[asterisk-users] Ideasip

[David @ULC]

I use IdeaSip with IPKall.

How may channels are open when we use IdeaSip ?
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Re: [asterisk-users] Important security alert: update your dialplans now!

[Warren Selby]

On Tue, Feb 16, 2010 at 6:28 PM, meetmecall  wrote:

> I didn't know about the function but from what I understand from the "show
> function FILTER"  output it doesn't validate a string but it cleans the
> string from not allowed characters. So TRIM(1234567890,01243567&505) results
> in  01243567505. If the length of the output string is shorter then the
> input string the call setup should stop because not allowed characters were
> stripped.  With some extra lines TRIM() will do as good as the macro I
> guess.  You can add some lines so someone trying to perform  number
> injection will be connected with an answering machine and be  requested to
> leave name and phone number ;-)
>
>
> Erik
>
>
One thing FILTER() will allow though is variable length dial strings, which
are needed in some parts of the world (as evidenced by earlier posts in this
thread).


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Re: [asterisk-users] insecure=invite - not working for different dial plan

[Joseph]

Thanks for the input.
I know that order in extension.conf makes a difference but I did not know that 
it applies to sip.conf as well.
I would like to find this article you have mentioned on WIKI what should I look 
for :-/?

--
Joseph

On 02/16/10 18:50, uzzi wrote:
>Order of configurations does make a difference so you may want to try with
>the same order as the one that works. Saw it on the voip-info.org wiki
>somewhere, but can't get you the link at the moment.
>
>In general, if you will be authenticating based on IP, you should leave
>username/secret out.
>
>Some more advanced users can correct me if I'm wrong.
>
>
>
>
>On Sun, Feb 14, 2010 at 7:35 PM, Joseph  wrote:
>
>> I'm using  "insecure=invite" with two different dial plans, it it working
>> with one dial plan but not with the other.
>> What other parameters could influence "insecure=invite"
>>
>> In sip.conf below "insecure=invite" is working OK
>> [pstn-1270]
>> type=friend
>> secret=spa3k
>> username=voice-1270
>> mailbox=369
>> host=dynamic
>> insecure=invite
>> canreinvite=no
>> disallow=all
>> allow=ulaw
>> allow=alaw
>> nat=no
>> context=incoming
>> callgroup=1
>> pickupgroup=1
>>
>>
>> In sip.conf below "insecure=invite" is NOT WORKING
>>
>> [pstn-]
>> type=friend
>> secret=256
>> insecure=invite
>> username=voice-
>> mailbox=622
>> context=incoming
>> host=dynamic
>> canreinvite=no
>> disallow=all
>> allow=ulaw
>> allow=alaw
>> nat=no
>> callgroup=1
>> pickupgroup=1
>>
>> Both dial plan loaded on the same asterisk using the same Audiocodes MP-114
>> What other variable would influence operation of  "insecure=invite" ?
>>
>> With the dial plan that  "insecure=invite" is not working, asterisk logs
>> show:
>> "... username mismatch, have <4>, digest has 
>> handle_request_invite: Failed to authenticate user "KMIEC J"
>>
>> --
>> Joseph

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Re: [asterisk-users] Asterisk listens on all NICs

[Landy Landy]

> See http://www.voip-info.org/wiki/index.php?page=Asterisk+config+sip.conf
> 
> 
> 
> Search for "bindaddr."
> Or "udpbindaddr" for 1.6.2+...also,
> "tcpbindaddr", "tlsbindaddr" if you plan
> on adding TCP/TLS SIP support to asterisk.
> 


Thanks to everyone who replied for clarifying.


  

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Re: [asterisk-users] Important security alert: update your dialplans now!

[meetmecall]

I didn't know about the function but from what I understand from the  
"show function FILTER"  output it doesn't validate a string but it  
cleans the string from not allowed characters. So  
TRIM(1234567890,01243567&505) results in  01243567505. If the length  
of the output string is shorter then the input string the call setup  
should stop because not allowed characters were stripped.  With some  
extra lines TRIM() will do as good as the macro I guess.  You can add  
some lines so someone trying to perform  number injection will be  
connected with an answering machine and be  requested to leave name  
and phone number ;-)



Erik


On 17 feb 2010, at 00:41, Warren Selby wrote:

On Tue, Feb 16, 2010 at 4:38 PM, meetmecall   
wrote:


Doesn't the built-in function FILTER() already do this?

*CLI> core show function FILTER
*CLI>

  -= Info about function 'FILTER' =-

[Synopsis]
Filter the string to include only the allowed characters

[Description]
Permits all characters listed in ,  filtering all  
others outs.
In addition to literally listing the characters,  you may also use  
ranges

of characters (delimited by a '-'
Hexadecimal characters started with a '\x'(i.e. \x20)
Octal characters started with a '\0' (i.e. \040)
Also '\t','\n' and '\r' are recognized.
NOTE: If you want the '-' character it needs to be prefixed with a   
'\'


[Syntax]
FILTER(allowed-chars,string)

[Arguments]
Not available

[See Also]
Not available



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Re: [asterisk-users] insecure=invite - not working for different dial plan

[uzzi]

Order of configurations does make a difference so you may want to try with
the same order as the one that works. Saw it on the voip-info.org wiki
somewhere, but can't get you the link at the moment.

In general, if you will be authenticating based on IP, you should leave
username/secret out.

Some more advanced users can correct me if I'm wrong.




On Sun, Feb 14, 2010 at 7:35 PM, Joseph  wrote:

> I'm using  "insecure=invite" with two different dial plans, it it working
> with one dial plan but not with the other.
> What other parameters could influence "insecure=invite"
>
> In sip.conf below "insecure=invite" is working OK
> [pstn-1270]
> type=friend
> secret=spa3k
> username=voice-1270
> mailbox=369
> host=dynamic
> insecure=invite
> canreinvite=no
> disallow=all
> allow=ulaw
> allow=alaw
> nat=no
> context=incoming
> callgroup=1
> pickupgroup=1
>
>
> In sip.conf below "insecure=invite" is NOT WORKING
>
> [pstn-]
> type=friend
> secret=256
> insecure=invite
> username=voice-
> mailbox=622
> context=incoming
> host=dynamic
> canreinvite=no
> disallow=all
> allow=ulaw
> allow=alaw
> nat=no
> callgroup=1
> pickupgroup=1
>
> Both dial plan loaded on the same asterisk using the same Audiocodes MP-114
> What other variable would influence operation of  "insecure=invite" ?
>
> With the dial plan that  "insecure=invite" is not working, asterisk logs
> show:
> "... username mismatch, have <4>, digest has 
> handle_request_invite: Failed to authenticate user "KMIEC J"
>
> --
> Joseph
>
> --
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Re: [asterisk-users] Asterisk listens on all NICs

[Warren Selby]

On Tue, Feb 16, 2010 at 5:26 PM, Steve Edwards wrote:

>
> See http://www.voip-info.org/wiki/index.php?page=Asterisk+config+sip.conf
>
> Search for "bindaddr."
>
> --
> Thanks in advance,
> -
> Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
> Newline  Fax: +1-760-731-3000
>
>
Or "udpbindaddr" for 1.6.2+...also, "tcpbindaddr", "tlsbindaddr" if you plan
on adding TCP/TLS SIP support to asterisk.

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Re: [asterisk-users] Asterisk listens on all NICs

[Philipp von Klitzing]

Hi!

> I know I can do it with iptables and block incoming connections to
> ports 5060-5070 from the internet but, wondering if it can be
> confiruged in asterisk. 

Iptables would be the right place, though.

Still: Look at 'permit'/'deny' in sip.conf, or use dialplan magic to 
check on IP addresses (f.e. SIPCHANINFO, SIPPEER). 
And of course have a look at 'bindaddress' in sip.conf.

Philipp


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Re: [asterisk-users] Asterisk listens on all NICs

[Philipp von Klitzing]

> I am puzzled and how asterisk listens to calls or connections from
> clients. When I do a netstat -nat I don't see asterisk listening on port
> 5060.

That is because you need to do 'netstat -nau'. Only very recently 
Asterisk has learned to do SIP over TCP.

Philipp


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Re: [asterisk-users] Important security alert: update your dialplans now!

[Warren Selby]

On Tue, Feb 16, 2010 at 4:38 PM, meetmecall  wrote:

> I have read the posts about the security issue and from what I
> understand there should be a check to make sure that the characters
> used are actually allowed. I wrote a very straightforward and not so
> rocket science kind of macro that will do the job I guess. Just two
> parameters, one with the allowed characters and one with the string to
> check. The allowed characters can be stored in a global variable if
> this set has a global character. It is just one extra line in the
> dialplan and a little bit extra cpu load.
>

Doesn't the built-in function FILTER() already do this?

*CLI> core show function FILTER
*CLI>

  -= Info about function 'FILTER' =-

[Synopsis]
Filter the string to include only the allowed characters

[Description]
Permits all characters listed in ,  filtering all others
outs.
In addition to literally listing the characters,  you may also use ranges
of characters (delimited by a '-'
Hexadecimal characters started with a '\x'(i.e. \x20)
Octal characters started with a '\0' (i.e. \040)
Also '\t','\n' and '\r' are recognized.
NOTE: If you want the '-' character it needs to be prefixed with a  '\'

[Syntax]
FILTER(allowed-chars,string)

[Arguments]
Not available

[See Also]
Not available



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Re: [asterisk-users] Asterisk listens on all NICs

[Steve Edwards]

On Tue, 16 Feb 2010, Landy Landy wrote:

> I am puzzled and how asterisk listens to calls or connections from 
> clients. When I do a netstat -nat I don't see asterisk listening on port 
> 5060.

man netstat. See what -t means.

