggestion.
> >
> >Did some checks, and thought that I could set a global variable to 
> >track the utilized bandwidth.
> >
> >Wish that there are plans for support to include variables like 
> >SIP_CODEC in other protocols.
> >  
> >
> Actually this sounds like a really nice idea. It would be 
> cool to have a way to start using less intensive bandwith 
> codecs for new calls when bandwith reaches a certain threshold.
> 
> For example:
> 
> - 0-40% bandwith: g711
> - 40-60% bandwith: g729
> - 60%-80% bandwith: g723
> - 80%-100% bandwith: drop new calls, or maybe use lpc10
> 
> It wouldn't help in SOHO usage but when using Asterisk as a 
> call termination gateway, it would help making the most out 
> of available bandwith. g711 is certainly better than g729 
> when you have the bandwith, and i'm pretty sure that even 
> lpc10 sounds better when on non-saturated bandwith compared 
> with g729 with some packet loss...
> 
> How would you go about implementing this?
> 
> Cheers,
> Jean-Michel.
> 
> --
> Jean-Michel Hiver - http://ykoz.net/
> Découvrez la Réunion des Technologies IP & Telecom
> TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE
> 
> 
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RE: [Asterisk-Users] Setting Codecs on the Fly

Simply check out the READMEs in asterisk/doc/ in your source directory.


> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Julio Arruda
> Sent: Wednesday, April 12, 2006 12:42 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Setting Codecs on the Fly
> 
> Douglas Garstang wrote:
> > Does anyone know if it's possible to set the codecs for a 
> number via an Asterisk command?
> > 
> > Ie, yes you can set the codecs in sip.conf for a user, but 
> I'd like to have a command that can set the same thing so 
> that it can be done without having to change sip.conf.
> > 
> > Essentially I want the user to be able to prefix a code to 
> their dialled number to set their preferred codec for that call.
> > 
> > Possible?
> 
> Humm..I wonder if what google returned for:
> 
> "asterisk set codec on a call"
> 
> http://www.voip-info.org/wiki-Asterisk+variables
> 
> Would help...Seeems that in fact, google is my friend:
> 
> "${SIP_CODEC}: Used to set the SIP codec for a call"
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RE: [Asterisk-Users] Setting Codecs on the Fly

2006-04-12 Thread Douglas Garstang

Ahhh a variable. I was looking for a command. Thanks, I'll try it out.

> -Original Message-
> From: Julio Arruda [mailto:[EMAIL PROTECTED]
> Sent: Wednesday, April 12, 2006 10:42 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Setting Codecs on the Fly
> 
> 
> Douglas Garstang wrote:
> > Does anyone know if it's possible to set the codecs for a 
> number via an Asterisk command?
> > 
> > Ie, yes you can set the codecs in sip.conf for a user, but 
> I'd like to have a command that can set the same thing so 
> that it can be done without having to change sip.conf.
> > 
> > Essentially I want the user to be able to prefix a code to 
> their dialled number to set their preferred codec for that call.
> > 
> > Possible?
> 
> Humm..I wonder if what google returned for:
> 
> "asterisk set codec on a call"
> 
> http://www.voip-info.org/wiki-Asterisk+variables
> 
> Would help...Seeems that in fact, google is my friend:
> 
> "${SIP_CODEC}: Used to set the SIP codec for a call"
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Re: [Asterisk-Users] Setting Codecs on the Fly

2006-04-12 Thread Julio Arruda


Douglas Garstang wrote:

Does anyone know if it's possible to set the codecs for a number via an 
Asterisk command?

Ie, yes you can set the codecs in sip.conf for a user, but I'd like to have a 
command that can set the same thing so that it can be done without having to 
change sip.conf.

Essentially I want the user to be able to prefix a code to their dialled number 
to set their preferred codec for that call.

Possible?


