date:20051107

Wasn't aware of it, but if quality is good, it makes sense since all  
I'm archiving is speech.


Will evaluate further.

Thanks,
Waldo

On Nov 7, 2005, at 7:14 PM, Mark Edwards wrote:


I would recommend vorbis speex for this.
You can get windows drivers to read speex files directly.

Vorbis are the same bunch that develops ogg.

Ogg and mp3 are more suited to music rather than speech.
Speex is a much better fit for speech archiving.

Mark


-Original Message-
From: BJ Weschke [mailto:[EMAIL PROTECTED]
Sent: Tuesday, 8 November 2005 5:52 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] MP3 or OGG

 You're probably not going to be violating any patent protections by
using OGG instead of MP3. As far as compression goes, I've found the
difference between the two of them to be negligible. I've always used
OGG when possible to stay "IP safe".

On 11/7/05, Waldo Rubinstein <[EMAIL PROTECTED]> wrote:

I'm trying to archive out call recordings and would appreciate some
feedback as to which audio compression is more recommended MP3 or
OGG. In the past, I've use lame to convert to MP3, but I noticed the
audio volume drops significantly. Is it just a setting on the command
line of lame or is OGG better? Which achieves higher compression
rates while maintaining call quality?

Thanks,
Waldo

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[Asterisk-Users] several beginner questions

2005-11-07 Thread Dane Reugger


Actually 1 beginner w/ multiple questions...

I'm getting ready to make my first jump into VoIP and the Asterisk PBX - 
Katrina has forced my hand much earlier than expected. My phone and ISP 
(Eatel) is leaving New Orleans so I've got just a couple weeks to get 
this done.


I read some good reviews on QuantumVoice - they don't seem like the 
cheapest and have answered every email I shot at them - any opinions?


I plan on using this with my 4 line (analog) Panasonic wireless base 
station w/ 5 wireless phones to start off with - Quantumvoice says I can 
have 5 concurrent calls but I won't need more than 3 or 4 for the 
foreseeable future - I will experiment with IP phones later -  any 
hardware recommendations?


Debian is my Linux flavor of choice - any good walk throughs, etc..?

The only way I'm gonna get a feel for this is if I start doing it but I 
don't want to paint myself in the corner or over spend - in fact I would 
like to spend as little as possible at this time but its a business 
system and I want it to work well - any /all advice appreciated.


TIA,
Dane Reugger
Crescent City Technologies
New Orleans, LA


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[Asterisk-Users]how to send fax using Spandsp

2005-11-07 Thread Andy Kuo

Hi,
 
I've been trying to get fax going for the last few days.
I can receive fax with Spandsp 0.0.2pre19 on Asterisk 1.0.9 with rxfax, but when I tried sending the received fax file to a fax machine, I either get "line error" or just a blank page.
 

Is anyone using Spandsp to send fax to fax machines on PSTN?
 
I've run out of things to try now, and I'd really appreciate if anyone can share some ideas/experiences here.
 
 
Thank you.
AK
 
 
 
 
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RE: [Asterisk-Users] MP3 or OGG

2005-11-07 Thread Mark Edwards

I would recommend vorbis speex for this. 
You can get windows drivers to read speex files directly.

Vorbis are the same bunch that develops ogg.

Ogg and mp3 are more suited to music rather than speech.
Speex is a much better fit for speech archiving.

Mark


-Original Message-
From: BJ Weschke [mailto:[EMAIL PROTECTED] 
Sent: Tuesday, 8 November 2005 5:52 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] MP3 or OGG

 You're probably not going to be violating any patent protections by
using OGG instead of MP3. As far as compression goes, I've found the
difference between the two of them to be negligible. I've always used
OGG when possible to stay "IP safe".

On 11/7/05, Waldo Rubinstein <[EMAIL PROTECTED]> wrote:
> I'm trying to archive out call recordings and would appreciate some
> feedback as to which audio compression is more recommended MP3 or
> OGG. In the past, I've use lame to convert to MP3, but I noticed the
> audio volume drops significantly. Is it just a setting on the command
> line of lame or is OGG better? Which achieves higher compression
> rates while maintaining call quality?
>
> Thanks,
> Waldo
>
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Re: [Asterisk-Users] libmfcr2 - spandsp.h: present but cannot be compiled

2005-11-07 Thread Steve Underwood


Jesus Mogollon wrote:


Hi all

When I try compiling libmfcr2 I get:

spandsp.h: present but cannot be compiled

Any ideas?


Either:

a) Ignore that message, and carry on. It works anyway.

b) Use a newer version of spandsp (pre21b eight now) which should no 
longer be generating that message.


Steve

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Re: [Asterisk-Users] Missing audio from Zaptel channels - SOLVED!?

2005-11-07 Thread Rod Bacon


For those who are interested, the problem appears to NOT exist in 1.2Beta2.

==
Rod Bacon
Empowered Communications
Ground Floor, 102 York St. South Melbourne
Victoria, Australia. 3205
Phone: +613 99401600Fax: +613 99401650
FWD: 512237   ICQ: 5662270
==


Rod Bacon wrote:
I have cross-posted this all over the place, and sent a copy directly to 
digium
support, in the hope of getting to the bottom of a problem that has me 
pulling

my hair out.

I currently have 2 production PSTN gateway servers, running asterisk 
1.2beta and
TE406P cards (upgraded 405 cards, with hardware echo cancelers that we 
recently
purchased on recommendation). We went to the beta version after 
installing the

cancelers, as 1.0.9 kept segfaulting with the cancelers installed. Our PRIs
terminate on a DMS100, at the same premises where our servers are 
co-located.


Also in my farm, I have a dedicated IVR server, a VOIP gateway 
(SIP/IAX/H.323)
and clustered MySQL servers running as FastAGI servers, to remove 
processor load

from the PSTN servers. All servers are connected via gigabit Ethernet, and
use IAX trunking for inter-server communications.

I have been through _everything_ possible to be sure that I don't have any
zaptel timing/irq problems (framebuffer, apic, acpi, smp irq affinity, irq
latency, etc. etc) and have good zttest results with no frame slips, 
pops or clicks.


After my PSTN gateway servers have been running for a few hours, I 
notice that
some missing audio creeps into the start of each call (makes no 
difference if

the call is ZAP-ZAP native bridge or ZAP-IAX). At best, you miss the first
syllable of the first word. At worst, you can miss the first 3 or 4 
seconds of
audio. Further investigation shows that asterisk is lagging after the 
second leg
of the call is answered (i.e. the time taken to bridge the channels gets 
longer). If the resultant call is a Zaptel native bridge, then the 
remaining audio is fine. If the resultant call is not zaptel natively 
bridged (eg. call is routed via another server, or asterisk remains in 
the media stream for another reason) then significant delay exists from 
one end of the call to another (simply put, asterisk seems to slow down).


If I restart asterisk (even without removing and reloading zaptel 
drivers), calls are OK again for a period (typically around 12 hours). A 
workaround is to simply to install a cron job that periodically restarts 
asterisk when it's idle,  but this is a less than ideal solution from my 
perspective.


Something is definitely changing over time. A memory leak? Runaway 
process? I
really need help in trying to troubleshoot this, as I've run completely 
out of

both patience and ideas.



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[Asterisk-Users] Very basic switching application -- bounty?

2005-11-07 Thread Eric Lyons


Eric Lyons wrote:


The basic function is to take an incoming DNIS/exten on one port, look
it up in the db, then dial out to another number on another port.


This is just basic dialplan work... why you would need a custom application?Hi, Kevin.  Yes, it *is* the most basic of dialplan 
configuration, but I don't want to reloadthe whole dialplan whenever the switch mapping changes or when adding/deleting a 
map.Natch, I looked into RealTime for extensions.conf, but this seems to generate at least nineidentical queries even if there's 
only one priority in the context.I think I might hack up app_addon_sql_mysql.c to do what I want -- but I half expected someoneto 
already have done it?Eric. 


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[Asterisk-Users] SIP domain support for authentication and virtual hosting

2005-11-07 Thread harry gaillac

Hello,

Where may i find documentation about SIP domain
support and dnsmgr.conf ,

Harry






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Re: [Asterisk-Users] Asterisk 1.2beta2 and UIP200

First off, if they are on the same network without any nat, then it is
not needed at all. Since this works well with pre 1.2b2 I would say
you should open up a but on the bug tracker at:
bugs.digium.com.
I did not yet update to 1.2bx so I have no way of confirming this.
Thank You.

On 11/7/05, Waldo Rubinstein <[EMAIL PROTECTED]> wrote:
> Ok.
>
> The keepalives work for other phones, but not the UIP200. I have a
> bunch of X-Lites, X-Pros, SPA-841s, and UIP200s. It works fine in all
> but the UIP200 (only in 1.2b2).
>
> As far as your questions:
>
> 1) They are on the same network and same netmask
> 2) They are not natted.
>
> Let me know what you find.
>
> Thanks,
> Waldo
>
> On Nov 7, 2005, at 4:58 PM, C F wrote:
>
> > If this is the case. then we now know what the problem is. The
> > keepalives from asterisk to the phones were not working in 1.2b2. The
> > question now is why?
> > Please work with this so that we can troubleshoot this to see if it's
> > a bug with 1.2b2 or not.
> > 1. Is the UIP200 on the same subnet as asterisk?
> > 2. if not, is the UIP200 or asterisk natted?
> >
> > In the meantime I will try to see on my 1.0.9 install if it works or
> > not with UIP200 phones.
> > Thank You.
> >
> > On 11/7/05, Waldo Rubinstein <[EMAIL PROTECTED]> wrote:
> >> I do have qualify=yes pretty much in all my sip entries. I just
> >> changed all the entries where I have a UIP200 to qualify=no and now
> >> they all work. The funny thing is that it worked with qualify=yes in
> >> 1.0.9 and 1.2b1
> >>
> >> Thanks,
> >> Waldo
> >>
> >> On Nov 7, 2005, at 1:29 PM, C F wrote:
> >>
> >>> I guess that somewhere in your settings you have a qualify on, or
> >>> that
> >>> 1.2 has it on by default. Do the following:
> >>> cd /etc/asterisk
> >>> grep ".*qualify.*" ./*
> >>> and see the output, if the only line that has qualify is that
> >>> qualify=no, then this looks like a bug to me. Please report back.
> >>>
> >>> On 11/7/05, Waldo Rubinstein <[EMAIL PROTECTED]> wrote:
>  Ok. That fixed it, but why? It works just fine in 1.0.9 and 1.2b1.
>  Very strange.
> 
>  Anyway, thanks.
> 
>  - Waldo
> 
>  On Nov 7, 2005, at 10:57 AM, C F wrote:
> 
> > The unreachable is the problem. Try adding a qualify=no to that
> > sip
> > entry.
> >
> > On 11/7/05, Waldo Rubinstein <[EMAIL PROTECTED]> wrote:
> >> Additionally:
> >>
> >> *CLI> sip show peer 100074
> >>
> >>   * Name   : 100074
> >>   Secret   : 
> >>   MD5Secret: 
> >>   Context  : qa
> >>   Subscr.Cont. : 
> >>   Language : en
> >>   AMA flags: Unknown
> >>   CallingPres  : Presentation Allowed, Not Screened
> >>   Callgroup:
> >>   Pickupgroup  :
> >>   Mailbox  : [EMAIL PROTECTED]
> >>   VM Extension : asterisk
> >>   LastMsgsSent : 0
> >>   Call limit   : 0
> >>   Dynamic  : Yes
> >>   Callerid : "Waldo Rubinstein" <211>
> >>   Expire   : 11077
> >>   Insecure : no
> >>   Nat  : No
> >>   ACL  : No
> >>   CanReinvite  : No
> >>   PromiscRedir : No
> >>   User=Phone   : No
> >>   Trust RPID   : No
> >>   Send RPID: No
> >>   DTMFmode : rfc2833
> >>   LastMsg  : 0
> >>   ToHost   :
> >>   Addr->IP : 10.0.10.236 Port 5060
> >>   Defaddr->IP  : 0.0.0.0 Port 5060
> >>   Def. Username: 100074
> >>   SIP Options  : (none)
> >>   Codecs   : 0x6 (gsm|ulaw)
> >>   Codec Order  : (ulaw,gsm)
> >>   Status   : UNREACHABLE
> >>   Useragent: Uniden SIP Phone p2 Ver BS4.63
> >>   Reg. Contact : sip:[EMAIL PROTECTED]:5060
> >>
> >> Thanks,
> >> Waldo
> >>
> >> On Nov 6, 2005, at 11:11 PM, C F wrote:
> >>
> >>> can you post the sip.conf for that uip200?
> >>>
> >>> On 11/6/05, Waldo Rubinstein <[EMAIL PROTECTED]> wrote:
>  When I dial the extension, I get this:
> 
>   -- Executing Dial("IAX2/gateway0-16386", "SIP/100074|20")
>  in new
>  stack
> == Everyone is busy/congested at this time (1:0/0/1)
> 
> 
>  When I do a sip show peer 100074, everything it shows
>  matches the
>  results of executing the same sip show peer on * 1.0.9 and
>  1.2b1,
>  except:
> 
> Status   : UNREACHABLE
> 
>  However, I can make any type of calls from them phone. I can
>  ping the
>  phone from the * server. It's just that * 1.2b2 can't reach
>  it, for
>  some reason.
> 
>  Thanks,
>  Waldo
> 
>  On Nov 6, 2005, at 1:37 PM, C F wrote:
> 
> > Whats the exact CLI output you are getting when calling that
> > extension?
> >
> > On 11/6/05, Waldo Rubinstein <[EMAIL PROTECTED]> wrote:
> >> Nope. It

Re: [Asterisk-Users] zaphfc not generally compatible with kernels >= 2.6.13

2005-11-07 Thread Gerald Dachs

On Mon, 7 Nov 2005 23:06:24 +0100
Gerald Dachs <[EMAIL PROTECTED]> wrote:

> Hi,
> 
> I am very new to asterisk so forgive me if I tell something stupid.