> Now, I'm testing a server with three network interfaces: two to the 
> internet doing load balancing and the other to our LAN. I would like 
> asterisk to only accept connections coming from our LAN but, can't find 
> where to configure this.
>
> I know I can do it with iptables and block incoming connections to ports 
> 5060-5070 from the internet but, wondering if it can be confiruged in 
> asterisk.

See http://www.voip-info.org/wiki/index.php?page=Asterisk+config+sip.conf

Search for "bindaddr."

-- 
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] Asterisk listens on all NICs

[C. Chad Wallace]

At 2:50 PM on 16 Feb 2010, Landy Landy wrote:

> Hello List.
> 
> I am puzzled and how asterisk listens to calls or connections from
> clients. When I do a netstat -nat I don't see asterisk listening on
> port 5060. Now, I'm testing a server with three network interfaces:
> two to the internet doing   load balancing and the other to our LAN.
> I would like asterisk to only accept connections coming from our LAN
> but, can't find where to configure this. 

Set bindaddr in sip.conf.

-- 

C. Chad Wallace, B.Sc.
The Lodging Company
http://www.lodgingcompany.com/
OpenPGP Public Key ID: 0x262208A0



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[asterisk-users] Asterisk listens on all NICs

[Landy Landy]

Hello List.

I am puzzled and how asterisk listens to calls or connections from clients. 
When I do a netstat -nat I don't see asterisk listening on port 5060. Now, I'm 
testing a server with three network interfaces: two to the internet doing   
load balancing and the other to our LAN. I would like asterisk to only accept 
connections coming from our LAN but, can't find where to configure this. 

I know I can do it with iptables and block incoming connections to ports 
5060-5070 from the internet but, wondering if it can be confiruged in asterisk.

Thanks.


  

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Re: [asterisk-users] Important security alert: update your dialplans now!

[Tommy Botten Jensen]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA512

I have tried to replicate this, but with no luck. If I use a SIP-client
that supports '&', I still get a reject from asterisk.

I am  writing a filter for lua (pbx_lua), but it's a bit hard when I
cannot reproduce and test this. I've tried with both 1.6.1-series and
1.6.2. Any ideas?

Thanks,

Tommy

Landy Landy skrev:
> I have this:
> 
> [menu]
> exten => _X.,1,answer()
> exten => _X.,2,wait(1)
> exten => _X.,n,GoTo(ivr,s,1)
> 
> 
> [default]
> include => record
> include => incoming
> include => menu
> 
> [local-dial]
> exten => _1XX,1,Verbose(". In local-dial context, dialing exten: ${EXTEN} 
> ."
> exten => _1XX,2,Dial(SIP/${EXTEN},20,tTmkKhHWw)
> exten => _1XX,n,voicemail(${EXTEN},u)
> exten => _1XX,n,Hangup()
> include => agents
> include => queue
> include => local-iax
> include => voicemail
> include => timeofday
> include => parkedcalls
> include => pickup
> include => to_client
> include => test-agi
> 
> include => menu
> 
> that goes to an ivr. Can this be a security bridge?
> 
> 
> 
> --- On Mon, 2/15/10, Tony Mountifield  wrote:
> 
>> From: Tony Mountifield 
>> Subject: Re: [asterisk-users] Important security alert: update your 
>> dialplans now!
>> To: asterisk-users@lists.digium.com
>> Date: Monday, February 15, 2010, 11:58 AM
>> In article <699ee941002150033t7c6e1be5xdba76cb0f68d5...@mail.gmail.com>,
>> Lenz Emilitri 
>> wrote:
>>> -=-=-=-=-=-
>>> -=-=-=-=-=-
>>>
>>> Or one could simply rewrite to:
>>>
>>> [incoming-from-voip]
>>> exten =>
>> XXX,1,Dial(${ext...@incoming-from-voip-old)
>>> exten =>
>> ,1,Dial(${ext...@incoming-from-voip-old)
>>> exten =>
>> X,1,Dial(${ext...@incoming-from-voip-old)
>>> exten =>
>> XX,1,Dial(${ext...@incoming-from-voip-old)
>>> [incoming-from-voip-old]
>>> exten => _X., 1, dial(SIP/${EXTEN})
>>>
>>> To avoid extensive rewriting and fix the current
>> issue.
>>> l.
>> Don't forget you still need the underscore to make X
>> magic:
>>
>> exten =>
>> _XXX,1,Dial(${ext...@incoming-from-voip-old)
>>
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=t0Wq
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Re: [asterisk-users] Important security alert: update your dialplans now!

[Landy Landy]

I have this:

[menu]
exten => _X.,1,answer()
exten => _X.,2,wait(1)
exten => _X.,n,GoTo(ivr,s,1)


[default]
include => record
include => incoming
include => menu

[local-dial]
exten => _1XX,1,Verbose(". In local-dial context, dialing exten: ${EXTEN} 
."
exten => _1XX,2,Dial(SIP/${EXTEN},20,tTmkKhHWw)
exten => _1XX,n,voicemail(${EXTEN},u)
exten => _1XX,n,Hangup()
include => agents
include => queue
include => local-iax
include => voicemail
include => timeofday
include => parkedcalls
include => pickup
include => to_client
include => test-agi

include => menu

that goes to an ivr. Can this be a security bridge?



--- On Mon, 2/15/10, Tony Mountifield  wrote:

> From: Tony Mountifield 
> Subject: Re: [asterisk-users] Important security alert: update your dialplans 
> now!
> To: asterisk-users@lists.digium.com
> Date: Monday, February 15, 2010, 11:58 AM
> In article <699ee941002150033t7c6e1be5xdba76cb0f68d5...@mail.gmail.com>,
> Lenz Emilitri 
> wrote:
> > -=-=-=-=-=-
> > -=-=-=-=-=-
> > 
> > Or one could simply rewrite to:
> > 
> > [incoming-from-voip]
> > exten =>
> XXX,1,Dial(${ext...@incoming-from-voip-old)
> > exten =>
> ,1,Dial(${ext...@incoming-from-voip-old)
> > exten =>
> X,1,Dial(${ext...@incoming-from-voip-old)
> > exten =>
> XX,1,Dial(${ext...@incoming-from-voip-old)
> > 
> > [incoming-from-voip-old]
> > exten => _X., 1, dial(SIP/${EXTEN})
> > 
> > To avoid extensive rewriting and fix the current
> issue.
> > l.
> 
> Don't forget you still need the underscore to make X
> magic:
> 
> exten =>
> _XXX,1,Dial(${ext...@incoming-from-voip-old)
> 
> etc.
> 
> Tony
> -- 
> Tony Mountifield
> Work: t...@softins.co.uk
> - http://www.softins.co.uk
> Play: t...@mountifield.org
> - http://tony.mountifield.org
> 
> -- 
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>http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [asterisk-users] Important security alert: update your dialplans now!

[meetmecall]

I have read the posts about the security issue and from what I  
understand there should be a check to make sure that the characters  
used are actually allowed. I wrote a very straightforward and not so  
rocket science kind of macro that will do the job I guess. Just two  
parameters, one with the allowed characters and one with the string to  
check. The allowed characters can be stored in a global variable if  
this set has a global character. It is just one extra line in the  
dialplan and a little bit extra cpu load.

It is used like this:



exten => _X.,1,Macro(alfa_num_check,1234567890,${EXTEN})
exten => _X.,n,Dial(SIP/315927xx/${EXTEN},40,t)
exten => _X.,n,Hangup()

If not allowed characters are used the call setup will end.

And the actual macro does just have 20 lines of Asterisk code. I'm  
sure there will be better solutions in the future but for the short  
time this might be useful. A clever scripter might be able to write a  
script that scans the dialplan line by line and adds the macro line  
above the _.,n,DIAL() lines and adjust the numbering of the line  
if needed.   I have done some testing and the code in this mail  
ishould work, at least it did on my Asterisk box.