Humm..I wonder if what google returned for:

"asterisk set codec on a call"

http://www.voip-info.org/wiki-Asterisk+variables

Would help...Seeems that in fact, google is my friend:

"${SIP_CODEC}: Used to set the SIP codec for a call"
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[Asterisk-Users] * 1.2.4 & 1.2.6: "Ringing" anamoly

2006-04-12 Thread Ronald Lewis

I was alerted the other day by of all people, my mom, that she wasn't hearing a "ring" when she dialed my number. Puzzled, I tried calling myself. The call connects, but there's dead silence until voicemail picks up. Calling internally, extensions worked perfectly. So, I figured, "another damned Broadvoice issue."

 
For kicks, I upgraded to 1.2.6 today, and the end result is the same. So, I went to the dialplan playground, and removed a few lines for testing. It turns out that if I playback a file before ringing an extension, ringing works fine. Without, dead silence.

 
Any ideas?
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Re: [Asterisk-Users] call center running Asterisk - sound quality-critical!

2006-04-12 Thread Kevin P. Fleming

Tamas wrote:

> Kevin, does MixMonitor have buffering? How big is the buffer? Is it
> possible to change the size? I guess, we are talking about buffering
> voice samples and writing only a bulk of them to disk (e.g. in every 50
> packets - 1second).

It buffers the data in memory, there is no fixed size. It _will_ attempt
to write out a mixed audio frame each time a matching pair of frames has
been received from both sides; changing that to only write after a
certain amount of data has been received would not be a significant
task. There is a risk of data loss, though, if you do that... but at
least MixMonitor does not sit in the channel read/write path like
Monitor does, so delays in writing the audio don't impact the audio
being bridged across the channels.
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Re: [Asterisk-Users] update - 512 Simultaneous Calls with Digital Recording

2006-04-12 Thread Waldo Rubinstein


Hey Henri,

Long time no talk. How far have you been able to scale oreka up to?  
How many simultaneous calls have you been able to record and under  
what hardware config?


Thanks,
Waldo

On Apr 12, 2006, at 11:12 AM, Henri Herscher wrote:


Another solution would be to use a dedicated recording server sniffing
RTP and signalling packets in the media path using software such as
http://www.oreka.org. Oreka automatically mixes both legs of an RTP
conversation to disk and GSM encodes the result in a separate thread
so that capture always has priority.

Cheers
Henri


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Re: [Asterisk-Users] call center running Asterisk - sound quality-critical!

2006-04-12 Thread Tamas

Kevin P. Fleming wrote:
> Matt Roth wrote:
>
>   
>> These statements seem contradictory.  I know of no way (short of a
>> custom patch) to tell Monitor() to mix the in and out legs prior to
>> writing them to disk.  On the other hand, MixMonitor() does just that
>> and I believe it also buffers the writes in a way that circumvents the
>> I/O bottleneck associated with Monitor().
>> 
>
> Both of these statements are correct
It seems, MixMonitor is usable again (since yesterday's svn commit) so
it can save at least saving of one channel. We would test now MixMonitor
for this reason.

Kevin, does MixMonitor have buffering? How big is the buffer? Is it
possible to change the size? I guess, we are talking about buffering
voice samples and writing only a bulk of them to disk (e.g. in every 50
packets - 1second).
If there is no such buffer, do you think implementing it can be a real
solution? Storing into RAM needs too big RAM which can be a problem
(e.g. if we want to use monitor() for storing 60 concurrent calls for
min. 15-20 minutes).

Regards,
T.

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Re: [Asterisk-Users] TE410P upgrade to TE411P => (solution to) no more fax carrier detection !

2006-04-12 Thread Kevin P. Fleming

Rob Lith wrote:

> Kevin - if you stop it from tone detection with 'vpmdtmfsupport=0' will it
> detect the fax cgn?

Yes, that was the point of my message; with that setting, the software
tone detector will be used, just as it was before the OP's VPM got
installed.
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Re: [Asterisk-Users] TE410P upgrade to TE411P => (solution to) no more fax carrier detection !

