Re: [Asterisk-Users] R2/MFC Mexico FREE calls to test chan_unicall

[Sam Njenga]

Hi
Am setting up * with R2/MfC support but am 90% done. I seem to be missing
something in my setup. Can you tell me what Linux distribution and the
packages you have used to complete your setup to a working level ?

/Sam

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

Sent: Thursday, January 13, 2005 1:22 AM
Subject: [Asterisk-Users] R2/MFC Mexico FREE calls to test chan_unicall


> Hi guys, I have one E1 with 30 channels in Mexico City, I guess that if
> i can fill this 30 channels with REAL traffic for 2 or 3 days I can
> find new bugs on chan_unicall or I can see how stable it can be. Im
> using R2/MFC with chan_unicall the patch that Steve Underwood wrote.
>
> I will let anyone make FREE LOCAL calls to Mexico City till saturday or
> maybe until monday to see how stable this can be with REAL traffic. Add
> this to your extensions.conf only gsm as a codec is going to be
> permitted.
>
> exten =>
> _,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED],90,Tt)
>
> --
> Saludos,
> Miguel Cavazos
>
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Re: [Asterisk-Users] snom220

[Altus Snyman]

Sorry
It works
Just had to reboot the phone

On Thu, 2005-01-13 at 08:40, Altus Snyman wrote:
> Good day all
> I got my snom 220 phone so that it displays on the buttons if someone is
> calling that extension
> I just added "exten => 403,hint,SIP/403" in my dialplan
> But
> These lights only comes on if someone calls that extension,not if that
> extension is busy are a call is made from that extension
> Can this be done?
> Please Help
> Altus
> 
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Re: [Asterisk-Users] Grandstream Bugetone 101 & mwi

[Ronald Wiplinger]

Paul Fielding wrote:
- Original Message - From: "Ronald Wiplinger" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Sent: Wednesday, January 12, 2005 11:13 PM
Subject: [Asterisk-Users] Grandstream Bugetone 101 & mwi


I tried to use message waiting indicator, by "Subscribe for MWI" in 
the web menu of the phone.

However, it does not light up / flash, even if a voice mail is waiting.
Where is the switch to turn it to?
bye
Ronald

you need to set 'mailbox=extention' in the sip phone's context in 
sip.conf

Paul
I have set that !!!
bye
Ronald
--
Ronald Wiplinger  (CEO of ELMIT)
http://www.elmit.com+886 (0) 939--77-55-16  or FWD 511208
- I'm a SpamCon Foundation Member, #694, Verify it at http://www.spamcon.org
PS: Spam prevention!
Our system is protected with a spam prevention program. 
If you send us an e-mail, our system will send you a confirmation message back. Just reply to this confirmation message please. 
After receiving this confirmation message, our system will send the hold message (one) and all future messages (after the received confirmation message) to me without asking you again.

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[Asterisk-Users] IAXy setup

[Ronald Wiplinger]

I have provisioned with iaxy.conf:
;
; IAXY Provisioning description
;
dhcp
codec: ulaw
server: 61.220.xx.xx
user: aaabbb
pass: cccddd
register
iax.conf:
=
[623]   ; IAXy
type=friend
host=dynamic
accountcode=aaabbb
disallow=all
allow=ulaw
secret=cccddd
callerid="IAXy at ELMIT" <623>
trunk=no
extensions.conf:

[inhouse]

PHONE_623=IAX2/aaabbb:[EMAIL PROTECTED]/623 ; 3 IAXy adapter
exten => 623,1,Dial(${PHONE_623},60,Ttrm)
exten => 623,2,Voicemail,u623
exten => 623,103,Voicemail,b623
[default]
...
include => inhouse   
What do I miss?
bye
Ronald
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[Asterisk-Users] snom220

[Altus Snyman]

Good day all
I got my snom 220 phone so that it displays on the buttons if someone is
calling that extension
I just added "exten => 403,hint,SIP/403" in my dialplan
But
These lights only comes on if someone calls that extension,not if that
extension is busy are a call is made from that extension
Can this be done?
Please Help
Altus

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[Asterisk-Users] ASTCC dimensioning

[Atif Rasheed]

hello there,
any one who used ASTCC in a real enviroment, or has successfully handled 
above 1k simultanous calls. need some evalution of ASTCC. if any one has 
such an experience please share it with the rest

thank you
Atif
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Re: [Asterisk-Users] Grandstream Bugetone 101 & mwi

[Paul Fielding]

you need to set 'mailbox=extention' in the sip phone's context in 
sip.conf

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

Sent: Wednesday, January 12, 2005 11:13 PM
Subject: [Asterisk-Users] Grandstream Bugetone 101 & mwi


I tried to use message waiting indicator, by "Subscribe for MWI" in the web 
menu of the phone.

However, it does not light up / flash, even if a voice mail is waiting.
Where is the switch to turn it to?
bye
Ronald
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Re: [Asterisk-Users] Looking for a wireless phone... wifi or traditional wireless ?

[Paul]

I saw that news but couldn't seem to find any price/availability info
yet.

I wonder how well they will work if everyone in the home or office has
one. I always liked the old-fashioned approach where switches were used
to explicitly select a channel. My guess would be that multiple wifi
sip phones will coexist better.

I have heard credible reports from my own customers of cordless phones
and wireless lan clashes. It was credible because I was on the phone
talking with them. When they switched to their cordless phone the
laptop PC would lose it's network connection. I also had them call me
using the cordless phone while the nearby laptop was turned off. When
they booted it up and opened a web browser I could hear the
interference at my end and sometimes we lost all sound. When the
customer switched back to his old 900 mhz cordless phone we didn't have
the problem anymore.

For those reasons I am desperately hoping that wifi sip phone prices
will rapidly fall.

Paul

James H. Thompson wrote:

  
  
  
  Uniden and Vtech both just announced cordless
phones with SIP ATAs built into the base station.
  You get better range and battery life compared to
a WiFi phone.
   
  Jim
   
  James H. Thompson
  [EMAIL PROTECTED]
  
  
-
Original Message - 
From:
Kim Lux 
To:
Asterisk Users Mailing
List - Non-Commercial Discussion 
Sent:
Wednesday, January 12, 2005 5:49 PM
Subject:
Re: [Asterisk-Users] Looking for a wireless phone... wifi or
traditional wireless ?



An unflattering zyxel review:

http://slacker.com/~nugget/asterisk3.php

I can't help but think my questions are out of place on this list... I'm
asking questions about SIP phones and everyone else is talking about
asterisk.  Sorry. 


On Wed, 2005-01-12 at 20:08 -0700, Kim Lux wrote:
> My wife wants a cordless phone for around the house.  We are going
to be
> using VOIP exclusively very shortly.  Our current cordless phone
is aged
> and on the verge of replacement.  The other phone we are going to
use is
> a SIP Budgetone.
> 
> Should I buy a SIP to POTS converter and a new cordless phone or a
wifi
> SIP phone ?   
> 
> Is anyone using the Pulver WiSIP phone ?  Any comments ?
> 
> How about the zyxel ?
> 
> Thanks
> 
-- 
Kim Lux,  Diesel Research Inc.


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Re: [Asterisk-Users] EuroISDN BRI 2 or 4 wires?

[Thomas Niesel]

On Wed, Jan 12, 2005 at 11:06:18PM +0100, Remco Barende wrote:
> Hi List!
> 
> Have a weird problem with ISDN in The Netherlands.
> 
> The line that is coming in from the telco is 2 wires. The line is 
> connected to an NT1 using the middle pair of a UTP connector. So far sop 
> good.

The incomming 2 wires are called UK0 here in Germany and they are connected to 
the
NT1 or NTBA.

> 
> However, the outgoing ports on the NT1, should they be wired with 2 wires 
> or 4 (2 pairs)?

The outgoing Line from the NT1/NTBA is a 4 wire (2pairs) bus called S0.

> 
> If I use a 4 wire cable and stick it in the NT1 I get 0 voltage on the 
> middle 2 wires, but i get approx 4 volts on the left 2 wires and another 
> 40 volts on the outer (right) 2 wires. Is it correct that 2 pairs should 
> be used to connect ISDN devices?

Yepp, if you can read german have a look at www.kabelfaq.de . There is a well 
explained
overview about isdn.

> 
> Sorry for this slightly offtopic issue. The house is fully 
> wired for one pair which means I may have a problem if i need 2 pairs.
> 
> Thanks!
> Remco
> 
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-- 
Tho/\/\as
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[Asterisk-Users] Grandstream Bugetone 101 & mwi

[Ronald Wiplinger]

I tried to use message waiting indicator, by "Subscribe for MWI" in the 
web menu of the phone.

However, it does not light up / flash, even if a voice mail is waiting.
Where is the switch to turn it to?
bye
Ronald
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Re: [Asterisk-Users] Is this a $50 wifi or wireless USB VOIP phone ?

[pjn]

Wired USB handset - quite plasticky looking/feeling really (no display)!
AND it is supplied with a software registration process which is a PITA
because each time you move the handset to a different computer you need
to unregister it from the old PC and re-register on the new PC.
I complained to the vendor that this critical piece of information was
missing from their website and was promptly ignored by them.
When I didn't faithfully follow the uninstall process to de-register the
software they did only take 12 hours to reactivate my software key but
shouldn't be there in the first place or at least should be disclosed
prior to purchase.
I have the skype version which supports a skype link and also a SIP link.
Have one sitting in front of me now. Not buying anymore!
..pjn
- Original Message - 
From: "Kim Lux" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Sent: Thursday, January 13, 2005 10:09 AM
Subject: [Asterisk-Users] Is this a $50 wifi or wireless USB VOIP phone ?


http://www.pcphoneline.com/skype
"The VPT1000 is NOT a simple last generation USB phone audio device but
is rather a next generation integrated gateway and USB phoneset with
simultaneous dual mode Skype and SIP calling support.  Skype is not
forecast to have "SkypeIn" available until June 2005 but you can have
the capability now via its built in SIP capabilities."
Is this a wireless USB phone ?  Does it support SIP and could it be used
to connect to any SIP server ?
Does anyone have experience with these ?
Thanks.
--
Kim Lux,  Diesel Research Inc.

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Re: [Asterisk-Users] Looking for a wireless phone... wifi ortraditional wireless ?

[Paul Fielding]

I think some people are missing the point.   You can't throw your cordless 
phone in your pocket, go to your office, hotel or buddie's house, turn it on 
and get a signal. You can with a WiFi phone, however


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

Sent: Wednesday, January 12, 2005 10:01 PM
Subject: Re: [Asterisk-Users] Looking for a wireless phone... wifi 
ortraditional wireless ?


I don't know why people keep making the statement about range with wifi
versus a cordless phone.  I can easily get a good wifi signal when I'm
over at my neighbors but can't get reception with our cordless (2.4GHz)
phone.  (Both receivers are at home...)  It seems to me that the wifi
range is at least as good as the phone range if not better.
Thanks for the tip on the SIP phones.
On Wed, 2005-01-12 at 18:53 -1000, James H. Thompson wrote:
Uniden and Vtech both just announced cordless phones with SIP ATAs
built into the base station.
You get better range and battery life compared to a WiFi phone.
Jim
James H. Thompson
[EMAIL PROTECTED]
- Original Message - 
From: Kim Lux
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Wednesday, January 12, 2005 5:49 PM
Subject: Re: [Asterisk-Users] Looking for a wireless phone...
wifi or traditional wireless ?


An unflattering zyxel review:
http://slacker.com/~nugget/asterisk3.php
I can't help but think my questions are out of place on this
list... I'm
asking questions about SIP phones and everyone else is talking
about
asterisk.  Sorry.
On Wed, 2005-01-12 at 20:08 -0700, Kim Lux wrote:
> My wife wants a cordless phone for around the house.  We are
going to be
> using VOIP exclusively very shortly.  Our current cordless
phone is aged
> and on the verge of replacement.  The other phone we are
going to use is
> a SIP Budgetone.
>
> Should I buy a SIP to POTS converter and a new cordless
phone or a wifi
> SIP phone ?
>
> Is anyone using the Pulver WiSIP phone ?  Any comments ?
>
> How about the zyxel ?
>
> Thanks
>
-- 
Kim Lux,  Diesel Research Inc.

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--
Kim Lux,  Diesel Research Inc.
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Re: [Asterisk-Users] Looking for a wireless phone... wifi or traditional wireless ?

[Kim Lux]

I don't know why people keep making the statement about range with wifi
versus a cordless phone.  I can easily get a good wifi signal when I'm
over at my neighbors but can't get reception with our cordless (2.4GHz)
phone.  (Both receivers are at home...)  It seems to me that the wifi
range is at least as good as the phone range if not better. 

Thanks for the tip on the SIP phones.   


On Wed, 2005-01-12 at 18:53 -1000, James H. Thompson wrote:
> Uniden and Vtech both just announced cordless phones with SIP ATAs
> built into the base station.
> You get better range and battery life compared to a WiFi phone.
>  
> Jim
>  
> James H. Thompson
> [EMAIL PROTECTED]
> 
> - Original Message - 
> From: Kim Lux 
> To: Asterisk Users Mailing List - Non-Commercial Discussion 
> Sent: Wednesday, January 12, 2005 5:49 PM
> Subject: Re: [Asterisk-Users] Looking for a wireless phone...
> wifi or traditional wireless ?
> 
> 
> 
> An unflattering zyxel review:
> 
> http://slacker.com/~nugget/asterisk3.php
> 
> I can't help but think my questions are out of place on this
> list... I'm
> asking questions about SIP phones and everyone else is talking
> about
> asterisk.  Sorry. 
> 
> 
> On Wed, 2005-01-12 at 20:08 -0700, Kim Lux wrote:
> > My wife wants a cordless phone for around the house.  We are
> going to be
> > using VOIP exclusively very shortly.  Our current cordless
> phone is aged
> > and on the verge of replacement.  The other phone we are
> going to use is
> > a SIP Budgetone.
> > 
> > Should I buy a SIP to POTS converter and a new cordless
> phone or a wifi
> > SIP phone ?   
> > 
> > Is anyone using the Pulver WiSIP phone ?  Any comments ?
> > 
> > How about the zyxel ?
> > 
> > Thanks
> > 
> -- 
> Kim Lux,  Diesel Research Inc.
> 
> 
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-- 
Kim Lux,  Diesel Research Inc.


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RE: [Asterisk-Users] SNOM 190 Configuration with Asterisk

[Colin Anderson]

Assumptions:

-Working DHCP server
-Good LAN, everyone's happy and can see each other
-Working DNS server on LAN
-IP of Asterisk server: 192.168.1.46


sip.conf: (remove comments)

[550] 'extension number
callerid="Joe Blow" <550>
canreinvite=no
context=from-internal 'default AMP context, salt to taste
dtmfmode=rfc8233
host=dynamic
mailbox=550
nat=never 'assumes behind the firewall or port forward SIP thru the firewall
port=5060
secret=123456
type=friend
username=550

In SNOM webconfig (click on "line 1")

Display name: Joe Blow
Account: 550
Password: 123456
Registrar: 192.168.1.46 'Insert IP or FQDN of Asterisk server here
Mailbox: *98 'Default AMP value, you can change in extensions.conf
Outbound proxy: 192.168.1.46 'Insert IP or FQDN of Asterisk server here

Leave all other values as default.