Friendly regards,




Erik

[macro-alfa_num_check]

; written by  : erik de wild
; company  : Tripple-oYour Asterisk migration partner
; e-mail: info   at   tripple-o.nl
; date   : 16 februari 2010
;
; everybody is free to use, improve and modify this code
; as long as credits will be addressed, no costs are charged
; and this header stays with the code.
;
exten => s,1,NoOp(This is a macro to check the content of the ${EXTEN}  
variable to cope with the vulnerabilily of number injection)
exten => s,n,Set(ALLOWED_CHARACTERS=${ARG1})
exten => s,n,Set(STRING_TO_CHECK=${ARG2})
exten => s,n,Set(LEN_ALLOWED_CHARACTERS=${LEN(${ALLOWED_CHARACTERS})})
exten => s,n,Set(LEN_STRING_TO_CHECK=${LEN(${STRING_TO_CHECK})})
exten => s,n,Set(COUNTER=0)

exten => s,n(start_loop),SET(CHAR_TO_CHECK=${STRING_TO_CHECK:$ 
{COUNTER}:1})
exten => s,n,NoOp(Character to check is ${CHAR_TO_CHECK}. This  
Character should be in this list of characters -> ${ALLOWED_CHARACTERS})
exten => s,n,Set(COUNTER2=0)
exten => s,n,NoOp(character to check: ${CHAR_TO_CHECK}-> $ 
{ALLOWED_CHARACTERS:${COUNTER2}:1})

exten => s,n(start_loop2),GotoIf($["${CHAR_TO_CHECK}" = "$ 
{ALLOWED_CHARACTERS:${COUNTER2}:1}"]?exit_loop2) ;
exten => s,n,Set(COUNTER2=$[${COUNTER2} + 1])
exten => s,n,GotoIf($["${COUNTER2}" = "${LEN_ALLOWED_CHARACTERS}"]? 
hangup)
; if the character to control is not in the list of allowed   
characters then the setup of the call has to stop

exten => s,n,Goto(start_loop2)
exten => s,n(exit_loop2),NoOp(character checked is allowed, go for the  
next charachter!!)

exten => s,n,Set(COUNTER=$[${COUNTER} + 1])
exten => s,n,GotoIf($["${COUNTER}" = "${LEN_STRING_TO_CHECK}"]? 
macroexit)

exten => s,n,Goto(start_loop)

exten => s,n(hangup),NoOp(the checked character is not in the allowed  
charachter string. Call setup has to stop)
exten => s,n,Hangup()

exten => s,n(macroexit),MacroExit()




On 14 feb 2010, at 00:04, Olle E. Johansson wrote:

> Friends,
>
> Last week, Hans Petter Selansky alerted us of a potential security  
> issue in all releases of Asterisk. In fact, it doesn't involve the  
> code, but the most common way to construct dialplans. If you have  
> something like this in your Asterisk, you need to update your  
> dialplans:
>
> [incoming-from-voip]
> exten => _X., 1, dial(SIP/${EXTEN})
>
> Many VoIP protocols support a large character set, that may cause  
> harm in your dialplan
> 
>
> I've written an article about this on my blog, where my summary says:
>
> "Because of a conflict between allowed characters in the called  
> number or name in many VoIP protocols and the way Asterisk handles  
> channel variables, there is a security risk hidden in many dialplans  
> based on examples provided over the years by the Asterisk  
> developers, trainers and community. The primary risk is that by  
> using an ampersand in the dialstring, a user can access protected  
> resources or misuse the pbx services. However, this character is not  
> the only problem, as other characters may cause unexpected or  
> problematic behaviour."
>
> There will be an Asterisk Security Advisory document coming out from  
> Digium soon, as well as updated documentation and examples within  
> the Asterisk source code tree. I strongly advise everyone to follow  
> these and stay updated. (I have no access to the ASA system myself  
> and can't generate an official security alert).
>
> For more information about this issue and some code examples of what  
> I personally currently think are good ways to prevent misuse of your  
> services, please read my blog article at
>
> http://www.voip-forum.com/

[asterisk-users] Handling Segmentation Faults / Crashes

[Robert Grignon]

Just curious to know how most of you deal with segmentation faults /
core dumps

These are a few of the things I have seen in regards to people dealing
with random crashes:

1. Apply Base OS updates
2. Recompile with DEBUG_THREADS and DON'T_OPTIMIZE turned on and look
for the cause of the seg fault in GDB next time it happens
3. Upgrade to a late version of asterisk


I have been running version 1.4.26 for a while and it has been
relatively stable except for the random segfaults (once every few
weeks). No errors or anything obvious in the logs, Server vitals are
fine, etc... I am contemplating upgrading but I hate to knee jerk
because sometimes you can introduce new problems with upgrading...

My front-end servers are pretty busy throughout the day and usually have
about 40 concurrent calls (80 channels) going at a time...

Just looking for some "best practices" in regards to this.

Thanks,

Robert

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[asterisk-users] chan_sip.c: Disconnecting call 'SIP/302-b720dd78' for lack of RTP activity in 301 seconds

[Danny Dias]

Hello My friends,

Today my asterisk stop working and i could see the following messags in
/var/log/asterisk/messages at the time that asterisk stop working:

[Feb 16 13:23:40] NOTICE[8230] chan_sip.c: Peer '324' is now Reachable. (2ms
/ 2000ms)
[Feb 16 13:24:41] NOTICE[8230] chan_sip.c: Disconnecting call
'SIP/302-b720dd78' for lack of RTP activity in 301 seconds
[Feb 16 13:25:54] NOTICE[8230] chan_sip.c: Disconnecting call
'SIP/346-b764bb28' for lack of RTP activity in 301 seconds
[Feb 16 13:26:00] NOTICE[8230] chan_sip.c: Disconnecting call
'SIP/317-b7664db0' for lack of RTP activity in 302 seconds
[Feb 16 13:26:43] NOTICE[8230] chan_sip.c: Disconnecting call
'SIP/324-b76510c0' for lack of RTP activity in 301 seconds
[Feb 16 13:28:19] NOTICE[8230] chan_sip.c: Peer '324' is now Reachable. (1ms
/ 2000ms)

It's important to say that asterisk stop working, i can not make internal
calls, not even voicemail, or pstn...but if you make an asterisk -r you
access the Asterisk CLI, but after minutes the CLI freezes and you can't
input any command, so to solve the problem i have to restart the server

What is happening here my friends? what should i do?

Thanks in advance for your help
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Re: [asterisk-users] Stupid question: Why Cmd Dial andQueuehaven'tsame options?

[Danny Nicholas]

In concept, how I would do #3 is to make an AGI that has a timer and warn
intervals, like this

-  exten => 100,1,AGI(timer.agi,3600,300) to set a timer for 60
minutes and play a message at 5 minute intervals.

 

In a post last year, Jared Smith posted a snippet to timeout a call.   It
shouldn't be hard to google for it.

 

The agent in a queue can transfer a call if the queue is set up correctly.

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of didier.cuffaut
Sent: Tuesday, February 16, 2010 12:37 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Stupid question: Why Cmd Dial
andQueuehaven'tsame options?

 

OK Danny, but of course, i don't mean 'exactely the same options'.hum i
don't really understand why it's possible for the caller to continue in his
dialplan (= cmd Dial option g) and it's not possiblef or the callee (as
option 'F transering' on DIAL).

 

Reading the source code in app_queue.c, i think it's possible to
initiate/monitor some action after caller/callee are bridged (and some
comment on the beginning of this app: "Optional monitoring of calls, started
when call is answered"), i'm wrong ?

 

Please, could you be more specific about 'item 3' ? some 'barge in' action
with AGI ? i'm Beotian (sorry for grek people)

 

One more time best regards

- Original Message - 

From: Danny Nicholas   

To: 'Asterisk Users Mailing List - 
Non-Commercial Discussion' 

Sent: Tuesday, February 16, 2010 5:20 PM

Subject: Re: [asterisk-users] Stupid question: Why Cmd Dial and
Queuehaven'tsame options?

 

"callee" in queue-land is always an agent who picks up the call instead of
getting it transferred.  In some concepts, queue is more like a conference
than a transferred/dialed call.  Item 3 could be accomplished with an
AGI/AMI command.

 


  _  


From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of didier.cuffaut
Sent: Tuesday, February 16, 2010 9:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Stupid question: Why Cmd Dial and Queue
haven'tsame options?

 

I apologize, my English isn't better than the last year

 

So, 

- why Queue has some options like 'caller can continue in his dialplan'
(Dial has g and F options for callee/caller) and not he same option for
'callee' ?

- no option in queue to send message (for only calle/caller) when bridged as
Dial has

- idem to announce periodically 'X minutes before hangup' (in Dial, not in
Queue)

 

Probabely, i've missed something. perhaps  some workaround ?

 

Thank's a lot for your attention

 


  _  


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Re: [asterisk-users] Important security alert: update your dialplans now!

[Philipp von Klitzing]

Hi!

> But, there's no reason we can't add other matching chars for
> handy things. A = alpha chars Y = alphanum chars, G = Graphical chars,

Pretty please!

Philipp


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Re: [asterisk-users] Issue with trying to dial two different servers at the same time.

[C. Chad Wallace]

At 9:04 AM on 16 Feb 2010, Steve Anness wrote:

[...]
> exten => 12109,1,Dial(iax2/castle-rock/109&iax2/colo/17128,20)
> 
> This has one flaw, if for whatever reason his home phone isn't
> connected, like it isn't now, than when I call 12109 I get his home
> voicemail as it picks up right away.
> 
> How can I program this so that even if the voicemail on 17128 or 109
> pick up it still rings the other phone?

I would give app_followme [1] a shot.  It requires the callee to hit
"1" to connect.  That way, an answering machine would never make it
through... unless the callee puts the tone for a "1" in his voicemail
greeting. :-)

You might also consider AMD [2] (answering machine detection), but I
don't know much about it.


[1] http://www.voip-info.org/wiki/view/Asterisk+cmd+FollowMe
[2] http://www.voip-info.org/wiki/view/Asterisk+cmd+AMD


-- 

C. Chad Wallace, B.Sc.
The Lodging Company
http://www.lodgingcompany.com/
OpenPGP Public Key ID: 0x262208A0



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[asterisk-users] Voicemail IMAP storage enhancement

[Hoggins!]

