Re: [Asterisk-Users] [EMAIL PROTECTED] and Cisco 7910

[Sergio Chersovani]

Javier Chia ha scritto:


Does anybody knows how to get this work?


ftp://ftp.berlios.de/pub/chan-sccp/chan_sccp-20050707.tar.gz
extract
cd
make
make install

edit the sccp.conf

Sergio
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[Asterisk-Users] h323 how to ?????

[Ronald_Wiplinger]

I try to get H323 to run, but have so far only partial success:

There is a Gatekeeper GK, where asterisk connects to.

The Gatekeeper sees Asterisk, and Asterisk sees the gatekeeper.

From the Network on the GK, asterisk is reachable via the number 
07033. I have an extension on asterisk 6002, which is reachable.


I try to call a number attached to the gatekeeper (070168177) with the 
dialing plan:


exten => _9070.,1,Set(CALLERID(number)=07033${CALLERIDNUM})
exten => _9070.,n,Dial(H323/${EXTEN:${TRUNKMSD}})
exten => _9070.,n,Hangup

CLI> shows:
*CLI>
   -- Executing Set("SIP/6002-9fac", "CALLERID(number)=070336002") 
in new stack

   -- Executing Dial("SIP/6002-9fac", "H323/070168177") in new stack
   -- Called 070168177
 == No one is available to answer at this time (1:0/0/0)
   -- Executing Hangup("SIP/6002-9fac", "") in new stack
 == Spawn extension (from-sip, 9070168177, 3) exited non-zero on 
'SIP/6002-9fac'


The gatekeeper sees nothing from that. I guess the syntax is wrong for 
dialing. How should it be?





Video connection:
I try to call from an H323 soft phone through the gatekeeper to call the 
extension 6003 (eyebeam)


H323 soft phone calls through GK Asterisk box without webcam installed:

   -- Executing Dial("H323/203.160.252.147-a44c", "SIP/8600") in new stack
Jul  8 13:51:37 WARNING[12674]: chan_sip.c:1742 create_addr: No such 
host: 8600
Jul  8 13:51:37 NOTICE[12674]: app_dial.c:977 dial_exec_full: Unable to 
create channel of type 'SIP' (cause 3)

 == Everyone is busy/congested at this time (1:0/0/1)
   -- Executing Answer("H323/203.160.252.147-a44c", "") in new stack
   -- Executing SetVar("H323/203.160.252.147-a44c", "TIMEOUT(digit)=5") 
in new stack
Jul  8 13:51:37 WARNING[12674]: pbx.c:5754 pbx_builtin_setvar_old: 
SetVar is deprecated, please use Set instead.

   -- Digit timeout set to 5
   -- Executing SetVar("H323/203.160.252.147-a44c", 
"TIMEOUT(response)=10") in new stack

   -- Response timeout set to 10
   -- Executing BackGround("H323/203.160.252.147-a44c", 
"demo-congrats") in new stack

   -- Playing 'demo-congrats' (language 'en')
 == CDR updated on H323/203.160.252.147-a44c
   -- Executing Dial("H323/203.160.252.147-a44c", "SIP/6003|60|trm") in 
new stack

   -- Called 6003
   -- Started music on hold, class 'default', on H323/203.160.252.147-a44c
   -- SIP/6003-e756 is ringing
   -- SIP/6003-e756 answered H323/203.160.252.147-a44c
   -- Stopped music on hold on H323/203.160.252.147-a44c
   -- Attempting native bridge of H323/203.160.252.147-a44c and 
SIP/6003-e756
Jul  8 13:52:16 WARNING[12674]: chan_sip.c:3203 process_sdp: Unknown SDP 
media type in offer: video 7156 RTP/AVP 105 34
Jul  8 13:52:16 WARNING[12674]: chan_h323.c:914 h323_indicate: Don't 
know how to indicate condition 17 on ooh323c_1
Jul  8 13:52:21 WARNING[12674]: chan_h323.c:914 h323_indicate: Don't 
know how to indicate condition 17 on ooh323c_1


No connection, not even audio!

sip.conf settings for 6003:

[6003]
type=friend
username=6003
secret=pwd
qualify=200
nat=yes
host=dynamic
canreinvite=yes
context=from-sip
callerid=Ronald Wiplinger <6003>  ; Full caller
dtmfmode=rfc2833
disallow=all
allow=ulaw
allow=alaw
allow=h261
allow=h263
allow=h263p 


Xten's settings:
Enable this SIP account
Display name:   Ronald at Leadtek
User name:  6003
Password: password
Authorization:  6003
Domain: 59.120.139.119

Domain Proxy:
x Register with domain

STUN server
x Manual override:stun.xten.com


Any hints are welcome


bye

Ronald




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RE: [Asterisk-Users] Re: Braodvoice - UK Non Geographic Numbers

[Andreas Sikkema]

[EMAIL PROTECTED] wrote:

> http://www.serviceview.bt.com/list/current/docs/Call_Charges.boo/00165.htm
> Of course these are BT retail rates but I fully expect wholesale
> rates based on call prefix will be available for carriers / ITSP

In some countries there's a company (companies?) providing access 
to a database which telcos can use to find the rates on this kind 
of numbers. 

-- 
Andreas Sikkema bbned NV
Van Vollenhovenstraat 33016 BE Rotterdam
t: +31 (0)10 2245544  f: +31 (0)10 413 65 45
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Re: [Asterisk-Users] IAX Transfers

[Stephen]

i have a similar problem and found out that # transfer will not work
only when two IAX phone are using the same codec. 

Moises Silva wrote:

  what does asterisk says in the console when you try to transfer from
the "buggy" phones??
asterisk -vvr

On 7/7/05, Brent Davidson <[EMAIL PROTECTED]> wrote:
  
  
I'm having a strange problem with transfers on IAX phones.  I have two
IAX phones behind my firewall that are extensions from my office phone
system.  Both phones can receive calls, but only one of the extensions
can do blind transfers by pressing the # key.  I have a similar problem
at the office.  Some of the phones can transfer calls, some of them
can't.  And my Zap lines can always transfer.

I have all of my IAX extensions configured exactly the same way in
iax.conf.  All handsets are configured the same way and runnign the same
firmware.  I thought at first that it was a problem with NAT, but none
of the office phones are behind firewalls.