Some tips:

-Make sure you have a working DNS server behind the firewall. 
-Make sure the DHCP server that gives the IP to the SNOM assigns the DNS
setting correctly
-Update the firmware from the shipping defaults (this is what you need the
DNS server for)
-Click on "SIP Trace" in web config. 

A working trace:

Sent to udp:192.168.1.46:5060 at 12/1/2005 21:46:17:500 (279 bytes):

SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.1.46:5060;branch=z9hG4bK6bbcd25b
From: "Asterisk" ;tag=as6d6578bc
To: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 NOTIFY
Content-Length: 0

A bad trace: (indicating bad account or secret password not matching
sip.conf)

Received from udp:192.168.1.46:5060 at 12/1/2005 21:42:42:030 (470 bytes):

SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.246:5060;branch=z9hG4bK-fcvcqu43c9va
From: "Joe Blow" ;tag=3y392m7gy8
To: "Joe Blow" ;tag=as3eae3493
Call-ID: [EMAIL PROTECTED]
CSeq: 362475 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: 
WWW-Authenticate: Digest realm="asterisk", nonce="069b0399"
Content-Length: 0

I guess I was lucky. My first SNOM worked perfectly, first try. I was
suprised. 

HTH.


-Original Message-
From: Ty Carter [mailto:[EMAIL PROTECTED]
Sent: Wednesday, January 12, 2005 6:34 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] SNOM 190 Configuration with Asterisk


Anyone have a suggestion for a configuration example using Asterisk and SNOM
190 SIP phone?

I have read both sets of documentation and for the life of me, I can't get
the phone to register and work.  I can use a IAX softphone and it works
perfectly.  It is just the SIP thing I guess.  Anyone have conf examples
they can share?

Thanks





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Re: [Asterisk-Users] Looking for a wireless phone... wifi or traditional wireless ?

[James H. Thompson]

Uniden and Vtech both just announced cordless phones with SIP 
ATAs built into the base station.
You get better range and battery life compared to a WiFi 
phone.
 
Jim
 
James H. Thompson[EMAIL PROTECTED]

  - Original Message - 
  From: 
  Kim 
  Lux 
  To: Asterisk Users Mailing List - 
  Non-Commercial Discussion 
  Sent: Wednesday, January 12, 2005 5:49 
  PM
  Subject: Re: [Asterisk-Users] Looking for 
  a wireless phone... wifi or traditional wireless ?
  An unflattering zyxel review:http://slacker.com/~nugget/asterisk3.phpI 
  can't help but think my questions are out of place on this list... 
  I'masking questions about SIP phones and everyone else is talking 
  aboutasterisk.  Sorry. On Wed, 2005-01-12 at 20:08 -0700, 
  Kim Lux wrote:> My wife wants a cordless phone for around the 
  house.  We are going to be> using VOIP exclusively very 
  shortly.  Our current cordless phone is aged> and on the verge of 
  replacement.  The other phone we are going to use is> a SIP 
  Budgetone.> > Should I buy a SIP to POTS converter and a new 
  cordless phone or a wifi> SIP phone ?   > > Is 
  anyone using the Pulver WiSIP phone ?  Any comments ?> > 
  How about the zyxel ?> > Thanks> -- Kim 
  Lux,  Diesel Research 
  Inc.___Asterisk-Users 
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Re: [Asterisk-Users] Operator Panels?

[Julien Goodwin]

On Wed, Jan 12, 2005 at 08:07:11AM -0600, Matt Schulte arranged a set of bits 
into the following:
> Ok, we're trying to use Asterisk as a PBX in our office. Our original
> plan was to use a Cisco 7960 with a 7914 attached. Short story is, no
> one updated chan_sccp in a long time and the 7914 is questionable at
> best anyway from what I've heard. We couldn't ever get chan_sccp to
> compile, I went to an older version of Asterisk and that broke some of
> our SIP devices. We tried using a couple soft panels listed on the Wiki,
> the only one easy enough to use was Asternic. And we found also that is
> buggy and doesn't function correctly with the new Asterisk. Any
> Suggestions?

What was your problem with chan_sccp? There's only one small issue I
know of in the code (already fixed, I just haven't committed it to CVS).

Although the biggest issue with using it would be that chan_sccp doesn't
yet have hint support (it's forthcoming once I get my new phone
delivered this week). 

Thanks,
Julien Goodwin 
chan_sccp developer


pgphcS3TLinYe.pgp
Description: PGP signature
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[Asterisk-Users] moh mp3 streaming problem

[Ken Godee]

asterisk v1.0.3, mpg123 v59r, shoutcast server.
When first starting asterisk all is fine, moh/mpg
processes start, can see asterisk client connections on shoutcast 
monitor as well and I've got mp3 streamed music on hold, cool!

After aprx. 32-105 seconds the asterisk client connections close on the 
shoutcast server. The moh/mpg processes are still running, but are now 
just looping a "buffer full?" of previous mp3 streamed music.

asterisk MP3Player works as expected.
mpg123 works fine from console, xmms too, etc.
Moh seems to have some type of time out.
Nothing in logs.
I know there's other people streaming MP3's to moh, is
this happening to you?
I've tried to peek thru the res_musiconhold.c file but
just can't figure it out. Class doesn't seem to
matter, mp3,custom, even httpmp3.
Any ideas how I can keep the MP3 stream open?
(hope I made sense)

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Re: [Asterisk-Users] Looking for a wireless phone... wifi or traditional wireless ?

[Kim Lux]

An unflattering zyxel review:

http://slacker.com/~nugget/asterisk3.php

I can't help but think my questions are out of place on this list... I'm
asking questions about SIP phones and everyone else is talking about
asterisk.  Sorry. 


On Wed, 2005-01-12 at 20:08 -0700, Kim Lux wrote:
> My wife wants a cordless phone for around the house.  We are going to be
> using VOIP exclusively very shortly.  Our current cordless phone is aged
> and on the verge of replacement.  The other phone we are going to use is
> a SIP Budgetone.
> 
> Should I buy a SIP to POTS converter and a new cordless phone or a wifi
> SIP phone ?   
> 
> Is anyone using the Pulver WiSIP phone ?  Any comments ?
> 
> How about the zyxel ?
> 
> Thanks
> 
-- 
Kim Lux,  Diesel Research Inc.


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[Asterisk-Users] pass through mode

[jeffrey johnson]

hi

1  does asterisk do full proxy to all calls,  or it can be configured
to do proxy signal  only ?  if yes,  how to configure to proxy signal
only?

2 G.729 codec license:  do i have to buy license if * doesn't do codec
conversion,  just proxy the calls?


Jeffrey
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RE: [Asterisk-Users] Looking for a wireless phone... wifi ortraditional wireless ?

[Michael Giagnocavo]

My advice, not having used a WiSIP phone, is to use an FXS port and plug a
normal cordless phone in. It'll save you all potential problems, and if you
don't like the phone, you can just plug any other one on the market in. 

In a home environment, I'm not quite sure I see the benefit of wifi. Esp. in
range -- I get much more range out of a normal phone that I do my wifi APs.

-Michael

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kim Lux
Sent: Wednesday, January 12, 2005 9:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Looking for a wireless phone... wifi ortraditional
wireless ?


My wife wants a cordless phone for around the house.  We are going to be
using VOIP exclusively very shortly.  Our current cordless phone is aged
and on the verge of replacement.  The other phone we are going to use is
a SIP Budgetone.

Should I buy a SIP to POTS converter and a new cordless phone or a wifi
SIP phone ?   

Is anyone using the Pulver WiSIP phone ?  Any comments ?

How about the zyxel ?

Thanks

-- 
Kim Lux,  Diesel Research Inc.


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[Asterisk-Users] IAX peering between two Asterisk servers, how?

[Adi Linden]

How do I setup IAX peering between two Asterisk servers? I found a few
examples for the IAX client side that conects to a service provider. But
what does the service provider end look like. I would also like to use md5
authentication.

Adi
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Re: [Asterisk-Users] RE: Polycom IP 500 Dial Issues

[Greg Boehnlein]

On Wed, 12 Jan 2005, Kai-Uwe Jensen wrote:
 
> On Wed, 12 Jan 2005, Paul Rodan wrote:
> 
> >> Yeah, it's a way for numbers to get sent faster, so you don't have to
> wait
> >> for the 3 second timeout before it gets transmitted to Asterisk. It's
> >> similar to the dial-plan in the Sipura devices. 
> >> 
> >> I don't know where it's mentioned in their documentation, maybe it's on
> >> their cd somewhere. I was just thrown a phone and told to figure it out,
> >> thanks to searching through the forum and a little help on the irc
> channel,
> >> I figured it out after a little bit of time. It's extremely customizable
> (so
> >> are the Sipura's).
> >
> > Any idea on docs? I found the following message from the mailing list that
> 
> > I'm going to work with tonight..
> > 
> > http://lists.digium.com/pipermail/asterisk-users/2004-May/045721.html
> 
>  My Polycom admin guide (pg. 115, 4.6.2.1.4.1) says the dialplan digitmap
>  is "compatible with the digit map feature of MGCP described in 2.1.5 of RFC
>  3435". The RFC can be found in many places, for example here:
>  http://www.faqs.org/rfcs/rfc3435.html

For archival/search purposes, I'll re-post the relevant portion of the RFC 
that deals with Digit maps. This looks like it will solve the problem!

2.1.5 Digit Maps

   The Call Agent can ask the gateway to collect digits dialed by the
   user.  This facility is intended to be used with residential gateways
   to collect the numbers that a user dials; it can also be used with

   trunking gateways and access gateways alike, to collect access codes,
   credit card numbers and other numbers requested by call control
   services.

   One procedure is for the gateway to notify the Call Agent of each
   individual dialed digit, as soon as they are dialed.  However, such a
   procedure generates a large number of interactions.  It is preferable
   to accumulate the dialed numbers in a buffer, and to transmit them in
   a single message.

   The problem with this accumulation approach, however, is that it is
   hard for the gateway to predict how many numbers it needs to
   accumulate before transmission.  For example, using the phone on our
   desk, we can dial the following numbers:

--
   |  0 |  Local operator |
   |  00|  Long distance operator |
   |    |  Local extension number |
   |  8xxx  |  Local number   |
   |  #xxx  |  Shortcut to local number at|
   ||  other corporate sites  |
   |  *xx   |  Star services  |
   |  91xx  |  Long distance number   |
   |  9011 + up to 15 digits|  International number   |
--

   The solution to this problem is to have the Call Agent load the
   gateway with a digit map that may correspond to the dial plan.  This
   digit map is expressed using a syntax derived from the Unix system
   command, egrep.  For example, the dial plan described above results
   in the following digit map:

  (0T|00T|[1-7]xxx|8xxx|#xxx|*xx|91xx|9011x.T)

   The formal syntax of the digit map is described by the DigitMap rule
   in the formal syntax description of the protocol (see Appendix A) -
   support for basic digit map letters is REQUIRED while support for
   extension digit map letters is OPTIONAL.  A gateway receiving a digit
   map with an extension digit map letter not supported SHOULD return
   error code 537 (unknown digit map extension).

   A digit map, according to this syntax, is defined either by a (case
   insensitive) "string" or by a list of strings.  Each string in the
   list is an alternative numbering scheme, specified either as a set of
   digits or timers, or as an expression over which the gateway will
   attempt to find a shortest possible match.  The following constructs
   can be used in each numbering scheme:

   * Digit:A digit from "0" to "9".
   * Timer:The symbol "T" matching a timer expiry.
   * DTMF: A digit, a timer, or one of the symbols "A", "B", "C",
   "D", "#", or "*".  Extensions may be defined.
   * Wildcard: The symbol "x" which matches any digit ("0" to "9").
   * Range:One or more DTMF symbols enclosed between square brackets
   ("[" and "]").
   * Subrange: Two digits separated by hyphen ("-") which matches any
   digit between and including the two.  The subrange
   construct can only be used inside a range construct,
   i.e., between "[" and "]".
   * Position: A period (".") which matches an arbitrary number,
   including zero, of occurrences of the preceding
   construct.

   A gateway that detects events to be matched against a digit map MUST
   do the following:

   1) A

[Asterisk-Users] Looking for a wireless phone... wifi or traditional wireless ?

[Kim Lux]

My wife wants a cordless phone for around the house.  We are going to be
using VOIP exclusively very shortly.  Our current cordless phone is aged
and on the verge of replacement.  The other phone we are going to use is
a SIP Budgetone.

Should I buy a SIP to POTS converter and a new cordless phone or a wifi
SIP phone ?   

Is anyone using the Pulver WiSIP phone ?  Any comments ?

How about the zyxel ?

Thanks

-- 
Kim Lux,  Diesel Research Inc.


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Re: [Asterisk-Users] Multiple gateways for same dial pattern

[Andrew Kohlsmith]

On January 10, 2005 03:09 pm, Adi Linden wrote:
> How can I setup Asterisk to place calls if the same dial pattern can be
> routed through several PRI gateways. I have one way that I tried:
>
> So what happens is that if all channels on 172.17.99.5 are in use calls
> are routed to 172.17.99.6 and if all channels are in use to 172.17.99.7.

I use this in my dialplan, as a snippet from my dialing macro:

exten => s,n,Dial(${PROVIDER1}/${ARG1},,g)
exten => s,n,Goto(dial-${DIALSTATUS},1)

exten => dial-CANCEL,1,Hangup
exten => dial-ANSWER,1,Hangup
exten => dial-NOANSWER,1,Hangup
exten => dial-BUSY,1,Busy
exten => dial-CONGESTION,1,Congestion
exten => dial-CHANUNAVAIL,1,Macro(dial-provider2,${ARG1})

dial-provider2's dialplan looks similar but will fall through to the next 
provider, and so on.  Works and is next to instantaneous on PRI and IAX2 if 
qualify is set.

With some more magic and maybe some database interaction you can have it fully 
dynamic and fall through to any number of levels you want.

-A.
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Re: [Asterisk-Users] Multiple gateways for same dial pattern

[Adi Linden]

Hi,

I just tried this using a couple of iax peers and it works quite well. But
I did have to alter my dialplan. In my iax.conf I added 'qualify=5000' for
the nufone and voipjet peers.