Hello,

I have several voicemail accounts that share the same IMAP mailbox (same
imapuser), and I'd like to store their messages in separate IMAP
subfolders. I tried to set 'imapfolder' settings on a per voicemail
account basis, but it does not seem to work. All messages keep being
stored in the same INBOX folder.

Should I file a feature request on the asterisk-dev list, or is there a
way to do this ?

Of course, I need to store all these in one single IMAP mailbox. The
easy solution would be to have separate IMAP users, but I can't.

Here is a piece of my voicemail.conf file :


[default]

1234 => ,Radio
Montaillou,email_addre...@host.com,,imapuser=repondeur|imappassword=123456|tz=european
1235 => ,Jeu Radio
Montaillou,email_addre...@host.com,,imapfolder=jeu|imapuser=repondeur|imappassword=123456|tz=european



Thanks in advance,

Hoggins!
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Re: [asterisk-users] call is not going to wrong "context"

[Joseph]

Audiocodes is not distinguishing/identifying correctly which device calls comes 
through IN; I think it just take the fist identifier base on internal 
registration. So the call goes to a wrong context. 

--
Joseph


On 02/16/10 10:48, Joseph wrote:
>I've Audiocodes MP-114 registered per-endpoint (2x FXO / 2x FXS)  but when 
>call comes on pstn- it goes to context "fax-incoming"
>
>in sip.conf:
>
>[pstn-]
>type=friend
>context=incoming
>...
>
>[pstn-9998]
>type=friend
>context=fax-incoming
>...
>
>the device register per end point just fine, so it can find "secret=xxx" 
>correctly but why the call is not forwarded to correct context.
>It seems to some limitation in Audiocodes since I had no problem with calls 
>going to incorrect context when I was using "Sipura/Linksys" devices.
>
>--
>Joseph

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Re: [asterisk-users] Stupid question: Why Cmd Dial and Queuehaven'tsame options?

[didier.cuffaut]

OK Danny, but of course, i don't mean 'exactely the same options'.hum i 
don't really understand why it's possible for the caller to continue in his 
dialplan (= cmd Dial option g) and it's not possiblef or the callee (as option 
'F transering' on DIAL).

Reading the source code in app_queue.c, i think it's possible to 
initiate/monitor some action after caller/callee are bridged (and some comment 
on the beginning of this app: "Optional monitoring of calls, started when call 
is answered"), i'm wrong ?

Please, could you be more specific about 'item 3' ? some 'barge in' action with 
AGI ? i'm Beotian (sorry for grek people)

One more time best regards
  - Original Message - 
  From: Danny Nicholas 
  To: 'Asterisk Users Mailing List - Non-Commercial Discussion' 
  Sent: Tuesday, February 16, 2010 5:20 PM
  Subject: Re: [asterisk-users] Stupid question: Why Cmd Dial and 
Queuehaven'tsame options?


  "callee" in queue-land is always an agent who picks up the call instead of 
getting it transferred.  In some concepts, queue is more like a conference than 
a transferred/dialed call.  Item 3 could be accomplished with an AGI/AMI 
command.

   


--

  From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of didier.cuffaut
  Sent: Tuesday, February 16, 2010 9:51 AM
  To: Asterisk Users Mailing List - Non-Commercial Discussion
  Subject: [asterisk-users] Stupid question: Why Cmd Dial and Queue haven'tsame 
options?

   

  I apologize, my English isn't better than the last year

   

  So, 

  - why Queue has some options like 'caller can continue in his dialplan' (Dial 
has g and F options for callee/caller) and not he same option for 'callee' ?

  - no option in queue to send message (for only calle/caller) when bridged as 
Dial has

  - idem to announce periodically 'X minutes before hangup' (in Dial, not in 
Queue)

   

  Probabely, i've missed something. perhaps  some workaround ?

   

  Thank's a lot for your attention

   



--


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Re: [asterisk-users] CODECS: Best practice question: Avoid transcode when calling out?

[Philipp von Klitzing]

Hi Karl,

that's funny you are asking this, am also currently looking at how to 
solve the g722 codec negotiation riddle, in my particular case to play 
nicely together with a KonfTel 300 IP conference phone.

> In other words, incoming calls are easy since codecs are negotiated
> from least-known (the remote party) to most-known (my endpoint) and my
> codecs can simply be preferred accordingly to match the remote. 

Look at setting the channel variable_SIP_CODEC - however it might or 
might not work depending on your version of Asterisk, see for example:

  https://issues.asterisk.org/view.php?id=13243

Here's a dialplan snippet that might give you another hint or two.

exten => 123,1,NoOp(-- Inbound read: ${CHANNEL(audioreadformat)} --)
exten => 123,n,NoOp(-- Inbound native: ${CHANNEL(audionativeformat)} --)
exten => 123,n,Set(WIDEBAND=0)
exten => 123,n,Set(WIDEBAND=${REGEX("g722" 
${SIPPEER(${SIPCHANINFO(peername)}:codecs)})})
exten => 123,n,ExecIf($[${WIDEBAND} = 1]|Set|_SIP_CODEC=g722)
exten => 123,n,Dial(SIP/abc123)

Please note the SPACE between ${REGEX("g722" and ${SIPPEER

> Outbound calls seem harder.  Our endpoints always negotiate g.722 between
> themselves and Asterisk and then Asterisk must transcode to the preferred
> codec of the REMOTE party.  Not ideal.

Together with the g722 transcoding patch for Asterisk 1.4.17 it does not 
work out, unfortunately. Currently I cannot make a statement on a more 
recent 1.4 release.

g722 patch: http://users.netplex.net/~andrew/asterisk/#g722

Older patch that I use for 1.4.17: 
http://users.netplex.net/~andrew/asterisk/g722-20080110.patch.gz

However I can successfully employ setting _SIP_CODEC if in the example 
above instead of "Dial()" I do a "MusicOnHold()" - both with or without a 
preceeding ANSWER; without means early audio playing of the native g722 
encoded MOH file. My snom starts out with alaw, and then we switch to 
g722.

> Is there an elegant way to do this?

Consider the codec negotiation patch? I'd be interested to hear about 
your results!

  http://www.b2bua.org/wiki/AsteriskCodecNegotiationPatch
  https://issues.asterisk.org/view.php?id=4825

Philipp


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Re: [asterisk-users] Important security alert: update your dialplans now!

[Tzafrir Cohen]

On Tue, Feb 16, 2010 at 10:53:16AM -0700, Steve Murphy wrote:

> (is there some escape mech in the syntax to let you say \NA\NCY? I
> haven't checked). 

[N]A[N]CY . Or, if we have it your way, [N][A][N]C[Y]

> But, there's no reason we can't add other matching chars
> for handy things.  A = alpha chars Y = alphanum chars, G = Graphical chars,
> whatever.

-- 
   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] Cannot built kmod-dahdi-linux for PAE kvariant from SRPM

[Jason Parker]

stephen.hindma...@bt.com wrote:
> rpmbuild --bb ~/localrpms/SPECS/dahdi-linux-kmod.spec
> 
snip
> 
> error: Failed build dependencies:
> 
> kernel-devel = 2.6.18-164.11.1.el5 is needed by 
> dahdi-linux-kmod-2.2.1-1_centos5.2.6.18_164.11.1.el5.i386
> 

Add a --target=i686 to your rpmbuild line.

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Re: [asterisk-users] Important security alert: update your dialplans now!

[Steve Murphy]

On Tue, Feb 16, 2010 at 3:01 AM, Olle E. Johansson  wrote:

>
> 16 feb 2010 kl. 09.43 skrev Tzafrir Cohen:
>
> > On Mon, Feb 15, 2010 at 09:40:31AM -0700, Steve Murphy wrote:
> >> On Mon, Feb 15, 2010 at 8:25 AM, Lenz Emilitri 
> wrote:
> >>
> >>> Yes but in any case you can enter all of the strings that reasonably
> match
> >>> - even if you have variable-length numbers, you will be able to
> determine
> >>> that a valid number be between 5 and 15 characters - or likely 2 to 20,
> all
> >>> numbers. A number of 156 characters is very likely to be a problem.
> >>>
> >>
> >> This is probably a stupid idea, because it could only be implemented in
> >> trunk, and won't help with current implementations,
> >> and I suggested it a long time ago already when I did the fast pattern
> >> matching code, but I don't THINK it would be all that
> >> hard to offer SOME regex syntax in patterns to help reduce the impact of
> >> these kinds of problems.
> >>
> >> Like using:
> >>
> >> [incoming-from-voip]
> >> exten => _X\{7-10\},1,Dial(${ext...@incoming-from-voip-old)
> >>
> >> instead of :
> >>
> >> [incoming-from-voip]
> >> exten => XXX,1,Dial(${ext...@incoming-from-voip-old)
> >> exten => ,1,Dial(${ext...@incoming-from-voip-old)
> >> exten => X,1,Dial(${ext...@incoming-from-voip-old)
> >> exten => XX,1,Dial(${ext...@incoming-from-voip-old)
> >>
> >> I put the \'s in front of the {}'s because we probably wouldn't want to
> >> change the
> >> behavior of exact matching, and there's some precedent for using such
> stuff
> >> in some implementations of regex, where \< matches the beginning of a
> word,
> >> etc.
> >>
> >> and, of course there would be the shorthand variants \{7-\} for seven or
> >> more; \{-10\} for 1-10.
> >> Some might argue 0-10. Whatever.
> >>
> >> I THINK this could be implemented in both the fast pattern matcher and
> the
> >> current slow one. I know it wouldn't be that bad to do in the fast
> pattern
> >> matcher.
> >> I hadn't really given the slow one (the current one) much thought.
> >
> > I think it would be very useful. One small point:
> >
> > The '.' is short. This helps making it pupular. X\{1-\} is much less
> > so.
> >
> > Another thing that I think would help: an equivalent of perl's \w:
> > something similar to 'X', but also matches letters. This is syntactic
> > sugar, but we need such sugar for readable dialplans.
> >
> Leif and I had a proposal years ago for an "alphaexten" that used perl
> regexps.
>
>
Yes, I know that many have proposed full regex  and regex-like extensions
for
the dialplan patterns, but there's one big rub. The current pattern matcher
is light and
fast compared to a full regex matcher.  The restrictions on constructs make
it a fairly
fast linear matcher without any loops or recursive behavior to slow it down.
We can currently use
rules to quantify the "best match" that makes it fairly predictable, and the
work on a
fast pattern matcher (O(log(pattern length))  instead of  O((num
patterns)^2) was possible.

