RE: [asterisk-users] Free seating Agents and logged in / logged outindication

[Alexander Topolanek]

Am Freitag, den 27.04.2007, 16:25 -0400 schrieb Dean Collins:
> A pop up on their pc display using Adhearsion to drive the resulting
> logged in/out popup?

Sorry, I forgot to tell that the Agents don't have PC's. Is it possible
to trigger the MWI from an AGI-script that is fired when an Agent is
logged out?

> > I would like to set up a call center with free seating agents.
> However I would like to indicate the agent status somehow on the
> terminal, to tell the agent if she has been logged out due to
> non-answer.

> 
-- 
Alexander Topolanek
http://www.topolanek.at

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Music on Hold issue with asterisk 1.4.2

[Steve Finkelstein]

Hi all,

I've compiled zaptel drivers and reconfigure asterisk afterwards from
source --with-zaptel.

Modules are loaded accordingly:

asterisk-1.4.2 # lsmod |grep z
Module  Size  Used by
ztdummy 5472  0
zaptel194504  5 ztdummy
crc_ccitt   3521  1 zaptel

my musiconhold.conf:

asterisk-1.4.2 # grep -v '^;' /etc/asterisk/musiconhold.conf

[default]
mode=quietmp3
directory=/var/lib/asterisk/mohmp3

and finally in my extensions.conf:

asterisk-1.4.2 # grep 100 /etc/asterisk/extensions.conf
exten => 100,1,MusicOnHold(30)
exten => 100,2,Hangup

When I dial 100 however, I receive the following:

-- Executing [EMAIL PROTECTED]:1] MusicOnHold("SIP/31337-007017f0",
"30") in new stack
[Apr 28 00:17:33] WARNING[30975]: res_musiconhold.c:947
local_ast_moh_start: No class: 30
[Apr 28 00:17:33] WARNING[30975]: res_musiconhold.c:575 moh0_exec:
Unable to start music on hold (class '30') on channel SIP/31337-007017f0
  == Spawn extension (internal, 100, 1) exited non-zero on
'SIP/31337-007017f0'

Thanks for any input.

- sf
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [asterisk-users] headsets for linksys/sipura phones?

[Nabeel Jafferali]

You can look for headsets made for Motorola cell phones. Also, Plantronics
has some compatible models - I can dig up part numbers if you're interested.


> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Per Jessen
> Sent: April 27, 2007 8:32 AM
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] headsets for linksys/sipura phones?
> 
> Erik Anderson wrote:
> 
> > On 4/26/07, Per Jessen <[EMAIL PROTECTED]> wrote:
> >> I was just browsing my local suppliers list of headsets - not a 
> >> single one with a single 2.5mm jack.  Either USB or 1-2 
> 3.5mm jacks.
> >>
> >> Can anyone recommend a headset that works with e.g. 
> SPA-921 and -941?
> > 
> > Try your local mobile phone supplier.  I used a headset 
> that came with 
> > one of my cell phones, and it worked great w/ my SPA-941.
> 
> Not a bad idea  - which make was this for?  None of my phones 
> (Ericsson,
> Nokia) have a 2.5mm socket, they're all special/proprietary.
> 
> 
> /Per Jessen, Zürich
> 
> ___
> --Bandwidth and Colocation provided by Easynews.com --
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [asterisk-users] Best Wireless bridge for Polycoms

[Nabeel Jafferali]

You can purchase the Linksys part PA100-NA and plug it into a WBP54G and
then ignore the power connector hanging off the WBP54G. 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of Mike
> Sent: April 27, 2007 3:00 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [asterisk-users] Best Wireless bridge for Polycoms
> 
> Michael and all those who replied,
>  
> This Linksys WBP54G does  seems to be what I need, but it 
> also seems very much made for Linksys phones.  Isn't there 
> some sort of equivalent thing that comes with it's own power 
> supply (at the cost of needing another outlet for the phone)?
>  
> Alternatively, where do I find an adapter for NA power that 
> turns into 2V 5A DC current?
>  
> Mike
> 
> 
> 
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Michael Graves
> Sent: Friday, April 27, 2007 13:45
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Best Wireless bridge for Polycoms
> 
> 
> --Original Message Text---
> From: Mike
> Date: Fri, 27 Apr 2007 10:24:05 -0400
> 
> Hi, 
> 
> I'm stuck doing an install with Polycoms at a small office 
> with no RJ-45. They went wireless 100%, poor them. I insist 
> on using Polycom unless it's impossible because that's what I 
> am standardized on for many reasons. 
> 
> What's the best way/device to turn a wired Polycom 501 (or 
> any Polycom for that matter) into a WiFi phone? 
> 
> Mike 
> 
> 
> Linksys makes a device spcifically for this role so that 
> their SPA series IP phones can be connected to WIFI.
> 
> http://www.linksys.com/servlet/Satellite?c=L_Product_C2&childp
> agename=US%2FLayout&cid=1139961537989&pagename=Linksys%2FCommo
> n%2FVisitorWrapper&lid=3798954250B11
> 
> This device take power from the phone via a simple coax plug 
> on the phone. Could easily be powered externally with a wall wart.
> 
> Michael
> 
> 
> 
> 

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] ZT_CHANCONFIG failed onchannel1:Nosuchdeviceoraddress

[CSB]

On Fri, Apr 27, 2007 at 07:11:48AM +1200, CSB wrote:

[snip]


As suggested earlier I replaced this with:
/etc/modprobe.d/zaptel
options wctdm opermode=NEWZEALAND honormode=1 boostringer=1 fastringer=1


[snip]


dmesg
Zapata Telephony Interface Registered on major 196
Zaptel Version: 1.2.17.1
Zaptel Echo Canceller: KB1
wctdm: Unknown parameter `honormode'


This is the problem


Updated
vi /etc/modprobe.d/zaptel
options wctdm opermode=NEWZEALAND fxshonormode=1 boostringer=1 fastringer=1

reboot
dmesg
Zapata Telephony Interface Registered on major 196
Zaptel Version: 1.2.17.1
Zaptel Echo Canceller: KB1
Zaptel Transcoder support loaded

/sbin/ztcfg -
Zaptel Configuration
==


Channel map:

Channel 01: FXO Kewlstart (Default) (Slaves: 01)
Channel 02: FXO Kewlstart (Default) (Slaves: 02)
Channel 03: FXS Kewlstart (Default) (Slaves: 03)
Channel 04: FXS Kewlstart (Default) (Slaves: 04)

4 channels configured.

ZT_CHANCONFIG failed on channel 1: No such device or address (6)

ls -l /sys/class/zaptel
total 0
drwxr-xr-x 2 root root 0 Apr 28 13:33 zapchannel
drwxr-xr-x 2 root root 0 Apr 28 13:33 zapctl
drwxr-xr-x 2 root root 0 Apr 28 13:33 zappseudo
drwxr-xr-x 2 root root 0 Apr 28 13:33 zaptimer
drwxr-xr-x 2 root root 0 Apr 28 13:33 zaptranscode

lsmod | grep ^zaptel
zaptel184612  2 zttranscode,wctdm

lspci
Card is not listed

ls /proc/zaptel
Nothing returned

ls -la /dev/zap
total 0
drwxr-xr-x  2 asterisk asterisk  140 Apr 28 13:33 .
drwxr-xr-x 11 root root 3660 Apr 28 13:33 ..
crw---  1 asterisk asterisk 196, 254 Apr 28 13:33 channel
crw---  1 asterisk asterisk 196,   0 Apr 28 13:33 ctl
crw---  1 asterisk asterisk 196, 255 Apr 28 13:33 pseudo
crw---  1 asterisk asterisk 196, 253 Apr 28 13:33 timer
crw-rw  1 asterisk asterisk 196, 250 Apr 28 13:33 transcode

vi /etc/udev/rules.d/50-udev.rules
# Section for zaptel device
KERNEL=="zapctl",   NAME="zap/ctl"
KERNEL=="zaptimer", NAME="zap/timer"
KERNEL=="zapchannel",   NAME="zap/channel"
KERNEL=="zappseudo",NAME="zap/pseudo"
KERNEL=="zap[0-9]*",NAME="zap/%n"

Any further help appreciated.

Cameron


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Call Pick Up

[Leonardo Kamache (Gmail)]

Two words for you... parking lot.
Try to transfer your call to extension 700 and see what hapens...




On 4/27/07, Jim Duda <[EMAIL PROTECTED]> wrote:

I use Asterisk in my house.  Each phone is a different extension.  I
really like the ability to have multiple simultaneous calls in the
house.  However, I do miss being able to be able to pick up a phone in a
different room.  Currently, I have to either transfer the call or
transfer the call to a "conference" extension to move around the house.

While a connection in progress on one extension, I would like to go to
any other phone, dial some extension number, in order to ether pick up
the call or join in an automatic conference.  In other words, make it
work like the old ma bell phone (when I want it to :-) )

Is this possible in Asterisk?

Thanks,

Jim

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Call Pick Up

[Jim Duda]

I use Asterisk in my house.  Each phone is a different extension.  I 
really like the ability to have multiple simultaneous calls in the 
house.  However, I do miss being able to be able to pick up a phone in a 
different room.  Currently, I have to either transfer the call or 
transfer the call to a "conference" extension to move around the house.


While a connection in progress on one extension, I would like to go to 
any other phone, dial some extension number, in order to ether pick up 
the call or join in an automatic conference.  In other words, make it 
work like the old ma bell phone (when I want it to :-) )


Is this possible in Asterisk?

Thanks,

Jim

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Unable to find a codec translation path from ilbc to ulaw

[dave cantera]

oliver,
what gateway provider are you referring to?doesn't your sip phone 
connect directly to * as your diagram indicated?

DSL providers should not be doing any codec anything!
daveC

Oliver Brandt wrote:

Hi!

As the upstream of my DSL-connection is very slow, I'd like my
sip-phones to use iLBC to connect to my *. My gateway provider only
allows ulaw. Hence, I'd like to use the follwing setup:

SIP-phone <--iLBC--> Asterisk <---ulaw> PSTN-Gateway

I get the following error:

"Unable to find a codec translation path from ilbc to ulaw"

Setup SIP-phone:
disallow=all
allow=ilbc

Setup PSTN-Gateway:
disallow=all
allow=ulaw

I've googled for overn an houre. But no luck. So I'd really apreciate
any help!

Thanks!
Oliver
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users



  


--
Building Strong Relationships w/ Intelligent Customer Service
--

Interlocking Business Solutions, LLC
856-380-0894 x5000


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Asterisk 1.4.4 Released

[The Asterisk Development Team]

The Asterisk.org development team has released Asterisk version 1.4.4.

A good number of significant bugs have been fixed in the past few days,
so a new release was made to get these fixes to the community as soon as
possible.  Some of the fixes include:

- Fix a crash in chan_zap
- Fix some cases where IAX2 calls would get dropped
- Merge a re-write of channel group counting support that fixes a lot of 
issues

- Fix some DTMF issues related to the use of chan_agent
- Fix a crash that occurs when using dialplan functions to set global 
variables


As always, a ChangeLog is available that provides a full list of 
changes.  The  releases are available for download from ftp.digium.com.


Thank you for your support of Asterisk.org!

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] CDR changes in 1.4.3?

