Re: [asterisk-users] voicemail - digits/1F does not exist in any format

[Philippe Lindheimer]

I've seen this before, in an ISDN card (can't recall which one) that defaults 
the incoming language to german. Since you don't have german, it defaults to 
english files but voicemail still runs through the german logic (e.g. 1F for 
femail). I reported a bug against this, it was silently killing the call - no 
error handling. I suggested that they check if the desired language is 
installed and if not, that within the app the 'temporarily change' the language 
to english so that it doesn't go off looking for sound files that are not 
there. I can't recall the bug number - but they didn't feel it was a reasonable 
approach ... different opinions I guess, they decided the behavior was 
accetable.

philippe


From: Per Jessen [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Date: Sun, 15 Apr 2007 14:55:31 +0200
Subject: Re: [asterisk-users] voicemail - digits/1F does not exist in any
 format

 Carlos Chavez wrote:

 I am assuming you are using a language other that English?  If so, do
 you have the language files installed in the correct place?  For
 asterisk 1.2 you need a structure like this:

No, I'm using English.  The default setup that came with 1.4.1.

The other sound files are in /var/lib/asterisk/sounds/digits: 

-rw-rw-r-- 1 per 1000 1353 Feb 20 23:05 0.gsm
-rw-rw-r-- 1 per 1000 1089 Feb 20 23:05 1.gsm
-rw-rw-r-- 1 per 1000 1023 Feb 20 23:05 10.gsm
-rw-rw-r-- 1 per 1000 1353 Feb 20 23:05 11.gsm
-rw-rw-r-- 1 per 1000 1155 Feb 20 23:05 12.gsm
-rw-rw-r-- 1 per 1000 1485 Feb 20 23:05 13.gsm
-rw-rw-r-- 1 per 1000 1485 Feb 20 23:05 14.gsm
-rw-rw-r-- 1 per 1000 1518 Feb 20 23:05 15.gsm
-rw-rw-r-- 1 per 1000 1617 Feb 20 23:05 16.gsm
-rw-rw-r-- 1 per 1000 1782 Feb 20 23:05 17.gsm
-rw-rw-r-- 1 per 1000 1551 Feb 20 23:05 18.gsm
-rw-rw-r-- 1 per 1000 1650 Feb 20 23:05 19.gsm
-rw-rw-r-- 1 per 1000  990 Feb 20 23:05 2.gsm
-rw-rw-r-- 1 per 1000 1254 Feb 20 23:05 20.gsm
-rw-rw-r-- 1 per 1000  990 Feb 20 23:05 3.gsm
-rw-rw-r-- 1 per 1000 1155 Feb 20 23:05 30.gsm
-rw-rw-r-- 1 per 1000 1089 Feb 20 23:05 4.gsm
-rw-rw-r-- 1 per 1000 1155 Feb 20 23:05 40.gsm
-rw-rw-r-- 1 per 1000 1122 Feb 20 23:05 5.gsm
-rw-rw-r-- 1 per 1000 1419 Feb 20 23:05 50.gsm
-rw-rw-r-- 1 per 1000 1155 Feb 20 23:05 6.gsm
-rw-rw-r-- 1 per 10000 Feb 20 23:05 60.gsm
-rw-rw-r-- 1 per 1000 1320 Feb 20 23:05 7.gsm
-rw-rw-r-- 1 per 1000 1485 Feb 20 23:05 70.gsm
-rw-rw-r-- 1 per 1000  891 Feb 20 23:05 8.gsm
-rw-rw-r-- 1 per 1000 1155 Feb 20 23:05 80.gsm
-rw-rw-r-- 1 per 1000 1254 Feb 20 23:05 9.gsm
-rw-rw-r-- 1 per 1000 1419 Feb 20 23:05 90.gsm



/Per Jessen, Zürich



   
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Re: [asterisk-users] error in FreePBX

[Philippe Lindheimer]

something is broken in your configuration. dialparties is returning with no 
extension to dial (which could be DND, could be the phone is occupied and no CW 
active, or several other factors). And your call to voicemail is failing which 
implies something is broken in that setup since it wouldn't go there unless you 
configured it with voicemail.

Get onto the IRC (online help in freepbx or #freepbx with another IRC client) 
and find one of the user who might be able to step you through the help. (or - 
delete all your extensions, re-install and recreate them). If the only problem 
were with the phone configuration (which is also a possibility) you would still 
hit voicemail properly.

p

[EMAIL PROTECTED] wrote:From: Carlos Jerónimo [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
Date: Thu, 29 Mar 2007 18:20:46 +0100
Subject: Re: [asterisk-users] error in FreePBX

 I haved read this but i not understand what the problem. mybe you
understand why this error, for all call, in a extensions configured in
asterisk/FreePbx.

ieeta-proj-04*CLI sip show peers
Name/username  HostDyn Nat ACL Port Status
6009/6009  192.168.9.54 D   N  5062 OK (1 ms)
6000/6000  192.168.1.211D   N  5061 OK (3 ms)
2 sip peers [2 online , 0 offline]
-- Executing Macro(SIP/6009-08197f70, exten-vm|6000|6000) in new stack
-- Executing Macro(SIP/6009-08197f70, user-callerid) in new stack
-- Executing NoOp(SIP/6009-08197f70, user-callerid: device
6009) in new stack
-- Executing GotoIf(SIP/6009-08197f70, 0?report) in new stack
-- Executing GotoIf(SIP/6009-08197f70, 0?start) in new stack
-- Executing Set(SIP/6009-08197f70, REALCALLERIDNUM=6009) in new stack
-- Executing NoOp(SIP/6009-08197f70, REALCALLERIDNUM is 6009)
in new stack
-- Executing Set(SIP/6009-08197f70, AMPUSER=6009) in new stack
-- Executing Set(SIP/6009-08197f70, AMPUSERCIDNAME=6009) in new stack
-- Executing GotoIf(SIP/6009-08197f70, 0?report) in new stack
-- Executing Set(SIP/6009-08197f70, CALLERID(all)=6009 6009)
in new stack
-- Executing Set(SIP/6009-08197f70, REALCALLERIDNUM=6009) in new stack
-- Executing NoOp(SIP/6009-08197f70, TTL:  ARG1: 6000) in new stack
-- Executing GotoIf(SIP/6009-08197f70, 0?continue) in new stack
-- Executing Set(SIP/6009-08197f70, _TTL=64) in new stack
-- Executing GotoIf(SIP/6009-08197f70, 1?continue) in new stack
-- Goto (macro-user-callerid,s,21)
-- Executing NoOp(SIP/6009-08197f70, Using CallerID 6009
6009) in new stack
-- Executing Set(SIP/6009-08197f70, FROMCONTEXT=exten-vm) in new stack
-- Executing Set(SIP/6009-08197f70, VMBOX=6000) in new stack
-- Executing Set(SIP/6009-08197f70, EXTTOCALL=6000) in new stack
-- Executing Set(SIP/6009-08197f70, CFUEXT=) in new stack
-- Executing Set(SIP/6009-08197f70, CFBEXT=) in new stack
-- Executing Set(SIP/6009-08197f70, RT=15) in new stack
-- Executing Macro(SIP/6009-08197f70, record-enable|6000|IN)
in new stack
-- Executing GotoIf(SIP/6009-08197f70, 0?2:4) in new stack
-- Goto (macro-record-enable,s,4)
-- Executing DeadAGI(SIP/6009-08197f70,
recordingcheck|20070329-181220|1175188340.3) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
-- AGI Script recordingcheck completed, returning 0
-- Executing NoOp(SIP/6009-08197f70, No recording needed) in new stack
-- Executing Macro(SIP/6009-08197f70, dial|15|tr|6000) in new stack
-- Executing DeadAGI(SIP/6009-08197f70, dialparties.agi) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
-- AGI Script dialparties.agi completed, returning 0
-- Executing NoOp(SIP/6009-08197f70, Returned from dialparties
with no extensions to call) in new stack
-- Executing NoOp(SIP/6009-08197f70, DIALSTATUS is ) in new stack
-- Executing GosubIf(SIP/6009-08197f70, 0?docfu|1) in new stack
-- Executing GosubIf(SIP/6009-08197f70, 0?docfb|1) in new stack
-- Executing NoOp(SIP/6009-08197f70, Voicemail is 6000) in new stack
-- Executing GotoIf(SIP/6009-08197f70, 0?s-|1) in new stack
-- Executing NoOp(SIP/6009-08197f70, Sending to Voicemail box
6000) in new stack
-- Executing Macro(SIP/6009-08197f70, vm|6000|) in new stack
-- Executing Macro(SIP/6009-08197f70, user-callerid|SKIPTTL)
in new stack
-- Executing NoOp(SIP/6009-08197f70, user-callerid: 6009 6009)
in new stack
-- Executing GotoIf(SIP/6009-08197f70, 0?report) in new stack
-- Executing GotoIf(SIP/6009-08197f70, 1?start) in new stack
-- Goto (macro-user-callerid,s,5)
-- Executing NoOp(SIP/6009-08197f70, REALCALLERIDNUM is 6009)
in new stack
-- Executing Set(SIP/6009-08197f70, AMPUSER=6009) in new stack
-- Executing Set(SIP/6009-08197f70, AMPUSERCIDNAME=6009) in new stack
-- Executing GotoIf(SIP/6009-08197f70, 0?report) in new 

Re: [asterisk-users] Answer Confirmation with SIP/AIX channels

[Philippe Lindheimer]

I have implemented the requested call confirmaiton feature in the freepbx 
followme and ringgroup applications (asterisk 1.2 for now). You can select to 
have confirmation and by default any external call (e.g. cellphone) will 
require such confirmation, any internal phone will not (unless you force it to) 
since you usually only want confirmation when leaving the system. It is 
supported in the simultaneous ring strategy (all phones ring at once).