But, when you move to full regex approaches, you lose all those nice
properties. You'd have
to move to a 'best match first' sort of strategy,  so you can move from a
O(n^2) type scenario,
and you lock yourself into an O(n^2/2) scenario. For large dialplans on a
busy system, you are
totally screwed, without any hope of improving the situation.

I tried in times past to propose a subset of regex patterns that would still
leave us with fast
pattern matching

murf

/O
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-- 
Steve Murphy
ParseTree Corp
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Re: [asterisk-users] rawplayer in asterisk 1.0.0

[Rob Hillis]

On 02/17/10 05:01, Steve Howes wrote:
> On 16 Feb 2010, at 17:36, Arjan Kroon | Mobillion wrote:
>   
>> We are using asterisk version 1.0.0.
>> 
> Wow.
>   
Yeah, that about sums it up.  A little googling reveals that Asterisk
1.0 was announced on January 14th, 2005 - over five years ago.

I would have thought that even if upgrading to 1.2 or 1.4 wasn't an
option that upgrading to a bugfixed release of 1.0 would have gone a
long way towards resolving this kind of problem.  After all, Asterisk
was a much less mature product at that stage - and common wisdom has
/always/ been to avoid the first two or three releases of a new version.
(though with Asterisk 1.0, there wasn't another stable version to run -
everything up to that point was developmental)

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[asterisk-users] Cannot built kmod-dahdi-linux for PAE kvariant from SRPM

[stephen.hindmarch]

I using the asterisk yum repository at 
http://packages.asterisk.org/centos/5/current to install a vanilla asterisk. 
All went well on my development server, which is a fully up to date CentOS5 
machine. But now I am trying to do the same with my production server, which is 
an OEL5 machine with the 2.6.18-128.el5PAE kernel. This throws a dependency 
error for kmod-dahdi-linux.

kmod-dahdi-linux-2.2.1-1_centos5.2.6.18_164.11.1.el5.i686 from asterisk-current 
has depsolving problems
  --> Missing Dependency: kernel-i686 = 2.6.18-164.11.1.el5 is needed by 
package kmod-dahdi-linux-2.2.1-1_centos5.2.6.18_164.11.1.el5.i686 
(asterisk-current)
Error: Missing Dependency: kernel-i686 = 2.6.18-164.11.1.el5 is needed by 
package kmod-dahdi-linux-2.2.1-1_centos5.2.6.18_164.11.1.el5.i686 
(asterisk-current)

Well I suppose I can't expect there to be a kmod package for every possible 
combination of dahdi version and kernel out there so I suppose I shall have to 
build my own. I cannot change the kernel version on the production server as it 
is out of my hands.

So I have been trying to use the source rpm. I tried using yum to download it 
but the SRPMS directory is not at the same level as the RPMS directory so yum 
cannot see it. Instead I used wget to download 
dahdi-linux-kmod-2.2.1-1_centos5.2.6.18_164.11.1.el5.src.rpm, installed in a 
local build area and tried to build it.

I installed the correct devel packages, that is both kernel-devel and 
kernel-PAE-devel. To begin with I just tried

rpmbuild --bb ~/localrpms/SPECS/dahdi-linux-kmod.spec

but that fails as the spec file relies on the internally defined kernel version

error: Failed build dependencies:
kernel-devel = 2.6.18-164.11.1.el5 is needed by 
dahdi-linux-kmod-2.2.1-1_centos5.2.6.18_164.11.1.el5.i386

A brief inspection of the spec file suggested I could override the kernel 
version

uname -r
2.6.18-128.el5PAE
rpmbuild --bb ~/localrpms/SPECS/dahdi-linux-kmod.spec -D "kversion `uname -r`"

But this fails because by the time the kernel name gets passed to the makefile 
the "PAE" has been lost:-

...
~/localrpms/BUILD/dahdi-linux-kmod-2.2.1/_kmod_build_ 
~/localrpms/BUILD/dahdi-linux-kmod-2.2.1
+ make KVERS=2.6.18-128.el5 modules
You do not appear to have the sources for the 2.6.18-128.el5 kernel installed.

Further inspection and I tried to override the kernel variant parameter with
rpmbuild --bb ~/localrpms/SPECS/dahdi-linux-kmod.spec -D "kversion `uname -r`" 
-D "kvariants PAE"

This allows the RPM to build but when I try to install it I get

error: Failed dependencies:
kernel-i386 = 2.6.18-128.el5PAE is needed by 
kmod-dahdi-linux-PAE-2.2.1-1_centos5.2.6.18_128.el5.i386

Why "kernel-i386" when the kernel package I am using is "kernel-PAE" and really 
this check should be using "uname -r" to test the kernel version?

So I think there are parts of the spec file that are broken for my needs. I 
would be happy help test any changes. In the mean time am I ok to do a 
"-nodeps" install of the package I have built?

Steve Hindmarch


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Re: [asterisk-users] How does holdtime get calculated for queues

[Rob Hillis]

On 02/17/10 03:39, Warren Selby wrote:
> I had a customer ask me this question today, and I was surprised to
> say I didn't have an exact answer for them.  They have a relatively
> small support queue for their business (three agents, and rarely more
> than one person in line at any given time in the queue, if all agents
> are on a call),  Their support calls can range anywhere from 30
> seconds to 30 minutes.  Occasionally, they'll get four or five calls
> in a short time span, and the first three that are answered may end up
> being rather long.  The people in line are listening to hold music and
> hearing the periodic hold announcements and estimated wait times (you
> are currently caller number 2 in line, your estimated wait time is
> ).  The complaint they're receiving is that the estimated hold
> time tends to jump around a bit - the example I was given was that a
> caller was told when they entered the queue, they were currently
> caller number 2 and the estimated hold time was less than 2 minutes. 
> The caller waited upwards of 10 minutes, and then hung up when the
> estimated wait time announcement told them the wait time was 8 more
> minutes.

I have a call centre that is very similar to yours - two or three
operators, normally minimal hold times with occasional busy times
(although in this case, it's when there's a special event on that the
volume of calls pick up rather than the length of each call) and found
that Asterisk's estimated hold time simply wasn't accurate enough.  In
my situation, I simply disabled the hold time and the complaints about
inaccurate wait times were replaced by lots of praise for the mere fact
that they were told "your call is number 3 in line".

The problem seems to be when Asterisk is used to long periods of next to
no waiting time - something you can't really fault Asterisk for since
it's not psychic.  I found that after 10 or 15 minutes and a half dozen
calls, the wait time became pretty accurate and stayed that way until
the queue emptied again.  It may be for the type of queues that we're
talking about here that trying to calculate hold times simply isn't
feasible.

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Re: [asterisk-users] rawplayer in asterisk 1.0.0

[Steve Howes]

On 16 Feb 2010, at 17:36, Arjan Kroon | Mobillion wrote:
> We are using asterisk version 1.0.0.

Wow.

S

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Re: [asterisk-users] Important security alert: update your dialplans now!

[Steve Murphy]

On Tue, Feb 16, 2010 at 1:43 AM, Tzafrir Cohen wrote:

> On Mon, Feb 15, 2010 at 09:40:31AM -0700, Steve Murphy wrote:
> > On Mon, Feb 15, 2010 at 8:25 AM, Lenz Emilitri 
> wrote:
> >
> > > Yes but in any case you can enter all of the strings that reasonably
> match
> > > - even if you have variable-length numbers, you will be able to
> determine
> > > that a valid number be between 5 and 15 characters - or likely 2 to 20,
> all
> > > numbers. A number of 156 characters is very likely to be a problem.
> > >
> >
> > This is probably a stupid idea, because it could only be implemented in
> > trunk, and won't help with current implementations,
> > and I suggested it a long time ago already when I did the fast pattern
> > matching code, but I don't THINK it would be all that
> > hard to offer SOME regex syntax in patterns to help reduce the impact of
> > these kinds of problems.
> >
> > Like using:
> >
> > [incoming-from-voip]
> > exten => _X\{7-10\},1,Dial(${ext...@incoming-from-voip-old)
> >
> > instead of :
> >
> > [incoming-from-voip]
> > exten => XXX,1,Dial(${ext...@incoming-from-voip-old)
> > exten => ,1,Dial(${ext...@incoming-from-voip-old)
> > exten => X,1,Dial(${ext...@incoming-from-voip-old)
> > exten => XX,1,Dial(${ext...@incoming-from-voip-old)
> >
> > I put the \'s in front of the {}'s because we probably wouldn't want to
> > change the
> > behavior of exact matching, and there's some precedent for using such
> stuff
> > in some implementations of regex, where \< matches the beginning of a
> word,
> > etc.
> >
> > and, of course there would be the shorthand variants \{7-\} for seven or
> > more; \{-10\} for 1-10.
> > Some might argue 0-10. Whatever.
> >
> > I THINK this could be implemented in both the fast pattern matcher and
> the
> > current slow one. I know it wouldn't be that bad to do in the fast
> pattern
> > matcher.
> > I hadn't really given the slow one (the current one) much thought.
>
> I think it would be very useful. One small point:
>
> The '.' is short. This helps making it pupular. X\{1-\} is much less
> so.
>

Very true, but the added syntax would not replace '.'. And they mean two
different things. X\{1-\} would mean one or more numbers, . means
any number of any character.