[Steve Murphy]

On Fri, 2007-04-27 at 11:32 -0400, Scott Lykens wrote:
> Hello all:
> 
> I upgraded to 1.4.3 last night and use MySQL for CDR.
> 
> I have noticed that 1.4.3 seems to log a lot of "crap" to CDR that
> 1.4.2 did not. I use a few macros in my dialplan to handle outgoing
> calls (lcr type stuff) and in addition to the proper CDR for the call
> itself I also have records to 's' in the same dest-context and entries
> to 's' in the default context. Up to 3 CDRs are generated for one
> outgoing call (SIP -> Zap channel) with one being the legit CDR and
> two being the type described above.
> 
> My dialplan executes a ResetCDR after calling the lcr macro so that
> the CDR is sane and accurate, however, it appears these "spurious" CDR
> entries are generated by the call the ResetCDR even though I do not
> call it with any options.
> 
> Am I missing something obvious here? I have read the ChangeLog but I
> didn't see anything that addressed this particular issue.
> 
> Thanks for the help.
> 
> sl

Scott--

I'm the guilty party. I've been trying to fix several CDR bugs,
involving stuff like missing times, missing changes in state (like
NO_ANSWER when the call was ANSWERED), etc.

CDR's are complicated by the fact that they record 3 events: "start",
"Answer", and "end" events. Add to that the fact that in most cases at
least two channels are involved, sometimes 4 or 5, or even more,
involving bridging, maquerading, parking, transfers, local channels,
AGI, conferences, and more...

Some cases were impossible to fix unless CDR's were attached to every
channel, 
and merged to collect the bits and pieces that sometimes were on the
wrong side of the bridge.

The result is that several more cases are more accurate, but also, that
rather uninteresting CDR's can be generated. In contemplating what could
be done to get rid of some of these, I sometimes have to ask, "is this
truly something we have to get rid of?"... In the meantime,
uninteresting CDR's with NO_ANSWER and billsec=0, should be easy to
filter out, right?

I will, in the coming days, look at some of the extraneous CDR's that
are generated, and see what I can do to get rid of them. It's not always
that simple.
If we ring a phone, for instance, and no-one answers it, is that truly,
really, something that no-one will ever be, could ever be, interested
in? (just a fer-instance).

I welcome your input. Complain up a storm. I'll try my best to make you
happy.

murf



-- 
Steve Murphy
Software Developer
Digium


smime.p7s
Description: S/MIME cryptographic signature
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Voicemail on Different Server

[Anthony Rodgers]

mount -o intr,nolock ought to do the trick. we're using those 
options now, but thankfully haven't had reason to find out if they work 
or not yet.


CP

Doug Garstang wrote:
No, you can get Asterisk and NFS to work fine together. It was in my 
past job, so I can't remember the exact settings, but there was some 
magic combination of NFS client mount settings that would cause 
Asterisk to return immediately, rather than hang, if there was an NFS 
communications problem.


Doug.



___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] incoming zaptel calls fail

[CSB]

Using the latest SVN of zaptel and asterisk, I can no longer receive  
incoming analog calls.  The caller just hears it ringing but Asterisk  
doesn't pick up.


I am seeing these error messages:

[Apr  9 09:38:02] WARNING[16564]: chan_zap.c:6470 ss_thread: CallerID  
returned with error on channel 'Zap/2-1'
[Apr  9 09:38:50] WARNING[16570]: chan_zap.c:6470 ss_thread: CallerID  
returned with error on channel 'Zap/2-1'
[Apr  9 09:39:34] WARNING[16580]: chan_zap.c:6470 ss_thread: CallerID  
returned with error on channel 'Zap/2-1'



Did you resolve this issue? If so, how?

Cameron
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [asterisk-users] Free seating Agents and logged in / logged outindication

[Dean Collins]

A pop up on their pc display using Adhearsion to drive the resulting
logged in/out popup?

 

 

Regards,

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
 +1-212-203-4357 Ph

    
 
 

 

> -Original Message-

> From: [EMAIL PROTECTED] [mailto:asterisk-users-

> [EMAIL PROTECTED] On Behalf Of Alexander Topolanek

> Sent: Friday, 27 April 2007 2:56 PM

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

> Subject: [asterisk-users] Free seating Agents and logged in / logged
outindication

> 

> Hi,

> 

> I would like to set up a call center with free seating agents. However
I

> would like to indicate the agent status somehow on the terminal, to
tell

> the agent if she has been logged out due to non-answer.

> 

> Does anyone has a good idea how this can be achived?

> 

> best regards

> --

> Alexander Topolanek

> http://www.topolanek.at

> 

> ___

> --Bandwidth and Colocation provided by Easynews.com --

> 

> asterisk-users mailing list

> To UNSUBSCRIBE or update options visit:

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



image001.gif
Description: image001.gif
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] chan_bluetooth as FXS?

[Yuan LIU]

Any way to use chan_bluetooth as FXS?

Yuan Liu


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [asterisk-users] can�t anserd the call

[Yuan LIU]

From: "Josu Lazkano Lete" <[EMAIL PROTECTED]>
Date: Fri, 27 Apr 2007 10:09:56 +0200

hello, I have instaled a analog line, and when I call on the console apears 
that:


I want to redirect the call to 101 extension.

*CLI> -- Starting simple switch on 'Zap/1-1'
  == Starting Zap/1-1 at default,s,1 failed so falling back to exten 's'
  == Starting Zap/1-1 at default,s,1 still failed so falling back to 
context 'default'

[good stuff sniffed]


mi configuration files are this:

extensions.conf:

[general]
static=yes
writeprotect=yes
;autofallthrough=yes
;clearglobalvars=no
;priorityjumping=no

[SOME]
exten => 101,1,Dial(SIP/101,30,Ttm)
exten => 101,2,Hangup

exten => 102,1,Dial(SIP/102,30,Ttm)
exten => 102,2,Hangup

[incoming]
exten => s,1,Wait(1)
exten => s,2,Answer()
exten => s,3,Dial(SIP/101,30,Ttm)

[outgoing]

exten =>_94XXX,1,Dial(ZAP/g1/${EXTEN},45,tTwW)
exten =>_94XXX,2,Hangup()
exten =>_94XXX,102,Hangup()

zapata.conf:


You have not specified a particular context for your Zap channels in 
zapata.conf, so any call initiated from Zap would go to [default].  You also 
specified that all SIP channels should use context [default].  However, you 
haven't created a [default] context in extensions.conf.


So either create a [default], or change contexts used by Zap and SIP to 
something you have in extensions.conf.


Yuan Liu


[channels]

signalling=fxs_ks
usecallerid=yes
callwaiting=no
threewaycalling=no
transfer=yes
cancallforward=yes
; valores validos 256(32ms),512(64ms),1024(128ms)
echocancel=yes
echotraining=yes
echocancelwhenbridged=no
rxgain=0
txgain=0
group=1
callgroup=1
pickupgroup=1
immediate=no
faxdetect=incoming
;busydetect=yes
;busycount=10
answeronpolarityswitch=yes
hanguponpolarityswitch=yes
polarityonanswerdelay=600
;callprogress=no
progzone=es
channel => 1

zaptel.conf:


loadzone=es
defaultzone=es
fxsks=1

sip.conf:

[general]
context=default
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes

[101]
type=friend
secret=some
qualify=yes
nat=no
host=dynamic
canreinvite=no
context=SOME

[102]
type=friend
secret=some
qualify=yes
nat=no
host=dynamic
canreinvite=no
context=SOME

thanks for all!!!




___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users



___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Fixed quantity calls per extension

[Steve Edwards]

On Fri, 27 Apr 2007, Eric "ManxPower" Wieling wrote:


equis software wrote:

Hi, is there any way to configure  a number of simultaneus calls per
extension.
I need to rerstrict the simultaneus calls per service ( in extension 33 I
answer Service 1 and in extension 37 I answer service 2.

Example:
No more than 3 simultaneus calls to extension 33
No more than 15 simultaneus calls to extension 37


Yes. Any of the following:

1) Check the documentation for your IP phone.
2) Use the applications shown by "show applications like group" in the CLI.
3) Talk to your VoIP provider
4) Use Queues


Or, the question says "simultaneus" (sp) which could be interpreted as a 
conference in which case meetme and meetmecount would do the trick.


Thanks in advance,

Steve Edwards  [EMAIL PROTECTED]  Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] execute commands after hangup

[Jerry Geis]

I have a few commands I wish to run after a hangup.
It looks like only the first 2 commands are run after hangup.

I am using 1.4.3

How can I get the entire loop to run 10 times. ( I know my example just 
has noop's but its an example).


exten => h,1,Set(i=1)
exten => h,n,While($[${i} < 10])
exten => h,n,Noop(jerry)
exten => h,n,Set(i=$[${i} + 1])
exten => h,n,EndWhile
exten => h,n,Noop(jerry)

The only other item to know is this is a call connected to console/dsp.

<< Hangup on console >>
 == Spawn extension (default, 1041, 4) exited non-zero on 
'SIP/devcentos64_to_bt610tMM-081febf8'
   -- Executing [EMAIL PROTECTED]:1] 
Set("SIP/devcentos64_to_bt610tMM-081febf8", "i=1") in new stack
   -- Executing [EMAIL PROTECTED]:2] 
While("SIP/devcentos64_to_bt610tMM-081febf8", "1") in new stack
 == Spawn extension (default, h, 2) exited non-zero on 
'SIP/devcentos64_to_bt610tMM-081febf8'


THanks,

Jerry
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Fixed quantity calls per extension

[equis software]

I´m trying this:

exten => 99,1,Set(GROUP(99) = GROUP99)
exten => 99,2,GotoIf($[${GROUP_COUNT(99)}>1]?103)
exten => 99,3,Goto(context2,s,1)
exten => 99,103,Hangup

but doesn't work...I call to extension 99 from two different phones and
Asterisk sends both to 'context2'.



On 4/27/07, Eric ManxPower Wieling <[EMAIL PROTECTED]> wrote:


equis software wrote:
> Hi, is there any way to configure  a number of simultaneus calls per
> extension.
> I need to rerstrict the simultaneus calls per service ( in extension 33
I
> answer Service 1 and in extension 37 I answer service 2.
>
> Example:
> No more than 3 simultaneus calls to extension 33
> No more than 15 simultaneus calls to extension 37

Yes. Any of the following:

1) Check the documentation for your IP phone.
2) Use the applications shown by "show applications like group" in the
CLI.
3) Talk to your VoIP provider
4) Use Queues
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk hosted Callwaiting???

[Alex Balashov]

Manu,

  Perhaps it is possible to do this by running the call through an RTP 
proxy (there are various) that supports inserts muxing outside audio

feeds into the media stream?  Then you would just need to implement
some sort of middleware layer to map Asterisk SIP channels to corresponding
call flows in the RTP proxy and run outside commands to make this happen, I 
assume.


  That said, I can't find any information specifically on this subject from 
a given source.  But try looking at various open-source RTP proxies.


  At present, it appears Asterisk only has the capability to "barge" into a 
channel on the software layer (or Zaptel kernelspace, in the case of 
ZapBarge()) for the purpose of listening in.


-- Alex

On Fri, 27 Apr 2007, Manu Mehta said something to this effect:


Hi,

Is it possible to host call waiting service on Asterisk for a SIP device?
What i am trying to achieve is that while a SIP user is busy on a call and
a new call for that user comes in, asterisk should play the call waiting
tone to that user.
I have a vague idea that if i can get hold of the existing bridged channel
when a subsequent call is received, i can then redirect that channel to
play tone.
The problem is how can i get hold of the bridged channel in the first
place? Also is there a better way of accomplish call waiting.