So you may use freepbx to do this, or take a look at how to do it in the 
dialplan, or contact me offline if you need something custom to your scenario.

philippel


From: [EMAIL PROTECTED]
CC: 
To: ASTERISK-USERS@LISTS.DIGIUM.COM
Date: Sun, 25 Mar 2007 5:44:32 -0500
Subject: [asterisk-users] Answer Confirmation with SIP/AIX channels

 
We need incoming calls to simultaneously ring SIP phones, and a cell phone
which is called via a SIP or IAX trunk.  When the cell phone answers we'd
like a brief prompt played (e.g. press # to accept call) and if # is pressed
connect the incoming call to the cell phone.

ZAP trunks have some of this functionality with the call confirmation option,
but we must use SIP or IAX trunks.

Follow-me does not appear to do this since we want simultaneous ring of all 
SIP phones, and the cell phone.  Follow-me appears to ring one device for
a period of time, then the second device for a period of time, then the third
...

The answer confirmation option is required to prevent cell phone company 
messages (e.g. the phone is turned off or out of the calling area) from
accepting the call (and stop ringing on all the SIP phones).

If this is not currently possible, we are interested in offering a bounty to
make it possible.

Thanks.
R Miller


 
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RE: [asterisk-users] Follow me on multiple numbers..

[Philippe Lindheimer]

I'm not sure what problem you are having, it should be fine on a proper freepbx 
install (and as best I know trixbox installs it properly although I don't use 
trixbox so can't say first hand). It should work fine, I know plenty of people 
who use it - I use it regularly. Feel free to try and catch me or someone else 
on the freepbx IRC to get help if you have it setup so you can take a look. 
(although I'll be gone all week cause of VON).
   
  philippel

  
From: Kevin Kiely [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Date: Sun, 18 Mar 2007 11:33:07 -0400
Subject: RE: [asterisk-users] Follow me on multiple numbers..

  

v\:* {behavior:url(#default#VML);}  o\:* {behavior:url(#default#VML);}  
w\:* {behavior:url(#default#VML);}  .shape {behavior:url(#default#VML);}
I tried to look at the code in Trixbox but when the option ‘confirm’ is 
selected in the follow me properties screen, no code is generated and the call 
goes dead.  Is there a trick to get the code generated?
   
   
  
-
  
  From: Philippe Lindheimer [mailto:[EMAIL PROTECTED] 
Sent: Saturday, March 17, 2007 12:20 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Follow me on multiple numbers..

   
  On 3/16/07, Ritesh Agrawal [EMAIL PROTECTED] wrote:

 Hi Folks,
  
   I want to setup a follow me routine so that asterisk can call me on the
   multiple numbers.
   I tried some of the samples at voip-info but there is a problem with those
   examples.
  
   I dont have coverage in my home area and my cell phone answering machine
   picks up the phone right away so my home phone never rings.
   I also want the caller to be able to leave a voicemail and the cell phone
   answering machine messes it all up.
   I have call screening setup so the call gets answered by the cell phone
   answering machine and it never accepts the call.
  
   I would appreciate if someone can help me with the setup.
  
  You can create a follow-me

 with 1.2 that requires you to confirm the call before
  answering the channel. If you need an example, go have a look at the code I
  generate in the dialplan in freepbx to do that exact thing when you choose 
call
  confirm. No need to go to 1.4 just for that.
  
  philippel



 
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Re: [asterisk-users] Follow me on multiple numbers..

[Philippe Lindheimer]

On 3/16/07, Ritesh Agrawal [EMAIL PROTECTED] wrote:

 Hi Folks,

 I want to setup a follow me routine so that asterisk can call me on the
 multiple numbers.
 I tried some of the samples at voip-info but there is a problem with those
 examples.

 I dont have coverage in my home area and my cell phone answering machine
 picks up the phone right away so my home phone never rings.
 I also want the caller to be able to leave a voicemail and the cell phone
 answering machine messes it all up.
 I have call screening setup so the call gets answered by the cell phone
 answering machine and it never accepts the call.

 I would appreciate if someone can help me with the setup.

You can create a follow-me with 1.2 that requires you to confirm the call before
answering the channel. If you need an example, go have a look at the code I
generate in the dialplan in freepbx to do that exact thing when you choose call
confirm. No need to go to 1.4 just for that.

philippel


 
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Re: [asterisk-users] Retain call control: Avoid letting call get

[Philippe Lindheimer]

Take a look at freepbx 2.2 beta. We have made both ringgroups and follow-me have a call confirmation option. When used, the ringgroup/follow-me extensions that are outside lines (like your cell phone) must confirm they want the call (press 1 to accept, 2 to decline). All the while the caller hears ringing (or MoH if chosen). If no answer, they are sent on to where ever else you want them to go (like asterisk vm, or ...)So it does what you are asking, if you don't care for freepbx, you can look at the dialplan it generates and get some good ideas. (It basically does what Dovid mentions).philippel  From: "Dovid B" [EMAIL PROTECTED]To: "Asterisk Users Mailing List - Non-Commercial Discussion"asterisk-users@lists.digium.comDate: Wed, 15 Nov 2006 02:36:53
 +0200Subject: Re: [asterisk-users] Retain call control: Avoid letting call getintocellular voicemail  You can create a macro that the person that is called has to press a key to take the call. If no key is pressed then you can send them to a menu where they can press one to leave a message or two to go back to the menu.On 11/14/06, joe a. [EMAIL PROTECTED] wrote:   Did not know how to make up a subject line for this.I have a dial
 plan that allows a caller can try my cell phone.And that's fine.If the call cannot be made, it sends caller back to voice menu. However, I'd like a way for the caller to elect to go back to the voice menu, if they end up getting the cell phone voice mail.Is that possible?joe a.___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users 

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[asterisk-users] talking caller ID

[Philippe Lindheimer]

Christian wrote:<[EMAIL PROTECTED]>From: "Christian" [EMAIL PROTECTED]To: asterisk-users@lists.digium.comDate: Wed, 08 Nov 2006 20:10:02 +0100Subject: [asterisk-users] talking caller ID Hi all,Lets say I have my incoming calls transfered to my mobile phone. When a call comes in, Asterisk will answer the call and ask the caller to hold the line while the call is being transfered.I know how to do this, but i dont want the caller to hear me answer the mobile phone. They can hear some music on hold. When I answer Asterisk will read the callerID to me and I can then decide if this call is important or not. If I press one on the mobile phone it will be connected, other wise it will be transfered to my voicemail. I think this can be done through some macro, but not sure how to do this.All the best and
 thanks,ChristianChristian,  this would be fairly straight forward. Take a look at the follow-me / ringgroup implementation of freepbx 2.2 (currently beta 2). It does call confirmation almost as you describe ('you have an incoming call, press 1 to accept, 2 to reject) while the phone rings or plays music to the caller. It would be rather trivial to tweak the dialplan that plays that message and play the callerid of the incoming call.  p 

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[asterisk-users] OT - Polycom https provisioning

[Philippe Lindheimer]

Hi,I've setup polycom https provisioning with an apache/linux server. However the log files aren't saved because there is nothing to process the http PUTS polycom uses. Does anyone have a secure solution they are using in this scenario so the phone log files can be saved?philippe 
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Re: [asterisk-users] Follow Me problems

[Philippe Lindheimer]

