[Asterisk-Users] Resolving QOS problems

[Chris Miller]

I'm looking for advise on troubleshooting QOS problems. After much 
searching and reading online (Google, Voip-Info Wiki, etc.) I don't feel 
any closer to finding the right tools to solve my problem. Any info you 
would like to share would be much appreciated, and I'm sure the thread 
will server others in the future.


The problem :
-

I'm having intermittent problems with the audio cutting out on calls. At 
the same time the audio problems occur, I often see these in the "full" 
log :


Received iseqno 122 not within window 123->123

These range from sounding like bad cell phone calls, to the audio track 
cutting out in one or both directions for up to 20-30 seconds.


I also see dropped calls that seem to be a result of the IAX connection 
going away.


The environment :
-

I've got an * server located at a data center with good connectivity, 10 
hops to my IAX provider, and ~34ms ping times. They (IAX provider) use 
Cogent which concerns me a bit, but I'm not ready to jump to conclusions 
just yet.


My IP phone is connected via "enhanced" DSL (static addresses, no PPPoE) 
and I'm 12 hops away from my * server. My DSL provider has direct 
connectivity and peering agreements with the data center my server is 
located in. I've set QOS priority on the LAN port (Linksys router) the 
phone is connected to, and I've dropped the MTU to 576 as suggested for 
lower speed links. (1.5Mbs/384kbps in my case). Both these changes 
seemed to make an improvement over previous calls. Currently I don't 
believe the bulk of my problems to be between the phone and the * 
server. testyourvoip.com tests consistently show a 4.4 score (the 
maximum for ulaw) and rarely shows errors.


Ulaw is the codec used for both the SIP calls and IAX trunk.

What I'm looking for :
--

I'm trying to determine the cause and location of the problem between my 
* server and the IAX provider (and possibly my IP phone), and see what 
if anything I can do to reduce the occurrence of these drop outs. I'm 
looking for a couple of things :


1. A method of monitoring RTP/IAX traffic QOS at the PBX in real time.
2. Tools that might be used to determine the location of the problem.
   I.E. An RTP/IAX "traceroute" tool.

What I'm hoping to find is something that either integrates directly 
with *, or captures live RTP/IAX traffic and provides real time 
statistics on calls.


What I've found :
-

I saw Telchemy's VQMON_EP product, but it's unclear how it would work 
with Asterisk. Many other companies in this market seem to leverage off 
of Telchemy's products.


http://www.telchemy.com/partners.html
http://www.voiptroubleshooter.com/tools/voiptr_tools.htm

All of the products above seem to be aimed at large enterprises with 
deep IT pockets. I wouldn't mind ponying up a reasonable sum for a tool 
that does the job, but I lack the time to thoroughly evaluate everything 
that may be out there.


I haven't found much on the open source front. I've seen "Windows RTP 
Quality Monitor" which might be useful, but it's beta and hasn't been 
updated in over a year.


It seems to me that Ethereal might be integrated with a graphical tool, 
and if nothing else provide postmortem statistics on a phone call.


Request for comments :
--

What are people using to troubleshoot these problems? What commercial 
software works for you? What open source projects are you using? Do you 
have suggestions on projects that might be glued together to provide 
this functionality?


Thanks in advance.

Chris
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Re: [Asterisk-Users] Re: Welcome to the "Asterisk-Users" mailing list

[Francesco Peeters]

On Tue, September 20, 2005 2:10, chuck gelm said:

> What is the URL of the "list archive"?
As listed at the bottom of each email on the list:
http://lists.digium.com/mailman/listinfo/asterisk-users

You can also use google: "SEARCH KEYWORDS site:lists.digium.com"

> The 'wiki' seems broken.
It's working fine from here

>
>   I want to connect to a single home POTS from an internet connection.
> i.e. I want to be able to make telephone calls on my home POTS
> from anywhere that I can obtain an internet connection with
> a device with a microphone and a speaker. e.g. My laptop and a
> headphone with boom microphone.  Does 'Asterisk' do this?
> Should I look elsewhere for this task?

It's a bit overkill, but it can certainly do it!

>
>   If this is answered in the list archive, please respond with the
> URL of the list archive.

As this is the premise of (*), being a software PABX, the entire list is
*full* of posts on this topic...

You might want to search for Asterisk @ Home (aka AAH, [EMAIL PROTECTED]) 
though...

>
> Best regards,
> Chuck
>

PS: Please do not reply to an existing email to start a new thread,
neither with or without changing subjects, etc. It really screws up
threaded email clients. I had to page back all the way to my first email
to find your question, as it was sorted under *my* welcome email... (I
only did it because it showed 1 unread email, and I do not like unread
emails hanging around. Usually I have less time and just click 'mark all
read' when this happens, and I know there are many more. Also your mail
would have never come up if I had the thread marked for deletion... So
it's not only courteous, it's also good sense! )

Just click new and send to [EMAIL PROTECTED] Cleaner,
easier and it doesn't screw up threading in mailclients... (You can add
the list to your address book too!)

Good luck!

-- 
Francesco Peeters

GPG Key = AA69 E7C6 1D8A F148 160C  D5C4 9943 6E38 D5E3 7704
If your program doesn't recognize my signature, please visit
http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate.
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[Asterisk-Users] "Stopping retransmission on" messages

[Chris Miller]

I'm seeing a number of these logged in "full" while my * system is idle, 
but I haven't found a good description of what they mean. Can someone 
oblige? I have a single SIP phone registered and an IAX trunk.


Chris

Sep 19 22:13:44 DEBUG[18720]: (Provisional) Stopping retransmission (but 
retaining packet) on '[EMAIL PROTECTED]' 
Request 732: Found
Sep 19 22:13:44 DEBUG[18720]: Stopping retransmission on 
'[EMAIL PROTECTED]' of Request 732: Found
Sep 19 22:13:44 DEBUG[18720]: (Provisional) Stopping retransmission (but 
retaining packet) on '[EMAIL PROTECTED]' 
Request 733: Found
Sep 19 22:13:44 DEBUG[18720]: Stopping retransmission on 
'[EMAIL PROTECTED]' of Request 733: Found

Sep 19 22:13:44 DEBUG[18720]: Registration successful
Sep 19 22:13:44 DEBUG[18720]: Cancelling timeout 1882
Sep 19 22:15:29 DEBUG[18720]: Scheduled a registration timeout # 1886
Sep 19 22:15:29 DEBUG[18720]: (Provisional) Stopping retransmission (but 
retaining packet) on '[EMAIL PROTECTED]' 
Request 734: Found
Sep 19 22:15:29 DEBUG[18720]: Stopping retransmission on 
'[EMAIL PROTECTED]' of Request 734: Found
Sep 19 22:15:29 DEBUG[18720]: (Provisional) Stopping retransmission (but 
retaining packet) on '[EMAIL PROTECTED]' 
Request 735: Found
Sep 19 22:15:29 DEBUG[18720]: Stopping retransmission on 
'[EMAIL PROTECTED]' of Request 735: Found


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[Asterisk-Users] TE410 stop responding

[Chee Foong]

Does anyone know why 
my TE410 stop respoding? usually the light on the card will show red when the 
cable is unplug from the card. But now it shows green even if the cable is not 
plugged in. The other TE410 card on the same machine works fine 
though.
 
The last message 
generated from the card is something saying the D channel link down. But i have 
this message before and the D channels will usually be up again 
automatically.
 
Is this a hardware 
issue or software one?
 
Anyone experience 
this?
 
 
CCF
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Re: [Asterisk-Users] Point to Point with Fritz Card ...

[Craig Guy]

You will need to use the mISDN drivers - the AVM CAPI drivers will not 
support PTP.  It is possible to use mISDN with chan_capi but chan_misdn 
would be easier.


Craig

- Original Message - 
From: "Joao Correia" <[EMAIL PROTECTED]>

To: 
Sent: Tuesday, September 20, 2005 4:57 AM
Subject: [Asterisk-Users] Point to Point with Fritz Card ...



Hello all,

Does anyone has any experience with Point to Point Fritz Card and 
Asterisk ?


I have a BRI access Point to Multipoint working fine but I can only  have 
3 numbers.


The phone telco said that if they change to Point to Point I can have  10 
numbers.


Does anyone has any experience with Point to Point ?

Best regards
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Re: [Asterisk-Users] ztdummy configuration help

[Lohit]

Hi Kurt,

I had the same problem, it seems that asterisk requires a timer for
meetme() app to work. You need to install ztdummy.o module into the
kernel and then restart asterisk.

Next obvious question will be, where do I get ztdummy??

Well, If you open the Makefile in zaptel source directory on line
number 61 you will see ztdummy commented. Please uncomment that and run
make and then make install, this will put the ztdummy.o module into
appropriate place in /lib/modules/

After that do a modprobe ztdummy and then restart asterisk.

That is it. You are ready for conference calls.

BTW here is what i am using.

zaptel-1.0.9.2
libpri-1.0.9
asterisk-1.0.9
Kernel version 2.4.18-14

-LohitOn 9/19/05, kurt x <[EMAIL PROTECTED]> wrote:
Upon setting up and configuring the my extension.conf, meetme.conf andfollowing the instruction outlined at this web page:http://www.voip-info.org/wiki-Asterisk+timer+ztdummy
I get the following errors when calling the meetme number.Executing Wait("SIP/216.53.118.2-f41196e0", "1") in new stack-- Executing MeetMe("SIP/216.53.118.2-f41196e0", "|sicp") in new stack
-- Playing 'conf-getconfno' (language 'en')  == Parsing '/etc/asterisk/meetme.conf': FoundSep 19 13:51:22 WARNING[14066]: chan_zap.c:757 zt_open: Unable to open'/dev/zap/pseudo': No such file or directory
Sep 19 13:51:22 ERROR[14066]: chan_zap.c:6687 chandup: Unable to dupchannel: No such file or directorySep 19 13:51:22 WARNING[14066]: app_meetme.c:227 build_conf: Unable toopen pseudo channel - trying device
Sep 19 13:51:22 WARNING[14066]: app_meetme.c:230 build_conf: Unable toopen pseudo device-- Playing 'conf-invalid' (language 'en')Sep 19 13:51:23 WARNING[14066]: file.c:554 ast_readaudio_callback:
Failed to write frame-- Playing 'conf-getconfno' (language 'en')Any help is greatly appreciated.Kurt___--Bandwidth and Colocation sponsored by 
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-- With warm regardsLohit
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[Asterisk-Users] Buy a digium hardware

[Leopoldo Rodríguez H]

Where i can buy a digium hardware TDM400P in Mexico

is there a hardware with more than 4 FXS/FXO ports (8, 12, 24)? that is supported by Asterisk*

Regards

Leopoldo
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Re: [Asterisk-Users] TDM400P stops answering

[Andy Howell]

> I also see this from time to time. Running:
> 
> Asterisk CVS-Nv1-0-9-09/02/05-14:07:10 built by [EMAIL PROTECTED] on a i686 
> running Linux
> 
> Setup a little cron job that, in the wee hours, does a: 
> 
> service asterisk restart
> 
> and then * will start answering the lines again. Not ideal, but..

Chris,

Thanks. I saw another post mentioning that the top slot of the Dell
Optiplex was not reliable to interupts. I kind of doubt that since there
are just working off the same riser. This a low profile desk-top model.

I switched it anyway. We'll see. If this does not do it, then I'll try
the updating the zaptel code as Kevin suggested.

There is also a utility called zttool. Looks like it should show me if
there are missed interupts.

Thanks,

Andy

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Re: [Asterisk-Users] Zap calls dropping just after answer

[Jesse Keating]

On Mon, 2005-09-19 at 16:37 -0700, Jesse Keating wrote:
> In my case, Zap/1 is the first channel of a 24 channel T1 line to a
> Fujitsu PBX.  All seems configured right as I can dial out and the cell
> phone will ring.  This is very frustrating, can anybody help out w/
> this?
> 

This was solved.  My T1 link to the PBX needed to use em signaling
instead of fxs_ks.

-- 
Jesse Keating
GameHouse -- Systems Engineer

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[Asterisk-Users] MWI indicator HINT on Snom thru IAX?

[Colin Anderson]

I have many remote locations that dial into a central server to retrieve
voicemail via IAX. Outbound calls are handled as SIP calls from a Snom to a
local (to them) Asterisk server that dials the main server thru IAX. I have
trained them to check their voicemail via the emailed WAV file, however some
of them are, how shall I put it, idiots*, and insist that they *have* to
have the MWI indicator light on their phone and / or a stutter tone to
indicate the presence of voicemail. Is there a way to propigate MWI hints
through IAX? I have read snippets in the list archives that indicate some
people are working on it, but other than that I haven't seen anything. Has
anyone done this, or is there a way I could fake it? tia

*Actually, it's largely unfair to say this. This is regular expected
behavior, it just frustrates me that they are unable to forget the old way
of doing things.
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Re: [Asterisk-Users] asterisk frequently dead

[stevanus]

Hi,

Thanks for the reply..
But the biggest problem here is that people using asterisk will get
dissapointed, sometimes mad because their call being dropped off when
asterisk is dead..

Any suggestions anyone?

Regards,

Stevanus

Moises Silva wrote:
this is not a solution, more a workaround, you can try
using svscan service, so when down will automagically briged up.
  
  On 9/15/05, stevanus <
[EMAIL PROTECTED]> wrote:
  Hi,

I've tried upgrade my asterisk to 1.0.9...
It's now seemed that asterisk is more stable but it's still dead by
itself occasionally..

Output from gdb yield this:

...

Reading symbols from /lib/libgcc_s.so.1...done.
Loaded symbols for /lib/libgcc_s.so.1

#0  0x00a597a2 in _dl_sysinfo_int80 () from /lib/ld-linux.so.2
(gdb)

...

Actually the information giving by gdb is far more detail..Just keep it
brief here to keep the space small.

If anyone want to help me, then I'll send it entirely..

Any comments/thoughts will be greatly appreciated.

Thanks,

Best regards,

Stevanus

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[Asterisk-Users] need example about sjphone with asterisk

[julien bossart]

Hi all,
 
I am new to this forum.Say hello to all.
I need some help to make a example using sjphone with
asterisk (which will fonction as SIP server).
 
I use Fedora core 4, asterisk release version 1.2.
Can you give me sip.conf and extension.conf example?
Thank you so much.
 
Julien
 
 
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Re: [Asterisk-Users] Voicemail

[Chris Coulthurst]

Oh btw: here's my voicemail.conf line for just getting the number, not the 
name:


emailbody=Fm:${VM_CIDNUM}\n${VM_DATE}\nDur:${VM_DUR}\n

Chris Coulthurst
[EMAIL PROTECTED]

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

To: "Damon Estep" <[EMAIL PROTECTED]>
Cc: "Asterisk Users Mailing List - Non-Commercial Discussion" 


Sent: Monday, September 19, 2005 6:55 AM
Subject: Re: [Asterisk-Users] Voicemail


AHHH HA!  Yes that answered all of my questions.

Ok.. everything is right with the world, except I still can't get the
'pager' e-mail address to not be
[EMAIL PROTECTED]

I have a serveremail= string and that gets set for e-mails.. but I
don't see anything about a pager string.   I set the pagerfromstring
but that also seems to not change the e-mail address used for pagers.

On 9/18/05, Damon Estep <[EMAIL PROTECTED]> wrote:

Download cvs head and look at
/usr/src/asterisk/configs/voicemail.conf.sample

All of the variables for email, page, etc are listed in the sample
files, it is more comprehensive than many of the other samples.

