Re: [asterisk-users] evaluate SIP response codes in dialplan

[Johansson Olle E]

 I still think we need a SIP_CAUSE channel variable. :-)

Then we need to start working on aggregation rules, like what if one  
IAX channel answers and one SIP channel is busy?

For SIP-only calls, we need to add a lot of code from proxy rules for  
call forking and response aggregation. It's not an
easy task.

/O

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[asterisk-users] followme order field

[Thomas Stein]

Hello.

Does someone know what order field means in followme.conf? The Doku says:

number= number to call[2nd #[3rd #]] [, timeout value in seconds [, 
order in follow-me] ]

So an example would be:

number= 123124125,10,?

It would be nice if someone could enlighten me.

cheers
t.

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Re: [asterisk-users] evaluate SIP response codes in dialplan

[Philipp Kempgen]

Johansson Olle E schrieb:


 I still think we need a SIP_CAUSE channel variable. :-)

 Then we need to start working on aggregation rules, like what if one  
 IAX channel answers and one SIP channel is busy?
 
 For SIP-only calls, we need to add a lot of code from proxy rules for  
 call forking and response aggregation. It's not an
 easy task.

I know it's not an easy task if you'd want it to be done properly.
But then again Asterisk is not a SIP softswitch but a PBX.  :-)
I've never seen people who are asking for SIP_CAUSE expect it
to work under all circumstances. All the use cases are pretty
simple:

Dial(SIP/buddy);  // single argument

When dialling to more than 1 SIP peer

Dial(SIP/busySIP/answers_the_call);

the best thing to do would be to store the last cause code that
we receive i.e. the one of the peer who answered.

In a multi-protocol situation

Dial(SIP/busyIAX/answers_the_call);

I don't expect SIP_CAUSE to be anything meaningful. It could be
set to 000 or somesuch.


   Philipp Kempgen

-- 
AMOOCON 2009, May 4-5, Rostock / Germany   -  http://www.amoocon.de
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied  -  http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
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[asterisk-users] how to cancel new recorded message from voicemail menu?

[Klaus Darilion]

Hi!

If a user has recorded a new voicemail message (e.g. unavailable 
message) then it is prompted with 3 choices.
1. accept recording
2. listen to the recorded message
3. rerecord the message

Isn't it possible to cancel the recording?

thanks
klaus

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[asterisk-users] Description of Zaptel/DAHDI E1 alarms

[Lukas Rypl]

 Hello,

 I am missing any description of zaptel/DAHDI alarms. The TE200 series
user manual contains only a description of LEDs states. These alarms
states are visible in zttool/dahditool or in astersick CLI (zap show
status) and I wonder what is the real meaning of these alarms for E1
channel.

Possible alarm states (based on zaptel.h 1.2):
 1. No alarms
 2. Recovering from alarm
 3. In loopback (local loopback or far end?)
 4. Yellow Alarm (is it only Far end Loss of Frame?)
 5. Red Alarm (Loss of Signal?)
 6. Blue Alarm (AIS?)
 7. Not Open


 Thank you for any help.

 Lukas Rypl


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Re: [asterisk-users] followme order field

[venkat siva]

Hi Thomas Stein

this is the syntax of follow me
exten = s,5,Macro(stdexten-followme,${ARG1},${ARG2})



On Mon, Jan 19, 2009 at 4:38 PM, Thomas Stein 
thomas.st...@knowledgetools.de wrote:

 Hello.

 Does someone know what order field means in followme.conf? The Doku says:

 number= number to call[2nd #[3rd #]] [, timeout value in seconds [,
 order in follow-me] ]

 So an example would be:

 number= 123124125,10,?

 It would be nice if someone could enlighten me.

 cheers
 t.

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Re: [asterisk-users] how to cancel new recorded message from voicemail menu?

[Philipp Kempgen]

Klaus Darilion schrieb:

 If a user has recorded a new voicemail message (e.g. unavailable 
 message) then it is prompted with 3 choices.
 1. accept recording
 2. listen to the recorded message
 3. rerecord the message
 
 Isn't it possible to cancel the recording?

You could hang up.
But users might not be aware of this simple solution.


   Philipp Kempgen

-- 
AMOOCON 2009, May 4-5, Rostock / Germany   -  http://www.amoocon.de
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied  -  http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
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[asterisk-users] indications.conf entry for Iceland

[Örn Arnarson]

Hi,

Not sure where to submit this to so I'll try here. Below is the toneset for
Iceland. Hopefully this can be added into the asterisk package.

[is]
description = Iceland
ringcadence = 1000,4000
dial = 425
busy = 425/250,0/250
ring = 425/1000,0/5000
congestion = 425+250/250,0/250
callwaiting = 600/100,0/100,600/100,0/9000
record = 1400/500,0/15000
info = !950/330,!1400/330,!1800/330,0
stutter =
!400/100,!0/100,!400/100,!0/100,!400/100,!0/100,!400/100,!0/100,!400/100,!0/100,!400/100,!0/100,400

Best regards,
Örn
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[asterisk-users] G729 codec

[michel freiha]

Dear All,

I have the following CPU info on my asterisk server:

Linux switch1.domain.net 2.6.18-92.1.22.el5 #1 SMP Tue Dec 16 12:03:43 EST
2008 i686 i686 i386 GNU/Linux

I need to install G729 on the asterisk server just to pass through and not
for encoding...Which G729 package do you advice me to install?
I tried several packages with no luck

Regards
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Re: [asterisk-users] G729 codec

[morteza kashani]

1)--Download G729 modules compatible with cpu model and asterisk version
 http://asterisk.hosting.lv/
2)--Change module
 rename to codec_g729.so
 copy to /usr/lib/asterisk/modules
 set permission 755
3)--
 restart asterisk
coonect to asterisk and type 'show translation'





From: michel freiha mich...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com; asterisk-users-boun...@lists.digium.com
Sent: Monday, January 19, 2009 3:33:38 PM
Subject: [asterisk-users] G729 codec


Dear All,

I have the following CPU info on my asterisk server:

Linux switch1.domain.net 2.6.18-92.1.22.el5 #1 SMP Tue Dec 16 12:03:43 EST 2008 
i686 i686 i386 GNU/Linux

I need to install G729 on the asterisk server just to pass through and not for 
encoding...Which G729 package do you advice me to install?
I tried several packages with no luck

Regards



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Re: [asterisk-users] G729 codec

[Daniel Ortiz]

please attach:

cat /proc/cpuinfo

2009/1/19 michel freiha mich...@gmail.com

 Dear All,

 I have the following CPU info on my asterisk server:

 Linux switch1.domain.net 2.6.18-92.1.22.el5 #1 SMP Tue Dec 16 12:03:43 EST
 2008 i686 i686 i386 GNU/Linux

 I need to install G729 on the asterisk server just to pass through and not
 for encoding...Which G729 package do you advice me to install?
 I tried several packages with no luck

 Regards

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Re: [asterisk-users] G729 codec

[morteza kashani]

if u have problem :

4)--Disable selinux
 Go to /etc/selinux/ and type (vim config)
 comment All lines
 reboot your linux





From: morteza kashani kasha...@yahoo.com
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Sent: Monday, January 19, 2009 3:39:28 PM
Subject: Re: [asterisk-users] G729 codec


1)--Download G729 modules compatible with cpu model and asterisk version
 http://asterisk.hosting.lv/
2)--Change module
 rename to codec_g729.so
 copy to /usr/lib/asterisk/modules
 set permission 755
3)--
 restart asterisk
coonect to asterisk and type 'show translation'





From: michel freiha mich...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com; asterisk-users-boun...@lists.digium.com
Sent: Monday, January 19, 2009 3:33:38 PM
Subject: [asterisk-users] G729 codec


Dear All,

I have the following CPU info on my asterisk server:

Linux switch1.domain.net 2.6.18-92.1.22.el5 #1 SMP Tue Dec 16 12:03:43 EST 2008 
i686 i686 i386 GNU/Linux

I need to install G729 on the asterisk server just to pass through and not for 
encoding...Which G729 package do you advice me to install?
I tried several packages with no luck

Regards


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Re: [asterisk-users] G729 codec

[Jon Weisman]

asterisk does pass thru out of the box, there is nothing to install.

in your sip.conf

just add the following:

disallow=all
allow=g729


this will force the peer to use g729 and the end points will take care of the 
codec assuming both end points support g729 to begin with.

-jon

  - Original Message - 
  From: michel freiha 
  To: Asterisk Users Mailing List - Non-Commercial Discussion ; 
asterisk-users-boun...@lists.digium.com 
  Sent: Monday, January 19, 2009 7:03 AM
  Subject: [asterisk-users] G729 codec


  Dear All,

  I have the following CPU info on my asterisk server:

  Linux switch1.domain.net 2.6.18-92.1.22.el5 #1 SMP Tue Dec 16 12:03:43 EST 
2008 i686 i686 i386 GNU/Linux

  I need to install G729 on the asterisk server just to pass through and not 
for encoding...Which G729 package do you advice me to install?
  I tried several packages with no luck

  Regards



--


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Re: [asterisk-users] G729 codec

[David fire]

hi
just for pass through you dont need any codec...


2009/1/19 michel freiha mich...@gmail.com

 Dear All,

 I have the following CPU info on my asterisk server:

 Linux switch1.domain.net 2.6.18-92.1.22.el5 #1 SMP Tue Dec 16 12:03:43 EST
 2008 i686 i686 i386 GNU/Linux

 I need to install G729 on the asterisk server just to pass through and not
 for encoding...Which G729 package do you advice me to install?
 I tried several packages with no luck

 Regards

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-- 
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(='.'=)This is Bunny. Copy and paste bunny into your
()_()signature to help him gain world domination.
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Re: [asterisk-users] G729 codec

[Thomas Kenyon]

On 1/19/2009 12:03, michel freiha wrote:
 Dear All,

 I have the following CPU info on my asterisk server:

 Linux switch1.domain.net http://switch1.domain.net 2.6.18-92.1.22.el5
 #1 SMP Tue Dec 16 12:03:43 EST 2008 i686 i686 i386 GNU/Linux

 I need to install G729 on the asterisk server just to pass through and
 not for encoding...Which G729 package do you advice me to install?
 I tried several packages with no luck

 Regards

If you only need it to be used as passthrough, you don't need any, just 
the format interpreter that comes with asterisk.

It is worth noting that if you have conference calls, you use the Page 
function or want to record calls, then you will need to install a codec 
(since in these situations the call is transcoded inside asterisk).

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[asterisk-users] IAX IP Phone

[bilal ghayyad]

Hi All;

Anyone knows an IAX IP Phone works fine and tested?

Does polycom support IAX IP Phone? 

Regards
Bilal


  

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Re: [asterisk-users] evaluate SIP response codes in dialplan

[Philipp Kempgen]

Johansson Olle E schrieb:

 Even if I think there's only one protocol for the future

Which is? :-) SIP? Maybe XMPP?


   Philipp Kempgen

-- 
AMOOCON 2009, May 4-5, Rostock / Germany   -  http://www.amoocon.de
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied  -  http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
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[asterisk-users] adding numbers in dialplan

[Ralf Träskman]

Hi

When we ned to  call 112 (emergency number) we need to add 0379 before 112 and 
464 after for it to work, how do I do that In my dialplan?
The caller should only dial 112 on the phone.

Regards
/ralf


Ralf Träskman, IT
AdLibris AB, Odengatan 106, 113 22 Stockholm, Sweden
Dir: +46-(0)707548074, vxl: +46-(0)85460 60 00, fax: +46-(0)85460 60 99
r...@adlibris.commailto:r...@adlibris.com 
www.adlibris.comhttp://www.adlibris.com/
P Please consider the environment before printing this e-mail

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Re: [asterisk-users] adding numbers in dialplan

[Daniel Ortiz]

 exten = 112,1,Dial(SIP/Provider/0379464${EXTEN})

bye

2009/1/19 Ralf Träskman r...@adlibris.com

  Hi



 When we ned to  call 112 (emergency number) we need to add 0379 before 112
 and 464 after for it to work, how do I do that In my dialplan?

 The caller should only dial 112 on the phone.