TIA,

Manu Mehta

A R I C E N T

Plot-17, Sector 18, Gurgaon 122015,
Haryana, India

Main +91.124.4095888 x3274
Fax  +91.124.4095912



***  Aricent-Private   ***

***  Aricent-Unclassified   ***
"DISCLAIMER: This message is proprietary to Aricent  and is intended solely for the use of 
the individual to whom it is addressed. It may contain privileged or confidential information and should not be 
circulated or used for any purpose other than for what it is intended. If you have received this message in error, 
please notify the originator immediately. If you are not the intended recipient, you are notified that you are strictly
prohibited from using, copying, altering, or disclosing the contents of this message. Aricent accepts no responsibility for 
loss or damage arising from the use of the information transmitted by this email including damage from virus."


--
Alex Balashov <[EMAIL PROTECTED]>
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [asterisk-users] Best Wireless bridge for Polycoms

[Mike]

Michael and all those who replied,
 
This Linksys WBP54G does  seems to be what I need, but it also seems very
much made for Linksys phones.  Isn't there some sort of equivalent thing
that comes with it's own power supply (at the cost of needing another outlet
for the phone)?
 
Alternatively, where do I find an adapter for NA power that turns into 2V 5A
DC current?
 
Mike

  _  

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael Graves
Sent: Friday, April 27, 2007 13:45
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Best Wireless bridge for Polycoms


--Original Message Text---
From: Mike
Date: Fri, 27 Apr 2007 10:24:05 -0400

Hi, 

I'm stuck doing an install with Polycoms at a small office with no RJ-45.
They went wireless 100%, poor them. I insist on using Polycom unless it's
impossible because that's what I am standardized on for many reasons. 

What's the best way/device to turn a wired Polycom 501 (or any Polycom for
that matter) into a WiFi phone? 

Mike 


Linksys makes a device spcifically for this role so that their SPA series IP
phones can be connected to WIFI.

http://www.linksys.com/servlet/Satellite?c=L_Product_C2&childpagename=US%2FL
ayout&cid=1139961537989&pagename=Linksys%2FCommon%2FVisitorWrapper&lid=37989
54250B11

This device take power from the phone via a simple coax plug on the phone.
Could easily be powered externally with a wall wart.

Michael



___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Fixed quantity calls per extension

[Gordon Henderson]

On Fri, 27 Apr 2007, equis software wrote:


Hi, is there any way to configure  a number of simultaneus calls per
extension.
I need to rerstrict the simultaneus calls per service ( in extension 33 I
answer Service 1 and in extension 37 I answer service 2.

Example:
No more than 3 simultaneus calls to extension 33
No more than 15 simultaneus calls to extension 37


in sip.conf:
  call-limit=3

etc.

Gordon
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Free seating Agents and logged in / logged out indication

[Alexander Topolanek]

Hi,

I would like to set up a call center with free seating agents. However I
would like to indicate the agent status somehow on the terminal, to tell
the agent if she has been logged out due to non-answer.

Does anyone has a good idea how this can be achived?

best regards
-- 
Alexander Topolanek
http://www.topolanek.at

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Fixed quantity calls per extension

[Eric \"ManxPower\" Wieling]

equis software wrote:

Hi, is there any way to configure  a number of simultaneus calls per
extension.
I need to rerstrict the simultaneus calls per service ( in extension 33 I
answer Service 1 and in extension 37 I answer service 2.

Example:
No more than 3 simultaneus calls to extension 33
No more than 15 simultaneus calls to extension 37


Yes. Any of the following:

1) Check the documentation for your IP phone.
2) Use the applications shown by "show applications like group" in the CLI.
3) Talk to your VoIP provider
4) Use Queues
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Problem of configuring musiconhold.conf file

[gc]

- Original Message - 
From: "Tzafrir Cohen" <[EMAIL PROTECTED]>

To: 
Sent: Friday, April 27, 2007 12:55 PM
Subject: Re: [asterisk-users] Problem of configuring musiconhold.conf file



On Fri, Apr 27, 2007 at 12:31:16PM -0400, gc wrote:

Asterisk 1.2.17

When try to play moh, I can only use old format in musiconhold.conf file 
to play moh like this:


[moh_files]
default => /var/lib/asterisk/mohmp3,r

If I use the new format like this:

[default]
mode=quietmp3
directory=/var/lib/asterisk/mohmp3

I hear no music at all.

Can anybody tell me what is wrong?


ls -l /var/lib/asterisk/mohmp3

Do you see any relevant messages in the CLI when a channel is on hold?


Here is the message from logger:
Apr 27 14:09:41 VERBOSE[11172] logger.c: -- Executing 
Wait("SIP/lycin.net-b79332f8", "2") in new stack
Apr 27 14:09:41 DEBUG[11088] chan_sip.c: Stopping retransmission on 
'[EMAIL PROTECTED]' of Response 1: Match Found
Apr 27 14:09:41 DEBUG[11172] channel.c: Generator got voice, switching to 
phase locked mode
Apr 27 14:09:41 DEBUG[11172] channel.c: Scheduling timer at 0 sample 
intervals
Apr 27 14:09:43 VERBOSE[11172] logger.c: -- Executing 
Queue("SIP/lycin.net-b79332f8", "queue1|t") in new stack
Apr 27 14:09:43 VERBOSE[11172] logger.c: -- Started music on hold, class 
'default', on channel 'SIP/lycin.net-b79332f8'
Apr 27 14:09:43 DEBUG[11172] channel.c: Scheduling timer at 160 sample 
intervals

Apr 27 14:09:43 DEBUG[11172] app_queue.c: Everyone is busy at this time
Apr 27 14:09:43 DEBUG[11172] channel.c: Generator got voice, switching to 
phase locked mode
Apr 27 14:09:43 DEBUG[11172] channel.c: Scheduling timer at 0 sample 
intervals

Apr 27 14:09:43 DEBUG[11172] channel.c: Auto-deactivating generator
Apr 27 14:09:43 VERBOSE[11172] logger.c: -- Stopped music on hold on 
SIP/lycin.net-b79332f8
Apr 27 14:09:43 DEBUG[11172] channel.c: Scheduling timer at 0 sample 
intervals

Apr 27 14:09:48 DEBUG[11172] app_queue.c: Everyone is busy at this time
Apr 27 14:09:53 DEBUG[11172] app_queue.c: Everyone is busy at this time

What is the 'Auto-deactivation generator' ? It seems that this one cause moh 
to stop immediatly.


Gary Chen 
___

--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Best Wireless bridge for Polycoms

[Blake Krone]

How do you overcome the 302 moved temporarily messages? I tried using
a wireless bridge with one of my Cisco phones & Polycom 301's and when
I tried to receive a call it would give me the 302 error.

On 4/27/07, Michael Graves <[EMAIL PROTECTED]> wrote:

--Original Message Text---
From: Mike
Date: Fri, 27 Apr 2007 10:24:05 -0400

Hi,

I'm stuck doing an install with Polycoms at a small office with no RJ-45.
They went wireless 100%, poor them. I insist on using Polycom unless it's
impossible because that's what I am standardized on for many reasons.

What's the best way/device to turn a wired Polycom 501 (or any Polycom for
that matter) into a WiFi phone?

Mike


Linksys makes a device spcifically for this role so that their SPA series IP
phones can be connected to WIFI.

http://www.linksys.com/servlet/Satellite?c=L_Product_C2&childpagename=US%2FLayout&cid=1139961537989&pagename=Linksys%2FCommon%2FVisitorWrapper&lid=3798954250B11

This device take power from the phone via a simple coax plug on the phone.
Could easily be powered externally with a wall wart.

Michael



___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:

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



___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Fixed quantity calls per extension

[equis software]

Hi, is there any way to configure  a number of simultaneus calls per
extension.
I need to rerstrict the simultaneus calls per service ( in extension 33 I
answer Service 1 and in extension 37 I answer service 2.

Example:
No more than 3 simultaneus calls to extension 33
No more than 15 simultaneus calls to extension 37
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] New VICIDIAL astGUIclient Release: 2.0.3

[Matt Florell]

Hello,

We've released another update to our astGUIclient suite: 2.0.3

http://astguiclient.sf.net/

The client suite runs on most modern web browsers on almost any
GUI-capable operating system, and it includes the astGUIclient
client-side web app which extends your phone's functionality and the
VICIDIAL call center suite.
This package is free and GPL.
 (the suite is not an asterisk configuration tool)
This package is geared towards Asterisk installations with SIP,IAX or
Zap phones and Zaptel, IAX or SIP trunks.

For this release, we have focused on fixing bugs and adding several
new features including many new administrative functions and more
campaign options. We have also tested the suite on Asterisk versions
through 1.2.18.

All client web-apps and administration pages are available in English,
Spanish, Greek and German, with rough translations of French, Polish,
Italian, Portuguese and Brazillian Portuguese for the client web-apps
only.

Check out the project blog for more information:
http://astguiclient.blogspot.com

Let me know what you think.

Thanks,



MATT---
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Best Wireless bridge for Polycoms

[Michael Graves]

--Original Message Text---
From: Mike
Date: Fri, 27 Apr 2007 10:24:05 -0400

Hi, 

I'm stuck doing an install with Polycoms at a small office with no RJ-45.  They 
went wireless 100%, poor them.  I insist on using Polycom unless it's 
impossible because that's what I am standardized on for many 
reasons. 

What's the best way/device to turn a wired Polycom 501 (or any Polycom for that 
matter) into a WiFi phone? 

Mike 


Linksys makes a device spcifically for this role so that their SPA series IP 
phones can be connected to WIFI.

http://www.linksys.com/servlet/Satellite?c=L_Product_C2&childpagename=US%2FLayout&cid=1139961537989&pagename=Linksys%2FCommon%2FVisitorWrapper&lid=3798954250B11

This device take power from the phone via a simple coax plug on the phone. 
Could easily be powered externally with a wall wart.

Michael



___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] CDR changes in 1.4.3?

[François Delawarde]

Happens to me again, SIP->Zap or SIP->SIPProvider with a quite simple 
dialplan, it generates an 's' record in the context of both sides just 
like if it was doing a per-channel CDR instead of a per-call...




Scott Lykens wrote:

Hello all:

I upgraded to 1.4.3 last night and use MySQL for CDR.

I have noticed that 1.4.3 seems to log a lot of "crap" to CDR that
1.4.2 did not. I use a few macros in my dialplan to handle outgoing
calls (lcr type stuff) and in addition to the proper CDR for the call
itself I also have records to 's' in the same dest-context and entries
to 's' in the default context. Up to 3 CDRs are generated for one
outgoing call (SIP -> Zap channel) with one being the legit CDR and
two being the type described above.

My dialplan executes a ResetCDR after calling the lcr macro so that
the CDR is sane and accurate, however, it appears these "spurious" CDR
entries are generated by the call the ResetCDR even though I do not
call it with any options.

Am I missing something obvious here? I have read the ChangeLog but I
didn't see anything that addressed this particular issue.