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Matt
> Sent: Sunday, September 18, 2005 6:53 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Voicemail
>
> While we are on the subject.. how do you modify the TXT message that
> gets send to the 'pager'... Is that hard coded.. or can that be
> changed?No variables I change in voicemail.conf seem to change the
> from address, etc.
>
> On 9/18/05, Damon Estep <[EMAIL PROTECTED]> wrote:
> >
> >
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of
> > [EMAIL PROTECTED]
> > Sent: Sunday, September 18, 2005 8:32 AM
> > To: asterisk-users@lists.digium.com
> > Subject: [Asterisk-Users] Voicemail
> >
> > When I receive a voicemail notify via e-mail I would like receive
not
> > the sender phone-number, but the sender name. Where can I configure
> > this and how? Is it possible to have some example?
> > Thank
> > Luca
> >
> > This will do it (in voicemail.conf) but I think the default does as
> > well. Are you actually getting caller ID name delivered when a call
> > comes it?
> >
> > emailbody=${VM_NAME} <${VM_MAILBOX}>\n\nYou have a new voicemail
message
> > from ${VM_CALLERID}.  The message is ${VM_DUR} long and was left on
> > ${VM_DATE}.
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RE: [Asterisk-Users] T.38 & Canreinvite (yes, again)

[Carlos Alperin]

What this has to do with Asterisk?

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Monday, September 19, 2005 4:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] T.38 & Canreinvite (yes, again)

I know this has been asked before, but I've checked the archives and I 
haven't found anybody that has given a definitive yes or no, just "yeah, 
it should work.".  If I have a T.38 gateway like a Cisco 5300 and a 
T.38 ATA (whatever model) and I have canreinvite=yes, should T.38 work?

I have it setup and it doesn't work, so I want to know if I am doing 
something wrong, or if it just won't work.  If I make a voice call, I 
see the media stream go from the gateway to the ata directly.  When I 
fax, I see the stream go that way as well, but it is g.729.  I see 
INVITE messages from my ATA that reference T.38, but they go to the * 
box, not the gateway and therefore * ignores it.  Any thoughts?

PA
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[Asterisk-Users] Call dropped 100% of time when incoming IAX routed to outgoing CAPI

[Christopher Mylonas]

Good day,

The unusual thing about this problem is that it doesn't occur just during a 
CAPI call, or just during an IAX/SIP call.  Only during IAX/CAPI
I'm having some trouble with the CAPI interface and it only occurs when a call 
comes in on an IAX channel and goes out the CAPI interface.

The capi debug in the asterisk console is below as well as the relevent parts 
of .conf files from the machines involved.

The scenario is this (extension numbers):

4074 is connected to a Siemens Hipath 3000 PBX.
The Hipath3000 has a cable running to asterisk at address 192.168.1.222 
connected to a wcte11xp zaptel card - Single span PRI.
The asterisk at 222 just mentioned ([EMAIL PROTECTED]) is on the LAN and it 
accepts calls 
from the Hipath3000 in a certain range and sends the calls to locally
connected SIP extensions.  When ext 4074 calls 4050 we've set up a test 
scenario where the Hipath3000 sends all calls for 4050 to [EMAIL PROTECTED], 
which then
has an IAX connection to another asterisk machine set up at 192.168.1.224 
([EMAIL PROTECTED]).  This call is then sent out the CAPI interface on the 
second
machine to the Hipath3000 and pointed to extension 4028.  This connection 
drops out 100% of the time at different times though - none longer than
one minute.  On this particular debug within asterisk the call got dropped 
during the ringing stage.

4074 dials (4050) hp3k pbx -> in thru zap [EMAIL PROTECTED] -> iax to [EMAIL 
PROTECTED] -> capi to 
hp3k pbx -> 4028



The CAPI device is an AVM C2 card on the 224 machine
chan_capi version is 0.5.4 on 192.168.1.224
asterisk version on 192.168.1.222 is CVS-HEAD and the chan_iax2.c files says 
revision 1.332
asterisk version on 192.168.1.224 is Asterisk 1.0.9


***CONF FILES


192.168.1.222's extension.conf
**
; DO NOT REMOVE
; FOR USE BY OLCS SYDNEY TO DIAL OUT VIA IAX2
;*
[iax-out]
exten => 4050,1,Dial(IAX2/iax-olcs-brisbane/4050)


192.168.1.224's extensions.conf
***
[iax-net-in]
exten = 4050,1,Dial(CAPI/g1/4028)


192.168.1.224's capi.conf
*
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8

[interfaces]
group=1
msn=4150
incomingmsn=*
controller=1
softdtmf=1
accountcode=
context=trunkincoming
devices=2


Just to re-iterate, the CAPI interface is fine when it is used as a normal 
operation from the extension.conf file,
i.e. exten => ,1,Dial(CAPI/g1/${EXTEN}
The problem only occurs when it is an IAX call coming in and being routed out 
the CAPI interface.


CAPI Debugging Enabled
-- Accepting AUTHENTICATED call from 192.168.1.222, requested format = 2, 
actual format = 2
-- Executing Dial("IAX2/iax-olcs-brisbane at iax-olcs-brisbane/2", 
"CAPI/g1/4028") in new stack
-- data = g1/4028
-- capi request group = 2
-- creating pipe for PLCI=0
  == CAPI Call CAPI/contr1/4028-5  (pres=0x00)
CONNECT_REQ ID=002 #0x3f55 LEN=0051
  Controller/PLCI/NCCI= 0x1
  CIPValue= 0x10
  CalledPartyNumber   = <80>4028
  CallingPartyNumber  = <00 80>4074
  CalledPartySubaddress   = default
  CallingPartySubaddress  = default
  BProtocol
   B1protocol = 0x1
   B2protocol = 0x1
   B3protocol = 0x0
   B1configuration= default
   B2configuration= default
   B3configuration= default
   GlobalConfiguration= default
  BC  = default
  LLC = default
  HLC = default
  AdditionalInfo
   BChannelinformation= <00 00>
   Keypadfacility = default
   Useruserdata   = default
   Facilitydataarray  = default
   SendingComplete= default

-- Called g1/4028
CONNECT_CONF ID=002 #0x3f55 LEN=0014
  Controller/PLCI/NCCI= 0x101
  Info= 0x0

  == received CONNECT_CONF PLCI = 0x101 INFO = 0
INFO_IND ID=002 #0x5755 LEN=0015
  Controller/PLCI/NCCI= 0x101
  InfoNumber  = 0x800d
  InfoElement = default

INFO_RESP ID=002 #0x5755 LEN=0012
  Controller/PLCI/NCCI= 0x101

  == info element SETUP ACK
INFO_IND ID=002 #0x5756 LEN=0017
  Controller/PLCI/NCCI= 0x101
  InfoNumber  = 0x1e
  InfoElement = <81 88>

INFO_RESP ID=002 #0x5756 LEN=0012
  Controller/PLCI/NCCI= 0x101

  == info element PI 81 88
 In-band information available
INFO_IND ID=002 #0x5757 LEN=0016
  Controller/PLCI/NCCI= 0x101
  InfoNumber  = 0x18
  InfoElement = <8a>

INFO_RESP ID=002 #0x5757 LEN=0012
  Controller/PLCI/NCCI= 0x101

  == info element CHANNEL IDENTIFIKATION 8a
INFO_IND ID=002 #0x5758 LEN=0015
  Controller/PLCI/NCCI   

Re: [Asterisk-Users] TDM400P Dialing Out - "Cannot be completed as dialed"

[John Novack (port)]

Barry King wrote:

  John Novack wrote:

  
  

   I'm not sure how I'm going to listen in, either.  The only thing I can think of is making myself a "lineman's handset" out of an old phone and plugging into the phone box outside. 

  

Not sure why you need to "go outside" to listen in. a simple 268 style
"t" adapter for modular plugs would do. I assume you are in the US .

  
  

Oh, hello.  Yeah, that would work.

  

Insert a .1 mF cap ( value not critical ) in series with the handst or
phone. You can't talk, but will be able to hear what's going on, and
you will find that Asterisk begins to dial as soon as it goes off hook

Ohers have reported that version 1.0.9 respnds to the "w" in the dial
string. Look through the archives for the last several weeks.
It doesn't work in my version of head from Feb 24, with dialpulse=yes
though.

Best just install the patches you found on the Wiki

Good luck

John Novack



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RE: [Asterisk-Users] unlocking cisco 7940 phone

[Brian]

Thanks... I will let you know how it works out.



Brian S. Reale
President
 
Colosa Inc.
Tel:  305.675.1400 ext. 201
Fax: 305.402.0282
www.colosa.com 
www.ProcessMaker.com 
 
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael Welter
Sent: Monday, September 19, 2005 5:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] unlocking cisco 7940 phone

I got five of these doctor's phones as well.

You need your server set-up with dhcp and tftp.  In the tftp directory, 
you'll need files SIP.cnf and SIPDefault.cnf available.  You'll 
also need the current SIP firmware load.

Then, you have to let the phone power up, then enter the password 
'cisco', the go to settings->3->28 (remove config) and also ->25 to 
enable DHCP.  Then hit 'yes'.

If your files are ok then the phone will reboot, upgrade its firmware, 
and read the new configuration.

I performed this procedure five times this morning and it worked every time!

5¢ please  :-)




Rich Adamson wrote:
>>Unfortunately, it didn't work.  Do you know any other ways?
>>
>>
>>
>>Brian S. Reale
>>President
>> 
>>Colosa Inc.
>>Tel:  305.675.1400 ext. 201
>>Fax: 305.402.0282
>>www.colosa.com 
>>www.ProcessMaker.com 
>> 
>> 
>>-Original Message-
>>From: [EMAIL PROTECTED]
>>[mailto:[EMAIL PROTECTED] On Behalf Of Tom Vile
>>Sent: Saturday, September 17, 2005 9:04 PM
>>To: Asterisk Users Mailing List - Non-Commercial Discussion
>>Subject: Re: [Asterisk-Users] unlocking cisco 7940 phone
>>
>>try going into settings and then unlock config and try typing in cisco
>>
>>On 9/17/05, Brian <[EMAIL PROTECTED]> wrote:
>>
>>> 
>>> 
>>>
>>>  
>>>
>>>I have a Cisco 7940 phone with a locked SIP configuration.  There is no
>>
>>tftp
>>
>>>server configured in network settings.  Does anyone know how to get this
>>>phone to upgrade its firmware and thereby unlock the SIP settings?  (The
>>
>>**#
>>
>>>combination on the Cisco manuals does not work). 
>>>
>>>  
>>>
>>>-brian 
> 
> 
> I just had to do one the other day; the phone shipped with mgcp installed
> with some Doctors name appearing on the screen. It had been locked down
> pretty tight.
> 
> The only way that I could do it is to use a sniffer (eg, ethereal) to
> see the IP addresses it was trying to reach.
> 
> I then put the phone on a network that had a linux server with that
> exact addressd on it. The phone happily read the tftp files that I
> had prestaged on the linux server, and upgraded itself to the sip image.
> 
> The upgrade process wiped out all passwords, etc, and reset the phone
> to factory defaults for sip (except for the password, which I specified
> in a tftp config file).
> 
> Once that was done, I could access the phone menues just fine and reset
> everything to the way that I wanted it.
> 
> If you're not comfortable reading packet traces, then you'll probably
> have to send the phone to someone that is.
> 
> 
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[Asterisk-Users] Sip and ISDN problem

[Andrew Nowrot]

Hi,

I'm quite new to all asterisk issues. Unfortunately I already have
some problems.

I use ISDN phone and my colleague has got a normal phone connected to
Linksys PAP-2 VoIP Gateway. These two phones are in the same call and
pickup group. With the phone connected to Linksys I'm able to pickup
all calls, but my ISDN does not work properly. When I dial *8 (pickup
extension configured in features.conf and in extension.conf as well)
on my ISDN phone, I can pickup the call to my colleague, but the phone
connected to Linksys does not stop ringing.
I tried everything pickup, pickupchan, answer and steal, but nothing worked.
Please help!!!

Andrew
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[Asterisk-Users] Re: Welcome to the "Asterisk-Users" mailing list

[chuck gelm]

"Unable to login to the MySQL database 'tiki' on 'localhost' as user
'tiki'
Go here to begin the installation process, if you haven't done so
already.
Lost connection to MySQL server during query"

Hi, Everyone:

What is the URL of the "list archive"?
The 'wiki' seems broken.

 I want to connect to a single home POTS from an internet connection.
i.e. I want to be able to make telephone calls on my home POTS
from anywhere that I can obtain an internet connection with
a device with a microphone and a speaker. e.g. My laptop and a
headphone with boom microphone.  Does 'Asterisk' do this?
Should I look elsewhere for this task?

 If this is answered in the list archive, please respond with the
URL of the list archive.

Best regards,
Chuck

[EMAIL PROTECTED] wrote:

Welcome to the Asterisk-Users@lists.digium.com mailing list! Before
posting to the Asterisk user list, please be sure your question hasn't
already been answered in the list archive or in the Asterisk wiki at
http://www.voip-info.org/wiki-Asterisk.  For more information see
asterisk.org Also, if you want to search the lists for a particular
topic, you can do so through Google, by using something along the
lines of: "asterisk rules site:lists.digium.com" (without
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If your postings don't seem to go through, please ensure that your
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Normally, Mailman will remind you of your lists.digium.com mailing
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[Asterisk-Users] Re: Most desireable Linux distribution for Asterisk?

[asterisk]

Thanks to all that responded. 

Having been biased toward FreeBSD for the last 8 years or so, we're not 
really tied by tradition to any particular Linux distro.  We have customer 
co-lo servers in our datacenter running pretty much all the major 
distributions, and none really seems to stand out in terms of performance 
and reliability.  Its all quite reliable so it stands to reason that 
personal preference and familiarity is usually the deciding factor. 

Slackware was the very first *nix I ever loaded onto a PC about 10 years 
ago.  Maybe its time to give it another spin!  :-) 

- D 


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[Asterisk-Users] Zap calls dropping just after answer

[Jesse Keating]

I've got a problem w/ zap calls being dropped right after they are
answered.  I have a log file:

http://pastebin.com/368526

Everything looks OK except for the 

DEBUG[25563] chan_zap.c: Exception on 9, channel 1

that seems to come up quite often.  As soon as the other end of the Zap
answers (my cell phone), and I can even hear a half second of noise, the
line goes dead and gets hungup.

In my case, Zap/1 is the first channel of a 24 channel T1 line to a
Fujitsu PBX.  All seems configured right as I can dial out and the cell
phone will ring.  This is very frustrating, can anybody help out w/
this?

-- 
Jesse Keating
GameHouse -- Systems Engineer

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[Asterisk-Users] Voicemail() application returning -1 on a hangup

[Robert G. Ristroph]

Hi,

   I am trying to insert a system() call right after the call to Voicemail() in
order to notify people that they have received new voicemail.  (In case you are
interested, I was following the tips here, but I have had to change a few
things:
http://www.voip-info.org/wiki-Asterisk+tips+callback ).

   My setup works when someone leaving a message hits # to exit the voicemail
after speaking, or lets it time out.  However, when they just hangup (which is
what 99 percent of people do) Voicemail() returns -1, which apparently stops
any further extensions from executing.

   Here is my vm macro:

[macro-vm]
exten => s,1,NoOp(At start of macro-vm---Arg1=${ARG1}  Arg2=${ARG2})
exten => s,2,Goto(s-${ARG2},1)
exten => s-BUSY,1,Voicemail(b${ARG1})   ; Voicemail Busy message
exten => s-BUSY,2,Hangup()
exten => _s-.,1,NoOp(Before Voicemail-u---Arg1=${ARG1}  Arg2=${ARG2})
exten => _s-.,2,Voicemail(u${ARG1}) ; Voicemail Unavailable message
exten => _s-.,3,NoOp(After Voicemail-u---Arg1=${ARG1}  Arg2=${ARG2})
exten => _s-.,4,system(${SCRIPTS_DIR}/voicemail_callback.sh ${ARG1})
exten => _s-.,5,NoOp(After systemArg1=${ARG1}  Arg2=${ARG2})
exten => _s-.,6,Hangup()

; this catches the error exit (n+101) from Voicemail, such was when a person
hangs up after recording a message
; instead of hitting #  (this doesn't work)
; exten => _s-.,103,system(${SCRIPTS_DIR}/voicemail_callback.sh ${ARG1})

exten => o,1,Background(one-moment-please)  ; 0 during vm message will
hangup
exten => o,2,GotoIf($["foo${FROM_DID}" =
"foo"]?from-pstn,s,1:from-pstn,${FROM_DID},1)
  
exten => h,1,system(${SCRIPTS_DIR}/voicemail_callback.sh ${ARG1})
exten => t,1,system(${SCRIPTS_DIR}/voicemail_callback.sh ${ARG1})

exten => a,2,VoiceMailMain(${ARG1})
exten => a,3,Hangup 

The NoOp after the Voicemail call never runs if you hang up, thus the
system() never has a chance to run.  The t,1 seems to work for when you time
out.  The h,1 which I thought might catch it when a person hangs up in
Voicemail(), doesn't.