2006-04-12 Thread Rob Lith

Kevin - if you stop it from tone detection with 'vpmdtmfsupport=0' will it detect the fax cgn?RegardsRobOn 12/04/06, Kevin P. Fleming <
[EMAIL PROTECTED]> wrote:[EMAIL PROTECTED]
 wrote:> I changed from a TE410P to a TE411P and fax carriers weren't detected> anymore !> I have tried everything (recompile zaptel+asterisk+spandsp ;> echocancel=yes/no ; rxgain=0/txgain=0 ; increase Wait() ; ...), nothing
> worked.> The only solution that worked for me was to install and use NVFaxDetect.For the moment, if you need FAX tone detection, you will need to use'vpmdtmfsupport=0' in your module configuration for loading the wct4xxp
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Re: [Asterisk-Users] iax2 show netstats

2006-04-12 Thread Benchev

> i've been using iax2 show netstats and i wonder if someone could explain
> what all these means, just in case i have them wrong.  Because i am looking
> for something that tells me that there is delay , and/or packet loss.
>
> LOCAL -  
> REMOTE  Channel  RTT  Jit  Del  Lost   % 
> Drop  OOO  Kpkts  Jit  Del  Lost   %  Drop  OOO  Kpkts IAX2/iaxBBG-16384  
> 1000   -10-1  -1 0   -1  000 0   0 00  
>0 IAX2/iaxBBG-16386 16   -10-1  -1 0   -1  10  
> 40 0   0 00  0
The new Jitterbuffer in Asterisk

Steve Kann
..
5) Testing and monitoring:
--
You can test the effectiveness of PLC and the new jitterbuffer's detection of 
loss by using 
the new CLI command "iax2 test losspct ".  This will simulate n percent 
packet loss 
coming _in_ to chan_iax2. You should find that with PLC and the new JB, 10 
percent packet 
loss should lead to just a tiny amount of distortion, while without PLC, it 
would lead to 
silent gaps in your audio.

"iax2 show netstats" shows you statistics for each iax2 call you have up.  
The columns are "RTT" which is the round-trip time for the last PING, and then 
a bunch of s
tats for both the local side (what you're receiving), and the remote side 
(what the other 
end is telling us they are seeing).  The remote stats may not be complete if 
the remote 
end isn't using the new jitterbuffer.

The stats shown are:
* Jit: The jitter we have measured (milliseconds)
* Del: The maximum delay imposed by the jitterbuffer (milliseconds)
* Lost: The number of packets we've detected as lost.
* %: The percentage of packets we've detected as lost recently.
* Drop: The number of packets we've purposely dropped (to lower latency).
* OOO: The number of packets we've received out-of-order
* Kpkts: The number of packets we've received / 1000.
...

Benchev
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Re: [Asterisk-Users] call center running Asterisk - sound quality-critical!

2006-04-12 Thread Kevin P. Fleming

Matt Roth wrote:

> These statements seem contradictory.  I know of no way (short of a
> custom patch) to tell Monitor() to mix the in and out legs prior to
> writing them to disk.  On the other hand, MixMonitor() does just that
> and I believe it also buffers the writes in a way that circumvents the
> I/O bottleneck associated with Monitor().

Both of these statements are correct.
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[Asterisk-Users] Setting Codecs on the Fly

2006-04-12 Thread Douglas Garstang

Does anyone know if it's possible to set the codecs for a number via an 
Asterisk command?

Ie, yes you can set the codecs in sip.conf for a user, but I'd like to have a 
command that can set the same thing so that it can be done without having to 
change sip.conf.

Essentially I want the user to be able to prefix a code to their dialled number 
to set their preferred codec for that call.

Possible?

Doug.
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[Asterisk-Users] CAPI Installation Eicon Diva Server

2006-04-12 Thread nkohl




Hi 

 
I've got a 
dell 2550 with an Eicon Diva server PRI card plugged into it. I can call out 
using the acopy2 test utility.
 
I'm having 
trouble with asterisk making calls however... my capi.conf and modules.conf looks correct by the wiki instructions - does anyone have any advice on where to 
look ? I can attach conf files etc. if needed.
 