Thanks for the help.

sl
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users



--

_

François Delawarde

Ingeniero de red

Tel: 918.03.92.51

E-mail: [EMAIL PROTECTED] 

_

WIRELESS MUNDI

http://www.wirelessmundi.com/

C/Isaac Newton, 1 - Oficina 26 · Parque Tecnológico de Madrid

28760 TRES CANTOS (Madrid)

Tlf./Fax: (+34) 918 03 92 51



La información contenida en este mensaje y en sus archivos adjuntos es 
CONFIDENCIAL y se dirige exclusivamente a sus destinatarios. Queda 
expresamente prohibida la utilización de la misma por cualquier persona 
distinta de los destinatarios de esta comunicación. Si usted ha recibido 
este mensaje por error le rogamos que lo comunique inmediatamente a 
WIRELESS MUNDI y lo borre al igual que todos sus documentos adjuntos. El 
correo electrónico no puede asegurar la confidencialidad ni la 
integridad de sus mensajes por lo que WIRELESS MUNDI no se hace 
responsable de tales errores u omisiones.


--0--

All information in this message and its attachments is confidential and 
may be legally privileged. Only intended recipients are authorized to 
use it. If you have received this transmission in error, please notify 
WIRELESS MUNDI immediately and delete this message and its attachments. 
E-mail transmissions are not guaranteed to be secure or error free and 
WIRELESS MUNDI does not accept liability for such errors or omissions.



___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Problems with Digium TE110P

[Antonio Carlos Theophilo Costa Junior]

Hi everyone

I have a Digium card TE110P and after plug it, turn on the computer and
configure, the LEDs don't light up in spite of what Digium FAQ says about the
LEDs:

When will the LED's light up on my TDM400P/TE110P/TE2XXP/TE4XXP?
... The TE110P LED's will light up RED when the span is configured and kernel
module is loaded. If configured correctly and a circuit or channel bank is
connected the LED should turn GREEN.
...

Below are the output of dmesg, lsmod, /etc/zaptel.conf and ztcfg -vvv. Did
anyone have this problem before?

Thanks
Antônio Theóphilo Júnior

[EMAIL PROTECTED]:~# dmesg | more
...
[42949385.79] Zapata Telephony Interface Registered on major 196
[42949385.79] Zaptel Version: 1.2.10 Echo Canceller: KB1
[42949385.84] e100: Intel(R) PRO/100 Network Driver, 3.4.14-k4-NAPI
[42949385.84] e100: Copyright(c) 1999-2005 Intel Corporation
[42949385.96] intel8x0_measure_ac97_clock: measured 61046 usecs
[42949385.96] intel8x0: clocking to 48000
[42949385.97] ACPI: PCI Interrupt :01:00.0[A] -> GSI 21 (level, low)
-> IRQ 209
[42949385.97] Controller version: 24
[42949385.97] FALC version: 
[42949385.97] TE110P: Setting up global serial parameters for E1 FALC V1.2
[42949385.97] TE110P: Successfully initialized serial bus for card
[42949385.97] Found a Wildcard: Digium Wildcard TE110P T1/E1
[42949386.00] ACPI: PCI Interrupt :01:08.0[A] -> GSI 20 (level, low)
-> IRQ 217
[42949386.02] e100: eth0: e100_probe: addr 0xff8fe000, irq 217, MAC addr
00:0C:F1:E7:1B:0B
[42949386.06] Registered tone zone 20 (Brazil)
[42949386.06] TE110P: Span configured for CAS/HDB3
[42949386.06] Calling startup (flags is 4099)
[42949386.10] CSLIP: code copyright 1989 Regents of the University of
California
[42949386.13] ISDN subsystem Rev:
1.1.2.3/1.1.2.3/1.1.2.2/1.1.2.3/1.1.2.2/1.1.2.2 loaded
[42949386.19] HiSax: Linux Driver for passive ISDN cards
[42949386.19] HiSax: Version 3.5 (module)
[42949386.19] HiSax: Layer1 Revision 2.46.2.5
[42949386.19] HiSax: Layer2 Revision 2.30.2.4
[42949386.19] HiSax: TeiMgr Revision 2.20.2.3
[42949386.19] HiSax: Layer3 Revision 2.22.2.3
[42949386.19] HiSax: LinkLayer Revision 2.59.2.4
[42949386.38] lp0: using parport0 (interrupt-driven).
...

[EMAIL PROTECTED]:~# lsmod
Module  Size  Used by
ipv6  287584  24
ext3  147848  1
jbd62996  1 ext3
dm_mod 63512  1
md_mod 76756  0
lp 13220  0
hisax 575824  0
isdn  153056  1 hisax
slhc8320  1 isdn
e100   43012  0
mii 7040  1 e100
wcte11xp   27424  0
zaptel192900  1 wcte11xp
crc_ccitt   3200  2 hisax,zaptel
snd_intel8x0   36380  0
snd_ac97_codec100512  1 snd_intel8x0
snd_ac97_bus3328  1 snd_ac97_codec
snd_pcm_oss56992  0
snd_mixer_oss  21248  1 snd_pcm_oss
snd_pcm96516  3 snd_intel8x0,snd_ac97_codec,snd_pcm_oss
snd_timer  27140  1 snd_pcm
i2c_i8106276  0
i2c_algo_bit   10760  1 i2c_i810
hw_random   6420  0
psmouse40836  0
snd59748  6
snd_intel8x0,snd_ac97_codec,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_timer
soundcore  11232  1 snd
snd_page_alloc 12296  2 snd_intel8x0,snd_pcm
i2c_core   23296  1 i2c_algo_bit
shpchp 50144  0
pci_hotplug30652  1 shpchp
intel_agp  25628  1
agpgart37580  1 intel_agp
parport_pc 38724  1
parport39624  2 lp,parport_pc
serio_raw   8580  0
pcspkr  3204  0
floppy 65252  0
evdev  11136  0
xfs   648156  10
exportfs7296  1 xfs
ide_generic 2432  0
ehci_hcd   34824  0
uhci_hcd   35984  0
usbcore   137732  3 ehci_hcd,uhci_hcd
ide_cd 36740  0
cdrom  42272  1 ide_cd
ide_disk   19968  13
piix   12420  1
generic 5892  0
thermal14728  0
processor  27080  1 thermal
fan 5764  0
capability  5896  0
commoncap   8192  1 capability
vga16fb14856  1
vgastate   11136  1 vga16fb
fbcon  8  70
tileblit3712  1 fbcon
font9216  1 fbcon
bitblit 7424  1 fbcon
softcursor  3200  1 bitblit

[EMAIL PROTECTED]:~# more /etc/zaptel.conf | egrep -v '^#'

span=1,0,0,cas,hdb3
cas=1-15:1001
dchan=16
cas=17-31:1001

loadzone=br
defaultzone=br

[EMAIL PROTECTED]:~# ztcfg -

Zaptel Configuration
==

SPAN 1: CAS/HDB3 Build-out: 0 db (CSU)/0

[asterisk-users] Re: Voicemail on Different Server, Voicemail with NFS

[JR Richardson]

I was afraid of an unavailable NFS mount hanging the app and I also
wanted to keep all of the communication over IAX for simplicity sake.  I
also hacked together my own "MWI over IAX".  I did write ups of how I
did both.

http://opensourcemadness.blogspot.com/2007/03/centralizing-asterisk-voic
email.html
http://opensourcemadness.blogspot.com/2007/03/propagating-asterisk-mwi-a
cross.html

- Jeremy


Nice hack.  This would work better in remote a server application,
different subnets, locations.  I like it, I'm adding to my bag of
tricks.  If you would, please post links on the wiki page for remote
vocemail/mwi.

Using NFS works great in a Cluster arrangement, all servers on the
same subnet, location.  I'll add to the wiki as well.

Thanks.

JR
--
JR Richardson
Engineering for the Masses
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] SIP<->H323 calls without proxying RTP

[Alex Balashov]

On Fri, 27 Apr 2007, Elman Efendiyev said something to this effect:


Hello,

Could somebody tell me is it possible to use asterisk without RTP proxying
in SIP<->H323 calls?
I mean exactly what canreinvite=yes option do in SIP<->SIP calls.
I don't need a transcoding, only a signaling conversion, and this is
possible with some softswitches, so i wondering what about asterisk.
Same question about H323<->H323 calls
I'm using NuFone Network's H323 cahhel


  As an addendum to this, I would be curious to know how to force Asterisk 
to behave like a signaling proxy[1] only, if possible.  "CANreinvite" 
doesn't mean "WILLreinvite" or "MUSTreinvite."


-- Alex

[1]  Yes, I know it's a B2BUA so it's not really a proxy.  But the intent
 here is to hand off the media path to the endpoints and not be
 involved in it.

--
Alex Balashov <[EMAIL PROTECTED]>
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Too many open files, asterisk crash

[Matthew J. Roth]

Rilawich Ango wrote:

Thanks for your reply.

You're welcome.  = )

What I ready do is:
add ulimit -n 65535 in safe_asterisk
increase value to 203380 in /proc/sys/fs/file-max
I don't think it can cause any problems, but I've never had to adjust 
anything in the /proc filesystem, and I'm administering a pretty busy 
server:


# date; asterisk -rx "show channels" | grep active
Fri Apr 27 12:55:44 EDT 2007
574 active channels
412 of 1000 max active calls (41.20% of capacity)

Both actions don't help much for the file descriptor growing.
It's possible that your version of Asterisk is leaking file 
descriptors.  Have you checked out the issue tracker at 
 to see if it's a known issue?  For perspective, 
Asterisk is currently using 2,260 file descriptors on my system.


# ls -l /proc/`cat /var/run/asterisk.pid`/fd/ | wc -l
2260

Do I need to reboot if I insert the following in /etc/security?
*   -   nofile  65535
I don't *think* a reboot should be necessary, but since limits.conf 
applies to login shells you should logout and log back in.  Note that 
while the '*' is documented as a wildcard for the domain, my copy of 
limits.conf doesn't mention using a '-' as the type:


# can have the two values:
#- "soft" for enforcing the soft limits
#- "hard" for enforcing hard limits
Can I identify or remove the file descriptors, which are unused, shown 
in lsof?


Can I reload some modules to reduce the unused file descriptor instead
of restart?
Once you properly configure your ulimits and restart Asterisk from a 
fresh login, you shouldn't have to take any additional steps.  If you 
continue to have problems and the number of file descriptors has 
actually exceeded 65,535, it's probably time to file a bug report.


Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Live conference call on now

[Dean Collins]

There is a live conference call on Asterisk regarding Adhearsion
occurring now. Check out www.X2Z.eu   for details.

 

 

Regards,

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
 +1-212-203-4357 Ph

    
 
 

 

 



image001.gif
Description: image001.gif
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] SIP<->H323 calls without proxying RTP

[Elman Efendiyev]

Hello,

Could somebody tell me is it possible to use asterisk without RTP proxying
in SIP<->H323 calls?
I mean exactly what canreinvite=yes option do in SIP<->SIP calls.
I don't need a transcoding, only a signaling conversion, and this is
possible with some softswitches, so i wondering what about asterisk.
Same question about H323<->H323 calls
I'm using NuFone Network's H323 cahhel

Thanks

--
Sincerely,
Elman Efendiyev
PROTECH INC.