The behavior of Voicemail() should allow you to differentiate between a real
error (out of disk space for voicemail, for example) and the most common
behaviour of message leavers (hanging up when done).  Perhaps that should be a
bug report.  But how can I work around this ?

Would it be easy to patch the C code of app_voicemail.c to not have this
behavior ?  I looked over it but I could not make headway on the first look.

--Rob

-- 
Robert G. Ristroph
Airlink Systems
[EMAIL PROTECTED]
(512) 231-1240 x103






This message was sent using IMP, the Internet Messaging Program.
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RE: [Asterisk-Users] kill a .call file

[Remco Barende]

A workaround is to disable call waiting on your phones (nasty workaround 
tho)


On Mon, 19 Sep 2005, Rene Kluwen wrote:


I have the same problem. Asterisk always makes two calls, even when the
first one went through successfully.

Rene Kluwen
Chimit

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Remco
Barende
Sent: maandag 19 september 2005 22:09
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] kill a .call file


On Mon, 19 Sep 2005, jltaylor wrote:


From my CLI:


Attempting call on SIP/gw/19857749166 for [EMAIL PROTECTED]:1
(Retry 114)
Attempting call on SIP/gw/19037747603 for [EMAIL PROTECTED]:1
(Retry 83)
Attempting call on SIP/gw/19857747603 for [EMAIL PROTECTED]:1
(Retry 80)

I want to stop it from any future attempts.

Any idea about a command to kill or where the data is stored?


This is an asterisk bug. I already filed it but they need a full trace and
I haven't had the time yet to do it.

It seems that * keeps retrying the call, even when it was succesfully
completed.

Annoying.




James Taylor
MetroTel
3505 Summerhill Road
Suite 11
Texarkana, Tx  75503
903-793-1956



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of trixter
http://www.0xdecafbad.com
Sent: Monday, September 19, 2005 1:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion;
[EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] kill a .call file


On Mon, 2005-09-19 at 13:43 -0500, jltaylor wrote:

Any means of killing a .call file that is in progress?



You mean once the call has begun?  You prolly want to hangup the
call ...

asterisk -rx "soft hangup "

Or is there something else that you wanted?


--
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378



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Re: [Asterisk-Users] TDM400P Dialing Out - "Cannot be completed as dialed"

[Barry King]

John Novack wrote:

>>though I'm also guessing that a lack of dialtone detection is causing my 
>>troubles.
>>
> And there seems to be no willingness to either fix or even consider it
> a bug.

>>  I'm not sure how I'm going to listen in, either.  The only thing I can 
>> think of is making myself a "lineman's handset" out of an old phone and 
>> plugging into the phone box outside. 
>>
> Not sure why you need to "go outside" to listen in. a simple 268 style
> "t" adapter for modular plugs would do. I assume you are in the US .


Oh, hello.  Yeah, that would work.

>> I'm worried that plugging the FXO jack into the
>>ComDial system would fry something, 
>>
> Can't imagine what. The FXO card in it's simplest form, the X100P is a
> recycled modem card using a specific chip set.
> How would you use a Comdial system to monitor anyway?


I'm presuming that some of the pairs in the phone line are powered,
since the comdial phones do a lot of tricks and aren't plugged into the
electrical outlet.  I'm concerned that this might fry something.

How to monitor?  The old phones are still hooked up until I work out all
the bugs in the new system, so I'd pick up line 1 with the phone and
then dial with asterisk.  Or dial first and then pick up.  *shrug*

>>and that's the only other way I can
>>think of to listen in, aside from maybe running the line through a fax 
>>machine.
>>  
>>
> You will want to be able to listen without starting dialtone.
>
> Any old telephone butt set or even a cheap battery powered transistor
> amplifier  would do as well.


Yeah, I've got a handset that will work, just no T connector.  I'll have
to run by Radio Shack.

>>Unless there's some software that does it.
>>
>>On the dialtone detection end of things,
>>http://www.voip-info.org/tiki-index.php?page=NVLineDetect
>>does seem to be available, but that would require me modifying emailing the 
>>guy and either going with cvs or modifying the gentoo ebuild I'm using (bleh).
>>
>>
> If that will give you dialtone detection, that's probably what you
> will have to do then!


Yep.  blah.

Thanks for your help!

Cheers,

Barry
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Re: [Asterisk-Users] Looking for firmware for Cisco 12sp+ and 30VIP

[Goran Dj]

Which firmware do you need? I have couple of them, but I don't 
know is it legal to share them over Internet :-))
 

  - Original Message - 
  From: 
  Stern, 
  Craig 
  To: asterisk-users@lists.digium.com 
  
   
  
  I have been 
  looking for the firmware for the 12sp+ and 30VIP and have been unable to find 
  it. Any help in locating would be much appreciated.
   
  Thanks
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[Asterisk-Users] MSNs don't work for me... :(

[Yoann Le Bihan]

Hi,

My * server is running FreeBSD. The wonderful isdn4bsd with chan_capi
port by hselaski works great, everything is perfect in a perfect
world... but !
These [EMAIL PROTECTED] MSNs don't work at all ! My extensions.conf is very
simple for my 5 MSNs (0353, 0416, 0618, 0619 and 0620) :

[rnis-in]
exten => 0353,1,Answer()
exten => 0353,2,Dial(SIP/600,,m)
exten => 0353,3,Hangup

exten => 0416,1,Dial(SIP/603)
exten => 0416,2,Hangup

exten => 0620,1,Dial(SIP/602)
exten => 0620,2,Hangup

exten => 0619,1,Dial(SIP/603)
exten => 0619,2,Hangup

exten => 0618,1,Dial(SIP/601)
exten => 0618,2,Hangup

And rnis-in is the context declared in capi.conf for incoming calls. Well.
But when a call arrives, I have this error in verbose mode :

-- creating pipe for PLCI=0x1208
-- started pbx on channel (callgroup=0)!
  == Starting CAPI/contr8/0620-1 at rnis-in,0620,1 failed so falling
back to exten 's'
  == Starting CAPI/contr8/0620-1 at rnis-in,s,1 still failed so
falling back to context 'default'
-- Executing Answer("CAPI/contr8/0620-1", "") in new stack
-- CAPI Answering for MSN 0620
-- Executing Dial("CAPI/contr8/0620-1", "SIP/600||m") in new stack
-- Called 600
-- Started music on hold, class 'default', on CAPI/contr8/0620-1
-- SIP/600-f5a7 is ringing
-- Stopped music on hold on CAPI/contr8/0620-1
  == Spawn extension (default, s, 2) exited non-zero on 'CAPI/contr8/0620-1'
-- CAPI Hangingup
-- removed pipe for PLCI = 0x1208

I'm really disappointed : the MSN is detected (0620). But * tells me
that rnis-in,0620,1 fails !... (the server was rebooted several times,
as I've been trying to solve this problem for at least one week)
And if I add an exten 's' in context 'rnis-in', every calls go to this
exten, never to 0620.

I also tried with and without including 'rnis-in' to 'default'. No
way. Doesn't work.

Is there someone to help me before suicide ? ;o)) please, give me all
your ideas... :)

Cheers,

YLB.
[EMAIL PROTECTED]
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[Asterisk-Users] hfc card unplug & plug not working?

[[EMAIL PROTECTED]]

Hello,

I have hfc-pci card with zaphfc driver from bristuff.

card is working with asterisk, but when i unplug line cable from card,
and then plug back to card in log i can see only "could not create Zap
channel" when dialing.

when I remove zaphfc&zaptel drivers and then modprobe them card with
asterisk works again and stop working when line is unplugged (then 
plugged back).


is it normal?
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Re: [Asterisk-Users] Looking for firmware for Cisco 12sp+ and 30VIP

[Shaun Ewing]

On 9/20/05, Stern, Craig <[EMAIL PROTECTED]> wrote:
> I have been looking for the firmware for the 12sp+ and 30VIP and have been
> unable to find it. Any help in locating would be much appreciated.
>  
> Thanks

What type of firmware?

The only firmware available is for SCCP/Callmanager and is a few years old.

-Shaun
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Re: [Asterisk-Users] AstriCon 2006 Location

[Shaun Ewing]

On 9/20/05, Darren Younger <[EMAIL PROTECTED]> wrote:
> Great Idea! I suggest Sydney :-)

No complaints from me there, I live in Sydney :-)

If it was in the US, I'd personally prefer the west coast. It's the
easiest and cheapest part of the US for me to get to.

-Shaun
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Re: [Asterisk-Users] Asterisk Keep Crashing need Help please

[Mark Edwards]

Hi.

I, personally,  would start this process by checking out a clean
version of CVS-HEAD and seeing if the problem still exists. If this is
too much of a jump all at once, at _least_ go to 1.0.9-Stable.

cheers,

MarkOn 9/20/05, Ka Lun Chan <[EMAIL PROTECTED]> wrote:
Hi All, 
 
My Asterisk Server crash at least once a day. The full and message
log files did not show any errors or indication on why the asterisk
service crashed. After looking at the core dump file, I am still unable
to identify the problem. Can someone please give me some guideance on
what I can do? Thank You All

 
Asterisk is loaded with version 1.0.5, PyAsterisk, and postgresql. Is used for IP Phone only with SIP protocols only. 
Server Spec: Dual Xeon 3.0 processor with 2G memory
Here is the core dump:
 
Program terminated with signal 11, Segmentation fault.#0  ed_next_char (el=0x6, c=-1073749366) at common.c:315315
return (CC_REFRESH);(gdb) bt#0  ed_next_char (el=0x6, c=-1073749366) at 
common.c:315#1  0x080b72db in ast_strcasestr (haystack=0x80fbf30 "èº\017\b î\022B\200ï\022B", needle=0xbfffe2dc "") at utils.c:389#2  0x0809cb32 in session_do (data="" at manager.c:252#3  0x42015504 in ?? ()
#4  0x0002 in ?? ()#5  0xbfffe674 in ?? ()#6  0xbfffe680 in ?? ()#7  0x4001582c in ?? ()#8  0x0002 in ?? ()#9  0x08052ac0 in __do_global_dtors_aux ()#10 0x08052ae1 in frame_dummy ()


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http://lists.digium.com/mailman/listinfo/asterisk-users-- regards,Mark P. EdwardsFWD: 667917
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Re: [Asterisk-Users] TDM400P Dialing Out - "Cannot be completed as dialed"

[John Novack]

Barry King wrote:

  "w" doesn't seem to make a difference, 

That has been my experience, though I am using pulsedial
There seems to be some undocumented differences between versions and
flavors of Asterisk that are not yet known.

  though I'm also guessing that a lack of dialtone detection is causing my troubles.

And there seems to be no willingness to either fix or even consider it
a bug.

I'm not sure how I'm going to listen in, either.  The only thing I can think of is making myself a "lineman's handset" out of an old phone and plugging into the phone box outside. 

Not sure why you need to "go outside" to listen in. a simple 268 style
"t" adapter for modular plugs would do. I assume you are in the US .

   I'm worried that plugging the FXO jack into the
ComDial system would fry something, 

Can't imagine what. The FXO card in it's simplest form, the X100P is a
recycled modem card using a specific chip set. 
How would you use a Comdial system to monitor anyway?

  and that's the only other way I can
think of to listen in, aside from maybe running the line through a fax machine.
  

You will want to be able to listen without starting dialtone.

Any old telephone butt set or even a cheap battery powered transistor
amplifier  would do as well.


  Unless there's some software that does it.

On the dialtone detection end of things,
http://www.voip-info.org/tiki-index.php?page=NVLineDetect
does seem to be available, but that would require me modifying emailing the guy and either going with cvs or modifying the gentoo ebuild I'm using (bleh).

  

If that will give you dialtone detection, that's probably what you will
have to do then!

  What is NANP?

  

North American Numbering Plan

JN


  Cheers,

Barry King

John Novack (port) wrote:

  
  
Of course check your dialplan first, but if that fails, monitor the
PSTN line with a buttset and/or digitgrabber, you probably will find
that Asterisk is dialing without waiting for dialtone, so the first
digit is not detected at the CO.
Some versions will allow a "w" to wait 1/2 second before starting to
dial, bur it doesn't always work, and there seems to be little
interest in detecting dialtone before dialing.
The NANP was SUPPOSED to require  7/10 digit local and 11 digit toll
dialing, but since that is now a state by state determination, there
is little "plan" left in the NANP

John Novack


  
  




  



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[Asterisk-Users] pridialplan per call or per channel group?

[Gary Reuter]

Hi,
Is it possible to set the pridialplan (or prilocaldialplan) on a call-by-call basis, or for a particular PRI channel-group?

>From my experimentation so far, it appears to me that the pridialplan
and prilocaldialplan are global in zapata.conf, unlike other options
such as switchtype, signalling, and context.

I currently have two PRIs from different providers (and different
switchtypes), and a third PRI port going to a Norstar MICS 4.x. 
I'm trying to get better dialplan integration with the Norstar by using
some of the features described at the beginning of the installer's
guide (ie, using 'private' or 'tie' line to link two remote MICS with
public-network PRIs at each end).
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[Asterisk-Users] [Fwd: ASTCC speaks and cut RTP channel => Kind of solution...

[Ricardo Poppi]

Hi all.

I´ve found a kind of solution (if we can call it this way...) and Im
reporting it here to help save some lives.

Editing into astcc.cgi I found where the parameters that set 60 and 30
seconds warning were and put zeros in its place. The last two
lots-of-zeros numbers at second line. So the zap trunk code of astcc.cgi
became like that:

==
 if ($res->{tech} eq "Zap") {
  $dialstr = "Zap/$res->{path}/$phone|30|HL(" . ($maxtime *
60 * 1000) . ":0:0)";
  $res = $AGI->exec("DIAL $dialstr");
  $answeredtime = $AGI->get_variable("ANSWEREDTIME");
  $dialstatus = $AGI->get_variable("DIALSTATUS");
  $callstart = localtime();
  return $dialstatus;
  }
==


And - at least until now... - everything is working fine. The credit is
being take from the cards in the right amount and no warnings are being
given when 60 and 30 seconds left. When credit finishes, the agi script
just finishes the call.

If somebody has a better way to do that, please let us know.

Rgs, Ricardo Poppi.


 Mensagem Original 
Assunto:ASTCC speaks and cut RTP channel
Data:   Fri, 09 Sep 2005 18:09:52 -0300
De: Ricardo Poppi <[EMAIL PROTECTED]>
Para:   asterisk-users@lists.digium.com



Hi list.

I have a fine running Ser+Asterisk environment and have just installed
ASTCC. It´s working fine either, including its caller-id authentication
feature (the one we pass the card-number as CALLERID variable and
number-to-dial as EXTEN variable).

The issue, a great one, is that when the credit is about one minute to
end, the ASTCC prompt gets into the call, says that "you have one minute
left..." and when it was suppose to leave and let the RTP traffic of the
original call be "reestablished", it never happens. The RTP packets  - I
could see that at asterisk debug screen - stop running and the call is
still signaled as active, but no media at all.

This is a serious problem I´m having and, as I could see, I´m not the
only one. Mr. Chilini reported that around jun 30th this year, as you
can see bellow: (I just added a comment at this voip-info page to see if
anyone could give some clues about that)

http://www.voip-info.org/tiki-index.php?page=ASTCCGuide#comments


Do anyone here in this list had any situation alike? Do you have any
clues do help me? (and others because it will be documented, of course).