Asterisk 
says it has 30 capi channels available, but my mistake may be in configuring the 
trunks... 
 
Nick

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Re: [Asterisk-Users] Texas User Group

2006-04-12 Thread Aaron Daniel

That may be the best idea.  Unfortunately we're such a huge state that 
it's going to be pretty hard to get everyone in the same room unless 
there's some big event going on.  Astricon may be a good time to get 
together in person though.


As for the site, a simple wiki may be best, and if everyone wants a forum 
(personally prefer mailing lists, easier to filter through, but that's 
just me) that'd be nifty as well.  Perhaps when this gets started, we may 
find more users in the state and do mini-sessions in different parts of 
the state.


Aaron

On Wed, 12 Apr 2006, Bruce Reeves wrote:


It sounds like what might be best is a Texas User group, since most of us
are spread out across our great state. With Astircon 2006 coming to Dallas
this year, we could all probably get together at that time. Mainly I would
like to see a user group in Texas because I am deploying a wide spread
asterisk setup in several cities across the state and Oklahoma and
Louisiana. It would be nice to know some possible local asterisk contacts.

I am willing to setup some space on my website or a new domain for a user
group if a state wide group sounds good. Any suggestions as to features for
the site? Wiki? Forums? ??

Bruce
Nortex Networks

On 4/12/06, Greg Camp <[EMAIL PROTECTED] > wrote:


 I'm in Lubbock.  A little closer to Amarillo than Dallas.



Thanks,
Greg


  --





--
Aaron Daniel
Computer Systems Technician
Sam Houston State University
[EMAIL PROTECTED]
(936) 294-4198
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Re: [Asterisk-Users] call center running Asterisk - sound quality-critical!

2006-04-12 Thread Matt Roth

Wai Wu wrote:

> You got to be kidding about 53 calls being recorded at sametime is an
> issue. I have done at least twice as many on my dual xeon 3.4Ghz system
> and had no problem as clients like to record every call that goes
> through the system.

Nope.  We took our system to MCI's development lab and ran it against an 
Abacus 5000.  Things fell apart on the 64 call test.  We looked at the 
logs and saw a massive amount of disk I/O, so we set up a RAM disk to 
write the recordings to.  We were then able to successfully test up to 
512 simultaneous calls.

Looking at this list and the wiki, you'll see that many other users ran 
into the same issue at around 60 simultaneous recordings via Monitor().

Tamas wrote:

> how do you record calls? Monitor app. or MixMonitor or something else?

Wai Wu wrote:

> Then again, in my system, the in and out channels are mixed first before
> they are written to the disk.
> Just good old monitor with no mixing onto the scsi drive.

These statements seem contradictory.  I know of no way (short of a 
custom patch) to tell Monitor() to mix the in and out legs prior to 
writing them to disk.  On the other hand, MixMonitor() does just that 
and I believe it also buffers the writes in a way that circumvents the 
I/O bottleneck associated with Monitor().

Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer
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Re: [Asterisk-Users] Performance: Xeon or Opteron?

2006-04-12 Thread John Novack




Rich Adamson wrote:



While talking with one of the sangoma folks very recently, he was 
rather emphatic the pci bus was designed to "share" interrupts. I was 
a little concerned as a test server had the wanpipe driver sharing an 
interrupt with libata and uhc1_hcd. His comment was "that's the way 
its suppose to work, sharing interrupts as needed". I've not had any 
recognizable issues with the A200D card at all, and faxing via a A200D 
fxs port to a A200D fxo (pstn) port functions 100% reliably.


What that would suggest is the TDM400 pci firmware (whether on card 
logic or whatever) is the source of at least part of the TDM400 shared 
interrupt issue. I don't have any digium T1/E1 cards laying around, 
but if memory serves correctly, the T1/E1 cards do not use the same 
pci controller chip. That would suggest the T1/E1 cards are less of an 
issue then with the TDM400 card.