___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Problem of configuring musiconhold.conf file

[Tzafrir Cohen]

On Fri, Apr 27, 2007 at 12:31:16PM -0400, gc wrote:
> Asterisk 1.2.17
> 
> When try to play moh, I can only use old format in musiconhold.conf file to 
> play moh like this:
> 
> [moh_files]
> default => /var/lib/asterisk/mohmp3,r
> 
> If I use the new format like this:
> 
> [default]
> mode=quietmp3
> directory=/var/lib/asterisk/mohmp3
> 
> I hear no music at all.
> 
> Can anybody tell me what is wrong?

ls -l /var/lib/asterisk/mohmp3

Do you see any relevant messages in the CLI when a channel is on hold?


-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Test

[C F]

Failed

On 4/26/07, gc <[EMAIL PROTECTED]> wrote:




___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:

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



___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [asterisk-users] Can asterisk record the duration of usersputting on hold?

[Alexander Lopez]

Cross posted from -users to -dev

I was looking at adding this functionality in last night.

I saw that in app_queue when a call is bridged it determines hold time.

Using the following:

holdtime = abs((now - qe->start) / 60);

and for queue.log the following:

(long) (callstart - qe->start)


My thoughts were that adding a timer to the hold in res_musiconhold
would allow us to calculate hold time while still being channel
agnostic.

Under the function of moh_alloc()

Do something like:
chan->holdtimestart = time_t


and under  moh_release()

chan->holdtimeend = time_t
chan->holdtimelast = (chan->holdtimeend - chan->holdtimestart)
chan->holdtime = chan->holdtime + chan->holdtimelast

chan->holdfreq = chan->holdfreq + 1


This would allow for a call to be placed on hold and have that time
addeded up as well as keep track of how many time a call was place on
hold.

It could then be reported as $CDR(callholdtime), this would be separate
from the value from app_queue or it could be inherited and then if an
agent placed a caller on hold it would add it in to the final number,
However being on hold 'waiting' to talk to an agent and being on hold
after an agent answers is two different values and should remain as
such.


Any thoughts

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Humberto Figuera
> Sent: Thursday, April 26, 2007 3:18 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Can asterisk record the duration of
> usersputting on hold?
> 
> Hi Xue Liangliang,
> 
> If you use queue's then look in queue_log
> 
> http://www.voip-info.org/wiki/index.php?page=Asterisk+log+queue_log
> 
> the COMPLETEAGENT and COMPLETECALLER events have this information.
> 
> COMPLETEAGENT(holdtime|calltime|origposition)
> The caller was connected to an agent, and the call was terminated
normally
> by the *agent*. The caller's hold time and the length of the call are
both
> recorded. The caller's original position in the queue is recorded in
> origposition.
> 
> COMPLETECALLER(holdtime|calltime|origposition)
> The caller was connected to an agent, and the call was terminated
normally
> by the *caller*. The caller's hold time and the length of the call are
> both
> recorded. The caller's original position in the queue is recorded in
> origposition.
> 
> 
> --
> Humberto Figuera - Using Linux 2.6.20
> Usuario GNU/Linux 369709
> Caracas - Venezuela
> GPG Key Fingerprint = 5AAC DF0C 00F4 2834 28BA  37AD 3364 01D1 74CA
0603
> ___
> --Bandwidth and Colocation provided by Easynews.com --
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Problem of configuring musiconhold.conf file

[gc]

Asterisk 1.2.17

When try to play moh, I can only use old format in musiconhold.conf file to 
play moh like this:

[moh_files]
default => /var/lib/asterisk/mohmp3,r

If I use the new format like this:

[default]
mode=quietmp3
directory=/var/lib/asterisk/mohmp3

I hear no music at all.

Can anybody tell me what is wrong?

Gary Chen




___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Unable to find a codec translation path from ilbc to ulaw

[Oliver Brandt]

Hi!

As the upstream of my DSL-connection is very slow, I'd like my
sip-phones to use iLBC to connect to my *. My gateway provider only
allows ulaw. Hence, I'd like to use the follwing setup:

SIP-phone <--iLBC--> Asterisk <---ulaw> PSTN-Gateway

I get the following error:

"Unable to find a codec translation path from ilbc to ulaw"

Setup SIP-phone:
disallow=all
allow=ilbc

Setup PSTN-Gateway:
disallow=all
allow=ulaw

I've googled for overn an houre. But no luck. So I'd really apreciate
any help!

Thanks!
Oliver
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Best Wireless bridge for Polycoms

[Dave Cotton]

On Fri, 2007-04-27 at 11:49 -0400, Alex Robar wrote:
> Hi Mike,
> 
> How close together are these phones? If you have a few clusters of
> them, you can use the Linksys WRT54G devices to act as wireless
> bridges (with some open source firmware - I use DD-WRT). Each device
> will give you 4 ports to plug into. It's not a particularly cost
> effective solution to provide one WRT54G per phone, but if they're
> clustered you could centralize one bridge and plug 4 phones into it. 

I don't know what the power supply voltage for the Polycoms is but
Linksys have the WBP54G to do just this with there phones it has a pass
through for the phone power supply.

 
-- 
Dave Cotton <[EMAIL PROTECTED]>


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] ZT_CHANCONFIG failed on channel1:Nosuchdeviceoraddress

[Tzafrir Cohen]

On Fri, Apr 27, 2007 at 07:11:48AM +1200, CSB wrote:

[snip]

> As suggested earlier I replaced this with:
> /etc/modprobe.d/zaptel
> options wctdm opermode=NEWZEALAND honormode=1 boostringer=1 fastringer=1

[snip]

> dmesg
> Zapata Telephony Interface Registered on major 196
> Zaptel Version: 1.2.17.1
> Zaptel Echo Canceller: KB1
> wctdm: Unknown parameter `honormode'

This is the problem

$ /sbin/modinfo wctdm | grep honor
parm:   fxshonormode:int

So if you actually need that parameter, it's called "fxshonormode".

options wctdm opermode=NEWZEALAND fxshonormode=1 boostringer=1 fastringer=1

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Best Wireless bridge for Polycoms

[Alex Robar]

Hi Mike,

How close together are these phones? If you have a few clusters of them, you
can use the Linksys WRT54G devices to act as wireless bridges (with some
open source firmware - I use DD-WRT). Each device will give you 4 ports to
plug into. It's not a particularly cost effective solution to provide one
WRT54G per phone, but if they're clustered you could centralize one bridge
and plug 4 phones into it.

Alex

On 4/27/07, Mike <[EMAIL PROTECTED]> wrote:


 Hi,

I'm stuck doing an install with Polycoms at a small office with no RJ-45.
They went wireless 100%, poor them.  I insist on using Polycom unless it's
impossible because that's what I am standardized on for many reasons.

What's the best way/device to turn a wired Polycom 501 (or any Polycom for
that matter) into a WiFi phone?

Mike



___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users





--
Alex Robar
[EMAIL PROTECTED]
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] zaptel/pri, early audio, dial()

[Håkon Nessjøen]

Hi,
 
Is it possible to have early audio while waiting for answer in a Dial()?
 
Say that I want to do this:
 
1,Progress()   // Establish early audio possibillites
2,Dial(SIP/user,20,z(repeated-musicfile))
 
Where z would be like a function for playing early audio.
Or z would just start MOH without Answer()'ing first.
 
Håkon Nessjøen
Loopback Systems AS
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] CDR changes in 1.4.3?

[Scott Lykens]

Hello all:

I upgraded to 1.4.3 last night and use MySQL for CDR.

I have noticed that 1.4.3 seems to log a lot of "crap" to CDR that
1.4.2 did not. I use a few macros in my dialplan to handle outgoing
calls (lcr type stuff) and in addition to the proper CDR for the call
itself I also have records to 's' in the same dest-context and entries
to 's' in the default context. Up to 3 CDRs are generated for one
outgoing call (SIP -> Zap channel) with one being the legit CDR and
two being the type described above.

My dialplan executes a ResetCDR after calling the lcr macro so that
the CDR is sane and accurate, however, it appears these "spurious" CDR
entries are generated by the call the ResetCDR even though I do not
call it with any options.

Am I missing something obvious here? I have read the ChangeLog but I
didn't see anything that addressed this particular issue.

Thanks for the help.

sl
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Best Wireless bridge for Polycoms

[Gordon Henderson]

On Fri, 27 Apr 2007, Mike wrote:


Hi,

I'm stuck doing an install with Polycoms at a small office with no RJ-45.
They went wireless 100%, poor them.  I insist on using Polycom unless it's
impossible because that's what I am standardized on for many reasons.

What's the best way/device to turn a wired Polycom 501 (or any Polycom for
that matter) into a WiFi phone?


I was going to suggest this:

  http://www.smartbridges.com/products/ap.asp

but it's discontinued, and their replacement units are all outdoor type 
things as far as I can see - but that's the sort of thing you're after - 
essentially a WiFi client device with an Ethernet port on it.


I know many cheap APs can be run in this mode, (often called client bridge 
mode, but it depends on the venduh) so it's worth while looking round for 
one that will sit on the desktop out of the way. You might want one with 
(eg) a 4-port ethernet switch which would then talk to 4 phones if they 
were in the same physical location (adjacent desks for exmaple)


Sometimes the client bridges do funny things to forge the outgoing MAC 
address though - the smartbridges did which confused older versions of 
DHCP server, so that might be a consideration.


Ah, a quick search reveals this:

  http://www.dlink.com/products/?sec=1&pid=292

I wasn't impressed with DLink Wi-Fi kit when I was using them in a 
community Wi-Fi broadband project some years ago, but maybe they've 
improved since.


Gordon
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Utilisation of multiple database tables in Asterisk

[David Florella]

Hi, 

 

I need to use a new table in my Asterisk database, to add new
data. I want to use the data of this new table in my Asterisk
app_voicemail.c source code. I want to know if someone has an idea how to do
it. 

 

Thank you.

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] How to configure a stun server for a sip peer

[Marco Ciacci]

Yes, i use Asterisk behind a retricted-cone NAT in many cases 
(portforwarding on router, nat=yes and externip=dyndns o static ip), 
but not ever is a possible solution, sometimes my client isn't 
router's owner (telco's router) and i can't do port forwarding.

I known that 1.4 has stun support, but i didn't find nothing :-(
Bye,

At 16.05 27/04/2007, you wrote:

On Fri, 27 Apr 2007, Marco Ciacci wrote:


HI all!
I'm looking for some infos to configure stun server support for a SIP peer.
I've installed Asterisk 1.4.3, but searching for stun support in 
chan_sip (sip.conf) i've found nothing, only a "misterious" externip = stun...

But where i have to put the ip of stun server?
No infos around Google and forum! :-)


I don't know is 1.4 supports a STUN server, 1.2 doesn't that I'm aware of.

What are you trying to achieve? If it's just asterisk behind NAT, then:

The externip= paramter in sip.conf lets asterisk know what the 
external IP address is, so it can put that in the outgoing SIP 
headers, and hope that incoming packets to that IP address are 
"magically" forwarded by the router. You need nat=yes too.


So, Eg. You have a router which has external facing IP address 
1.2.3.4 and doing NAT to internal hosts on 192.168.1.x. you put 
externip=1.2.3.4 in the sip.conf file and arrange the router to 
port-forward incoming 5060 to the asterisk box on 192.168.1.x. At 
the same time, you need to get the router to port-forward RTP on 
1-2 by default to the asterisk box too.