 Regards

 /ralf



 

 Ralf Träskman, IT
 AdLibris AB, Odengatan 106, 113 22 Stockholm, Sweden
 Dir: +46-(0)707548074, vxl: +46-(0)85460 60 00, fax: +46-(0)85460 60 99
 r...@adlibris.com www.adlibris.com
 P *Please consider the environment before printing this e-mail*



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Re: [asterisk-users] adding numbers in dialplan

[Daniel Ortiz]

sorry try with:

 exten = 112,1,Dial(SIP/Provider/0379${EXTEN}464)



2009/1/19 Daniel Ortiz zate...@gmail.com

  exten = 112,1,Dial(SIP/Provider/0379464${EXTEN})

 bye

 2009/1/19 Ralf Träskman r...@adlibris.com

  Hi



 When we ned to  call 112 (emergency number) we need to add 0379 before 112
 and 464 after for it to work, how do I do that In my dialplan?

 The caller should only dial 112 on the phone.



 Regards

 /ralf



 

 Ralf Träskman, IT
 AdLibris AB, Odengatan 106, 113 22 Stockholm, Sweden
 Dir: +46-(0)707548074, vxl: +46-(0)85460 60 00, fax: +46-(0)85460 60 99
 r...@adlibris.com www.adlibris.com
 P *Please consider the environment before printing this e-mail*



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Re: [asterisk-users] adding numbers in dialplan

[Ralf Träskman]

Hi

Thanks

/ralf

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Daniel Ortiz
Sent: den 19 januari 2009 14:27
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] adding numbers in dialplan


sorry try with:

 exten = 112,1,Dial(SIP/Provider/0379${EXTEN}464)



2009/1/19 Daniel Ortiz zate...@gmail.commailto:zate...@gmail.com

 exten = 112,1,Dial(SIP/Provider/0379464${EXTEN})

bye

2009/1/19 Ralf Träskman r...@adlibris.commailto:r...@adlibris.com

Hi



When we ned to  call 112 (emergency number) we need to add 0379 before 112 and 
464 after for it to work, how do I do that In my dialplan?

The caller should only dial 112 on the phone.



Regards

/ralf





Ralf Träskman, IT
AdLibris AB, Odengatan 106, 113 22 Stockholm, Sweden
Dir: +46-(0)707548074, vxl: +46-(0)85460 60 00, fax: +46-(0)85460 60 99
r...@adlibris.commailto:r...@adlibris.com 
www.adlibris.comhttp://www.adlibris.com/
P Please consider the environment before printing this e-mail



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[asterisk-users] How to add SipAddHeader in outgoing call file.

[Mian M Asif]

How to add SipAddHeader in outgoing call file.
I am implementing a Callback scenario, in which a user makes a call to
Local Access Number. The  system have to callback to the user. During
callback a call file is generated. All I want, is to add
SipAddHeader(pchargingvector,val) in outgoing Invite.
How can I achieve this?
Please help me, where can I add SipAddHeader() in below dialplan.


exten = _X.,1,wait(1)
exten = _X.,2,Set(outCallerID=${exten:1})
exten = _X.,3,Busy(1)
exten = _X.,4,Hangup()

exten = h,1,GotoIf($[${InvalidUser} = 1]?20:2)
exten = h,2,DeadAGI(STD/STD-CBLeg1-RadAuth.pl|${SIP_HEADER(Call-ID)})
exten = h,3,Set(CALLERID(number)=${CALLERID(number)})
exten = h,4,System(echo channel:
SIP/${callback...@${lcr_terminator_std}  /tmp/${CALLERID(number)})
exten = h,5,System(echo context: STD-callback-leg2  /tmp/${CALLERID(number)})
exten = h,6,System(echo extension: s  /tmp/${CALLERID(number)})
exten = h,7,System(echo priority: 1  /tmp/${CALLERID(number)})
exten = h,8,System(echo callerid: ${outCallerID} 
/tmp/${CALLERID(number)}) ; Your CallerID goes here
exten = h,9,System(echo maxretries: 0  /tmp/${CALLERID(number)})
exten = h,10,System(echo retrytime: 3  /tmp/${CALLERID(number)})
exten = h,11,System(echo Set: confID=${confID}  /tmp/${CALLERID(number)})
exten = h,12,System(echo Set: calltime=${calltime}  /tmp/${CALLERID(number)})
exten = h,13,System(echo Set: CallBackNo=${CALLERID(number)} 
/tmp/${CALLERID(number)})
exten = h,14,System(echo Set: Leg1CallID=${Leg1CallID} 
/tmp/${CALLERID(number)})
exten = h,15,System(echo sleep 5  /tmp/${CALLERID(number)}.2)
exten = h,16,System(echo mv /tmp/${CALLERID(number)}
/var/spool/asterisk/outgoing  /tmp/${CALLERID(number)}.2)
exten = h,17,System(chmod 775 /tmp/${CALLERID(number)}.2)
exten = h,18,System(/tmp/${CALLERID(number)}.2)
exten = h,19,NoOp(Hanging up ...!!)
exten = h,20,Hangup()

[STD-callback-leg2]

exten = s,1,NoOp(Entering callback-leg2)
exten = s,2,Set(CALLERID(number)=${CallBackNo})

;-- The Script Authorizes the user on Basis of Caller ID--
;-- Plays an IVR, gets destination Phno in SIP_Dest variable -
exten = s,3,Set(TIME_NOW=${EPOCH})
exten = 
s,4,DeadAGI(STD/STD-CBLeg2-RadAuthAcc.pl|${confID}|${calltime}|${TIME_NOW}|${SIP_HEADER(Call-ID)}|${Leg1CallID})
exten = s,5,hangup()

Regards,
Asif

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[asterisk-users] How to overwrite CDR(dst) value in h priority?

[Zeeshan Zakaria]

Hi everyone,

In one of my contexts I run h priority in which I need to change the
CDR(dst) value. But it doesn't work and in the CDR dst field is recorded as
h.

Context abc {

111 = {
...
...
...
};

h = {
Set(CDR(dst)='111');
NoOp(${CDR(dst)});
Hangup();
};

};

Can anybody give me an idea how to accomplish this task? In my CDR I need to
see 111 in the dst field and not h.

Thanks

-- 
Zeeshan A Zakaria
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Re: [asterisk-users] Cisco 7941G-GE with Asterisk and CTPSEP odyssee

[Shamus Rask]

I have just got a Cisco 7941G and am experiencing the exact same  
problem (phone is requesting .tlv file from TFTP server and never asks  
for .cnf.xml file). The phone originally had SCCP on it, but I  
downloaded and flashed with the latest Cisco SIP image (8.4(3)  
released 2009-01-13). In reading your message below, it looks like you  
were going to try an incremental upgrade–did you have any success with  
this?

cheers,
   Shamus

Update and revision:
I now downloaded the oldest gettable SIP firmware for 7941/61, i.e.
8.0.2. I always get the same behaviour. But I realized it never got to
the SIP image completely loaded status.
I bought this phone and it had - no wonder - an SCCP image installed.
When plugging that into an ethernet port the first thing it does is
requesting an IP address and afterwards the CTLSEPmac.tlv file. In the
status section I see an SCCP firmware entry. When I do a factory reset
(that should be the right way to get the SIP firmware on such a phone,
right?) it now loads the term41.default.loads and some other files and
then reboots and requests the CTLSEPmac.tlv file. The firmware entry
in the status section now says term41.default.loads. Getting over this
CTLSEP step should bring the phone to load the SIP41XXX.loads file, I
assume.
But as I am not getting over this step it stays in the
term41.default.loads step, unfortunately.
Does that ring a bell to anyone? Does anyone of you have had the same
situation? In which state did you get the 7961G? SCCP? And how did you
manage to load SIP firmware onto it?

Christophorus
  I do have to answer to your suggestion of renaming the  
CTLSEPmac.tlv
  to SEPmac. The phone is still requesting CTLSEPmac.tlv and as it
  cannot find that it goes into a loop. I also let the phone do that  
the
  whole weekend so there should be no iterative process in requesting  
the
  files as I read in some howtos. Any further ideas?
  I also read that it is possible to connect and configure the phone by
  ssh. So after flashing the phone with a SIP image there should be  
some
  default username/password combination which I did not manage to  
find out
  yet. Does anyone know?
  I now am going to revert to an older release to try that. I will  
report
  any success as well as misses.
  Thanks again, Christophorus
 
   This should result in the same problem. The CTLSEPmac file is the
   first that is requested on the TFTP server. But I am going to try  
that.
   Regards and thanks,
  
   Christophorus
Try naming the empty file:
SEP0019E7D16CD6.tlv
   
Not
CTLSEP0019E7D16CD6.tlv
   
-Original Message-
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of
Christophorus Laube
Sent: Friday, January 04, 2008 10:26 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Cisco 7941G-GE with Asterisk and  
CTPSEP
odyssee
   
Thanks for the hint. I just tried that although I only see my  
worries
coming true: the CTLSEPmac.tlv file is the first one the phone
requests when booting, no possibility to set something  
different as the
SEPmac.cnf.xml should be loaded after the successful load of  
the CTL
file. And thus the phone is looping with Configuring IP and  
CTLFile
failure. Can I set this option by ssh?
Thanks a lot and in advance,
   
Christophorus
 In your SEPmac.cnf.xml file look for the setting below and  
set it to
 0:

 deviceSecurityMode0/deviceSecurityMode

 -Original Message-
 From: asterisk-users-bounces at lists.digium.com
 [mailto:asterisk-users-bounces at lists.digium.com] On Behalf  
Of Glenn
Cobb
 Sent: Friday, January 04, 2008 9:37 AM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: Re: [asterisk-users] Cisco 7941G-GE with Asterisk  
and CTPSEP
 odyssee

 Here is a little more info...

 I hooked up the 7971G-GE to my pc and grabbed this with tera- 
term. Its
 the
 console output during the CTL update process. I am using  
SIP70.8-3-3.

 NOT 09:28:45.969295 DHCP: Restart - delay = 1
 NOT 09:28:45.981198 DHCP: Sending Release...
 NOT 09:28:49.000449 DHCP:  dhcpSendReq: status 0x12301000
 NOT 09:28:49.001281 DHCP: Sending Request...
 NOT 09:28:49.015673 DHCP: ACK received
 NOT 09:28:49.016517 DHCP: Succeeded
 NOT 09:28:49.058273 DHCP: IP Address -- 10.10.10.247
 NOT 09:28:49.059129 DHCP: Subnet Mask - 255.255.255.0
 NOT 09:28:49.059960 DHCP: Default Gwy -
 NOT 09:28:49.073169 PAE: SIGIPCFG received...
 NOT 09:28:49.075897 ESP: send ADMIN, logging = 1, shell = 0,  
ipconfig
=
 1
 WRN 09:28:49.120127 SECD: WARN:getCTLInfo: ** phone has no CTL
 WRN 09:28:49.127292 SECD: WARN:getCTLInfo: ** phone has no CTL
 NOT 09:28:49.140946 CDP-D: catchipcfg getdhcpinfo IP:  
a0a0af7  Chng:1
 NOT 

Re: [asterisk-users] G729 codec

[michel freiha]

Dear Sir,

kindly find below my CPU info...I just need which package should i install

[r...@switch1 modules]# cat /proc/cpuinfo
processor   : 0
vendor_id   : GenuineIntel
cpu family  : 15
model   : 2
model name  : Intel(R) Xeon(TM) CPU 3.20GHz
stepping: 5
cpu MHz : 3199.424
cache size  : 1024 KB
physical id : 0
siblings: 1
core id : 0
cpu cores   : 1
fdiv_bug: no
hlt_bug : no
f00f_bug: no
coma_bug: no
fpu : yes
fpu_exception   : yes
cpuid level : 2
wp  : yes
flags   : fpu vme de pse tsc msr pae mce cx8 apic mtrr pge mca cmov
pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe cid xtpr
bogomips: 6401.00

processor   : 1
vendor_id   : GenuineIntel
cpu family  : 15
model   : 2
model name  : Intel(R) Xeon(TM) CPU 3.20GHz
stepping: 5
cpu MHz : 3199.424
cache size  : 1024 KB
physical id : 3
siblings: 1
core id : 0
cpu cores   : 1
fdiv_bug: no
hlt_bug : no
f00f_bug: no
coma_bug: no
fpu : yes
fpu_exception   : yes
cpuid level : 2
wp  : yes
flags   : fpu vme de pse tsc msr pae mce cx8 apic mtrr pge mca cmov
pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe cid xtpr
bogomips: 6397.21

Regards

On Mon, Jan 19, 2009 at 2:10 PM, Daniel Ortiz zate...@gmail.com wrote:

 please attach:

 cat /proc/cpuinfo

 2009/1/19 michel freiha mich...@gmail.com

 Dear All,

 I have the following CPU info on my asterisk server:

 Linux switch1.domain.net 2.6.18-92.1.22.el5 #1 SMP Tue Dec 16 12:03:43
 EST 2008 i686 i686 i386 GNU/Linux

 I need to install G729 on the asterisk server just to pass through and not
 for encoding...Which G729 package do you advice me to install?
 I tried several packages with no luck

 Regards

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Re: [asterisk-users] Call file in the future

[didier.cuffaut]

First, thanks for your help

Ok, i going to do a script and call ot with only one 'System' (cf Gordon 
Henderson) and take a look to 'incron' (T Cohen)

Just need some explanations:

1) If the call file 'failed', an 'exitstatus' is happendGood 
How to check/get these $ and put in in an * $ ? (of course, the call file have 
to have archive= yes and go to 'outgoing-done')
sorry, i'm not a linux guru and it's not a pure Asterisk pb. Anyway, could 
someone show me the complete exact way and syntax to do this?