That's good to know, but considering the response from Digium on the 
TDM400 ( try another motherboard) when there didn't seem to even be an 
int. sharing issue, the card just couldn't be seen at all , and the 
support I received from Sangoma on a recent FXS issue that was resolved 
within a few days, I would tend to go with Sangoma for the T1 card, if 
and when I have the need.


John Novack

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Re: [Asterisk-Users] call center running Asterisk - sound quality - critical!

2006-04-12 Thread Henri Herscher

If you don't want to worry about * handling the full recording of all
traffic, you can potentially do this on a separate server on the RTP
path using http://www.oreka.org.

Cheers
Henri

On 10/04/06, Dov Bigio <[EMAIL PROTECTED]> wrote:
>
> Hi,
>
> I am using Asterisk for a call center on a Dual Xeon machine..
>
> I currently have
>
> 109 active channels
> 53 active calls
>
> Every body is complaining about quality and cpu is around 80% idle.
>
> Is there any tuning I can do???
>
> Besides that, Asterisk normally goes down once or twice per day...
>
> Thank you
>
> Dov
>
>
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Re: [Asterisk-Users] URL in Queue App / Determining the DID/Queue at Agent's Phone

2006-04-12 Thread Steve Feinstein

Thanks!, I will definitely take a look at that.  We were hoping not to 
have to do AGI in the client, but if we have to, we have to.  It'll 
probably be useful for other things down the road.


-Steve Feinstein
GatherWorks Inc.

BJ Weschke wrote:

On 4/12/06, Steve Feinstein <[EMAIL PROTECTED]> wrote:
  

I'd like for our custom soft phone to be able to know what queue, and/or
what DID is calling an Agent's phone before the agent picks up.  The
agent is using the AGENTCALLBACKLOGIN.  One agent can be in multiple
queues so it would be nice if they could get a pop up window telling
them who's on the line before they pick up and hear the announcement
telling them that.  I'd like to lose the announcment all together.

It seems like that the phone can easily know what extension was dialed
to make it ring, but at best that's the phone client's extension (The
server dialed it via the Local/ interface), and at worst it's 's'.  Is
there anyway I can know the DID of the person who called into the Queue?

I've done ethereal traces and it seems like the DID, that actually
called the agent/phone is no where to be found.
I've tried also to use the URL string in the Queue() application, but
the server doesn't seem to send it.  (I've also tried having the client
send a URL, and it seems to get sent, yet the server doesn't seem to
forward it.  It seems to just get lost).

Has anyone gotten the URL in the Queue application to work?  And if it
does, it it delivered to the phone before, or after the phone answers?

Any hacks,tips,tricks,pointers, would be most appreciated.




http://bugs.digium.com/view.php?id=6843

 Here's code to fire off an AGI to do pretty much anything you need to
do on the calling channel after a Queue Member has been assigned to
it.

--
Bird's The Word Technologies, Inc.
http://www.btwtech.com/
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begin:vcard
fn:Steve Feinstein
n:Feinstein;Steve
org:GatherWorks Inc.
adr:;;27 Dupaw Gould Road;Brookline;NH;03033;USA
email;internet:[EMAIL PROTECTED]
tel;work:+1 (603) 672-1472
x-mozilla-html:TRUE
url:http://www.gatherworks.com/
version:2.1
end:vcard

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Re: [Asterisk-Users] update - 512 Simultaneous Calls with Digital Recording

2006-04-12 Thread Henri Herscher

Another solution would be to use a dedicated recording server sniffing
RTP and signalling packets in the media path using software such as
http://www.oreka.org. Oreka automatically mixes both legs of an RTP
conversation to disk and GSM encodes the result in a separate thread
so that capture always has priority.