Thanks in advance,

Ricardo Poppi.



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[Asterisk-Users] H.263 Format video

[Areski K]

Anyone have a clue about how asterisk handle the h263 format?
I tried to play some h263 file or even convert them myself and
I didnt succeed to play them through Asterisk.
It seems that it works only if I record the video with Voicemail.

Any advice on this ?
Thx in advance,
/Areski
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RE: [Asterisk-Users] IAX dialplan problem?

[Colin Anderson]

 > In the above example, you have to press 1,2 
or 3 before it will tell you that. I was wanting it to tell me before 
I  > pressed those numbers. 
 
You 
have to get to a context before you can do any action  so you have to dial *something* to get 
to that context. However, with Zapata interfaces (i.e. you are using a 
Digium TDM card + a regular POTS phone) you can specify immediate=yes in 
zapata.conf so as soon as the phone goes off-hook it will be thrown into that 
extension's 's' priority. However, a VoIP phone or an IAXy would have to be set 
to auto-dial an extension as soon as it goes off hook, and this is dependent on 
the model of phone that you have. There are smarter people than me on the list 
that can comment on how this would work with an IAXy.Can 
an IAX device be setup so that it appears to be an inbound DID, and if so how 
can I do that with the IAXy?Sort of. An IAX device has to be defined in 
IAX.conf with a default context, or Asterisk won't know what context to hand the 
call off to when the IAX device tries to instantiate a call on your Asterisk 
box. You can simulate an "inbound" call simply by making the context entry in 
IAX.conf the same context as what is used for inbound calls. So if your default 
context for PSTN calls is "from-pstn" you would make the context entry for the 
IAXy to "from-pstn" and any time the IAXy dials, Asterisk would throw the call 
into the "from-pstn" context and attempt to match the extension dialled from the 
IAXy to an entry in the "from-pstn" context. If there was no direct match, 
Asterisk would try the 's' extension. If the 's' extension would not yield a 
match, Asterisk would return 404 to the IAXy. 
 
 
hth
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Re: [Asterisk-Users] TDM400P Dialing Out - "Cannot be completed as dialed"

[Barry King]

"w" doesn't seem to make a difference, though I'm also guessing that a
lack of dialtone detection is causing my troubles.  I'm not sure how I'm
going to listen in, either.  The only thing I can think of is making
myself a "lineman's handset" out of an old phone and plugging into the
phone box outside.  I'm worried that plugging the FXO jack into the
ComDial system would fry something, and that's the only other way I can
think of to listen in, aside from maybe running the line through a fax
machine.

Unless there's some software that does it.

On the dialtone detection end of things,
http://www.voip-info.org/tiki-index.php?page=NVLineDetect
does seem to be available, but that would require me modifying emailing
the guy and either going with cvs or modifying the gentoo ebuild I'm
using (bleh).

What is NANP?

Cheers,

Barry King

John Novack (port) wrote:

> Of course check your dialplan first, but if that fails, monitor the
> PSTN line with a buttset and/or digitgrabber, you probably will find
> that Asterisk is dialing without waiting for dialtone, so the first
> digit is not detected at the CO.
> Some versions will allow a "w" to wait 1/2 second before starting to
> dial, bur it doesn't always work, and there seems to be little
> interest in detecting dialtone before dialing.
> The NANP was SUPPOSED to require  7/10 digit local and 11 digit toll
> dialing, but since that is now a state by state determination, there
> is little "plan" left in the NANP
>
> John Novack
>

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Re: [Asterisk-Users] TDM400P Dialing Out - "Cannot be completed as dialed"

[Barry King]

I'm replacing an old ComDial system.  It required a 9 to dial out, but
never 10 digits.

relevant bits from extension.conf :
;exten => _X.,1,Dial(ZAP/g1/w${EXTEN},60,R)
exten => _X.,1,Dial(ZAP/2/w${EXTEN},60,R)
;exten => _X.,3,Dial(ZAP/g1/${EXTEN},60,t)

I've tried the commented varieties w/ the same results.

Cheers,

Barry

MF Hulber wrote:

> I can't see anything immediately wrong here.  Maybe it's something in
> your dialplan?  It sounds like Asterisk is dialing out but the PSTN
> doesn't like the number you are dialing.  Are you in an area that
> requires 10 digits or does not like if you dial 11 for local calls?
>
> From what I can tell, the 805 area code requires 11 digits when dialing.
>
> MARK.
>
> Barry King wrote:
>
>> I've tried to google this issue with no resolution.
>>
>> I'm having the same issue as this person:
>> http://lists.digium.com/pipermail/asterisk-users/2004-August/058280.html
>>
>> Basically, anytime I try to dial out on my TDM400P w/ FXO, I get "we're
>> sorry, but your call cannot be completed as dialed."
>>
>> When I "debug channel Zap/x-x", I get a whole bunch of this:  [ TYPE:
>> Null Frame (5) SUBCLASS: N/A (0) ] [Zap/2-1]
>>
>> I recieve calls just fine.
>>
>> I've tried dialing the Zap group and the specific channel, all with the
>> same results.
>>
>> Any ideas?  zapata and zaptel configs follow.
>>
>> zapata.conf:
>> [channels]
>> context=local-in
>> relaxdtmf=yes
>> language=en
>> signalling=fxs_ks
>> ;rxwink=300
>> usecallerid=yes
>> useincomingcalleridonzaptransfer=yes
>> callerid=asreceived
>> group=1
>> immediate=yes
>> echocancel=64
>> echocancelwhenbridged=no
>> echotraining=800
>> rxgain=9.0
>> txgain=1.0
>> channel => 1-4
>>
>> zaptel.conf:
>>
>> fxsks=1-4
>> defaultzone=us
>> loadzone=us
>>
>> Regards,
>> Barry King
>> King Computer Solutions 
>

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RE: [Asterisk-Users] OT: Hardware Interrupts; Who is it?

[Kevin Collins]

Matthew,

I see similar timer interrupts from using ztdummy as my timing reference, 
should be about 1k per sec.
 

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Boehm
Sent: Monday, September 19, 2005 4:52 PM
To: Asterisk Users
Subject: Re: [Asterisk-Users] OT: Hardware Interrupts; Who is it?

   CPU0   CPU1
  0: 85 1703809954IO-APIC-edge  timer
  8:  0  0IO-APIC-edge  rtc
  9:  0  1   IO-APIC-level  acpi
 14:  0 31IO-APIC-edge  ide0
177:  0   17840313   IO-APIC-level  megaraid
185:  0 1817423967   IO-APIC-level  eth0
193:  0   40198530   IO-APIC-level  eth1
201:  0 3507106255   IO-APIC-level  wanpipe1, wanpipe2, wanpipe3,
wanpipe4
NMI:  0  0
LOC: 1633394197 1633394188
ERR:  0
MIS:  0

Any idea what "LOC" and "timer" are? I did "watch -n1 cat /proc/interrupts"
for about a minute. The largest movers were "timer", "LOC" and "eth0". The
others either never change or hardly changed.

-Matthew

> From: Sig Lange <[EMAIL PROTECTED]>
> Reply-To: <[EMAIL PROTECTED]>, Asterisk Users Mailing List - Non-Commercial
> Discussion 
> Date: Mon, 19 Sep 2005 15:34:26 -0400
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
> Subject: Re: [Asterisk-Users] OT: Hardware Interrupts; Who is it?
> 
> It means you have a piece of hardware that is generating a lot of interupts.
> Try a few commands like this:
> 
> # vmstat 1
> # watch -n1 cat /proc/interrupts
> Go through lspci -vb and disable hardware that's not being used.
> 
> Watch the numbers as they increase. Also check for ERR and MIS
> 
> 
> On 9/19/05, Rich Adamson <[EMAIL PROTECTED]> wrote:
>> 
>> 
>>> OK. Perhaps I was not clear. Please read my original post again:
>>> 
>>> I've been trying to diagnose why my server has a constant idle time
>> of 90%
>>> even when nothing is running.
>>> 
>>> I did not say "90% usage" I said "90% idle". Meaning I have a constant
>> CPU
>>> usage of 10%.
>>> 
>>> This 10% measurement comes from the "hardware interrupt" (hi) from
>> within
>>> "top":
>>> 
>>> Cpu(s): 3.8% us, 2.1% sy, 0.0% ni, 85.5% id, 0.2% wa, 8.0% hi, 0.4% si
>>> 
>>> 
>>> Even when all other percentages are at 0%, hi remains around 10%. How
>> can I
>>> figure out what is causing all these interrupts?
>> 
>> I don't have a clue other then to unload drivers, etc.
>> 
>> 
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>> 
> 
> 
> 
> -- 
> Sig Lange
> http://www.signuts.net/
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Re: [Asterisk-Users] unlocking cisco 7940 phone

[Michael Welter]

I got five of these doctor's phones as well.

You need your server set-up with dhcp and tftp.  In the tftp directory, 
you'll need files SIP.cnf and SIPDefault.cnf available.  You'll 
also need the current SIP firmware load.


Then, you have to let the phone power up, then enter the password 
'cisco', the go to settings->3->28 (remove config) and also ->25 to 
enable DHCP.  Then hit 'yes'.


If your files are ok then the phone will reboot, upgrade its firmware, 
and read the new configuration.


I performed this procedure five times this morning and it worked every time!

5¢ please  :-)




Rich Adamson wrote:

Unfortunately, it didn't work.  Do you know any other ways?



Brian S. Reale
President

Colosa Inc.
Tel:  305.675.1400 ext. 201
Fax: 305.402.0282
www.colosa.com 
www.ProcessMaker.com 



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tom Vile
Sent: Saturday, September 17, 2005 9:04 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] unlocking cisco 7940 phone

try going into settings and then unlock config and try typing in cisco

On 9/17/05, Brian <[EMAIL PROTECTED]> wrote:





 


I have a Cisco 7940 phone with a locked SIP configuration.  There is no


tftp


server configured in network settings.  Does anyone know how to get this
phone to upgrade its firmware and thereby unlock the SIP settings?  (The


**#

combination on the Cisco manuals does not work). 

 

-brian 



I just had to do one the other day; the phone shipped with mgcp installed
with some Doctors name appearing on the screen. It had been locked down
pretty tight.

The only way that I could do it is to use a sniffer (eg, ethereal) to
see the IP addresses it was trying to reach.

I then put the phone on a network that had a linux server with that
exact addressd on it. The phone happily read the tftp files that I
had prestaged on the linux server, and upgraded itself to the sip image.

The upgrade process wiped out all passwords, etc, and reset the phone
to factory defaults for sip (except for the password, which I specified
in a tftp config file).

Once that was done, I could access the phone menues just fine and reset
everything to the way that I wanted it.

If you're not comfortable reading packet traces, then you'll probably
have to send the phone to someone that is.


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Re: [Asterisk-Users] AstriCon 2006 Location

[Robert Hajime Lanning]

> On Sun, Sep 18, 2005 at 11:32:00AM -0500, Brian Capouch wrote:
>> Yes, but what would one do there?
>>
>> One who doesn't gamble, drink, or carouse, that is.
>>
>> I am making my first trip to LV later this Fall, and I dread it.
>> I can't imagine what I'll be able to find to do when I'm not at
>> the conference.
>
> It's ok, I don't either  :-)  I was actually kind of wondering
> the same thing.  I'm sure there's something to do that doesn't
> involve all of that.

If you are willing to spend some money (not on gambling...)

I went and saw the shows.

Magician Lance Burton
Cirque du Soleil - "Mystere" "Ka" "O" "Zumanity"
Blue Man Group

Try going to http://www.lasvegas24hours.com/
On the left, put in the dates you will be there and it will find
the shows available.

-- 
And, did Guloka think the Ulus were too ugly to save?
 -Centauri

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Re: [Asterisk-Users] Most desireable Linux distribution for Asterisk?

[Matt Florell]

We have over 12 Asterisk servers running in production right now on
Slackware. We have used RedHat, Fedora, CentOS and  Mandrake in
the past as well(we have since switched all servers to Slackware). For
us Slackware runs better and gives us more control. That and we like
installing everything from source.

MATT--On 9/19/05, Nathan Pralle <[EMAIL PROTECTED]> wrote:
>>>We have a few Asterisk boxes running under Fedora C3 with no issues.>>>Before we roll into full production mode, we're wondering if the gurus>>>prefer any particular Linux distribution?
Asterisk is really, really good at running on just about anything.Where you get into distributions that are better or otherwise is whenyou add in 3rd party utilities, tools, and applications.  If you are
using a bunch of those, check for THEIR requirements, as thedistribution might matter more.I'm using Slackware on my Asterisk boxes and it runs like a dream.  Ihave also managed Asterisk on FreeBSD, Debian, and RedHat w/o problems.
Nathan

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[Asterisk-Users] Point to Point with Fritz Card ...

[Joao Correia]

Hello all,

Does anyone has any experience with Point to Point Fritz Card and  
Asterisk ?


I have a BRI access Point to Multipoint working fine but I can only  
have 3 numbers.


The phone telco said that if they change to Point to Point I can have  
10 numbers.


Does anyone has any experience with Point to Point ?

Best regards
Joao Correia 
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Re: [Asterisk-Users] AstriCon 2006 Location

[Asterisk]

A couple from Europe - London or even better Edinburgh.

Gary

Rich Adamson wrote:

The best place for Astri Con 2006 would definatly be
Omaha, Nebraska! ;)  very central

...ah one could hope.



or Lincoln, better facilities ;)


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Re: [Asterisk-Users] IAX dialplan problem?

[John Crowhurst]

On 9/19/05, Colin Anderson <[EMAIL PROTECTED]> wrote:







[default]
 
exten 
=> 1,1,Playback(YouPressedOne)
exten 
=> 1,2,Playback(BecauseYouPressed1IWillCallCarol)
exten 
=> 1,3,Goto(first-ivr,s,1)
 

exten 
=> 2,1,Playback(YouPressedTwo)
exten 
=> 2,2,Playback(BecauseYouPressed1IWillCallCarol)
exten 
=> 2,3,Goto(second-ivr,s,1)
 
exten => 
3,1,Playback(YouPressedTwo)

exten 
=> 
3,2,Playback(BecauseYouPressed3IWillSendYouToVoicemail)
exten 
=> 3,3,Goto(third-ivr,s,1)
 
[first-ivr]
 
exten 
=> s,1,Dial(InsertDialStatementHere)

exten 
=> s,2,Playback(Goodbye)
exten 
=> s,3,Hangup
 

[second-ivr]
 
exten 
=> s,1,Dial(InsertDialStatementHere)

exten 
=> s,2,Playback(Goodbye)
exten => 
s,3,Hangup
 
[third-ivr]
 
exten 
=> s,1,VoiceMailMain(default)
exten => 
s,2,Playback(Goodbye)
exten => 
s,3,Hangup
 
Get it? In a context (which is the name 
specified in brackets) , the first number in a statement indicates the 
extension, which Asterisk interprets through a DTMF tone (you press buttons to 
dial something) or a DID number for inbound calls (in the case of a PRI) and 
executes statements according to the priority, which is the next number. 


In the above example, you have to press 1,2 or 3 before it will tell
you that. I was wanting it to tell me before I pressed those numbers.

Can an IAX device be setup so that it appears to be an inbound DID, and if so how can I do that with the IAXy?


--
John

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[Asterisk-Users] Dial time limit doesn't work when calling party transfers

[P.]

Hi,

I'm using the AbsoluteTimeout and Dial (with L() option) commands to set
a timeout for my calls, but when the calling user transfers a call the
timeout doesn't work and the call last until hanging-up.

If the call is not transfered the limit works just fine.

¿How can I make this work?

Thanks in advance.