Heck, besides N\{2-4\}, Z\{3-7\}, I guess we could even do .\{2-100\}, which
could mean 'match everything at the end of the string, but only if there's 2
to
100 of them', or something like that. Whatever is the most handy.


>
> Another thing that I think would help: an equivalent of perl's \w:
> something similar to 'X', but also matches letters. This is syntactic
> sugar, but we need such sugar for readable dialplans.
>

Could be done, but it ends up getting in the way of exact matching;
right now, using X,N,Z keeps you from exact matching those characters,
(is there some escape mech in the syntax to let you say \NA\NCY? I
haven't checked). But, there's no reason we can't add other matching chars
for handy things.  A = alpha chars Y = alphanum chars, G = Graphical chars,
whatever. We just have to watch those backslashes, because if we use them as
an escape to mean literals in some situations, and as a notation to mean a
special function in others, then it starts getting confusing real quick.
But all this kind of thing Could Be Done.

murf


>
> --
>   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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>



-- 
Steve Murphy
ParseTree Corp
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[asterisk-users] call is not going to wrong "context"

[Joseph]

I've Audiocodes MP-114 registered per-endpoint (2x FXO / 2x FXS)  but when call 
comes on pstn- it goes to context "fax-incoming"

in sip.conf:

[pstn-] 
type=friend
context=incoming
...

[pstn-9998] 
type=friend
context=fax-incoming
...

the device register per end point just fine, so it can find "secret=xxx" 
correctly but why the call is not forwarded to correct context.
It seems to some limitation in Audiocodes since I had no problem with calls 
going to incorrect context when I was using "Sipura/Linksys" devices.

-- 
Joseph

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[asterisk-users] rawplayer in asterisk 1.0.0

[Arjan Kroon | Mobillion]

 

Hi,

 

We are using asterisk version 1.0.0.

 

For queue'ing we use the rawplayer script to play a music file in the
background.

 

Now we see that after a while all the sessions on our Linux environment
will be taken by the rawplayer process.

An example of such a session is (done with ps -ax|grep rawplayer)

24785 ?Z  0:00 [rawplayer ]

 8415 ?Z  0:00 [rawplayer ]

13821 ?Z  0:00 [rawplayer ]

18868 ?Z  0:00 [rawplayer ]

22950 ?Z  0:00 [rawplayer ]

 

The only thing to get rid of these sessions is to restart asterisk and
then kill all rawplayer sessions

 

Does anybody have the same problem with this problem.

 

A way is to upgrade asterisk, but this is not now the solution for us.

 

The code for the rawplayer is: /usr/bin/rawplayer

#!/bin/sh

for name in $@; do

 cat $name ;

done

 

 

Regards,

 

Arjan Kroon

Mobillion BV

 

 

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[asterisk-users] CODECS: Best practice question: Avoid transcode when calling out?

[Karl Fife]

What is the current best practice to avoid transcoding on an outgoing call 
to a
party whose codec preference is not known in advance?

In other words, incoming calls are easy since codecs are negotiated from
least-known (the remote party) to most-known (my endpoint) and my codecs can
simply be preferred accordingly to match the remote.

Outbound calls seem harder.  Our endpoints always negotiate g.722 between 
themselves and Asterisk and then Asterisk must transcode to the preferred 
codec of the REMOTE party.  Not ideal.

It's easy if I'm calling to the PSTN--since I know in advance the 
preferences
of my ITSP, but I'm stumped about how to avoid transcoding when calling a
party whose codec preferences are not known in advance.

Is there an elegant way to do this?

-Karl


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[asterisk-users] How does holdtime get calculated for queues

[Warren Selby]

I had a customer ask me this question today, and I was surprised to say I
didn't have an exact answer for them.  They have a relatively small support
queue for their business (three agents, and rarely more than one person in
line at any given time in the queue, if all agents are on a call),  Their
support calls can range anywhere from 30 seconds to 30 minutes.
Occasionally, they'll get four or five calls in a short time span, and the
first three that are answered may end up being rather long.  The people in
line are listening to hold music and hearing the periodic hold announcements
and estimated wait times (you are currently caller number 2 in line, your
estimated wait time is ).  The complaint they're receiving is that the
estimated hold time tends to jump around a bit - the example I was given was
that a caller was told when they entered the queue, they were currently
caller number 2 and the estimated hold time was less than 2 minutes.  The
caller waited upwards of 10 minutes, and then hung up when the estimated
wait time announcement told them the wait time was 8 more minutes.

This is on a queue with a 30 minute wait time (the customer specifically
asked for this for all of their queues, so that's not changing).  After the
timeout limit is reached (i.e the command is Queue(support1800)), the
dialplan jumps to voicemail.

So, how does asterisk calculate holdtime for queues?  Why would it go up the
longer a caller is in line?  Where does it determine the initial wait time,
etc?

-- 
Thanks,
--Warren Selby
http://www.selbytech.com
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Re: [asterisk-users] Stupid question: Why Cmd Dial and Queue haven'tsame options?

[Danny Nicholas]

"callee" in queue-land is always an agent who picks up the call instead of
getting it transferred.  In some concepts, queue is more like a conference
than a transferred/dialed call.  Item 3 could be accomplished with an
AGI/AMI command.

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of didier.cuffaut
Sent: Tuesday, February 16, 2010 9:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Stupid question: Why Cmd Dial and Queue
haven'tsame options?

 

I apologize, my English isn't better than the last year

 

So, 

- why Queue has some options like 'caller can continue in his dialplan'
(Dial has g and F options for callee/caller) and not he same option for
'callee' ?

- no option in queue to send message (for only calle/caller) when bridged as
Dial has

- idem to announce periodically 'X minutes before hangup' (in Dial, not in
Queue)

 

Probabely, i've missed something. perhaps  some workaround ?

 

Thank's a lot for your attention

 

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[asterisk-users] Users of the SMS application?

[Tilghman Lesher]

Is anybody currently using the SMS application?  We've had some reports of the
current SMS application not working, and we're really not equipped on this
side of the pond (USA) to work on figuring out what the problem is, or even if
there is a problem at all.  I've attempted to get the SMS application to talk
to itself, but that does not seem to work.

Because of our inability to effectively work on this module, we're considering
moving the SMS application and related utilities into "unsupported" mode.

So if there's anybody who is successfully using the SMS application, we'd
really appreciate your assistance on this issue report:
https://issues.asterisk.org/view.php?id=16508

-- 
Tilghman Lesher
Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
Check us out at: www.digium.com & www.asterisk.org

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Re: [asterisk-users] [asterisk-dev] Maximum call handling capacity on single server

[David Backeberg]

On Mon, Feb 15, 2010 at 11:05 AM, Amit Patkar | Avhan Technologies
Pvt. Ltd.  wrote:
> I have a server with Quad Core Xeon 2.4GHz and 4GB RAM. I want to use it for
> PSTN-IP gateway. What is the maximum call handling capacity I can achieve
> with this server?

You can handle a lot of pure sip calls. You don't say anything about
the PCI / PCI-E bus on that machine, and purely speaking, nobody here
knows whether that server can even physically terminate several cards.

> I want at least 480 concurrent PSTN-IP calls. That mean I will have to
> install minimum 4 x 4E1 cards and run 480 G.711 RTP sessions. No call
> recording. No IVR. Pure gateway functionality. Can I achieve this capacity
> with given server configuration?
> If not, what kind of server is required to achieve this capacity.
>
> Has anyone done this? Please share results.

If anybody has done this, they would run into the problems the others
mentioned. If you can afford the phone bills you're going to incur
with this setup, you're also going to be able to afford at least
making 2x the capacity you really need for redundancy and business
continuity purposes.

Also, you're math is bad. If you're talking US, T1/PRI, you're only
terminating 23 channels per T1/PRI, so if you need 480 channels,
you're going to need 21 PRI's meaning 6x 4-port cards.

At which point, you should seriously consider buying a hardware
appliance like a Cisco 3845, and really you should buy two and split
the lines over those two.

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[asterisk-users] Stupid question: Why Cmd Dial and Queue haven't same options?