<[EMAIL PROTECTED]>From: "Time Bandit" [EMAIL PROTECTED]To: "Asterisk Users Mailing List - Non-Commercial Discussion" asterisk-users@lists.digium.comDate: Tue, 7 Nov 2006 08:53:51 -0500Subject: Re: [asterisk-users] Follow Me problems  Today we appear to have discovered our first bug.  We have an extension setup to "followme" by ringing that extension + an external cell # (ringall).  If nobody answers after 20 seconds the "destination if no answer" is set to go to the extensions voicemail in the "followme" module. The problem is it just keeps ringing forever.  If we delete the followme it forwards to the voicemail as per the default SIP extension configuration with voicemail enabled. Anyone run into this?  Is there a workaround?  Any advice would be
 greatly appreciated as always. Our configuration is: Supermicro Pentium D 2.66 Server with 2x512MB Memory 3ware 8006-2LP Hardware RAID 1 Sangoma A200D with 8fxo (latest firmware/drivers as of last week) CentOS 4.4 Asterisk 1.2.13 Zaptel 1.2.10 FreePBX 2.1.3When Asterisk dial the Cell phone, it goes out on the ZAP channel(Sangoma A200D), so as soon as it hit that channel, the call isconsidered answered even if the cell phone never actually pickup thecall. I didn't play with the "followme" module myself but that is whatI suspect is happening. Just watch the console and you should seesomething like "Zap/1-1 answered ..."That sound probably correct. Take a look at 2.2 beta2. You can choose to require confirmation on external numbers (press 1 to accept, 2 to decline). That way it is not treated as answered. It works very very well
 and has some additional enhancements.philippe<[EMAIL PROTECTED]><[EMAIL PROTECTED]> 


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RE: [asterisk-users] Asterisk in Seattle

[Philippe Lindheimer]

The mail system somewhere seems to have eaten some of the digetst versions of this list that are sent to me (jumped from 24 to 29). So - in case this didn't make it out, just expressing my interest. Were there many others around here who responded that I must have missed?philippePhilippe Lindheimer [EMAIL PROTECTED] wrote: If anyone wants to try to start a users group in the Seattle Area, I'm interested. (Although not in July).philippe 
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RE: [asterisk-users] Asterisk in Seattle

[Philippe Lindheimer]

If anyone wants to try to start a users group in the Seattle Area, I'm interested. (Although not in July).philippe<[EMAIL PROTECTED]>From: "calvis" [EMAIL PROTECTED]To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" asterisk-users@lists.digium.comDate: Wed, 5 Jul 2006 15:16:06 -0700Subject: RE: [asterisk-users] Asterisk in SeattleSeattle is way overdue for an Asterisk’s user group.From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom Lynn Sent: Wednesday, July 05, 2006 2:15 PM To: Asterisk  Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk in Seattle  I don't know of anybody using it inbusiness, but I'm curious to find out if there are any user groups formed or forming in the Seattle Area.On 7/5/06, Douglas Garstang [EMAIL PROTECTED] wrote:   All,  Anyone know of any companies (small, large) that are using, experimenting with, deploying, and so on, Asterisk in Washington state,
 most likely in and around Seattle? I'm curious from an employment perspective. :)   Doug. ___ --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 
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Re: [Asterisk-Users] Digium Hardware Reliability

[Philippe Lindheimer]

I would love to see some feedback on this as well. I've lost exact count now, but think I've seen about 5-6 failures on their cards TDM400P and TDM2400P cards, mostly FXO but in once case FXS. And I don't deal with that many systems, which makes this really concerning. I've started a thread on the Asterisk Forum to get more feedback on the Sangoma cards as an alternative. I'm finding it hard to think this experience is a total fluke - it would be great to hear other people's experience though - good or bad.philippe<[EMAIL PROTECTED]><[EMAIL PROTECTED]><[EMAIL PROTECTED]>From: "M.Hockings" [EMAIL PROTECTED]To: asterisk-users@lists.digium.comDate: Thu, 29 Jun 2006 21:38:20 -0400Subject: [Asterisk-Users] Digium Hardware Reliability How reliable is Digium hardware in general.?  My new TDM400P just
 died.I am trying to determine if I have a lemon.  This a new PC with a Digium TDM400P in it with a single FXO and single FXS card just stopped working today.  It has been running less than three weeks with the the FXS card and has the FXO card in it only for about a week.  Today the power went out due to a mis-configuration on my part the UPS shut down before the machine shut down.  Now, I would not think this should be a problem but the Digium card no longer responds.  lspci does not show it either so I presume it deadSo, at over 2x the cost is Sangoma hardware more sturdy than the Digium stuff?Right now we are back using the POTS phones with the nice new SPA-922's looking like cute paperweights.Mike (totally UNimpressed with Digium)___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or
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Re: [Asterisk-Users] Digium Hardware Reliability

[Philippe Lindheimer]

Andrew,you seem to be assuming a lot. These were spread out across different parts of the country (US), on projects I was involved with but deployed by more than compentent telco and engineering colleagues of mine. And ... in the majority of the cases, they were DOA (not a transient issue, noisy line or not). The warranty is there and Digium or their resellers make good - but the delays in the project and the lossed time are still real. Once working, they do seem to continue working fine.So ... don't try to read too much into it. That is why I am very interested in seeing what others are finding.pFrom: Andrew Kohlsmith [EMAIL PROTECTED]To: asterisk-users@lists.digium.comDate: Fri, 30 Jun 2006 08:49:07 -0400Subject: Re: [Asterisk-Users] Digium Hardware Reliability On Friday 30 June 2006
 02:24, Philippe Lindheimer wrote: I would love to see some feedback on this as well. I've lost exact count now, but think I've seen about 5-6 failures on their cards TDM400P and TDM2400P cards, mostly FXO but in once case FXS. And I don't deal with thatThen put proper telco line protection in place!  Good lord, it's blindingly obvious to me that you seem to be in a particularly harsh environment and that the protection on the FXO modules was not designed for the type of transient disturbances you're experiencing.-A. 
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RE: [Asterisk-Users] need help troubleshooting clipping and garbledVOIP calls

[Philippe Lindheimer]

Interesting web tool, but the results are completely misleading and wrong on my system that I just tested it on 5-6 times. And this is a connection that I use regularly for VoIP traffic, multiple channels (although just a few) and I am very familiar with the characteristics of the internet paths to various providers and other trunks I am connected to. So ... from at least one datapoint, doesn't seem reliable.As far as the QoS issue - there are some reasonable links on the wiki that describe some techniques you can use with the Cisco box and you will also want to investigate if there are issues between you and your provider. Usually though - the issues are at the edge.Oh - and if Asterisk isn't running as root, the ToS settings won't happen for the outbound packets. (In case you are using that for your ACLs on Cisco for QoS rules). There is a patch that I have never tried. I find it simpler to just use iptables and create a few simple rules to set the ToS for
 outbound packets based on SIP/IAX and RTP ports per your configuration.philippe<[EMAIL PROTECTED]><[EMAIL PROTECTED]><[EMAIL PROTECTED]>From: "James Hawks" [EMAIL PROTECTED]To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" asterisk-users@lists.digium.comDate: Thu, 29 Jun 2006 13:39:28 -0700Subject: RE: [Asterisk-Users] need help troubleshooting clipping and garbledVOIP calls Sounds like a QoS issue with your DSL provider. If you go tohttp://www.bandwidth.com/tools/voipTest it might give you some insight.James
 Hawks-Original Message-From: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED] On Behalf Of T. ShawSent: Thursday, June 29, 2006 1:27 PMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] need help troubleshooting clipping and garbledVOIPcallsHello all,I have a problem with call quality with my Asterisk setup. I'm doing VOIP only so far, but have a zaptel TDM400P in the box not being used. The problem i'm having is that when calls are placed, connected, and the far-endis reporting that they are experiencing clipping, choppy, and garbled voice conversations. So bad that we have to resort to using our cell phones. This entire setup is still being built, but any phone attached is experiencing this. Call volume is almost nil (under 20 total incoming calls a day). This is a small business setup. The server is used exclusively for Asterisk, so it
 isn't a fileserver, or anything else.The setup is as such:ipphone  ---cisco 2900XL switch  Cisco 2621 router --- dsl modem --DSL --- VOIPproviderI've configured the switch and the router to set priority and qos to prioritize voice traffic above data.Funny thing is, there is not data REALLY hitting the network. I have setup 2vlans, data vlan, and voice vlan. There are two work stations on the network, and neither is being used to hit the internet heavily (office is still being setup).Any pointers or suggestions anyone have for me as to were to look for this poor quality?It seems only the Far-end (called party), is hearing this and not the calling party.I haven't tried switching out the phones because we only have 1 type, and any of the phones i used exhibit these problems. I will try softphones to see if it is truly a "networking" issue or
 Phone issue.Is anyone using a cisco 2900 switch or router and care to provide config samples of their COS/QOS setup?Thanks!Terrelle Shaw___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo 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:  
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RE: [Asterisk-Users] Trixbox maunual configuration