My asterisk version is 1-0-9-07 and here's an example of one of the
macros I use:

[macro-long-distance]
; Macro for long distance calls
exten=>s,1,Authenticate(4567)
exten=>s,2,AbsoluteTimeout(${ARG3})
exten=>s,3,SetAccount(2)
exten=>s,4,Dial(${ARG1},${ARG2}|TtL(${ARG3}000:3))
exten=>s,5,Congestion(5)
exten=T,1,ResetCDR(w)
exten=T,2,NoCDR
exten=T,3,Hangup
exten=t,1,ResetCDR(w)
exten=t,2,NoCDR
exten=t,3,Hangup


-- 
Alejandro Ríos P. 

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Re: [Asterisk-Users] Silence suppression /RTP VAD and Asterisk? Dropped calls on IP-500

[Jeremy Gault]

david.j as wrote:



problems.  They tell me that I need to disable silence suppression on
the IP-500 because Asterisk won't support it and either Asterisk or
the MG thinks the call is dropped in abstence of RTP packets.

However I can't find any reference to this in the Polycom manual. 
Does anyone know how to turn VAD/Silence suppression on the Polycom --
or if it even supports it?  Is there a work around with Asterisk? 
 

We have it disabled here.  It is in the ipmid.cfg file.  Open up your 
ipmid.cfg file and do a search for VAD.  We have a line like this:




To be more specific, look in your ipmid.cfg under the  section.  
That is where you should find it.  Most likely, voice.vadEnable="1" in 
your installation.  Change this to 0, save the file, and reboot the 
phones.  If VAD (Voice Activity Detetion) is indeed the cause of your 
problem, it should go away after updating the config and rebooting.


 Jeremy

--
Jeremy Gault, KD4NED<[EMAIL PROTECTED]>
Network Administrator, WinWorld Corporation
Member: Bradley County ACS/RACES/SkyWarn
voice: +1.423.473.8084  fax: +1.423.472.9465
fwd: 461771 msn msgr: [EMAIL PROTECTED]

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Re: [Asterisk-Users] OT: Hardware Interrupts; Who is it?

[Matthew Boehm]

   CPU0   CPU1
  0: 85 1703809954IO-APIC-edge  timer
  8:  0  0IO-APIC-edge  rtc
  9:  0  1   IO-APIC-level  acpi
 14:  0 31IO-APIC-edge  ide0
177:  0   17840313   IO-APIC-level  megaraid
185:  0 1817423967   IO-APIC-level  eth0
193:  0   40198530   IO-APIC-level  eth1
201:  0 3507106255   IO-APIC-level  wanpipe1, wanpipe2, wanpipe3,
wanpipe4
NMI:  0  0
LOC: 1633394197 1633394188
ERR:  0
MIS:  0

Any idea what "LOC" and "timer" are? I did "watch -n1 cat /proc/interrupts"
for about a minute. The largest movers were "timer", "LOC" and "eth0". The
others either never change or hardly changed.

-Matthew

> From: Sig Lange <[EMAIL PROTECTED]>
> Reply-To: <[EMAIL PROTECTED]>, Asterisk Users Mailing List - Non-Commercial
> Discussion 
> Date: Mon, 19 Sep 2005 15:34:26 -0400
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
> Subject: Re: [Asterisk-Users] OT: Hardware Interrupts; Who is it?
> 
> It means you have a piece of hardware that is generating a lot of interupts.
> Try a few commands like this:
> 
> # vmstat 1
> # watch -n1 cat /proc/interrupts
> Go through lspci -vb and disable hardware that's not being used.
> 
> Watch the numbers as they increase. Also check for ERR and MIS
> 
> 
> On 9/19/05, Rich Adamson <[EMAIL PROTECTED]> wrote:
>> 
>> 
>>> OK. Perhaps I was not clear. Please read my original post again:
>>> 
>>> I've been trying to diagnose why my server has a constant idle time
>> of 90%
>>> even when nothing is running.
>>> 
>>> I did not say "90% usage" I said "90% idle". Meaning I have a constant
>> CPU
>>> usage of 10%.
>>> 
>>> This 10% measurement comes from the "hardware interrupt" (hi) from
>> within
>>> "top":
>>> 
>>> Cpu(s): 3.8% us, 2.1% sy, 0.0% ni, 85.5% id, 0.2% wa, 8.0% hi, 0.4% si
>>> 
>>> 
>>> Even when all other percentages are at 0%, hi remains around 10%. How
>> can I
>>> figure out what is causing all these interrupts?
>> 
>> I don't have a clue other then to unload drivers, etc.
>> 
>> 
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> 
> 
> 
> -- 
> Sig Lange
> http://www.signuts.net/
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Re: [Asterisk-Users] Most desireable Linux distribution for Asterisk?

[Paul]

José Pablo Ezequiel Fernández wrote:


On Monday 19 September 2005 16:06, Michiel van Baak wrote:
 


On 14:05, Mon 19 Sep 05, Asterisk Lists wrote:
   


We have a few Asterisk boxes running under Fedora C3 with no issues.
Before we roll into full production mode, we're wondering if the gurus
prefer any particular Linux distribution?

There are pre-packaged Asterisk solutions out there based on WBEL,
CentOS and a few others, so there doesn't seem to be a "de-facto best
choice", but I figured it was worth asking.
 


Hi,

There is no real answer to your question.
just use one you're most familiar with.

Personally I wouldn't survive without Debian ;)
   


And I would with it, I prefeer Gentoo.
Not tring to start a flamewar, just to show that there are various 
distributions that are good for asterisk, almost any mainstream distro 
targetted at servers (I wouldn't expect Ark Linux to be a good distro for 
Asterisk) may be a good distro, and the big ones like Debian and Gentoo 
already have some Asterisk packages.
 

Debian has always been great for my server needs. I wouldn't bash other 
distros in general. I would avoid using a distro that focuses mostly on 
being a user-friendly workstation for something like asterisk.


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RE: [Asterisk-Users] unlocking cisco 7940 phone

[Rich Adamson]

> Unfortunately, it didn't work.  Do you know any other ways?
> 
> 
> 
> Brian S. Reale
> President
>  
> Colosa Inc.
> Tel:  305.675.1400 ext. 201
> Fax: 305.402.0282
> www.colosa.com 
> www.ProcessMaker.com 
>  
>  
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Tom Vile
> Sent: Saturday, September 17, 2005 9:04 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] unlocking cisco 7940 phone
> 
> try going into settings and then unlock config and try typing in cisco
> 
> On 9/17/05, Brian <[EMAIL PROTECTED]> wrote:
> >  
> >  
> > 
> >   
> > 
> > I have a Cisco 7940 phone with a locked SIP configuration.  There is no
> tftp
> > server configured in network settings.  Does anyone know how to get this
> > phone to upgrade its firmware and thereby unlock the SIP settings?  (The
> **#
> > combination on the Cisco manuals does not work). 
> > 
> >   
> > 
> > -brian 

I just had to do one the other day; the phone shipped with mgcp installed
with some Doctors name appearing on the screen. It had been locked down
pretty tight.

The only way that I could do it is to use a sniffer (eg, ethereal) to
see the IP addresses it was trying to reach.

I then put the phone on a network that had a linux server with that
exact addressd on it. The phone happily read the tftp files that I
had prestaged on the linux server, and upgraded itself to the sip image.

The upgrade process wiped out all passwords, etc, and reset the phone
to factory defaults for sip (except for the password, which I specified
in a tftp config file).

Once that was done, I could access the phone menues just fine and reset
everything to the way that I wanted it.

If you're not comfortable reading packet traces, then you'll probably
have to send the phone to someone that is.


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RE: [Asterisk-Users] T.38 & Canreinvite (yes, again)

[Joshua Colp - Asterlink]

Hello,

Asterisk does not act as a SIP Proxy as you may have in mind. Each call is
treated independently, that is - codec capabilities of one call don't go to
the other one during a reinvite. Only the IP address and Port go.

Joshua Colp

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Monday, September 19, 2005 5:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] T.38 & Canreinvite (yes, again)

I know this has been asked before, but I've checked the archives and I 
haven't found anybody that has given a definitive yes or no, just "yeah, 
it should work.".  If I have a T.38 gateway like a Cisco 5300 and a 
T.38 ATA (whatever model) and I have canreinvite=yes, should T.38 work?

I have it setup and it doesn't work, so I want to know if I am doing 
something wrong, or if it just won't work.  If I make a voice call, I 
see the media stream go from the gateway to the ata directly.  When I 
fax, I see the stream go that way as well, but it is g.729.  I see 
INVITE messages from my ATA that reference T.38, but they go to the * 
box, not the gateway and therefore * ignores it.  Any thoughts?

PA
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Re: [Asterisk-Users] unlocking cisco 7940 phone

[chouck]

Yeah, if it's before a sip upgrade I think you go to the tftp field, and hit 
*0 or *#, I'd google it.  Try it a few times and edit should appear.  If 
that doesn't work setup a dhcp server that hands out your tftp address and 
upgrade it this way, then once it's upgraded unlock's password will be 
"cisco", unless of course you changed it in the profile.


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

Sent: Monday, September 19, 2005 4:10 PM
Subject: RE: [Asterisk-Users] unlocking cisco 7940 phone



Unfortunately, it didn't work.  Do you know any other ways?



Brian S. Reale
President

Colosa Inc.
Tel:  305.675.1400 ext. 201
Fax: 305.402.0282
www.colosa.com
www.ProcessMaker.com


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tom Vile
Sent: Saturday, September 17, 2005 9:04 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] unlocking cisco 7940 phone

try going into settings and then unlock config and try typing in cisco

On 9/17/05, Brian <[EMAIL PROTECTED]> wrote:






I have a Cisco 7940 phone with a locked SIP configuration.  There is no

tftp

server configured in network settings.  Does anyone know how to get this
phone to upgrade its firmware and thereby unlock the SIP settings?  (The

**#

combination on the Cisco manuals does not work).



-brian
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--
Tom Vile
Baldwin Technology Solutions, Inc
Consulting - Web Design - VoIP Telephony
Phone: 518-631-2855 x205
Fax: 518-631-2856
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RE: [Asterisk-Users] kill a .call file

[Rene Kluwen]

I have the same problem. Asterisk always makes two calls, even when the
first one went through successfully.

Rene Kluwen
Chimit

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Remco
Barende
Sent: maandag 19 september 2005 22:09
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] kill a .call file


On Mon, 19 Sep 2005, jltaylor wrote:

>> From my CLI:
>
> Attempting call on SIP/gw/19857749166 for [EMAIL PROTECTED]:1
> (Retry 114)
> Attempting call on SIP/gw/19037747603 for [EMAIL PROTECTED]:1
> (Retry 83)
> Attempting call on SIP/gw/19857747603 for [EMAIL PROTECTED]:1
> (Retry 80)
>
> I want to stop it from any future attempts.
>
> Any idea about a command to kill or where the data is stored?

This is an asterisk bug. I already filed it but they need a full trace and
I haven't had the time yet to do it.

It seems that * keeps retrying the call, even when it was succesfully
completed.

Annoying.


>
> James Taylor
> MetroTel
> 3505 Summerhill Road
> Suite 11
> Texarkana, Tx  75503
> 903-793-1956
>
>
>> -Original Message-
>> From: [EMAIL PROTECTED]
>> [mailto:[EMAIL PROTECTED] Behalf Of trixter
>> http://www.0xdecafbad.com
>> Sent: Monday, September 19, 2005 1:59 PM
>> To: Asterisk Users Mailing List - Non-Commercial Discussion;
>> [EMAIL PROTECTED]
>> Subject: Re: [Asterisk-Users] kill a .call file
>>
>>
>> On Mon, 2005-09-19 at 13:43 -0500, jltaylor wrote:
>>> Any means of killing a .call file that is in progress?
>>>
>>
>> You mean once the call has begun?  You prolly want to hangup the
>> call ...
>>
>> asterisk -rx "soft hangup "
>>
>> Or is there something else that you wanted?
>>
>>
>> --
>> Trixter http://www.0xdecafbad.com Bret McDanel
>> UK +44 870 340 4605   Germany +49 801 777 555 3402
>> US +1 360 207 0479 or +1 516 687 5200
>> FreeWorldDialup: 635378
>>
>
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RE: [Asterisk-Users] IAX dialplan problem?

[Colin Anderson]

[default]
 
exten 
=> 1,1,Playback(YouPressedOne)
exten 
=> 1,2,Playback(BecauseYouPressed1IWillCallCarol)
exten 
=> 1,3,Goto(first-ivr,s,1)
 

exten 
=> 2,1,Playback(YouPressedTwo)
exten 
=> 2,2,Playback(BecauseYouPressed1IWillCallCarol)
exten 
=> 2,3,Goto(second-ivr,s,1)
 
exten => 
3,1,Playback(YouPressedTwo)

exten 
=> 
3,2,Playback(BecauseYouPressed3IWillSendYouToVoicemail)
exten 
=> 3,3,Goto(third-ivr,s,1)
 
[first-ivr]
 
exten 
=> s,1,Dial(InsertDialStatementHere)

exten 
=> s,2,Playback(Goodbye)
exten 
=> s,3,Hangup
 

[second-ivr]
 
exten 
=> s,1,Dial(InsertDialStatementHere)

exten 
=> s,2,Playback(Goodbye)
exten => 
s,3,Hangup
 
[third-ivr]
 
exten 
=> s,1,VoiceMailMain(default)
exten => 
s,2,Playback(Goodbye)
exten => 
s,3,Hangup
 
Get it? In a context (which is the name 
specified in brackets) , the first number in a statement indicates the 
extension, which Asterisk interprets through a DTMF tone (you press buttons to 
dial something) or a DID number for inbound calls (in the case of a PRI) and 
executes statements according to the priority, which is the next number. 

 
The first number 'exten => 
1' allows you to group related statements together and 
bind them to an extension number ( which could be internal / outbound, 
could be external / inbound) and the second number 'exten => 1,1' 
specifies the order in which the statements are executed. After a 
statement is executed, and it could be executed incorrectly or correctly, or it 
could have executed successfully or unsucessfully, it will then process the next 
line in the context until it runs out of stuff to do. 
 
When you dial out, Asterisk determines the 
context in which your phone is to be run in, and tries to match what you dialled 
to an extension specified in the context for your phone. If there is no match, 
it tries the s extension. If the s extension does not exist or is invalid, 
Asterisk returns a busy tone and "404 not found" on your phone. 
 
When someone dials in, Asterisk tries to 
match the DID number against an extension in the inbound calls context. If there 
is no match, it tries the s extension in the inbound calls context. If the s 
extension does not exist or is invalid, Asterisk rejects the call. 
 
The s extension is a special kind of 
extension. It means, "Execute these statements regardless of the extension 
specified in the context unless told otherwise, or if there is no match for the 
extension requested".  In the example above, pressing 1 or 2 will do 
something, but if you press any other digit, Asterisk will hang up the call. 
Why? Because in the [default] context, I specified something for Asterisk to do 
if I press 1 or 2, but I didn't specify it for any other number 
and I didn't specify an 's' extension which would have handled a keypress other 
than 1 or 2. 

  -Original Message-From: John Crowhurst 
  [mailto:[EMAIL PROTECTED]Sent: Monday, September 19, 2005 1:23 
  PMTo: Asterisk-Users@lists.digium.comSubject: Re: 
  [Asterisk-Users] IAX dialplan problem?
  On 9/19/05, Thameem 
  Ansari <[EMAIL PROTECTED]> 
  wrote:
  You 
are doing correct. But you have to explain what you want to do? As per your 
second configuration, if you dial 1 then it will ringPost exactly what 
you are trying to accomplish?
  I don't actually have 
  dialplan created, as I'm trying to see if I can get it to work. 
  What I would like to do is present me with a menu when I first 
  press the connect button on the phone, give me a set of options like 'press 1 
  to call Carol', 'press 2 to call Jenny', 'press * for voicemail', 
  etc.--John
  -Thameem

On 
9/19/05, John Crowhurst < [EMAIL PROTECTED]> 
wrote:

  Hello, I'm a newbie to the 
  asterisk system.I'm trying to configure a dialplan so that when I 
  use my IAXy it will prompt me with an IVR and then send me off to 
  different things like dial and voicemail from that.I've tried 
  various combinations but I can't seem to get it to work properly. Here is 
  an example:[default]exten => s,1,Answerexten => 
  s,2,RingingIt gives me a dialtone and waits for an input, but if I 
  do:[default]exten => 1,1,Answerexten => 
  1,2,RingingAnd then dial 1 it rings...Am I doing something 
  wrong? Any suggestions or pointers gratefully received.Thanks in 
  advance,John___--Bandwidth 
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Re: [Asterisk-Users] Orinoco Injectors

[Michael Loftis]

--On September 16, 2005 6:06:20 PM -0400 Darren Wright 
<[EMAIL PROTECTED]> wrote:



Has anyone gotten the Lucent / Orinoco injectors (AE-1, AE-6, AE-12) to
work with the Cisco 79* series phones?