It has happend, my post was stupid

> I am investigating currently a problem with zaphfc. I get only very few 
> interrupts,
> they don't get lost, the interrupt count increases only very slowly.
> 
> I really don't know where to look for the problem, so I looked here and there 
> and found
> the following line in zaphfc.c from bristuff-0.2.0-RC8o:
> schedule_timeout((30 * HZ) / 1000); // wait 30 ms

schedule_timeout is a kernel call that gets ticks as arg and not ms.

Gerald
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Re: [Asterisk-Users] Asterisk 1.2beta2 and UIP200


Ok.

The keepalives work for other phones, but not the UIP200. I have a  
bunch of X-Lites, X-Pros, SPA-841s, and UIP200s. It works fine in all  
but the UIP200 (only in 1.2b2).


As far as your questions:

1) They are on the same network and same netmask
2) They are not natted.

Let me know what you find.

Thanks,
Waldo

On Nov 7, 2005, at 4:58 PM, C F wrote:


If this is the case. then we now know what the problem is. The
keepalives from asterisk to the phones were not working in 1.2b2. The
question now is why?
Please work with this so that we can troubleshoot this to see if it's
a bug with 1.2b2 or not.
1. Is the UIP200 on the same subnet as asterisk?
2. if not, is the UIP200 or asterisk natted?

In the meantime I will try to see on my 1.0.9 install if it works or
not with UIP200 phones.
Thank You.

On 11/7/05, Waldo Rubinstein <[EMAIL PROTECTED]> wrote:

I do have qualify=yes pretty much in all my sip entries. I just
changed all the entries where I have a UIP200 to qualify=no and now
they all work. The funny thing is that it worked with qualify=yes in
1.0.9 and 1.2b1

Thanks,
Waldo

On Nov 7, 2005, at 1:29 PM, C F wrote:

I guess that somewhere in your settings you have a qualify on, or  
that

1.2 has it on by default. Do the following:
cd /etc/asterisk
grep ".*qualify.*" ./*
and see the output, if the only line that has qualify is that
qualify=no, then this looks like a bug to me. Please report back.

On 11/7/05, Waldo Rubinstein <[EMAIL PROTECTED]> wrote:

Ok. That fixed it, but why? It works just fine in 1.0.9 and 1.2b1.
Very strange.

Anyway, thanks.

- Waldo

On Nov 7, 2005, at 10:57 AM, C F wrote:

The unreachable is the problem. Try adding a qualify=no to that  
sip

entry.

On 11/7/05, Waldo Rubinstein <[EMAIL PROTECTED]> wrote:

Additionally:

*CLI> sip show peer 100074

  * Name   : 100074
  Secret   : 
  MD5Secret: 
  Context  : qa
  Subscr.Cont. : 
  Language : en
  AMA flags: Unknown
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup:
  Pickupgroup  :
  Mailbox  : [EMAIL PROTECTED]
  VM Extension : asterisk
  LastMsgsSent : 0
  Call limit   : 0
  Dynamic  : Yes
  Callerid : "Waldo Rubinstein" <211>
  Expire   : 11077
  Insecure : no
  Nat  : No
  ACL  : No
  CanReinvite  : No
  PromiscRedir : No
  User=Phone   : No
  Trust RPID   : No
  Send RPID: No
  DTMFmode : rfc2833
  LastMsg  : 0
  ToHost   :
  Addr->IP : 10.0.10.236 Port 5060
  Defaddr->IP  : 0.0.0.0 Port 5060
  Def. Username: 100074
  SIP Options  : (none)
  Codecs   : 0x6 (gsm|ulaw)
  Codec Order  : (ulaw,gsm)
  Status   : UNREACHABLE
  Useragent: Uniden SIP Phone p2 Ver BS4.63
  Reg. Contact : sip:[EMAIL PROTECTED]:5060

Thanks,
Waldo

On Nov 6, 2005, at 11:11 PM, C F wrote:


can you post the sip.conf for that uip200?

On 11/6/05, Waldo Rubinstein <[EMAIL PROTECTED]> wrote:

When I dial the extension, I get this:

 -- Executing Dial("IAX2/gateway0-16386", "SIP/100074|20")
in new
stack
   == Everyone is busy/congested at this time (1:0/0/1)


When I do a sip show peer 100074, everything it shows  
matches the
results of executing the same sip show peer on * 1.0.9 and  
1.2b1,

except:

   Status   : UNREACHABLE

However, I can make any type of calls from them phone. I can
ping the
phone from the * server. It's just that * 1.2b2 can't reach
it, for
some reason.

Thanks,
Waldo

On Nov 6, 2005, at 1:37 PM, C F wrote:


Whats the exact CLI output you are getting when calling that
extension?

On 11/6/05, Waldo Rubinstein <[EMAIL PROTECTED]> wrote:

Nope. It isn't active. I even factory reseted the phone but
still the
same. One more piece of information: it works just fine in
1.2b1. I
beginning to think it could be a bug in 1.2b2.

Any other ideas/suggestions?

Thanks,
Waldo

On Nov 5, 2005, at 9:10 PM, C F wrote:


You sure that the DND (Do Not Disturb) button is not active
on the
UIP200?

On 11/4/05, Waldo Rubinstein <[EMAIL PROTECTED]> wrote:

I am running * 1.2b2 with some UIP200 phones and a bunch of
X-Pro
phones.

All phones register fine with * and I can place outbound
calls
with
no problem.

I can call from the X-Pro to any other X-Pro. I can call  
from

UIP200
to any other X-Pro. However, the UIP200 cannot receive  
calls.

Every
time I call the UIP200, the CLI says Everyone is Busy/
Congested and
sends the call to voicemail.

Everything is in the same network. I have in sip.conf
localnet=10.0.10.0/24

and in each UIP200 sip profile
nat=never

What's wrong?

I have the same configuration in * 1.0.9 and it works just
fine.
Could the SIP protocol be broken in 1.2b2?

Thanks,
Waldo

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_

[Asterisk-Users] sill looking for a provider

2005-11-07 Thread info

OOPPS!  Looks like someone just broke voipjet's tos

gw at adcomcorp.com gw at adcomcorp.com wrote on
Sat Nov 5 11:36:46 CST 2005 



 I tend to agree with you, my experience with Teliax has been decent,
and getting better.  If only I could get to them at under 20ms though,
right now my latency is about 75ms whereas voipjet comes through at
19ms.

Greg
--

https://www.voipjet.com/tos.php
NON-DISCLOSURE: ALL CUSTOMERS USING VOIPJET'S SERVICE ARE SPECIFICALLY 
PROHIBITED FROM DISCLOSING TO OTHERS THAT THEY USE VOIPJET'S SERVICE, THIS 
INCLUDES BUT IS NOT LIMITED TO, END USERS. CUSTOMERS MAY NOT DISCLOSE USE OF OR 
PAYMENTS TO VOIPJET ON PERSONAL, CORPORATE, LEGAL, ACCOUNTING AND OTHER 
DOCUMENTS AND COMMUNICATIONS UNLESS SPECIFICALLY REQUIRED TO DO SO BY LAW


Has anyone else read these TOS'es???  Some are pretty funny.


Thomas Herlihy
Scaletta Moloney Armoring
Chicago, IL USA
708.924.0099
Skype VoIP @ HerlsOne
Free World Dialup 647717
[EMAIL PROTECTED]
www.scaletta.com
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[Asterisk-Users] zaphfc not generally compatible with kernels >= 2.6.13

2005-11-07 Thread Gerald Dachs

Hi,

I am very new to asterisk so forgive me if I tell something stupid.

I am investigating currently a problem with zaphfc. I get only very few 
interrupts,
they don't get lost, the interrupt count increases only very slowly.

I really don't know where to look for the problem, so I looked here and there 
and found
the following line in zaphfc.c from bristuff-0.2.0-RC8o:
schedule_timeout((30 * HZ) / 1000); // wait 30 ms

IIRC the default HZ in 2.6.13 (or was it 2.6.14?) is 250. In our kernel HZ is 
100.
So the wait gets too short, should the driver not check that CONFIG_HZ_1000 is 
set?
I am not sure that this is the reason for my problem, but I build currently a 
new kernel
and will test it.

Gerald 
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Re: [Asterisk-Users] Asterisk 1.2beta2 and UIP200

If this is the case. then we now know what the problem is. The
keepalives from asterisk to the phones were not working in 1.2b2. The
question now is why?
Please work with this so that we can troubleshoot this to see if it's
a bug with 1.2b2 or not.
1. Is the UIP200 on the same subnet as asterisk?
2. if not, is the UIP200 or asterisk natted?

In the meantime I will try to see on my 1.0.9 install if it works or
not with UIP200 phones.
Thank You.

On 11/7/05, Waldo Rubinstein <[EMAIL PROTECTED]> wrote:
> I do have qualify=yes pretty much in all my sip entries. I just
> changed all the entries where I have a UIP200 to qualify=no and now
> they all work. The funny thing is that it worked with qualify=yes in
> 1.0.9 and 1.2b1
>
> Thanks,
> Waldo
>
> On Nov 7, 2005, at 1:29 PM, C F wrote:
>
> > I guess that somewhere in your settings you have a qualify on, or that
> > 1.2 has it on by default. Do the following:
> > cd /etc/asterisk
> > grep ".*qualify.*" ./*
> > and see the output, if the only line that has qualify is that
> > qualify=no, then this looks like a bug to me. Please report back.
> >
> > On 11/7/05, Waldo Rubinstein <[EMAIL PROTECTED]> wrote:
> >> Ok. That fixed it, but why? It works just fine in 1.0.9 and 1.2b1.
> >> Very strange.
> >>
> >> Anyway, thanks.
> >>
> >> - Waldo
> >>
> >> On Nov 7, 2005, at 10:57 AM, C F wrote:
> >>
> >>> The unreachable is the problem. Try adding a qualify=no to that sip
> >>> entry.
> >>>
> >>> On 11/7/05, Waldo Rubinstein <[EMAIL PROTECTED]> wrote:
>  Additionally:
> 
>  *CLI> sip show peer 100074
> 
>    * Name   : 100074
>    Secret   : 
>    MD5Secret: 
>    Context  : qa
>    Subscr.Cont. : 
>    Language : en
>    AMA flags: Unknown
>    CallingPres  : Presentation Allowed, Not Screened
>    Callgroup:
>    Pickupgroup  :
>    Mailbox  : [EMAIL PROTECTED]
>    VM Extension : asterisk
>    LastMsgsSent : 0
>    Call limit   : 0
>    Dynamic  : Yes
>    Callerid : "Waldo Rubinstein" <211>
>    Expire   : 11077
>    Insecure : no
>    Nat  : No
>    ACL  : No
>    CanReinvite  : No
>    PromiscRedir : No
>    User=Phone   : No
>    Trust RPID   : No
>    Send RPID: No
>    DTMFmode : rfc2833
>    LastMsg  : 0
>    ToHost   :
>    Addr->IP : 10.0.10.236 Port 5060
>    Defaddr->IP  : 0.0.0.0 Port 5060
>    Def. Username: 100074
>    SIP Options  : (none)
>    Codecs   : 0x6 (gsm|ulaw)
>    Codec Order  : (ulaw,gsm)
>    Status   : UNREACHABLE
>    Useragent: Uniden SIP Phone p2 Ver BS4.63
>    Reg. Contact : sip:[EMAIL PROTECTED]:5060
> 
>  Thanks,
>  Waldo
> 
>  On Nov 6, 2005, at 11:11 PM, C F wrote:
> 
> > can you post the sip.conf for that uip200?
> >
> > On 11/6/05, Waldo Rubinstein <[EMAIL PROTECTED]> wrote:
> >> When I dial the extension, I get this:
> >>
> >>  -- Executing Dial("IAX2/gateway0-16386", "SIP/100074|20")
> >> in new
> >> stack
> >>== Everyone is busy/congested at this time (1:0/0/1)
> >>
> >>
> >> When I do a sip show peer 100074, everything it shows matches the
> >> results of executing the same sip show peer on * 1.0.9 and 1.2b1,
> >> except:
> >>
> >>Status   : UNREACHABLE
> >>
> >> However, I can make any type of calls from them phone. I can
> >> ping the
> >> phone from the * server. It's just that * 1.2b2 can't reach
> >> it, for
> >> some reason.
> >>
> >> Thanks,
> >> Waldo
> >>
> >> On Nov 6, 2005, at 1:37 PM, C F wrote:
> >>
> >>> Whats the exact CLI output you are getting when calling that
> >>> extension?
> >>>
> >>> On 11/6/05, Waldo Rubinstein <[EMAIL PROTECTED]> wrote:
>  Nope. It isn't active. I even factory reseted the phone but
>  still the
>  same. One more piece of information: it works just fine in
>  1.2b1. I
>  beginning to think it could be a bug in 1.2b2.
> 
>  Any other ideas/suggestions?
> 
>  Thanks,
>  Waldo
> 
>  On Nov 5, 2005, at 9:10 PM, C F wrote:
> 
> > You sure that the DND (Do Not Disturb) button is not active
> > on the
> > UIP200?
> >
> > On 11/4/05, Waldo Rubinstein <[EMAIL PROTECTED]> wrote:
> >> I am running * 1.2b2 with some UIP200 phones and a bunch of
> >> X-Pro
> >> phones.
> >>
> >> All phones register fine with * and I can place outbound
> >> calls
> >> with
> >> no problem.
> >>
> >> I can call from the X-Pro to any other X-Pro. I can call from
> >> UIP200
> >> to any other X-Pro. However, the UIP200 cannot receive calls.
> >> Every
> >>

Re: [Asterisk-Users] Problems with DTMF on Polycomm Phones

2005-11-07 Thread Doug


At 15:36 11/7/2005, Krishna Sumanth Chava wrote:

hi,

Would like to have help in fixing the DTMF problem i am facing on Polycomm 
Soundpoint IP Phones


I am having the following network setup..