Then, external SIP devices use the external IP address 1.2.3.4 and 
it all "just works". (you may have issues if the external SIP 
devices are behind NAT, but that's another issue)


If asterisk supports STUN, then what the external STUN server will 
do is tell asterisk what it's external IP address is. (and maybe some more)


I'm sure there's a good reason for SIP having it's IP address 
encoded into it, but I can't think of one right now )-:


Gordon
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

--
This message has been scanned for viruses and
dangerous content by MailScanner, and is
believed to be clean.


--

Marco Ciacci

Asterisk Admin
Windows Server & Linux Admin
Security & Networking
@
Silog Srl. (Siena)
Tel. 0577271840
[EMAIL PROTECTED]


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [asterisk-users] Re: Voicemail on Different Server, Voicemail with NFS

[Porier, Jeremy M.]

I was afraid of an unavailable NFS mount hanging the app and I also
wanted to keep all of the communication over IAX for simplicity sake.  I
also hacked together my own "MWI over IAX".  I did write ups of how I
did both.

http://opensourcemadness.blogspot.com/2007/03/centralizing-asterisk-voic
email.html
http://opensourcemadness.blogspot.com/2007/03/propagating-asterisk-mwi-a
cross.html

- Jeremy

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of JR
Richardson
Sent: Thursday, April 26, 2007 6:32 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Re: Voicemail on Different Server,Voicemail
with NFS



> -Original Message-
> From: JR Richardson [mailto:[EMAIL PROTECTED]
> Sent: Saturday, June 17, 2006 2:30 PM
> To: asterisk-users@lists.digium.com; Douglas Garstang
> Subject: Voicemail with NFS (working, I think)
> 
> I'm using a stand-alone VM server and exporting the VM files ro for 
> MWI function only.  All my registration servers mount the remote NFS 
> share just to check MWI, all read-write functions to the VM files 
> occur on the VM server only.
> 
> On the registration servers, I mounted the remote VM share with this 
> in my fstab.conf:
> 
> 10.10.14.124:/var/spool/asterisk/voicemail  /mnt/vmserver   nfs
>  soft,nolock,timeo=1,retrans=1,bg,intr
> 
> all on one line of course
> 
> [10.10.14.124:/var/spool/asterisk/voicemail] is the voicemail server 
> and [/mn/vmserver] is the mounted location, [nfs] is the file system 
> type,
> 
> options: (the important stuff)
> [soft] If  an NFS file operation has a major timeout, then report an 
> I/O error to the calling program.  The default is to continue retrying

> NFS  file  operations indefinitely.
> [nolock] does not use locking.
> [timeo=1] redces the nfs share minor timeouts to 1 tenth of a second 
> [retrans=1] reduces the number of minor timeouts and retransmitts 
> needed to cause a major timeout.
> [bg] background the re-mount operation [intr] interupt the nfs file 
> operation for the calling program when a major timeout occurs
> 
> When mounting the nfs without these options, Asterisk became 
> immediately unstable when the remote nfs dropped off-line, but also 
> regained is operation immediately when the nfs became available again.
> 
> When mouting the nfs with these options, asterisk kept on ticking 
> without interruption, WMI was unavailable of course, but all other 
> functions worked.  The MWI came back on immediately after the nfs 
> share was restored.
> 
> Of course this is a hack and I'm patiently waiting for Digium to 
> implement the Remote MWI over IAX and incorporate into stable 
> releases.
> 
> Hope this helps.
> 
> JR
> 
> --
> JR Richardson
> Engineering for the Masses

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Best Wireless bridge for Polycoms

[Mike]

Hi,
 
I'm stuck doing an install with Polycoms at a small office with no RJ-45.
They went wireless 100%, poor them.  I insist on using Polycom unless it's
impossible because that's what I am standardized on for many reasons.
 
What's the best way/device to turn a wired Polycom 501 (or any Polycom for
that matter) into a WiFi phone?
 
Mike
 
 
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] No Audio with SIP to only one provider whenswitching servers

[Hadar Pedhazur]

Brad Sumrall wrote:

I would not rule your firewall out as the problem!
Port 5060 is only the authentication port, the rtp stream is normally 10,000
thru 20,000.
Some of your phone may have STUN modules on them.

Open 10,000 thru 20,000 and 5060 on the firewall.
Stick some holes in it for testing purposes.
Verify ports are open with telnet:port number "both ways", telnet is your
friend.
If it works, close the holes up and consult your firewall docs

Brad


Thanks for the response Brad (and Brian Capouch as well in a 
separate note!).


I was offline all day yesterday, but I can do more testing today.

Of course, it's quite possible that it's the firewall. That said, 
all other providers (including SIP) work, so it would have to be a 
reasonably tight number of ports that are open to the other 
providers, and a different set of ports that are closed that 
StanaPhone is trying to communicate on.


Anyway, more testing on the way ;-)

BTW, I run Shorewall (which is a cover for IPTABLES), and it 
usually logs every dropped packet, and I see _no_ rejections in 
the log file for source IP from StanaPhone and destination UDP 
ports on my machine. I'm running the same Shorewall "rules" 
(different version of Shorewall and different OS on the two linux 
boxes) on the box that works with StanaPhone...


Thanks again to both of you for the responses!


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Hadar Pedhazur
Sent: Wednesday, April 25, 2007 6:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] No Audio with SIP to only one provider
whenswitching servers

I have been running Asterisk for years on a machine with a public 
IP. Most recently, I have been running 1.2.17, from the day it 
came out, with no (noticeable) problems.


Yesterday, I switched over to a new server that is on the same 
public subnet, one higher than the original server.


I built 1.2.17 from source on that machine (as I did on the old 
server). My firewall on the new machine is configured identically 
to the old one as well.


All of my IAX connections just worked. All but one of my SIP 
connections just worked as well (which is why I can't believe it's 
a firewall issue).


StanaPhone, which I use for 2 incoming DIDs, registers correctly, 
and rings my phones correctly when a call comes in. However, once 
answered, there is dead silence in both directions, on 100% of the 
calls.


There isn't any problem on StanaPhone's side (which has provided a 
_fantastic_ service ever since I signed up!), because I can 
connect to them with X-Lite and receive calls with audio. More 
importantly, if I fire up Asterisk on the old server, it still 
works!!! I can connect with X-Lite to the new server, so the new 
server definitely accepts SIP connections, and audio works.


It's _not_ a codec problem. I verified that on both the working 
and non-working servers the connection is established with ulaw on 
both sides.


I have dumped the "peer" and the "channel" on both, while the call 
was active, and they look identical to me, except for the random 
bits associated with a particular connection. Here are the ones 
from the machine that fails:


*CLI> sip show peer XX


   * Name   : XX
   Secret   : 
   MD5Secret: 
   Context  : default
   Subscr.Cont. : 
   Language :
   AMA flags: Unknown
   CallingPres  : Presentation Allowed, Not Screened
   Callgroup:
   Pickupgroup  :
   Mailbox  :
   VM Extension : asterisk
   LastMsgsSent : 32767/65535
   Call limit   : 0
   Dynamic  : No
   Callerid : "" <>
   Expire   : -1
   Insecure : port,invite
   Nat  : RFC3581
   ACL  : No
   CanReinvite  : No
   PromiscRedir : No
   User=Phone   : No
   Trust RPID   : No
   Send RPID: No
   DTMFmode : rfc2833
   LastMsg  : 0
   ToHost   : sip.stanaphone.com
   Addr->IP : 204.147.183.18 Port 5060
   Defaddr->IP  : 0.0.0.0 Port 0
   Def. Username: 12345678
   SIP Options  : (none)
   Codecs   : 0x4 (ulaw)
   Codec Order  : (ulaw)
   Status   : OK (20 ms)
   Useragent:
   Reg. Contact :

new*CLI> sip show channel 
[EMAIL PROTECTED]


   * SIP Call
   Direction:  Outgoing
   Call-ID: [EMAIL PROTECTED]
   Our Codec Capability:   4
   Non-Codec Capability:   1
   Their Codec Capability:   4
   Joint Codec Capability:   4
   Format  ulaw
   Theoretical Address:204.147.183.18:5060
   Received Address:   204.147.183.18:5060
   NAT Support:RFC3581
   Audio IP:   AAA.BBB.CCC.DDD (local)
   Our Tag:as360c7ca5
   Their Tag:  0bd46ffd48e4fbffb3a68f13f8ad2599
   SIP User agent:
   Username:   87654321
   Peername:   12345678
   Original uri:   sip:204.147.183.55:1024
   Need Destroy:   0
   Last Message:   Tx: ACK
   Promiscuous Redir:  No
   Route: 

Re: [asterisk-users] How to configure a stun server for a sip peer

[Gordon Henderson]

On Fri, 27 Apr 2007, Marco Ciacci wrote:


HI all!
I'm looking for some infos to configure stun server support for a SIP peer.
I've installed Asterisk 1.4.3, but searching for stun support in chan_sip 
(sip.conf) i've found nothing, only a "misterious" externip = stun...

But where i have to put the ip of stun server?
No infos around Google and forum! :-)


I don't know is 1.4 supports a STUN server, 1.2 doesn't that I'm aware of.

What are you trying to achieve? If it's just asterisk behind NAT, then:

The externip= paramter in sip.conf lets asterisk know what the external IP 
address is, so it can put that in the outgoing SIP headers, and hope that 
incoming packets to that IP address are "magically" forwarded by the 
router. You need nat=yes too.


So, Eg. You have a router which has external facing IP address 1.2.3.4 and 
doing NAT to internal hosts on 192.168.1.x. you put externip=1.2.3.4 in 
the sip.conf file and arrange the router to port-forward incoming 5060 to 
the asterisk box on 192.168.1.x. At the same time, you need to get the 
router to port-forward RTP on 1-2 by default to the asterisk box 
too.


Then, external SIP devices use the external IP address 1.2.3.4 and it all 
"just works". (you may have issues if the external SIP devices are behind 
NAT, but that's another issue)


If asterisk supports STUN, then what the external STUN server will do is 
tell asterisk what it's external IP address is. (and maybe some more)


I'm sure there's a good reason for SIP having it's IP address encoded into 
it, but I can't think of one right now )-:


Gordon
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] How to configure a stun server for a sip peer

[Marco Ciacci]

HI all!
I'm looking for some infos to configure stun server support for a SIP peer.
I've installed Asterisk 1.4.3, but searching for stun support in 
chan_sip (sip.conf) i've found nothing, only a "misterious" externip = stun...

But where i have to put the ip of stun server?
No infos around Google and forum! :-)
Thank all, regards


--

Marco Ciacci

Asterisk Admin
Windows Server & Linux Admin
Security & Networking 


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] dialplan / problem with extension-length > 1

[Michael Kamleitner]

On 4/26/07, Michael Kamleitner <[EMAIL PROTECTED]> wrote:




On 4/26/07, Yuan LIU <[EMAIL PROTECTED]> wrote:
>
> >From: "Michael Kamleitner" <[EMAIL PROTECTED]>
> >Date: Wed, 25 Apr 2007 17:47:34 +0200
> >
> >however, I've continued to experiment again and again, and strangely it
>
> >seemed to work _some_ times, even when passing 4digit-extensions. now I
> >think I got the solution: it seems I have to press the extension digits
> a
> >little bit longer! let's say I hold each button at least 0.5sec,
> everything
> >works great. if I do a quick dial, asterisk seems to "loose" digits.
> >
> >any ideas why this might be?
>
> >From which channel do you make the call? (Zap? SIP?)  Looks like a DTMF
>
> detection problem.  If ZAP, you better use longer tone.  You can try
> relaxeddtmf in zapata.conf, but people generally recommend against
> it.  The
> card you use also matters.  Heavy echo could also interfere with DTMF.
>
> If SIP, the symptom you described would happen only to inband DTMF.  Try
> not
> to use inband if you can help it.
>


neither of both - I'm using a plain cellphone connected via bluetooth
(latest chan_cellphone-patch). maybe that's the root of my troubles...?

anyhow, after further testing I tend to beliebe that it's not the
_duration_ of actually pressing the key, but rather that asterisk requires a
minimum period of silence (~0.5sec) _between_ the key-presses to recognizes
everything correctly.

is there anything I can tweak according chan_cellphone?