[macro-gw-voipjet]
;
; This is the VoipJet IAX peer
;
; Use: Macro(gw-voipjet,${EXTEN}))
; Requires 'qualify=' statement in iax.conf for fallthru to work
;
exten => s,1,SetCIDNum(${PSTN_CIDNUM})
exten => s,2,Dial,IAX2/[EMAIL PROTECTED]/${ARG1}
exten => s,3,Goto(s-${DIALSTATUS},1)
exten => s-BUSY,1,Busy
exten => s-CHANUNAVAIL,1,Noop
exten => _s-.,1,Congestion

[macro-gw-nufone]
;
; This is the NuFone IAX peer
;
; Use: Macro(gw-nufone,${EXTEN}))
; Requires 'qualify=' statement in iax.conf for fallthru to work
;
exten => s,1,SetCIDNum(${PSTN_CIDNUM})
exten => s,2,Dial,IAX2/[EMAIL PROTECTED]/${ARG1}
exten => s,3,Goto(s-${DIALSTATUS},1)
exten => s-BUSY,1,Busy
exten => s-CHANUNAVAIL,1,Noop
exten => _s-.,1,Congestion

[outbound-longdistance]
;
exten => _91NXXNXX,1,Macro(gw-voipjet,${EXTEN:1}) ; VoipJet.com NANPA
exten => _91NXXNXX,2,Macro(gw-nufone,${EXTEN:1})  ; NuFone NANPA
exten => _91NXXNXX,3,Congestion
;
exten => _9011.,1,Macro(gw-voipjet,${EXTEN:1}); VoipJet.com WORLD
exten => _9011.,2,Macro(gw-nufone,${EXTEN:1}) ; NuFone WORLD
exten => _9011.,3,Congestion

Thank you!
Adi

On Mon, 10 Jan 2005, Matt Hess wrote:

> Just a thought I had on this.. Why not setup a sip peer entry in
> sip.conf with a qualify statement in it and send the call to the peer
> entry? That way asterisk will know if the peer is alive or not and I
> would think it would skip that particular peer accordingly.. that or
> it's really late and I've misunderstood what qualify is for.. but it
> seems to want to work how I imagine it would.. and from a small test I
> just tried it appears to do just that.. but who knows..
>

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RE: [Asterisk-Users] Is this a $50 wifi or wireless USB VOIP phone ?

[Jeff Glassman]

It is a USB attached phone

It needs to be used with a soft phone

It does work as a handset for x-ten type soft phone or their own softphone 


Jeff

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Kim Lux
Sent: Wednesday, January 12, 2005 9:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Is this a $50 wifi or wireless USB VOIP phone
?


http://www.pcphoneline.com/skype


"The VPT1000 is NOT a simple last generation USB phone audio device but
is rather a next generation integrated gateway and USB phoneset with
simultaneous dual mode Skype and SIP calling support.  Skype is not
forecast to have "SkypeIn" available until June 2005 but you can have
the capability now via its built in SIP capabilities."

Is this a wireless USB phone ?  Does it support SIP and could it be used
to connect to any SIP server ? 

Does anyone have experience with these ?

Thanks. 

-- 
Kim Lux,  Diesel Research Inc.


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Re: [Asterisk-Users] Is this a $50 wifi or wireless USB VOIP phone ?

[Kim Lux]

I just realized that there is nothing wireless about this phone.  I
think it is just a wired USB phone that looks like a wireless phone that
will be wireless is your laptop is wireless.  

I think the picture of the guy holding the phone in his hand vaguely
shows a cord in his palm.  


On Wed, 2005-01-12 at 19:09 -0700, Kim Lux wrote:
> http://www.pcphoneline.com/skype
> 
> 
> "The VPT1000 is NOT a simple last generation USB phone audio device but
> is rather a next generation integrated gateway and USB phoneset with
> simultaneous dual mode Skype and SIP calling support.  Skype is not
> forecast to have "SkypeIn" available until June 2005 but you can have
> the capability now via its built in SIP capabilities."
> 
> Is this a wireless USB phone ?  Does it support SIP and could it be used
> to connect to any SIP server ? 
> 
> Does anyone have experience with these ?
> 
> Thanks. 
> 
-- 
Kim Lux,  Diesel Research Inc.


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Re: [Asterisk-Users] Ports to open behind a NAT

[Howard Lowndes]

On Thu, 2005-01-13 at 11:51, [EMAIL PROTECTED] wrote:
> > >From searching the list archive I have come up with the following list
> > 
> > 22 for SSH  Should this be TCP, UDP or Both?
> TCP
> > 5060TCP Only

No, UDP

> > 1 -2 UDP Only

This last is for RTP and can be anything in the unpriv range.  It
depends what your SIP device is configured to and also what * is
configured to.

Personally I set my SIP devices to use src 3, and use * to use src
29000-2 to try to keep the open range as small as reasonable.

> > 
> > Is this info correct or is there other ports or port type  corrections
> > above?
> Yes, for SIP it is correct
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-- 
Howard.
LANNet Computing Associates;
Your Linux people 
--
"When you just want a system that works, you choose Linux;
when you want a system that just works, you choose Microsoft."
--
"Flatter government, not fatter government;
Get rid of the Australian states."


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Re: [Asterisk-Users] Setting channel display in SIP

[Howard Lowndes]

On Thu, 2005-01-13 at 12:38, Adam Goryachev wrote:
> On Thu, 2005-01-13 at 11:24 +1100, Howard Lowndes wrote:
> > On Thu, 2005-01-13 at 11:11, Adam Goryachev wrote:
> > > On Thu, 2005-01-13 at 10:20 +1100, Howard Lowndes wrote:
> > > > I have a situation where I need to know which Zap channel an incoming
> > > > call is on, so that the call can be answered appropriately when a SIP
> > > > phone displays the channel.  These Zap calls are coming in over PSTN and
> > > > don't have caller ID.
> > > > 
> > > > As far as I can make out my SIP phones (WuChuan HOP-1002) display the
> > > > user part from the SIP "From:" header as the second line on the
> > > > display.  If the call comes from another SIP phone then this shows as
> > > > the phone's number, but when the call comes in over the Zap channels
> > > > then it gets generated as "asterisk".
> > > 
> > > AFAIK, this is the default callerid asterisk uses when it doesn't
> > > receive callerid.
> > > Try adding setcallerid in your dialplan,
> > 
> > I tried setcidname in the dialplan without success, so I will try this
> > suggestion.
> 
> Play with combinations of setcallerid and setcidnum and setcidname ...
> see the wiki to correctly format your examples.

I have actually got a bit more cunning that this by using sipgetheader()
and sipaddheader().

The default user name is "asterisk", hard coded in chan_sip.c, so what I
did was to use sipgetheader() to get the From: header, then I cut() it
at the ":" character and the "@" character and checked the string
between these two characters.  If the string was "asterisk" then I did
sipaddheader(From: ${PIECE_BEFORE}:[EMAIL PROTECTED]).

OK, so it adds a second From: header, but as it gets added after the
original it doesn't seem to matter because it works and
"replacement-string" is what gets displayed on the phone, which is what
I want.  I also don't see that the tag= in the header makes any
difference either.

Can anyone see any probs I am likely to encounter using this?



> 
> > >  or callerid in your zapata.conf
> > > for each channel.
> > 
> > Tried that - no dice.
> 
> Send your zapata.conf file so we can see what you tried. AFAICT,
> asterisk is sometimes picky with the formatting of the callerid info.
> 
> Regards,
> Adam
-- 
Howard.
LANNet Computing Associates;
Your Linux people 
--
"When you just want a system that works, you choose Linux;
when you want a system that just works, you choose Microsoft."
--
"Flatter government, not fatter government;
Get rid of the Australian states."


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Re: [Asterisk-Users] AMP Anyone?

[Steve Totaro]

There is a mailing list dedicated to AMP that would be better suited to your
line of questions.

To answer your question, you can create your own files and name them
whatever you want and then use include statements.  The most you will have
to re-input are the include statements.

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

Sent: Wednesday, January 12, 2005 8:39 PM
Subject: RE: [Asterisk-Users] AMP Anyone?


Cool thanks.  I got the system running today, and you were right, it was
pretty easy.  There are several things I would like to change. What
files can be changed manually without AMP clobbering them?


David A. Morrow
Technical Systems Lead
Autodata Solutions Company
[EMAIL PROTECTED]
http://www.autodata.net
Tel: (519) 951-6079
Fax: (519) 451-6615

< Poor planning on your part does not necessarily consitute an emergency
on my part. >

This message has originated from Autodata Solutions. The attached
material is the Confidential and Proprietary Information of Autodata
Solutions. This email and any files transmitted with it are confidential
and intended solely for the use of the individual or entity to whom they
are addressed. If you have received this email in error please delete
this message and notify the Autodata system administrator at
[EMAIL PROTECTED] 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dennis
Boylan
Sent: Wednesday, January 12, 2005 6:05 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] AMP Anyone?

On Wed, Jan 12, 2005 at 11:59:00AM -0500, Dave Morrow wrote:
> Thanks for the information Dennis, it is much appreciated.  I think I
> am going to start from scratch (with AMP) also.  It's just a bit of a
> pain is all.  Do you have any expertise in regards to keeping current
> with * when new versions come out?  How does this impact AMP?
>

I've updated both AMP and asterisk multiple times.  Other than the
overwriting of my hooks (includes get killed), it went seemlessly.  I
have the issue where I want to have some extensions which don't have
voicemail.  So, I've added them via include files.

- Dennis
>
>
> David A. Morrow
> Technical Systems Lead
> Autodata Solutions Company
> [EMAIL PROTECTED]
> http://www.autodata.net
> Tel: (519) 951-6079
> Fax: (519) 451-6615
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[Asterisk-Users] Is this a $50 wifi or wireless USB VOIP phone ?

[Kim Lux]

http://www.pcphoneline.com/skype


"The VPT1000 is NOT a simple last generation USB phone audio device but
is rather a next generation integrated gateway and USB phoneset with
simultaneous dual mode Skype and SIP calling support.  Skype is not
forecast to have "SkypeIn" available until June 2005 but you can have
the capability now via its built in SIP capabilities."

Is this a wireless USB phone ?  Does it support SIP and could it be used
to connect to any SIP server ? 

Does anyone have experience with these ?

Thanks. 

-- 
Kim Lux,  Diesel Research Inc.


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Re: [Asterisk-Users] ASTCC configuration problem

[Steve Totaro]

> > > > > If nothing else, my efforts are documented for anyone else in the
> > > > > same boat.
> > > > > It seems that you can debug agi by typing agi debug at the *
command
> > > line.
> > > > > Amazing!  Here is the output.  I am assuming that since astcc-tone
> > didnt
> > > > > play, the problem lies there.  Thoughts?
> > > >
> > > > Steve,
> > > >
> > > > I just remembered...
> > > >
> > > > The Makefile puts all its sounds into /usr/share/asterisk/sounds but
*
> > > needs
> > > > the sounds in /var/lib/asterisk/sounds.
> > > >
> > > >
> > > > Karl
> > >
> > > Karl,
> > >
> > > Thanks alot!  Everything is working now but I dont get any
> > audio in either
> > > direction.  If I dial the same external number from the same
extension,
> > > everything is fine but going through the AGI breaks the audio.
> >
> > To clarify, I get audio just fine while entering the pin and phone
number
> > but when the call connects over my IAX provider there is no audio
> > in either
> > direction.
>
> Steve,
>
> I sounds like the problem may be in your "trunks" table definition of the
> IAX2 connection.
>
> Unfortunately, I am only using TDM interfaces so I do not know what the
path
> field should contain for IAX.
> For Zap interfaces, I simply put the group I want to use.  I.e. "g2".
>
>
> Karl

My final solution after seeing some other posts about CVS update fixing
different issues was to update.  It all works perfectly now.  Thanks again.

Steve

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[Asterisk-Users] RE: Polycom IP 500 Dial Issues

[Kai-Uwe Jensen]

On Wed, 12 Jan 2005, Paul Rodan wrote:

>> Yeah, it's a way for numbers to get sent faster, so you don't have to
wait
>> for the 3 second timeout before it gets transmitted to Asterisk. It's
>> similar to the dial-plan in the Sipura devices. 
>> 
>> I don't know where it's mentioned in their documentation, maybe it's on
>> their cd somewhere. I was just thrown a phone and told to figure it out,
>> thanks to searching through the forum and a little help on the irc
channel,
>> I figured it out after a little bit of time. It's extremely customizable
(so
>> are the Sipura's).
>
> Any idea on docs? I found the following message from the mailing list that

> I'm going to work with tonight..
> 
> http://lists.digium.com/pipermail/asterisk-users/2004-May/045721.html

 My Polycom admin guide (pg. 115, 4.6.2.1.4.1) says the dialplan digitmap
 is "compatible with the digit map feature of MGCP described in 2.1.5 of RFC
 3435". The RFC can be found in many places, for example here:
 http://www.faqs.org/rfcs/rfc3435.html

-- kuj


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RE: [Asterisk-Users] AMP Anyone?

[Dave Morrow]

Cool thanks.  I got the system running today, and you were right, it was
pretty easy.  There are several things I would like to change. What
files can be changed manually without AMP clobbering them? 


David A. Morrow
Technical Systems Lead
Autodata Solutions Company
[EMAIL PROTECTED]
http://www.autodata.net
Tel: (519) 951-6079
Fax: (519) 451-6615 

< Poor planning on your part does not necessarily consitute an emergency
on my part. >

This message has originated from Autodata Solutions. The attached
material is the Confidential and Proprietary Information of Autodata
Solutions. This email and any files transmitted with it are confidential
and intended solely for the use of the individual or entity to whom they
are addressed. If you have received this email in error please delete
this message and notify the Autodata system administrator at
[EMAIL PROTECTED] 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dennis
Boylan
Sent: Wednesday, January 12, 2005 6:05 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] AMP Anyone?

On Wed, Jan 12, 2005 at 11:59:00AM -0500, Dave Morrow wrote:
> Thanks for the information Dennis, it is much appreciated.  I think I 
> am going to start from scratch (with AMP) also.  It's just a bit of a 
> pain is all.  Do you have any expertise in regards to keeping current 
> with * when new versions come out?  How does this impact AMP?
>  

I've updated both AMP and asterisk multiple times.  Other than the
overwriting of my hooks (includes get killed), it went seemlessly.  I
have the issue where I want to have some extensions which don't have
voicemail.  So, I've added them via include files.

- Dennis
> 
> 
> David A. Morrow
> Technical Systems Lead
> Autodata Solutions Company
> [EMAIL PROTECTED]
> http://www.autodata.net
> Tel: (519) 951-6079
> Fax: (519) 451-6615
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Re: [Asterisk-Users] Setting channel display in SIP

[Adam Goryachev]

On Thu, 2005-01-13 at 11:24 +1100, Howard Lowndes wrote:
> On Thu, 2005-01-13 at 11:11, Adam Goryachev wrote:
> > On Thu, 2005-01-13 at 10:20 +1100, Howard Lowndes wrote:
> > > I have a situation where I need to know which Zap channel an incoming
> > > call is on, so that the call can be answered appropriately when a SIP
> > > phone displays the channel.  These Zap calls are coming in over PSTN and
> > > don't have caller ID.
> > > 
> > > As far as I can make out my SIP phones (WuChuan HOP-1002) display the
> > > user part from the SIP "From:" header as the second line on the
> > > display.  If the call comes from another SIP phone then this shows as
> > > the phone's number, but when the call comes in over the Zap channels
> > > then it gets generated as "asterisk".
> > 
> > AFAIK, this is the default callerid asterisk uses when it doesn't
> > receive callerid.
> > Try adding setcallerid in your dialplan,
> 
> I tried setcidname in the dialplan without success, so I will try this
> suggestion.

Play with combinations of setcallerid and setcidnum and setcidname ...
see the wiki to correctly format your examples.

> >  or callerid in your zapata.conf
> > for each channel.
> 
> Tried that - no dice.

Send your zapata.conf file so we can see what you tried. AFAICT,
asterisk is sometimes picky with the formatting of the callerid info.

Regards,
Adam
-- 
 -- 
Adam Goryachev
Website Managers
Ph:  +61 2 8304 [EMAIL PROTECTED]
Fax: +61 2 9345 4396www.websitemanagers.com.au

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Re: [Asterisk-Users] What is the best and easiest flavor to be usedwith Asterisk.