I have my Asterisk PBX server connected to the Cisco 3620 Router with an 
ethernet cable  which inturn is connected with a T1 circuit to my SIP 
Provider..


i have DTMF working on the Cisco 7940 phones but have problems with 
working on Polycomms.. i am using G711u  codec.


Will Appreciate your help if you can give me some comments and suggestions..



No problems here with Polycom 501s.  Using rfc2833 in "extensions"
in Asterisk.  Changing to "inband" sometimes helps.

What are your other settings?

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[Asterisk-Users] Harald Baron/EBAROH/CH/Ascom ist nicht anwesend.

2005-11-07 Thread Harald Baron


Ich werde ab  07.11.2005 nicht im Büro sein. Ich kehre zurück am
04.12.2005.

Ich bin vom 8.11.05 bis 4.12.05 nicht erreichbar und werde die Emails
sobald als möglich  bearbeiten. Dringende Anfragen bitte an Andreas
Widrig/CZWIAN/CH/Ascom oder Ralf Knobel/CZKNOR/AScom senden.

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[Asterisk-Users] libmfcr2 - spandsp.h: present but cannot be compiled

2005-11-07 Thread Jesus Mogollon

Hi all

When I try compiling libmfcr2 I get:

spandsp.h: present but cannot be compiled

Any ideas?
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[Asterisk-Users] new sip domain support and REGISTER requests

2005-11-07 Thread Günther Starnberger


hello,

since beta2 there is the new sip domain support - but somehow this 
feature is still a bit unclear for me.


routing to different contexts based on the domain in the extension.conf 
seems to be rather trivial, but is it possible to do the following things:


a) allow two users with the same name but different domains to register 
to asterisk. e.g. [EMAIL PROTECTED] and [EMAIL PROTECTED]


b) if a) is not possible: is it possible to set the outgoing domain when 
a user places a call (the opposite of fromdomain).


bye,
gst
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[Asterisk-Users] Problems with DTMF on Polycomm Phones

2005-11-07 Thread Krishna Sumanth Chava

hi,
 
Would like to have help in fixing the DTMF problem i am facing on Polycomm Soundpoint IP Phones
 
I am having the following network setup..
 
I have my Asterisk PBX server connected to the Cisco 3620 Router with an ethernet cable  which inturn is connected with a T1 circuit to my SIP Provider..
 
i have DTMF working on the Cisco 7940 phones but have problems with working on Polycomms.. i am using G711u  codec.
 
Will Appreciate your help if you can give me some comments and suggestions..
 
Thanks
krishna
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Re: [Asterisk-Users] Stopping Asterisk from forwarding calls?

2005-11-07 Thread pdhales

A workaround (bvut slighty messy) would be to set up two lines on each
phone.
Standard calls (which can be forwarded) can go the first (main)
line/extension.
Group calls go to line 2.

PaulH

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

Sent: Tuesday, November 08, 2005 6:18 AM
Subject: [Asterisk-Users] Stopping Asterisk from forwarding calls?


> The first time I asked this to the list I didn't do a great job of it so
> I'm posting again with more details.
>
> Problem: when ringing multiple extensions, if one user has their phone
> forwarded directly to voicemail, it stops the whole group from ringing
> because the voicemail picks up immediately.
>
> Also, after hours incoming calls are to ring all extensions so anyone
> can pickup. But if one person in the office has their phone forwarded
> the same problem occurs.
>
> What we need is for asterisk, when ringing multiple extensions, to
> completely ignore the forward requests and just ring the remaining
> phones.
>
> Reading the source code I see there are two parameters for channels,
> allowredir_in & allowredir_out. These offer me some hope that Asterisk
> has the ability but I couldn't figure out what these do or how to make
> use of them (I'm not a C programmer so maybe its just a red herring?).
>
> -- 
> John Lange
>
>
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[Asterisk-Users] [OTAnn] Feedback

2005-11-07 Thread shenanigans

I was interested in getting feedback from current mail group users.We have mirrored your mail list in a new application that provides a more aggregated and safe environment which utilizes the power of broadband.Roomity.com v 1.5 is a web 2.01 community webapp. Our newest version adds broadcast video and social networking such as favorite authors and an html editor.It?s free to join and any feedback would be appreciated.S.--Broadband interface (RIA) + mail box saftey = Asterisk_Users_List.roomity.com*Your* clubs, no sign up to read, ad supported; try broadband internet. ~~1131397871736~~--___
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Re: [Asterisk-Users] TDM400 FXO vs FXS Interrupt performance

2005-11-07 Thread pdhales

Considering that I am a user 'down under' that's even funnier.

PaulH

- Original Message - 
From: "Kevin P. Fleming" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<[EMAIL PROTECTED]>
Sent: Tuesday, November 08, 2005 2:12 AM
Subject: Re: [Asterisk-Users] TDM400 FXO vs FXS Interrupt performance


> [EMAIL PROTECTED] wrote:
> > I was pretty unhappy to see that the new cards had RJ12 sockets - you
can
> > put RJ12 into RJ45, but not the other way round...
> >
> > But I do know that a lot of people would ask if RJ12 would fit, so it
might
> > have been to cut down on support calls.
>
> Definitely not :-) It was done to appease the certification officials in
> a couple of places, including (IIRC), an unnamed carrier in the land
> 'down under' 
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[Asterisk-Users] Can't make calls from Asterisk IAX to other IAX

2005-11-07 Thread chawki hammoud

Hi:

I have been having this problem for sometime that I am
not able to solve and I hope someone can help. 

I can make VOIP calls between my Asterisk box and my
VOIP provider using sip channel without a problem. But
when I attempt to make a call using IAX, the call get
accepted and then get a hangup message:

This is the message I get when I attempt to make an
IAX call:

 Executing Dial("OSS/dsp",
"IAX2/callshopcompany/0017046872001") in new stack
-- Called callshopcompany/0017046872001
-- Call accepted by 213.61.187.150 (format gsm)
-- Format for call is gsm
-- Hungup 'IAX2/callshopcompany/1'
  == No one is available to answeer at this time



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Re: [Asterisk-Users] asterisks talking to asterisks

2005-11-07 Thread Mark Phillips

Indeed. They prefer to talk IAX protocol and can do everything you would 
expect them to do.


I'm a consultant in NJ. Contact me off list and I'll discuss with you 
how to do it.


Mark
973 828 1625



Jason Brashear wrote:

I have a request. I have a server in Texas
And one in NJ.
Is it possible for the system in Texas to log into the system in NJ so that 
Extensions can call each other?

-J


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--

Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
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Re: [Asterisk-Users] Stopping Asterisk from forwarding calls?

2005-11-07 Thread Tad Heckaman

I use [EMAIL PROTECTED], and if I have one of my Cisco phones forwarded to
my cell phone, when the phones ring in a ring group, it never forwards.
You may want to look at the latest configs that comes with
[EMAIL PROTECTED] and see if theres some special dialplans thats doing what
your looking for. 

Keep in mind I am using the call forward on the phone, and not the built in call forward in the dialplan.On 11/7/05, Kevin P. Fleming <
[EMAIL PROTECTED]> wrote:John Lange wrote:> Reading the source code I see there are two parameters for channels,
> allowredir_in & allowredir_out. These offer me some hope that Asterisk> has the ability but I couldn't figure out what these do or how to make> use of them (I'm not a C programmer so maybe its just a red herring?).
Those are entirely unrelated.At this time there is no method available to make Asterisk ignoreincoming '302 REDIRECT' from SIP phones. It may be possible to sendthose 'forward' requests to a context that has no valid extensions in
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Re: [Asterisk-Users] asterisks talking to asterisks

 Technically, yes.

On 11/7/05, Rob Lith <[EMAIL PROTECTED]> wrote:
> Wouldn't IAX be more efficient as you can trunk simultaneous calls and save
> bandwidth?
>
> Rob
>
>
> On 11/7/05, Andy Kuo < [EMAIL PROTECTED]> wrote:
> >
> > I do that through SIP.
> >
> > Assuming your TX extensions are 10XX, and NJ extensons are 20XX
> > On your NJ box...
> > sip.conf
> > [gwtx]
> > type=friend
> > secret=x
> > host=10.11.12.13(your TX IP)
> >
> > extensions.conf
> > [toTX]
> > exten => _10XX,1,Dial(SIP/[EMAIL PROTECTED])
> >
> > On your TX box
> > sip.conf
> > [gwnj]
> > type=friend
> > secret=x
> > host=20.21.22.23 (your IP for your NJ gateway)
> >
> > extensions.conf
> > [toNJ]
> > exten => _20XX,1,Dial(SIP/[EMAIL PROTECTED])
> >
> > I think you should be able to do similar using IAX too.
> > I don't know if this helps.  I'm still quite new to Asterisk too.
> >
> > Good Luck.
> > AK
> >
> >
> >
> >
> >
> > On 11/6/05, Jason Brashear < [EMAIL PROTECTED]> wrote:
> >
> > > I have a request. I have a server in Texas
> > > And one in NJ.
> > > Is it possible for the system in Texas to log into the system in NJ so
> that
> > > Extensions can call each other?
> > > -J
> > >
> > >
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> > >
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> >
> >
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> >
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> >
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> >
> >
>
>
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RE: [Asterisk-Users] Re: Help with dialplan to allow breakout to DISA

2005-11-07 Thread Alexander O. Lopez


> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Frank Tarczynski
> Sent: Monday, November 07, 2005 2:26 PM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] Re: Help with dialplan to allow
> breakout to DISA
>
> Yes, I know.
>
> BUT, I want the line to work as normal for incoming calls AND
> allow the user to breakout.
>
> So how do I merge:
> [incoming]
> exten => 1000,1,Ringing
> exten => 1000,2,Answer
> exten => 1000,n,Dial(IAX,iaxy/20)
> exten => 1000,n,Voicemail()
> exten => 1000,n,Hangup
>
> AND
>
> exten => *, 1, Authenticate(PASSWORD)
> exten => *, 2, DISA(no-password|DESTINATION_CONTEXT)
> exten => *, 3, Hangup
>
> to have Asterisk answer the line as normal but also react to
> the user pressing "*"?
>
> I've tried putting' all of the above in the same context but
> it doesn't work when I call in and press "*".
>
> Frank
>
> >
> > Message: 10
> > Date: Mon, 7 Nov 2005 12:45:05 -0500
> > From: Rusty Dekema <[EMAIL PROTECTED]>
> > Subject: Re: [Asterisk-Users] Help with dialplan to allow
> breakout to
> > DISA
> > To: asterisk-users@lists.digium.com
> > Message-ID:
> > <[EMAIL PROTECTED]>
> > Content-Type: text/plain; charset="iso-8859-1"
> >
> > I do it this way:
> >
> > exten => *, 1, Authenticate(PASSWORD)
> > exten => *, 2, DISA(no-password|DESTINATION_CONTEXT)
> > exten => *, 3, Hangup
> >
> > It seems to work fine...
> >
> > -Rusty
> >
> >
> >
> > On 11/7/05, Frank Tarczynski <[EMAIL PROTECTED]> wrote:
> >>
> >> I'm trying to set-up a dialplan for incoming calls that allows a
> >> breakout by pressing something like "*". Users would then
> be able to
> >> get an inside dial tone for voicemail, outgoing calls, etc.
> >>
> >> I've been struggling with Waitexten(), Disa() in the
> dialplan but not
> >> having much luck.
> >>
> >> Are there any good documents out there to assist me in this?
> >>
> >> Frank
> >>
>

I don't know if you are using DID and the exten => 1000 is because of this
but you can do this.

Try using a instead of *.

That would pop you into DISA when you press (a)sterisk from Voicemail.