Using something as: $ egrep -vw (^#|^) file | awk -F   '{ print $2 }'  
(or some use of awk)

2) From my first post, are these lines  OK or wrong? (syntax error?)
  tmsp = the delay in future.. say 100 seconds

  exten= ra,n,System(NOW='date %S')

  exten= ra,n,System(let NOW=$NOW+$tmsp)

  exten= ra,n,System(TOUCH_TMSP='date -d 1970-01-01 $NOW sec GMT+1 
+%Y%m%d%H%M. %S)NOTE THE 'M. %S'



  *

  or this way ?

   

  exten= ra,n,Set(touchtime=$[${EPOCH} + ${tmsp}])

  exten= ra,n,Set(TOUCH_TMSP=${STRFM(${touchtime},GMT+1,%C%y%m%d%H%M%S)

  * 


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Re: [asterisk-users] evaluate SIP response codes in dialplan

[Johansson Olle E]

19 jan 2009 kl. 11.10 skrev Philipp Kempgen:

 Johansson Olle E schrieb:


 I still think we need a SIP_CAUSE channel variable. :-)

 Then we need to start working on aggregation rules, like what if one
 IAX channel answers and one SIP channel is busy?

 For SIP-only calls, we need to add a lot of code from proxy rules for
 call forking and response aggregation. It's not an
 easy task.

 I know it's not an easy task if you'd want it to be done properly.
 But then again Asterisk is not a SIP softswitch but a PBX.  :-)
 I've never seen people who are asking for SIP_CAUSE expect it
 to work under all circumstances. All the use cases are pretty
 simple:
Well, but if we implement a half-done implementation, we will get a
ton of bug reports within days... We can't do it like that, Philipp.

(Well, looking at TLS/TCP in 1.6 I guess we can do anything... ;-) )

/O


   Dial(SIP/buddy);  // single argument

 When dialling to more than 1 SIP peer

   Dial(SIP/busySIP/answers_the_call);

 the best thing to do would be to store the last cause code that
 we receive i.e. the one of the peer who answered.

 In a multi-protocol situation

   Dial(SIP/busyIAX/answers_the_call);

 I don't expect SIP_CAUSE to be anything meaningful. It could be
 set to 000 or somesuch.


   Philipp Kempgen

 -- 
 AMOOCON 2009, May 4-5, Rostock / Germany   -  http://www.amoocon.de
 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
 AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied  -  http://www.amooma.de
 Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
 -- 

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---
* Olle E Johansson - o...@edvina.net
* Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden




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[asterisk-users] Asterisk On Solaris

[Ali Jawad]

Hi All
I got Asterisk to run on Solaris however I do need it to run in
realtime mode I.e. with the res_mysql file.
Did anyone succeed in this ?
Regards

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Re: [asterisk-users] IAX IP Phone

[David]

bilal ghayyad wrote:
 Hi All;

 Anyone knows an IAX IP Phone works fine and tested?

 Does polycom support IAX IP Phone? 

 Regards
 Bilal


   

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I am using an at-530, works fine;
http://www.atcom.cn/En_products_At530.html

-- 
powered by Gentoo/GNU Linux
http://linuxcrazy.com


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Re: [asterisk-users] How to overwrite CDR(dst) value in h priority?

[Steve Murphy]

On Mon, 2009-01-19 at 08:45 -0500, Zeeshan Zakaria wrote:
 Hi everyone,
 
 In one of my contexts I run h priority in which I need to change the
 CDR(dst) value. But it doesn't work and in the CDR dst field is
 recorded as h.
 
 Context abc {
 
 111 = {
 ...
 ...
 ...
 };
 
 h = {
 Set(CDR(dst)='111');
 NoOp(${CDR(dst)});
 Hangup();
 };
 
 };
 
 Can anybody give me an idea how to accomplish this task? In my CDR I
 need to see 111 in the dst field and not h.

CDR is specifically written to only allow certain fields to be modified.
dst wasn't one of them.

If you get rid of the h extension entirely, it won't cause an update
of the CDR. If I had to keep to the h exten (because it did other things
than just try to reset a value which was set by running the h-exten), 
then I'd get rid of the Hangup() call, because it's useless. (The
h-exten
is being run because of a hangup situation in the first place.)

murf

-- 
Steve Murphy m...@digium.com
Digium


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Re: [asterisk-users] pstn hangs up: MWI no message waiting ??

[Doug Bailey]

- sean darcy seandar...@gmail.com wrote:

 OK. Calmer now. If fact a 410 would have the same problem.
 
 I'll make the fix on our machines. Should I file a bug, or does the 
 169154 commit already fix it?
 
 sean
 
 

The issues has been corrected in trunk and the 1.6.1 branch.  
Sicne we have addressed the issue and if it works for you, then 
I don't see the need for a bug report.  If you have any other concerns 
or have any other problems with it, then go ahead and submit a bug report. 

Doug 

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Re: [asterisk-users] IAX IP Phone

[bilal ghayyad]

Dear David;

At what price u get it?

Did u test it with IAX and SIP? Are u sure it is good? As really I did not deal 
with chinese phone until now and I found it fine.

Regards
Bilal


--- On Mon, 1/19/09, David da...@linuxcrazy.com wrote:

 From: David da...@linuxcrazy.com
 Subject: Re: [asterisk-users] IAX IP Phone
 To: bilmar...@yahoo.com, Asterisk Users Mailing List - Non-Commercial 
 Discussion asterisk-users@lists.digium.com
 Date: Monday, January 19, 2009, 9:42 AM
 bilal ghayyad wrote:
  Hi All;
 
  Anyone knows an IAX IP Phone works fine and tested?
 
  Does polycom support IAX IP Phone? 
 
  Regards
  Bilal
 
 

 
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 http://lists.digium.com/mailman/listinfo/asterisk-users
 

 I am using an at-530, works fine;
 http://www.atcom.cn/En_products_At530.html
 
 -- 
 powered by Gentoo/GNU Linux
 http://linuxcrazy.com


  

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Re: [asterisk-users] How to overwrite CDR(dst) value in h priority?

[Zeeshan Zakaria]

The reason why I introduced h priority here is that I needed to get the
variable CDR(duration) for DeadAGI script which I am also running in h
priority. Without h priority, I was getting correct CDR(dst) value but not
correct CDR(duration) value even if I tried to run DeadAGI after Hangup().

Current situation is that I have to sacrifise either on CDR(duration) or on
CDR(dst) for the same call. But I am sure there must be a way to get this
information because afterall asterisk has this information and it writes it
in the CDR after call completion. And I also need these two variables after
a call is hungup so I can do something with them in my AGI acript.

Any idea how can this be done?

-- 
Zeeshan A Zakaria


 CDR is specifically written to only allow certain fields to be modified.
 dst wasn't one of them.

 If you get rid of the h extension entirely, it won't cause an update
 of the CDR. If I had to keep to the h exten (because it did other things
 than just try to reset a value which was set by running the h-exten),
 then I'd get rid of the Hangup() call, because it's useless. (The
 h-exten
 is being run because of a hangup situation in the first place.)

 murf

 --
 Steve Murphy m...@digium.com
 Digium

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On Mon, Jan 19, 2009 at 10:16 AM, Steve Murphy m...@digium.com wrote:
On Mon, 2009-01-19 at 08:45 -0500, Zeeshan Zakaria wrote:
 Hi everyone,

 In one of my contexts I run h priority in which I need to change the
 CDR(dst) value. But it doesn't work and in the CDR dst field is
 recorded as h.

 Context abc {

 111 = {
 ...
 ...
 ...
 };

 h = {
 Set(CDR(dst)='111');
 NoOp(${CDR(dst)});
 Hangup();
 };

 };

 Can anybody give me an idea how to accomplish this task? In my CDR I
 need to see 111 in the dst field and not h.
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Re: [asterisk-users] Text messaging and Asterisk

[Pascal Bruno]

Is it possible for asterisk to send sms through a GSM gateway, tor example
the Portech MV-37X?
If yes, any examples of configurations would be really apreciated.



On Tue, Oct 14, 2008 at 11:13 PM, Steve Totaro 
stot...@totarotechnologies.com wrote:

 The most flexible way but will require a bit of work and scales SMS modem
 per SMS per second.

 Install kannel and configure it to work with your SMS modem (many cell
 phones work just fine for sending and receiving).  It does not have to go on
 the asterisk box, just a box you can hit with HTTP or HTTPs.

 Make sure you have lynx installed

 In your Asterisk dialplan use system(lynx
 http://ipofyoursmsserverusernamepasswordnumbermessage) that is not the
 exact syntax but it is all documented but everything in the SMS in encoded
 in the URL.

 With five T-mobile phones, I can send five a second, it seems linear, it
 may be possible to increase throughput, I just got it working and left it at
 that.

 Messages queue until there is an available modem, sent in order.

 PLUS it is MUCH cheaper (at least in the US) than these aggregators.  With
 a T-Mobile family plan, 1,000 voice minutes, and unlimited SMS runs me about
 $135/mo.  I guess it depends on volume of SMS you are sending.

 Thanks,
 Steve Totaro


 On Tue, Oct 14, 2008 at 11:46 PM, C. Savinovich 
 c.savinov...@itntelecom.com wrote:


  Thanks, excellent point.  Furthermore, a google search on fastsms.conf
 yielded the existence of a couple of 'Asterisk SMS gateways'..wow

 CS


 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Drew Gibson
 Sent: Tuesday, October 14, 2008 2:22 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Text messaging and Asterisk

 C. Savinovich wrote:
Can somebody please give a pointer to a complete neophyte (like me) on
  text messaging, what product can I use to send and automatic text
 message
 to
  a cell phone from within the asterisk dialplan? (the part of the
 dialplan
 I
  have down, the part of the text message no)
 
  Thanks
  C. Savinovich
 
 

 I don't use it but on my Asterisk 1.4 slug there was a file
 /etc/asterisk/fastsms.conf which had info about connecting to SMS
 services for about 4c per txt.

 regards,

 Drew


 --
 Drew Gibson

 Systems Administrator
 OANDA Corporation
 www.oanda.com


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 --
 Thanks,
 Steve Totaro
 +18887771888 (Toll Free)
 +12409381212 (Cell)
 +12024369784 (Skype)

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Re: [asterisk-users] evaluate SIP response codes in dialplan

[Philipp Kempgen]

Johansson Olle E schrieb:
 19 jan 2009 kl. 11.10 skrev Philipp Kempgen:
 Johansson Olle E schrieb:

 I still think we need a SIP_CAUSE channel variable. :-)

 Then we need to start working on aggregation rules, like what if one
 IAX channel answers and one SIP channel is busy?

 For SIP-only calls, we need to add a lot of code from proxy rules for
 call forking and response aggregation. It's not an
 easy task.

 I know it's not an easy task if you'd want it to be done properly.
 But then again Asterisk is not a SIP softswitch but a PBX.  :-)
 I've never seen people who are asking for SIP_CAUSE expect it
 to work under all circumstances. All the use cases are pretty
 simple:
 Well, but if we implement a half-done implementation, we will get a
 ton of bug reports within days... We can't do it like that, Philipp.