Cheers
Henri

On 05/04/06, Isaac Xiao <[EMAIL PROTECTED]> wrote:
> Matthew, thanks for your feedback and advice.
> > what I actually experienced was the complete breakdown of Asterisk at
> > around 60 concurrent recordings without it (the reality).
>
> The drive for saving your voice recordings is the same as your OS
> (Asterisk)? What do you think that save the voice recordings to a
> dedicated drive rather than the one which Asterisk program (OS) locates?
> I also think about using GSM format (Monitor(gsm,${CALLFILENAME}, mb))
> rather than WAV, PCM. In this case, it will use more CPU, but I/O of
> hard disk is reduced dramatically as you mentioned that it is I/O
> bottleneck issue, not CPU (In my case, I want to use P4 Dual core CPU or
> extreme edition). In order to reduce the CPU usage, we can have two leg
> files mixed after peak time.
>
> Matt mentioned about fragmented free space. I googled about Linux
> defragment topic. People always talk about that Linux doesn't need to
> defragment, it can handle it by itself very well. Not sure how true it
> is.
>
> I am looking a solution to record expanding simultaneous calls in the
> future in a call centre which accepts calls from our global branches. If
> I find the good solution, I definitely post it to the community.
>
> Cheers,
> Isaac Xiao
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Re: [Asterisk-Users] Performance: Xeon or Opteron?

2006-04-12 Thread Rich Adamson


Kristian Kielhofner wrote:

Rich Adamson wrote:
Yep, there is a lot of chatter about how hardware "x" performs with 
Asterisk and while I/O is the primary mover, most designs today will 
handle the modest Asterisk install easily.   I've got a site where 
they use 6 lines and 15 users on a 500Mhz CPU w/512MB RAM and boot 
off a 2GB flash disk.


VERY modest and absolutely dominates that particular install.

Only in the larger installs will hardware be an issue, but even then 
it doesn't take much hardware (from a server perspective) to handle 
a LOT of Asterisk traffic.


RandyW

The worst problem will be older hardware that doesn't play well with 
Digium cards. The TDM400 is the one I have some experience with, and 
even motherboards that are PCI 2.2 don't always see the TDM400

The Sangoma A200 seems more forgiving.
I have to wonder if the T1/E1 cards suffer  in a similar manner?



While talking with one of the sangoma folks very recently, he was 
rather emphatic the pci bus was designed to "share" interrupts. I was 
a little concerned as a test server had the wanpipe driver sharing an 
interrupt with libata and uhc1_hcd. His comment was "that's the way 
its suppose to work, sharing interrupts as needed". I've not had any 
recognizable issues with the A200D card at all, and faxing via a A200D 
fxs port to a A200D fxo (pstn) port functions 100% reliably.


What that would suggest is the TDM400 pci firmware (whether on card 
logic or whatever) is the source of at least part of the TDM400 shared 
interrupt issue. I don't have any digium T1/E1 cards laying around, 
but if memory serves correctly, the T1/E1 cards do not use the same 
pci controller chip. That would suggest the T1/E1 cards are less of an 
issue then with the TDM400 card.




The single port T1/E1 card (te110p) and the TDM400 both use the TigerJet 
320.


I guess they both would have the same issues then. ;)

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[Asterisk-Users] Newbie MOH and call transfer question

2006-04-12 Thread kevin ling

 
Hi,

I use the AAH2.7 (asterisk version 1.2.5). When someone call me and I pickup
the phone. If I want to transfer to another extension. Then I dial the "#"
key the system will play the onhold music. After I dial the extension
number. The system stop play onhold music and play ringtone. Is it possiable
keep play onhold music until someone pickup the phone? Appreciate any input.
Thanks.

Kevin


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RE: [Asterisk-Users] call center running Asterisk - sound quality-critical!

Just good old monitor with no mixing onto the scsi drive. 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tamas
Sent: Wednesday, April 12, 2006 4:55 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] call center running Asterisk - sound
quality-critical!