Alex


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Re: [Asterisk-Users] Stopping Asterisk from forwarding calls?

2005-11-07 Thread Kevin P. Fleming


John Lange wrote:


Reading the source code I see there are two parameters for channels,
allowredir_in & allowredir_out. These offer me some hope that Asterisk
has the ability but I couldn't figure out what these do or how to make
use of them (I'm not a C programmer so maybe its just a red herring?).


Those are entirely unrelated.

At this time there is no method available to make Asterisk ignore 
incoming '302 REDIRECT' from SIP phones. It may be possible to send 
those 'forward' requests to a context that has no valid extensions in 
it, but I don't think we even support that at this time.

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Re: [Asterisk-Users] asterisks talking to asterisks

2005-11-07 Thread Gavin Spurgeon

There is a Step-by-Step HOW-TO on the Brilliant
voip-info.org site about connecting * servers..
The HOW-TO is Titled "Asterisk Connect 2 servers"
and can be found @ 
http://www.voip-info.org/tiki-index.php?page=Asterisk+Connect+2+servers

Hope This Helps...

Best Regards


Gavin Spurgeon
Assistant Systems Administrator
[EMAIL PROTECTED]
http://www.leighctc.kent.sch.uk 
Tel: 01322 620501
Fax: 01322 620599
IS HelpDesk : Ext 541


-- 
This message has been scanned for viruses and
dangerous content by the Systems @ the LeighCTC,
and is believed to be clean.

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Re: [Asterisk-Users] asterisks talking to asterisks

2005-11-07 Thread Rob Lith

Wouldn't IAX be more efficient as you can trunk simultaneous calls and save bandwidth?RobOn 11/7/05, Andy Kuo <
[EMAIL PROTECTED]> wrote:I do that through SIP.
 
Assuming your TX extensions are 10XX, and NJ extensons are 20XX
On your NJ box...
sip.conf
[gwtx]
type=friend
secret=x
host=10.11.12.13(your TX IP)
 
extensions.conf
[toTX]
exten => _10XX,1,Dial(SIP/[EMAIL PROTECTED])
 
On your TX box
sip.conf
[gwnj]
type=friend
secret=x
host=20.21.22.23 (your IP for your NJ gateway)
 
extensions.conf
[toNJ]
exten => _20XX,1,Dial(SIP/[EMAIL PROTECTED])
 
I think you should be able to do similar using IAX too.
I don't know if this helps.  I'm still quite new to Asterisk too.
 
Good Luck.
AK
 
 
 
 
On 11/6/05, Jason Brashear <
[EMAIL PROTECTED]> wrote:
I have a request. I have a server in Texas
And one in NJ.Is it possible for the system in Texas to log into the system in NJ so that
Extensions can call each other?-J___--Bandwidth and Colocation sponsored by 
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Re: [Asterisk-Users] Re: Asterisk-Users Digest, Vol 16, Issue 44

2005-11-07 Thread Rob Lith

I'm 15,000kms away and 9 hours time zone away yet I get superb same day response from the folk at Digium. Bending over backwards to help in anyway.I'm in South Africa.Met the Digium team at Astricon in Anaheim and I can say that while our business is probably a rounding error compared to what is done the the local market we are treated no less importantly.
Well done guys and keep it up.Rob On 11/7/05, patty McHenry <[EMAIL PROTECTED]> wrote:

The 104d has been available for a few weeks. I've had one for 4 weeks working with Sangoma on the driver side. My echo issues are a thing of the past. I had a few issues configuring, but it turned out to be Asterisk configuration- not Sangoma configuration. You must download their latest drivers. Their tech support is a
9.9 out of 10. These guys know what they are doing.I'm trying to run a business here. Why should I compromise my business with Digium's 406P solution if they take 4 days to get back to me, or sell me soemthing that doesn't work in the machine I chose to use?
What I really want is the Sangoma analog FXS/FXO hardware!!!
Pat

Yahoo! FareChase - Search multiple travel sites in one click.

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RE: [Asterisk-Users] Asterisk 1.2beta2 and UIP200

2005-11-07 Thread Anton Krall

Some ATAs do not like the qualify, I have some MTA102 and that's the case
with those, if I enable qualify, the ata doesn't work with asterisk, if I
disable qualify, the ATA works without problems. 

|-Original Message-
|From: [EMAIL PROTECTED] 
|[mailto:[EMAIL PROTECTED] On Behalf Of C F
|Sent: Monday, November 07, 2005 9:57 AM
|To: Asterisk Users Mailing List - Non-Commercial Discussion
|Subject: Re: [Asterisk-Users] Asterisk 1.2beta2 and UIP200
|
|The unreachable is the problem. Try adding a qualify=no to 
|that sip entry.
|
|On 11/7/05, Waldo Rubinstein <[EMAIL PROTECTED]> wrote:
|> Additionally:
|>
|> *CLI> sip show peer 100074
|>
|>   * Name   : 100074
|>   Secret   : 
|>   MD5Secret: 
|>   Context  : qa
|>   Subscr.Cont. : 
|>   Language : en
|>   AMA flags: Unknown
|>   CallingPres  : Presentation Allowed, Not Screened
|>   Callgroup:
|>   Pickupgroup  :
|>   Mailbox  : [EMAIL PROTECTED]
|>   VM Extension : asterisk
|>   LastMsgsSent : 0
|>   Call limit   : 0
|>   Dynamic  : Yes
|>   Callerid : "Waldo Rubinstein" <211>
|>   Expire   : 11077
|>   Insecure : no
|>   Nat  : No
|>   ACL  : No
|>   CanReinvite  : No
|>   PromiscRedir : No
|>   User=Phone   : No
|>   Trust RPID   : No
|>   Send RPID: No
|>   DTMFmode : rfc2833
|>   LastMsg  : 0
|>   ToHost   :
|>   Addr->IP : 10.0.10.236 Port 5060
|>   Defaddr->IP  : 0.0.0.0 Port 5060
|>   Def. Username: 100074
|>   SIP Options  : (none)
|>   Codecs   : 0x6 (gsm|ulaw)
|>   Codec Order  : (ulaw,gsm)
|>   Status   : UNREACHABLE
|>   Useragent: Uniden SIP Phone p2 Ver BS4.63
|>   Reg. Contact : sip:[EMAIL PROTECTED]:5060
|>
|> Thanks,
|> Waldo
|>
|> On Nov 6, 2005, at 11:11 PM, C F wrote:
|>
|> > can you post the sip.conf for that uip200?
|> >
|> > On 11/6/05, Waldo Rubinstein <[EMAIL PROTECTED]> wrote:
|> >> When I dial the extension, I get this:
|> >>
|> >>  -- Executing Dial("IAX2/gateway0-16386", "SIP/100074|20") in 
|> >> new stack
|> >>== Everyone is busy/congested at this time (1:0/0/1)
|> >>
|> >>
|> >> When I do a sip show peer 100074, everything it shows matches the 
|> >> results of executing the same sip show peer on * 1.0.9 and 1.2b1,
|> >> except:
|> >>
|> >>Status   : UNREACHABLE
|> >>
|> >> However, I can make any type of calls from them phone. I can ping 
|> >> the phone from the * server. It's just that * 1.2b2 can't 
|reach it, 
|> >> for some reason.
|> >>
|> >> Thanks,
|> >> Waldo
|> >>
|> >> On Nov 6, 2005, at 1:37 PM, C F wrote:
|> >>
|> >>> Whats the exact CLI output you are getting when calling that 
|> >>> extension?
|> >>>
|> >>> On 11/6/05, Waldo Rubinstein <[EMAIL PROTECTED]> wrote:
|>  Nope. It isn't active. I even factory reseted the phone 
|but still 
|>  the same. One more piece of information: it works just fine in 
|>  1.2b1. I beginning to think it could be a bug in 1.2b2.
|> 
|>  Any other ideas/suggestions?
|> 
|>  Thanks,
|>  Waldo
|> 
|>  On Nov 5, 2005, at 9:10 PM, C F wrote:
|> 
|> > You sure that the DND (Do Not Disturb) button is not active on 
|> > the UIP200?
|> >
|> > On 11/4/05, Waldo Rubinstein <[EMAIL PROTECTED]> wrote:
|> >> I am running * 1.2b2 with some UIP200 phones and a bunch of 
|> >> X-Pro phones.
|> >>
|> >> All phones register fine with * and I can place 
|outbound calls 
|> >> with no problem.
|> >>
|> >> I can call from the X-Pro to any other X-Pro. I can call from 
|> >> UIP200 to any other X-Pro. However, the UIP200 cannot receive 
|> >> calls.
|> >> Every
|> >> time I call the UIP200, the CLI says Everyone is Busy/ 
|> >> Congested and sends the call to voicemail.
|> >>
|> >> Everything is in the same network. I have in sip.conf
|> >> localnet=10.0.10.0/24
|> >>
|> >> and in each UIP200 sip profile
|> >> nat=never
|> >>
|> >> What's wrong?
|> >>
|> >> I have the same configuration in * 1.0.9 and it works 
|just fine.
|> >> Could the SIP protocol be broken in 1.2b2?
|> >>
|> >> Thanks,
|> >> Waldo
|> >>
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|> >> Asterisk-Users@lists.digium.com 
|> >> http://lists.digium.com/mailman/listinfo/asterisk-users
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|> 
|>

Re: [Asterisk-Users] DNS Server Failure wreaks havoc

2005-11-07 Thread Mojo with Horan & Company, LLC

I installed a caching dns server on the * box itself 'cause when the 
external dns stopped resolving guess where my emailed voicemails went? 
Ya, I don't know either.  :P  They weren't in the mailq but showed up 
just a little while later when the names began resolving again :)


Brian Capouch wrote:
I don't think this is a new issue--I've seen it talked about on the list 
before.  I don't know if I've ever seen anyone post a fix.


My DNS server went out last night in a horrendous storm when an upstream 
link went down.  The madness is that the behavior of the whole server, 
including the part that's handling my POTS lines, gets wigged out on a 
DNS failure, making the whole system unusable.  I have two questions; 
being able to solve either would be wonderful:


* Is it true that if I hand-resolve the server names in all the config 
files, and then use those instead of the hostnames, this problem won't 
occur?  That's not exactly optimal, of course, since it defeats the 
whole purpose of dynamic name binding.  But it's hard to explain to my 
SOHO customers, who don't really need any IP-based functionality 
(although I give all of them some complimentary minutes on nufone) why 
their phones go down when the Internet is down.


* Is it true that there's no way to get applications in Linux, generally 
speaking, to try more than a single server when doing a name resolve? 
Only the first server listed in /etc/resolv.conf (on my systems, anyway) 
seems to ever get consulted.


I think both of these situations are pretty serious failings, if in fact 
they're failings in the systems and not this bedeviled cranium.


Thanks.

B.

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--
Mojo <[EMAIL PROTECTED]>
Office Manger, Horan & Company, LLC
(907) 747- x112
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Re: [Asterisk-Users] MP3 or OGG


Thanks
- Waldo

On Nov 7, 2005, at 1:52 PM, BJ Weschke wrote:


 You're probably not going to be violating any patent protections by
using OGG instead of MP3. As far as compression goes, I've found the
difference between the two of them to be negligible. I've always used
OGG when possible to stay "IP safe".

On 11/7/05, Waldo Rubinstein <[EMAIL PROTECTED]> wrote:

I'm trying to archive out call recordings and would appreciate some
feedback as to which audio compression is more recommended MP3 or
OGG. In the past, I've use lame to convert to MP3, but I noticed the
audio volume drops significantly. Is it just a setting on the command
line of lame or is OGG better? Which achieves higher compression
rates while maintaining call quality?

Thanks,
Waldo

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Re: [Asterisk-Users] Asterisk 1.2beta2 and UIP200

I do have qualify=yes pretty much in all my sip entries. I just  
changed all the entries where I have a UIP200 to qualify=no and now  
they all work. The funny thing is that it worked with qualify=yes in  
1.0.9 and 1.2b1


Thanks,
Waldo

On Nov 7, 2005, at 1:29 PM, C F wrote:


I guess that somewhere in your settings you have a qualify on, or that
1.2 has it on by default. Do the following:
cd /etc/asterisk
grep ".*qualify.*" ./*
and see the output, if the only line that has qualify is that
qualify=no, then this looks like a bug to me. Please report back.

On 11/7/05, Waldo Rubinstein <[EMAIL PROTECTED]> wrote:

Ok. That fixed it, but why? It works just fine in 1.0.9 and 1.2b1.
Very strange.

Anyway, thanks.

- Waldo

On Nov 7, 2005, at 10:57 AM, C F wrote:


The unreachable is the problem. Try adding a qualify=no to that sip
entry.