I'm not sure if the are the statndard POE or not


Cisco's phones are not standard POE.  They reversed the polarity, and I 
think they run the power hot all the time.  Can't remember specifically.



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[Asterisk-Users] Silence suppression /RTP VAD and Asterisk? Dropped calls on IP-500

[david.j as]

Hello,

I run Asterisk in a 100% VOIP installation with the Polycom IP-500 phones.
Every once and a while I have problems with either dropped calls
between Asterisk and my provider,  or invalid RTP audio streams with
phones behind NAT.  I have had a few Asterisk developers look into my
installation and even my provider check my setup but still am having
problems.  They tell me that I need to disable silence suppression on
the IP-500 because Asterisk won't support it and either Asterisk or
the MG thinks the call is dropped in abstence of RTP packets.

 However I can't find any reference to this in the Polycom manual. 
Does anyone know how to turn VAD/Silence suppression on the Polycom --
or if it even supports it?  Is there a work around with Asterisk? 
Thanks in advance.

David
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[Asterisk-Users] T.38 & Canreinvite (yes, again)

[[EMAIL PROTECTED]]

I know this has been asked before, but I've checked the archives and I 
haven't found anybody that has given a definitive yes or no, just "yeah, 
it should work.".  If I have a T.38 gateway like a Cisco 5300 and a 
T.38 ATA (whatever model) and I have canreinvite=yes, should T.38 work?


I have it setup and it doesn't work, so I want to know if I am doing 
something wrong, or if it just won't work.  If I make a voice call, I 
see the media stream go from the gateway to the ata directly.  When I 
fax, I see the stream go that way as well, but it is g.729.  I see 
INVITE messages from my ATA that reference T.38, but they go to the * 
box, not the gateway and therefore * ignores it.  Any thoughts?


PA
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Re: [Asterisk-Users] Pinging ...

[Alan Bunch]

Nope, no alarm monitor, theses are new installs and no alarm.

alabun

Ariel Batista wrote:


Alan Bunch wrote:


Ok, if I missed something in the wiki please point me there with the
correct search terms.

Asterisk 1.0.7  (AAH really)

4 co lines from Bellsouth into a Diguim T400P.

Polycom 501 x 4 on the desktops.

My problem is on calls to or from the CO I hear a beeping every 12
seconds.



Sounds like your main line has the Alarm monitor on it.  Have you 
check that out?



You can set your watch to it. Could this be a "call recording in
progress" tone.
I have not made any effort to turn on call recording.  If it is I'll
go figure out how to
turn it off.  Could this be echo cancelation "retraining"

Any guesses from anyone here. I just want the beeping to stop.  Well I
don't really care but my users sure do 8-)

One more thing, this machine is remotely managed and is in a distant
city.  If I really have to go I can.

Thank in advance
Alan
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Re: [Asterisk-Users] Differ between "private" and "out of area"?

[Rich Adamson]

> I have CID identificator connected in parallel with X101P/Asterisk, and
> it displays "Private" for hidden calls, and "Out of area" for calls from
> rural areas (with old phone systems). But Asterisk always deliver
> CALLERIDNUM="", CALLERIDNAME="" in both cases.

Paste the section from zapata.conf that handles the x101p.


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RE: [Asterisk-Users] unlocking cisco 7940 phone

[Brian]

Unfortunately, it didn't work.  Do you know any other ways?



Brian S. Reale
President
 
Colosa Inc.
Tel:  305.675.1400 ext. 201
Fax: 305.402.0282
www.colosa.com 
www.ProcessMaker.com 
 
 
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tom Vile
Sent: Saturday, September 17, 2005 9:04 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] unlocking cisco 7940 phone

try going into settings and then unlock config and try typing in cisco

On 9/17/05, Brian <[EMAIL PROTECTED]> wrote:
>  
>  
> 
>   
> 
> I have a Cisco 7940 phone with a locked SIP configuration.  There is no
tftp
> server configured in network settings.  Does anyone know how to get this
> phone to upgrade its firmware and thereby unlock the SIP settings?  (The
**#
> combination on the Cisco manuals does not work). 
> 
>   
> 
> -brian 
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> 


-- 
Tom Vile
Baldwin Technology Solutions, Inc
Consulting - Web Design - VoIP Telephony
Phone: 518-631-2855 x205
Fax: 518-631-2856
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RE: [Asterisk-Users] kill a .call file

[Remco Barende]

On Mon, 19 Sep 2005, jltaylor wrote:


From my CLI:


Attempting call on SIP/gw/19857749166 for [EMAIL PROTECTED]:1
(Retry 114)
Attempting call on SIP/gw/19037747603 for [EMAIL PROTECTED]:1
(Retry 83)
Attempting call on SIP/gw/19857747603 for [EMAIL PROTECTED]:1
(Retry 80)

I want to stop it from any future attempts.

Any idea about a command to kill or where the data is stored?


This is an asterisk bug. I already filed it but they need a full trace and 
I haven't had the time yet to do it.


It seems that * keeps retrying the call, even when it was succesfully 
completed.


Annoying.




James Taylor
MetroTel
3505 Summerhill Road
Suite 11
Texarkana, Tx  75503
903-793-1956



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of trixter
http://www.0xdecafbad.com
Sent: Monday, September 19, 2005 1:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion;
[EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] kill a .call file


On Mon, 2005-09-19 at 13:43 -0500, jltaylor wrote:

Any means of killing a .call file that is in progress?



You mean once the call has begun?  You prolly want to hangup the
call ...

asterisk -rx "soft hangup "

Or is there something else that you wanted?


--
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378



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Re: [Asterisk-Users] kill a .call file

[Jeremy Gault]

If I understand correctly what you are wanting to do, I think you can 
simply delete the call file from the Asterisk outgoing spool directory 
and it will stop it from trying again.


If I'm wrong, someone correct me on this. :)

 Jeremy

jltaylor wrote:


From my CLI:


Attempting call on SIP/gw/19857749166 for [EMAIL PROTECTED]:1
(Retry 114)
Attempting call on SIP/gw/19037747603 for [EMAIL PROTECTED]:1
(Retry 83)
Attempting call on SIP/gw/19857747603 for [EMAIL PROTECTED]:1
(Retry 80)

I want to stop it from any future attempts.

Any idea about a command to kill or where the data is stored?

James Taylor
MetroTel
3505 Summerhill Road
Suite 11
Texarkana, Tx  75503
903-793-1956


 


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of trixter
http://www.0xdecafbad.com
Sent: Monday, September 19, 2005 1:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion;
[EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] kill a .call file


On Mon, 2005-09-19 at 13:43 -0500, jltaylor wrote:
   


Any means of killing a .call file that is in progress?

 


You mean once the call has begun?  You prolly want to hangup the
call ...

asterisk -rx "soft hangup "

Or is there something else that you wanted?


--
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378

   



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--
Jeremy Gault, KD4NED<[EMAIL PROTECTED]>
Network Administrator, WinWorld Corporation
Member: Bradley County ACS/RACES/SkyWarn
voice: +1.423.473.8084  fax: +1.423.472.9465
fwd: 461771 msn msgr: [EMAIL PROTECTED]

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Re: [Asterisk-Users] Most desireable Linux distribution for Asterisk?

[Nathan Pralle]

We have a few Asterisk boxes running under Fedora C3 with no issues.
Before we roll into full production mode, we're wondering if the gurus
prefer any particular Linux distribution?


Asterisk is really, really good at running on just about anything. 
Where you get into distributions that are better or otherwise is when 
you add in 3rd party utilities, tools, and applications.  If you are 
using a bunch of those, check for THEIR requirements, as the 
distribution might matter more.


I'm using Slackware on my Asterisk boxes and it runs like a dream.  I 
have also managed Asterisk on FreeBSD, Debian, and RedHat w/o problems.


Nathan

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-
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Give the Director a Serpent Deflector
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Re: [Asterisk-Users] AstriCon 2006 Location

[Matt Fredrickson]

On Sun, Sep 18, 2005 at 11:32:00AM -0500, Brian Capouch wrote:
> Senad J wrote:
> >>If you are looking for the maximum number of cheap flights from around
> >>the world, and plenty of convention and room space, the answer is Las
> >>Vegas :-)
> >
> >
> >I would definitively agree!
> >
> 
> Yes, but what would one do there?
> 
> One who doesn't gamble, drink, or carouse, that is.
> 
> I am making my first trip to LV later this Fall, and I dread it.  I 
> can't imagine what I'll be able to find to do when I'm not at the 
> conference.

It's ok, I don't either  :-)  I was actually kind of wondering the same
thing.  I'm sure there's something to do that doesn't involve all of that.

-- 
Matthew Fredrickson
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Re: [Asterisk-Users] Most desireable Linux distribution for Aster isk?

[Ariel Batista]

Colin Anderson wrote:

There is no real answer to your question.
just use one you're most familiar with.




I use RH allot so I am now using CentOS mostly. It's Red Hat EL GPL code and 
so far everything I have runs on it without issues. Great OS.



Second that, using FC2 for me, and it's the Devil You Know, right? I
have a good handle on the RedHat Way which the Debian guys say is
stupid, but it's not, it's just different. All of the little RedHat-y
things you get used to and when you use a different distro, it's
like, WTF?? I like YUM and RPM's, so RedHat it is, for me.

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Re: [Asterisk-Users] Pinging ...

[Ariel Batista]

Alan Bunch wrote:

Ok, if I missed something in the wiki please point me there with the
correct search terms.

Asterisk 1.0.7  (AAH really)

4 co lines from Bellsouth into a Diguim T400P.

Polycom 501 x 4 on the desktops.

My problem is on calls to or from the CO I hear a beeping every 12
seconds.


Sounds like your main line has the Alarm monitor on it.  Have you check that 
out?



You can set your watch to it. Could this be a "call recording in
progress" tone.
I have not made any effort to turn on call recording.  If it is I'll
go figure out how to
turn it off.  Could this be echo cancelation "retraining"

Any guesses from anyone here. I just want the beeping to stop.  Well I
don't really care but my users sure do 8-)

One more thing, this machine is remotely managed and is in a distant
city.  If I really have to go I can.

Thank in advance
Alan
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Re: [Asterisk-Users] Differ between "private" and "out of area"?

[Goran Dj]

I have CID identificator connected in parallel with X101P/Asterisk, and
it displays "Private" for hidden calls, and "Out of area" for calls from
rural areas (with old phone systems). But Asterisk always deliver
CALLERIDNUM="", CALLERIDNAME="" in both cases.

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[Asterisk-Users] Asterisk Keep Crashing need Help please

[Ka Lun Chan]

Hi All, 
 
My Asterisk Server crash at least once a day. The full and message log files did not show any errors or indication on why the asterisk service crashed. After looking at the core dump file, I am still unable to identify the problem. Can someone please give me some guideance on what I can do? Thank You All

 
Asterisk is loaded with version 1.0.5, PyAsterisk, and postgresql. Is used for IP Phone only with SIP protocols only. 
Server Spec: Dual Xeon 3.0 processor with 2G memory
Here is the core dump:
 
Program terminated with signal 11, Segmentation fault.#0  ed_next_char (el=0x6, c=-1073749366) at common.c:315315 return (CC_REFRESH);(gdb) bt#0  ed_next_char (el=0x6, c=-1073749366) at 
common.c:315#1  0x080b72db in ast_strcasestr (haystack=0x80fbf30 "èº\017\b î\022B\200ï\022B", needle=0xbfffe2dc "") at utils.c:389#2  0x0809cb32 in session_do (data="" at manager.c:252#3  0x42015504 in ?? ()
#4  0x0002 in ?? ()#5  0xbfffe674 in ?? ()#6  0xbfffe680 in ?? ()#7  0x4001582c in ?? ()#8  0x0002 in ?? ()#9  0x08052ac0 in __do_global_dtors_aux ()#10 0x08052ae1 in frame_dummy ()
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[Asterisk-Users] Re: Pinging ...

[Noah Miller]

Hi Alan -


Asterisk 1.0.7  (AAH really)
4 co lines from Bellsouth into a Diguim T400P.
Polycom 501 x 4 on the desktops.

My problem is on calls to or from the CO I hear a "pinging" (thing  
sonar

ping in a submarine) every 12 seconds.  You can set your watch to it.
COuld this be a "call recording in progress" tone.  I have not made  
any

effort to turn on call recording.  If it is I'll go figure out how to
turn it off.  Could this be echo cancelation "retraining"  Am I just
u8nder dosing on my morning meds ?

Any guesses from anyone here. I just want the "pinging" to stop.

One more thing, this machine is remotely managed and is in a distant
city.  If I really have to go I can.


My best guess would be rf interference local to the data center where  
the server is.  For a while, I had a cordless phone near one of our  
asterisk servers with analog lines, and it made some pretty weird  
sounds during calls.


- Noah

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RE: [Asterisk-Users] Most desireable Linux distribution for Aster isk?

[Colin Anderson]

>There is no real answer to your question.
>just use one you're most familiar with.

Second that, using FC2 for me, and it's the Devil You Know, right? I have a
good handle on the RedHat Way which the Debian guys say is stupid, but it's
not, it's just different. All of the little RedHat-y things you get used to
and when you use a different distro, it's like, WTF?? I like YUM and RPM's,
so RedHat it is, for me. 

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RE: [Asterisk-Users] OT: Hardware Interrupts; Who is it?

[Kevin Collins]

Cat /proc/interrupts and watch the devices 

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Boehm
Sent: Monday, September 19, 2005 3:16 PM
To: Asterisk Users
Cc: Rich Adamson
Subject: Re: [Asterisk-Users] OT: Hardware Interrupts; Who is it?

OK. Perhaps I was not clear. Please read my original post again:

 I've been trying to diagnose why my server has a constant idle time of 90%
 even when nothing is running.

I did not say "90% usage" I said "90% idle". Meaning I have a constant CPU
usage of 10%.

This 10% measurement comes from the "hardware interrupt" (hi) from within
"top":

Cpu(s):  3.8% us,  2.1% sy,  0.0% ni, 85.5% id,  0.2% wa,  8.0% hi,  0.4% si


Even when all other percentages are at 0%, hi remains around 10%. How can I
figure out what is causing all these interrupts?

-Matthew


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[Asterisk-Users] Error with VOIP-INFO

[Crystal Stream, Incorporated]

when I searched for _**6XX the site returned:

An error occured in a database query!