I guess you're right.
Give them an inch and they will request a mile.
SIP_CAUSE_HALFBAKED could do the trick. ;-)

 (Well, looking at TLS/TCP in 1.6 I guess we can do anything... ;-) )

;-)


   Philipp Kempgen

-- 
AMOOCON 2009, May 4-5, Rostock / Germany   -  http://www.amoocon.de
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied  -  http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
-- 

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Re: [asterisk-users] indications.conf entry for Iceland

[Jared Smith]

On Mon, 2009-01-19 at 11:51 +, Örn Arnarson wrote:
 Not sure where to submit this to so I'll try here. Below is the
 toneset for Iceland. Hopefully this can be added into the asterisk
 package.

Could you please add it to the request tracker at
http://bugs.digium.com, so that it doesn't get lost before a developer
has the opportunity to address it?


-- 
Jared Smith
Digium, Inc. | Training Manager 




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[asterisk-users] Fring and Asterisk

[Olivier]

Hi,

Is anyone using Fring as a SIP client to an Asterisk server ?
A prospective customer of mine is asking to integrate its iphones with an
Asterisk server and after googling, I still have some unanswered questions :

1. Which codecs are available when calling from fring ?
2. Is it easy and natural to change your presence status (available, busy,
...) with Fring or will users prefer to use another software (bundled with
iPhones) or to do nothing at all ?
3. Is it possible to add custom presence status in Fring client ?
4. Is it possible and recommended to limit Fring usage to WiFi presence ?
5. Would fring replies to Qualify messages ?

Regards
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Re: [asterisk-users] Description of Zaptel/DAHDI E1 alarms

[Jared Smith]

On Mon, 2009-01-19 at 11:10 +0100, Lukas Rypl wrote:
  I am missing any description of zaptel/DAHDI alarms. The TE200 series
 user manual contains only a description of LEDs states. These alarms
 states are visible in zttool/dahditool or in astersick CLI (zap show
 status) and I wonder what is the real meaning of these alarms for E1
 channel.

I can't speak for all the possible states in the T1/E1 card driver, but
I can state that typically in T1s and E1s you have three different
general alarm states: RED alarms, YELLOW alarms, and BLUE alarms.  (This
is a brief synopsis of the information we cover in the Asterisk Advanced
training class.)

Red alarm
---
Your T1/E1 port will go into red alarm when it maintain synchronization
with the remote switch.  A red alarm typically indicates either a
physical wiring problem, loss of connectivity, or a framing and/or
line-coding mismatch with the remote switch.  When your T1/E1 port loses
sync, it will transmit a yellow alarm to the remote switch to indicate
that it's having a problem receiving signal from the remore switch.
(The easy way to remember this is that the R in red stands for right
here and receive... indicating that we're having a problem right here
receiving the signal from the remote switch.)

Yellow alarm or RAI (Remote Alarm Indication)
---
Your T1/E1 port will go into yellow alarm when it receives a signal from
the remote switch that the port on that remote switch is in red alarm.
This essentially means that the remote switch is not able to maintain
sync with you, or is not receiving your transmission.  (The easy way to
remember this is that the Y in yellow stands for yonder... indicating
that the remote switch (over yonder) isn't able to see what you're
sending.)

Blue alarm or AIS (Alarm Indication Signal)
---
Your T1/E1 port will go into blue alarm when it receives all unframed 1s
on all timeslots from the remote switch.  This is a special signal to
indicate that the remote switch is having problems with it's upstream
connection.  As far as I know, dahdi_tool/zttool and Asterisk don't
correctly indicate a blue alarm (at least I've never seen them indicate
one).  The easy way to remember this is that streams are blue, so a blue
alarm indicates a problem upstream from the switch you're connected to.

I hope the explanation helps.


-- 
Jared Smith
Digium, Inc. | Training Manager 




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Re: [asterisk-users] IAX IP Phone

[Joseph]

On Mon, 19 Jan 2009, bilal ghayyad wrote:

Hi All;

Anyone knows an IAX IP Phone works fine and tested?

Does polycom support IAX IP Phone? 

Regards
Bilal

How about IAX2 adapter from digium?
I've been uing it and it works very well. 

-- 
#Joseph
GPG KeyID: ED0E1FB7

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Re: [asterisk-users] Text messaging and Asterisk

[Gordon Henderson]

On Mon, 19 Jan 2009, Pascal Bruno wrote:

 Is it possible for asterisk to send sms through a GSM gateway, tor example
 the Portech MV-37X?
 If yes, any examples of configurations would be really apreciated.

AIUI, the Portechs can recieve TXTs and you can see them via their Web 
interface.. I don't think you can send with them.

However, I can get asterisk to send TXT messages via a GSM gateway I have 
on the serial port of the server using the Linux command-line SMS 
programs. That's easy enough.

   System(putsms 44mobilenumber 'message goes here')

However, it takes my Siemens TC35 GSM terminal about 4.5 seconds to send 
each message which might stall a dial-plan somewhat... But there are 
various message handling systems avalable that will queue outgoing 
messages, etc. so the send command can return instantly to the dialplan 
with sending going on in the background.

Gordon

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[asterisk-users] Server freeze kernel panic

[Plugworld]

Hi All

I'm having some serious kernel panic while using digium cards.

 

It may be related to IRQ shared. 

Can this cause a lot of drop call and bad voice quality ?

Do you guys know if there is a way I can assign one IRQ for each digium card
?

 

Thanks a lot.

 

Here is the output of /var/log/syslog

kernel: [ 3821.982893] Uhhuh. NMI received for unknown reason 20.

kernel: [ 3821.982938] Do you have a strange power saving mode enabled?

kernel: [ 3821.982964] Dazed and confused, but trying to continue

 

dahdi_hardware 

pci::04:08.0 wct4xxp+ d161:0420 Wildcard TE420 (4th Gen)

pci::06:08.0 wct4xxp+ d161:0420 Wildcard TE420 (4th Gen)

pci::08:02.0 wct4xxp+ d161:0410 Wildcard TE410P (3rd Gen)

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[asterisk-users] Need help registering Cisco 7960 Phones on Asterisk

[Zeeshan Zakaria]

Hi everyone,

I googled this followed the instructions, but it hasn't work for me yet.

I have universal setting in SIPDefault.cnf and phone specific settings in
SIPXX.cnf. But it doesn't get registered.

I need to register it on two different asterisk boxes. So my
SIPXX.cnf looks like this:

phone_label: Zeeshan A Zakaria

line1_name: 523
line1_displayname: Zeeshan A Zakaria
line1_authname: 523
line1_password: 523
line1_shortname: x523

line2_name: 523
line2_displayname: Zeeshan
line2_authname: 523
line2_password: 523
line2_shortname: x523

line3_name: 224
line3_displayname: Zeeshan
line3_authname: 224
line3_password: 224
line3_shortname: x224

SIPDefault.cnf contains default settings along with proxy info like this:

proxy1_address: xxx.xxx.xxx.xxx
proxy1_port: 5060

proxy2_address: xxx.xxx.xxx.xxx
proxy2_port: 5060

proxy3_address: xxx.xxx.xxx.xxx
proxy3_port: 5060


Same settings work fine from Grandstream phone, and X-lite. What am I
missing on this Cisco phone configuration?

-- 
Zeeshan A Zakaria
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Re: [asterisk-users] Server freeze kernel panic

[Luis Morales]

on kernel boot parameters do it:

acpi=off

Regards,

Luis Morales

On Mon, Jan 19, 2009 at 12:25 PM, Plugworld plugwo...@micnes.com wrote:
 Hi All

 I'm having some serious kernel panic while using digium cards.



 It may be related to IRQ shared.

 Can this cause a lot of drop call and bad voice quality ?

 Do you guys know if there is a way I can assign one IRQ for each digium card
 ?



 Thanks a lot.



 Here is the output of /var/log/syslog

 kernel: [ 3821.982893] Uhhuh. NMI received for unknown reason 20.

 kernel: [ 3821.982938] Do you have a strange power saving mode enabled?

 kernel: [ 3821.982964] Dazed and confused, but trying to continue



 dahdi_hardware

 pci::04:08.0 wct4xxp+ d161:0420 Wildcard TE420 (4th Gen)

 pci::06:08.0 wct4xxp+ d161:0420 Wildcard TE420 (4th Gen)

 pci::08:02.0 wct4xxp+ d161:0410 Wildcard TE410P (3rd Gen)

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-- 
-
Luis Morales
Consultor de Tecnologia
Cel: +(58)416-4242091
-
Empieza por hacer lo necesario, luego lo que es posible... y de
pronto estarás haciendo lo imposible

Leonardo Da'Vinci
-

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Re: [asterisk-users] How to overwrite CDR(dst) value in h priority?

[Tilghman Lesher]

On Monday 19 January 2009 09:34:43 am Zeeshan Zakaria wrote:
 The reason why I introduced h priority here is that I needed to get the
 variable CDR(duration) for DeadAGI script which I am also running in h
 priority. Without h priority, I was getting correct CDR(dst) value but not
 correct CDR(duration) value even if I tried to run DeadAGI after Hangup().

 Current situation is that I have to sacrifise either on CDR(duration) or on
 CDR(dst) for the same call. But I am sure there must be a way to get this
 information because afterall asterisk has this information and it writes it
 in the CDR after call completion. And I also need these two variables after
 a call is hungup so I can do something with them in my AGI acript.

 Any idea how can this be done?

If you're using the cdr_adaptive_odbc backport (for 1.4), you can work
around this limitation by using aliases:

cdr_adaptive_odbc.conf:
[first]
dsn=mysql1
alias dst = does_not_exist
alias realdst = dst

extensions.conf:
exten = _X.,1,Set(CDR(realdst)=${EXTEN})
...

If you're using 1.6, there isn't a problem, because CDR(duration) will never
return zero except during the first half second of a call.

-- 
Tilghman

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Re: [asterisk-users] Description of Zaptel/DAHDI E1 alarms

[Tzafrir Cohen]

On Mon, Jan 19, 2009 at 04:30:37PM +, Jared Smith wrote:
 On Mon, 2009-01-19 at 11:10 +0100, Lukas Rypl wrote:
   I am missing any description of zaptel/DAHDI alarms. The TE200 series
  user manual contains only a description of LEDs states. These alarms
  states are visible in zttool/dahditool or in astersick CLI (zap show
  status) and I wonder what is the real meaning of these alarms for E1
  channel.
 
 I can't speak for all the possible states in the T1/E1 card driver, but
 I can state that typically in T1s and E1s you have three different
 general alarm states: RED alarms, YELLOW alarms, and BLUE alarms.  (This
 is a brief synopsis of the information we cover in the Asterisk Advanced
 training class.)

[snip]

 I hope the explanation helps.

You can also find it in the README of DAHDI:

http://docs.tzafrir.org.il/dahdi-linux/#_alarm_types

Commments would be welcomed

-- 
   Tzafrir Cohen
icq#16849755  jabber:tzafrir.co...@xorcom.com
+972-50-7952406   mailto:tzafrir.co...@xorcom.com
http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir

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Re: [asterisk-users] IAX IP Phone

[Jeff LaCoursiere]

On Mon, 19 Jan 2009, Joseph wrote:

 On Mon, 19 Jan 2009, bilal ghayyad wrote:

 Hi All;

 Anyone knows an IAX IP Phone works fine and tested?

 Does polycom support IAX IP Phone?

 Regards
 Bilal

 How about IAX2 adapter from digium?
 I've been uing it and it works very well.


Wow, that has NOT been my experience, though it has been a few years 
(2005) since I used them.  The ones I purchased were first of all 
expensive.  They overheated and froze up often.  Only a single port.  No 
dual ethernet option.  Provisioning is a PITA.  Codec support was minimal. 
I thought at the time that being IAX I wouldn't have to worry about NAT 
issues and that was worth the extra difficulties, but I have been using 
Linksys PAP2Ts ever since and never looked back.  And it has been over a 
year since I had any NAT issue to deal with, though have now installed 
them in hundreds of different configurations.

Perhaps these things have been rectified since...

j

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[asterisk-users] Interesting observation

[Darrick Hartman]

I have an interesting observation which I thought I'd pass along to save 
other people from spending time trying to 'fix' it.

One of my clients uses Charter's so called business phone service. 
They provide 'analog' phone lines over IP.  In general, they've worked 
OK.  End users were saying that the phone are cutting out at times. 
What I've observed is they actually do cut out (meaning all inbound 
audio is momentarily lost) if a loud noise is created on the local end. 
  This client has a machine shop so you can imagine that at times it 
does get quite loud.