On 11/7/05, Waldo Rubinstein <[EMAIL PROTECTED]> wrote:

Additionally:

*CLI> sip show peer 100074

  * Name   : 100074
  Secret   : 
  MD5Secret: 
  Context  : qa
  Subscr.Cont. : 
  Language : en
  AMA flags: Unknown
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup:
  Pickupgroup  :
  Mailbox  : [EMAIL PROTECTED]
  VM Extension : asterisk
  LastMsgsSent : 0
  Call limit   : 0
  Dynamic  : Yes
  Callerid : "Waldo Rubinstein" <211>
  Expire   : 11077
  Insecure : no
  Nat  : No
  ACL  : No
  CanReinvite  : No
  PromiscRedir : No
  User=Phone   : No
  Trust RPID   : No
  Send RPID: No
  DTMFmode : rfc2833
  LastMsg  : 0
  ToHost   :
  Addr->IP : 10.0.10.236 Port 5060
  Defaddr->IP  : 0.0.0.0 Port 5060
  Def. Username: 100074
  SIP Options  : (none)
  Codecs   : 0x6 (gsm|ulaw)
  Codec Order  : (ulaw,gsm)
  Status   : UNREACHABLE
  Useragent: Uniden SIP Phone p2 Ver BS4.63
  Reg. Contact : sip:[EMAIL PROTECTED]:5060

Thanks,
Waldo

On Nov 6, 2005, at 11:11 PM, C F wrote:


can you post the sip.conf for that uip200?

On 11/6/05, Waldo Rubinstein <[EMAIL PROTECTED]> wrote:

When I dial the extension, I get this:

 -- Executing Dial("IAX2/gateway0-16386", "SIP/100074|20")
in new
stack
   == Everyone is busy/congested at this time (1:0/0/1)


When I do a sip show peer 100074, everything it shows matches the
results of executing the same sip show peer on * 1.0.9 and 1.2b1,
except:

   Status   : UNREACHABLE

However, I can make any type of calls from them phone. I can
ping the
phone from the * server. It's just that * 1.2b2 can't reach  
it, for

some reason.

Thanks,
Waldo

On Nov 6, 2005, at 1:37 PM, C F wrote:


Whats the exact CLI output you are getting when calling that
extension?

On 11/6/05, Waldo Rubinstein <[EMAIL PROTECTED]> wrote:

Nope. It isn't active. I even factory reseted the phone but
still the
same. One more piece of information: it works just fine in
1.2b1. I
beginning to think it could be a bug in 1.2b2.

Any other ideas/suggestions?

Thanks,
Waldo

On Nov 5, 2005, at 9:10 PM, C F wrote:


You sure that the DND (Do Not Disturb) button is not active
on the
UIP200?

On 11/4/05, Waldo Rubinstein <[EMAIL PROTECTED]> wrote:

I am running * 1.2b2 with some UIP200 phones and a bunch of
X-Pro
phones.

All phones register fine with * and I can place outbound  
calls

with
no problem.

I can call from the X-Pro to any other X-Pro. I can call from
UIP200
to any other X-Pro. However, the UIP200 cannot receive calls.
Every
time I call the UIP200, the CLI says Everyone is Busy/
Congested and
sends the call to voicemail.

Everything is in the same network. I have in sip.conf
localnet=10.0.10.0/24

and in each UIP200 sip profile
nat=never

What's wrong?

I have the same configuration in * 1.0.9 and it works just
fine.
Could the SIP protocol be broken in 1.2b2?

Thanks,
Waldo

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RE: [Asterisk-Users] asterisks talking to asterisks

2005-11-07 Thread Paul

Standard IAX link found on wiki


> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Jason Brashear
> Sent: Sunday, November 06, 2005 11:13 AM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: [Asterisk-Users] asterisks talking to asterisks
> 
> I have a request. I have a server in Texas
> And one in NJ.
> Is it possible for the system in Texas to log into the system in NJ so
> that
> Extensions can call each other?
> -J
> 
> 
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Re: [Asterisk-Users] asterisks talking to asterisks

2005-11-07 Thread Andy Kuo

I do that through SIP.
 
Assuming your TX extensions are 10XX, and NJ extensons are 20XX
On your NJ box...
sip.conf
[gwtx]
type=friend
secret=x
host=10.11.12.13(your TX IP)
 
extensions.conf
[toTX]
exten => _10XX,1,Dial(SIP/[EMAIL PROTECTED])
 
On your TX box
sip.conf
[gwnj]
type=friend
secret=x
host=20.21.22.23 (your IP for your NJ gateway)
 
extensions.conf
[toNJ]
exten => _20XX,1,Dial(SIP/[EMAIL PROTECTED])
 
I think you should be able to do similar using IAX too.
I don't know if this helps.  I'm still quite new to Asterisk too.
 
Good Luck.
AK
 
 
 
 
On 11/6/05, Jason Brashear <[EMAIL PROTECTED]> wrote:
I have a request. I have a server in TexasAnd one in NJ.Is it possible for the system in Texas to log into the system in NJ so that
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[Asterisk-Users] Re: Help with dialplan to allow breakout to DISA

2005-11-07 Thread Frank Tarczynski

Yes, I know.

BUT, I want the line to work as normal for incoming calls AND allow the
user to breakout.

So how do I merge:
[incoming]
exten => 1000,1,Ringing
exten => 1000,2,Answer
exten => 1000,n,Dial(IAX,iaxy/20)
exten => 1000,n,Voicemail()
exten => 1000,n,Hangup

AND

exten => *, 1, Authenticate(PASSWORD)
exten => *, 2, DISA(no-password|DESTINATION_CONTEXT)
exten => *, 3, Hangup

to have Asterisk answer the line as normal but also react to the user
pressing "*"?

I've tried putting' all of the above in the same context but it doesn't
work when I call in and press "*".

Frank

>
> Message: 10
> Date: Mon, 7 Nov 2005 12:45:05 -0500
> From: Rusty Dekema <[EMAIL PROTECTED]>
> Subject: Re: [Asterisk-Users] Help with dialplan to allow breakout to
>   DISA
> To: asterisk-users@lists.digium.com
> Message-ID:
>   <[EMAIL PROTECTED]>
> Content-Type: text/plain; charset="iso-8859-1"
>
> I do it this way:
>
> exten => *, 1, Authenticate(PASSWORD)
> exten => *, 2, DISA(no-password|DESTINATION_CONTEXT)
> exten => *, 3, Hangup
>
> It seems to work fine...
>
> -Rusty
>
>
>
> On 11/7/05, Frank Tarczynski <[EMAIL PROTECTED]> wrote:
>>
>> I'm trying to set-up a dialplan for incoming calls that allows a
>> breakout
>> by pressing something like "*". Users would then be able to get an
>> inside
>> dial tone for voicemail, outgoing calls, etc.
>>
>> I've been struggling with Waitexten(), Disa() in the dialplan but not
>> having much luck.
>>
>> Are there any good documents out there to assist me in this?
>>
>> Frank
>>


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[Asterisk-Users] asterisk 1.2b2 compiling problem

2005-11-07 Thread Don Pobanz

I just checked out asterisk 1.2b2 for zaptel, libpri, asterisk and 
asterisk-sounds. Zaptel and libpri compile fine with a 'make clean' and 
'make install'. However even after a make clean, the asterisk 'make 
install' does not finish on my redhat 7.3 system. 
CVS-D2005.09.12.05.00.00-09/14/05-02:05:11 is currently running.


Here are the last few lines before erroring out.

chan_agent.c:1684: parse error before `char'
chan_agent.c:1701: `agent_goodbye' undeclared (first use in this function)
chan_agent.c:1701: (Each undeclared identifier is reported only once
chan_agent.c:1701: for each function it appears in.)
chan_agent.c:1708: `tmpoptions' undeclared (first use in this function)
chan_agent.c:1714: `update_cdr' undeclared (first use in this function)
chan_agent.c:1732: `context' undeclared (first use in this function)
chan_agent.c:1737: `play_announcement' undeclared (first use in this 
function)

chan_agent.c:1864: `filename' undeclared (first use in this function)
make[1]: *** [chan_agent.o] Error 1
make[1]: Leaving directory `/usr/src/asterisk/channels'
make: *** [subdirs] Error 1

Any ideas?

Don Pobanz
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RE: [Asterisk-Users] CentOS vs. Vanilla Kernel

2005-11-07 Thread Bryan J. Smith

Ryan Amos <[EMAIL PROTECTED]> wrote:
> The default CentOS kernel has worked fine for me.
> Just an FYI; CentOS uses the RedHat EL kernel source to
> build... It's pretty heavily patched so if you want to use
> the latest stable, download the SRPMs from RedHat/CentOS
> and patch in the kernel.org patches.

It would be easier to patch in those patches already merged
in the Rawhide (Fedora Development) kernels.  Especially if
you rebuild from SRPM proper.

Just a clarification, I'm not advocating using the Rawhide
kernels.  If there is one place where Fedora
Development/Core/Legacy differ heavily with Red Hat
Enterprise Linux, it's at the kernel.  But the patches from
Rawhide kernels would probably be a far better fit for the
RHEL kernels.

> But yeah, stick with the CentOS kernel unless you have
> problems. 

Agreed.  Way too much is added/removed/changed.

-- 
Bryan J. Smith| Sent from Yahoo Mail
mailto:[EMAIL PROTECTED] |  (please excuse any
http://thebs413.blogspot.com/ |   missing headers)
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[Asterisk-Users] Stopping Asterisk from forwarding calls?

2005-11-07 Thread John Lange

The first time I asked this to the list I didn't do a great job of it so
I'm posting again with more details.

Problem: when ringing multiple extensions, if one user has their phone
forwarded directly to voicemail, it stops the whole group from ringing
because the voicemail picks up immediately.

Also, after hours incoming calls are to ring all extensions so anyone
can pickup. But if one person in the office has their phone forwarded
the same problem occurs.

What we need is for asterisk, when ringing multiple extensions, to
completely ignore the forward requests and just ring the remaining
phones.

Reading the source code I see there are two parameters for channels,
allowredir_in & allowredir_out. These offer me some hope that Asterisk
has the ability but I couldn't figure out what these do or how to make
use of them (I'm not a C programmer so maybe its just a red herring?).

-- 
John Lange


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RE: [Asterisk-Users] Change Asterisk User

2005-11-07 Thread Ryan Amos

Use group permissions. Add the apache user to the asterisk group and
give the group the appropriate read and/or write access. IMO this is the
easiest way to get around the apache permissions thing, and probably the
Right Way (tm)

-Ryan

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of amaury
BOSSE
Sent: Monday, November 07, 2005 12:44 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Change Asterisk User

Thanks for your answer,
I am working on Debian Sarge but I have compiled Astersik 1.0.9 myself
without .deb Packages.
I need to access to voicemail and sound files from my web-interface (php
and cgi/perl) but I can't change Apache user because of others
applications.
Asterisk creates files under Asterisk user and I have to access them
from www-data user.
Do you have other solution? I have tried using sudo but it doesn't seem
to work.

Regards,
Amaury


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Re: [Asterisk-Users] Getting ztdummy to load on startup for X100P

2005-11-07 Thread Mojo with Horan & Company, LLC

I'm not sure where in your startup process asterisk gets loaded.  I load 
my asterisk from my rc.local file, so I can of course control when 
ztdummy would be loaded in relation to asterisk.


Tzafrir Cohen wrote:

On Fri, Nov 04, 2005 at 11:43:37AM -0900, Mojo with Horan & Company, LLC wrote:


Try putting a line at the very bottom of /etc/rc.d/rc.local like
/sbin/modprobe ztdummy



Which means ztdummy gts loaded only after asterisk is run?



--
Mojo <[EMAIL PROTECTED]>
Office Manger, Horan & Company, LLC
(907) 747- x112
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Re: [Asterisk-Users] MP3 or OGG

 You're probably not going to be violating any patent protections by
using OGG instead of MP3. As far as compression goes, I've found the
difference between the two of them to be negligible. I've always used
OGG when possible to stay "IP safe".

On 11/7/05, Waldo Rubinstein <[EMAIL PROTECTED]> wrote:
> I'm trying to archive out call recordings and would appreciate some
> feedback as to which audio compression is more recommended MP3 or
> OGG. In the past, I've use lame to convert to MP3, but I noticed the
> audio volume drops significantly. Is it just a setting on the command
> line of lame or is OGG better? Which achieves higher compression
> rates while maintaining call quality?
>
> Thanks,
> Waldo
>
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RE: [Asterisk-Users] CentOS vs. Vanilla Kernel

2005-11-07 Thread Ryan Amos

The default CentOS kernel has worked fine for me.

Just an FYI; CentOS uses the RedHat EL kernel source to build... It's
pretty heavily patched so if you want to use the latest stable, download
the SRPMs from RedHat/CentOS and patch in the kernel.org patches.

But yeah, stick with the CentOS kernel unless you have problems. 

-Ryan

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Julian
Lyndon-Smith
Sent: Monday, November 07, 2005 12:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] CentOS vs. Vanilla Kernel

HW: HP DL360 1GB Ram Single 3GHz Pentium (with Hyper-threading turned
off).
OS: CentOS 4.2
Dual Embedded NIC enabled
USB disabled
serial disabled
printer disabled
2x73GB SCSI in HW Raid 1

What is the opinion of this fine list  - should I use the default CentOS

kernel (2.6.9-22.0.1.EL) or download from kernel.org the latest stable 
(2.6.14)

Anyone got any clues / hints / tips on what should go into the kernel ?

All views and comments appreciated :)

Julian.

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Re: [Asterisk-Users] asterisk-1.2-bêta2 | presen ce/subscription support in the SIP channel driver

 There could be 1 of 100 reasons that's causing this not to work.