[didier.cuffaut]

I apologize, my English isn't better than the last year

So, 
- why Queue has some options like 'caller can continue in his dialplan' (Dial 
has g and F options for callee/caller) and not he same option for 'callee' ?
- no option in queue to send message (for only calle/caller) when bridged as 
Dial has
- idem to announce periodically 'X minutes before hangup' (in Dial, not in 
Queue)

Probabely, i've missed something. perhaps  some workaround ?

Thank's a lot for your attention

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Re: [asterisk-users] Issue with trying to dial two different servers at the same time.

[William Stillwell (Lists)]

Do a qeuee, add each as a station in the quee..


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Anness
Sent: Tuesday, February 16, 2010 10:04 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Issue with trying to dial two different servers at
the same time.

Okay, so my issue isn't really a technical one but more of needing
advice on the best way to program this.  I have a user in Colorado who
works from home but frequents our office in Colorado.  All of our
remote users connect to a server in Dallas the users at the HQ in
Colorado connect to a separate server in the Colorado office that is
on the private network.  I have an IAX trunk between the two servers.
I also have an IAX trunk from my local server here in Texas.  I am
using this dial command to try to ring both phones at the same time:

exten => 12109,1,Dial(iax2/castle-rock/109&iax2/colo/17128,20)

This has one flaw, if for whatever reason his home phone isn't
connected, like it isn't now, than when I call 12109 I get his home
voicemail as it picks up right away.

How can I program this so that even if the voicemail on 17128 or 109
pick up it still rings the other phone?

Thank you all for your advice.

-- 
Steve Anness

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Re: [asterisk-users] Important security alert: update your?dialplans now!

[Leif Madsen]

Tilghman Lesher wrote:
> On Monday 15 February 2010 18:01:11 Vinícius Fontes wrote:
 He probably means AgentCallbackLogin
>>> While it has been deprecated, that hasn't been removed, either.  If
>>> an
>>> enterprising person would like to try to fix it, I don't have an
>>> objection.
>> Wasn't AgentCallBackLogin() removed in 1.6.1?
> 
> I didn't think we did, but apparently so.
> 

I wrote a document on how to do this in the dialplan. It's in the doc/ 
subdirectory in Asterisk 1.6.2.

Leif.

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[asterisk-users] Issue with trying to dial two different servers at the same time.

[Steve Anness]

Okay, so my issue isn't really a technical one but more of needing
advice on the best way to program this.  I have a user in Colorado who
works from home but frequents our office in Colorado.  All of our
remote users connect to a server in Dallas the users at the HQ in
Colorado connect to a separate server in the Colorado office that is
on the private network.  I have an IAX trunk between the two servers.
I also have an IAX trunk from my local server here in Texas.  I am
using this dial command to try to ring both phones at the same time:

exten => 12109,1,Dial(iax2/castle-rock/109&iax2/colo/17128,20)

This has one flaw, if for whatever reason his home phone isn't
connected, like it isn't now, than when I call 12109 I get his home
voicemail as it picks up right away.

How can I program this so that even if the voicemail on 17128 or 109
pick up it still rings the other phone?

Thank you all for your advice.

-- 
Steve Anness

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Re: [asterisk-users] call transfer

[Gergo Csibra]

Tuesday, February 16, 2010, 12:55:12 PM, cool wrote:

> call comes it should be received by extenion 2000, n if person wants to
> talk to Sales, receptionist should put the caller on hold than connect
> to Sales i.e exten 2001, while on hold the caller should hear music on
> hold,now sale exten can take his call n talk to it.same with Accounts
> ext 2002.
...
> what to next

To have call transfer in your asterisk setup, YOU need to read some
documentation. Start here: http://www.voip-info.org

-- 
Best regards,
 Gergomailto:csi...@gmail.com


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Re: [asterisk-users] SIP RTP ports not released when channel is hung up

[Armin Schindler]

On Tue, 16 Feb 2010, Marcus Hunger wrote:

Hi,

did you see this one: https://issues.asterisk.org/view.php?id=16774 ? It looks 
related to your issue.


Oh thanks, I missed that one.
It really looks related. I have added a note.

Thanks,
Armin


Best regards, Marcus

On Fri, Feb 12, 2010 at 12:04 PM, Armin Schindler  wrote:
  On Fri, 12 Feb 2010, Armin Schindler wrote:
   I had a look at
     netstat -nuap
   and it shows that a lot of ports are still assigned, even if there 
is no
   channel in use.
   But "sip show channels" show a lot of (unused) entries with no
   codec/Format and "Last Message" like INVITE, REGISTER, OPTIONS.
  >> REGISTER and OPTIONS allocate no RTP ports, so those are not a 
problem. If
  >> you have a SIP channel that has a last message being INVITE and still 
say
  >> you have no calls, you have a problem right there.
  >
  > I just see these entries with "sip show channels", but cannot tell if
  > e.g. the REGISTER listed channels have RTP ports allocated.
  > Who can I find out which SIP channel allocated which port?
  > Or which SIP channel belongs to the ports I see with 'netstat -nuap'?

I just made a test to confirm:
After a restart of asterisk (to have a clean state with no sip channels
activ and no RTP port allocated), I can confirm that:
- REGISTER and OPTION listed sip channels don't use RTP ports
- after some calls (e.g. SIP to SIP) the RTP ports are freed immediately
  (looks like this is the case on hangup before answer).
- after some other calls, the RTP ports are freed after about 20-30 seconds
  after hangup.
So basically all is correct.

> I do have a sip channels like
>  172.21.4.114    666    0430c3a638e  00102/0  0x0 (nothing)    No   Init: 
INVITE
> in 'sip show channels' and they don't go away for a long time.
> Shouldn't there be a timeout to destroy such a channel even if somehow
> the phone was 'disconnected' in during a call?
>
>>> If the channels exists even after 32 seconds after BYE, and BYE was
>>> signaled correctly, I would file a bug report.

It really looks like that there is a case where the sip channel is not
destroyed and that is the cause of the problem.
I will try to reproduce this.
Any ideas?

Armin


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Marcus Hunger - hun...@sipgate.de
Telefon: +49 (0)211-63 55 55-61
Telefax: +49 (0)211-63 55 55-22

sipgate GmbH - Gladbacher Str. 74 - 40219 Düsseldorf
HRB Düsseldorf 39841 - Geschäftsführer: Thilo Salmon, Tim Mois
Steuernummer: 106 / 5724 / 7147, Umsatzsteuer-ID: DE219349391

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Re: [asterisk-users] call transfer

[Brian]

On Tue, 2010-02-16 at 17:25 +0530, cool dude wrote:
> call transfer
> 
> call transfer from reception to other extensions.
> 
> Question: Details of Extensions
> 
> Reception - 2000
> Sales - 2001
> Accounts - 2002
> 
> any call comes it should be received by extenion 2000, n if person
> wants to talk to Sales, receptionist should put the caller on hold
> than connect to Sales i.e exten 2001, while on hold the caller should
> hear music on hold,now sale exten can take his call n talk to it.same
> with Accounts ext 2002.
> 
> 
> 
> vi /etc/asterisk/sip.conf
> 
> [general]
> port = 5060
> bindaddr = 0.0.0.0
> context = others
> 
> 
> [2000]
> type=friend
> context=reception
> secret=1234
> host=dynamic
> 
> [2001]
> type=friend
> context=sales
> secret=1234
> host=dynamic
> 
> 
> [2002]
> type=friend
> context=accounts
> secret=1234
> host=dynamic
> ~ 
> ##
> 
> vi /etc/asterisk/extension.conf
> 
> [from-zaptel]
> exten => s,1,wait(2)
> exten => s,n,Dial(SIP/2000,20)
> 
> 
> what to next
> 
> 
> __
What have you tried? Which links/mans have you read to set up music on
hold? Are any of them wrong at all? 


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[asterisk-users] call transfer

[cool dude]

call transfer




call transfer from reception to other extensions.

Question: Details of Extensions

Reception - 2000
Sales - 2001
Accounts - 2002

any
call comes it should be received by extenion 2000, n if person wants to
talk to Sales, receptionist should put the caller on hold than connect
to Sales i.e exten 2001, while on hold the caller should hear music on
hold,now sale exten can take his call n talk to it.same with Accounts
ext 2002.



vi /etc/asterisk/sip.conf

[general]
port = 5060
bindaddr = 0.0.0.0
context = others


[2000]
type=friend
context=reception
secret=1234
host=dynamic

[2001]
type=friend
context=sales
secret=1234
host=dynamic


[2002]
type=friend
context=accounts
secret=1234
host=dynamic
~
##

vi /etc/asterisk/extension.conf

[from-zaptel]
exten => s,1,wait(2)
exten => s,n,Dial(SIP/2000,20)


what to next


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Re: [asterisk-users] Empty SIP Packet

[Olle E. Johansson]

16 feb 2010 kl. 10.40 skrev Alexandru Oniciuc:

> Hello list,
>  
> debugging SIP, I found many empty lines like:
>  
> <--- SIP read from UDP://XXX.XXX.XXX.XXX:5060 --->
> <->
>  
> The IP address above corresponds to one of my accounts, which 
> is behind a firewall.
>  
> Is that normal, maybe some firewall that tries to keep a port 
> open, or is my firewall cleaning the SIP Packet?
>  
> Thanks in advance & best regards,
>  
It's one way of sending a SIP keepalive, many phones do this to keep NAT open.