I spent a few hours trying to different setting in the Polycom phones, 
but finally thought I'd try plugging an analog headset into the Charter 
CPE device directly.  The same behavior was experienced.  It appears 
they have a 'feature' which cuts out the incoming audio if a loud noise 
(simulated by blowing into the receiver) is experienced outgoing.

Pretty much going to a true analog service is the only solution that I 
can think of.  Would be interested if anyone has other thoughts.

Darrick

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Re: [asterisk-users] How to overwrite CDR(dst) value in h priority?

[Zeeshan Zakaria]

Thanks for this info. I am using Asterisk 1.4. I'll try this method and hope
it'll solve my problem in h priority.


On Mon, Jan 19, 2009 at 12:18 PM, Tilghman Lesher 
tilgh...@mail.jeffandtilghman.com wrote:

 On Monday 19 January 2009 09:34:43 am Zeeshan Zakaria wrote:
  The reason why I introduced h priority here is that I needed to get the
  variable CDR(duration) for DeadAGI script which I am also running in h
  priority. Without h priority, I was getting correct CDR(dst) value but
 not
  correct CDR(duration) value even if I tried to run DeadAGI after
 Hangup().
 
  Current situation is that I have to sacrifise either on CDR(duration) or
 on
  CDR(dst) for the same call. But I am sure there must be a way to get this
  information because afterall asterisk has this information and it writes
 it
  in the CDR after call completion. And I also need these two variables
 after
  a call is hungup so I can do something with them in my AGI acript.
 
  Any idea how can this be done?

 If you're using the cdr_adaptive_odbc backport (for 1.4), you can work
 around this limitation by using aliases:

 cdr_adaptive_odbc.conf:
 [first]
 dsn=mysql1
 alias dst = does_not_exist
 alias realdst = dst

 extensions.conf:
 exten = _X.,1,Set(CDR(realdst)=${EXTEN})
 ...

 If you're using 1.6, there isn't a problem, because CDR(duration) will
 never
 return zero except during the first half second of a call.

 --
 Tilghman

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-- 
Zeeshan A Zakaria
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Re: [asterisk-users] Digium TE220 supported protocol

[Benoit]

Laurent a écrit :
 Le 19.01.2009 08:50, Benoit a écrit :
   
 Laurent a écrit :
 

   
 Well, the telcos techs said a straight cable should do the trick, but 
 since i didn't get any isdn link up
 with the straight, i built a crossover like what you described, with no 
 luck either.

 

 Did you check (like with a multimeter or something similar) the
 connectivity of your cable ? the first E1 crossover cable I made
 had a problem (entirely my own fault) and I thought it didn't
 work. The way I checked was by connecting the two ports of the
 Digium card with the crossover cable, and when I saw the LEDs turn
 red I knew my cable was not good.
   
Yes !

That was it, i tested my cable with a multimeter, but i have been caught
by the mirror
effect on the diagram. Rebuilding the cable using correct numbering and
voila!
The link is up now !

Thanks a lot (and a trick for the telco tech that assured me a straight
cable would work).


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Re: [asterisk-users] Interesting observation

[Tim Nelson]

My understanding is that Charter 'telephone' doesn't use IP at all but rather 
uses some additional frequency spectrum on their cable network. Hence, the 
reason why faxing with their service is reliable unlike other providers who are 
*actually* using VoIP.

It sounds like they're suffering from clipping of some sort or almost a 
half-duplex audio situation. Weird. I've seen some el-cheapo cordless phones 
behave this way but never a 'business' solution. Eek.

Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332 x105

- Darrick Hartman dhart...@djhsolutions.com wrote:

 I have an interesting observation which I thought I'd pass along to
 save 
 other people from spending time trying to 'fix' it.
 
 One of my clients uses Charter's so called business phone service. 
 They provide 'analog' phone lines over IP.  In general, they've worked
 
 OK.  End users were saying that the phone are cutting out at times.
 
 What I've observed is they actually do cut out (meaning all inbound
 
 audio is momentarily lost) if a loud noise is created on the local
 end. 
   This client has a machine shop so you can imagine that at times it 
 does get quite loud.
 
 I spent a few hours trying to different setting in the Polycom phones,
 
 but finally thought I'd try plugging an analog headset into the
 Charter 
 CPE device directly.  The same behavior was experienced.  It appears 
 they have a 'feature' which cuts out the incoming audio if a loud
 noise 
 (simulated by blowing into the receiver) is experienced outgoing.
 
 Pretty much going to a true analog service is the only solution that I
 
 can think of.  Would be interested if anyone has other thoughts.
 
 Darrick
 
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Re: [asterisk-users] IAX IP Phone

[David]

bilal ghayyad wrote:
 Dear David;

 At what price u get it?

 Did u test it with IAX and SIP? Are u sure it is good? As really I did not 
 deal with chinese phone until now and I found it fine.

 Regards
 Bilal


 --- On Mon, 1/19/09, David da...@linuxcrazy.com wrote:

   
 From: David da...@linuxcrazy.com
 Subject: Re: [asterisk-users] IAX IP Phone
 To: bilmar...@yahoo.com, Asterisk Users Mailing List - Non-Commercial 
 Discussion asterisk-users@lists.digium.com
 Date: Monday, January 19, 2009, 9:42 AM
 bilal ghayyad wrote:
 
 Hi All;

 Anyone knows an IAX IP Phone works fine and tested?

 Does polycom support IAX IP Phone? 

 Regards
 Bilal


   

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 I am using an at-530, works fine;
 http://www.atcom.cn/En_products_At530.html

 -- 
 powered by Gentoo/GNU Linux
 http://linuxcrazy.com
 


   

   
I use it now, at first I used it with IAX2 for about 4 months until I
figured out my nat issues and now I use it with sip, I mainly use it for
my podcast interviews and use asterisk to do the recording.
Here is where I purchased the phone, never had a problem, the sound
quality is OK, I really don't know any difference as this is my first IP
phone :)

http://cgi.ebay.com/ATCOM-VoIP-IP-Phone-AT-530-2Ports-support-2-SIP-IAX2_W0QQitemZ300286465044QQcmdZViewItemQQptZCOMP_Telecom_IP_Telephony?_trksid=p3286.m20.l1116

-- 
powered by Gentoo/GNU Linux
http://linuxcrazy.com


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Re: [asterisk-users] Interesting observation

[Frank Bulk]

Tim:

Are you referring to the older-style cable telephony where they had an
analog carrier on the cable plant, or PacketCable VoIP?

Frank

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Nelson
Sent: Monday, January 19, 2009 12:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Interesting observation

My understanding is that Charter 'telephone' doesn't use IP at all but
rather uses some additional frequency spectrum on their cable network.
Hence, the reason why faxing with their service is reliable unlike other
providers who are *actually* using VoIP.

It sounds like they're suffering from clipping of some sort or almost a
half-duplex audio situation. Weird. I've seen some el-cheapo cordless phones
behave this way but never a 'business' solution. Eek.

Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332 x105

- Darrick Hartman dhart...@djhsolutions.com wrote:

 I have an interesting observation which I thought I'd pass along to
 save
 other people from spending time trying to 'fix' it.

 One of my clients uses Charter's so called business phone service.
 They provide 'analog' phone lines over IP.  In general, they've worked

 OK.  End users were saying that the phone are cutting out at times.

 What I've observed is they actually do cut out (meaning all inbound

 audio is momentarily lost) if a loud noise is created on the local
 end.
   This client has a machine shop so you can imagine that at times it
 does get quite loud.

 I spent a few hours trying to different setting in the Polycom phones,

 but finally thought I'd try plugging an analog headset into the
 Charter
 CPE device directly.  The same behavior was experienced.  It appears
 they have a 'feature' which cuts out the incoming audio if a loud
 noise
 (simulated by blowing into the receiver) is experienced outgoing.

 Pretty much going to a true analog service is the only solution that I

 can think of.  Would be interested if anyone has other thoughts.

 Darrick

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[asterisk-users] Suggestions on how to create a hunt or hunt like (rollover, multi-line) group or where to get one?

[Alfred Monticello]

I have about 5 incoming USA SIP lines, but my provider does not have any sort 
of roll-over or huntgroup feature. Does anybody have an idea on how I can 
create a general number that will ring to the next available, non-busy SIP line 
that I have? Is there a provider out there that would do this? 

Any suggestions would be greatly welcome.

Thank you.


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Re: [asterisk-users] Interesting observation

[David Gibbons]

snip
My understanding is that Charter 'telephone' doesn't use IP at all but
rather uses some additional frequency spectrum on their cable network.
Hence, the reason why faxing with their service is reliable unlike other
providers who are *actually* using VoIP.
/snip

I think what you're referring to is the general hesitance of the cable 
providers to call their phone service VOIP service. VOIP still has a negative 
connotation with most regular folks, so they don't want to negative PR.

I'm don't have any facts, but I'll bet you a penny that they don't have a 
proprietary system using something /OTHER/ than IP to send encapsulated voice 
over 'additional frequency spectrum'. That would be prohibitively expensive to 
develop and pointless from a technical standpoint, given that IP telephony is 
already set to deploy and relatively mature.

The reliability of faxing is based soley on network jitter and latency and 
codec compression. I've found that taking the compression out of the mix (using 
g.711 ulaw) and controlling the jitter and latency (something that's easy to do 
on a private network like theirs with QOS) causes faxing to be pretty darn 
reliable.

--Dave

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Re: [asterisk-users] Interesting observation

[Tim Nelson]

- David Gibbons d...@videon-central.com wrote:
 I think what you're referring to is the general hesitance of the cable
 providers to call their phone service VOIP service. VOIP still has a
 negative connotation with most regular folks, so they don't want to
 negative PR.

True.
 
 I'm don't have any facts, but I'll bet you a penny that they don't
 have a proprietary system using something /OTHER/ than IP to send
 encapsulated voice over 'additional frequency spectrum'. That would be
 prohibitively expensive to develop and pointless from a technical
 standpoint, given that IP telephony is already set to deploy and
 relatively mature.
 

True.
 The reliability of faxing is based soley on network jitter and latency
 and codec compression. I've found that taking the compression out of
 the mix (using g.711 ulaw) and controlling the jitter and latency
 (something that's easy to do on a private network like theirs with
 QOS) causes faxing to be pretty darn reliable.
 
 --Dave
Also True.

I'd be willing to bet *TWO* pennies that you're correct. I certainly was not 
coming into the conversation as an expert, just stating what I'd read/heard of 
their service... hence the My understanding is that... beginning to the 
email. :-)

Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332 x105

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Re: [asterisk-users] IAX IP Phone

[Steve Edwards]

On Mon, 19 Jan 2009, Jeff LaCoursiere wrote:

 On Mon, 19 Jan 2009, Joseph wrote:

 On Mon, 19 Jan 2009, bilal ghayyad wrote:

 Anyone knows an IAX IP Phone works fine and tested?

 How about IAX2 adapter from digium? I've been uing it and it works very 
 well.

 Wow, that has NOT been my experience, though it has been a few years 
 (2005) since I used them.  The ones I purchased were first of all 
 expensive.  They overheated and froze up often.  Only a single port. 
 No dual ethernet option.  Provisioning is a PITA.  Codec support was 
 minimal. I thought at the time that being IAX I wouldn't have to worry 
 about NAT issues and that was worth the extra difficulties, but I have 
 been using Linksys PAP2Ts ever since and never looked back.  And it has 
 been over a year since I had any NAT issue to deal with, though have now 
 installed them in hundreds of different configurations.

 Perhaps these things have been rectified since...

I've had an Iaxy2 (s101i) for several years. It's always worked fine for 
me. It does generate a very slight amount of heat. Just enough so you know 
it's plugged in -- as if the overly-bright blue registration LED wasn't a 
clue.

The original iaxprov command line tool was a bit of a bother, but the 
iaxprov Asterisk command is better since it uses a centralized 
iaxprov.conf file to provision the devices. I prefer devices that 
request (via TFTP) configuration, but once configured you're done.