[Philippe Lindheimer]

As pointed out, just build your own system. If you understand the Freepbx dialplan, you can usually do almost anything you want in _custom files including redefining contexts in such a way that upgrades do not wipe them out. It's simply a matter of spending some time to see what is being done and then extending it. On the other hand - there are some more difficult scenarios to get around and plenty of other good reasons to just role your own. You can have one, the other or the best of both if that has value to you and you are willing to understand the dialplan, config and how to integrate into and work with it. (I use all three, depending on the scenario)pFrom: "Mimmus" [EMAIL PROTECTED]To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" asterisk-users@lists.digium.comDate: Wed, 28 Jun 2006
 15:04:53 +0200Subject: RE: [Asterisk-Users] Trixbox maunual configuration   I can confirm this.  AMP/TrixBox is a wonderful project but if you like to tweak  something or you became a moreexperienced user, it will became  soonas a straitjacket.  I'm still struggling to clean AMP config files to work with  a plain Asterisk install.  From: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED] On Behalf Of JordanNovakSent: Wednesday, June 28, 2006 2:24 PMTo:asterisk-users@lists.digium.comSubject: [Asterisk-Users] Trixboxmaunual
 configuration I love the added apps installedwith trixbox, ARI, Web-Meetme, FOP, and Reports are great. FreePBX on theother hand, is nearly impossible to do everything with. Trying to edit theconfigs manually proves impossible due to the excessive use of includes andmacros. It is kind of like watching someone try to bite their own ear off. Hasanybody tried to wipe all the configs clean and program the switch manually.Will this interfere with the other apps. I would wipe out extensions.conf,voicemail.conf, IAX.conf SIP.conf queues.conf and agents.conf. I do not wantto use the FreePBX again after this. I am not trying to put down FreePBX, Iknow a lot of people have worked very hard on this. It just over complicatesthings for me.  JordanNovak   CommunicationsTechnician   ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users 
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Re: [Asterisk-Users] Remote employees using Polycom 501 lose

[Philippe Lindheimer]

The Polycom's need to have their registration time lowered. Set it to 60 seconds which will re-register every 30 seconds. The polycom doesn't have any sort of 'keep alive' feature to keep the NAT holes open. There is information on the wiki fruther describing this and how to set it up if you don't know where to look.pFrom: "Von L." [EMAIL PROTECTED]To: asterisk-users@lists.digium.comDate: Wed, 28 Jun 2006 12:04:40 -0400Subject: [Asterisk-Users] Remote employees using Polycom 501 lose ability to receive incoming calls after few minutes. Hello,Here is a breakdown of the issue I am experiencing. I have three remoteemployees, in various states, who have Polycom 501 phones. They areunable to receive incoming calls after a few minutes of the phones beingplugged in. They work immediately after
 being plugged in, but they losethe ability shortly thereafter. They can always make outbound calls, butonly to real phone numbers, not extensions.They each have NAT routers, and I have triple checked that they haveopened/forwarded the correct ports, basically 5060-3 UDP. Once theyplug the phone it (power and ethernet) I see on the CLI console of theasterisk server that the phones register:Asterisk CVS-v1-0-12/01/04-18:46:01, Copyright (C) 1999-2004 Digium.Written by Mark Spencer <[EMAIL PROTECTED]>=Connected to Asterisk CVS-v1-0-12/01/04-18:46:01 currently running onbell (pid = 3652)nell*CLIVerbosity is at least 10-- Registered SIP '3015' at XXX.XXX.XXX.XXX port 1500 expires 3600Here is the top part of my
 sip.conf;_;sip.conf;_[general]port=5060bindaddr=0.0.0.0externip=XXX.XXX.XXX.XXXlocalnet=XXX.XXX.XXX.XXX/255.255.255.248canreinvite=notos=reliabilitysrvlookup=yesdisallow=allallow=ulawdtmfmode=rfc2833nat=yesignoreregexpire=yesI know it has something to do with the NAT because if I plug my Polycomdirectly into my cable modem, thus making it sit on the Internet andhave a real IP, everything works just fine.I am curious what I am missing.Thanks.Von L.___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users 
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Re: [Asterisk-Users] Asterisk home on VMWare time sync issues

[Philippe Lindheimer]

Take a look at /etc/grub.conf and on the line(s) that look something like:kernel /vmlinuz-2.6.9-34.0.1.EL ro root=LABEL=/ add clock=pit so that it looks something like:kernel /vmlinuz-2.6.9-34.0.1.EL ro root=LABEL=/ clock=pit You will also want to install VMWare Tools and then in your ".vmx" vmware file you will want to set:tools.synctime = "TRUE"That line will not be there (I don't think) if you have not installed the VMWare Tools.Beyond that - if you still have problems, turn off the NTP
 daemon.p<[EMAIL PROTECTED]><[EMAIL PROTECTED]>From: Al Lougher [EMAIL PROTECTED]To: asterisk-users@lists.digium.comDate: Fri, 23 Jun 2006 11:29:41 -0700 (PDT)Subject: [Asterisk-Users] Asterisk home on VMWare time sync issues Hello - I am using AH on VMWARE. I have noticed that the date and time periodically loses sync with the server system time. Does anyone know why this would be, or where if I need to change a setting somewhere so it keeps time properly?Thanks!  Alan. 
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[Asterisk-Users] Passing DID to external number?

[Philippe Lindheimer]

You already posted this. I answered it yesterday also?p<[EMAIL PROTECTED]><[EMAIL PROTECTED]>From: "Brian McCarey" [EMAIL PROTECTED]To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" asterisk-users@lists.digium.comDate: Fri, 23 Jun 2006 10:27:19 +0100Subject: [Asterisk-Users] Passing DID to external number?  Hi,  We run a small  switchboard using Asterisk and Free PBX.  We have two main  extensions and two ring groups. The first ring group rings the two internal  extensions. If the internal extensions do not pick up the call after 15 seconds  then the second ring group kicks in which should ring the two internal  extensions plus two external numbers.  Firstly, how do I  pass the DID number of an incoming call to the external number so that the  external number sees the incoming number and not the voip dial out  number?  Secondly, when the  second ring group kicks in only one of the external numbers dials when both  internal extensions and both external numbers should ring according to the ring  group setting. Any ideals what's going wrong?  Kind  regards  Brian.  UK 
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Subject: [Asterisk-Users] Passing DID to external number?

[Philippe Lindheimer]

For the DID's the easiest way for you to trasmit the incoming DID is to create custom extensions for the external numbers that access the external trunk directly. (e.g. they should NOT go to Loca/. or it will not retain the orignal CID in freepbx- which is effectively how it is being sent when you put x# in the ring list).As far as why it is only ringing one of your external numbers, I can only guess (with the limited information) that you may only have a single outbound channel available on your trunk so the second number is getting rejected? (Look in the log and see if it tells you that or if not provides additonal clues).p<[EMAIL PROTECTED]><[EMAIL PROTECTED]>From: "Brian McCarey" [EMAIL PROTECTED]To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
 asterisk-users@lists.digium.comDate: Thu, 22 Jun 2006 18:40:25 +0100Subject: [Asterisk-Users] Passing DID to external number? Hi,  We run a small  switchboard using Asterisk and Free PBX.  We have two main  extensions and two ring groups. The first ring group rings the two internal  extensions. If the internal extensions do not pick up the call after 15 seconds  then
 the second ring group kicks in which should ring the two internal  extensions plus two external numbers.  Firstly, how do I  pass the DID number of an incoming call to the external number so that the  external number sees the incoming number and not the voip dial out  number?  Secondly, when the  second ring group kicks in only one of the external numbers dials when both  internal extensions and both external numbers should ring according to the ring  group setting. Any ideals what's going wrong?  Kind  regards  Brian.  UK 
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Re: [Asterisk-Users] Canreinvite

[Philippe Lindheimer]