[Ryan Cavanaugh]

Roger Hanson wrote:
I'm using CentOS - which is another Red Hat Enterprise clone, like WBEL
www.centos.org
I've had no problems of any kind with the OS
- Original Message - From: "Imran Sadiq" <[EMAIL PROTECTED]>
To: 
Sent: Tuesday, January 11, 2005 9:58 PM
Subject: [Asterisk-Users] What is the best and easiest flavor to be 
usedwith Asterisk.

Could anyone please advise me on the best flavor of Linux on which
Asterisk is easiest to install.

I am currently using RH8.0, everything over the IP works fine but when I
want to call a physical line I can only have conversation for about 3
sec and everything freezes after that.

I have to hard reset the machine to bring it back up. Any suggestions
will be greatly appreciated.

Thanks

 
Imran Sadiq Systems Engineer

Tel:
+64 9 377 8282
  "World Class Support for any business 
Fax:
+64 9 377 7900
   with between 7 and 70 computers." 
Mob:
027  286  9269

LANcom 
Technology Limited  : 25 Union St, Auckland,
New Zealand


 


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No virus found in this incoming message.
Checked by AVG Anti-Virus.
Version: 7.0.300 / Virus Database: 265.6.10 - Release Date: 1/10/2005
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I 'm using CentOS also. No problems so far. I played with WhiteBox but 
moved over to CentOS because I got sick of waiting for yum to pull down 
updates.

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[Asterisk-Users] SNOM 190 Configuration with Asterisk

[Ty Carter]

Anyone have a suggestion for a configuration example using Asterisk and SNOM
190 SIP phone?

I have read both sets of documentation and for the life of me, I can't get
the phone to register and work.  I can use a IAX softphone and it works
perfectly.  It is just the SIP thing I guess.  Anyone have conf examples
they can share?

Thanks





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RE: [Asterisk-Users] Unofficial Broadvoice-users query/offer and DIDrouting question

[Vitalie Apostu]

Let me know how we can post message in your mailing list

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mike Cathey
Sent: Wednesday, January 12, 2005 9:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Unofficial Broadvoice-users query/offer and
DIDrouting question

Let me start by stating that I've been thinking about setting up an
unofficial broadvoice-users mailing list for a while.  I've asked Broadvoice
techs about it a couple times by phone and email and it looks like they
haven't passed it up to the people who would make that decision or said
people aren't interested in creating/maintaining one.
If there's enough interest (read 10 or more existing BV customers) in one,
I'd be willing to host it.  Interested parties please contact me _offlist_.
Please let me know if you think a forum/BB or ML would be preferred.

My main interest in setting up the list is to be able to troubleshoot issues
without bugging (a) BV (since they don't officially support * and don't lock
us out of their network) and (b) asterisk-users.

--Topic change--

If any BV customer with a 312/625 DID reads this and has had incoming call
issues, please let me know.  I've had incoming call routing stop working no
less than 3 times since 07/2004.  By stop working, I mean BV support (and
other customers) can call me through their softswitch(es), but calls through
the PSTN won't go through.  I've tested from 3 different NPA/NXXs and 3
separate PSTN providers in the past when it 'broke' and all were unable to
complete the call.  This seems like it could be an issue that's isolated to
the provider they're getting the DID from.  I acquired another did from them
that they're getting from a different provider (Global Crossing) so that I
can test this theory without bugging them.  312/625 is utilized by Global
Naps according to nanpa.com.

Cheers,

Mike

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[Asterisk-Users] Volume in line for music-on-hold

[Vitalie Apostu]

How to increase volume in line for music on hold?

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Re: [Asterisk-Users] not sharing IRQ's

[timebandit001]

> just to make sure:
> when i have zaptel devices on my box and i also use meetme and iax2,
> do i need to have USB device enabled and it's modules loaded?
No
your zaptel device will provide the needed hardware timer

the USB timer hack is for when you don't have any digium card
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[Asterisk-Users] BT keeps open sip channels

[Robert Rozman]

Hi,

I've switched to fresh install and unfortunately changed two things at the
time:
- used fresh AMP install
- upgraded Grandstream bt100 to latest beta from Grandstream web site .18

I have one local IAX2 extension (IAX phone) and one bt 100 extension. I
cannot make calls between those two.

With 'sip show channels' I can see 1-3 active SIP channels without doing
anything on Grandstream. Dial script (dialparties.agi) erases call to BT
cause it's like 'on the phone' condition  for Asterisk regarding opened sip
channels

Are opened sip channels common on Grandstreams ?  Is anyone running .18
version without these problems ?
Where should I start looking for solutions?

Thanks in advance,

regards,

Rob.

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RE: [Asterisk-Users] Xfering a call

[Michael Levenson]

Well that didn't workI now get this error


Jan 12 16:56:21 NOTICE[4989]: app_dial.c:746 dial_exec: Unable to create
channel of type 'SIP'
  == Everyone is busy/congested at this time
-- Executing VoiceMail("IAX2/[EMAIL PROTECTED]:4569/5", "b") in new
stackJan 12 16:56:21 WARNING[4989]: app_voicemail.c:1539 leave_voicemail: No
entry in voicemail config file for ''
-- Timeout on IAX2/[EMAIL PROTECTED]:4569/5
  == CDR updated on IAX2/[EMAIL PROTECTED]:4569/5
-- Executing Goto("IAX2/[EMAIL PROTECTED]:4569/5", "#|1") in new
stack
-- Goto (home,#,1)
-- Executing Playback("IAX2/[EMAIL PROTECTED]:4569/5", "sai-thanks")
in new stack
Jan 12 16:56:31 WARNING[4989]: file.c:475 ast_openstream: File sai-thanks
does not exist in any format
Jan 12 16:56:31 WARNING[4989]: file.c:779 ast_streamfile: Unable to open
sai-thanks (format ulaw): No such file or directory
Jan 12 16:56:31 WARNING[4989]: app_playback.c:83 playback_exec:
ast_streamfile failed on IAX2/[EMAIL PROTECTED]:4569/5 for sai-thanks
-- Executing Hangup("IAX2/[EMAIL PROTECTED]:4569/5", "") in new stack
  == Spawn extension (home, #, 2) exited non-zero on
'IAX2/[EMAIL PROTECTED]:4569/5'
-- Hungup 'IAX2/[EMAIL PROTECTED]:4569/5'

This user does have an entry in the voicemail.conf file..
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson
Sent: Wednesday, January 12, 2005 3:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Xfering a call

> I'm having an issue when I transfer a call to another SIP extension it
sees
> that the sip phone is not there and goes to voicemail but in my case it
> transfers to the main voicemail instead of the users voicemail.
> 
> Here is what my SIP extensions look like in the extension.conf file
> 
> exten => 3957,1,Dial(${Theresa},20,Tt)
> exten => 3957,2,VoicemailMain2(u${TheresaVM})
> exten => 3957,3,Hangup
> exten => 3957,102,VoicemailMain2(b${TheresaVM})
> exten => 3957,103,Hangup

Change the above from VoicemailMain2 to Voicemail and it will work
as expected.

The 3,Hangup isn't required... remove it. 103 isn't actually needed
either.


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Re: [Asterisk-Users] Polycom IP 500 Dial Issues

[Greg Boehnlein]

On Wed, 12 Jan 2005, Andrei (MPI) wrote:

> Greg Boehnlein wrote:
> 
> >Hello,
> > I have a mixture of Polycom SP IP 500 and 300 phones. I have been 
> >reading through the administration manual to try and solve this problem, 
> >but I do not seem to be able to find the answers to my question. I figured 
> >I would ask here and see if anyone has some suggestions.
> >
> >The problem is kind of annoying. After dialing 4 digits, the phone seems 
> >to pause and miss the 5th digit, often requiring the user to re-dial the 
> >5th digit several times.
> >
> >I'm not sure if this is some sort of "Early Dial" feauture trying to match 
> >on a 4 figit extension, but I would like any help that people can provide.
> >
> >  
> >
> It's because of the dialplan configuration in phone MAC-config file. 
> Easiest way is to get default conf file that came with your firmware and 
> replace it, modifying settings as needed.
> 
> Andrei

Andrei,
I am using the default file that came with the firmware version 
1.3.1. I've only made very slight modifications to it, most of which is 
specific to the phonexxx.cfg file specifically for registration, MWI and 
line issues. I'll dig deeper into this tonight when I get a chance.

-- 
Vice President of N2Net, a New Age Consulting Service, Inc. Company
 http://www.n2net.net Where everything clicks into place!
 KP-216-121-ST



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RE: [Asterisk-Users] Polycom IP 500 Dial Issues

[Greg Boehnlein]

On Wed, 12 Jan 2005, Paul Rodan wrote:

> Yeah, it's a way for numbers to get sent faster, so you don't have to wait
> for the 3 second timeout before it gets transmitted to Asterisk. It's
> similar to the dial-plan in the Sipura devices. 
> 
> I don't know where it's mentioned in their documentation, maybe it's on
> their cd somewhere. I was just thrown a phone and told to figure it out,
> thanks to searching through the forum and a little help on the irc channel,
> I figured it out after a little bit of time. It's extremely customizable (so
> are the Sipura's).

Any idea on docs? I found the following message from the mailing list that 
I'm going to work with tonight..

http://lists.digium.com/pipermail/asterisk-users/2004-May/045721.html

-- 
Vice President of N2Net, a New Age Consulting Service, Inc. Company
 http://www.n2net.net Where everything clicks into place!
 KP-216-121-ST



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Re: [Asterisk-Users] Ports to open behind a NAT

[timebandit001]

> >From searching the list archive I have come up with the following list
> 
> 22 for SSH  Should this be TCP, UDP or Both?
TCP
> 5060TCP Only
> 1 -2 UDP Only
> 
> Is this info correct or is there other ports or port type  corrections
> above?
Yes, for SIP it is correct
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[Asterisk-Users] Bristuff 0.20RC3 loses connectivity after short line interruption?

[Remco Barende]

I installed bristuff0.20-RC3 (which includes * 1.0.3 stable)
It works fine until I disconnect the phone jack for the ISDN line. Even 
after plug it back in asterisk still reports that it could not create a 
zap channel when I try to dial out and the line gives an engaged tone when 
I try to dial.

Re-starting asterisk doesn't solve this, I have to stop asterisk, unload 
the modules, reload the modules and start asterisk again.

I assume this is a bug, not a feature (should I e-mail it to Junghanns 
directly??)?

I know the telco here in holland and I will lose the line for a short 
period every once in a while and it's annoying when the line doesn't come 
back up.

Or did I forget some setting to recover from such a situation?
Thanks!
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RE: [Asterisk-Users] BroadVoice

[Mike Cathey]

On Tue, 2005-01-11 at 10:59 -0500, Vitalie Apostu wrote:
> Can you give me example of sip.conf and extention.conf which work with
> broadvoice? I want users who registered with Messenger through sip to be
> able to make a call thought broadvoice.

* for BV setup guide:

http://www.broadvoice.com/support_install_asterisk.html

Except use this patch instead of the one they link to:

http://edvina.net/broadvoice/broadvoicesip2.txt

Cheers,

Mike

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Re: [Asterisk-Users] Setting channel display in SIP

[Howard Lowndes]

On Thu, 2005-01-13 at 11:11, Adam Goryachev wrote:
> On Thu, 2005-01-13 at 10:20 +1100, Howard Lowndes wrote:
> > I have a situation where I need to know which Zap channel an incoming
> > call is on, so that the call can be answered appropriately when a SIP
> > phone displays the channel.  These Zap calls are coming in over PSTN and
> > don't have caller ID.
> > 
> > As far as I can make out my SIP phones (WuChuan HOP-1002) display the
> > user part from the SIP "From:" header as the second line on the
> > display.  If the call comes from another SIP phone then this shows as
> > the phone's number, but when the call comes in over the Zap channels
> > then it gets generated as "asterisk".
> 
> AFAIK, this is the default callerid asterisk uses when it doesn't
> receive callerid.
> Try adding setcallerid in your dialplan,

I tried setcidname in the dialplan without success, so I will try this
suggestion.

>  or callerid in your zapata.conf
> for each channel.

Tried that - no dice.

> 
> 
> Regards,
> Adam
> 
> 
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Howard.
LANNet Computing Associates;
Your Linux people 
--
"When you just want a system that works, you choose Linux;
when you want a system that just works, you choose Microsoft."
--
"Flatter government, not fatter government;
Get rid of the Australian states."


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[Asterisk-Users] Trouble building appradius

[Anthony Hill]

I am having trouble building appradius from http://appradius.minitelecom.org/

I configure, make, make install cpprad-1.0, but when I configure, then
make appradius I get :-

obelix:/usr/src/appradius/appradius1.0 # make
make[1]: Entering directory `/usr/src/appradius/appradius1.0/lib'
make[1]: Nothing to be done for `all'.
make[1]: Leaving directory `/usr/src/appradius/appradius1.0/lib'
make[1]: Entering directory `/usr/src/appradius/appradius1.0/etc'
make[1]: Nothing to be done for `all'.
make[1]: Leaving directory `/usr/src/appradius/appradius1.0/etc'
make[1]: Entering directory `/usr/src/appradius/appradius1.0/inc'
make[1]: Nothing to be done for `all'.
make[1]: Leaving directory `/usr/src/appradius/appradius1.0/inc'
make[1]: Entering directory `/usr/src/appradius/appradius1.0/src'
gcc -c -I../inc -O -Wall -I/usr/local/include -g -O2 app_radius.c
In file included from app_radius.c:17:
/usr/include/asterisk/lock.h: In function `ast_mutex_init':
/usr/include/asterisk/lock.h:311: error: `PTHREAD_MUTEX_RECURSIVE' undeclared 
(first use in this function)
/usr/include/asterisk/lock.h:311: error: (Each undeclared identifier is 
reported only once
/usr/include/asterisk/lock.h:311: error: for each function it appears in.)
make[1]: *** [app_radius.o] Error 1

..anyone know what that is about ?

Cheers.
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Re: [Asterisk-Users] What's the easiest way to get * to call PSTN?

[TC]

yah sorry dyslexic
this is what you want
http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&category=71505&item=5742009370
&rd=1
and i'd guess this will close $2000-2500 the  1- TNT-SL-CT1 will do 8 t1
but you will need to add an extra  1-APX8-SL-96DSP  to handle a full 8 t1
Pri
that should be about 1k extra

- Original Message -
From: Michael B. Murdock
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Wednesday, January 12, 2005 3:30 PM
Subject: Re: [Asterisk-Users] What's the easiest way to get * to call PSTN?


Nevermind... I presume you are refering to the Lucent (Ascend) MAX TNT WAN
Access Switch (correct??)

-- Mike

- Original Message -
From: Michael B. Murdock
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Wednesday, January 12, 2005 5:23 PM
Subject: Re: [Asterisk-Users] What's the easiest way to get * to call PSTN?


Who makes the TNT-Max ??

-- Mike

- Original Message -
From: TC
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Wednesday, January 12, 2005 5:01 PM
Subject: Re: [Asterisk-Users] What's the easiest way to get * to call PSTN?


ebayed TNT-Max with SIP for mere 8 pri you might get that for 3K or better
I know i was offered a 4 pri tnt-max for us1500
- Original Message -
From: Michael B. Murdock
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Wednesday, January 12, 2005 1:39 PM
Subject: Re: [Asterisk-Users] What's the easiest way to get * to call PSTN?