It gets my vote for a just works device and it's great to travel with as 
long as you remember to use a transformer instead of an adapter in 
countries (England) that insist on delivering excessive voltage :)

Thanks in advance,

Steve Edwards  sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000

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[asterisk-users] [somewhat OT] seeking ideas/input for my thesis

[sp4rc]

Hello VoIP guys

Sorry for being somewhat off-topic. At the moment I am studying
informatics in the seventh semester and I need to start thinking about
my thesis. As I am very interested in VoIP technologies I thought about
picking this as my main topic. So far I have only little experience in
this area. I have been fiddling around with siproxd and pfSense and have
red the one or the other packet dump containing SIP and RTP traffic, had
a look into codecs, STUN, etc... but very cursorily, and that's the
reason why I am quite unsure on which track to go. I think I am quite
familiar with many network protocols and devices... so here comes the
question of the questions:

What would be a great project for my thesis to work on in the VoIP
field? What are topics that still need special development? The time
frame should be around 300 hours but don't take this value too
seriously... 

An idea: contact synchronisation via SIP
Are there any (working or concept) extensions on using SIP to synchronize 
contacts
in the way icq does it? (server-side contacts)

Any ideas are welcome!
/sp4rc


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Re: [asterisk-users] [somewhat OT] seeking ideas/input for my thesis

[Alex Balashov]

sp4rc wrote:


 An idea: contact synchronisation via SIP
 Are there any (working or concept) extensions on using SIP to synchronize 
 contacts
 in the way icq does it? (server-side contacts)

The OpenSER/Kamailio/OpenSIPS technology stack along with XCAP and XMPP 
provides pretty good solutions to this:

http://www.kamailio.org/pub/OpenSER-Summit-2008/15-openser_summit_2008-QuoVadis-anca_vamanu-presence.pdf

-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (678) 237-1775

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Re: [asterisk-users] compare Linksys SPA8000 and Grandstream GXW4008

[Vieri]

Thanks!

I've just ordered a Linksys SPA8000 to try it out and compare it with my 
Grandstream GXW4008 devices.

They are similar feature-wise. Linksys/Cisco should theoretically be a lot more 
stable/reliable...

The only thing I'm missing in the SPA8000 but is available in the GXW4008 
devices are the dual ethernet ports (LAN/WAN).
However, I saw a video (youtube I think) where I saw a second NIC on the 
SPA8000 (called aux). Is it only for admin use or can it be used just like 
the other nic, to register to a PBX?

I'd like to give the SPA8000 NICs IP addresses on 2 different subnets, both 
connected to the same PBX server (but with two NICs) via 2 different switches 
(registration via DNS SRV). That's how I setup the GXW4008 and they work great 
if any one of the switches fails. I'm wondering if I can do that with the 
SPA8000 and what the second NIC on that device is for.

Vieri

--- On Sun, 1/18/09, C F shma...@gmail.com wrote:

 Linksys has far better products than Grandstream. I
 wouldn't even put
 them in the same email. Let alone on a subject line.
 It's like asking: Door stoppers vs Phones.
 
 On Sun, Jan 18, 2009 at 10:41 PM, Positively Optimistic
 positivelyoptimis...@gmail.com wrote:
  We have used a lot of the GXP400x series..  In my
 option, they have a high
  failure rate...we've been testing the SPA8000s
 in our lab...   my
  opinion is that the architecture, everything from the
 software to the metal
  chassis is superb to the grandstream.   The SPA8000
 has a fan built in for
  cooling and a 25 pin amphenol connector which makes
 connection to a 66block
  a breeze.We plan to use them as customer CPE as we
 further deploy our
  FTTx service offering...
 
  I'm hoping that linksys will release a 24 port
 version to compete with
  Grandstream's offering there...
 
  JC



  

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Re: [asterisk-users] Interesting observation

[Brent Vrieze]

I investigated Charter for our business phone systems and asked many of 
these questions of the sales person.  I was told they have a dedicated 
part of the bandwidth available that is used just for phone traffic. 

I could break out my college networking book and get you the frequency 
break down as far as what is used for IP and what is used for TV and why 
the upload and down load speeds are asymmetrical if I was motivated but 
I am not so you will have to take this for what it is worth. 

As cable is not a point to point system (cable is shared bandwidth for 
all users on that cable) that means all phone users will be using the 
same piece of spectrum on that cable.  This means that too many phone 
calls on that line at the same time could affect a Charter phone call. 

I do not know if they use analog or digital signals for the phones but 
if we use the cell phone system as an example they took down all analog 
towers because they could service more phones on the same bandwidth with 
digital.  I would assume that would hold true for the spectrum on a 
cable as well.  I would also find it hard to believe that they would not 
use off the shelf technology.

That being said my brothers in-laws are using it and are having no 
problems what so ever.



David Gibbons wrote:
 snip
 My understanding is that Charter 'telephone' doesn't use IP at all but
 rather uses some additional frequency spectrum on their cable network.
 Hence, the reason why faxing with their service is reliable unlike other
 providers who are *actually* using VoIP.
 /snip

 I think what you're referring to is the general hesitance of the cable 
 providers to call their phone service VOIP service. VOIP still has a negative 
 connotation with most regular folks, so they don't want to negative PR.

 I'm don't have any facts, but I'll bet you a penny that they don't have a 
 proprietary system using something /OTHER/ than IP to send encapsulated voice 
 over 'additional frequency spectrum'. That would be prohibitively expensive 
 to develop and pointless from a technical standpoint, given that IP telephony 
 is already set to deploy and relatively mature.

 The reliability of faxing is based soley on network jitter and latency and 
 codec compression. I've found that taking the compression out of the mix 
 (using g.711 ulaw) and controlling the jitter and latency (something that's 
 easy to do on a private network like theirs with QOS) causes faxing to be 
 pretty darn reliable.

 --Dave

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-- 
Brent T. Vrieze
CIM Automation
Softare Engineer
507-216-0465


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Re: [asterisk-users] Interesting observation

[J. Oquendo]

Digital Phone Service is a Fancy Marketing term Meaning Expensive VoIP
http://ezinearticles.com/?Digital-Phone-Service-is-a-Marketing-Term-for-Relabled,-Expensive-VoIPid=262018

Pure VoIP vs. Telephone and Cable VoIP
http://www.tmcnet.com/news/2006/08/16/1809766.htm

A telephone call over IP is what it is... Voice over IP.


=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
J. Oquendo
SGFA, SGFE, C|EH, CNDA, CHFI, OSCP

Enough research will tend to support your
conclusions. - Arthur Bloch

A conclusion is the place where you got
tired of thinking - Arthur Bloch

227C 5D35 7DCB 0893 95AA  4771 1DCE 1FD1 5CCD 6B5E
http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x5CCD6B5E


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[asterisk-users] looking for Asterisk experts

[Meftah Tayeb]

hi my friend,
i have to start a new company to provide asterisk Installation / 
configuration to Small / mediom business

i'm looking for a asterisk expert to start with me
salary: 50%
i have a online store and is ready to use
please Call Me for mor informations:
Make a Sip Call sip:vsdev...@iptel.org
phone: 00213550055636
irc nick name (FreeNode): DelphiWorld
Yahoo: vsdev2006
thanks
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Re: [asterisk-users] Interesting observation

[David Gibbons]

snip
I'd be willing to bet *TWO* pennies that you're correct. I certainly was not 
coming into the conversation as an expert, just stating what I'd read/heard of 
their service... hence the My understanding is that... beginning to the 
email. :-)
/snip

Fair enough. I get worked up when I hear the cable companies calling their 
phone service anything other than VOIP :).

I'm going to hold off on going on a 2-page rant about the cable companies, 
their false advertising, awful performance, sub-par quality and terrible 
customer service. Then again, I heard Verizon has been known to burn down your 
house when they install FiOS... Yikes!

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Re: [asterisk-users] looking for Asterisk experts

[Alex Balashov]

One problem to overcome is that your competitors are:

1) Literate.

2) Post to the right mailing lists.

Meftah Tayeb wrote:

 hi my friend,
 i have to start a new company to provide asterisk Installation / 
 configuration to Small / mediom business
 i'm looking for a asterisk expert to start with me
 salary: 50%
 i have a online store and is ready to use
 please Call Me for mor informations:
 Make a Sip Call sip:vsdev...@iptel.org
 phone: 00213550055636
 irc nick name (FreeNode): DelphiWorld
 Yahoo: vsdev2006
 thanks
 
 
 
 
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-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (678) 237-1775

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Re: [asterisk-users] looking for Asterisk experts

[David Gibbons]

snip
One problem to overcome is that your competitors are:

1) Literate.

2) Post to the right mailing lists.

Meftah Tayeb wrote:
/snip

Ha ha ha ha.

So, you're saying you don't want the job?

LOL.

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Re: [asterisk-users] looking for Asterisk experts

[Alex Balashov]

David Gibbons wrote:
 snip
 One problem to overcome is that your competitors are:
 
 1) Literate.
 
 2) Post to the right mailing lists.
 
 Meftah Tayeb wrote:
 /snip
 
 Ha ha ha ha.
 
 So, you're saying you don't want the job?
 
 LOL.

Well, actually, it would've been more proper and literacy-affirming to say:

One problem to overcome is that your competitors:

1) Are literate.

2) Post to the right mailing lists.

But, no, I think I'll pass.

-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (678) 237-1775

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Re: [asterisk-users] looking for Asterisk experts

[Doug Lytle]

Alex Balashov wrote:
 One problem to overcome is that your competitors are:

 1) Literate.

 2) Post to the right mailing lists.

   

That'd be 2 me thinks.

Doug


-- 
 
Ben Franklin quote:

Those who would give up Essential Liberty to purchase a little Temporary 
Safety, deserve neither Liberty nor Safety.


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[asterisk-users] Problems With Playback of Audio On SIP Only System

[Brian Alexander]

I have been installing Asterisk as a SIP only system (no Digium Hardware)
for demonstration purposes. SIP users can connect to menus and voicemail
fine but the audio quality is terrible. The stock voicemail problems are bad
but basically understandable - voice menus recorded through the
asterisk-gui-2.0 are difficult to even understand.

The phone I am testing with is a Polycom SountPoint IP 430 SIP. I have
configured the phone for ulaw to be it primary codec and set disallow all
and allow ulaw in the users.conf.

When that did not work I guessed that something was wrong with dahdi_dummy
but dahdi_test is showing results around 99.987%.

Here are the details of what software I have been using:
asterisk-1.4 (r168975)
dahdi-linux-complete 2.1.0 (r 5662)
asterisk-gui-2.0 (r4446)

The linux kernel is 2.6.24.6 built with 1000 Hz timer.

Thank you for your help, I am a stumped.

-Brian
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Re: [asterisk-users] Problems With Playback of Audio On SIP Only System

[Mark Michelson]

Brian Alexander wrote:
 I have been installing Asterisk as a SIP only system (no Digium 
 Hardware) for demonstration purposes. SIP users can connect to menus and 
 voicemail fine but the audio quality is terrible. The stock voicemail 
 problems are bad but basically understandable - voice menus recorded 
 through the asterisk-gui-2.0 are difficult to even understand.
 
 The phone I am testing with is a Polycom SountPoint IP 430 SIP. I have 
 configured the phone for ulaw to be it primary codec and set disallow 
 all and allow ulaw in the users.conf.
 
 When that did not work I guessed that something was wrong with 
 dahdi_dummy but dahdi_test is showing results around 99.987%.
 
 Here are the details of what software I have been using:
 asterisk-1.4 (r168975)
 dahdi-linux-complete 2.1.0 (r 5662)
 asterisk-gui-2.0 (r4446)
 
 The linux kernel is 2.6.24.6 built with 1000 Hz timer.
 
 Thank you for your help, I am a stumped.
 
 -Brian

If you are using gsm prompts and gcc version 4.2 or higher, then you may be 
experiencing the optimizer bug that gcc has with gsm audio. The workarounds for 
this are to use a different format for sounds or to set the DONT_OPTIMIZE flag 
in menuselect. If you want an optimized build and gsm formatted sounds, then 
you 
could always attempt downgrading your gcc version to 4.1 or earlier.

Mark Michelson

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Re: [asterisk-users] Asterisk Appliance

[Lincoln King-Cliby]

From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] On Behalf Of Bob Pierce 
[pier...@westmancom.com]
Sent: Tuesday, January 13, 2009 5:36 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Asterisk Appliance
 
 I'm looking for some info on the Asterisk Appliance.
 