I've found help at the chan_cellphone page... thx binsa!
(http://bugs.digium.com/view.php?id=8919)
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Too many open files, asterisk crash

[Jason Fuermann]

I've had mixed results with changing ulimit and not restarting asterisk. 
Best bet is to stop and start asterisk so that it calls a new shell


Rilawich Ango wrote:

Thanks for your reply.

What I ready do is:
add ulimit -n 65535 in safe_asterisk
increase value to 203380 in /proc/sys/fs/file-max
Both actions don't help much for the file descriptor growing.

What I want to know is:

Do I need to reboot if I insert the following in /etc/security?
*   -   nofile  65535

Can I identify or remove the file descriptors, which are unused, shown 
in lsof?


Can I reload some modules to reduce the unused file descriptor instead
of restart?


On 4/27/07, Matthew J. Roth <[EMAIL PROTECTED]> wrote:

Rilawich,

Here are a couple of my old posts that document how to guarantee that
Asterisk starts with an increased number of file descriptors available:

Too many open files
 


Asterisk Open File Limit
 



Good luck,

Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Asterisk hosted Callwaiting???

[Manu Mehta]

Hi,

Is it possible to host call waiting service on Asterisk for a SIP device?
What i am trying to achieve is that while a SIP user is busy on a call and 
a new call for that user comes in, asterisk should play the call waiting 
tone to that user.
I have a vague idea that if i can get hold of the existing bridged channel 
when a subsequent call is received, i can then redirect that channel to 
play tone. 
The problem is how can i get hold of the bridged channel in the first 
place? Also is there a better way of accomplish call waiting.

TIA,

Manu Mehta
 
A R I C E N T
 
Plot-17, Sector 18, Gurgaon 122015,
Haryana, India
 
Main +91.124.4095888 x3274
Fax  +91.124.4095912

 

***  Aricent-Private   ***

***  Aricent-Unclassified   ***
"DISCLAIMER: This message is proprietary to Aricent  and is intended solely for 
the use of 
the individual to whom it is addressed. It may contain privileged or 
confidential information and should not be 
circulated or used for any purpose other than for what it is intended. If you 
have received this message in error, 
please notify the originator immediately. If you are not the intended 
recipient, you are notified that you are strictly
prohibited from using, copying, altering, or disclosing the contents of this 
message. Aricent accepts no responsibility for 
loss or damage arising from the use of the information transmitted by this 
email including damage from virus."
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Call prority (QUEUE_PRO) in the queues

[gc]

Whenever I turn the weight option on, it locked the *.  It happens several 
times a day ( abount every two to three hours). When this happen, the 
incoming call can still connect to * but will not hear any music on hold. If 
I issue the 'show channels' command, it shows the connected channels 
continues going up and never released and eventfully * will run out of file 
descriptors and completely lock the *. When this happen, I have to use 
'kill -9 ' to kill * .  When I turn the weight option off, everything works 
fine. I searched the web and several people have the same problem. I also 
found a patch to fix this problem. Right now I am running * using this 
patch. It is up and running for about 24 hours and everything looks good 
right now. Some people have concern about this patch so it has never been 
put into * release. Since we do really need weight option, we have no choice 
but try this patch.


Gary Chen

- Original Message - 
From: "BJ Weschke" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 


Sent: Thursday, April 26, 2007 5:36 PM
Subject: Re: [asterisk-users] Call prority (QUEUE_PRO) in the queues



On 4/26/07, gc <[EMAIL PROTECTED]> wrote:



Suppose I have one agent login into two different queue and there are 
calls

waiting in both queues. If the calls in one queue has higher call prority
(set QUEUE_PRO to higher value) than the calls in other queue, will the
agent get the higher prority call first or the QUEUE_PRO has no effect?
We have an Asterisk server( 1.2.17 with CentOS) running as ACD. We are
having problem using weight option in the queue. I figure maybe I can use
QUEUE_PRO instead.



Queue priority will, unfortunately, only cover one queue. It cannot
cover and account for priorities of calls from more than one queue.
You will want "weight" for that. What's the problem you were having
with it?

BJ


--
Bird's The Word Technologies, Inc.
http://www.btwtech.com/
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users 

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] headsets for linksys/sipura phones?

[Per Jessen]

Erik Anderson wrote:

> On 4/26/07, Per Jessen <[EMAIL PROTECTED]> wrote:
>> I was just browsing my local suppliers list of headsets - not a
>> single
>> one with a single 2.5mm jack.  Either USB or 1-2 3.5mm jacks.
>>
>> Can anyone recommend a headset that works with e.g. SPA-921 and -941?
> 
> Try your local mobile phone supplier.  I used a headset that came with
> one of my cell phones, and it worked great w/ my SPA-941.

Not a bad idea  - which make was this for?  None of my phones (Ericsson,
Nokia) have a 2.5mm socket, they're all special/proprietary.


/Per Jessen, Zürich

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: AW: [asterisk-users] 7970 sip success

[Zachary Whitley]

I also have

nat=no
qualify=no

I haven't checked to see if they're necessary. I think I've read some
suggestions that the phone needs to be on the same subnet as the
asterisk server but I haven't been able to check that either. 

On Fri, 2007-04-27 at 09:19 +0200, René Enskat wrote:
> Mmm i have set it in my MySQL Database in the row: Variables
>  buggymwi = yes 
> 
> But can't see MWI
> 
> 
> Regards rene
> 
> -Ursprüngliche Nachricht-
> Von: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Im Auftrag von Zachary
> Whitley
> Gesendet: Freitag, 27. April 2007 00:09
> An: Asterisk Users Mailing List - Non-Commercial Discussion
> Betreff: Re: [asterisk-users] 7970 sip success
> 
> MWI also works with Asterisk 1.4.2 with buggymwi=yes in sip.conf
> 
> ___
> --Bandwidth and Colocation provided by Easynews.com --
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> 
> 
> ___
> --Bandwidth and Colocation provided by Easynews.com --
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
> 

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Re: MeetMe + IAX2 + Asterisk 1.2.18 = calls dropped

[Tony Mountifield]

In article <[EMAIL PROTECTED]>,
Dinesh Nair <[EMAIL PROTECTED]> wrote:
> 
> is there a patch for this against 1.2.18 ? it would sure help those who're
> tracking the release tarballs instead of having to svn and compile it. 

Have a look at:
http://svn.digium.com/view/asterisk/branches/1.2/channels/chan_iax2.c?r1=61866&r2=62037

The change is so trivial is would be easy to apply by hand.

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Re: FYI - PRS fraud

[Benny Amorsen]

> "SIP" == SIP  <[EMAIL PROTECTED]> writes:

SIP> Premium Rate Services think like 900 and 976 numbers in the
SIP> US, but not every country allocates a particular block of numbers
SIP> or prefixes to its premium rate services, so with some, they're
SIP> pretty close to impossible to block.

Perhaps someone could make a list and put it into ENUM form? It would
be great to be able to blacklist all those numbers.


/Benny


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

RE: [asterisk-users] Problem with SuSe 10.0 and zaptel 1.2.17

[Lee Archer]

It was fixed in 1.2.17.1. 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir
Cohen
Sent: 26 April 2007 21:05
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Problem with SuSe 10.0 and zaptel 1.2.17

On Wed, Apr 25, 2007 at 08:57:37AM +0100, Lee Archer wrote:
> I installed zaptel 1.2.17 and shortly afterwards got a problem of 
> calls not clearing properly.  I ran dmesg which showed
> 
>   Unable to handle kernel NULL pointer dereference at virtual
address 009c
>   printing eip:
>   f8a79fa8
>   *pde = 
>   Oops:  [#1]
>   Modules linked in: zttranscode button battery ac ipv6 edd
wcte11xp zaptel crc_ccitt i2c_i801 i2c_core tg3 generic shpchp
pci_hotplug parport_pc lp parport dm_mod ext3 jbd sg fan thermal
processor 3w_ piix sd_mod scsi_mod ide_disk ide_core
>   CPU:0
>   EIP:0060:[]Tainted: G U VLI
>   EFLAGS: 00010082   (2.6.13-15.15-default)
>   EIP is at zt_chanandpseudo_ioctl+0xd28/0xf70 [zaptel]
>   eax:    ebx: f74403ac   ecx:    edx: 
>   esi: b723f2b0   edi: f749ca78   ebp: 0046   esp: f50b3e28
>   ds: 007b   es: 007b   ss: 0068
>   Process asterisk (pid: 5430, threadinfo=f50b2000 task=f7bbf060)
>   Stack: 462f0587  41a0d314  01ff 0001
0246 0001
>    f50b3f38  005b 0001
dfcf089c f50b3ebc
>  f50b3efc f61ae400 f6a0d3b4 005b 005b f6a0d314
 0001
>   Call Trace:
>[] generic_file_aio_write+0x58/0xc0
>[] ext3_file_write+0x1b/0x93 [ext3]
>[] do_sync_write+0xb6/0x110
>[] zt_ioctl+0x93/0x100 [zaptel]
>[] zt_ioctl+0x0/0x100 [zaptel]
>[] do_ioctl+0x4e/0x60
>[] vfs_ioctl+0x4f/0x1c0
>[] sys_ioctl+0x37/0x70
>[] sysenter_past_esp+0x54/0x79
>   Code: ff 89 f8 89 f1 e8 75 88 77 c7 31 ff c7 85 94 06 00 00 00
00 00 
> 00 e9 77 f4 ff ff 8b 4c 24 24 e9 e4 f8 ff ff 8b 04 95 20 0d aa f8 <8b>

> 80 9c 00 00 00 e8 5d a9 6c c7 8b 44 24 20 8b 04 85 20 0d aa
> 
> I've since installed zaptel 1.2.16 again and it's fine.  Is anyone
else getting this problem?

Not me, but others do. Try 1.2.17.1 .


-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
###

This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange.
For more information, connect to http://www.f-secure.com/
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] 2 cards in a server

[Wilson Pickett]

I think you need to explain "control the call pass through those
cards" a little please.

On 4/27/07, Rilawich Ango <[EMAIL PROTECTED]> wrote:

Hi all,
  I have 2 cards, they are x100p and TDM400p (2 FXO and 2 FXS), in a
server.  Is it possible to control the call pass through those cards?
Any example for me to reference?
ango
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Can asterisk record the duration of users putting on hold?

[Xue Liangliang]

Hi, the holdtime in queue log entry is not what we want, that holdtime
only records the duration that caller stay in the queue before an
agent answers. However what we want is the duration that agent put the
customers on hold(i.e music on hold, for SIP, the device will send a
re-Invite as I attached last time),  actually I already find a way, in
sip.conf, there is a option callevents, set to yes, the hold and
unhold event will send to manager interface. In version 1.4, manager
interface can add a timestamp header for every event, that will help
to realize this report feature.