We are also looking for a high density SIP<->TDM gateway (signalling &
media) as an alternative to putting the ISDN PRI cards in the * box. Ideally
it should support up to 8 ISDN Pri's with NFAS on the TDM side and
100baseT/1000baseT on the IP side.

Has anyone had experience with this type of config?

-- Mike

- Original Message -
From: Walid Azab
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Sent: Wednesday, January 12, 2005 12:18 PM
Subject: RE: [Asterisk-Users] What's the easiest way to get * to call PSTN?


We have Asterisk CVS 1.0.2. I intend to connect Asterisk to Cisco 3745
unless there is a better way. Asterisk is not configured with any H/W. Cisco
3745 will accordingly send the call to the softswitch. PGW2200 which
controls our AS5300.

Walid




From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Damon Estep
Sent: Wednesday, January 12, 2005 3:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] What's the easiest way to get * to call PSTN?


You have not specified what type of lines you wish to use, POTS, PRI,
T1-CAS, E1, ISDN/BRI ???




From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Walid Azab
Sent: Wednesday, January 12, 2005 5:11 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] What's the easiest way to get * to call PSTN?

Hi,

I just want to know what is the easiest way to have Asterisk route calls to
PSTN. Hope any one can help me.

PS: Any solution using a Cisco device is preferable.


Thanks
Walid



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Re: [Asterisk-Users] New SIP Phone Config

[Howard Lowndes]

On Thu, 2005-01-13 at 10:40, John Dunham wrote:
> Just checking if anyone has experence with Integrated Networks IN1002 phone.

You might like to try aredfox.com and see if there is anything there
that might suit.  I have HOP1002 phones and I am using the "1002" as a
clue here.

> We just got 100 of them in and no manual or passowrd to program the phone.
> Also need some direction on the * sip.conf if anyone has experence with
> these phones.
> 
> Thanks,
> John Dunham
> 
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-- 
Howard.
LANNet Computing Associates;
Your Linux people 
--
"When you just want a system that works, you choose Linux;
when you want a system that just works, you choose Microsoft."
--
"Flatter government, not fatter government;
Get rid of the Australian states."


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Re: [Asterisk-Users] Setting channel display in SIP

[Adam Goryachev]

On Thu, 2005-01-13 at 10:20 +1100, Howard Lowndes wrote:
> I have a situation where I need to know which Zap channel an incoming
> call is on, so that the call can be answered appropriately when a SIP
> phone displays the channel.  These Zap calls are coming in over PSTN and
> don't have caller ID.
> 
> As far as I can make out my SIP phones (WuChuan HOP-1002) display the
> user part from the SIP "From:" header as the second line on the
> display.  If the call comes from another SIP phone then this shows as
> the phone's number, but when the call comes in over the Zap channels
> then it gets generated as "asterisk".

AFAIK, this is the default callerid asterisk uses when it doesn't
receive callerid.
Try adding setcallerid in your dialplan, or callerid in your zapata.conf
for each channel.


Regards,
Adam


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Re: [Asterisk-Users] linphone -> NAT -> * -> NAT -> firefly woes.

[Bruno Hertz]

OK, I'm coming to think linphone is bullshitting me.

I now tried the following call paths

 firefly -> * -> iaxcomm works
 firefly -> * -> linphone works
 sjphone -> * -> iaxcomm works, especially sip->iax works
 sjphone -> * -> linphone works

The opposite paths work too except

 linphone -> * -> firefly as said in my orig post, but also
 linphone -> * -> sjphone fails.

Guess it's about time to contact the linphone people.

In case I get that issue resolved I'll post the solution
here, too.

Thanks, Bruno.


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Re: [Asterisk-Users] Dial Out Errors

[Matt Riddell]

Scheda wrote:
If anyone knows of a linux applicable IAX softphone,
I'd be more than willing to give it a shot, but I haven't found one so
far.
Have you tried iaxcomm?
http://iaxclient.sourceforge.net/iaxcomm/
--
Cheers,
Matt Riddell
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Re: [Asterisk-Users] Re: test-ignore

[Matt Riddell]

Christian Savinovich wrote:
  I can't believe you!!, how an incredibly rude person you are.  In two 
paragraphs you manage to imply that you belong to the group of gurus in this 
list (my respects to you, oh major guru), that I don't know LookOut, and you 
suggest I should drop learning linux and asterisk (not even knowing if I'm an 
expert on both)
  Look, please leave me alone.  Don't bother me.  Get a life.  I started this thread with the title 
"test-ignore", and in the body I wrote "This is a test, please disregard".  
Hellooo.  Duh.  If that bothers you my friend, only some rest can help you.  For all I 
know, some of those people sending test mails to the list might be implementing new features into 
asterisk.  Or maybe if they get to sucessfully filter their mail they will be more productive to 
the list.
  My five minutes are way wasted, thank you.
As are you obviously!  :-)
You're lucky you didn't get 10,000 messages saying that the test worked!
hehe, now there's an idea - if you get an html mail or a test message, 
everybody should reply off-list to the person to inform them.  That way 
they will see how much bandwidth is used for their simple test.

Oh an BTW: generally not a good idea to berate Critchy!
:-)
--
Cheers,
Matt Riddell
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Re: [Asterisk-Users] Unicall errors

[Steve Underwood]

Sam Njenga wrote:
Hi Steve
Did that but still the same error :-(
PS. There is now unicall-0.0.2pre3. What are the changes in it ?
/Sam
 

pre3 has some bug fixes in the heart if the R2 protocol. The only file 
different between pre2 and pre3 is the libmfcr2 tar file. There were a 
couple of things where abandoned calls which could cause a crash.

Regards,
Steve
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Re: [Asterisk-Users] Xfering a call

[Rich Adamson]

> I'm having an issue when I transfer a call to another SIP extension it sees
> that the sip phone is not there and goes to voicemail but in my case it
> transfers to the main voicemail instead of the users voicemail.
> 
> Here is what my SIP extensions look like in the extension.conf file
> 
> exten => 3957,1,Dial(${Theresa},20,Tt)
> exten => 3957,2,VoicemailMain2(u${TheresaVM})
> exten => 3957,3,Hangup
> exten => 3957,102,VoicemailMain2(b${TheresaVM})
> exten => 3957,103,Hangup

Change the above from VoicemailMain2 to Voicemail and it will work
as expected.

The 3,Hangup isn't required... remove it. 103 isn't actually needed
either.


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Re: [Asterisk-Users] Re: [Asterisk-biz] SS7 and Asterisk solution

[Michael Welter]

Matt Riddell wrote:
Steve Underwood wrote:
Matthew Boehm wrote:
I know myself, SS7 will be a make or break for our continued use of
Asterisk.
  

   Our make/break is FoIP support. If Asterisk had some form of T.38 for
reliable fax transmission..or even just T38 pass-thru..
 

One down, one to go. The T.38 support will be free software, though. :-)
Steve

Just thought I post that you are truly a star Steve!
:-)
Keep up the good work!
Hear Hear!
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[Asterisk-Users] New SIP Phone Config

[John Dunham]

Just checking if anyone has experence with Integrated Networks IN1002 phone.
We just got 100 of them in and no manual or passowrd to program the phone.
Also need some direction on the * sip.conf if anyone has experence with
these phones.

Thanks,
John Dunham

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Re: [Asterisk-Users] Re: [Asterisk-biz] SS7 and Asterisk solution

[Matt Riddell]

Steve Underwood wrote:
Matthew Boehm wrote:
I know myself, SS7 will be a make or break for our continued use of
Asterisk.
  

   Our make/break is FoIP support. If Asterisk had some form of T.38 for
reliable fax transmission..or even just T38 pass-thru..
 

One down, one to go. The T.38 support will be free software, though. :-)
Steve
Just thought I post that you are truly a star Steve!
:-)
Keep up the good work!
--
Cheers,
Matt Riddell
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Re: [Asterisk-Users] Cant receive calls after network goes down and up

[Steve Totaro]

You can change the setting.  I set mine for 
every 1 min on a small system.  The phones always work.

  
  What is the register 
  interval in the grandstreams? The qualify=yes should keep the connection alive 
  as long as Asterisk is up, but if it goes down and then comes back up, the 
  phone has to re-register with Asterisk before asterisk can keep the connection 
  alive. 
   
   
   
  
  
  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of David NortonSent: Wednesday, January 12, 2005 4:44 
  PMTo: 'Asterisk Users Mailing List - Non-Commercial 
  Discussion'Subject: [Asterisk-Users] Cant receive 
  calls after network goes down and up
   
  Hi,
   
  I have several Grandstream phones 
  connected to Asterisk, some behind NAT and others not. If I reboot all the 
  phones, everything is fine. Should the connection go down, and then come back 
  again, those behind a NAT are still able to make calls, but are unable to 
  receive calls.
   
      -- Executing 
  Dial("SIP/1239-ba74", "SIP/1242|60|t") in new 
  stack
  Jan 12 23:45:19 NOTICE[21576]: 
  app_dial.c:803 dial_exec: Unable to create channel of type 'SIP' (cause 
  3)
    == Everyone is 
  busy/congested at this time (1:0/1/0)
   
  However, extension 1242 is still 
  able to call 1239? 
   
  Is this a configuration problem in 
  Asterisk or in the phones?
   
  Please 
  help
   
  Regards
   
  David 
  Norton
  
  

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[Asterisk-Users] Xfering a call

[Michael Levenson]

I'm having an issue when I transfer a call to another SIP extension it sees
that the sip phone is not there and goes to voicemail but in my case it
transfers to the main voicemail instead of the users voicemail.

Here is what my SIP extensions look like in the extension.conf file

exten => 3957,1,Dial(${Theresa},20,Tt)
exten => 3957,2,VoicemailMain2(u${TheresaVM})
exten => 3957,3,Hangup
exten => 3957,102,VoicemailMain2(b${TheresaVM})
exten => 3957,103,Hangup

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RE: [Asterisk-Users] Xorcom Rapid CD for Production?

[Jeff R Glassman]

Did you ever have success copying your configs to the Xorcom box?


Jeff

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Michael
Graves
Sent: Monday, January 03, 2005 10:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Xorcom Rapid CD for Production?


Hi All,

The past day or so I've setup a new * server based upon the Xorcom
Rapid ISO. It did as promised; wiped the base system, installed Debian
OS, installed Asterisk with a dummy configuration. So far so good.

If I could get the config from my existing * server migrated to the
Xorcom box the I'd be ready to roll. Essentially, it would be the same
as I have now on Fedora Core 1, but with a text mode management shell
and reboots into everything fully working without an user intervention.
I could put a 1 Gb CF card in a CF to IDE adapter and have an HD free
server, or just use a small HD instead.

However, I can't seem to get my current * configs to load to the Xorcom
box. I tried to sftp from my Windows desktop but my Windows ssh/sftp
client (www.privateshell.com) would not connect to the Xorcom box. I
confirmed that sshd was installed and running by trying Putty, as
described in the Xorcom faq. With Putty and its sftp counterpart I
could one-by-one upload the conf files from my existing server to the
Xorcom box, but it killed the * install on the Xorcom box.

Anybody have any experience turning the Xorcom Rapid installation into
a production installation for a small office? It would seem that I only
need to setup the configs appropriately, but something is going
dramatically wrong and I can't see it at the moment.

Michael

--
Michael Graves   [EMAIL PROTECTED]
Sr. Product Specialist  www.pixelpower.com
Pixel Power Inc. [EMAIL PROTECTED]

o713-861-4005
o800-905-6412
c713-201-1262



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[Asterisk-Users] IAX2 dropped calls: need debug suggestions

[hwstar]

Hi,

I'm trying to determine why IAX2 calls are getting dropped after a 4-24 hours 
of continuous connect time. My project requires that calls stay up for days at 
a time. When I turn on IAX2 debugging, I see "max retries exceeded" for control 
frames just before the connection is dropped.

My test setup is:

Zap Phone => Local Asterisk Server/IAX2/GSM => NAT => Internet => NAT(port 
mapped) => Remote Asterisk server/IAX2/GSM => Echo App

I also tried it without NAT on another pair of systems and it made no 
difference:

Zap Phone => Local Asterisk Server/IAX2/GSM => Internet =>  => Remote Asterisk 
server/IAX2/GSM => Echo App

Note: An SSH connection stays up indefinitely to the from the calling server to 
the called server. 

I wrote a small UDP echo client and server, and installed it on both ends, and 
noticed that a few packets are dropped every hour, but the connection seems 
solid with this simple client and server combo.

How resilient is the IAX2 protocol to receiving out-of-order packets, and how 
well does it deal with short term packet loss?

Has anyone else experienced problems with long duration calls over IAX2?

Thanks

Steve.






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Re: [Asterisk-Users] What's the easiest way to get * to call PSTN?

[Michael B. Murdock]

Nevermind... I presume you are refering to the 
Lucent (Ascend) MAX TNT WAN Access Switch (correct??)
 
-- Mike
 

  - Original Message - 
  From: 
  Michael 
  B. Murdock 
  To: Asterisk Users Mailing List - 
  Non-Commercial Discussion 
  Sent: Wednesday, January 12, 2005 5:23 
  PM
  Subject: Re: [Asterisk-Users] What's the 
  easiest way to get * to call PSTN?
  
  Who makes the TNT-Max ??
   
  -- Mike
   
  
- Original Message - 
From: 
TC 
To: Asterisk Users Mailing List - 
Non-Commercial Discussion 
Sent: Wednesday, January 12, 2005 5:01 
PM
Subject: Re: [Asterisk-Users] What's 
the easiest way to get * to call PSTN?

ebayed TNT-Max with SIP for mere 8 pri you 
might get that for 3K or better
I know i was offered a 4 pri tnt-max for 
us1500

  - Original Message - 
  From: 
  Michael B. Murdock 
  To: Asterisk Users Mailing List 
  - Non-Commercial Discussion 
  Sent: Wednesday, January 12, 2005 
  1:39 PM
  Subject: Re: [Asterisk-Users] What's 
  the easiest way to get * to call PSTN?
  
  We are also looking for a high density 
  SIP<->TDM gateway (signalling & media) as an alternative to 
  putting the ISDN PRI cards in the * box. Ideally it should support up to 8 
  ISDN Pri's with NFAS on the TDM side and 100baseT/1000baseT on the IP 
  side.
   
  Has anyone had experience with this type of 
  config?
   
  -- Mike
   
  
- Original Message - 
From: 
Walid Azab 

To: 'Asterisk Users Mailing 
List - Non-Commercial Discussion' 
Sent: Wednesday, January 12, 2005 
12:18 PM
Subject: RE: [Asterisk-Users] 
What's the easiest way to get * to call PSTN?

We have Asterisk CVS 1.0.2. I intend to connect Asterisk 
to Cisco 3745 unless there is a better way. Asterisk is not 
configured with any H/W. Cisco 3745 will accordingly send the call 
to the softswitch. PGW2200 which controls our 
AS5300.
 
Walid


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of 
Damon EstepSent: Wednesday, January 12, 2005 3:25 
PMTo: Asterisk Users Mailing List - Non-Commercial 
DiscussionSubject: RE: [Asterisk-Users] What's the easiest 
way to get * to call PSTN?