 I understand it has a gui, but can I still do all the dialplan config
 that I'm used of doing by hand outside of the gui? If I really wanted
 to, could I even ignore that the device has a gui and do all my config
 in the files? I guess I'm just wondering if it will be as flexible as a
 'vanilla' asterisk install from source on a linux system.
 
 Also, from those who are using these devices, what has your experience
 been? Are they stable? Do they seem to have enough horsepower and
 storage space for an SMB with up to 50 phones? Some older specs stated
 they would be appropriate for businesses with 2-50 users, while the
 current spec on the Digium site states they are appropriate for 2-20
 users.

Sorry for the delay in responding -- I'm just now having time to catch up on a 
month of the list, but it doesn't appear anyone else has responded. 

We have an Asterisk Appliance in a remote office (and I have one at home) and 
it's a mistake we would not make again. Concept wise it's a nice idea, but 
using Asterisk built from source is so much easier, more flexible, and less 
stressful. I guess our biggest issue with the appliance isn't hardware but 
administrative: While I found no reference to it before we purchased the 
Appliances (maybe I didn't look hard enough, maybe it's not documented anywere 
that's readily findable) Digium doesn't support any confiuguration not 
generated using the GUI wizard... Digium doesn't support FTPing files onto or 
off of the appliance, Digium doesn't support... 

Stability wise I don't have any complaints, it's never crashed on us except 
when I tried FTPing config files off so I could edit them. We only have two 
full-time users at the remote office, and I have only me at home (6 total 
extentions at the office for visiting staff, 7 extensions at home). There is a 
wierd FXO caller ID issue that we're fighting on one of the POTS lines, but 
Digium essentially refused to support it because we aren't running anything 
close to what the GUI could build for us automagically (unified dialplan 
across three sites, either site can call out on the other site's POTS lines, 
call queuing that crosses sites, feature codes to force calls to extensions 
direct-to-voicemail (no ring ever) and ring-for-eternity (no voicemail ever, 
etc.) [in fairness it looks like its a CO issue because the problem follows the 
line, the other POTS lines don't exhibit the issue, though all lines work with 
a standard caller ID box]. 

I'm weary of making changes to that box because while you can edit config files 
through the GUI, if you aren't very careful it seems like parts of a context 
will rearange itself (e.g. part of extension s will end up in the middle of 
something completely different).

We tried to upgrade the firmware once and it screwed things up to the point 
where for a day the site had no telephone service (oh, suprise! If you have any 
custom stuff in extension.conf and upgrade the firmware it will spin off into 
an infinite loop orbit until you factory default the thing... and once we got 
control of the infinite loop issue we couldn't get the config that had been 
working just beautifully to work at all, so we punted and rolled back to the 
previous FW version) 

Basically, my experience with the appliance it it's a beautiful little box and 
if they'd just ditch the GUI and give a moderately user friendly commandline 
text editor I'd be on it in a second, but with the appliance in it's current 
state it took me less time to 

* Unbox a Dell PowerEdge 1950
* Install AEX804E card
* Rack a Dell PowerEdge 1950
* Install server version of Linux from Distro CD 
* Download Aserisk and Zaptel sources, compile, configure, and install
* Build Dialplan by hand from scratch
(and I think, not including labor costs this was also close to the same cost, 
if not cheaper than the appliance) 

Then the amount of time I've spent trying to get the Appliance to do what we 
want -- and what we want for the thing really isn't that complicated (really 
just Take POTS lines in, handle local switching for a handful of extensions so 
that that traffic doesn't wind up on our WAN/VPN connecions, and act as a 
SSU-- we aren't even doing queues on it!). 

Plus making changes to the Dell version of things is also quicker and less 
nerve wracking..

As you can probably tell I don't care much for the Appliance based on our 
understanding of what it was. On the other hand, though, if all you have is one 
site and just need a basic SOHO PBX it's a decent contender... the support more 
than anything is what's left a bad taste in my mouth.

Re: [asterisk-users] Problems With Playback of Audio On SIP Only System

[Brian Alexander]

Mark,

Thanks - that was the problem I was having. Is there somewhere I could have
looked to have discovered the problem on my own? I would never have guessed
that on my own and my searches had not found it either.

Thanks again,
-Brian


On Mon, Jan 19, 2009 at 7:00 PM, Mark Michelson mmichel...@digium.comwrote:

 Brian Alexander wrote:
  I have been installing Asterisk as a SIP only system (no Digium
  Hardware) for demonstration purposes. SIP users can connect to menus and
  voicemail fine but the audio quality is terrible. The stock voicemail
  problems are bad but basically understandable - voice menus recorded
  through the asterisk-gui-2.0 are difficult to even understand.
 
  The phone I am testing with is a Polycom SountPoint IP 430 SIP. I have
  configured the phone for ulaw to be it primary codec and set disallow
  all and allow ulaw in the users.conf.
 
  When that did not work I guessed that something was wrong with
  dahdi_dummy but dahdi_test is showing results around 99.987%.
 
  Here are the details of what software I have been using:
  asterisk-1.4 (r168975)
  dahdi-linux-complete 2.1.0 (r 5662)
  asterisk-gui-2.0 (r4446)
 
  The linux kernel is 2.6.24.6 built with 1000 Hz timer.
 
  Thank you for your help, I am a stumped.
 
  -Brian

 If you are using gsm prompts and gcc version 4.2 or higher, then you may be
 experiencing the optimizer bug that gcc has with gsm audio. The workarounds
 for
 this are to use a different format for sounds or to set the DONT_OPTIMIZE
 flag
 in menuselect. If you want an optimized build and gsm formatted sounds,
 then you
 could always attempt downgrading your gcc version to 4.1 or earlier.

 Mark Michelson

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Re: [asterisk-users] Call file in the future

[Steve Edwards]

On Mon, 19 Jan 2009, didier.cuffaut wrote:

 2) From my first post, are these lines OK or wrong? (syntax error?)

  tmsp = the delay in future.. say 100 seconds
  exten= ra,n,System(NOW='date %S')
  exten= ra,n,System(let NOW=$NOW+$tmsp)
  exten= ra,n,System(TOUCH_TMSP='date -d 1970-01-01 $NOW sec GMT+1 
 +%Y%m%d%H%M. %S)   NOTE THE 'M. %S'

  *

  or this way ?

  exten= ra,n,Set(touchtime=$[${EPOCH} + ${tmsp}])
  exten= ra,n,Set(TOUCH_TMSP=${STRFM(${touchtime},GMT+1,%C%y%m%d%H%M%S)

  *

Each invocation of system() executes a separate process. The environment 
variables do not survive across processes. This method will not work.

Setting a channel variable and then passing it will work.

Your choices are to use system() or agi(). I'm leaning towards system() 
because the script/executable does not interact with Asterisk and may have 
value to you as a stand-alone command line utility.

You can write either in whatever language you are comfortable with. My 
sharpest tool is C but if execution speed is not important any scripting 
language (like shell) will do.

I'm a big fan of the getopt facility as it does all the nasty command line 
parsing for you so your utilities have a consistent, self-documenting look 
and feel. I even use it when I write AGIs. A year from now, which would 
you rather re-discover in your dialplan:

exten = 
s,n,agi(schedule-future-call,--archive,--max-retries=2,--offset=${TMSP},--retry-time=60,--wait-time=20)

or

exten = s,n,agi(schedule-future-call,${TMSP},60,,,20,a,,2)

I think I would pass the offset rather than the absolute. I don't like to 
clutter up my dialplans too much.

Thanks in advance,

Steve Edwards  sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000

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Re: [asterisk-users] Description of Zaptel/DAHDI E1 alarms

[Lyle Giese]

Lukas Rypl wrote:
  Hello,

  I am missing any description of zaptel/DAHDI alarms. The TE200 series
 user manual contains only a description of LEDs states. These alarms
 states are visible in zttool/dahditool or in astersick CLI (zap show
 status) and I wonder what is the real meaning of these alarms for E1
 channel.

 Possible alarm states (based on zaptel.h 1.2):
  1. No alarms
  2. Recovering from alarm
  3. In loopback (local loopback or far end?)
  4. Yellow Alarm (is it only Far end Loss of Frame?)
  5. Red Alarm (Loss of Signal?)
  6. Blue Alarm (AIS?)
  7. Not Open


  Thank you for any help.

  Lukas Rypl


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Notes:
3 In loopback means I have been asked locally to provide a loopback of
some sort to somebody.

4 Yellow means the far end does not like the signal received for what
ever reason and the far end is trying to tell me the circuit is broken.

5 Red means I don't like the signal received. Could be framing issues,
could be CRC errors, could be no signal, could be ?

6 Blue alarm means I am receiving AIS or all ones signal, can be framed
or unframed.

Lyle Giese
LCR Computer Services, Inc.



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Re: [asterisk-users] [somewhat OT] seeking ideas/input for my thesis

[John Todd]

On Jan 19, 2009, at 12:35 PM, sp4rc wrote:

 Hello VoIP guys

 Sorry for being somewhat off-topic. At the moment I am studying
 informatics in the seventh semester and I need to start thinking about
 my thesis. As I am very interested in VoIP technologies I thought  
 about
 picking this as my main topic. So far I have only little experience in
 this area. I have been fiddling around with siproxd and pfSense and  
 have
 red the one or the other packet dump containing SIP and RTP traffic,  
 had
 a look into codecs, STUN, etc... but very cursorily, and that's the
 reason why I am quite unsure on which track to go. I think I am quite
 familiar with many network protocols and devices... so here comes the
 question of the questions:

 What would be a great project for my thesis to work on in the VoIP
 field? What are topics that still need special development? The time
 frame should be around 300 hours but don't take this value too
 seriously...

 An idea: contact synchronisation via SIP
 Are there any (working or concept) extensions on using SIP to  
 synchronize contacts
 in the way icq does it? (server-side contacts)

 Any ideas are welcome!
 /sp4rc


I suppose there are a lot of questions here, actually, since this is a  
fairly broad topic area you've mentioned.

  - are you looking to write code to solve a problem?
  - are you looking for a particularly vexing problem about which to  
write an analysis paper but write no code?
  - in what areas have you done work already?  Signal analysis? Packet  
protocols? Hotel management?  (the last one is facetious but actually  
is not entirely non-relevant - Asterisk is lacking a good open-source  
SMDR interface.)


So here are some projects you might look into:

  - Work with Kristian Kielhofner and make a better signal analysis  
engine for his Recqual system
  - SMDR for Asterisk 
(http://lists.digium.com/pipermail/asterisk-users/2004-June/042854.html 
)
  - steganographic audio multiplexing (http://stegano.net/)
  - audio encryption/decryption codecs (analog PSTN compatible)
  - RTP multiplexing for bandwidth savings (see 
http://lists.digium.com/pipermail/asterisk-dev/2008-December/thread.html#35814)
  - open-source ZRTP implementation

There's a start.  Asterisk is a good platform for testing lots of new  
ideas.  Let us know what you might be interested in, and I'm sure  
there will be good comments from the crowd.

JT


---
John Todd   email:jt...@digium.com
Digium, Inc. | Asterisk Open Source Community Director
445 Jan Davis Drive NW -  Huntsville AL 35806  -   USA
direct: +1-256-428-6083 http://www.digium.com/




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Re: [asterisk-users] Fring and Asterisk

[D Tucny]

2009/1/20 Olivier oza-4...@myamail.com

 Hi,

 Is anyone using Fring as a SIP client to an Asterisk server ?


Yes, testing it...



 A prospective customer of mine is asking to integrate its iphones with an
 Asterisk server and after googling, I still have some unanswered questions :

 1. Which codecs are available when calling from fring ?


I believe it offers GSM, ilbc, ulaw and alaw...



 2. Is it easy and natural to change your presence status (available, busy,
 ...) with Fring or will users prefer to use another software (bundled with
 iPhones) or to do nothing at all ?


 3. Is it possible to add custom presence status in Fring client ?


On the version running on my nokia phone, there seems to be no obvious way
to change status...



 4. Is it possible and recommended to limit Fring usage to WiFi presence ?


The options on the version I have are: Wifi first, 3G/GPRS first, Wifi only,
3G/GPRS only, Always ask... So, it is possible... It does work over GPRS,
but quality is noticably lower than over Wifi...