How have you confirmed that they did not reinvite? The channels are still controlled by Asterisk (sip signalling), it is the rtp streams that go direct. You can do a sip show channel 146b518a4cd  on the specific channel to see where the rtp streams are going. Or ... if this is the only active channel on the box, just do a rtp debug. If the rtp stream is going through asterisk, it will be very obvious. If not, you won't see a constant flow of rtp debug messages going on.pFrom: "Il Neofita" [EMAIL PROTECTED]To: "Asterisk Users Mailing List - Non-Commercial Discussion" asterisk-users@lists.digium.comDate: Sun, 18 Jun 2006 05:01:20 -0400Subject: Re: [Asterisk-Users] Canreinvite This is the dial in extensionsexten = _40001,1,Dial(SIP/40001,30) exten =
 _40002,1,Dial(SIP/40002,30)  From: "Il Neofita" [EMAIL PROTECTED]To: "Asterisk Users Mailing List - Non-Commercial Discussion" asterisk-users@lists.digium.comDate: Sun, 18 Jun 2006 05:22:35 -0400Subject: Re: [Asterisk-Users] Canreinvite cosa vedo a console -- Executing Dial("SIP/40001-3760", "SIP/40002|30") in new stack -- Called 40002 -- SIP/40002-4753 is ringing -- SIP/40002-4753 answered SIP/40001-3760  -- Attempting native bridge of SIP/40001-3760 and SIP/40002-4753srvlinux*CLI sip show channelsPeer User/ANR Call ID Seq (Tx/Rx) Form Hold Last Message82.X2.XX3.X3
 40002 146b518a4cd 00103/0 alaw No Tx: ACK 82.X2.XX3.X3 40001 CBD1DB85-8B 00102/30987 alaw No Tx: ACK2 active SIP channels 
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Re: [Asterisk-Users] MOS Scores and LCR

[Philippe Lindheimer]

I don't have any links, but there has been work done to 'measure' MOS scores and I believe they are a little more sophisticated than simply tracking latency, jitter and packet loss. An example of one box that does measure/predict MOS is Edgewater Network's Edgemarc. (I have no experience with it so I don't know how good of a job it does nor if it uses any of the more 'sophisticated' measures).p<[EMAIL PROTECTED]>-- Trixter http://www.0xdecafbad.com Bret McDanelBelfast IE +44 28 9099 6461DE +49 801 777 555 3402Utrecht NL +31 306 553058  US WA +1 360 207 0479US NY +1 516 687 5200  FreeWorldDialup: 635378http://www.trxtel.com the VoIP provider that pays you!From: Daniel Salama [EMAIL PROTECTED]To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.comDate: Sat, 17 Jun 2006 01:55:28 -0400Subject: Re: [Asterisk-Users] MOS Scores and LCR Thanks for the lecture. Yes, I thought MOS was more of a perception  type of measurement, but I can't say I know enough to opinion-ate and  thus the reason for the question.Also, thanks for the links. They seem helpful. Since I have several  scripts in Cacti and Nagios, I'm gonna see if I can come up with  something that could create some performance data per provider. Then  I'll give it a such at integrating that with Asterisk, unless someone  out there has done something like it.Thanks,DanielOn Jun 17, 2006, at 2:00 AM, trixter aka Bret McDanel wrote: On Sat, 2006-06-17 at 01:26 -0400, Daniel Salama wrote: Is there any tool that can do LCR for Asterisk but also take into account MOS scores? Is it possible to automatically
 generate MOS scores on random "calls" so as to keep an updated database on a per provider, per destination, per time-of-day score? Hopefully, with that information we can create a better LCR module or script? MOS (Mean Opinion Score) is generally a bunch of people sitting there listening to audio and rating it 1-5 (there is a newer method that is "twice as good" becuase it goes 1-10, basically all values are   double). Its their opinion.  This generally cant be dont automagically and   still be MOS.  You can try to track frame drops and other things on your end to rate call quality and try to come up with something, but that technically isnt MOS. AFAIK asterisk doesnt keep statistics of jitter, frame drops or   anything else, that might be a good project for someone to take on,   especially if you have multiple
 providers so you can rate quality in a more   meaningful way.  The human ear really isnt the best tool for much of this. http://searchnetworking.techtarget.com/sDefinition/  0,,sid7_gci786677,00.html http://www.tmcnet.com/tmcnet/articles/2005/voice-quality-  measurement-voip-alan-clark-telchemy.htm http://channels.lockergnome.com/it/archives/  20050715_voipqos_mos_mean_opinion_score_explained.phtml --  Trixter http://www.0xdecafbad.com Bret McDanel Belfast IE +44 28 9099 6461DE +49 801 777 555 3402 Utrecht NL +31 306 553058  US WA +1 360 207 0479 US NY +1 516 687 5200  FreeWorldDialup: 635378 http://www.trxtel.com the VoIP provider that pays you! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To
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Re: [Asterisk-Users] FreePBX 2.1.0: Manually rewriting

[Philippe Lindheimer]

do you have selinux enabled? It should not be.pp.s. - if it comes to re-installing, you can backup all your settings with the freepbx backup utility and then restore so that you don't have to re-enter everything.From: "Lachek Butalek" [EMAIL PROTECTED]To: "Asterisk Users Mailing List - Non-Commercial Discussion" asterisk-users@lists.digium.comDate: Thu, 8 Jun 2006 12:24:24 -0400Subject: [Asterisk-Users] FreePBX 2.1.0: Manually rewriting extensions_additional.conf Figuring I knew what I was doing (I didn't - surprise) I added atotally unnecessary line in /etc/asterisk/extensions_additional.conf acouple of days ago. Troubleshooting a dialing rule issue, I'm nowrealizing that FreePBX is updating its database with the new settingsbut is not rewriting/updating
 extensions_additional.conf with thechanges I'm making.I've tried renaming the file, changing its ownership, changing itspermissions, restarting the portal, all without any success. Webresources on this issue claim the opposite problem - that customchanges to extensions_additional.conf will be automatically rewrittenevery time FreePBX/AMP is updated. If that was true, I'd be done -unfortunately, it seems this is not the case.I really don't want to reinstall FreePBX and redo my entireconfiguration again... :(Any assistance would be greatly appreciated. __Do You Yahoo!?Tired of spam?  Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___
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Re: [Asterisk-Users] Recommended Web Interface

[Philippe Lindheimer]

Dakota,freepbx is a web application and associated core dialplan that allows you to do many things on top of asterisk by generating the dialplan customizations ontop of the base that it provides. Once you spend some time understanding it, you can usually do most things that you want within the gui and almost anything else in custom dialplan applications that you can write and will coexist with freepbx. The dialplan foundation is a bit 'fat' because of the rich features set and potential that it can provide but allows for much flexibility. It is not for everyone as it clearly has its pros and cons, but there should be very little, if anything, that you couldn't do using this as a base that you can do on a 'raw' system (since you can always write custom dialplan code).[EMAIL PROTECTED], now named trixbox, has been using freepbx (or amp, which was the previous name for freepbx)
 as its main interface and dialplan, and has also added several other packages as you have mentioned.Freepbx does not provide an environment to easily run mulitple businesses on a single server. It does provide an ability to give different levels of access to different freepbx 'users.' However, I personally do not believe it is a good interface for end users. It may be reasonable to give a non-telephony IT admin or other knowledgable customer access to do certain basic functions, but beyond that it is not really geared for end users, IMHO. However - I think going forward you will see more end users portals that will provide access to change their settings within such an environment, so that eventually you may be able to have an enduser portal where they can set their features (forward, cw, dnd, follow-me settings, voicemail, etc.) on the web in addition to what they can do from the phone.philippe 
 From: "Dakota Burns" [EMAIL PROTECTED]To: asterisk-users@lists.digium.comDate: Sat, 3 Jun 2006 15:07:20 -0500Subject: [Asterisk-Users] Recommended Web InterfaceI'm currently reviewing the latest release of FreePBX (formerly known as [EMAIL PROTECTED]). Do either of you know whether FreePBX is robust enough to handle multiple clients, or have any recommendations on front-end Web interface to manage client config  provide clients access to manage their level of access (similar to how Vonage, Teliax, and others provide client access to their web management console)? The latest FreePBX is module driven - pretty cool. I've plans to step through the "Asterisk: Future of Telephony" with old laptop in order to get a different view of Asterisk. Am used to Linux  CLI -- do either of you have any preferences?
 My guess is that some purists may look at FreePBX as a lesser product but ... I think it's simply a product base built right on top of Asterisk to help new Asterisk people hit the ground running (and provides some extras such as SugarCRM, Credit Card app, etc.). Thanks,Dakota __Do You Yahoo!?Tired of spam?  Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___
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[Asterisk-Users] Re: Asterisk-Users Digest, Vol 23, Issue 4

[Philippe Lindheimer]