Any ideas?

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Re: [Asterisk-Users] Sipura SPA-841 Volume Oscillation Problem

[Vahan Yerkanian]

Greetings,

I'm experiencing the same problem. It manifests itself mostly in noisy 
environments - as soon as there is some increase of the ambient noise 
the volume in the headpiece or the "speakerphone" decreases immediately, 
and starts to randomly increase/decrease for some time after the ambient 
noise gets low. This is 100% repeatable if you start the conversation by 
using speakerphone. As soon as you switch to the handset, the defect 
disappears. Now the problem is that 5% of calls via headset have the 
same problem.


I am using the latest firmware for the SPA-841.

Javier Ergas wrote:

Hi all,

 

The problem is on the volume of the voice sent by the SPA-841. I think 
the echo cancel algorithm sets a limit to the microphone when detects 
sounds or noise from the earphone. This problem generates an oscillation 
on the voice volume sent by the phone and even turns it off completely 
for very little lapses of time making the communication very 
uncomfortable. I manage three different implementations with Asterisk 
and Sipura SPA-841 on different clients and network topologies, and on 
every one we are experiencing the same situation.


 


Thanks,

jergas

 





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begin:vcard
fn:Vahan Yerkanian
n:Yerkanian;Vahan
org:ARMINCO Global Telecommunications;Head, Research & Development dept.
adr:;;28, Isahakian ave., PO BOX 10;Yerevan;;375009;Armenia
email;internet:[EMAIL PROTECTED]
x-mozilla-html:FALSE
url:http://www.arminco.com/
version:2.1
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[Asterisk-Users] Routing DID calls to external lines

[Syed Akbar]

I am trying to route incoming DID call (on a analog channel) through
Asterisk to an outside (analog) line. My extensions.conf is something like
the following:

exten => 500,1,Dial,Zap/g1/3105551010

In this case the incoming DID call extension is 500. I am able to dial out
and connect with the incoming call, however, the voice conversation is only
one way. The called party is not able to hear the calling party.

Does anyone have any suggestions of how to better route the incoming DID
calls to external POTS lines. Basically what I am trying to do is to forward
DID calls to specific external PSTN phone numbers. A one to one forwarding
scheme.

Syed Akbar

Alico Systems Inc
www.alicosystems.com
Tel: 562-436-1510 

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[Asterisk-Users] wholesale DID's?

[Plexicomm Admin]

We would like to use Asterisk to deploy VoIP to our broadband internet
access customers.
Which VoIP providers (that are reliable & stable) provide wholesale
DID's?

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Re: [Asterisk-Users] "Set" syntax equivalent of DBDel?

[Trevor Peirce]

Brian Capouch wrote:

Set(DB(family/key)=) sets the value for the key to null, but that 
doesn't appear to be equivalent to removing the key entirely.


Or maybe DBDel isn't deprecated, like the other two are.


It's not deprecated.  There is no code yet for a DBDel type function.



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Re: [Asterisk-Users] Net2Phone equipment and different VOIP providers

[Guillermo Salas M]

On Tue, 2005-06-28 at 16:14 -0700, bph wrote:
> Hello we are a small call center with only 8 lines we use max4 and the
> 2-2 port gateways from net2phone . There equipment is good but we are
> getting hit by lower cost competition. We need to be able to compete.
> We have a couple of providers who are 50% less in some cases even
> more. So it makes sense that we would like to be able to compete .
> Since we have spent quite a bit of money on existing equipment that is
> in good operating condition we don't want to just throw it away but
> rather use it with the new provider. Has anyone out there been able to
> use the Max or 2 port gateway on other service providers. If so we
> would greatly appreciate you letting us know how. We are not experts
> by any means but we will try anything that works . I hope someone
> outher can help.

You need to change the firmware of the n2p hardware.

Contact me off the list if you want the H323 firmware. I think there is
not SIP firmware for this equipement.

>  
> Julio Periz
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Re: [Asterisk-Users] Looking for link.exe to compile G729 codec

[Vaniah Voip]

Obelix wrote:

>Quoting Tony Hoyle <[EMAIL PROTECTED]>:
>
>After downloading the framework and reading the firefly G729 notes, it turns 
>out
>the program I want is lib.exe, which is not in the distribution.
>
>I have a lib.exe from a Visual Studio 6 CD. Will that suffice?
>
>I also found some instructions here
>http://sapdb.2scale.net/moin.cgi/MS_20C_2b_2b_20Toolkit which allow the lib.exe
>to be built from some files in this code below.
>
>@echo off
>if "%1" == "" goto nocmd
>echo %* > cmdline.tmp
>link /lib @cmdline.tmp
>del cmdline.tmp
>goto end
>:nocmd
>link /lib
>:end
>
>Are any of these options likely to work?
>
>In fact considering everything is it only Microsoft's lib.exe which can do the
>job?
>
>
>
>
>  
>
>>Obelix wrote:
>>
>>
>>>I want to compile the G729 codec to try it out with firefly.
>>>I don't have Visual C++ 6 compiler. Is there a way I can obtain the
>>>  
>>>
>>link.exe
>>
>>
>>>alone for use with cygwin, or a substitute program?
>>>
>>>I don't look forward to installing the whole Visual C++ just for the
>>>  
>>>
>>link.exe
>>
>>
>>The .net framework SDK apparently has it... just ignore all the .net
>>bits and use nmake/cl/link from it.
>>
>>Tony
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>>
>>
>
>
>
>
>
>This message was sent using IMP, the Internet Messaging Program.
>
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>  
>
Email me off list if you want the dll.
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Re: [Asterisk-Users] Looking for link.exe to compile G729 codec

[Obelix]

Quoting Tony Hoyle <[EMAIL PROTECTED]>:

After downloading the framework and reading the firefly G729 notes, it turns out
the program I want is lib.exe, which is not in the distribution.

I have a lib.exe from a Visual Studio 6 CD. Will that suffice?

I also found some instructions here
http://sapdb.2scale.net/moin.cgi/MS_20C_2b_2b_20Toolkit which allow the lib.exe
to be built from some files in this code below.