Regards,
Liangliang

On 4/27/07, Humberto Figuera <[EMAIL PROTECTED]> wrote:

Hi Xue Liangliang,

If you use queue's then look in queue_log

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

the COMPLETEAGENT and COMPLETECALLER events have this information.

COMPLETEAGENT(holdtime|calltime|origposition)
The caller was connected to an agent, and the call was terminated normally
by the *agent*. The caller's hold time and the length of the call are both
recorded. The caller's original position in the queue is recorded in
origposition.

COMPLETECALLER(holdtime|calltime|origposition)
The caller was connected to an agent, and the call was terminated normally
by the *caller*. The caller's hold time and the length of the call are both
recorded. The caller's original position in the queue is recorded in
origposition.


--
Humberto Figuera - Using Linux 2.6.20
Usuario GNU/Linux 369709
Caracas - Venezuela
GPG Key Fingerprint = 5AAC DF0C 00F4 2834 28BA  37AD 3364 01D1 74CA 0603
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users




--
Regards!
Liangliang
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] MeetMe + IAX2 + Asterisk 1.2.18 = calls dropped

[Dinesh Nair]

On Thu, 26 Apr 2007 12:39:32 -0400, Dave Miller wrote:

> Dave Miller wrote on 4/26/07 11:46 AM:
> > We upgraded our asterisk server to 1.2.18 last night to pick up the
> > security update.  Since then, any calls coming in on IAX2 links get
> > dropped if they try to enter a MeetMe conference room.
> > 
> > The log shows this:
> > 
> > Apr 26 08:33:16 NOTICE[27362]: chan_iax2.c:3167 iax2_read: I should
> > never be called! Hanging up.
> > 
> > I've temporarily worked around it by switching our inbound provider to
> > use SIP instead of IAX, but that's not an ideal solution.
> 
> Quick turnaround on the bug tracker, bug is resolved fixed already :)
> 
> http://bugs.digium.com/view.php?id=9600
> 
> guess that'll be fixed in the next release.
> 

is there a patch for this against 1.2.18 ? it would sure help those who're
tracking the release tarballs instead of having to svn and compile it. 

-- 
Regards,   /\_/\   "All dogs go to heaven."
[EMAIL PROTECTED](0 0)   http://www.openmalaysiablog.com/
+==oOO--(_)--OOo==+
| for a in past present future; do|
|   for b in clients employers associates relatives neighbours pets; do   |
|   echo "The opinions here in no way reflect the opinions of my $a $b."  |
| done; done  |
+=+
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Too many open files, asterisk crash

[Rilawich Ango]

Thanks for your reply.

What I ready do is:
add ulimit -n 65535 in safe_asterisk
increase value to 203380 in /proc/sys/fs/file-max
Both actions don't help much for the file descriptor growing.

What I want to know is:

Do I need to reboot if I insert the following in /etc/security?
*   -   nofile  65535

Can I identify or remove the file descriptors, which are unused, shown in lsof?

Can I reload some modules to reduce the unused file descriptor instead
of restart?


On 4/27/07, Matthew J. Roth <[EMAIL PROTECTED]> wrote:

Rilawich,

Here are a couple of my old posts that document how to guarantee that
Asterisk starts with an increased number of file descriptors available:

Too many open files

Asterisk Open File Limit


Good luck,

Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] 2 cards in a server

[Rilawich Ango]

Hi all,
 I have 2 cards, they are x100p and TDM400p (2 FXO and 2 FXS), in a
server.  Is it possible to control the call pass through those cards?
Any example for me to reference?
ango
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Too many open files, asterisk crash

[Knud Müller]

Alex Balashov wrote:


On Thu, 26 Apr 2007, Knud Müller said something to this effect:


Tzafrir Cohen wrote:


On Thu, Apr 26, 2007 at 08:43:17AM -0500, Jason Fuermann wrote:

1024 open files will get you around 120 concurrent calls. 



8 file-handles per call? Why is that?


Depends on what channel type you use and what you do in the dialplan. 
A file handle is also used for an open socket. I always ran into that 
kind of problem when conferencing and play soundfiles to the 
conference members. Try before starting asterisk to ulimit your shell 
to a higher value of file handles. AFAIK its ulimit -n.



  Some aspects of the limit on concurrent file descriptors are still 
on the kernel side, unfortunately, depending on what you're doing with 
them.



This is kind of irrelevant for all using linux..
Yep, we have a solaris (10)  box. Asterisk installed fine. But has a 
much tighter open file limit.
Solaris started ages ago with a one byte file handle (256 Open Files) 
they decided to be binary compatible in newer versions. When the need 
came for more open files the fopen() method can handle more files under 
solaris, but the open method  is still limited to 256 Files. The 
asterisk source code showed us that both open methods are spread widely 
over the sources. We started to change into fopen wherever possible but 
been still unable to overcome these limits on solaris by now.



For instance, if any kind of asynchronous polling of descriptors is done
with select(), know that FD_SETSIZE is hard-coded into the kernel 
headers.


  If it's a simple issue of open files, there should be no problem 
increasing it on the ulimit side.  But more complex operations involving
concurrent use of descriptors and sockets may require changes to 
hard-coded parameters.


-- Alex

--
Alex Balashov <[EMAIL PROTECTED]>



___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users
 




--
Knud A. Müller


___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] can´t anserd the call

[Josu Lazkano Lete]

hello, I have instaled a analog line, and when I call on the console apears 
that:

I want to redirect the call to 101 extension.

*CLI> -- Starting simple switch on 'Zap/1-1'
  == Starting Zap/1-1 at default,s,1 failed so falling back to exten 's'
  == Starting Zap/1-1 at default,s,1 still failed so falling back to context 
'default'
Apr 27 08:15:53 WARNING[3494]: pbx.c:2377 __ast_pbx_run: Channel 'Zap/1-1' sent 
into invalid extension 's' in context 'default', but no invalid handler
-- Hungup 'Zap/1-1'
-- Starting simple switch on 'Zap/1-1'
Apr 27 08:15:58 NOTICE[3497]: chan_zap.c:6223 ss_thread: Got event 18 (Ring 
Begin)...
Apr 27 08:16:00 NOTICE[3497]: chan_zap.c:6223 ss_thread: Got event 2 
(Ring/Answered)...
  == Starting Zap/1-1 at default,s,1 failed so falling back to exten 's'
  == Starting Zap/1-1 at default,s,1 still failed so falling back to context 
'default'
Apr 27 08:16:00 WARNING[3497]: pbx.c:2377 __ast_pbx_run: Channel 'Zap/1-1' sent 
into invalid extension 's' in context 'default', but no invalid handler
-- Hungup 'Zap/1-1'


mi configuration files are this:

extensions.conf:

[general]
static=yes
writeprotect=yes
;autofallthrough=yes
;clearglobalvars=no
;priorityjumping=no

[SOME]
exten => 101,1,Dial(SIP/101,30,Ttm)
exten => 101,2,Hangup

exten => 102,1,Dial(SIP/102,30,Ttm)
exten => 102,2,Hangup

[incoming]
exten => s,1,Wait(1)
exten => s,2,Answer()
exten => s,3,Dial(SIP/101,30,Ttm)

[outgoing]

exten =>_94XXX,1,Dial(ZAP/g1/${EXTEN},45,tTwW)
exten =>_94XXX,2,Hangup()
exten =>_94XXX,102,Hangup()

zapata.conf:

[channels]

signalling=fxs_ks
usecallerid=yes
callwaiting=no
threewaycalling=no
transfer=yes
cancallforward=yes
; valores validos 256(32ms),512(64ms),1024(128ms)
echocancel=yes
echotraining=yes
echocancelwhenbridged=no
rxgain=0
txgain=0
group=1
callgroup=1
pickupgroup=1
immediate=no
faxdetect=incoming
;busydetect=yes
;busycount=10
answeronpolarityswitch=yes
hanguponpolarityswitch=yes
polarityonanswerdelay=600
;callprogress=no
progzone=es
channel => 1

zaptel.conf:


loadzone=es
defaultzone=es
fxsks=1

sip.conf:

[general]
context=default
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes

[101]
type=friend
secret=some
qualify=yes
nat=no
host=dynamic
canreinvite=no
context=SOME

[102]
type=friend
secret=some
qualify=yes
nat=no
host=dynamic
canreinvite=no
context=SOME

thanks for all!!!___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk 1.4 Conference with G.722

[TienSen Chong]

I haven't tried the app_conference yet. I want to know if the conference is
consisting of 3 users with G.722, does the app_conference perform
transcoding? If it is not, then app_conference will solve the issue of
having conference consists of only G.722 user since no transcoding is
needed. Is my understanding correct?

Regards,
chong


On 4/26/07, Thomas Kenyon <[EMAIL PROTECTED]> wrote:


TienSen Chong wrote:
> Hi all,
>
> I am having problem with conference call (meetme feature) using G.722
> phone. G.722 phone to phone is working fine. I suspect this is due to
> the fact that Asterisk 1.4 only support G.722 passthrough.
>
This will be the case, Meetme transcodes the audio (to slin iirc), where
it mixes it.

> Any ideas how this problem can be fixed.
>
Have you tried using app_conference?
To be honest, I don't know how you would be able to have more than 2
people in a call without some transcoding going on.
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Attended Transfer of a queue call fails

[Alexander Topolanek]

Hi,

I'm using Grandstreams as the agents phone of a queue. Attended
transfers in a normal situation (direct call to the extension) work
fine, but when the agent has a queue call and tries to transfer it to
another sip extension the called party is hung up. 

Transfer is to pick another line on the Grandstream, dial the other
extension, wait for a pickup, and then press the transfer button + the
parked line on the Grandstream. I can reproduce that in several
asterisk, and the inbound call comes from an mISDN channel or a sip
channel, and with similar results using a SNOM 190 terminal

Asterisk is a 1.2.14-BRIstuffed-0.3.0-PRE-1x built on Gentoo

any ideas?

best regards
-- 
Alexander 

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

AW: [asterisk-users] 7970 sip success

[René Enskat]

Mmm i have set it in my MySQL Database in the row: Variables
 buggymwi = yes

But can't see MWI


Regards rene

-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Im Auftrag von Zachary
Whitley
Gesendet: Freitag, 27. April 2007 00:09
An: Asterisk Users Mailing List - Non-Commercial Discussion
Betreff: Re: [asterisk-users] 7970 sip success

MWI also works with Asterisk 1.4.2 with buggymwi=yes in sip.conf

___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users



___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Calllog

[Suity Zsolt]

Asterisk wrote:

Hi guys,

 

I have an IVR configured in my PBX, which callers use to browse thru the 
list of stores. Once they choose a store, the call gets redirected to 
that store (obviously using Dial() application). Now, my question is:


 

Each of this calls is logged in the calllog as one entry. How could I 
configure my dialplan so that that portion of the call, which is in fact 
just browsing thru the IVR, gets logged as one entry, and that portion 
of the call, which is an outbound call, as another entry?



Try, ForkCDR
You would have two entries in log. The first line will hold the total 
times (IVR+Dial) and the second line will hold the length of Dial.




--
Suich
___
--Bandwidth and Colocation provided by Easynews.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users