 Let's start out by you posting your relevant sections of sip.conf and
extensions.conf and then do a "sip show subscriptions" from the CLI
and give us the results of that as well.

On 11/7/05, harry gaillac <[EMAIL PROTECTED]> wrote:
> Hello,
>
> I configure Polycom ip300 for presence but when status
> change notify is no sent to subscriber !?
>
> Why ?
>
> Regards
> Harry
>
>
>
>
>
>
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RE: [Asterisk-Users] Change Asterisk User

2005-11-07 Thread amaury BOSSE

Thanks for your answer,
I am working on Debian Sarge but I have compiled Astersik 1.0.9 myself
without .deb Packages.
I need to access to voicemail and sound files from my web-interface (php
and cgi/perl) but I can't change Apache user because of others
applications.
Asterisk creates files under Asterisk user and I have to access them
from www-data user.
Do you have other solution? I have tried using sudo but it doesn't seem
to work.

Regards,
Amaury


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Re: [Asterisk-Users] Re: Double DTMF with tdm card

2005-11-07 Thread Bart Fisher

Just wanted to let the group know this problem is fixed (for me).  Mark 
log-on to my system and found a "bug" in chan_zap.c on Saturday night and 
made the correction - I believe the change is available for download by now 
at zaptel 1.0.9.2, or CVS Head.  He stated that recent changes "unmask" the 
bug and the change will slightly improve TE410P performance


Thanks for you help!

Bart


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


Sent: Thursday, November 03, 2005 6:20 PM
Subject: Re: [Asterisk-Users] Re: Double DTMF with tdm card



I just heard back from Mark.  I volunteered my system to used for testing.


From Mark:

"Generally, issues which involve Digium hardware should go through
technical support, even if it's a newly introduced problem, because they
can help narrow down the nature of the failure, what might have changed,
etc.  If you or a representative of this "group" want to fill this role
instead, I'm happy to work with you, but I need the situation labbed up in
an environment where the problem can be demonstrated, where I can remotely
log in, and where I can edit, recompile, and test in real time (i.e. not
on a production server).  If you want to set all this up and contact me
with login details and a number where I can see the problem occur, then
when it's ready, I can work with you directly.

Mark"


Bart

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


Sent: Thursday, November 03, 2005 2:41 PM
Subject: Re: [Asterisk-Users] Re: Double DTMF with tdm card


"If" in fact it is the exact same issue, then I'd suggest creating a 
feature

request to add "disable dtmf detection after answer supervision" and post
it to the -dev list (which is what Kevin is suggesting now). You will 
need
to be explain the wanted functionality in terms that non-telephone 
technical

folks can understand. I'd suggest a zapata.conf configuration option that
is something like "ignore-dtmf-after-answersup" with a default value of
however it works today (=no).

Think about that carefully as the option set to =yes will disable dtmf
from interacting with your internal * ivr (assuming you have one).
What you want is kind of related to a pass-thru connection and not
necessarily for a connection terminating within *. There might be other
ways to handle your objective.

This same issue comes up in other cases where interaction with an 
external

ivr is needed, some airlines automated systems, etc.

I honestly believe the exact same thing should apply to iax2 incoming
trunks as well. Not so sure about sip trunks.

I'd agree with your statement relative to digium support being contacted,
but if the boss-man suggests it, there might be an unstated reason for
that. If properly worded (and with the supporting documentation that you
heard the problem with a T1 analyzer), they might be able to help support
the need for some kind of option.



This is exactly what is happening...  It's bad news...  In my case the 
T1 is
connected to a PBX Voice Mail.  So, double dialing really messes up 
thing

like when entering a passcode.  Where passcode "1234" arrives as
"11223344" - no good.   This would always be an issue in cases where the
call is Tandem thru Asterisk.

In fact, I can't see any reason to repeat the digits when the signal is
"inband" and/or Zap Bridged call. -  And why was it changed from 1.0.9?
Makes no sense.

It seems an easy fix, maybe a digit time-out parameter or disable 
sending

after answer supervision has been achieved.

Given what you say, Digium Support won't be able to fix without code
changes - I don't know what Mark is thinking here.

Bart

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

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

Sent: Thursday, November 03, 2005 1:17 PM
Subject: Re: [Asterisk-Users] Re: Double DTMF with tdm card


>I might be able to shed a little light on this...
>
> Asterisk is constantly listening for dtmf tones on most channels. Its
> either listening for inband or rfc-out-of-band, depending upon how the
> attached device is defined and how asterisk def's for that device is
> defined. For pstn interfaces, the "cards" don't listen for any dtmf, 
> but

> rather the zap sutff is listening.
>
> If a call is generated from some external source (coming into *), the
> dtmf will be inband once a channel is answered. For commercial 
> telephone
> equipment, once a channel is answered, the telephone equipment no 
> longer

> listens for dtmf (its simply passed inband). Not so with asterisk, and
> this point has been argued with Mark some time ago; asterisk still
> listens and trys to handle the dtmf, translating to rfc2833 as it 
> thinks

> is necessary.
>
> So, it sounds like you have an answered T1 call where * is still 
> trying
> to handle dtmf (regener

Re: [Asterisk-Users] CentOS vs. Vanilla Kernel

2005-11-07 Thread Jesse Keating

On Mon, 2005-11-07 at 18:17 +, Julian Lyndon-Smith wrote:
> 
> What is the opinion of this fine list  - should I use the default CentOS 
> kernel (2.6.9-22.0.1.EL) or download from kernel.org the latest stable 
> (2.6.14)
> 
> Anyone got any clues / hints / tips on what should go into the kernel ?
> 
> All views and comments appreciated :)

Depends.  Do you want to spend your time using the system and working on
Asterisk, or do you want to spend your time tracking kernel changes,
patching security fixes, tracking down kernel bugs, breaking rpm deps
and working around that, etc, etc, etc...

Red Hat puts a lot of work into making sure their kernel is solid and
secure.  They backport security fixes and bug fixes into their stable
tree, 2.6.9.  In my opinion, I'd rather let the folks that know the
kernel work on it rather than spend my limited time on it.

-- 
Jesse Keating
GameHouse -- Systems Engineer

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Re: [Asterisk-Users] CentOS vs. Vanilla Kernel

2005-11-07 Thread Jason Pyeron

my $0.02 if you are going w/ RHEL use one of the kernel rpms provided. You 
can always add a module rpm to supplement it. Once you roll your own there 
might be better distros for you, since you are going to break the 
rpm/up2date features that make RHEL a desirable product.


On Mon, 7 Nov 2005, Julian Lyndon-Smith wrote:


HW: HP DL360 1GB Ram Single 3GHz Pentium (with Hyper-threading turned off).
OS: CentOS 4.2
Dual Embedded NIC enabled
USB disabled
serial disabled
printer disabled
2x73GB SCSI in HW Raid 1

What is the opinion of this fine list  - should I use the default CentOS 
kernel (2.6.9-22.0.1.EL) or download from kernel.org the latest stable 
(2.6.14)


Anyone got any clues / hints / tips on what should go into the kernel ?

All views and comments appreciated :)

Julian.

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Re: [Asterisk-Users] asterisk as SIP gateway

2005-11-07 Thread Peter Petrov


Miloš Kocbek wrote:

I want to enable access to some context in asterisk without authentication.


In sip.conf:

[username]
type=friend
host=x.x.x.x
context=context_for_this_user




--
Regards,
Peter Petrov
[EMAIL PROTECTED]
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Re: [Asterisk-Users] Asterisk 1.2beta2 and UIP200

I guess that somewhere in your settings you have a qualify on, or that
1.2 has it on by default. Do the following:
cd /etc/asterisk
grep ".*qualify.*" ./*
and see the output, if the only line that has qualify is that
qualify=no, then this looks like a bug to me. Please report back.

On 11/7/05, Waldo Rubinstein <[EMAIL PROTECTED]> wrote:
> Ok. That fixed it, but why? It works just fine in 1.0.9 and 1.2b1.
> Very strange.
>
> Anyway, thanks.
>
> - Waldo
>
> On Nov 7, 2005, at 10:57 AM, C F wrote:
>
> > The unreachable is the problem. Try adding a qualify=no to that sip
> > entry.
> >
> > On 11/7/05, Waldo Rubinstein <[EMAIL PROTECTED]> wrote:
> >> Additionally:
> >>
> >> *CLI> sip show peer 100074
> >>
> >>   * Name   : 100074
> >>   Secret   : 
> >>   MD5Secret: 
> >>   Context  : qa
> >>   Subscr.Cont. : 
> >>   Language : en
> >>   AMA flags: Unknown
> >>   CallingPres  : Presentation Allowed, Not Screened
> >>   Callgroup:
> >>   Pickupgroup  :
> >>   Mailbox  : [EMAIL PROTECTED]
> >>   VM Extension : asterisk
> >>   LastMsgsSent : 0
> >>   Call limit   : 0
> >>   Dynamic  : Yes
> >>   Callerid : "Waldo Rubinstein" <211>
> >>   Expire   : 11077
> >>   Insecure : no
> >>   Nat  : No
> >>   ACL  : No
> >>   CanReinvite  : No
> >>   PromiscRedir : No
> >>   User=Phone   : No
> >>   Trust RPID   : No
> >>   Send RPID: No
> >>   DTMFmode : rfc2833
> >>   LastMsg  : 0
> >>   ToHost   :
> >>   Addr->IP : 10.0.10.236 Port 5060
> >>   Defaddr->IP  : 0.0.0.0 Port 5060
> >>   Def. Username: 100074
> >>   SIP Options  : (none)
> >>   Codecs   : 0x6 (gsm|ulaw)
> >>   Codec Order  : (ulaw,gsm)
> >>   Status   : UNREACHABLE
> >>   Useragent: Uniden SIP Phone p2 Ver BS4.63
> >>   Reg. Contact : sip:[EMAIL PROTECTED]:5060
> >>
> >> Thanks,
> >> Waldo
> >>
> >> On Nov 6, 2005, at 11:11 PM, C F wrote:
> >>
> >>> can you post the sip.conf for that uip200?
> >>>
> >>> On 11/6/05, Waldo Rubinstein <[EMAIL PROTECTED]> wrote:
>  When I dial the extension, I get this:
> 
>   -- Executing Dial("IAX2/gateway0-16386", "SIP/100074|20")
>  in new
>  stack
> == Everyone is busy/congested at this time (1:0/0/1)
> 
> 
>  When I do a sip show peer 100074, everything it shows matches the
>  results of executing the same sip show peer on * 1.0.9 and 1.2b1,
>  except:
> 
> Status   : UNREACHABLE
> 
>  However, I can make any type of calls from them phone. I can
>  ping the
>  phone from the * server. It's just that * 1.2b2 can't reach it, for
>  some reason.
> 
>  Thanks,
>  Waldo
> 
>  On Nov 6, 2005, at 1:37 PM, C F wrote:
> 
> > Whats the exact CLI output you are getting when calling that
> > extension?
> >
> > On 11/6/05, Waldo Rubinstein <[EMAIL PROTECTED]> wrote:
> >> Nope. It isn't active. I even factory reseted the phone but
> >> still the
> >> same. One more piece of information: it works just fine in
> >> 1.2b1. I
> >> beginning to think it could be a bug in 1.2b2.
> >>
> >> Any other ideas/suggestions?
> >>
> >> Thanks,
> >> Waldo
> >>
> >> On Nov 5, 2005, at 9:10 PM, C F wrote:
> >>
> >>> You sure that the DND (Do Not Disturb) button is not active
> >>> on the
> >>> UIP200?
> >>>
> >>> On 11/4/05, Waldo Rubinstein <[EMAIL PROTECTED]> wrote:
>  I am running * 1.2b2 with some UIP200 phones and a bunch of
>  X-Pro
>  phones.
> 
>  All phones register fine with * and I can place outbound calls
>  with
>  no problem.
> 
>  I can call from the X-Pro to any other X-Pro. I can call from
>  UIP200
>  to any other X-Pro. However, the UIP200 cannot receive calls.
>  Every
>  time I call the UIP200, the CLI says Everyone is Busy/
>  Congested and
>  sends the call to voicemail.
> 
>  Everything is in the same network. I have in sip.conf
>  localnet=10.0.10.0/24
> 
>  and in each UIP200 sip profile
>  nat=never
> 
>  What's wrong?
> 
>  I have the same configuration in * 1.0.9 and it works just
>  fine.
>  Could the SIP protocol be broken in 1.2b2?
> 
>  Thanks,
>  Waldo
> 
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[Asterisk-Users] Re: Help with dialplan to allow breakout to DISA

2005-11-07 Thread Brent Torrenga

I have my dialplan setup the same, only with 0 instead of * as the
extension. What would the reason be, after authenticating, that I get a
dialtone, as expected, but no response to any DTMF tones I input? It is as
if the DISA works, gives me tone, but is unresponsive? The destination
context is exactly the same as any of my internal extensions, too...