Hi,

how do you record calls? Monitor app. or MixMonitor or something else?
How does your storage backend looks like?
What kind of channels do you use? Do you record IAX2 channels?
Regards,
 Tamas

Wai Wu wrote:
> You got to be kidding about 53 calls being recorded at sametime is an
issue. I have done at least twice as many on my dual xeon 3.4Ghz system
and had no problem as clients like to record every call that goes
through the system. Then again, in my system, the in and out channels
are mixed first before they are written to the disk.
>
> 
>
> From: [EMAIL PROTECTED] on behalf of Matt Roth
> Sent: Tue 4/11/2006 5:25 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] call center running Asterisk - sound
quality- critical!
>
>
>
>  >>On 4/10/06, Dov Bigio <[EMAIL PROTECTED]> wrote:
>  >>
>  >>Hi,
>  >>
>  >>I am using Asterisk for a call center on a Dual Xeon machine..
>  >>
>  >>I currently have
>  >>
>  >>109 active channels
>  >>53 active calls
>  >>
>  >>Every body is complaining about quality and cpu is around 80% idle.
>  >>
>  >>Is there any tuning I can do???
>  >>
>  >>Besides that, Asterisk normally goes down once or twice per day...
>  >>
>  >>Thank you
>  >>
>  >>Dov
>  >>
>  > C F wrote:
>  >
>  >From what you say it sounds that the problem is not with asteisk, 
> but  >the way it's configured. Asterisk should *never* go down that
often.
>  >Asterisk as a normal PBX should run without a restart for as long as

> >there is power to the box, in the case of a call center if I would  
> >hear of a restart once a week I would accept it, but still would look

> >for ways of improving it beyond that.
>  >
>  >You complain about call quality, what type of phones are thes? What

> >codec? are they all local?
>  >
>
> Dov,
>
> I agree with the first response.  Your system is failing at an 
> abnormal rate.  Please share more information about your setup so that

> we can help you.  Hardware, software, OS, configuration...there's no 
> such thing as too many details when trying to work out these problems 
> via a mailing list.
>
> Information about what tasks you are asking Asterisk to perform and 
> how you have it configured to do so is vital.  In particular, I'm 
> curious to know if you're recording the calls using the Monitor() 
> application?  53 concurrent calls being recorded directly to disk is 
> about where things start to go south (it's an I/O bottleneck, not 
> CPU).  If you have a Digium card in the box, make sure that it's not 
> sharing an interrupt with any other hardware.  The list and the wiki 
> both have plenty of information to help you with that.
>
> Matthew Roth
> InterMedia Marketing Solutions
> Software Engineer and Systems Developer
>   
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RE: [Asterisk-Users] SipXPhone

2006-04-12 Thread Greg Camp









Mark,

 

I could not get SipXPhone working
either.  We've been using this SDK and really like it: http://www.worksoutsoftware.com/

 

The pricing is seems decent as well.

 



Thanks,
Greg



 











From: Mark Hayward [mailto:[EMAIL PROTECTED] 
Sent: Wednesday, April 12, 2006
3:21 AM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: [Asterisk-Users]
SipXPhone



 

Has anybody managed to
get SipXPhone working with asterisk? I just cannot get it to work. It just
keeps reporting an authentication failure even though all the details seem
correct. The same settings work fine in X-Lite. Failing that, are there any
opensource or reasonably priced SIP SDKs that people can recommend?




I have the log from the SipXPhone. It says authentication required, yet it is
definitely using the correct user/pass. 

Thanks,










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RE: [Asterisk-Users] Bandwidth Management

I think this belongs to the development mail-list. 

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jean-Michel Hiver
Sent: Wednesday, April 12, 2006 12:05 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Bandwidth Management

Andy Tan a écrit :

>Hi Alex,
>
>thanks for the suggestion.
>
>Did some checks, and thought that I could set a global variable to 
>track the utilized bandwidth.
>
>Wish that there are plans for support to include variables like 
>SIP_CODEC in other protocols.
>  
>
Actually this sounds like a really nice idea. It would be cool to have a way to 
start using less intensive bandwith codecs for new calls when bandwith reaches 
a certain threshold.

For example:

- 0-40% bandwith: g711
- 40-60% bandwith: g729
- 60%-80% bandwith: g723
- 80%-100% bandwith: drop new calls, or maybe use lpc10

It wouldn't help in SOHO usage but when using Asterisk as a call termination 
gateway, it would help making the most out of available bandwith. g711 is 
certainly better than g729 when you have the bandwith, and i'm pretty sure that 
even lpc10 sounds better when on non-saturated bandwith compared with g729 with 
some packet loss...