@echo off
if "%1" == "" goto nocmd
echo %* > cmdline.tmp
link /lib @cmdline.tmp
del cmdline.tmp
goto end
:nocmd
link /lib
:end

Are any of these options likely to work?

In fact considering everything is it only Microsoft's lib.exe which can do the
job?




> Obelix wrote:
> > I want to compile the G729 codec to try it out with firefly.
> > I don't have Visual C++ 6 compiler. Is there a way I can obtain the
> link.exe
> > alone for use with cygwin, or a substitute program?
> >
> > I don't look forward to installing the whole Visual C++ just for the
> link.exe
> >
> The .net framework SDK apparently has it... just ignore all the .net
> bits and use nmake/cl/link from it.
>
> Tony
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This message was sent using IMP, the Internet Messaging Program.

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RE: [Asterisk-Users] Asterisk/Grandstream Budgetone disconnect issue

[Storm D. J. Petersen]

Title: Asterisk/Grandstream Budgetone disconnect issue








Might want to try updating your firmware
on the Grandstream.  I use this version, works great: http://gs-firmware.gratissip.dk/firmwares/1.0.6.7/

 

Cheers,

 

 

S.









From:
[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
On Behalf Of Bates, Curtis
Sent: Thursday, July 07, 2005 7:13
AM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users]
Asterisk/Grandstream Budgetone disconnect issue



 

I
am setting up a small Asterisk system for use at home.  I have one
Budgetone 101, one Cisco 7960 and two Xten lite softphones.  So far
everything is working expect for an issue with the Budgetone.  When a call
is placed between the Budgetone and any other phone, the call is setup and
sounds good.  If I hang-up on the Budgetone, everything is ok.  If I
hang up on the other phone, the Budgetone give me a busy signal, and does not
hang up.  Nothing is showing up in the messages log.  Calls between
the other phones work ok.  Any ideas?

Thanks.




-
A.G. Edwards & Sons' outgoing and incoming e-mails are electronically
archived and subject to review and/or disclosure to someone other 
than the recipient.

-






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Re: [Asterisk-Users] Announce incoming callerID via paging/intercom?

[C F]

Search the lists it's all over every 2-3 weeks. Here is the what to
look for. the M option in dial application
(http://www.voip-info.org/wiki-asterisk+cmd+dial).

On 7/7/05, EddyG <[EMAIL PROTECTED]> wrote:
> Greetings!
> 
> I was wondering if it is possible (using something like a group of
> Sipura SPA-841 IP phones) to have * announce information about the
> calling party via the SPA-841's speaker to a selected set of
> extensions that aren't set to "Do Not Disturb"... i.e., have * say the
> number, or perhaps have Festival speak the name, etc.
> 
> If so, any hints and tips on how you'd go about configuring this?
> 
> I have some old Northern Telecom POTS phones (9516CW's) that do this
> and it is a pretty useful feature...
> 
> Thanks!
> Eddy
> 
> 
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Re: [Asterisk-Users] Extension Problems

[Richard Adamson]

Better read up on why a sip phone should register with asterisk. Do a 'sip show 
peers' and that will be the list of phones that can "receive" calls.
---

I've double checked this.  Everything is logging in fine, because I can 
make calls, check my voicemail, everything except recieve calls on the 
SIP devices.

David Phelan wrote:

> 
>
>
>
>From: [EMAIL PROTECTED]
>[mailto:[EMAIL PROTECTED] On Behalf Of Jeremi Bergman
>Sent: Friday, 8 July 2005 9:12 AM
>To: Asterisk Users Mailing List - Non-Commercial Discussion
>Subject: [Asterisk-Users] Extension Problems
>
>
>The extensions I've created in AAH, when dialed, always go straight to
>voicemail.
>
>I may be missing a step... I'm simply adding it in the "Extensions" part of
>AAH.
>
>I can dial out with my extension, and recieve the voicemail notification, 
so
>I know i'm logged in, or so I thought...
>
>This is SIP 210 logging in and 220 making a call to 210
>
>
> 
>
><--SNIP -->
>
> 
>
>Looks Like an Authentication Issue to me
>Chack the Username and password on the sip device and AAH
>
>Dave
>
>
>
>  
>

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RE: [Asterisk-Users] Problems to leave messages in Asterisk

[Carlos Alperin]

Try to change the format to

format=wav49|gsm|wav

It looks like you choose the wrong format. Standard it uses gsm.

Regards,

Carlos Alperin

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jack Towards
Sent: Thursday, July 07, 2005 8:35 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Problems to leave messages in Asterisk

Hi all!
  I'm using Asterisk as a SIP server with 2 ip phones and it
works great, the only think that I can´t make it wotk is:

1. I leave the messages but when I receive them in my mailbox , and
open them I hear only noises
 ---voicemail.conf---
[general]
format=wav49
servermail=asterisk
attach=yes
maxmessage=180
pbxskip=yes
fromstring=The Asterisk PBX

[default]

123 => 123,peter,[EMAIL PROTECTED],attach=yes
321 => 321,jack,[EMAIL PROTECTED],attach=yes


 


2. When I try to play an mp3 with MP3player in extensions.conf I only
hear noises


Any help will be Great!!


Thanks
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Re: [Asterisk-Users] Extension Problems

[Jeremi Bergman]

I've double checked this.  Everything is logging in fine, because I can 
make calls, check my voicemail, everything except recieve calls on the 
SIP devices.


David Phelan wrote:






From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jeremi Bergman
Sent: Friday, 8 July 2005 9:12 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Extension Problems


The extensions I've created in AAH, when dialed, always go straight to
voicemail.

I may be missing a step... I'm simply adding it in the "Extensions" part of
AAH.

I can dial out with my extension, and recieve the voicemail notification, so
I know i'm logged in, or so I thought...

This is SIP 210 logging in and 220 making a call to 210




<--SNIP -->



Looks Like an Authentication Issue to me
Chack the Username and password on the sip device and AAH

Dave



 



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Re: [Asterisk-Users] "Set" syntax equivalent of DBDel?

[Nicolás Gudiño]

> I found an unanswered mail in the archives that implied that perhaps
> there is no direct way to delete a DB entry with the new "Set" syntax.
> 
> Set(DB(family/key)=) sets the value for the key to null, but that
> doesn't appear to be equivalent to removing the key entirely.
> 
> Or maybe DBDel isn't deprecated, like the other two are.
> 
> Anyone know the score?