You have not 
specified what type of lines you wish to use, POTS, PRI, T1-CAS, E1, 
ISDN/BRI ???
 





From: 
[EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Walid AzabSent: Wednesday, January 12, 2005 
5:11 AMTo: 'Asterisk 
Users Mailing List - Non-Commercial Discussion'Subject: [Asterisk-Users] What's 
the easiest way to get * to call 
PSTN?
 

Hi,

 

I just want to know what is 
the easiest way to have Asterisk route calls to PSTN. Hope any one can 
help me.

 

PS: Any solution using a 
Cisco device is preferable.

 

 

Thanks

Walid



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RE: [Asterisk-Users] BroadVoice Troubles

[Rich Adamson]

> > My question is simply, has anyone received a deposit from 
> > these people once you return the equipment in good order? 
> > I've been unable to contact them now for almost 2 whole months.
> 
> Get in line.  Refunds are difficult it seems -- best bet is to go
> through the credit card co.  I cancelled a line three days before I was
> charged for it, gave them a week for the refund before I filed a
> chargeback with my CC.

For others that might be considering this, change your service to
BYOD-Lite, then cancel in a day or two. Their web site automatically
switches everthing, issues a credit for the unused original service,
and a charge for the $5.95/mo plan. Then don't use their service.

End result: max $5.95 (but the first month is likely to result in
a _credit_ to your credit card :)



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[Asterisk-Users] Setting channel display in SIP

[Howard Lowndes]

I have a situation where I need to know which Zap channel an incoming
call is on, so that the call can be answered appropriately when a SIP
phone displays the channel.  These Zap calls are coming in over PSTN and
don't have caller ID.

As far as I can make out my SIP phones (WuChuan HOP-1002) display the
user part from the SIP "From:" header as the second line on the
display.  If the call comes from another SIP phone then this shows as
the phone's number, but when the call comes in over the Zap channels
then it gets generated as "asterisk".

I tried playing with the realm field in sip.conf, which defaults to
"asterisk", but changing that doesn't do anything for me.

So, my question is:  How do I set the user part of the SIP From: header
to be the Zap channel?


-- 
Howard.
LANNet Computing Associates;
Your Linux people 
--
"When you just want a system that works, you choose Linux;
when you want a system that just works, you choose Microsoft."
--
"Flatter government, not fatter government;
Get rid of the Australian states."


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Re: [Asterisk-Users] What's the easiest way to get * to call PSTN?

[Michael B. Murdock]

Who makes the TNT-Max ??
 
-- Mike
 

  - Original Message - 
  From: 
  TC 
  To: Asterisk Users Mailing List - 
  Non-Commercial Discussion 
  Sent: Wednesday, January 12, 2005 5:01 
  PM
  Subject: Re: [Asterisk-Users] What's the 
  easiest way to get * to call PSTN?
  
  ebayed TNT-Max with SIP for mere 8 pri you might 
  get that for 3K or better
  I know i was offered a 4 pri tnt-max for 
  us1500
  
- Original Message - 
From: 
Michael 
B. Murdock 
To: Asterisk Users Mailing List - 
Non-Commercial Discussion 
Sent: Wednesday, January 12, 2005 1:39 
PM
Subject: Re: [Asterisk-Users] What's 
the easiest way to get * to call PSTN?

We are also looking for a high density 
SIP<->TDM gateway (signalling & media) as an alternative to 
putting the ISDN PRI cards in the * box. Ideally it should support up to 8 
ISDN Pri's with NFAS on the TDM side and 100baseT/1000baseT on the IP 
side.
 
Has anyone had experience with this type of 
config?
 
-- Mike
 

  - Original Message - 
  From: 
  Walid Azab 
  
  To: 'Asterisk Users Mailing List 
  - Non-Commercial Discussion' 
  Sent: Wednesday, January 12, 2005 
  12:18 PM
  Subject: RE: [Asterisk-Users] What's 
  the easiest way to get * to call PSTN?
  
  We have Asterisk CVS 1.0.2. I intend to connect Asterisk 
  to Cisco 3745 unless there is a better way. Asterisk is not 
  configured with any H/W. Cisco 3745 will accordingly send the call to 
  the softswitch. PGW2200 which controls our AS5300.
   
  Walid
  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Damon 
  EstepSent: Wednesday, January 12, 2005 3:25 PMTo: 
  Asterisk Users Mailing List - Non-Commercial DiscussionSubject: 
  RE: [Asterisk-Users] What's the easiest way to get * to call 
  PSTN?
  
  
  You have not 
  specified what type of lines you wish to use, POTS, PRI, T1-CAS, E1, 
  ISDN/BRI ???
   
  
  
  
  
  
  From: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Walid AzabSent: Wednesday, January 12, 2005 
  5:11 AMTo: 'Asterisk 
  Users Mailing List - Non-Commercial Discussion'Subject: [Asterisk-Users] What's the 
  easiest way to get * to call PSTN?
   
  
  Hi,
  
   
  
  I just want to know what is 
  the easiest way to have Asterisk route calls to PSTN. Hope any one can 
  help me.
  
   
  
  PS: Any solution using a Cisco 
  device is preferable.
  
   
  
   
  
  Thanks
  
  Walid
  
  

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[Asterisk-Users] Re: EuroISDN BRI 2 or 4 wires? (Remco Barende)

[HBK]

Hi
ISDN wire:
From phone company you receive on two wire, this is called "U" 
interface on this you can connect only one device, normaly the NT1 box.
On the NT1 there is a "S/T" bus that allows several devices (phones) 
connected (in "TE" mode)!
Yes S/T is four wire !

HB
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Re: [Asterisk-Users] linphone -> NAT -> * -> NAT -> firefly woes.

[Bruno Hertz]

On Wed, 2005-01-12 at 14:39 -0800, Erik Espinoza wrote:
> Did you enable passthrough for the rtp ports on the asterisk box?
> 
> I had the same problem until I enabled udp 1:2 on the firewall.

I did. That's why linphone -> * echo test works.

Maybe I made some progress however, by logging linphone output and
comparing the successful echo test to the unsuccessful iax bridge.

On echo test I see:

(linphone:5450): LinphoneCore-WARNING **: payload PCMA is not usable or enabled.
(linphone:5450): LinphoneCore-WARNING **: This remote sip phone did not answer 
properly to my sdp offer!
MediaStreamer-Message: ms_filter_add_link: OssRead,0 -> GSMEncoder,0
MediaStreamer-Message: ms_filter_add_link: GSMEncoder,0 -> RTPSend,0
MediaStreamer-Message: ms_filter_add_link: RTPRecv,0 -> GSMDecoder,0
MediaStreamer-Message: ms_filter_add_link: GSMDecoder,0 -> OssWrite,0
MediaStreamer-Message: Opening sound card in capture mode with 
stereo=0,rate=8000,bits=16
MediaStreamer-Message: dsp blocksize is 512.
MediaStreamer-Message: Opening sound card in playback mode with 
stereo=0,rate=8000,bits=16

whereas on linphone -> * -> firefly:

(linphone:5456): LinphoneCore-WARNING **: payload PCMA is not usable or enabled.
(linphone:5456): LinphoneCore-WARNING **: This remote sip phone did not answer 
properly to my sdp offer!
MediaStreamer-Message: Mediastreamer processing thread is exiting.

I.e. on echo test linphone does select the gsm codec, while with
iax bridge the media stream is canceled immediately, hence it stops
sending data as properly reported by *. Maybe it's a codec issue,
I'm just in the process of investigating ... just thinking, doesn't
* transcode between channel legs if necessary, could it be I disabled
that by accident (?) ...

Thanks, Bruno.


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Re: [Asterisk-Users] AMP Anyone?

[Dennis Boylan]

On Wed, Jan 12, 2005 at 11:59:00AM -0500, Dave Morrow wrote:
> Thanks for the information Dennis, it is much appreciated.  I think I am
> going to start from scratch (with AMP) also.  It's just a bit of a pain
> is all.  Do you have any expertise in regards to keeping current with *
> when new versions come out?  How does this impact AMP?
>  

I've updated both AMP and asterisk multiple times.  Other than the overwriting
of my hooks (includes get killed), it went seemlessly.  I have the issue where
I want to have some extensions which don't have voicemail.  So, I've added
them via include files.

- Dennis
> 
> 
> David A. Morrow
> Technical Systems Lead
> Autodata Solutions Company
> [EMAIL PROTECTED]
> http://www.autodata.net
> Tel: (519) 951-6079
> Fax: (519) 451-6615 
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Re: [Asterisk-Users] What's the easiest way to get * to call PSTN?

[TC]

ebayed TNT-Max with SIP for mere 8 pri you might 
get that for 3K or better
I know i was offered a 4 pri tnt-max for 
us1500

  - Original Message - 
  From: 
  Michael 
  B. Murdock 
  To: Asterisk Users Mailing List - 
  Non-Commercial Discussion 
  Sent: Wednesday, January 12, 2005 1:39 
  PM
  Subject: Re: [Asterisk-Users] What's the 
  easiest way to get * to call PSTN?
  
  We are also looking for a high density 
  SIP<->TDM gateway (signalling & media) as an alternative to putting 
  the ISDN PRI cards in the * box. Ideally it should support up to 8 ISDN Pri's 
  with NFAS on the TDM side and 100baseT/1000baseT on the IP side.
   
  Has anyone had experience with this type of 
  config?
   
  -- Mike
   
  
- Original Message - 
From: 
Walid Azab 
To: 'Asterisk Users Mailing List - 
Non-Commercial Discussion' 
Sent: Wednesday, January 12, 2005 12:18 
PM
Subject: RE: [Asterisk-Users] What's 
the easiest way to get * to call PSTN?

We have Asterisk CVS 1.0.2. I intend to connect Asterisk 
to Cisco 3745 unless there is a better way. Asterisk is not configured 
with any H/W. Cisco 3745 will accordingly send the call to the 
softswitch. PGW2200 which controls our AS5300.
 
Walid


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Damon 
EstepSent: Wednesday, January 12, 2005 3:25 PMTo: 
Asterisk Users Mailing List - Non-Commercial DiscussionSubject: 
RE: [Asterisk-Users] What's the easiest way to get * to call 
PSTN?


You have not 
specified what type of lines you wish to use, POTS, PRI, T1-CAS, E1, 
ISDN/BRI ???
 





From: 
[EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Walid AzabSent: Wednesday, January 12, 2005 5:11 
AMTo: 'Asterisk Users 
Mailing List - Non-Commercial Discussion'Subject: [Asterisk-Users] What's the 
easiest way to get * to call PSTN?
 

Hi,

 

I just want to know what is the 
easiest way to have Asterisk route calls to PSTN. Hope any one can help 
me.

 

PS: Any solution using a Cisco 
device is preferable.

 

 

Thanks

Walid



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Re: [Asterisk-Users] linphone -> NAT -> * -> NAT -> firefly woes.

[Erik Espinoza]

Did you enable passthrough for the rtp ports on the asterisk box?

I had the same problem until I enabled udp 1:2 on the firewall.

On Wed, 12 Jan 2005 22:06:18 +0100, Bruno Hertz <[EMAIL PROTECTED]> wrote:
> Hi folks
> 
> an issue I don't understand. I'm running * stable 1.0.3 on public
> internet, with following iax.conf / sip.conf entries:
> 
> iax.conf
> 
>  [100]
>  type=friend
>  username=Foo
>  context=default
>  auth=md5,plaintext,rsa
>  secret=secret
>  host=dynamic
>  callerid="Foo" <100>
>  qualify=no
> 
> sip.conf
> 
>  [10]
>  type=friend
>  username=Bar
>  context=default
>  callerid=Bar <10>
>  host=dynamic
>  secret=secret
>  nat=yes
>  canreinvite=no
> 
> On iax exten 10 I register firefly, on sip exten 100 linphone,
> both behind nat.
> 
> Now, calls I can do is e.g.
> firefly -> * -> linphone
> linphone -> * echo test (copied this from demo and put it on exten 600)
> 
> but what wouldn't properly work is is sip to iax bridging
> linphone -> * -> firefly
> 
> More specifically, firefly rings properly, but when I press Accept
> it just keeps ringing, and finally * tells me that linphone didn't
> send any frames:
> 
> channel.c:2655 ast_channel_bridge: Didn't get a frame from channel: 
> SIP/10-e8bd
> Jan 12 21:55:02 DEBUG[22661]: channel.c:2725 ast_channel_bridge: Bridge stops 
> bridging channels SIP/10-e8bd and IAX2/100/2
> 
> Doing my tcpdumps I checked that there's really no data sent by linphone,
> while nothing is dropped by firewalls either.
> 
> Did anyone experience similar troubles? A hint about how to resolve or further
> debug this would sure be appreciated.
> 
> Another point I'm wondering about is why, in that same connection, the
> caller id handed to firefly is just "10", and not the one specified
> in sip.conf, i.e. "Bar <10>".
> 
> I tested all that stuff also with iaxcomm, i.e. pure iax bridging
> iaxcomm -> NAT -> * -> NAT -> firefly
> and here, everything works OK, calls in both ways and caller id
> transmission.
> 
> Thanks, Bruno.
> 
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Re: [Asterisk-Users] Using asterisk to convert H.323 to SIP?

[sgup015]

We used the Nufone Implementation along with the OH323 implementation but, none
perform to a commercial level.

If there is a stable product out there, we would be keen on utilising it.
Quoting Paul Belanger <[EMAIL PROTECTED]>:

> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> James,
>
> Could you point me to the right source for some configuration settings?
> ~ I have searched http://voip-info.org, but could not see anything.
>
> PB
>
> James W. Coberly wrote:
> | We have a stabilized product for H323 to SIP conversion using *.  We
> | have tested it up to 500 lines per box without many troubles using G729.
> | Depending upon the hardware selected,  it should scale into the DS3+
> | range.
> |
> | James-
> |
> |
> | On Wed, 2005-01-12 at 13:34 -0800, William Boehlke wrote:
> |
> |>Yes, it is. Ugly but possible.
> |>
> |>
> |>-Original Message-
> |>From: [EMAIL PROTECTED]
> |>[mailto:[EMAIL PROTECTED] On Behalf Of Paul
> Belanger
> |>Sent: Wednesday, January 12, 2005 1:06 PM
> |>To: asterisk-users@lists.digium.com
> |>Subject: [Asterisk-Users] Using asterisk to convert H.323 to SIP?
> |>
> | Hello all,
> |
> | I was looking for some information about using Asterisk to convert an
> | incoming H.323 call to and outgoing SIP call.  Is this possible?
> |
> | PB
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>
>
>
> -BEGIN PGP SIGNATURE-
> Version: GnuPG v1.2.5 (MingW32)
> Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org
>
> iD8DBQFB5aF5YDbdAkxc7yQRAu8BAJ4zgIp2VTy0XZ4ea0vYe9/dGkJtnQCcC0SY
> 22JLyY+ZxyhxNGgIkc7R2G8=
> =JmB+
> -END PGP SIGNATURE-
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[Asterisk-Users] Re: Polycom IP500 - problems with multiple simultaneous calls

[ewr]

Well, thank you to Tor for the SetGroup/Checkgroup config.  It works
well!  (Thanks to John for the contexts/dialplan version, too).
Unfortunately, the phone doesn't audibly ring when the second call is
coming in (just a visual prompt), and you have to press the line
appearance button, and then the Answer softkey to actually pick up the
second call, but I think it is still an improvement over the Call
Waiting (at least logically, for my receptionists).