How would you go about implementing this?

Cheers,
Jean-Michel.

--
Jean-Michel Hiver - http://ykoz.net/
Découvrez la Réunion des Technologies IP & Telecom
TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE


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Re: [Asterisk-Users] Performance: Xeon or Opteron?

2006-04-12 Thread Kristian Kielhofner


Rich Adamson wrote:
Yep, there is a lot of chatter about how hardware "x" performs with 
Asterisk and while I/O is the primary mover, most designs today will 
handle the modest Asterisk install easily.   I've got a site where 
they use 6 lines and 15 users on a 500Mhz CPU w/512MB RAM and boot 
off a 2GB flash disk.


VERY modest and absolutely dominates that particular install.

Only in the larger installs will hardware be an issue, but even then 
it doesn't take much hardware (from a server perspective) to handle a 
LOT of Asterisk traffic.


RandyW

The worst problem will be older hardware that doesn't play well with 
Digium cards. The TDM400 is the one I have some experience with, and 
even motherboards that are PCI 2.2 don't always see the TDM400

The Sangoma A200 seems more forgiving.
I have to wonder if the T1/E1 cards suffer  in a similar manner?



While talking with one of the sangoma folks very recently, he was rather 
emphatic the pci bus was designed to "share" interrupts. I was a little 
concerned as a test server had the wanpipe driver sharing an interrupt 
with libata and uhc1_hcd. His comment was "that's the way its suppose to 
work, sharing interrupts as needed". I've not had any recognizable 
issues with the A200D card at all, and faxing via a A200D fxs port to a 
A200D fxo (pstn) port functions 100% reliably.


What that would suggest is the TDM400 pci firmware (whether on card 
logic or whatever) is the source of at least part of the TDM400 shared 
interrupt issue. I don't have any digium T1/E1 cards laying around, but 
if memory serves correctly, the T1/E1 cards do not use the same pci 
controller chip. That would suggest the T1/E1 cards are less of an issue 
then with the TDM400 card.




The single port T1/E1 card (te110p) and the TDM400 both use the TigerJet 
320.


--
Kristian Kielhofner

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Re: [Asterisk-Users] Performance: Xeon or Opteron?

2006-04-12 Thread Rich Adamson

Yep, there is a lot of chatter about how hardware "x" performs with 
Asterisk and while I/O is the primary mover, most designs today will 
handle the modest Asterisk install easily.   I've got a site where 
they use 6 lines and 15 users on a 500Mhz CPU w/512MB RAM and boot off 
a 2GB flash disk.


VERY modest and absolutely dominates that particular install.

Only in the larger installs will hardware be an issue, but even then 
it doesn't take much hardware (from a server perspective) to handle a 
LOT of Asterisk traffic.


RandyW

The worst problem will be older hardware that doesn't play well with 
Digium cards. The TDM400 is the one I have some experience with, and 
even motherboards that are PCI 2.2 don't always see the TDM400

The Sangoma A200 seems more forgiving.
I have to wonder if the T1/E1 cards suffer  in a similar manner?


While talking with one of the sangoma folks very recently, he was rather 
emphatic the pci bus was designed to "share" interrupts. I was a little 
concerned as a test server had the wanpipe driver sharing an interrupt 
with libata and uhc1_hcd. His comment was "that's the way its suppose to 
work, sharing interrupts as needed". I've not had any recognizable 
issues with the A200D card at all, and faxing via a A200D fxs port to a 
A200D fxo (pstn) port functions 100% reliably.


What that would suggest is the TDM400 pci firmware (whether on card 
logic or whatever) is the source of at least part of the TDM400 shared 
interrupt issue. I don't have any digium T1/E1 cards laying around, but 
if memory serves correctly, the T1/E1 cards do not use the same pci 
controller chip. That would suggest the T1/E1 cards are less of an issue 
then with the TDM400 card.


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Re: [Asterisk-Users] Texas User Group