I was wondering the same thing myself... I guess dbdel is not
deprecated, and that is confusing because dbput is deprecated. Maybe
this should be posted to the -dev list because I don't think that
-users is being heavily monitored by Digium.

-- 
Nicolás Gudiño
Buenos Aires - Argentina
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Re: [Asterisk-Users] Emergency Asterisk Guru Help needed EMERGENCY

[Jeffrey Starin]

Perfect!

Thanks!

B.

Kevin P. Fleming wrote:


Jeffrey Starin wrote:


However, I don't believe a full restore is needed -- I just need to know
the names of the directories under /var/spool/asterisk and re-create
them (I hope!).  Can some kind soul give me some direction or tell me
the directory structure under /var/spool/asterisk?



Do a 'make samples' on another system or some other safe place, and 
copy over the contents of /var/spool/asterisk.

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[Asterisk-Users] Problems to leave messages in Asterisk

[Jack Towards]

Hi all!
  I'm using Asterisk as a SIP server with 2 ip phones and it
works great, the only think that I can´t make it wotk is:

1. I leave the messages but when I receive them in my mailbox , and
open them I hear only noises
 ---voicemail.conf---
[general]
format=wav49
servermail=asterisk
attach=yes
maxmessage=180
pbxskip=yes
fromstring=The Asterisk PBX

[default]

123 => 123,peter,[EMAIL PROTECTED],attach=yes
321 => 321,jack,[EMAIL PROTECTED],attach=yes


 


2. When I try to play an mp3 with MP3player in extensions.conf I only
hear noises


Any help will be Great!!


Thanks
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[Asterisk-Users] [EMAIL PROTECTED] and Cisco 7910

[Javier Chia]

Hi,
 
Does anybody knows how to get this work?
I have been searching all over and can´t find the way.
 
Using chan_sccp tge phone can log in and has line tone. But the problem is that it is not able to receive nor dial out.
 
Any help would be apprecieted.
 
Thanks
 
Javier
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RE: [Asterisk-Users] Extension Problems

[David Phelan]

 



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jeremi Bergman
Sent: Friday, 8 July 2005 9:12 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Extension Problems


The extensions I've created in AAH, when dialed, always go straight to
voicemail.

I may be missing a step... I'm simply adding it in the "Extensions" part of
AAH.

I can dial out with my extension, and recieve the voicemail notification, so
I know i'm logged in, or so I thought...

This is SIP 210 logging in and 220 making a call to 210


 

<--SNIP -->

 

Looks Like an Authentication Issue to me
Chack the Username and password on the sip device and AAH

Dave



-- 
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.323 / Virus Database: 267.8.10/43 - Release Date: 6/07/2005
 

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[Asterisk-Users] Force SIP Proxy use

[Jason Frisch]

Hi again,

I don't know if I am asking the wrong questions or just nobody knows, 
but I will try

again anyway because I am quickly running out of hair to pull out...

Is there any setting in asterisk that will force proxy-authentication on 
every call?


Please help :-(

Jason Frisch

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[Asterisk-Users] CDR Record question.

[Paul Dracevich]

I have two asterisk boxes, and both have a PSTN/POTS gateway
so I am able to dial into the phone local network. When I place a call on box a
to go to box b then into the pstn I want to be able
to send back to box a CDR record where that call ended at.

 

 <--->  < 
>  < -- > 

 

The PSTN connection has an IP address and a
identifier attached to it I want theses details passed back to CDR table on
point A








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[Asterisk-Users] Extension Problems

[Jeremi Bergman]

The extensions I've created in AAH, when dialed, always go straight to
voicemail.

I may be missing a step... I'm simply adding it in the "Extensions"
part of AAH.

I can dial out with my extension, and recieve the voicemail
notification, so I know i'm logged in, or so I thought...

This is SIP 210 logging in and 220 making a call to 210

---
asterisk1*CLI> 
 
Sip read: 
REGISTER sip:192.168.0.50 SIP/2.0 
Via: SIP/2.0/UDP 192.168.0.8:5060;branch=z9hG4bK-6681167b 
From: Jeremi Bergman [EMAIL PROTECTED]>;tag=49a471fb8d817603o0 
To: Jeremi Bergman [EMAIL PROTECTED]> 
Call-ID: [EMAIL PROTECTED] 
CSeq: 1 REGISTER 
Max-Forwards: 70 
Contact: Jeremi Bergman [EMAIL PROTECTED]:5060>;expires=3600 
User-Agent: Sipura/SPA3000-2.0.10(GWc) 
Content-Length: 0 
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER 
Supported: x-sipura 
 
 
12 headers, 0 lines 
Using latest request as basis request 
Sending to 192.168.0.8 : 5060 (non-NAT) 
Transmitting (no NAT): 
SIP/2.0 100 Trying 
Via: SIP/2.0/UDP 192.168.0.8:5060;branch=z9hG4bK-6681167b 
From: Jeremi Bergman [EMAIL PROTECTED]>;tag=49a471fb8d817603o0 
To: Jeremi Bergman [EMAIL PROTECTED]>;tag=as0373da7c 
Call-ID: [EMAIL PROTECTED] 
CSeq: 1 REGISTER 
User-Agent: Asterisk PBX 
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER 
Contact: [EMAIL PROTECTED]> 
Content-Length: 0 
 
asterisk1*CLI> 
to 192.168.0.8:5060 
Transmitting (no NAT): 
SIP/2.0 401 Unauthorized 
Via: SIP/2.0/UDP 192.168.0.8:5060;branch=z9hG4bK-6681167b 
From: Jeremi Bergman [EMAIL PROTECTED]>;tag=49a471fb8d817603o0 
To: Jeremi Bergman [EMAIL PROTECTED]>;tag=as0373da7c 
Call-ID: [EMAIL PROTECTED] 
CSeq: 1 REGISTER 
User-Agent: Asterisk PBX 
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER 
Contact: [EMAIL PROTECTED]> 
WWW-Authenticate: Digest realm="asterisk", nonce="04ab00ad" 
Content-Length: 0 
 
 
to 192.168.0.8:5060 
Scheduling destruction of call '[EMAIL PROTECTED]'
in 15000 ms 
asterisk1*CLI> 
 