>I do it this way:
>
>exten => *, 1, Authenticate(PASSWORD)
>exten => *, 2, DISA(no-password|DESTINATION_CONTEXT)
>.exten => *, 3, Hangup..
>
>It seems to work fine...
>
>-Rusty


Sincerely,

Brent A. Torrenga
[EMAIL PROTECTED]

Torrenga Engineering, Inc.
907 Ridge Road
Munster, Indiana 46321-1771

219.836.8918 Voice
219.836.1138 Facsimile 

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[Asterisk-Users] asterisk-1.2-bêta2 | pres ence/subscription support in the SIP channel driver

2005-11-07 Thread harry gaillac

Hello,

I configure Polycom ip300 for presence but when status
change notify is no sent to subscriber !?

Why ?

Regards
Harry 






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[Asterisk-Users] Speex codec problems

2005-11-07 Thread Branko Samardzic

I am trying to tweak my Asterisk servers to talk to each other using Speex
codec.
I downloaded and installed speex and speex devel libraries, recompiled
asterisk (including make clean), did set up speex codec as only one allowed
on both sides. Sounds enough.
However, conversations are not Speex encoded!!! It is codec 64 (16 bit
Signed Linear PCM) all the time.
Any clue as to why Asterisk don't want to kick in Speex into play?
BTW One asterisk (initiator) is HEAD version, another is asterisk-1.0.9.

Any help is wappreciated.

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Re: [Asterisk-Users] TDM400 FXO vs FXS Interrupt performance

2005-11-07 Thread Andrew Kohlsmith

On Monday 07 November 2005 12:57, George Gardiner wrote:
> Most modern installations/buildings are wired with RJ45, as are the patch
> panels.  RJ12 is a real pain - I had to chop up patch leads and put RJ12
> sockets on the end.  Very messy and a waste of time.   

We just moved in to a new building.  While you're right in the sense that 
there's cat5 and rj45 everywhere, *every* phone port is RJ11.  I've never 
seen it otherwise.

Up in the equipment room the telco is all terminated to BIX, and there are 
special BIX strips that have BIX on the back and 12 (I think) RJ11 on the 
front.  There are also similar BIX strips that do 6 RJ45 on the front but our 
data termination is all done on 19" patch panels with BIX on the back and 24 
(I think) RJ45 on the front.

-A.
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[Asterisk-Users] AGI environment dump callerid

2005-11-07 Thread bbench

Hi,
Since * 1.2-beta1 (incl CVS HEAD) there is a change in the
callerid's output to STDERR when an AGI environment
dump is requested:

Asterisk CVS HEAD built by root @ chick on a i686 running
Linux on 2005-11-06 16:35:14 UTC
AGI Environment Dump:
 -- accountcode =
 -- callerid = 1234689
 -- calleridname = Callee Name
 -- callingani2 = 0
 -- callingpres = 0
 -- callingtns = 0
 -- callington = 0
 -- channel = SIP/22-f55e
 -- context = default
 -- dnid = 19147858756
 -- enhanced = 0.0
 -- extension = 19147858756
 -- language = en
 -- priority = 1
 -- rdnis = unknown
 -- request = dump.agi
 -- type = SIP
 -- uniqueid = 1131381756.13

but ... Connected to Asterisk 1.0.9 currently running on dog
(pid = 28360)
AGI Environment Dump:
 -- accountcode =
 -- callerid = "Callee Name" <1234689>
 -- channel = SIP/22-9351
 -- context = default
 -- dnid = 19147858756
 -- enhanced = 0.0
 -- extension = 19147858756
 -- language = en
 -- priority = 1
 -- rdnis = unknown
 -- request = dump.agi
 -- type = SIP
 -- uniqueid = 1131381457.0

Thus my question was "which is the future-to-be" callerid
format?
1.  -- callerid = 1234689
 -- calleridname = Callee Name
OR
2. -- callerid = "Callee Name" <1234689>
Nothing wrong with that in general since clid, as
${CDR(clid)}, is still being written correctly in 1.0.7,
1.0.9,
1.2-beta1&2 and CVS HEAD in the usual cdr database/table,
and in any custom table through
$dbh->quote($callerid).

However, since * 1.2-beta1 (incl CVS HEAD), when
AGI(perl) script try $callerid=$input{callerid} it results
to $dbh->quote($callerid) "calleridnum"(by
default it appears eq to "callerid"), only.

/* Obviously, because in res_agi.c "$Revision: 1.53 $":
fdprintf(fd, "agi_callerid: %s\n", chan->cid.cid_num ?
chan->cid.cid_num : "unknown");
fdprintf(fd, "agi_calleridname: %s\n", chan->cid.cid_name ?
chan->cid.cid_name : "unknown"); */

Changing to "$callerid=$input{calleridname}" is inserted as
requested.

Trying to group both callerid attributes results in an empty
string.

Playing with the dilaplan yet damages ${CDR(clid)}
record.

Any thoughts?
benchev





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[Asterisk-Users] CentOS vs. Vanilla Kernel

2005-11-07 Thread Julian Lyndon-Smith


HW: HP DL360 1GB Ram Single 3GHz Pentium (with Hyper-threading turned off).
OS: CentOS 4.2
Dual Embedded NIC enabled
USB disabled
serial disabled
printer disabled
2x73GB SCSI in HW Raid 1

What is the opinion of this fine list  - should I use the default CentOS 
kernel (2.6.9-22.0.1.EL) or download from kernel.org the latest stable 
(2.6.14)


Anyone got any clues / hints / tips on what should go into the kernel ?

All views and comments appreciated :)

Julian.

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Re: [Asterisk-Users] TDM400 FXO vs FXS Interrupt performance

2005-11-07 Thread George Gardiner

 
Most modern installations/buildings are wired with RJ45, as are the patch panels.  RJ12 is a real pain - I had to chop up patch leads and put RJ12 sockets on the end.  Very messy and a waste of time.  
 
 On Sun, 6 Nov 2005 22:04:48 -0500, Andrew Kohlsmith wrote:> On Sunday 06 November 2005 21:46, [EMAIL PROTECTED] wrote:>>> I was pretty unhappy to see that the new cards had RJ12 sockets ->> you can put RJ12 into RJ45, but not the other way round... You've gotta be shitting me.>> Why on earth do you want RJ45 jacks for POTS connections?  Sure it> fits but it's a loose fit to start and you get absolutely zero> advantages unless you count being able to make a screwy cable a> good thing.  :-)>



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Re: [Asterisk-Users] sill looking for a provider

2005-11-07 Thread Dinesh Nair




On 11/06/05 02:31 Dustin Goodwin said the following:
Of course it's hard for me to see the return route with 
traceroute. I assume the return path from their host takes on some 
bizarre route that adds a lot of latency. 


try a traceroute with lft. lft gives you the different AS/BGP routers your 
packet will pass thru, and is a good tool to isolate latency problems.


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Re: [Asterisk-Users] Help with dialplan to allow breakout to DISA

2005-11-07 Thread Rusty Dekema

I do it this way: 

exten => *, 1, Authenticate(PASSWORD)
exten => *, 2, DISA(no-password|DESTINATION_CONTEXT)
exten => *, 3, Hangup

It seems to work fine...

-Rusty

On 11/7/05, Frank Tarczynski <[EMAIL PROTECTED]> wrote:
I'm trying to set-up a dialplan for incoming calls that allows a breakoutby pressing something like "*".  Users would then be able to get an insidedial tone for voicemail, outgoing calls, etc.I've been struggling with Waitexten(), Disa() in the dialplan but not
having much luck.Are there any good documents out there to assist me in this?Frank___--Bandwidth and Colocation sponsored by 
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[Asterisk-Users] Help with dialplan to allow breakout to DISA

2005-11-07 Thread Frank Tarczynski

I'm trying to set-up a dialplan for incoming calls that allows a breakout
by pressing something like "*".  Users would then be able to get an inside
dial tone for voicemail, outgoing calls, etc.

I've been struggling with Waitexten(), Disa() in the dialplan but not
having much luck.

Are there any good documents out there to assist me in this?

Frank

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Re: [Asterisk-Users] References?

2005-11-07 Thread Chad Scott


Hrm.  Perhaps I should have actually responded off-list...  DOH! :D

On Nov 7, 2005, at 9:11 AM, Chad Scott wrote:


Matt,

Sorry for the response off-list...

Would you be willing to talk to the "powers that be" for about 30  
minutes about your experiences with Asterisk?  I don't know what  
questions they're planning to ask, but they're likely to be  
centered around reliability and supportability as those are their  
major paranoia points.


-Chad

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Re: [Asterisk-Users] References?

2005-11-07 Thread Chad Scott


Matt,

Sorry for the response off-list...

Would you be willing to talk to the "powers that be" for about 30  
minutes about your experiences with Asterisk?  I don't know what  
questions they're planning to ask, but they're likely to be centered  
around reliability and supportability as those are their major  
paranoia points.


-Chad

On Nov 3, 2005, at 1:08 PM, Matt wrote:


I can not say that we are using it for a call center, as we use a
NorHell switch for that.. but we will be migrating to Asterisk.
However, we do use it to provide VoIP to all of our customers, and
even customers on other broadband networks.

On 11/3/05, Chad Scott <[EMAIL PROTECTED]> wrote:

All,

I've been pushing hard for the use of Asterisk for the corporate
phone solution at the company I work for.  Unfortunately, this
decision is completely out of my hands, although I've been applying
gentle influence and pressure where I can.

The management for this project would like reference accounts that
utilize Asterisk for their telephony solution and are happy with it.
Ideally, the reference accounts would be around 500 seats in size and
have some sort of call center and/or outbound sales calling.

Anyone want to volunteer for this?

If I can get Asterisk in here it would be HUGE but this is currently
standing in my way.

I know there *must* be installations out there this size and larger,
I just don't know who they are... help me out!

Thanks,
-Chad
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Re: [Asterisk-Users] Change Asterisk User

2005-11-07 Thread Tzafrir Cohen

On Mon, Nov 07, 2005 at 05:27:00PM +0100, Amaury BOSSE wrote:
> Hi all,
> 
> I would like to start asterisk with a different user than "asterisk" in
> order to use the same than my apache server.
> 

Hmmm, you basically need to run apache's user to the Asterisk group.

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Re: [Asterisk-Users] bristuff-0.2.0-RC8n problems and kernel panic

2005-11-07 Thread Tzafrir Cohen

On Mon, Nov 07, 2005 at 03:43:03PM +0100, gincantalupo wrote:
> Hi,
> 
> I had some problems to with a quadBRI with a 2.6 kernel debian distro.
> Have you tried to insmod the zaptel.ko module instead of modprobing?
> It worked for me, hope it will work for you too.
> 
> Giorgio Incantalupo

Could you please give more details?

One thing you should try to do is remove the automatic run of ztcfg at
module load time. Practically: rem-out all the lines in
/etc/modprobe.d/zaptel . 

There is some black-magic claim that if you un ztcfg more than once it
may cause a problem to a configured zaphfc module.

Don't forget to run ztcfg manually (or in an init.d script) later.

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Re: [Asterisk-Users] Change Asterisk User

2005-11-07 Thread Paul

If you give us more info it is easier to help. For example, if you are 
using a standard debian sarge setup I could help you and be sure to give 
you the right advice.


However you might want to think carefully about this type of change. 
There are other approaches such as setting ownership and permissions for 
files and directories the webserver needs access to.


Amaury BOSSE wrote:


Hi all,

I would like to start asterisk with a different user than “asterisk” 
in order to use the same than my apache server.


I have tried to change it in /etc/init.d/asterisk but when I change 
USER, asterisk doesn’t start.


Has someone already start asterisk under other user that “asterisk”?

Thanks

Amaury



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Re: [Asterisk-Users] compiling problems

2005-11-07 Thread Elio Rojano





Some problems happened with precompiled kernels. 
If you compile your own vanilla kernel, I'm sure that you haven't this
issues.

Remember, if you use 2.6 kernel, you can need udev and hotplug systems
to better performance.

I allways use Debian with vanilla kernel that I compile, and I haven't
problems neither 2.4 nor 2.6 kernels on single or dual procesors (32 or
64bits)

I hope that it helps you.




FaberK wrote:

  The problem is the 2.6. I know that there is compability also with
that kernel, but in my small experience, I've got not these problems
with 2.4.
Now, I've got to migrate Asterisk into a Dual Xeon 3.0 4Gb RAM.
What distro would you use?

Until now, I've tested CentOS 3.4 Server with no problem, but not on
this kind of server.
With Fedora 3, too many problems, concerning the kernel 2.6.

Suggestions?

Thanks

2005/11/6, Tzafrir Cohen <[EMAIL PROTECTED]>:
  
  
On Sat, Nov 05, 2005 at 07:29:18PM +0100, FaberK wrote:


  Fedora Core 3
kernel-0-2.6.9-1.667 and kernel-2.6.12-1.1380 (same results)
Sangoma 102
Concerning udev, I've read that it uses hotplug and if I'm not wrong,
I remember that zaptel got conflicts with hotplug. But maybe I'm
confusing (terrible headache!)
Thanks a lot!
  

zaptel should not conflict with hotplug if the specific hardware driver
module is well-written (e.g: declares PCI IDs it will identify). This
will mean that hotplug will try using it automatically.