Context:
File/tiki-searchresults.php
Url
/tiki-searchresults.php?words=_**6XX&where=pages&search=go
Query:
SELECT COUNT(*) FROM tiki_comments c, tiki_pages p
WHERE c.objectType = "wiki page" AND
p.pageName=c.object AND (UPPER(c.title) REGEXP
'.*_**6XX.*' OR UPPER(c.data) REGEXP '.*_**6XX.*')
Values:

array(5) {
  [0]=>
  array(6) {
["file"]=>
string(46)
"/var/www/html/tikiwiki-1.8.5/lib/tikidblib.php"
["line"]=>
int(109)
["function"]=>
string(9) "sql_error"
["class"]=>
string(9) "searchlib"
["type"]=>
string(2) "->"
["args"]=>
array(3) {
  [0]=>
  &string(185) "SELECT COUNT(*) FROM tiki_comments
c, tiki_pages p WHERE c.objectType = "wiki page" AND
p.pageName=c.object AND (UPPER(c.title) REGEXP
'.*_**6XX.*' OR UPPER(c.data) REGEXP '.*_**6XX.*')"
  [1]=>
  &NULL
  [2]=>
  &bool(false)
}
  }
  [1]=>
  array(6) {
["file"]=>
string(46)
"/var/www/html/tikiwiki-1.8.5/lib/searchlib.php"
["line"]=>
int(106)
["function"]=>
string(6) "getone"
["class"]=>
string(9) "searchlib"
["type"]=>
string(2) "->"
["args"]=>
array(1) {
  [0]=>
  &string(185) "SELECT COUNT(*) FROM tiki_comments
c, tiki_pages p WHERE c.objectType = "wiki page" AND
p.pageName=c.object AND (UPPER(c.title) REGEXP
'.*_**6XX.*' OR UPPER(c.data) REGEXP '.*_**6XX.*')"
}
  }
  [2]=>
  array(6) {
["file"]=>
string(46)
"/var/www/html/tikiwiki-1.8.5/lib/searchlib.php"
["line"]=>
int(110)
["function"]=>
string(5) "_find"
["class"]=>
string(9) "searchlib"
["type"]=>
string(2) "->"
["args"]=>
array(5) {
  [0]=>
  &array(10) {
["from"]=>
string(29) "tiki_comments c, tiki_pages p"
["name"]=>
string(7) "c.title"
["data"]=>
string(6) "c.data"
["hits"]=>
string(6) "p.hits"
["lastModif"]=>
string(13) "c.commentDate"
["href"]=>
string(31) "tiki-index.php?page=%s#comments"
["id"]=>
array(2) {
  [0]=>
  string(10) "p.pageName"
  [1]=>
  string(10) "c.threadId"
}
["pageName"]=>
string(33) "CONCAT(p.pageName, ": ", c.title)"
["search"]=>
array(2) {
  [0]=>
  string(7) "c.title"
  [1]=>
  string(6) "c.data"
}
["filter"]=>
string(50) "c.objectType = "wiki page" AND
p.pageName=c.object"
  }
  [1]=>
  &string(6) "_**6XX"
  [2]=>
  ∫(0)
  [3]=>
  &string(2) "10"
  [4]=>
  &bool(false)
}
  }
  [3]=>
  array(6) {
["file"]=>
string(46)
"/var/www/html/tikiwiki-1.8.5/lib/searchlib.php"
["line"]=>
int(156)
["function"]=>
string(5) "_find"
["class"]=>
string(9) "searchlib"
["type"]=>
string(2) "->"
["args"]=>
array(5) {
  [0]=>
  &array(10) {
["from"]=>
string(29) "tiki_comments c, tiki_pages p"
["name"]=>
string(7) "c.title"
["data"]=>
string(6) "c.data"
["hits"]=>
string(6) "p.hits"
["lastModif"]=>
string(13) "c.commentDate"
["href"]=>
string(31) "tiki-index.php?page=%s#comments"
["id"]=>
array(2) {
  [0]=>
  string(10) "p.pageName"
  [1]=>
  string(10) "c.threadId"
}
["pageName"]=>
string(33) "CONCAT(p.pageName, ": ", c.title)"
["search"]=>
array(2) {
  [0]=>
  string(7) "c.title"
  [1]=>
  string(6) "c.data"
}
["filter"]=>
string(50) "c.objectType = "wiki page" AND
p.pageName=c.object"
  }
  [1]=>
  &string(6) "_**6XX"
  [2]=>
  ∫(0)
  [3]=>
  &string(2) "10"
  [4]=>
  &bool(true)
}
  }
  [4]=>
  array(6) {
["file"]=>
string(51)
"/var/www/html/tikiwiki-1.8.5/tiki-searchresults.php"
["line"]=>
int(165)
["function"]=>
string(10) "find_wikis"
["class"]=>
string(9) "searchlib"
["type"]=>
string(2) "->"
["args"]=>
array(4) {
  [0]=>
  &string(6) "_**6XX"
  [1]=>
  ∫(0)
  [2]=>
  &string(2) "10"
  [3]=>
  &bool(true)
}
  }
}


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[Asterisk-Users] Pinging ...

[Alan Bunch]

Ok, if I missed something in the wiki please point me there with the
correct search terms.

Asterisk 1.0.7  (AAH really)

4 co lines from Bellsouth into a Diguim T400P.

Polycom 501 x 4 on the desktops.

My problem is on calls to or from the CO I hear a beeping every 12 
seconds. 
You can set your watch to it. Could this be a "call recording in 
progress" tone.
I have not made any effort to turn on call recording.  If it is I'll go 
figure out how to
turn it off.  Could this be echo cancelation "retraining" 

Any guesses from anyone here. I just want the beeping to stop.  Well I 
don't really care but my users sure do 8-)


One more thing, this machine is remotely managed and is in a distant
city.  If I really have to go I can.

Thank in advance
Alan
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Re: [Asterisk-Users] OT: Hardware Interrupts; Who is it?

[Sig Lange]

It means you have a piece of hardware that is generating a lot of interupts. Try a few commands like this:

# vmstat 1
# watch -n1 cat /proc/interrupts 
Go through lspci -vb and disable hardware that's not being used. 

Watch the numbers as they increase. Also check for ERR and MIS
On 9/19/05, Rich Adamson <[EMAIL PROTECTED]> wrote:
> OK. Perhaps I was not clear. Please read my original post again:>>  I've been trying to diagnose why my server has a constant idle time of 90%>  even when nothing is running.
>> I did not say "90% usage" I said "90% idle". Meaning I have a constant CPU> usage of 10%.>> This 10% measurement comes from the "hardware interrupt" (hi) from within
> "top":>>
Cpu(s):  3.8% us,  2.1% sy,  0.0% ni,
85.5% id,  0.2% wa,  8.0% hi,  0.4% si>>> Even when all other percentages are at 0%, hi remains around 10%. How can I> figure out what is causing all these interrupts?I don't have a clue other then to unload drivers, etc.
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http://lists.digium.com/mailman/listinfo/asterisk-users-- Sig Langehttp://www.signuts.net/
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RE: [Asterisk-Users] kill a .call file

[jltaylor]

>From my CLI:

Attempting call on SIP/gw/19857749166 for [EMAIL PROTECTED]:1
(Retry 114)
Attempting call on SIP/gw/19037747603 for [EMAIL PROTECTED]:1
(Retry 83)
Attempting call on SIP/gw/19857747603 for [EMAIL PROTECTED]:1
(Retry 80)

I want to stop it from any future attempts.

Any idea about a command to kill or where the data is stored?

James Taylor
MetroTel
3505 Summerhill Road
Suite 11
Texarkana, Tx  75503
903-793-1956


> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of trixter
> http://www.0xdecafbad.com
> Sent: Monday, September 19, 2005 1:59 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion;
> [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] kill a .call file
>
>
> On Mon, 2005-09-19 at 13:43 -0500, jltaylor wrote:
> > Any means of killing a .call file that is in progress?
> >
>
> You mean once the call has begun?  You prolly want to hangup the
> call ...
>
> asterisk -rx "soft hangup "
>
> Or is there something else that you wanted?
>
>
> --
> Trixter http://www.0xdecafbad.com Bret McDanel
> UK +44 870 340 4605   Germany +49 801 777 555 3402
> US +1 360 207 0479 or +1 516 687 5200
> FreeWorldDialup: 635378
>

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Re: [Asterisk-Users] Most desireable Linux distribution for Asterisk?

[José Pablo Ezequiel Fernández]

On Monday 19 September 2005 16:06, Michiel van Baak wrote:
> On 14:05, Mon 19 Sep 05, Asterisk Lists wrote:
> > We have a few Asterisk boxes running under Fedora C3 with no issues.
> > Before we roll into full production mode, we're wondering if the gurus
> > prefer any particular Linux distribution?
> >
> > There are pre-packaged Asterisk solutions out there based on WBEL,
> > CentOS and a few others, so there doesn't seem to be a "de-facto best
> > choice", but I figured it was worth asking.
>
> Hi,
>
> There is no real answer to your question.
> just use one you're most familiar with.
>
> Personally I wouldn't survive without Debian ;)
And I would with it, I prefeer Gentoo.
Not tring to start a flamewar, just to show that there are various 
distributions that are good for asterisk, almost any mainstream distro 
targetted at servers (I wouldn't expect Ark Linux to be a good distro for 
Asterisk) may be a good distro, and the big ones like Debian and Gentoo 
already have some Asterisk packages.
-- 
José Pablo Ezequiel Fernández
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Re: [Asterisk-Users] IAX dialplan problem?

[John Crowhurst]

On 9/19/05, Thameem Ansari <[EMAIL PROTECTED]> wrote:
You are doing correct. But you have to explain what you want to do?
As per your second configuration, if you dial 1 then it will
ringPost exactly what you are trying to accomplish?
I don't actually have dialplan created, as I'm trying to see if I can get it to work. 

What I would like to do is present me with a menu when I first
press the connect button on the phone, give me a set of options like
'press 1 to call Carol', 'press 2 to call Jenny', 'press * for
voicemail', etc.

--
John

-Thameem
On 9/19/05, John Crowhurst <
[EMAIL PROTECTED]> wrote:

Hello, I'm a newbie to the asterisk system.

I'm trying to configure a dialplan so that when I use my IAXy it will
prompt me with an IVR and then send me off to different things like
dial and voicemail from that.

I've tried various combinations but I can't seem to get it to work properly. Here is an example:

[default]
exten => s,1,Answer
exten => s,2,Ringing

It gives me a dialtone and waits for an input, but if I do:

[default]
exten => 1,1,Answer
exten => 1,2,Ringing

And then dial 1 it rings...

Am I doing something wrong? Any suggestions or pointers gratefully received.

Thanks in advance,

John




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Re: [Asterisk-Users] OT: Hardware Interrupts; Who is it?

[Rich Adamson]

> OK. Perhaps I was not clear. Please read my original post again:
> 
>  I've been trying to diagnose why my server has a constant idle time of 
>  90%
>  even when nothing is running.
> 
> I did not say "90% usage" I said "90% idle". Meaning I have a constant CPU
> usage of 10%.
> 
> This 10% measurement comes from the "hardware interrupt" (hi) from within
> "top":
> 
> Cpu(s):  3.8% us,  2.1% sy,  0.0% ni, 85.5% id,  0.2% wa,  8.0% hi,  0.4% si
> 
> 
> Even when all other percentages are at 0%, hi remains around 10%. How can I
> figure out what is causing all these interrupts?

I don't have a clue other then to unload drivers, etc.


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RE: [Asterisk-Users] Asterisk monitoring availability

[Sherwood McGowan]

I personally use Zabbix, and custom scripts that call 
asterisk -rx or the manager API and return numeric digits. For instance, I use 
"asterisk -rx 'sip show peers' | grep --text -i 'OK' | wc -l" to show me how 
many users are connected at once, and test for no data return after 60 seconds 
for deadlock catching. (Asterisk does not always play nice and return the data 
you expect). 
 
Another example: "asterisk -rx 'show channels' | grep 
--text -i 'active call' | awk '{print $1}' " (if I remember correctly) gives me 
my number of active calls. 
 
Hope this was somewhat helpful.

  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Sig 
  LangeSent: Monday, September 19, 2005 2:58 PMTo: 
  Asterisk Users Mailing List - Non-Commercial DiscussionSubject: 
  [Asterisk-Users] Asterisk monitoring availability
  I'm attempting to implement an acceptable asterisk monitoring 
  program. One method is with a simple program i've written (Adapted from the 
  perl one in the wiki).One problem with this solution is the timeout I 
  expect a packet response. Currently it's set to 1 second, and I still 
  sometimes miss an IAX ping response. Which throws false alarms to my other 
  programs, SMS's, etc. What have others done to monitor asterisks 
  availability?Thanks-- Sig Langehttp://www.signuts.net/ 

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Re: [Asterisk-Users] OT: Hardware Interrupts; Who is it?

[Matthew Boehm]

OK. Perhaps I was not clear. Please read my original post again:

 I've been trying to diagnose why my server has a constant idle time of 90%
 even when nothing is running.

I did not say "90% usage" I said "90% idle". Meaning I have a constant CPU
usage of 10%.

This 10% measurement comes from the "hardware interrupt" (hi) from within
"top":

Cpu(s):  3.8% us,  2.1% sy,  0.0% ni, 85.5% id,  0.2% wa,  8.0% hi,  0.4% si


Even when all other percentages are at 0%, hi remains around 10%. How can I
figure out what is causing all these interrupts?

-Matthew


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RE: [Asterisk-Users] Differ between "private" and "out of area"?

[trixter http://www.0xdecafbad.com]

On Mon, 2005-09-19 at 13:48 -0500, jltaylor wrote:
> "out-of-area" is displayed for calls that originate from LECs that have not
> implemented caller id.

Or for companies that dont share it, this is sometimes the issue for
foreign originated calls.  Caller id is sent via SS7, and some companies
that do have caller id, and do have SS7 for other aspects for some
reason do not transfer it globally.  I have seen this from calls from
the UK to the US for example (but not all such calls).

In America the FCC basically requires that if you have caller id support
you must pass caller id data, so most companies in the US pass caller
id.  The federal government however always seems to pass 000-000-, I
guess to keep it 'private' but not trigger any privacy blockers so the
call goes through.  MCI sales team used to pass 'out of area' for the
same reason.  


-- 
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378


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Re: [Asterisk-Users] Most desireable Linux distribution for Asterisk?

[Michiel van Baak]

On 14:05, Mon 19 Sep 05, Asterisk Lists wrote:
> We have a few Asterisk boxes running under Fedora C3 with no issues. 
> Before we roll into full production mode, we're wondering if the gurus 
> prefer any particular Linux distribution?
> 
> There are pre-packaged Asterisk solutions out there based on WBEL, 
> CentOS and a few others, so there doesn't seem to be a "de-facto best 
> choice", but I figured it was worth asking.
> 

Hi,

There is no real answer to your question.
just use one you're most familiar with.

Personally I wouldn't survive without Debian ;)
-- 
Michiel van Baak
http://michiel.vanbaak.info
[EMAIL PROTECTED]
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D

"Why is it drug addicts and computer afficionados are both called users?"

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Re: [Asterisk-Users] IAX dialplan problem?

[Thameem Ansari]

You are doing correct. But you have to explain what you want to do?
As per your second configuration, if you dial 1 then it will
ringPost exactly what you are trying to accomplish?

-Thameem
On 9/19/05, John Crowhurst <[EMAIL PROTECTED]> wrote:
Hello, I'm a newbie to the asterisk system.

I'm trying to configure a dialplan so that when I use my IAXy it will
prompt me with an IVR and then send me off to different things like
dial and voicemail from that.

I've tried various combinations but I can't seem to get it to work properly. Here is an example:

[default]
exten => s,1,Answer
exten => s,2,Ringing

It gives me a dialtone and waits for an input, but if I do:

[default]
exten => 1,1,Answer
exten => 1,2,Ringing

And then dial 1 it rings...

Am I doing something wrong? Any suggestions or pointers gratefully received.

Thanks in advance,

John




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Re: [Asterisk-Users] kill a .call file

[trixter http://www.0xdecafbad.com]

On Mon, 2005-09-19 at 13:43 -0500, jltaylor wrote:
> Any means of killing a .call file that is in progress?
> 

You mean once the call has begun?  You prolly want to hangup the
call ...

asterisk -rx "soft hangup "

Or is there something else that you wanted?


-- 
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378


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[Asterisk-Users] Asterisk monitoring availability

[Sig Lange]

I'm attempting to implement an acceptable asterisk monitoring program.
One method is with a simple program i've written (Adapted from the perl
one in the wiki).

One problem with this solution is the timeout I expect a packet
response. Currently it's set to 1 second, and I still sometimes miss an
IAX ping response. Which throws false alarms to my other programs,
SMS's, etc. What have others done to monitor asterisks availability?