Sip read: 
REGISTER sip:192.168.0.50 SIP/2.0 
Via: SIP/2.0/UDP 192.168.0.8:5060;branch=z9hG4bK-bde1320f 
From: Jeremi Bergman [EMAIL PROTECTED]>;tag=49a471fb8d817603o0 
To: Jeremi Bergman [EMAIL PROTECTED]> 
Call-ID: [EMAIL PROTECTED] 
CSeq: 2 REGISTER 
Max-Forwards: 70 
Authorization: Digest
username="210",realm="asterisk",nonce="04ab00ad",uri="sip:[EMAIL PROTECTED]",algorithm=MD5,response="4b5484b65bc24fc38c8cdff7684a9452"; 
Contact: Jeremi Bergman [EMAIL PROTECTED]:5060>;expires=3600 
User-Agent: Sipura/SPA3000-2.0.10(GWc) 
Content-Length: 0 
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER 
Supported: x-sipura 
 
 
13 headers, 0 lines 
Using latest request as basis request 
Sending to 192.168.0.8 : 5060 (non-NAT) 
Transmitting (no NAT): 
SIP/2.0 100 Trying 
Via: SIP/2.0/UDP 192.168.0.8:5060;branch=z9hG4bK-bde1320f 
From: Jeremi Bergman [EMAIL PROTECTED]>;tag=49a471fb8d817603o0 
To: Jeremi Bergman [EMAIL PROTECTED]>;tag=as0373da7c 
Call-ID: [EMAIL PROTECTED] 
CSeq: 2 REGISTER 
User-Agent: Asterisk PBX 
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER 
Contact: [EMAIL PROTECTED]> 
Content-Length: 0 
 
 
to 192.168.0.8:5060 
Transmitting (no NAT): 
SIP/2.0 200 OK 
Via: SIP/2.0/UDP 192.168.0.8:5060;branch=z9hG4bK-bde1320f 
From: Jeremi Bergman [EMAIL PROTECTED]>;tag=49a471fb8d817603o0 
To: Jeremi Bergman [EMAIL PROTECTED]>;tag=as0373da7c 
Call-ID: [EMAIL PROTECTED] 
CSeq: 2 REGISTER 
User-Agent: Asterisk PBX 
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER 
Expires: 3600 
Contact: [EMAIL PROTECTED]:5060>;expires=3600 
Date: Wed, 06 Jul 2005 15:53:45 GMT 
Content-Length: 0 
 
 
to 192.168.0.8:5060 
Scheduling destruction of call '[EMAIL PROTECTED]'
in 15000 ms 
11 headers, 2 lines 
Reliably Transmitting: 
NOTIFY sip:[EMAIL PROTECTED]:5060
SIP/2.0 
Via: SIP/2.0/UDP 192.168.0.50:5060;branch=z9hG4bK70c85742 
From: "Unknown" [EMAIL PROTECTED]>;tag=as60a8b668 
To: [EMAIL PROTECTED]:5060> 
Contact: [EMAIL PROTECTED]> 
Call-ID: [EMAIL PROTECTED] 
CSeq: 102 NOTIFY 
User-Agent: Asterisk PBX 
Event: message-summary 
Content-Type: application/simple-message-summary 
Content-Length: 42 
 
Messages-Waiting: no 
Voice-Message: 0/0 
(no NAT) to 192.168.0.8:5060 
Scheduling destruction of call '[EMAIL PROTECTED]' in
15000 ms 
asterisk1*CLI> 
 
Sip read: 
SIP/2.0 200 OK 
To: [EMAIL PROTECTED]:5060>;tag=e365bb5ba561bc23i0 
From: "Unknown" [EMAIL PROTECTED]>;tag=as60a8b668 
Call-ID: [EMAIL PROTECTED] 
CSeq: 102 NOTIFY 
Via: SIP/2.0/UDP 192.168.0.50:5060;branch=z9hG4bK70c85742 
Server: Sipura/SPA3000-2.0.10(GWc) 
Content-Length: 0 
 
 
8 headers, 0 lines 
Destroying call '[EMAIL PROTECTED]' 
Destroying call '[EMAIL PROTECTED]' 
asterisk1*CLI> 
asterisk1*CLI> 
 
Sip read: 
INVITE sip:[EMAIL PROTECTED]
SIP/2.0 
Via: SIP/2.0/UDP
192.168.0.2:5060;rport;branch=z9hG4bKCF6A42E8A4F84FF1B1FE8F2D8EA78724 
From: Jeremi Bergman [EMAIL PROTECTED]>;tag=2051996763 
To: [EMAIL PROTECTED]> 
Contact: [EMAIL PROTECTED]:5060> 
Call-ID: [EMAIL PROTECTED] 
CSeq: 44887 INVITE 
Max-Forwards: 70 
Content-Type: applicati

Re: [Asterisk-Users] MeetMe hardware dimensioning

[Denis Galvão - iSolve]

Hi William.

On 07 de jul de 2005, at 18:39, William Boehlke wrote:

If your users are business people they ratio to 1100 simultaneous  
business

calls and you will need  6-9 Lintel servers, again depending on the
conferencing load and the transcoding.


I think that I will be in this case. That is a PalTalk like project.

What is your opnion about the separation of the services? Would you  
use the 6-9 lintel to handle each one a separate service, or your  
plan is to have some redundancy?


What is the hardware configuration that you recomend for each server?  
Xeon 3Ghz each?


Thanks.

Denis.

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Re: [Asterisk-Users] changing "Nobody picked up in 30000 ms"

[Darren Wiebe]

Is this in the astcc.agi file?  I know I answered a question like that a 
few days ago but I don't know if this is the same.  If it is, let me 
know and I'll send you a patched file.


Darren Wiebe
[EMAIL PROTECTED]

wassim darwish wrote:


how to edit the time "3 ms" for ringing  to "4
ms", i ve tried but i dindt know how,so please help me please.



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[Asterisk-Users] Sipura SPA-841 Volume Oscillation Problem

[Javier Ergas]

Hi all,

 

The problem is on
the volume of the voice sent by the SPA-841. I think the echo cancel algorithm
sets a limit to the microphone when detects sounds or noise from the earphone. This
problem generates an oscillation on the voice volume sent by the phone and even
turns it off completely for very little lapses of time making the communication
very uncomfortable. I manage three different implementations with Asterisk and Sipura
SPA-841 on different clients and network topologies, and on every one we are
experiencing the same situation.