I use also SetGroup and Checkgroup to roll 4 lines on a Polycom 600 for a 
receptionist.  Just to let you know, you can press the line-appearance 
button twice to pick up the call, instead of hitting the line button, then 
the Answer softkey.  I would also like to figure out how to make the phone 
*ring* when you're already on another line, but haven't had a chance to 
seriously explore it yet.

Eric
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[Asterisk-Users] Queue and penalties

[Florian Overkamp]

Hi,

I'm trying to have a queue with members that work like this:

Member = Local/[EMAIL PROTECTED]
Member = Local/[EMAIL PROTECTED],1 ; Penalty!

And a dialplan that looks like this:

[context]
Exten = 101,1,DBget(Channel=QM/101)
Exten = 101,2,Dial(${Channel})
Exten = 101,102,Busy

And similar for 102.

Now, if 101 does not have its corresponding database entry, it will return a
Busy signal. However, if this happens, Member 102 is still never called. 

Is there something I'm missing ?

(Note, Yes, this is what agents are for, but in the version I'm tied to
(stable) agents are unable to do some other things I need. This could be a
workaround, if it behaved properly)

Florian


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[Asterisk-Users] R2/MFC Mexico FREE calls to test chan_unicall

[Miguel Cavazos]

Hi guys, I have one E1 with 30 channels in Mexico City, I guess that if 
i can fill this 30 channels with REAL traffic for 2 or 3 days I can 
find new bugs on chan_unicall or I can see how stable it can be. Im 
using R2/MFC with chan_unicall the patch that Steve Underwood wrote.

I will let anyone make FREE LOCAL calls to Mexico City till saturday or 
maybe until monday to see how stable this can be with REAL traffic. Add 
this to your extensions.conf only gsm as a codec is going to be 
permitted.

exten => 
_,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED],90,Tt)

--
Saludos,
Miguel Cavazos
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RE: [Asterisk-Users] So many Asterisk Patches - Which do I choose and use?

[Paul Rodan]

Which MOH patch should I use? What is the best link to the best mpg123
replacement patch? Something with instructions please.

I found the original MOH patch, but then I remember somebody mentioning
another good patch that included the MOH patch within it, but I can't find
that post/link, anybody?

Any other CPU or performance saving or crash-prevening patch's that need
applying to the latest CVS-Stable?



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian S.
Adelson
Sent: Wednesday, January 12, 2005 12:08 PM
To: John Middleton; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] So many Asterisk Patches - Which do I choose
anduse?

To get CVS-STABLE:


export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot
cvs login
cvs checkout -r v1-0 zaptel libpri asterisk 

> type these 3 command inorder to get CVS HEAD.
> export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot
> cvs login
> cvs checkout zaptel libpri asterisk 
> 
> 
> 
> On Wed, 12 Jan 2005 16:20:06 +, John Middleton
> <[EMAIL PROTECTED]> wrote:
> > When you say CVS HEAD is the the same as stable? where do you get it
> > from and what params do you use?
> > 
> > 
> > On Wed, 12 Jan 2005 17:03:34 +, Niksa Baldun
<[EMAIL PROTECTED]> wrote:
> > > There is no easy answer to your question. If you ask me, I prefer not
to use
> > > any patches, except that I am forced to use bristuff because I have
quadBRI
> > > ISDN cards. Bristuff patches Zaptel in order to enable using quadBRI
and
> > > octoBRI cards, and also  adds some features to *. More info on
> > > www.junghanns.net.
> > >
> > > Like you said, really valuable patches will make it to the CVS sooner
or
> > > later, so I prefer to wait because it makes installation and
maintenance
> > > easier.
> > >
> > > I use Gentoo with 2.6 kernel. I am not sure whether you will get any
> > > benefits from upgrading, but I didn't have any problems with it
(except that
> > > I had to migrate from devfs to udev, but that issue exists with 2.4
kernel
> > > too).
> > >
> > >
> > > Paul Rodan wrote:
> > >
> > >
> > > Ok,
> > >
> > >
> > >
> > > I usually use the latest stable CVS, with no patches or modifications.
If
> > > figured if there was a worthwhile patch, Mark would have already
included
> > > it. However, there was that neat patch about being able to press a
certain
> > > key and it'd begin recording in mid-stream, that was an awesome
feature and
> > > I patched my latest features.c file with that patch. But I keep seeing
> > > mentions of other patches, specifically something about the MOH patch,
the
> > > BRISTUFFED patch, and now I'm hearing about a "Super Parking Lot"
patch? For
> > > now I've been using the mpg123 method, it tends to work for me, but if
I can
> > > save CPU/RAM and other troubles by using another format, which one do
I go
> > > with? What is BRISTUFFED? And if I'm right, the super parking lot
patch
> > > allows for call parking based on context, a way to break it apart,
instead
> > > of making it universal across the whole system (where can I find this
> > > patch)?
> > >
> > >
> > >
> > > So I'm going to ask the question, if I were to install the latest CVS
Stable
> > > tonight, which patches should I install on it before compiling? Also,
I'm
> > > using Gentoo Linux, with the 2.4.26-r9 gentoo kernel. I've seen issues
with
> > > people making Asterisk work perfectly with the 2.6 kernel so I've
stayed
> > > clear of it, but I still see people fighting to make it work and such,
I saw
> > > one post a while back about the benefits using Asterisk w/ the 2.6
kernel,
> > > can somebody please refresh my memory? What are the benefits of using
> > > Asterisk with the 2.6 kernel? I'm trying to get the most out of my
system.
> > >
> > >
> > >
> > > Any help in making tonights compile/upgrade go perfect would be
greatly
> > > appreciated.
> > >
> > >
> > >
> > > Thanks,
> > >
> > > Paul
> > >
> > ___
> > Asterisk-Users mailing
> > > list
> > Asterisk-Users@lists.digium.com
> > 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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> > >
> > >
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> >
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Re: [Asterisk-Users] Asterisk server stopped - "0-order allocation failed " errors in the log

[Steven Critchfield]

On Wed, 2005-01-12 at 15:50 -0500, Paul Rodan wrote:
> Hey, 
> 
>  
> 
> One of my remote Asterisk servers which has been up and running for a
> couple of months now suddenly stopped. I didn’t realize it because my
> monitoring system only pings the machine. But my remote office
> complained, I checked “iax2 show peers” and saw they weren’t checked
> in. So I pinged the IP of their Asterisk server and got a response.
>  Tried to SSH in but it timed out. Machine must be in some kind of
> locked state. So I had them reboot it, it’s now up and working fine,
> but the log files show:
> 
>  
> 
> …
> 
> Jan 12 15:10:00 palmtrace-nuvoip CRON[30145]: (root) CMD (test
> -x /usr/sbin/run-crons && /usr/sbin/run-crons )
> 
> Jan 12 15:12:27 palmtrace-nuvoip tftpd[7279]: Serving ata000d28383ca9
> to 10.9.8.49:6616
> 
> Jan 12 15:12:27 palmtrace-nuvoip tftpd[21012]: Serving atadefault.cfg
> to 10.9.8.49:6617
> 
> Jan 12 15:13:43 palmtrace-nuvoip __alloc_pages: 0-order allocation
> failed (gfp=0x1d2/0)
> 
> Jan 12 15:13:43 palmtrace-nuvoip VM: killing process mpg123
> 
> Jan 12 15:13:48 palmtrace-nuvoip __alloc_pages: 0-order allocation
> failed (gfp=0x1f0/0)

> Jan 12 23:29:07 palmtrace-nuvoip syslog-ng[17735]: syslog-ng version
> 1.6.5 starting

> 
> The only clue is the mpg123 error. Do you think it’s possible all my
> memory got allocated? It’s running Asterisk
> CVS-v1-0-10/21/04-00:18:55 

I'm pretty sure you ran out of memory. I don't think mpg123 caused it. 

> Should I upgrade it w/ the MOH patch? Anyways, any advice or help
> would be appreciated.

Advice would be to first to ensure you don't use HTML email. 

Next, If you thought it was out of memory, it would probably be a good
idea to also mention hardware specs including RAM and swap space. 

After that, you might want to look around and see if you might have been
hit by some kind of worm. Look for any other processes that decided to
fail around the same time. I know of a few people who are experiencing
unexplained ssh probes. 
-- 
Steven Critchfield <[EMAIL PROTECTED]>

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Re: [Asterisk-Users] Using asterisk to convert H.323 to SIP?

[Paul Belanger]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
James,
Could you point me to the right source for some configuration settings?
~ I have searched http://voip-info.org, but could not see anything.
PB
James W. Coberly wrote:
| We have a stabilized product for H323 to SIP conversion using *.  We
| have tested it up to 500 lines per box without many troubles using G729.
| Depending upon the hardware selected,  it should scale into the DS3+
| range.
|
| James-
|
|
| On Wed, 2005-01-12 at 13:34 -0800, William Boehlke wrote:
|
|>Yes, it is. Ugly but possible.
|>
|>
|>-Original Message-
|>From: [EMAIL PROTECTED]
|>[mailto:[EMAIL PROTECTED] On Behalf Of Paul
Belanger
|>Sent: Wednesday, January 12, 2005 1:06 PM
|>To: asterisk-users@lists.digium.com
|>Subject: [Asterisk-Users] Using asterisk to convert H.323 to SIP?
|>
| Hello all,
|
| I was looking for some information about using Asterisk to convert an
| incoming H.323 call to and outgoing SIP call.  Is this possible?
|
| PB
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-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.5 (MingW32)
Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org
iD8DBQFB5aF5YDbdAkxc7yQRAu8BAJ4zgIp2VTy0XZ4ea0vYe9/dGkJtnQCcC0SY
22JLyY+ZxyhxNGgIkc7R2G8=
=JmB+
-END PGP SIGNATURE-
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RE: [Asterisk-Users] Cant receive calls after network goes down and up

[David Norton]

qualify is not set in sip.conf at all. What
should the value be, or should it just be set to yes?

 

The register interval is 60 minutes. The
Asterisk server is not going down, but the connection between the phone and the
server might go down for a few minutes, and when it comes back up the problem
occurs.

 

 









From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
On Behalf Of Paul Rodan
Sent: Wednesday, January 12, 2005
11:53 PM
To: 'Asterisk Users Mailing List -
Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Cant
receive calls after network goes down and up



 

What is the register interval in the grandstreams?
The qualify=yes should keep the connection alive as long as Asterisk is up, but
if it goes down and then comes back up, the phone has to re-register with
Asterisk before asterisk can keep the connection alive. 

 

 

 









From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David Norton
Sent: Wednesday, January 12, 2005
4:44 PM
To: 'Asterisk
 Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Cant
receive calls after network goes down and up



 

Hi,

 

I have several Grandstream phones connected to Asterisk,
some behind NAT and others not. If I reboot all the phones, everything is fine.
Should the connection go down, and then come back again, those behind a NAT are
still able to make calls, but are unable to receive calls.

 

    -- Executing
Dial("SIP/1239-ba74", "SIP/1242|60|t") in new stack

Jan 12 23:45:19 NOTICE[21576]: app_dial.c:803 dial_exec:
Unable to create channel of type 'SIP' (cause 3)

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

 

However, extension 1242 is still able to call 1239? 

 

Is this a configuration problem in Asterisk or in the
phones?

 

Please help

 

Regards

 

David Norton





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

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[Asterisk-Users] getting * to start on suse 9.1

[eamonn doyle]

hello all,

I have been lurking here for a while learning what I can, and reading
quite a bit, using the fine reference at asteriskdocs.org I have been
building my test server on an older compaq dl380, PIII-700, 1.2gig mem
[X not running].

My problems begin when trying to start * for the first time, I have
purchased a TDM400P card with 4 FXO modules from digium and that is
installed and seemingly working fine from what I can see.

I made a singe entry in zaptel.conf adding "fxsks=1-4"
then
modprobe zaptel
and
modprobe wcfxs

var/log/messages reports:

Jan 12 16:03:42 asterisk kernel: Zapata Telephony Interface Registered
on major 196
Jan 12 16:03:51 asterisk kernel: Freshmaker version: 71
Jan 12 16:03:51 asterisk kernel: Freshmaker passed register test
Jan 12 16:03:51 asterisk kernel: Module 0: Installed -- AUTO FXO (FCC
mode)
Jan 12 16:03:51 asterisk kernel: Module 1: Installed -- AUTO FXO (FCC
mode)
Jan 12 16:03:51 asterisk kernel: Module 2: Installed -- AUTO FXO (FCC
mode)
Jan 12 16:03:51 asterisk kernel: Module 3: Installed -- AUTO FXO (FCC
mode)
Jan 12 16:03:51 asterisk kernel: Found a Wildcard TDM: Wildcard TDM400P
REV E/F (4 modules)
Jan 12 16:03:51 asterisk kernel: Registered tone zone 0 (United States /
North America)


So from what I can see all seems well up to this point.

when I start * I get the following at the very end where it craps out:
[I am assuming this is where the problem is]

[app_privacy.so]Jan 12 16:58:13 WARNING[8395]: loader.c:309
ast_load_resource: /usr/lib/asterisk/modules/app_privacy.so: undefined
symbol: ast_strlen_zero
Jan 12 16:58:13 WARNING[8395]: loader.c:501 load_modules: Loading module
app_privacy.so failed!
Ouch ... error while writing audio data: : Broken pipe

This does not mean much to me, and searching on wiki and google only
seem to give references to code examples which are Irish to me.

I have not started the * configuration process yet as it seems I should
be able to get * started prior to that step.

Any help for this problem is greatly appreciated.

Thanks
Eamonn


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[Asterisk-Users] EuroISDN BRI 2 or 4 wires?

[Remco Barende]

Hi List!
Have a weird problem with ISDN in The Netherlands.
The line that is coming in from the telco is 2 wires. The line is 
connected to an NT1 using the middle pair of a UTP connector. So far sop 
good.

However, the outgoing ports on the NT1, should they be wired with 2 wires 
or 4 (2 pairs)?

If I use a 4 wire cable and stick it in the NT1 I get 0 voltage on the 
middle 2 wires, but i get approx 4 volts on the left 2 wires and another 
40 volts on the outer (right) 2 wires. Is it correct that 2 pairs should 
be used to connect ISDN devices?

Sorry for this slightly offtopic issue. The house is fully 
wired for one pair which means I may have a problem if i need 2 pairs.

Thanks!
Remco
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RE: [Asterisk-Users] Using asterisk to convert H.323 to SIP?

[James W. Coberly]

We have a stabilized product for H323 to SIP conversion using *.  We
have tested it up to 500 lines per box without many troubles using G729.
Depending upon the hardware selected,  it should scale into the DS3+
range.