Thanks
-- Sig Langehttp://www.signuts.net/
#include 
#include 
#include 
#include 
#include 
#include 


int udpping(char *host, int port, int timeout_sec, int timeout_usec) {
	int rc = 0;
	int r, s;
	char  msg[1024];
   struct hostent *hp;
	struct sockaddr_in destaddr;

	fd_set rfds;
   struct timeval timeout;
   struct hostent *hostname;

	s = socket(PF_INET, SOCK_DGRAM, 17);

	memset(&destaddr, 0, sizeof(destaddr));

	destaddr.sin_port = htons(port);

   // destaddr.sin_addr.s_addr = inet_addr(host);
   //
  
   hostname = gethostbyname(host);

   destaddr.sin_addr.s_addr = ((struct in_addr *)(hostname->h_addr))->s_addr;

	// 80 00 00 0000 00 00 00 00 00 06 1e
	strcpy(msg, "\x80\x00\x00\x00\x00\x00\x00\x00\x00\x00\x06\x1e");

	timeout.tv_sec = timeout_sec;
	timeout.tv_usec = timeout_usec;

	FD_ZERO(&rfds);
	FD_SET(s, &rfds);

	sendto(s, msg, 12, MSG_NOSIGNAL, (struct sockaddr *) &destaddr, sizeof(destaddr) );

	r = select(s + 1, &rfds, NULL, NULL, &timeout);

	if (r) {
		r = recv(s, msg, 1024, 0);
		if (r) rc = 1;
	}

	return rc;
}
int main(int argc, char **argv) {
	int r;

   if (argc < 2) {
  printf("Usage: %s \n", argv[0]);
  exit(100);
   }

	r = udpping(argv[1], 4569, 1, 0);

   r = ! r;

	exit(r);
}
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RE: [Asterisk-Users] Differ between "private" and "out of area"?

[jltaylor]

"out-of-area" is displayed for calls that originate from LECs that have not
implemented caller id.

James Taylor
MetroTel
3505 Summerhill Road
Suite 11
Texarkana, Tx  75503
903-793-1956


> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Rich
> Adamson
> Sent: Monday, September 19, 2005 12:19 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Differ between "private" and "out of
> area"?
>
>
> I don't believe you can trust the keywords that may or maynot be in the
> calleridname. The telco folks will frequently honor anything that a
> company wants inserted as a name (assuming a reasonable request).
>
> So, even if you get the correct logic in place for asterisk code,
> the end result is most likely not going to give you what you want.
>
> I know a telco tech that will change the libd database to say
> "God Calling", place a call to a buddy, then change it back to the
> original string after the call. Also, some itsp's allow you to change
> that string to anything reasonable.
>
> 
>
> > Yes, I know that, but, how to distinguish between them at incoming call?
> >
> >
> > - Original Message -
> >
> > A private call is a call that someone has specifically blocked.   An
> > "out of area" or "unknown" call is simply a call that the caller-id
> > did not come through on correctly, for some reason.
> >
> > On 9/18/05, Goran Dj. <[EMAIL PROTECTED]> wrote:
> > > Is there any method to make difference between Hidden ("Private") and
> > > unknown ("Out of area") incoming calls on ZAP/x101p? I want to block
> > any
> > > hidden call, and to allow unknow calls, but ZAP channel (X101P) always
> > > delivering empty CALLERID=""<> in both cases.
>
>
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Re: [Asterisk-Users] SIP audio port usage

[Rich Adamson]

> I know that SIP is using port 5060 for session initiation, but which port
> does it use for audio ? is it dynamically assigned ?

Its dynamically assigned on a per-call basis. 

Asterisk assigns the port based on contents of rtp.conf.

Remote sip phones assign port numbers based on whatever the manufacturer
happened to choose (no industry standard). E.g., Cisco uses 32,768 to
something around 40,000, while xlite uses something in the area of 8,000.
The various manufacturers are not consistent at all.



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[Asterisk-Users] kill a .call file

[jltaylor]

Any means of killing a .call file that is in progress?

James Taylor
MetroTel
3505 Summerhill Road
Suite 11
Texarkana, Tx  75503
903-793-1956

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[Asterisk-Users] IAX dialplan problem?

[John Crowhurst]

Hello, I'm a newbie to the asterisk system.

I'm trying to configure a dialplan so that when I use my IAXy it will
prompt me with an IVR and then send me off to different things like
dial and voicemail from that.

I've tried various combinations but I can't seem to get it to work properly. Here is an example:

[default]
exten => s,1,Answer
exten => s,2,Ringing

It gives me a dialtone and waits for an input, but if I do:

[default]
exten => 1,1,Answer
exten => 1,2,Ringing

And then dial 1 it rings...

Am I doing something wrong? Any suggestions or pointers gratefully received.

Thanks in advance,

John



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[Asterisk-Users] Re: ztdummy configuration help

[Stefan Tichy]

On Mon, Sep 19, 2005 at 01:53:16PM -0400, kurt x wrote:
> Upon setting up and configuring the my extension.conf, meetme.conf and
> following the instruction outlined at this web page:
> http://www.voip-info.org/wiki-Asterisk+timer+ztdummy
> 
> I get the following errors when calling the meetme number.
> 
> Executing Wait("SIP/216.53.118.2-f41196e0", "1") in new stack
> -- Executing MeetMe("SIP/216.53.118.2-f41196e0", "|sicp") in new stack
> -- Playing 'conf-getconfno' (language 'en')
>   == Parsing '/etc/asterisk/meetme.conf': Found
> Sep 19 13:51:22 WARNING[14066]: chan_zap.c:757 zt_open: Unable to open
> '/dev/zap/pseudo': No such file or directory

The device /dev/zap/pseudo (and probably ctl, timer, channel) is
missing. Asterisk process needs write permissions for this device.

If you are using kernel 2.6 and udev check README.udev (as described
in the wiki mentioned in your mail) 


-- 
Stefan Tichy   <[EMAIL PROTECTED]>
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[Asterisk-Users] Looking for firmware for Cisco 12sp+ and 30VIP

[Stern, Craig]

I have been looking 
for the firmware for the 12sp+ and 30VIP and have been unable to find it. Any 
help in locating would be much appreciated.
 
Thanks===This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the sender immediately. This message may contain confidential information and is intended only for the individual named. If you are not the named addressee you should not disseminate, distribute or copy this e-mail.
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[Asterisk-Users] Re: SIP audio port usage

[Stefan Tichy]

On Mon, Sep 19, 2005 at 12:22:35PM -0400, Adrien Laurent wrote:
> I know that SIP is using port 5060 for session initiation, but which port
> does it use for audio ? is it dynamically assigned ?

RTP protocol is used for audio. Port range is defined in /etc/asterisk/rtp.conf


-- 
Stefan Tichy   <[EMAIL PROTECTED]>
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[Asterisk-Users] Most desireable Linux distribution for Asterisk?

[Asterisk Lists]

We have a few Asterisk boxes running under Fedora C3 with no issues. 
Before we roll into full production mode, we're wondering if the gurus 
prefer any particular Linux distribution?


There are pre-packaged Asterisk solutions out there based on WBEL, 
CentOS and a few others, so there doesn't seem to be a "de-facto best 
choice", but I figured it was worth asking.


- D
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[Asterisk-Users] HooDaHek Version 0.5 Release

[Nathan E. Pralle]

HooDaHek 0.5 has been released.

NEW AS OF VERSION 0.5
 
- Changed the format of the incoming call notification to be on one line, not 
two. 
- Changed how the script sleeps between call notifications -- it now goes and 
outputs the line to everyone in the list, pauses, and then looks for a second 
line. Much better than before, where it was sleeping after every line. 
- Changed the bot to use Unix::Syslog to log messages about its activities.

Information and download link here:
   http://www.nathanpralle.com/software/hoodahek.html

Enjoy.

Nathan

-- 

Interesting things abide:
http://www.nathanpralle.com

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[Asterisk-Users] ztdummy configuration help

[kurt x]

Upon setting up and configuring the my extension.conf, meetme.conf and
following the instruction outlined at this web page:
http://www.voip-info.org/wiki-Asterisk+timer+ztdummy

I get the following errors when calling the meetme number.

Executing Wait("SIP/216.53.118.2-f41196e0", "1") in new stack
-- Executing MeetMe("SIP/216.53.118.2-f41196e0", "|sicp") in new stack
-- Playing 'conf-getconfno' (language 'en')
  == Parsing '/etc/asterisk/meetme.conf': Found
Sep 19 13:51:22 WARNING[14066]: chan_zap.c:757 zt_open: Unable to open
'/dev/zap/pseudo': No such file or directory
Sep 19 13:51:22 ERROR[14066]: chan_zap.c:6687 chandup: Unable to dup
channel: No such file or directory
Sep 19 13:51:22 WARNING[14066]: app_meetme.c:227 build_conf: Unable to
open pseudo channel - trying device
Sep 19 13:51:22 WARNING[14066]: app_meetme.c:230 build_conf: Unable to
open pseudo device
-- Playing 'conf-invalid' (language 'en')

Sep 19 13:51:23 WARNING[14066]: file.c:554 ast_readaudio_callback:
Failed to write frame
-- Playing 'conf-getconfno' (language 'en')

Any help is greatly appreciated.

Kurt
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Re: RE : [Asterisk-Users] Asterisk realtime beta

[Matthew Boehm]

That is not a limitation of the Asterisk RealTime Architecture. That is a
limitation of libiodbc.

-Matthew


> From: Olivier Taylor <[EMAIL PROTECTED]>
> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
> Date: Mon, 19 Sep 2005 18:09:38 +0200
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> 
> Subject: RE : [Asterisk-Users] Asterisk realtime beta
> 
> The limitation is that it doesn't work on freebsd, probably due to
> libiodbc...
> That's a limitation, isn't it?
> 
> Olivier
> 
> -Message d'origine-
> De : [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] De la part de Matthew Boehm
> Envoyé : lundi 19 septembre 2005 16:42
> À : Asterisk Users
> Objet : Re: [Asterisk-Users] Asterisk realtime beta
> 
> 
> So, you admit that you can do what you want using RealTime Static, but you
> are just unwilling to do so. So, how is that a limitation if you 'can' do
> it?
> 
> -Matthew
> 
>> From: Urban <[EMAIL PROTECTED]>
>> Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
>> 
>> Date: Mon, 19 Sep 2005 11:03:09 +0200
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> 
>> Subject: Re: [Asterisk-Users] Asterisk realtime beta
>> 
>> Matthew Boehm wrote:
>> 
 I currently not use it due to some limitations in * realtime .

 
>>> 
>>>Such as?
>>> 
>>> -Matthew
>>> 
>>> 
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>>>  
>>> 
>> My current configuration uses a lot of include statements to split up
>> the context's such as security contexts included per extension (allow
>> national, internation calls etc). Since realtime does not have this
>> type of feature (if you not using static) I decided it was to much
>> work to redeisgn the dialplan at the moment.
>> 
>> /urban
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> 
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Re: [Asterisk-Users] realtime audio for asterisk using jack

[Esben Stien]

Matt Riddell <[EMAIL PROTECTED]> writes:

> Is it possible that JACK creates an emulated alsa/oss layer for non
> JACK connections?

I can now confirm that jackifying iaxcomm works. The kfusd and
oss2jack now works on linux-2.6.13.

http://fort.xdas.com/~kor/oss2jack/

I still have latency problems as described in my other threads on the
iaxclient list. It has gone down to about half (300ms) since last
report.

I will do an exact measurement tonight most probably and report back
(on the iaxclient mailinglist).

So for the search engine:

Running voip applications with jack (jackd) (oss2jack) works with
linux 2.6.13. You can also run kphone and asterisk, though some noise
problems still exist. In the case of asterisk, it's very much in need
to get this to work so that you can plug voip into jack and process
the signal before it is sent and before it reach the speakers.

It would of course be great if a real jack driver were implemented in
asterisk, but few developers have shown interest.

To get help with these issues, please try #lad and the
linux-audio-user mailinglist.

There is also a oss2jack mailinglist being set up.

We finally made it;). 

-- 
Esben Stien is [EMAIL PROTECTED] s  a 
 http://www. s tn m
  irc://irc.  b  -  i  .   e/%23contact
  [sip|iax]:   e e 
   jid:b0ef@n n
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[Asterisk-Users] Fwd: Asterisk in Spanish

[Sebastian Milioto]

Great info!!!. Thank you all guys.

Regards,

Sebastian 

-- Forwarded message --
From: Sergio Serrano <[EMAIL PROTECTED]>
Date: Mon, 19 Sep 2005 18:38:55 +0200
Subject: RE: [Asterisk-Users] Asterisk in Spanish
To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial
Discussion 

 

Try in www.asterisk-es.org

-Mensaje original-
De: Sebastian Milioto [mailto:[EMAIL PROTECTED] 
Enviado el: lunes, 19 de septiembre de 2005 15:08
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: [Asterisk-Users] Asterisk in Spanish

Hi all,

I've been installing [EMAIL PROTECTED] and (of course) all the "answering
machine" (I don't sure that's the right word in english, "preatendedora" in
spanish) speech is in enlgish languaje.
Is there anyway to download all those .gsm files speaked in spanish?
Or may be another site which contain this kind of stuff (.wav, .gsm files
for answering machines in spanish)?


Thank you very much,

Regards,

Sebastian Milioto
Telecommunications Engineer
IM: [EMAIL PROTECTED]
e-mail: [EMAIL PROTECTED]
Mobile: 549 3571 543658
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[Asterisk-Users] i4l ring indication problem, again...

[Omadon]

I can't find solution anywhere. I googled and find people with the same
problem but there was no answers on how to fix this.

I have W6692 based PCI cards that uses hisax driver (card type=36).
Card is working fine under asterisk with i4l modem driver for incoming
calls. If I want to dial out using some sip phone I don't get ring
indication. Phone is ringing and I hear only silence until someone
answers the phone. Using r in Dial command kind of fixes this problem
but not completely. Example: I call someone on mobile phone and he is busy,
I want get busy right away, first i will hear ringing (until calls reach
mobile phone) and then i get busy. This confuses people allot.

So my question is how to get ring indication on i4l. I tried different
i4l cards, some of them where usb and all have the same ring indication
problem.

And yes I have /etc/asterisk/indications.conf, I tried to change countries
but it didn't help.

Please don't tell me to try CAPI or bristuff, I know they work but I want
to make this work (this small usb isdn is very cool)

Is there anybody that has a working configuration for i4l that doesn't
have ring indication problem (I'm using 2.4.29 kernel).

Thanks

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Re: [Asterisk-Users] Differ between "private" and "out of area"?

[Rich Adamson]

I don't believe you can trust the keywords that may or maynot be in the
calleridname. The telco folks will frequently honor anything that a
company wants inserted as a name (assuming a reasonable request).

So, even if you get the correct logic in place for asterisk code, 
the end result is most likely not going to give you what you want.

I know a telco tech that will change the libd database to say
"God Calling", place a call to a buddy, then change it back to the
original string after the call. Also, some itsp's allow you to change
that string to anything reasonable.



> Yes, I know that, but, how to distinguish between them at incoming call?
> 
> 
> - Original Message - 
> 
> A private call is a call that someone has specifically blocked.   An
> "out of area" or "unknown" call is simply a call that the caller-id
> did not come through on correctly, for some reason.
> 
> On 9/18/05, Goran Dj. <[EMAIL PROTECTED]> wrote:
> > Is there any method to make difference between Hidden ("Private") and
> > unknown ("Out of area") incoming calls on ZAP/x101p? I want to block
> any
> > hidden call, and to allow unknow calls, but ZAP channel (X101P) always
> > delivering empty CALLERID=""<> in both cases.


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[Asterisk-Users] SIP audio port usage

[Adrien Laurent]

Hi,

I know that SIP is using port 5060 for session initiation, but which port
does it use for audio ? is it dynamically assigned ?

Thanks,

Adrien

--
Adrien Laurent - CIO
www.modulis.ca
514-284-2020 ext 202
[EMAIL PROTECTED]


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