 

Thanks,

jergas

 






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Re: [Asterisk-Users] Pickup pin

[Ronald Wiplinger]

Moises Silva wrote:


i would do that with Agi application. To give you a clue, this is the
flow of events:

- some one calls 777666 ( the local extension )
- the first priority in 777666 calls Agi application 
- the agi application will connect to asterisk manager and will

Originate a call to the remote extension, so when the remote extension
answers will be asked to dial the conference NIP to join a specific or
random conference. Next, agi will Originate other call to the true
local extensión, but this one will not be asked to dial NIP to enter
the conference.
- agi ends, and here is where im not sure what to do, the easy thing
is just send the caller to the same conference without a NIP, and the
caller should wait to the local or remote extension to Join the
conference. But it can be a little bit boring to wait without knowing
what is going on. But that is a way i can bet it works.
- The alternative is to try to catch  some manager event that can tell
us when the local or remote extension has entered the conference, and
only then send the caller to the conference. otherwise just put him on
music on hold or something.

Hope it gives you an idea.
 



Nice idea, I got in the meantime another one, which would also serve the 
purpose:
Put the remote phone into a queue and let the remote phone register this 
agent when I arrive there, and unregister when I leave.

Agent can be setup with a pin.

What do you think about this approach?


bye

Ronald Wiplinger


best regards

On 6/30/05, Ronald Wiplinger <[EMAIL PROTECTED]> wrote:
 


I want to let an additional remote extension number ring, if a call
comes in to a local number.

However, this remote extension is in another company.

I would like to make only a signaling to the remote phone, forward the
local phone to a conference room (with pin). The remote phone should
call than the conference room an key in the pin to be able to talk to
the caller.
How can I do that? Or do you have another solution to key in a pin to be
able to pickup a call?


bye

Ronald

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RE: [Asterisk-Users] changing "Nobody picked up in 30000 ms"

[Jay Milk]

http://www.voip-info.org/wiki-Asterisk+cmd+dial

Is it April 1st again already?

> -Original Message-
> From: wassim darwish [mailto:[EMAIL PROTECTED] 
> Sent: Thursday, July 07, 2005 2:08 PM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] changing "Nobody picked up in 3 ms"
> 
> 
> how to edit the time "3 ms" for ringing  to "4
> ms", i ve tried but i dindt know how,so please help me please.
> 
> 
>   
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> Have fun online with music videos, cool games, IM and more. 
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Re: [Asterisk-Users] NOTICE[180235]: chan_zap.c:7358 pri_dchannel: PRI got event: 6 on Primary D-channel of span 1

[Adam Dobrin]

Also, around the same time, I isolated the IRQ that my zaptel cards were 
on. (so neither zaptel card shared its IRQ).  


you can see what IRQ's are in use with

lspci -vb

This is more likely to be the cause of the fix.


Adam Dobrin wrote:


Lance,

I was in a similar situation, though i was rec'ing the event 6 
message, i noticed no degradation of sound and so ignored it.  I've 
since removed a *load* of unused modules, and it appears that the 
message is no longer coming in.  I had read that some people were only 
getting the message after the machine had been up for a few days.. 
I'll check back then.


This is what i added to modules.conf:
noload => res_musiconhold.so
noload => pbx_wilcalu.so
noload => app_image.so
noload => app_url.so
noload => app_adsiprog.so
noload => app_getcpeid.so
noload => app_milliwatt.so
noload => app_zapateller.so
noload => app_festival.so
noload => app_lookupblacklist.so
noload => app_random.so
noload => app_ices.so
noload => app_nbscat.so
noload => app_zapras.so
noload => codec_adpcm.so
noload => cdr_sqlite.so



Lance Grover wrote:


Does anyone have comment on this?


I am getting:
NOTICE[180235]: chan_zap.c:7358 pri_dchannel: PRI got event: 6 on
Primary D-channel of span 1

on my asterisk box and it seems to be causing a poping sound in the
phones, I am wondering if anyone can shed some light on this.  I have
scanned the archives and get possibilities ranging form motherboards,
to pri, to loaded module problems.  Can someone tell me the best way
to start tracking this down?

--
Thanks,

Lance Grover
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RE: [Asterisk-Users] Long Distance

[Carlos Alperin]

Ok,

IT looks like you arrive to the theatre when the movie started 15 minutes
ago.

If your phones are going to be distributed on a large area (nationwide), yes
you are going to save a lot calling to your extensions, instead of a phone
number.

If you plan to connect your Asterisk to one VoIP Provider, the rate should
be flat nationwide (That is how we do).

If you plan to access the Public Network by yourself, there is no way to
save the costs of your connection to the PSTN.

Regards,

Carlos Alperin
Senior System Engineer
Seneca Communications, LLC
[EMAIL PROTECTED]

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Don Brearley
Sent: Thursday, July 07, 2005 3:33 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Long Distance


Hello Everyone,

Pardon me if im sounding like a total idiot, but im new to this and have to
ask.

Numerous people have been telling me that I will be able to somehow
do long distance calling for free when I roll out Asterisk.. and yet none
of them can explain to me how exactly that will be.

So.. I'll ask the community at large..  is this total BS or is there
actually a way to reduce my long distance charges by rolling this out?

I appreciate any info provided..  Thanks!

-Don Brearley


PS:  I'm planning on deploying asterisk on a 300-phone line campus, and this
is
all in the planning stage at this point.

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Re: [Asterisk-Users] IAX2 Trunking - CVS-Head

[Steve Kann]

Kris Boutilier wrote:

  
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]]On Behalf Of Clive
Sent: Thursday, July 07, 2005 2:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] IAX2 Trunking - CVS-Head

Is anyone successfully using iax2 trunking with CVS head ?

The reason I am asking is that I have heard there may be some audio 
problems, which I would like to know about before sending customer's 
calls over a iax2 trunked connection.