Kevin,since voicemail doesn't support saving in g729 format (as far as I have seen last time I looked into the code), it would need to transcode the g729 to wav or something else at this point to save the voicemail. Isn't that why it is failing when it hits the voicemail system? (sure sounds like what the error is complaining about).philippe  [EMAIL PROTECTED] From: "Kevin P. Fleming" [EMAIL PROTECTED]To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.comDate: Thu, 01 Jun 2006 10:30:32 -0500Subject: Re: [Asterisk-Users] Re: G729, voicemail, no codec_g729Steven wrote: The codec is not just for transcoding audio. It is required to read and write it
 as well.Not true. It's possible to do playback of compressed files withouthaving that codec installed. It should also be possible to record them.___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [Asterisk-Users] Re: G729, voicemail, no codec_g729

[Philippe Lindheimer]

Sorry for the repost - forgot to put the proper subject last time.Kevin,since voicemail doesn't support saving in g729 format (as far as I have seen last time I looked into the code), it would need to transcode the g729 to wav or something else at this point to save the voicemail. Isn't that why it is failing when it hits the voicemail system? (sure sounds like what the error is complaining about).philippe  [EMAIL PROTECTED] From: "Kevin P. Fleming" [EMAIL PROTECTED]To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.comDate: Thu, 01 Jun 2006 10:30:32 -0500Subject: Re: [Asterisk-Users] Re: G729, voicemail, no codec_g729Steven wrote: The codec is not just
 for transcoding audio. It is required to read and write it as well.Not true. It's possible to do playback of compressed files withouthaving that codec installed. It should also be possible to record them.___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [Asterisk-Users] Converting Voicemail wav to mp3

[Philippe Lindheimer]

Aaron,any chance you've gotten that mp3 email file such that a blackberry unit can listen to it? (I've experimented but the blackberry just doesn't like mp3 attachments, just links?)thanks,philippe  From: Aaron Daniel [EMAIL PROTECTED]To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.comDate: Thu, 1 Jun 2006 13:50:20 -0500 (CDT)Subject: Re: [Asterisk-Users] Converting Voicemail wav to mp3Not sure about storing it, but we have a patch in place that converts any voicemails to mp3 format before emailing it.On Thu, 1 Jun 2006, Douglas Garstang wrote: Anyone know if a way to have voicemail files stored as mp3's? Thanks, Doug.
 ___ --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-- Aaron DanielComputer Systems TechnicianSam Houston State University[EMAIL PROTECTED](936) 294-4198
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Re: [Asterisk-Users] Handset recommendations

[Philippe Lindheimer]

I am also a VERY happy with the Polycom 501. I will disagree with the comment "They Just Work." when it comes to NAT. Once they are setup properly (e.g. set registration timeout to 60 sec so it registers every 30 seconds and keeps NAT holes open) then they work fine. There is good info on the wiki on how to set them up.Question I have on the Aastra - isn't it 2.4Ghz, so potential interference with wireless?One thing to consider with Polycom or other phones, is you can get a wireless headset arrangement as a compromise. Another option is to get a second extension with an ATA and your favorite 5.8Ghz cordless phone.p  From: Andrew Kohlsmith [EMAIL PROTECTED]To:
 asterisk-users@lists.digium.comDate: Wed, 31 May 2006 06:34:31 -0400Subject: Re: [Asterisk-Users] Handset recommendationsOn Tuesday 30 May 2006 23:13, George A. Roberts IV wrote: Any other thoughts on good reasonably priced handsets? This is for just a couple of people who work from home offices and will be connecting to an Asterisk server hosted in a datacenter.I am a *huge* fan of the Polycom ip501. The 301 works just as well, but the display is significantly crappier. If you've got the cash, go 601.I have never used Cisco, but I've used the cheaper phones enough to know that this is one place were spending a little more is WELL worth it. And I know from personal experience that the Polycom phones have *zero* issues with being behind NAT and talking to a public-IP Asterisk box. No firewall configuration, no screwing around whatsoever. They Just Work.-A.
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Re: [Asterisk-Users] Handset recommendations

[Philippe Lindheimer]

Andrew,it depends on the routers/firewalls involved. There are plenty of people who have problems if they don't set this up and it is something that Polycom has acknowledged (not providing any type of keepalive or similar) and may fix in future firmware. However - there is also very good documentation available and furthermore, many others fall in your situation where it does just work.pFrom: Andrew Kohlsmith [EMAIL PROTECTED]To: asterisk-users@lists.digium.comDate: Wed, 31 May 2006 10:34:00 -0400Subject: Re: [Asterisk-Users] Handset recommendationsOn Wednesday 31 May 2006 10:12, Philippe Lindheimer wrote: I am also a VERY happy with the Polycom 501. I will disagree with the comment "They Just Work." when it comes
 to NAT. Once they are setup properly (e.g. set registration timeout to 60 sec so it registers every 30 seconds and keeps NAT holes open) then they work fine. There is good info on the wiki on how to set them up.I made *no* registration changes to the default values. Perhaps your router had its NAT timeout window set really short? I am using a totally-factory-standard WRT54G.-A.
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[Asterisk-Users] Ring-Answer with Polycom 501 and Asterisk

[Philippe Lindheimer]

Peter,the configurations that I have seen do not auto-answer on CID. You need to set the Alertinfo field in the sip header in order to make this work. The polycoms do have an ability to customize the ring based on the caller which is set in the telephone's inernal directory. You may be able to accomlish what you are trying this way although it is not something I have tried or looked into.I have the following (allowing any 2XX extension to be autoanswered by dialing 12XX). (I'm thinking though that there may be a new way to do Alertinfo - you may want to look around but this currently works):exten = _12XX,1,SetVar(_ALERT_INFO="Ring Answer")exten = _12XX,n,Goto(${EXTEN:1},1)  p  From: "Peter Ketteridge" [EMAIL PROTECTED]To:
 asterisk-users@lists.digium.comDate: Mon, 29 May 2006 17:56:14 +0800Subject: [Asterisk-Users] Ring-Answer with Polycom 501 and AsteriskHi GuysThis has been discussed a little in the list before so my apologies for sendig it again but I have done what others have done in the list but to no avail.I have configured Asterisk to send the callerID of extension phones as "firstname lastname" and that seems to work well and extensions show calls originating on other extensions in this
 format.I set the following in sip.cfg for one of the phones:alertInfo voIpProt.SIP.alertInfo.1.value="John Smith" voIpProt.SIP.alertInfo.1.class="4"/So when John Smith calls that phone it should ring-answeras per Polycoms documentation but it doesnt it jsut rings as per usual then goes to voicemail can anyone point me in the right direction?I want to make all extension to extension calls auto answerand go straight to speakerphone (handsfree) without the receiver having to press any button on the phone.RegardsPeter
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RE: [Asterisk-Users] macro-dial

[Philippe Lindheimer]

Domenico,as I mentioned: "...and extensions are not necessarily what you think they areeither." AMP/Freepbx 'virtualizes' extensions. The basic concept is that there are users and then there are devices. A user can have multiple devices. The default shipping mode provides the 'extensions' tab which ends up creating a user with the sam extension number as the device that you assign them. However, if you flip to 'deviceanduser' mode (see /etc/amportal.conf - AMPEXTENSIONS=) you will see that you now can control users separate from devices and you can assign multiple devices to a single user or you can make a device adhoc allowing any user to login to the device and it becomes their phone until they logout.So as I mentioned, it isn't that simple, it is the reason for all the various callerid macros, dialparties.agi, etc. that is there. If you want more detail, in addition to digging in as you have, you
 may want to move over to the freepbx.org site and/or the IRC. If you want to get rid of dialparties, maybe you can get the entire functionality into a dialplan format (and probably improve performance) and then submit it back. But as you've probably seen, dialparties itself is inegrally interwoven with macro-dial and the various other interdependencies throughout the dial plan, astdb, etc.p  From: "Mimmus" [EMAIL PROTECTED]To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"asterisk-users@lists.digium.comDate: Thu, 25 May 2006 10:21:46 +0200Subject: RE: [Asterisk-Users] macro-dialHi,I digged in dialparties.agi and found that apart from DND, hunt-group,status, etc its main function is looking up real device(s) for any user fromAstDB. In fact, AMP/FreePBX keep a
 long list of entries in AstDB for anydevice/user.I'm interested in knowing how people on this list manage link between anextension and the real device (SIP, Zap, etc).ThanksFrom: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED] On Behalf Of PhilippeLindheimerSent: Wednesday, May 24, 2006 7:42 PMTo: asterisk-users@lists.digium.comSubject: Re: [Asterisk-Users] macro-dialIt's not that simple. dialparties is fundamental to the wholedialplan in AMP/freepbx and accomplishes a lot of the features such as huntgroups, DND, etc. And extensions are not necessarily what you think they areeither. If you don't like it, you'd probably be better off writing your owndialplan or alternatively, rewrite it's entire functionality outside of anagi and then submit the mod to freepbx to streamline freepbx
 more.pFrom: "Mimmus" <[EMAIL PROTECTED]>To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"Date: Wed, 24 May 2006 18:00:36 +0200Subject: [Asterisk-Users] macro-dialHi,I'm trying to edit an AMP-derived dialplan: the macro "dial" usesthe AGIscript "dialparties.agi" to find the extension to call.I'd like to drop this script: does anyone can explain me what is itsmainjob?Thanks-- Domenico Viggiani
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RE: [Asterisk-Users] macro-dial