 5. Would fring replies to Qualify messages ?


Yes

One thing to note about fring, the device establishes a connection using
fring's proprietary protocols to fring servers, fring then establishes SIP
connections from those servers... So, even if connected to the office Wifi
connection, you could experience connectivity issues or high latency as a
result of a potentially long path involved for the traffic to travel...

d
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Re: [asterisk-users] Need help registering Cisco 7960 Phones on Asterisk

[D Tucny]

2009/1/20 Zeeshan Zakaria zisha...@gmail.com

 Hi everyone,

 I googled this followed the instructions, but it hasn't work for me yet.

 I have universal setting in SIPDefault.cnf and phone specific settings in
 SIPXX.cnf. But it doesn't get registered.

 I need to register it on two different asterisk boxes. So my
 SIPXX.cnf looks like this:

 phone_label: Zeeshan A Zakaria

 line1_name: 523
 line1_displayname: Zeeshan A Zakaria
 line1_authname: 523
 line1_password: 523
 line1_shortname: x523

 line2_name: 523
 line2_displayname: Zeeshan
 line2_authname: 523
 line2_password: 523
 line2_shortname: x523

 line3_name: 224
 line3_displayname: Zeeshan
 line3_authname: 224
 line3_password: 224
 line3_shortname: x224

 SIPDefault.cnf contains default settings along with proxy info like this:

 proxy1_address: xxx.xxx.xxx.xxx
 proxy1_port: 5060

 proxy2_address: xxx.xxx.xxx.xxx
 proxy2_port: 5060

 proxy3_address: xxx.xxx.xxx.xxx
 proxy3_port: 5060


 Same settings work fine from Grandstream phone, and X-lite. What am I
 missing on this Cisco phone configuration?


That all looks fine, though I don't think the ports need quotes...

You do have
proxy_register: 1
too don't you?

If so, check the debug output from sip set debug on, or, sip set debug peer
224 to see if it's reaching the server but not authenticating etc... Also,
you can log into the phone using telnet to check status and restart
registration (and many more things)

d
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Re: [asterisk-users] Fring and Asterisk

[John Todd]

On Jan 19, 2009, at 6:29 PM, D Tucny wrote:

 2009/1/20 Olivier oza-4...@myamail.com
 Hi,

 Is anyone using Fring as a SIP client to an Asterisk server ?

 Yes, testing it...


 A prospective customer of mine is asking to integrate its iphones  
 with an Asterisk server and after googling, I still have some  
 unanswered questions :

 1. Which codecs are available when calling from fring ?

 I believe it offers GSM, ilbc, ulaw and alaw...


 2. Is it easy and natural to change your presence status (available,  
 busy, ...) with Fring or will users prefer to use another software  
 (bundled with iPhones) or to do nothing at all ?

 3. Is it possible to add custom presence status in Fring client ?

 On the version running on my nokia phone, there seems to be no  
 obvious way to change status...


 4. Is it possible and recommended to limit Fring usage to WiFi  
 presence ?

 The options on the version I have are: Wifi first, 3G/GPRS first,  
 Wifi only, 3G/GPRS only, Always ask... So, it is possible... It does  
 work over GPRS, but quality is noticably lower than over Wifi...


 5. Would fring replies to Qualify messages ?

 Yes

 One thing to note about fring, the device establishes a connection  
 using fring's proprietary protocols to fring servers, fring then  
 establishes SIP connections from those servers... So, even if  
 connected to the office Wifi connection, you could experience  
 connectivity issues or high latency as a result of a potentially  
 long path involved for the traffic to travel...

 d


Yes, it's disappointing that Fring doesn't release the media.  I tried  
it a few times, but the latency was unacceptable since the hairpin  
apparently has very long legs from a millisecond-delay perspective.

I had it working once, but now for some reason I get authentication  
errors (repeated REGISTERs and they don't seem to see my replies) and  
I don't really have the time to work it out.  Anyone having similar  
problems or is this my local problem?

JT



---
John Todd   email:jt...@digium.com
Digium, Inc. | Asterisk Open Source Community Director
445 Jan Davis Drive NW -  Huntsville AL 35806  -   USA
direct: +1-256-428-6083 http://www.digium.com/




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Re: [asterisk-users] Fring and Asterisk

[D Tucny]

2009/1/20 John Todd jt...@digium.com


 On Jan 19, 2009, at 6:29 PM, D Tucny wrote:

  2009/1/20 Olivier oza-4...@myamail.com
  Hi,
 
  Is anyone using Fring as a SIP client to an Asterisk server ?
 
  Yes, testing it...
 
 
  A prospective customer of mine is asking to integrate its iphones
  with an Asterisk server and after googling, I still have some
  unanswered questions :
 
  1. Which codecs are available when calling from fring ?
 
  I believe it offers GSM, ilbc, ulaw and alaw...
 
 
  2. Is it easy and natural to change your presence status (available,
  busy, ...) with Fring or will users prefer to use another software
  (bundled with iPhones) or to do nothing at all ?
 
  3. Is it possible to add custom presence status in Fring client ?
 
  On the version running on my nokia phone, there seems to be no
  obvious way to change status...
 
 
  4. Is it possible and recommended to limit Fring usage to WiFi
  presence ?
 
  The options on the version I have are: Wifi first, 3G/GPRS first,
  Wifi only, 3G/GPRS only, Always ask... So, it is possible... It does
  work over GPRS, but quality is noticably lower than over Wifi...
 
 
  5. Would fring replies to Qualify messages ?
 
  Yes
 
  One thing to note about fring, the device establishes a connection
  using fring's proprietary protocols to fring servers, fring then
  establishes SIP connections from those servers... So, even if
  connected to the office Wifi connection, you could experience
  connectivity issues or high latency as a result of a potentially
  long path involved for the traffic to travel...
 
  d


 Yes, it's disappointing that Fring doesn't release the media.  I tried
 it a few times, but the latency was unacceptable since the hairpin
 apparently has very long legs from a millisecond-delay perspective.

 I had it working once, but now for some reason I get authentication
 errors (repeated REGISTERs and they don't seem to see my replies) and
 I don't really have the time to work it out.  Anyone having similar
 problems or is this my local problem?


I've just had a look, and can confirm that it is working here... even quite
a while after the fring client has been closed :/

Fring seems to send CR and LF twice in one packet every 10 seconds, not
relevant here, but, a bit odd...

One thing that might be relevant, it does use NAT, at least some of the
time, so asterisk needs to be aware of that... Address combinations I've
seen in the past hour...
91.151.216.12:52931
212.150.129.13:52924 - 172.16.8.15:52924
91.151.216.10:53219
91.151.216.4:52796

d
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Re: [asterisk-users] followme order field

[D Tucny]

2009/1/19 Thomas Stein thomas.st...@knowledgetools.de

 Hello.

 Does someone know what order field means in followme.conf? The Doku says:

 number= number to call[2nd #[3rd #]] [, timeout value in seconds [,
 order in follow-me] ]

 So an example would be:

 number= 123124125,10,?

 It would be nice if someone could enlighten me.


As I understand it, Follow-me can take multiple number lines in the
config... (I've not used follow-me though, this is just from looking at
app_followme.c)

number = 123124125,10,1 ; Would call 123, 124  125 first (all at the
same time as the same syntax in a Dial string would do), trying for 10
seconds
number = 126,10,3 ; Would call 126 third, trying for 10 seconds
number = 127,10,2 ; Would call 127 second, trying for 10 seconds

In the example I've provided, 127 would be called before 126 due to the use
of the order field... If the order was not specified then 126 would be
called before 127 due to the order of the lines in the file...

Again, I've not used it, so not 100% sure that's how it works, but, that's
what it looks like to me... Hope it helps...

d
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Re: [asterisk-users] Need help registering Cisco 7960 Phones on Asterisk

[Yehavi Bourvine]

 From my experience it won't register to the second box, only to the first
one. Why? god knows...

__Yehavi:

2009/1/20 D Tucny d...@tucny.com

  2009/1/20 Zeeshan Zakaria zisha...@gmail.com

 Hi everyone,

 I googled this followed the instructions, but it hasn't work for me yet.

 I have universal setting in SIPDefault.cnf and phone specific settings in
 SIPXX.cnf. But it doesn't get registered.

 I need to register it on two different asterisk boxes. So my
 SIPXX.cnf looks like this:

 phone_label: Zeeshan A Zakaria

 line1_name: 523
 line1_displayname: Zeeshan A Zakaria
 line1_authname: 523
 line1_password: 523
 line1_shortname: x523

 line2_name: 523
 line2_displayname: Zeeshan
 line2_authname: 523
 line2_password: 523
 line2_shortname: x523

 line3_name: 224
 line3_displayname: Zeeshan
 line3_authname: 224
 line3_password: 224
 line3_shortname: x224

 SIPDefault.cnf contains default settings along with proxy info like this:

 proxy1_address: xxx.xxx.xxx.xxx
 proxy1_port: 5060

 proxy2_address: xxx.xxx.xxx.xxx
 proxy2_port: 5060

 proxy3_address: xxx.xxx.xxx.xxx
 proxy3_port: 5060


 Same settings work fine from Grandstream phone, and X-lite. What am I
 missing on this Cisco phone configuration?


 That all looks fine, though I don't think the ports need quotes...

 You do have
 proxy_register: 1
 too don't you?

 If so, check the debug output from sip set debug on, or, sip set debug peer
 224 to see if it's reaching the server but not authenticating etc... Also,
 you can log into the phone using telnet to check status and restart
 registration (and many more things)

 d


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Re: [asterisk-users] Need help registering Cisco 7960 Phones on Asterisk

[D Tucny]

That's not my experience...
e.g.

SIP Phone show register

LINE REGISTRATION TABLE
Proxy Registration: ENABLED, state: REGISTERED
line  APR  state  timer   expires proxy:port
  ---  -  --  --
 
1 111  REGISTERED 115 98  192.168.1.1:5060
2 111  REGISTERED 115 98  192.168.1.12:5060
3 ...  NONE   0   0   undefined:0
4 ...  NONE   0   0   undefined:0
5 ...  NONE   0   0   undefined:0
6 ...  NONE   0   0   undefined:0
1-BU  111  REGISTERED 115 98  192.168.1.1:5060

Note: APR is Authenticated, Provisioned, Registered

I can see registers on both servers...

d

2009/1/20 Yehavi Bourvine yehavi.bourv...@gmail.com

 From my experience it won't register to the second box, only to the first
 one. Why? god knows...

 __Yehavi:

 2009/1/20 D Tucny d...@tucny.com

  2009/1/20 Zeeshan Zakaria zisha...@gmail.com

 Hi everyone,

 I googled this followed the instructions, but it hasn't work for me yet.

 I have universal setting in SIPDefault.cnf and phone specific settings in
 SIPXX.cnf. But it doesn't get registered.

 I need to register it on two different asterisk boxes. So my
 SIPXX.cnf looks like this:

 phone_label: Zeeshan A Zakaria

 line1_name: 523
 line1_displayname: Zeeshan A Zakaria
 line1_authname: 523
 line1_password: 523
 line1_shortname: x523

 line2_name: 523
 line2_displayname: Zeeshan
 line2_authname: 523
 line2_password: 523
 line2_shortname: x523

 line3_name: 224
 line3_displayname: Zeeshan
 line3_authname: 224
 line3_password: 224
 line3_shortname: x224

 SIPDefault.cnf contains default settings along with proxy info like this:

 proxy1_address: xxx.xxx.xxx.xxx
 proxy1_port: 5060

 proxy2_address: xxx.xxx.xxx.xxx
 proxy2_port: 5060

 proxy3_address: xxx.xxx.xxx.xxx
 proxy3_port: 5060


 Same settings work fine from Grandstream phone, and X-lite. What am I
 missing on this Cisco phone configuration?


 That all looks fine, though I don't think the ports need quotes...

 You do have
 proxy_register: 1
 too don't you?

 If so, check the debug output from sip set debug on, or, sip set debug
 peer 224 to see if it's reaching the server but not authenticating etc...
 Also, you can log into the phone using telnet to check status and restart
 registration (and many more things)

 d


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Re: [asterisk-users] followme order field

[Thomas Stein]

On Tuesday 20 January 2009 06:17:14 D Tucny wrote:

 number = 123124125,10,1 ; Would call 123, 124  125 first (all at the
 same time as the same syntax in a Dial string would do), trying for 10
 seconds
 number = 126,10,3 ; Would call 126 third, trying for 10 seconds
 number = 127,10,2 ; Would call 127 second, trying for 10 seconds

Ah, now i understand. Thank you.

cheers
t.

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