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Re: [Asterisk-Users] Change Asterisk User

2005-11-07 Thread Jason Pyeron


there is a lot more to changing the user than just su'ing

you need to change the permissions on a lot of files too.


On Mon, 7 Nov 2005, Amaury BOSSE wrote:


Hi all,

I would like to start asterisk with a different user than "asterisk" in
order to use the same than my apache server.



I have tried to change it in /etc/init.d/asterisk but when I change USER,
asterisk doesn't start.



Has someone already start asterisk under other user that "asterisk"?



Thanks



Amaury




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[Asterisk-Users] Change Asterisk User

2005-11-07 Thread Amaury BOSSE









Hi all,

I would like to start asterisk with a different user
than “asterisk” in order to use the same than my apache server.

 

I have tried to change it in /etc/init.d/asterisk but
when I change USER, asterisk doesn’t start.

 

Has someone already start asterisk under other user
that “asterisk”?

 

Thanks

 

Amaury






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Re: [Asterisk-Users] asterisks talking to asterisks

 Yes. Most certainly. Take a look at IAX (Inter Asterisk eXchange)
protocol to enable this functionality for you with minimal impact on
your firewall/NAT setups.

On 11/6/05, Jason Brashear <[EMAIL PROTECTED]> wrote:
> I have a request. I have a server in Texas
> And one in NJ.
> Is it possible for the system in Texas to log into the system in NJ so that
> Extensions can call each other?
> -J
>
>
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Re: [Asterisk-Users] Asterisk 1.2beta2 and UIP200

Ok. That fixed it, but why? It works just fine in 1.0.9 and 1.2b1.  
Very strange.


Anyway, thanks.

- Waldo

On Nov 7, 2005, at 10:57 AM, C F wrote:

The unreachable is the problem. Try adding a qualify=no to that sip  
entry.


On 11/7/05, Waldo Rubinstein <[EMAIL PROTECTED]> wrote:

Additionally:

*CLI> sip show peer 100074

  * Name   : 100074
  Secret   : 
  MD5Secret: 
  Context  : qa
  Subscr.Cont. : 
  Language : en
  AMA flags: Unknown
  CallingPres  : Presentation Allowed, Not Screened
  Callgroup:
  Pickupgroup  :
  Mailbox  : [EMAIL PROTECTED]
  VM Extension : asterisk
  LastMsgsSent : 0
  Call limit   : 0
  Dynamic  : Yes
  Callerid : "Waldo Rubinstein" <211>
  Expire   : 11077
  Insecure : no
  Nat  : No
  ACL  : No
  CanReinvite  : No
  PromiscRedir : No
  User=Phone   : No
  Trust RPID   : No
  Send RPID: No
  DTMFmode : rfc2833
  LastMsg  : 0
  ToHost   :
  Addr->IP : 10.0.10.236 Port 5060
  Defaddr->IP  : 0.0.0.0 Port 5060
  Def. Username: 100074
  SIP Options  : (none)
  Codecs   : 0x6 (gsm|ulaw)
  Codec Order  : (ulaw,gsm)
  Status   : UNREACHABLE
  Useragent: Uniden SIP Phone p2 Ver BS4.63
  Reg. Contact : sip:[EMAIL PROTECTED]:5060

Thanks,
Waldo

On Nov 6, 2005, at 11:11 PM, C F wrote:


can you post the sip.conf for that uip200?

On 11/6/05, Waldo Rubinstein <[EMAIL PROTECTED]> wrote:

When I dial the extension, I get this:

 -- Executing Dial("IAX2/gateway0-16386", "SIP/100074|20")  
in new

stack
   == Everyone is busy/congested at this time (1:0/0/1)


When I do a sip show peer 100074, everything it shows matches the
results of executing the same sip show peer on * 1.0.9 and 1.2b1,
except:

   Status   : UNREACHABLE

However, I can make any type of calls from them phone. I can  
ping the

phone from the * server. It's just that * 1.2b2 can't reach it, for
some reason.

Thanks,
Waldo

On Nov 6, 2005, at 1:37 PM, C F wrote:


Whats the exact CLI output you are getting when calling that
extension?

On 11/6/05, Waldo Rubinstein <[EMAIL PROTECTED]> wrote:

Nope. It isn't active. I even factory reseted the phone but
still the
same. One more piece of information: it works just fine in  
1.2b1. I

beginning to think it could be a bug in 1.2b2.

Any other ideas/suggestions?

Thanks,
Waldo

On Nov 5, 2005, at 9:10 PM, C F wrote:

You sure that the DND (Do Not Disturb) button is not active  
on the

UIP200?

On 11/4/05, Waldo Rubinstein <[EMAIL PROTECTED]> wrote:
I am running * 1.2b2 with some UIP200 phones and a bunch of  
X-Pro

phones.

All phones register fine with * and I can place outbound calls
with
no problem.

I can call from the X-Pro to any other X-Pro. I can call from
UIP200
to any other X-Pro. However, the UIP200 cannot receive calls.
Every
time I call the UIP200, the CLI says Everyone is Busy/
Congested and
sends the call to voicemail.

Everything is in the same network. I have in sip.conf
localnet=10.0.10.0/24

and in each UIP200 sip profile
nat=never

What's wrong?

I have the same configuration in * 1.0.9 and it works just  
fine.

Could the SIP protocol be broken in 1.2b2?

Thanks,
Waldo

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[Asterisk-Users] asterisks talking to asterisks

2005-11-07 Thread Jason Brashear

I have a request. I have a server in Texas
And one in NJ.
Is it possible for the system in Texas to log into the system in NJ so that 
Extensions can call each other?
-J


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Re: [Asterisk-Users] Asterisk 1.2beta2 and UIP200

The unreachable is the problem. Try adding a qualify=no to that sip entry.

On 11/7/05, Waldo Rubinstein <[EMAIL PROTECTED]> wrote:
> Additionally:
>
> *CLI> sip show peer 100074
>
>   * Name   : 100074
>   Secret   : 
>   MD5Secret: 
>   Context  : qa
>   Subscr.Cont. : 
>   Language : en
>   AMA flags: Unknown
>   CallingPres  : Presentation Allowed, Not Screened
>   Callgroup:
>   Pickupgroup  :
>   Mailbox  : [EMAIL PROTECTED]
>   VM Extension : asterisk
>   LastMsgsSent : 0
>   Call limit   : 0
>   Dynamic  : Yes
>   Callerid : "Waldo Rubinstein" <211>
>   Expire   : 11077
>   Insecure : no
>   Nat  : No
>   ACL  : No
>   CanReinvite  : No
>   PromiscRedir : No
>   User=Phone   : No
>   Trust RPID   : No
>   Send RPID: No
>   DTMFmode : rfc2833
>   LastMsg  : 0
>   ToHost   :
>   Addr->IP : 10.0.10.236 Port 5060
>   Defaddr->IP  : 0.0.0.0 Port 5060
>   Def. Username: 100074
>   SIP Options  : (none)
>   Codecs   : 0x6 (gsm|ulaw)
>   Codec Order  : (ulaw,gsm)
>   Status   : UNREACHABLE
>   Useragent: Uniden SIP Phone p2 Ver BS4.63
>   Reg. Contact : sip:[EMAIL PROTECTED]:5060
>
> Thanks,
> Waldo
>
> On Nov 6, 2005, at 11:11 PM, C F wrote:
>
> > can you post the sip.conf for that uip200?
> >
> > On 11/6/05, Waldo Rubinstein <[EMAIL PROTECTED]> wrote:
> >> When I dial the extension, I get this:
> >>
> >>  -- Executing Dial("IAX2/gateway0-16386", "SIP/100074|20") in new
> >> stack
> >>== Everyone is busy/congested at this time (1:0/0/1)
> >>
> >>
> >> When I do a sip show peer 100074, everything it shows matches the
> >> results of executing the same sip show peer on * 1.0.9 and 1.2b1,
> >> except:
> >>
> >>Status   : UNREACHABLE
> >>
> >> However, I can make any type of calls from them phone. I can ping the
> >> phone from the * server. It's just that * 1.2b2 can't reach it, for
> >> some reason.
> >>
> >> Thanks,
> >> Waldo
> >>
> >> On Nov 6, 2005, at 1:37 PM, C F wrote:
> >>
> >>> Whats the exact CLI output you are getting when calling that
> >>> extension?
> >>>
> >>> On 11/6/05, Waldo Rubinstein <[EMAIL PROTECTED]> wrote:
>  Nope. It isn't active. I even factory reseted the phone but
>  still the
>  same. One more piece of information: it works just fine in 1.2b1. I
>  beginning to think it could be a bug in 1.2b2.
> 
>  Any other ideas/suggestions?
> 
>  Thanks,
>  Waldo
> 
>  On Nov 5, 2005, at 9:10 PM, C F wrote:
> 
> > You sure that the DND (Do Not Disturb) button is not active on the
> > UIP200?
> >
> > On 11/4/05, Waldo Rubinstein <[EMAIL PROTECTED]> wrote:
> >> I am running * 1.2b2 with some UIP200 phones and a bunch of X-Pro
> >> phones.
> >>
> >> All phones register fine with * and I can place outbound calls
> >> with
> >> no problem.
> >>
> >> I can call from the X-Pro to any other X-Pro. I can call from
> >> UIP200
> >> to any other X-Pro. However, the UIP200 cannot receive calls.
> >> Every
> >> time I call the UIP200, the CLI says Everyone is Busy/
> >> Congested and
> >> sends the call to voicemail.
> >>
> >> Everything is in the same network. I have in sip.conf
> >> localnet=10.0.10.0/24
> >>
> >> and in each UIP200 sip profile
> >> nat=never
> >>
> >> What's wrong?
> >>
> >> I have the same configuration in * 1.0.9 and it works just fine.
> >> Could the SIP protocol be broken in 1.2b2?
> >>
> >> Thanks,
> >> Waldo
> >>
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> >> http://lists.digium.com/mailman/listinfo/asterisk-users
> >> To UNSUBSCRIBE or update options visit:
> >>http://lists.digium.com/mailman/listinfo/asterisk-users
> >>
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> >>> http://lists.digium.com/mailman/l

Re: [Asterisk-Users] Dropping last digit from phone number

Here is what I do:
${EXTEN:0:$[${LEN(${EXTEN})} - 1]}
that should give you for the following
exten => 123456789,1,Noop(${EXTEN:0:$[${LEN(${EXTEN})} - 1]})
12345678

Hope this helps.

On 11/7/05, Bartosz Piec <[EMAIL PROTECTED]> wrote:
> Erik napisał(a):
> > exten => _XX*,1,NoOp(${EXTEN:0:-1})
>
> exten => _XX*,1,NoOp(${EXTEN:0:2})
> :)
>
> It works, thanks.
>
> --
> Best regards,
> Bartosz Piec
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Re: Re: [Asterisk-Users] call from asterisk to SIP cisco 5300

2005-11-07 Thread Ivan Vershigora


sorry, i didnt write i have voip  peer

so i have sloved thy problem, nubder like

#00#7091222
*00*7091222
*777
doesnt work
Cisco says
dpMatchPeersMoreArg: Match Dest. pattern; called ()

and when i tries to dial *777*777
it says

dpMatchPeersMoreArg: Match Dest. pattern; called (777)

But I cant understand why CISCO cant understand this "MAGIC" # and * :)


I think you should set dial-peer voice 21 voip with incoming called number
#00#..\* too, this catch this call and the dial peer 22 send it.

Adam

Cytowanie Ivan Vershigora <[EMAIL PROTECTED]>:


i dial on my phone to to 8091222
and convert it on asterisk to #00#7091222
But Cisco says 404

cisco peer=
!
dial-peer voice 22 pots
huntstop
preference 5
destination-pattern #00#..\*
translate-outgoing calling 1
direct-inward-dial
port 0:D
prefix 810
!


peer in sip.conf==
[krdvox]
context=from-sip
type=peer
host=123.123.123.123
canreinvite=yes
dtmfmode=inband


extensions.conf==
exten => _.,1,SetCallerID("861273" <8612731107>[|a])
exten => _.,2,Dial(SIP/#00#7${EXTEN:[EMAIL PROTECTED],60)
exten => _.,3,Congestion


Asterisk says===
-- Executing Dial("SIP/201-2966", "SIP/[EMAIL PROTECTED]|60") in
new stack
   -- Called [EMAIL PROTECTED]
   -- Got SIP response 404 "Not Found" back from XXX.XXX.XXX.XXX
   -- SIP/krdvox-3910 is circuit-busy
 == Everyone is busy/congested at this time
===

==CISCO debug ccsip ===
Nov  3 16:10:03.516: Received:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK6697eb34
From: "861273" ;tag=as74db268c
To: 
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: CSCO/6
Date: Thu, 03 Nov 2005 13:10:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 235

.

Nov  3 16:10:03.524: MatchNextPeer: Peer 999 matched
Nov  3 16:10:03.524: Using Voice Class Codec, tag=1

.

Disconnect Cause (SIP)   : 404

===
Nov  3 16:10:03.524: MatchNextPeer: Peer 999 matched

Peer 999- wrong one !!!
why he cant find dial-peer voice 22



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Pozdrawiam,
Adam Rybak






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Re: [Asterisk-Users] Asterisk 1.2beta2 and UIP200