/O
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Re: [asterisk-users] SIP RTP ports not released when channel is hung up

[Marcus Hunger]

Hi,

did you see this one: https://issues.asterisk.org/view.php?id=16774 ? It
looks related to your issue.

Best regards, Marcus

On Fri, Feb 12, 2010 at 12:04 PM, Armin Schindler  wrote:

> On Fri, 12 Feb 2010, Armin Schindler wrote:
>  I had a look at
>    netstat -nuap
>  and it shows that a lot of ports are still assigned, even if there is
> no
>  channel in use.
>  But "sip show channels" show a lot of (unused) entries with no
>  codec/Format and "Last Message" like INVITE, REGISTER, OPTIONS.
> >> REGISTER and OPTIONS allocate no RTP ports, so those are not a problem.
> If
> >> you have a SIP channel that has a last message being INVITE and still
> say
> >> you have no calls, you have a problem right there.
> >
> > I just see these entries with "sip show channels", but cannot tell if
> > e.g. the REGISTER listed channels have RTP ports allocated.
> > Who can I find out which SIP channel allocated which port?
> > Or which SIP channel belongs to the ports I see with 'netstat -nuap'?
>
> I just made a test to confirm:
> After a restart of asterisk (to have a clean state with no sip channels
> activ and no RTP port allocated), I can confirm that:
> - REGISTER and OPTION listed sip channels don't use RTP ports
> - after some calls (e.g. SIP to SIP) the RTP ports are freed immediately
>   (looks like this is the case on hangup before answer).
> - after some other calls, the RTP ports are freed after about 20-30 seconds
>   after hangup.
> So basically all is correct.
>
> > I do have a sip channels like
> >  172.21.4.1146660430c3a638e  00102/0  0x0 (nothing)No
> Init: INVITE
> > in 'sip show channels' and they don't go away for a long time.
> > Shouldn't there be a timeout to destroy such a channel even if somehow
> > the phone was 'disconnected' in during a call?
> >
> >>> If the channels exists even after 32 seconds after BYE, and BYE was
> >>> signaled correctly, I would file a bug report.
>
> It really looks like that there is a case where the sip channel is not
> destroyed and that is the cause of the problem.
> I will try to reproduce this.
> Any ideas?
>
> Armin
>
>
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-- 
Dipl.-Inf. (FH)
Marcus Hunger - hun...@sipgate.de
Telefon: +49 (0)211-63 55 55-61
Telefax: +49 (0)211-63 55 55-22

sipgate GmbH - Gladbacher Str. 74 - 40219 Düsseldorf
HRB Düsseldorf 39841 - Geschäftsführer: Thilo Salmon, Tim Mois
Steuernummer: 106 / 5724 / 7147, Umsatzsteuer-ID: DE219349391

www.sipgate.de - www.sipgate.at - www.sipgate.co.uk
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Re: [asterisk-users] Important security alert: update your dialplans now!

[Olle E. Johansson]

16 feb 2010 kl. 09.43 skrev Tzafrir Cohen:

> On Mon, Feb 15, 2010 at 09:40:31AM -0700, Steve Murphy wrote:
>> On Mon, Feb 15, 2010 at 8:25 AM, Lenz Emilitri  wrote:
>> 
>>> Yes but in any case you can enter all of the strings that reasonably match
>>> - even if you have variable-length numbers, you will be able to determine
>>> that a valid number be between 5 and 15 characters - or likely 2 to 20, all
>>> numbers. A number of 156 characters is very likely to be a problem.
>>> 
>> 
>> This is probably a stupid idea, because it could only be implemented in
>> trunk, and won't help with current implementations,
>> and I suggested it a long time ago already when I did the fast pattern
>> matching code, but I don't THINK it would be all that
>> hard to offer SOME regex syntax in patterns to help reduce the impact of
>> these kinds of problems.
>> 
>> Like using:
>> 
>> [incoming-from-voip]
>> exten => _X\{7-10\},1,Dial(${ext...@incoming-from-voip-old)
>> 
>> instead of :
>> 
>> [incoming-from-voip]
>> exten => XXX,1,Dial(${ext...@incoming-from-voip-old)
>> exten => ,1,Dial(${ext...@incoming-from-voip-old)
>> exten => X,1,Dial(${ext...@incoming-from-voip-old)
>> exten => XX,1,Dial(${ext...@incoming-from-voip-old)
>> 
>> I put the \'s in front of the {}'s because we probably wouldn't want to
>> change the
>> behavior of exact matching, and there's some precedent for using such stuff
>> in some implementations of regex, where \< matches the beginning of a word,
>> etc.
>> 
>> and, of course there would be the shorthand variants \{7-\} for seven or
>> more; \{-10\} for 1-10.
>> Some might argue 0-10. Whatever.
>> 
>> I THINK this could be implemented in both the fast pattern matcher and the
>> current slow one. I know it wouldn't be that bad to do in the fast pattern
>> matcher.
>> I hadn't really given the slow one (the current one) much thought.
> 
> I think it would be very useful. One small point:
> 
> The '.' is short. This helps making it pupular. X\{1-\} is much less
> so.
> 
> Another thing that I think would help: an equivalent of perl's \w: 
> something similar to 'X', but also matches letters. This is syntactic
> sugar, but we need such sugar for readable dialplans.
> 
Leif and I had a proposal years ago for an "alphaexten" that used perl regexps.

/O
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[asterisk-users] Empty SIP Packet

[Alexandru Oniciuc]

Hello list,

debugging SIP, I found many empty lines like:

<--- SIP read from UDP://XXX.XXX.XXX.XXX:5060 --->
<->

The IP address above corresponds to one of my accounts, which 
is behind a firewall.

Is that normal, maybe some firewall that tries to keep a port 
open, or is my firewall cleaning the SIP Packet?

Thanks in advance & best regards,

Alex
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Re: [asterisk-users] Important security alert: update your dialplans now!

[Tzafrir Cohen]

On Mon, Feb 15, 2010 at 09:40:31AM -0700, Steve Murphy wrote:
> On Mon, Feb 15, 2010 at 8:25 AM, Lenz Emilitri  wrote:
> 
> > Yes but in any case you can enter all of the strings that reasonably match
> > - even if you have variable-length numbers, you will be able to determine
> > that a valid number be between 5 and 15 characters - or likely 2 to 20, all
> > numbers. A number of 156 characters is very likely to be a problem.
> >
> 
> This is probably a stupid idea, because it could only be implemented in
> trunk, and won't help with current implementations,
> and I suggested it a long time ago already when I did the fast pattern
> matching code, but I don't THINK it would be all that
> hard to offer SOME regex syntax in patterns to help reduce the impact of
> these kinds of problems.
> 
> Like using:
> 
> [incoming-from-voip]
> exten => _X\{7-10\},1,Dial(${ext...@incoming-from-voip-old)
> 
> instead of :
> 
> [incoming-from-voip]
> exten => XXX,1,Dial(${ext...@incoming-from-voip-old)
> exten => ,1,Dial(${ext...@incoming-from-voip-old)
> exten => X,1,Dial(${ext...@incoming-from-voip-old)
> exten => XX,1,Dial(${ext...@incoming-from-voip-old)
> 
> I put the \'s in front of the {}'s because we probably wouldn't want to
> change the
> behavior of exact matching, and there's some precedent for using such stuff
> in some implementations of regex, where \< matches the beginning of a word,
> etc.
> 
> and, of course there would be the shorthand variants \{7-\} for seven or
> more; \{-10\} for 1-10.
> Some might argue 0-10. Whatever.
> 
> I THINK this could be implemented in both the fast pattern matcher and the
> current slow one. I know it wouldn't be that bad to do in the fast pattern
> matcher.
> I hadn't really given the slow one (the current one) much thought.

I think it would be very useful. One small point:

The '.' is short. This helps making it pupular. X\{1-\} is much less
so.

Another thing that I think would help: an equivalent of perl's \w: 
something similar to 'X', but also matches letters. This is syntactic
sugar, but we need such sugar for readable dialplans.

-- 
   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] OT- Using TR-069

[Olle E. Johansson]

16 feb 2010 kl. 08.54 skrev Olivier:

> Hi,
> 
> Phone vendors (Snom, Thomson-Technicolor, ...) are on the way to support 
> TR-069 (see http://en.wikipedia.org/wiki/TR-069).
> 
> Has someone experienced with TR-069 ?
> What do you think of this protocol set ?
> 
And the SIP forum is about to release their UA Config framework. So there are a 
few to choose from.

/O ;-)
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Re: [asterisk-users] OT- Using TR-069

[Leonja Cerebro]

Hi,
sorry if the question seems silly, but for some reason, all these phones,
modems, ATAs use TR-069 instead of SNMP ?
I have experienced it, but as user, for a small time.

Regards


On 16 February 2010 09:54, Olivier  wrote:

> Hi,
>
> Phone vendors (Snom, Thomson-Technicolor, ...) are on the way to support
> TR-069 (see http://en.wikipedia.org/wiki/TR-069).
>
> Has someone experienced with TR-069 ?
> What do you think of this protocol set ?
>
> Regards
>
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[asterisk-users] OT- Using TR-069

[Olivier]

Hi,

Phone vendors (Snom, Thomson-Technicolor, ...) are on the way to support
TR-069 (see http://en.wikipedia.org/wiki/TR-069).

Has someone experienced with TR-069 ?
What do you think of this protocol set ?

Regards
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