[Philippe Lindheimer]

I understand, seems like it might be easier to write a new dialplan from scratch though, vs. running into all sorts of strange issues? On the other hand, doing it your way will make you understand what freepbx is doing, which migh provide for your own ideas on how to do or not to do things in your own dialplan.p  From: "Mimmus" [EMAIL PROTECTED]To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"asterisk-users@lists.digium.comDate: Thu, 25 May 2006 18:25:15 +0200Subject: RE: [Asterisk-Users] macro-dialPhilippe,  I understand what you say...  I'd like to free myself from AMP/Freepbx because I feel better if I have only'vi-made' configuration files I can tweak.  I'd like also to have macro-dial entirely in the dialplan without AGI script but without losing call-forwarding, do-not-disturb, etc. functionalities.At this moment, I cleaned up a lot of things but still have dialparties.agi. I hope to thrash it in some future, when I will be able to rewrite all logic in the diaplan.Thanks  Domenico  
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RE: [Asterisk-Users] FreePBX virtualization

[Philippe Lindheimer]

A revision of what? of Freepbx? Can you elaborate?p  To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"asterisk-users@lists.digium.comDate: Thu, 25 May 2006 14:34:00 -0400Subject: RE: [Asterisk-Users] FreePBX virtualizationWe have a revision of this that we use in house. We are interested inworking with others on a version 2 skipping some of the mistakes of ourfirst version and using a better model.-Original Message-From: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED] On Behalf Of Daniel SalamaSent: Thursday, May 25, 2006 2:23 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] FreePBX virtualizationAny alternate open-source solutions?On May
 25, 2006, at 2:17 PM, Douglas Garstang wrote: Yes, but it fast becomes a provisioning and management nightmare. -Original Message- From: Kerry Garrison [mailto:[EMAIL PROTECTED] Sent: Thursday, May 25, 2006 12:07 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] FreePBX virtualization You can by creating different contexts and using the Administrators function allow them to modify some of the settings themselves. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Daniel Salama Sent: Thursday, May 25, 2006 10:42 AM To: Non-Commercial Discussion Asterisk
 Subject: [Asterisk-Users] FreePBX virtualization Does FreePBX support virtualization of its services? For example, can I use it to provide virtual PBX to different clients under the same instance of FreePBX? Or is it more geared to single office-type installation? Thanks, Daniel ___ --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 list To UNSUBSCRIBE or update
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Re: [Asterisk-Users] Placing call files in

[Philippe Lindheimer]

actually it sounds like a permission issue. You said you are doing it as root, but what is asterisk running as. I've found it is very sensitive, even to ownership. Make sure the owner:group is set to what Asterisk is running as before copying. Then, I've never had problems copying vs. moving - although I could imagine it might create problems in a race condition case.p  From: Tzafrir Cohen [EMAIL PROTECTED]To: asterisk-users@lists.digium.comDate: Wed, 24 May 2006 09:26:51 -0400Subject: Re: [Asterisk-Users] Placing call files in/var/spool/asterisk/outgoing/ does not workOn Wed, May 24, 2006 at 03:06:54PM +0200, Maxim Vexler wrote: Hello everyone  I'm trying to make asterisk get a call out using the .call system. The setup is [EMAIL PROTECTED] 2.6  This is the
 content of the file is :  Channel: Zap/g0/052MYPHONE MaxRetries: 2 RetryTime: 60 WaitTime: 30 # # Assuming that your local extensions are kept in the # context called [extensions] # Context: ext-local Extension: 210 Priority: 1   I'm coping (as root) from /root/call to  /var/spool/asterisk/outgoing/max.callyou should mv the file (and in the same filesystem, so 'rename' is used)  This is what tunes up in the console :  May 24 08:57:27 WARNING[10618]: pbx_spool.c:347 scan_service: Unable to open /var/spool/asterisk/outgoing/max.call: Permission denied, deleting May 24 08:57:27 WARNING[10618]: pbx_spool.c:389 scan_thread: Failed to scan service '/var/spool/asterisk/outgoing/max.call'   What am I doing
 wrong ?Letting asterisk read it before it is complete.-- Tzafrir
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Re: [Asterisk-Users] macro-dial

[Philippe Lindheimer]

It's not that simple. dialparties is fundamental to the whole dialplan in AMP/freepbx and accomplishes a lot of the features such as hunt groups, DND, etc. And extensions are not necessarily what you think they are either. If you don't like it, you'd probably be better off writing your own dialplan or alternatively, rewrite it's entire functionality outside of an agi and then submit the mod to freepbx to streamline freepbx more.pFrom: "Mimmus" [EMAIL PROTECTED]To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"asterisk-users@lists.digium.comDate: Wed, 24 May 2006 18:00:36 +0200Subject: [Asterisk-Users] macro-dialHi,I'm trying to edit an AMP-derived dialplan: the macro "dial" uses the AGIscript "dialparties.agi" to find the extension to call.I'd like to drop this script: does anyone can explain me what is its mainjob?Thanks-- Domenico
 Viggiani
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Re: [Asterisk-Users] macro-dial

[Philippe Lindheimer]

It also provides the 'hunt' functionality and implements the different ring strategies.From: Avi Miller [EMAIL PROTECTED]To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.comDate: Thu, 25 May 2006 07:29:46 +1000Subject: Re: [Asterisk-Users] macro-dialMimmus wrote: I'd like to drop this script: does anyone can explain me what is its main job?Dialparties.agi is used to test all of the submitted destinations for Call-Waiting and Call-Forward settings before passing the final extension(s) that can be called back to Asterisk.-- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London /2/340 Gore Street T: +61 (0) 3 9235 5400Fitzroy, VIC F: +61 (0) 3 9235 54443065 W: http://www.squiz.net/
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Re: [Asterisk-Users] Default dialplan??

[Philippe Lindheimer]

Aaron,There are probably plenty of ways to do this, off the top of my head, if you add a 'include = go-to-pbx' context within the context where your Asterisk patterns are, and there is no match, Asterisk will then begin to check the 'include' contexts in order. (It does not even look at them if it can find a match in the current context. So ... put such an include with a 'catch all' dial plan within 'go-to-pbx' that will handle any patterns that don't match and send them off to your alternate pbx.philippe  From: "Aaron Paxson" [EMAIL PROTECTED]To: asterisk-users@lists.digium.comDate: Thu, 18 May 2006 09:26:50 -0400Subject: [Asterisk-Users] Default dialplan??Hey all!I've got my Asterisk box tied into my PBX. Currently, if a call comes into my PBX, and can't find the extension, it forwards it through my Asterisk trunk to Asterisk.This works great!Is there a special dialplan function (or common usage pattern) that can do the same thing in Asterisk? i.e. If it can't find the extension, send it out Zap/g1?My dialplan works with patterns, but patterns isn't what I need here. Is anyone doing anything like this?Thanks!  ~~Aaron
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[Asterisk-Users] Re: DISA SPA3000 issues

[Philippe Lindheimer]

Just tried it on mine, worked fine:Cellphone Call - POTS - SPA3000 - Asterisk - DISA - TelasipAs an FYI, I have my SPA3000 setup with INFO for the DTMF. When I originally installed it, I couldn't get the DTMF digits to work coming in using AUTO, which is why I have it using INFO (needs to be set on both the SPA and in Asterisk).I'm running the SPA300 with Software Version: 2.0.13(GWg), Hardware Version 2.0.1(4e16).philippeFrom: Dave Hawkes [EMAIL PROTECTED]To: asterisk-users@lists.digium.comDate: Wed, 17 May 2006 13:44:43 -0400Subject:
 [Asterisk-Users] Re: DISA  SPA3000 issuesI have this exact same issue with the SPA3000, I'm assuming it must be a SPA3000 bug?Dave HawkesAlchaemist wrote: Hi,  These days I run into something quite odd. I have an [EMAIL PROTECTED] that was modified to meet our requirements. We have a completely funtional DISA which we use pretty much all the  time. I works flawlessly with incomming SIP calls from several providers,  IAX calls from FWD and with ZAP.  Recently we came out with a situation where it doesn't work... with  a SPA3000 PSTN Line. You can call, navigate de IVR, log in into our app, and then when  you go to DISA, and DISA plays the dialtone... whatever you dial is not  recognized...  This was REALLY odd... so I made a network capture with Ethereal,  and... the SPA actually STOPS sending the RTP Events after the
 second  dialtone...  To verify this, I created an IVR which played the dialtone, and  verified that it was true no RTP DTMF events (RFC2833) are sent after  the SPA listens the second dialtone.  I just reviewed the 87 pages PDF of the SPA3000... and didn't find  anything about such "feature". Now I am going to try to figure out if it has something to do with  the tones recognition of the SPA. I the meanwhile I had to write a little DISA-like app, based on  something I found on this forum, without the dialtone.  Did anyone find out anything about this issue before?  REGARDS!!! Alchaemist  ___ --Bandwidth and Colocation provided by Easynews.com --  Asterisk-Users mailing list To UNSUBSCRIBE or update options
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[Asterisk-Users] How to tell if RTP stream is has been reinvited?