  
  {clip}

I have been using IAX2 trunking, combined with newjitterbuffer and trunktimestamps, since it came out earlier this year. At the moment I'm running HEAD-05/25/05 on all my servers and use g726 with trunk frequency of 40ms. All calls are Asterisk to Asterisk, no other IAX devices attached.
  

Cool :)  I'm happy, at least, to hear it's working well for people!  (I
guess no news is good news..).

-SteveK



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RE: RE : [Asterisk-Users] HowTo start DIAL by a sillenttrainingas Wfor modems

[Carlos Alperin]

The dialplan is on Extensions.conf

Except that there exist any #include additional to it.

Regards,

Carlos

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bill Wesson
Sent: Thursday, July 07, 2005 6:03 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: RE : [Asterisk-Users] HowTo start DIAL by a sillenttrainingas
Wfor modems

 
Hello list,

What file do you put the 'w' in?

Thanks,
--Bill



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling
aka ManxPower
Sent: Tuesday, June 28, 2005 12:54 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: RE : [Asterisk-Users] HowTo start DIAL by a sillent trainingas
Wfor modems

If you are dialing out an analog Zap FXO port then you use the "w" 
option.  Dial(Zap/1/ww5551212)

[EMAIL PROTECTED] wrote:

> Hello the list, hello Doug,
> 
> Thank you, but I don't see any correct reply in this page.
> I want to have a "silent header" of 1 or 2 seconds between to pick up 
> the line and before to start to sned the DTMF numbering, because my 
> RTC provider doesn't give the prompt tone or listen the DTMF before this
time.
> If I place a "normal" call, it fails, because the first tones of my 
> Asterisk numbering sequence are missed.
> 
> If any idea...
> 
> Best Regards,
> Francois BERGERET,
> France.
> 
> 
> -Message d'origine-
> De : [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] De la part de Doug 
> Lytle Envoyé : mardi 28 juin 2005 12:25 À : Asterisk Users Mailing 
> List - Non-Commercial Discussion Objet : Re: [Asterisk-Users] HowTo 
> start DIAL by a sillent training as Wfor modems
> 
> 
> 
> [EMAIL PROTECTED] wrote:
> 
> 
>>Hi all the list,
>>
>>I am searching how to insert few seconds of silence just before to 
>>send the DTMF sequence via a FXO WildCard X101P to PSTN.
>>
>>I remember that Hayes compatible modems knows a special character "W" 
>>that do a 1 sec pause. Is it possible to do something like this in 
>>DIAL line sequence ?
>>
>> 
>>
> 
> This should help:
> 
> http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+Dial
> 
> 
> Doug
> 
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RE: [Asterisk-Users] IAXphone -> ip address -> extension number.

[Carlos Alperin]

The only reason for try SIP, is to find where the problem is.

You can use what you prefer, if you can made it works.

Any chance to see both SIP & IAX.conf?

Thanks,

Carlos

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Zoltan Szecsei
Sent: Thursday, July 07, 2005 6:01 PM
To: Time Bandit; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] IAXphone -> ip address -> extension number.


>My opinion is to keep using IAX, because, like you concluded, it's a
>better protocol.
>
>hth
>  
>


hth? - well, only if you can give me some pointers as to what I should 
be looking at to make it work :-)

(all right, yes it does help: you've given me confidence in my 
conviction, but not helped in making me realise  where I've been dorf. :-) )

zoltan.

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Re: [Asterisk-Users] NOTICE[180235]: chan_zap.c:7358 pri_dchannel: PRI got event: 6 on Primary D-channel of span 1

[Adam Dobrin]

Lance,

I was in a similar situation, though i was rec'ing the event 6 message, 
i noticed no degradation of sound and so ignored it.  I've since removed 
a *load* of unused modules, and it appears that the message is no longer 
coming in.  I had read that some people were only getting the message 
after the machine had been up for a few days.. I'll check back then.


This is what i added to modules.conf:
noload => res_musiconhold.so
noload => pbx_wilcalu.so
noload => app_image.so
noload => app_url.so
noload => app_adsiprog.so
noload => app_getcpeid.so
noload => app_milliwatt.so
noload => app_zapateller.so
noload => app_festival.so
noload => app_lookupblacklist.so
noload => app_random.so
noload => app_ices.so
noload => app_nbscat.so
noload => app_zapras.so
noload => codec_adpcm.so
noload => cdr_sqlite.so



Lance Grover wrote:


Does anyone have comment on this?


I am getting:
NOTICE[180235]: chan_zap.c:7358 pri_dchannel: PRI got event: 6 on
Primary D-channel of span 1

on my asterisk box and it seems to be causing a poping sound in the
phones, I am wondering if anyone can shed some light on this.  I have
scanned the archives and get possibilities ranging form motherboards,
to pri, to loaded module problems.  Can someone tell me the best way
to start tracking this down?

--
Thanks,

Lance Grover
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RE: RE : [Asterisk-Users] HowTo start DIAL by a sillent trainingas Wfor modems

[Bill Wesson]

 
Hello list,

What file do you put the 'w' in?

Thanks,
--Bill



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling
aka ManxPower
Sent: Tuesday, June 28, 2005 12:54 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: RE : [Asterisk-Users] HowTo start DIAL by a sillent trainingas
Wfor modems

If you are dialing out an analog Zap FXO port then you use the "w" 
option.  Dial(Zap/1/ww5551212)

[EMAIL PROTECTED] wrote:

> Hello the list, hello Doug,
> 
> Thank you, but I don't see any correct reply in this page.
> I want to have a "silent header" of 1 or 2 seconds between to pick up 
> the line and before to start to sned the DTMF numbering, because my 
> RTC provider doesn't give the prompt tone or listen the DTMF before this
time.
> If I place a "normal" call, it fails, because the first tones of my 
> Asterisk numbering sequence are missed.
> 
> If any idea...
> 
> Best Regards,
> Francois BERGERET,
> France.
> 
> 
> -Message d'origine-
> De : [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] De la part de Doug 
> Lytle Envoyé : mardi 28 juin 2005 12:25 À : Asterisk Users Mailing 
> List - Non-Commercial Discussion Objet : Re: [Asterisk-Users] HowTo 
> start DIAL by a sillent training as Wfor modems
> 
> 
> 
> [EMAIL PROTECTED] wrote:
> 
> 
>>Hi all the list,
>>
>>I am searching how to insert few seconds of silence just before to 
>>send the DTMF sequence via a FXO WildCard X101P to PSTN.
>>
>>I remember that Hayes compatible modems knows a special character "W" 
>>that do a 1 sec pause. Is it possible to do something like this in 
>>DIAL line sequence ?
>>
>> 
>>
> 
> This should help:
> 
> http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+Dial
> 
> 
> Doug
> 
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--
Always do right. This will gratify some people and astonish the rest.
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Re: [Asterisk-Users] IAXphone -> ip address -> extension number.