James-


On Wed, 2005-01-12 at 13:34 -0800, William Boehlke wrote:
> Yes, it is. Ugly but possible.
> 
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Paul Belanger
> Sent: Wednesday, January 12, 2005 1:06 PM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] Using asterisk to convert H.323 to SIP?
> 
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
> 
> Hello all,
> 
> I was looking for some information about using Asterisk to convert an
> incoming H.323 call to and outgoing SIP call.  Is this possible?
> 
> PB
> -BEGIN PGP SIGNATURE-
> Version: GnuPG v1.2.5 (MingW32)
> Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org
> 
> iD8DBQFB5ZE2YDbdAkxc7yQRAmFxAKCYwA24FtnnbSxtgm6oSM5KtNUr7wCglDcH
> gcBg9EdQqJWuIvp9mr53sXE=
> =P8Mc
> -END PGP SIGNATURE-
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RE: [Asterisk-Users] Changes to manager outputs - A discussion

[Peter Svensson]

On Wed, 12 Jan 2005, Asterisk wrote:

> Sometimes things are so obvious that you miss them. "just view a single LF
> as the field separator and a double LF as the record separator" is, of
> course, the point that makes me look soo stupid.

Note that the order of the elements is only defined by their tags. You can 
not rely on a specific order and just ignore the tags. Reordering the tags 
once the records are isolated should not be too hard, if you wish to 
import to a spreadsheet.

Peter


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RE: [Asterisk-Users] Cant receive calls after network goes down and up

[Paul Rodan]

What is the register interval in the
grandstreams? The qualify=yes should keep the connection alive as long as
Asterisk is up, but if it goes down and then comes back up, the phone has to
re-register with Asterisk before asterisk can keep the connection alive. 

 

 

 









From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David Norton
Sent: Wednesday, January 12, 2005
4:44 PM
To: 'Asterisk
 Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Cant
receive calls after network goes down and up



 

Hi,

 

I have several Grandstream phones connected to Asterisk,
some behind NAT and others not. If I reboot all the phones, everything is fine.
Should the connection go down, and then come back again, those behind a NAT are
still able to make calls, but are unable to receive calls.

 

    -- Executing
Dial("SIP/1239-ba74", "SIP/1242|60|t") in new stack

Jan 12 23:45:19 NOTICE[21576]: app_dial.c:803 dial_exec:
Unable to create channel of type 'SIP' (cause 3)

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

 

However, extension 1242 is still able to call 1239? 

 

Is this a configuration problem in Asterisk or in the
phones?

 

Please help

 

Regards

 

David Norton






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Re: [Asterisk-Users] (UN)structured E1

[Peter Svensson]

On Wed, 12 Jan 2005, Alex G Robertson wrote:

> If I got the matter, unstructured framing is used for Data (2M full) and 
> structured for "64k circuits".

You can run data over a channelized link, or even over pri. There are lots 
of flexibility in how an E1/T1 can be configured. For voice there are only 
a few configurations that are still common, but for data the number of 
permutations is enormous.

Peter


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Re: [Asterisk-Users] Unofficial Broadvoice-users query/offer and DID routing question

[Mark Musone]

i'll join the list. my DID is 716, but i continuously have issues with
BV..the good thing is i've kinda learned a lot about how they are
setup. The bad thing is i'm still plagued with horrible jitter/warble,
downtime, and dial-out capability..



On Wed, 12 Jan 2005 09:44:59 -0500, Mike Cathey
<[EMAIL PROTECTED]> wrote:
> Let me start by stating that I've been thinking about setting up an
> unofficial broadvoice-users mailing list for a while.  I've asked
> Broadvoice techs about it a couple times by phone and email and it looks
> like they haven't passed it up to the people who would make that
> decision or said people aren't interested in creating/maintaining one.
> If there's enough interest (read 10 or more existing BV customers) in
> one, I'd be willing to host it.  Interested parties please contact me
> _offlist_.  Please let me know if you think a forum/BB or ML would be
> preferred.
> 
> My main interest in setting up the list is to be able to troubleshoot
> issues without bugging (a) BV (since they don't officially support * and
> don't lock us out of their network) and (b) asterisk-users.
> 
> --Topic change--
> 
> If any BV customer with a 312/625 DID reads this and has had incoming
> call issues, please let me know.  I've had incoming call routing stop
> working no less than 3 times since 07/2004.  By stop working, I mean BV
> support (and other customers) can call me through their softswitch(es),
> but calls through the PSTN won't go through.  I've tested from 3
> different NPA/NXXs and 3 separate PSTN providers in the past when it
> 'broke' and all were unable to complete the call.  This seems like it
> could be an issue that's isolated to the provider they're getting the
> DID from.  I acquired another did from them that they're getting from a
> different provider (Global Crossing) so that I can test this theory
> without bugging them.  312/625 is utilized by Global Naps according to
> nanpa.com.
> 
> Cheers,
> 
> Mike
> 
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RE: [Asterisk-Users] Cant receive calls after network goes down and up

[Tenorio, Leandro]

That's probably a timeout problem in the nat 
box.
 


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of David 
NortonSent: Wednesday, January 12, 2005 6:44 PMTo: 
'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: 
[Asterisk-Users] Cant receive calls after network goes down and 
up


Hi,
 
I have several Grandstream phones connected to Asterisk, some behind NAT 
and others not. If I reboot all the phones, everything is fine. Should the 
connection go down, and then come back again, those behind a NAT are still able 
to make calls, but are unable to receive calls.
 
    -- Executing Dial("SIP/1239-ba74", "SIP/1242|60|t") in new 
stack
Jan 12 23:45:19 NOTICE[21576]: app_dial.c:803 dial_exec: Unable to create channel of type 'SIP' (cause 
3)
  == Everyone is 
busy/congested at this time (1:0/1/0)
 
However, extension 1242 is still 
able to call 1239? 
 
Is this a configuration problem in 
Asterisk or in the phones?
 
Please 
help
 
Regards
 
David 
Norton
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Re: [Asterisk-Users] SS7 and Asterisk solution

[Peter Svensson]

On Wed, 12 Jan 2005, Matthew Boehm wrote:

> ahh..american arrogance. I assumed you were in the US.
> We pay $2000 a month for DS3/SS7 to national carrier. We will soon be
> dropping the SS7 and turning that voice DS3 into a bandwidth DS3. We will
> still use the carrier but all calls will terminate to them VoIP. It will
> save us over $2000 a month.

Interesting. For us in Sweden it is actually cheaper to connect to our 
carrier through PRIs than through VoIP. We have two fibre pairs instead of 
one, and it is still cheaper. Well, they had six pairs since that is the 
minimum cables they install.

Add to that the lower latency etc of a tdm solution. I guess it is a scale
thing - when lots (most?) companies are connected via isdn technology the
cost goes down.

Peter


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[Asterisk-Users] Cant receive calls after network goes down and up

[David Norton]

Hi,

 

I have several Grandstream phones
connected to Asterisk, some behind NAT and others not. If I reboot all the
phones, everything is fine. Should the connection go down, and then come back
again, those behind a NAT are still able to make calls, but are unable to
receive calls.

 

    -- Executing Dial("SIP/1239-ba74", "SIP/1242|60|t")
in new stack

Jan 12 23:45:19 NOTICE[21576]:
app_dial.c:803 dial_exec: Unable to create channel of
type 'SIP' (cause 3)

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

 

However, extension 1242 is still able to call 1239? 

 

Is this a configuration problem in Asterisk or in the
phones?

 

Please help

 

Regards

 

David Norton






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[Asterisk-Users] Come join the Asterisk Bookclub

[Nick Bachmann]

Greetings all-
For whatever reason of personal insanity, I've decided to start an 
Asterisk bookclub.  Basically, we'll pick three books every month (a 
users book, a developers book, and another general interest book) and 
then read and discuss on IRC in the #asterisk-bookclub channel. 

The users' book will be something related to telephony or the technology 
related to being an Asterisk administrator.  The developers' book will 
be, similarly, related to software development with a particular focus 
on telephony, drivers, and the related technology an Asterisk developer 
should know about.  The general interest book will be something fiction 
or nonfiction that is interesting to the folks who hang around here :-).

To get things rolling, here are our books for January:
Users: _Ethernet: The Definitive Guide_, from O'Reilly.  ISBN 1565926609
Developers: _Secure Programming Cookbook for C and C++_, also from ORA. 
ISBN 0596003943**
General Interest: _Cuckoo's Egg_, by Cliff Stoll. ISBN 0743411463

So, head out to your favorite bookstore and join #asterisk-bookclub!
See ya on IRC,
Nick
(IRC hermie)
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RE: [Asterisk-Users] BroadVoice Troubles

[Jay Milk]

> My question is simply, has anyone received a deposit from 
> these people once you return the equipment in good order? 
> I've been unable to contact them now for almost 2 whole months.

Get in line.  Refunds are difficult it seems -- best bet is to go
through the credit card co.  I cancelled a line three days before I was
charged for it, gave them a week for the refund before I filed a
chargeback with my CC.

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RE: [Asterisk-Users] Using asterisk to convert H.323 to SIP?

[William Boehlke]

Yes, it is. Ugly but possible.


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul Belanger
Sent: Wednesday, January 12, 2005 1:06 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Using asterisk to convert H.323 to SIP?

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Hello all,

I was looking for some information about using Asterisk to convert an
incoming H.323 call to and outgoing SIP call.  Is this possible?

PB
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Re: [Asterisk-Users] What's the easiest way to get * to call PSTN?

[Michael B. Murdock]

We are also looking for a high density 
SIP<->TDM gateway (signalling & media) as an alternative to putting 
the ISDN PRI cards in the * box. Ideally it should support up to 8 ISDN Pri's 
with NFAS on the TDM side and 100baseT/1000baseT on the IP side.
 
Has anyone had experience with this type of 
config?
 
-- Mike
 

  - Original Message - 
  From: 
  Walid Azab 
  To: 'Asterisk Users Mailing List - 
  Non-Commercial Discussion' 
  Sent: Wednesday, January 12, 2005 12:18 
  PM
  Subject: RE: [Asterisk-Users] What's the 
  easiest way to get * to call PSTN?
  
  We have Asterisk CVS 1.0.2. I intend to connect Asterisk 
  to Cisco 3745 unless there is a better way. Asterisk is not configured 
  with any H/W. Cisco 3745 will accordingly send the call to the 
  softswitch. PGW2200 which controls our AS5300.
   
  Walid
  
  
  From: [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Damon 
  EstepSent: Wednesday, January 12, 2005 3:25 PMTo: 
  Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: 
  [Asterisk-Users] What's the easiest way to get * to call 
  PSTN?
  
  
  You have not 
  specified what type of lines you wish to use, POTS, PRI, T1-CAS, E1, ISDN/BRI 
  ???
   
  
  
  
  
  
  From: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED] On Behalf Of Walid AzabSent: Wednesday, January 12, 2005 5:11 
  AMTo: 'Asterisk Users 
  Mailing List - Non-Commercial Discussion'Subject: [Asterisk-Users] What's the 
  easiest way to get * to call PSTN?
   
  
  Hi,
  
   
  
  I just want to know what is the 
  easiest way to have Asterisk route calls to PSTN. Hope any one can help 
  me.
  
   
  
  PS: Any solution using a Cisco 
  device is preferable.
  
   
  
   
  
  Thanks
  
  Walid
  
  

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[Asterisk-Users] SIP Authenication (Simple, Digest, ACL)

[Paul Belanger]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello,
I have been successful in getting Digest authentication to work with my
Mitel 5055 IP Phones, however I'm wondering if Asterisk still supports
Simple authentication?  I know it has been depreciated in the RFC, but I
have some phones with don't support Digest.  Or how about an ACL?
PB
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Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org
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[Asterisk-Users] linphone -> NAT -> * -> NAT -> firefly woes.

[Bruno Hertz]

Hi folks

an issue I don't understand. I'm running * stable 1.0.3 on public
internet, with following iax.conf / sip.conf entries:

iax.conf

 [100]
 type=friend
 username=Foo
 context=default
 auth=md5,plaintext,rsa
 secret=secret
 host=dynamic
 callerid="Foo" <100>
 qualify=no

sip.conf

 [10]
 type=friend
 username=Bar
 context=default
 callerid=Bar <10>
 host=dynamic
 secret=secret
 nat=yes
 canreinvite=no

On iax exten 10 I register firefly, on sip exten 100 linphone,
both behind nat.

Now, calls I can do is e.g.
firefly -> * -> linphone
linphone -> * echo test (copied this from demo and put it on exten 600)

but what wouldn't properly work is is sip to iax bridging
linphone -> * -> firefly

More specifically, firefly rings properly, but when I press Accept
it just keeps ringing, and finally * tells me that linphone didn't
send any frames:

channel.c:2655 ast_channel_bridge: Didn't get a frame from channel: SIP/10-e8bd
Jan 12 21:55:02 DEBUG[22661]: channel.c:2725 ast_channel_bridge: Bridge stops 
bridging channels SIP/10-e8bd and IAX2/100/2

Doing my tcpdumps I checked that there's really no data sent by linphone,
while nothing is dropped by firewalls either.

Did anyone experience similar troubles? A hint about how to resolve or further
debug this would sure be appreciated.

Another point I'm wondering about is why, in that same connection, the
caller id handed to firefly is just "10", and not the one specified
in sip.conf, i.e. "Bar <10>".

I tested all that stuff also with iaxcomm, i.e. pure iax bridging
iaxcomm -> NAT -> * -> NAT -> firefly
and here, everything works OK, calls in both ways and caller id
transmission.

Thanks, Bruno.


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Re: [Asterisk-Users] MySQL Realtime Driver

[Matthew Boehm]

You can use RealTime to store the mgcp.conf file. This does not get you
"realtime" abilities as you still need to reload mgcp when u make a change.

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

Sent: Wednesday, January 12, 2005 2:25 PM
Subject: Re: [Asterisk-Users] MySQL Realtime Driver


> > On Fri, 10 Dec 2004 10:01:00 -0600, Matthew Boehm <[EMAIL PROTECTED]>
> > wrote:
> > > > Yes and No. You need to realize this isn't "Asterisk and MySQL".
This
> > is
> > > > "Asterisk and RealTime using MySQL".  You can also have "Asterisk
and
> > > > RealTime using ODBC" etc..
> > > >
> > > > It is NOT the database that supports features. It is RealTime that
> > supports
> > > > features.
> > > >
> > > > RealTime is still in DEVELOPMENT. and more apps are slowly being
added
> > with
> > > > RealTime abilities.
> > > >
> > > > Currently, the only officially supported RealTime configs are
> > "sipfriends",
> > > > "iaxfriends", "voicemail" and "extensions".
>
> Any thoughts on whether their will be realtime support for mgcp.conf (on
> a line by line basis) in the near future.
>
> Regards
> MIKE
>
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[Asterisk-Users] Using asterisk to convert H.323 to SIP?

[Paul Belanger]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello all,
I was looking for some information about using Asterisk to convert an
incoming H.323 call to and outgoing SIP call.  Is this possible?
PB
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.2.5 (MingW32)
Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org
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gcBg9EdQqJWuIvp9mr53sXE=
=P8Mc
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[Asterisk-Users] calling an extension after a voicemail is left

[Joel Duffield]

Hi All

I am setting up * for use as a voicemail. I have discovered that if I dial the 
phone system and send "#+Extension+messagenumber" (dtmf) that the "msg" light 
will 
come on on the phones, if 00 messages the light will go off. they are an old 
tie 
onyx vs system. So how can I get asterisk to pick up a line and send these 
digits 
after a voicemail has been left, even if the person hangs up the phone? would I 
need a script to do this, I'm a noobie at writing scripts? Any experience or 
advice 
would be greatly appreciated.

Thanks

Joel Duffield



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