[Philippe Lindheimer]

I do a sip debug on the appropriate channel or IP address and look at the SIP messages. Would be great if there were an easier way though?pFrom: "Brent Torrenga" [EMAIL PROTECTED]To: asterisk-users@lists.digium.comDate: Mon, 15 May 2006 12:52:19 -0500Subject: [Asterisk-Users] How to tell if RTP stream is has been reinvited?Howdy,How can you tell if RTP traffic has been reinvited/is bypassing an * server?Sincerely,Brent A. Torrenga[EMAIL PROTECTED]Torrenga Engineering, Inc.907 Ridge RoadMunster, Indiana 46321-1771+1 219 836 8918 x325 Voice+1 219 836 1138 Facsimilewww.torrenga.com
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Re: [Asterisk-Users] number that starts with star on PAP2

[Philippe Lindheimer]

You need to modify the dialplan within the PAP2 unit to allow that as a valid number or it won't pass it on. Take a look at the following, it is not specifically for the PAP2 but all the dialplan information should apply:http://www.sipura.com/Documents/SipuraSPAUserGuidev2.0.9.pdf  From: "Time Bandit" [EMAIL PROTECTED]To: "Asterisk Users Mailing List - Non-Commercial Discussion"asterisk-users@lists.digium.comDate: Fri, 5 May 2006 09:48:41 -0400Subject: Re: [Asterisk-Users] number that starts with star on PAP2  Why I did to mine is modify all the internal "Vertical Service  Activation Codes" to be "**x" instead of "*x". There is probably a  better way, but this worked for
 me. We tried that, but users report they are still having the same problem (the site is located in a different country so I can't check myself).Sorry, I don't have my PAP2 under hand, but this is all I did, changedevery *xx to **xx and it worked.Something that may help you ishttp://www.netphonedirectory.com/pap2_dialplan.htm Philippe Lindheimer wrote:  Yes - that's your problem. You need to porgram the dialpan in the PAP2  appropriately, disable functions you don't want, etc.  We were trying to dial *100, and there wasn't anything in either of the Codes section that started with *1. Do we have to disable every function that starts with a star to get anything to work? Also, is a function disabled by clearing it?I didn't try that so I don't know. Just make sure that you changedevery single "Vertical Service Activation Codes" to a
 double *. If youstill can't fix it, let me know and I will get back my PAP2 and try tohelp youhth
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RE: [Asterisk-Users] Call Center Phone with Auto Answer

[Philippe Lindheimer]

I don't see any reason you can't use a polycom. You should be able to solve your problem multiple ways. You can simply put the default ring on the Polycom to autoanswer if that is the sole purpose, give it a second extension to be used in the queue that is programmed to autoanswer, as a couple of examples, or design your dialplan such that the appropriate _ALERT_INFO variable is set where the queue is concerned.pFrom: "Kevin Savoy" [EMAIL PROTECTED]To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"asterisk-users@lists.digium.comDate: Fri, 5 May 2006 15:31:41 -0500Subject: RE: [Asterisk-Users] Call Center Phone with Auto AnswerThe problem with what is in wiki is that these calls are being sent to aqueue. There is no way to have the queue dial the
 preceding digit that I canthink of that would trigger this. In the example shown he has an 8 dialedbefore the extension. How would I get Asterisk to dial an 8 before sendingthe call to the logged in agent in the queue?Thanks-Original Message-From: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED] On Behalf Of Mike ClarkSent: Friday, May 05, 2006 2:55 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Call Center Phone with Auto AnswerKevin Savoy wrote: Can anyone recommend a phone to use in an inbound call center  environment that has an auto answer feature? We don't want the agents  having to acknowledge the call. The call should just activate on the  headphones. We have tried Grandstream 2000, Polycom 301, 501 and 601.  None of these support it.My Polycom phones support
 auto-answer. This link should get you started.http://www.voip-info.org/wiki-Polycom+auto-answer+configMike___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [Asterisk-Users] number that starts with star on PAP2

[Philippe Lindheimer]

 In the PAP2's setup there are all of these "Vertical Service Activation  Codes" that start with star and "Outbound Call Codec Selection Codes",  also the setup menu is accessed by pressing star four times, could they  be intefering with dialing numbers that start with a star? And is there  any way to get *8 and *XXX to dial?  Yes - that's your problem. You need to porgram the dialpan in the PAP2 appropriately, disable functions you don't want, etc.p
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Re: [Asterisk-Users] FreePBX in production?

[Philippe Lindheimer]

Rich Adamson wrote: Let's see if I can summarize various recent postings relative to the  broader topic of whether FreePBX/AAH is production-ready.It's not proper to put FreePBX/AAH in the same breath. AAH puts FreePBX ontop of their build, along with a bunch of other software. Although AAH gets 'most of the credit,' the 'value' that most users are exposed to is FreePBX (or AMP). Not to trivialize the 'integration' work that AAH does. However - I say don't put them together because all too often I have seen AAH drop things on top that break FreePBX. If you want a pure system, build your own ISO and drop FreePBX on top of it.Rich Adamson wrote: Seems the general consensus is that AAH and/or FreePBX is considered  production ready if the functionality embedded in AMP (primarily)  happens to fit the specific small
 business requirements...and "Craig" wrote: Too many limitations in terms of having a flexible diaplan. What would be nice though is if they were to produce a 'lite' version that gave a gui interface to add/change/move things - sip.conf, voicemail.conf, meetme.conf but staying well away from extensions.conf  What one considers 'prodcution ready' is a very subjective evaluation. However, I will say that I have yet to find something I can't do on a system that FreePBX has. What I mean by that is that I can modify any macro or part of the core dial plan I don't like by overriding it in the _custom file as well as add any custom dialplans or other configuration that I need. I'm sure there is something out there I will run into where this doesn't work - but I haven't hit it yet. (And if I do, I'll do the needed changes and submit it back to FreePBX). 
   I think it is fair to say that if you really understand Asterisk and then spend a small amount of time understanding what FreePBX is doing, you can easily accomplish the best of both worlds. If you don't have that level of understanding, you may be thankful for those 'magic scripts' that are making a working system behind the scene from the FreePBX GUI.If you are not interested in the fundamental functionality and 'fat/rich' dailplan, then you are better off using something else though.Remco Barende wrote: There are still some basic things missing (for example if you don't use  voicemail it is not possible to set a destination for the call if not  answered, you have to create a ring group for each extension to work  around it, this is a major issue)  Remco - take a look at the Follow Me module I added. It is
 basically a presonal ring group for each extension. If you want to do the above, just define the Follow-Me settings to ring your own extension (or more if you want) and then choose any destination you want. It effectively does 'creat a ring group for each extensions' that wants one, but it does it in such a way as to be separate and work side by side with normal ringgroups, and there is a direct link between it and the extension (or user) so that navigation is very easy as you can bounce back and forth with a single mouse click.p
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Re: [Asterisk-Users] Polycom NTP issue

[Philippe Lindheimer]

Kerry Garrison wrote: I am ready to pull my hair out. I cannot seem to get the Polycoms to  read the time properly. Regardless of the server they are pointed to our  the offset, i am getting the correct time, but 24 hours ahead. So for  today it is showing Friday April 28 but with the correct time. Any clues?  Kerry,Don't know if this will help, but I'm using a Windows Server 2003 DHCP server for my Polycom 501. In the config I use:002 Time Offset (UTC offset in seconds): 0x8f80  004 Time Server (Array of time server addresses, by preference): my list of servers  (note my list is some of the external standard ntp servers out there).p  
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