[Zoltan Szecsei]

My opinion is to keep using IAX, because, like you concluded, it's a
better protocol.

hth
 




hth? - well, only if you can give me some pointers as to what I should 
be looking at to make it work :-)


(all right, yes it does help: you've given me confidence in my 
conviction, but not helped in making me realise  where I've been dorf. :-) )


zoltan.

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[Asterisk-Users] Announce incoming callerID via paging/intercom?

[EddyG]

Greetings!

I was wondering if it is possible (using something like a group of
Sipura SPA-841 IP phones) to have * announce information about the
calling party via the SPA-841's speaker to a selected set of
extensions that aren't set to "Do Not Disturb"... i.e., have * say the
number, or perhaps have Festival speak the name, etc.

If so, any hints and tips on how you'd go about configuring this?

I have some old Northern Telecom POTS phones (9516CW's) that do this
and it is a pretty useful feature...

Thanks!
Eddy


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Re: [Asterisk-Users] Re: Braodvoice - UK Non Geographic Numbers

[Steve Kennedy]

On Thu, Jul 07, 2005 at 03:42:54PM -0400, Mark Phillips wrote:

> My take on this is that they are protecting themselves against fraud.
> Discounting the freefone numbers for a while, the "national rate" 
> numbers are charged at variying rates and so how is a company to know 
> just what they are gonna get charged.

0845 numbers aren't variable - they're local rate.
0844 can vary but towards zero.
087  can vary, but the providers should publish the rates.

Steve

-- 
NetTek Ltd  Fax +44-(0)20 7483 2455
Skype / In  stevekennedyuk / UK +442088167166 / US +13106518226
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RE: [Asterisk-Users] problem with iax2 and 2 peers behind nat

[Carlos Alperin]

That is correct. That should work, can you
send me more info about extensions.conf. Something looks to be missing here.

 

Thanks,

 

Carlos

 









From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of SAT MADRID
Sent: Thursday, July 07, 2005 1:44
PM
To: Asterisk
 Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] problem
with iax2 and 2 peers behind nat



 



Hi carlos, the dialplan its the same, i have only change the
line dial[sip/peer] by dial[aix2/peer].







- Original Message - 





From: Carlos
Alperin 





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





Sent: Thursday, July 07,
2005 6:51 PM





Subject: RE:
[Asterisk-Users] problem with iax2 and 2 peers behind nat





 



Do you have different dialplan for IAX
& SIP?, that shoudn’t depend on the protocol used.

 









From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of SAT MADRID
Sent: Thursday, July 07, 2005
12:27 PM
To: Asterisk
 Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users]
problem with iax2 and 2 peers behind nat



 



HI all, thanks Carlos, now its all working, but i have other
cuestion, how y transfer call to other peer, when i try sip y do it pressing
the # key but with iax it is not working.







- Original Message - 





From: Carlos
Alperin 





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





Sent: Wednesday, July
06, 2005 7:06 PM





Subject: RE:
[Asterisk-Users] problem with iax2 and 2 peers behind nat





 



Juan, 

 

That is not going to work. Asterisk
shouldn’t be behind a NAT to get registration of boxes behind NAT.

 

Put the asterisk on DMZ zone of their
router to make that happen.

 

Carlos Alperin

[EMAIL PROTECTED]

 









From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of SAT MADRID
Sent: Wednesday, July 06, 2005
12:52 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] problem
with iax2 and 2 peers behind nat



 



 





 





Hi all,





 





i have a problem with 2 peers conecting to an asterisk
machine, both are conected behind nat without any port mapping in the router,
and the * is conected behind other nat with the port 4569 mapped to it address,
the problem is:





 





when a peer register to the asterisk the other cant register
and viceversa, only gets registration the first one, im using firefly and a
hardphone from wuchuan, itried with 2 firefly and the error its the same, it
could be because the 2 peers are going to the internet with the same ip
addres(both behind nat)? if i conect both peers in the same lan there is no
problem so i think it cpuld be a problem with nat, i dont konw if i had to
change some configuration in iax.conf.





 





Thanks.





 





Juan Lopez.





 [EMAIL PROTECTED]




 
  
  
  Mensaje analizado y protegido, tecnologia antivirus www.trendmicro.es
  
 


 


 
  
  _
  Mensaje analizado y protegido, tecnologia antivirus www.trendmicro.es
  
 








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RE: [Asterisk-Users] IAX2 Trunking - CVS-Head

[Kris Boutilier]

> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Clive
> Sent: Thursday, July 07, 2005 2:08 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] IAX2 Trunking - CVS-Head
> 
> Is anyone successfully using iax2 trunking with CVS head ?
> 
> The reason I am asking is that I have heard there may be some audio 
> problems, which I would like to know about before sending customer's 
> calls over a iax2 trunked connection.
> 
{clip}

I have been using IAX2 trunking, combined with newjitterbuffer and 
trunktimestamps, since it came out earlier this year. At the moment I'm running 
HEAD-05/25/05 on all my servers and use g726 with trunk frequency of 40ms. All 
calls are Asterisk to Asterisk, no other IAX devices attached.

I don't have IAX native transfers enabled as it seemed to be suspect in my 
environment - needs further research.

Everything has been working fine for quite a while now, though I've not tried a 
very recent edition of head.

Hope that helps.

Kris Boutilier
Information Services Coordinator
Sunshine Coast Regional District
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RE: [Asterisk-Users] MeetMe hardware dimensioning

[William Boehlke]

If your 3500 users are consumers representing perhaps 175 simultaneous calls
you should be fine with three servers. Depending on conferencing load and
transcoding, two may be enough. You should have n+1 servers for redundency,
all with RAID. Then you don't need to worry about subsystem failures. 

If your users are business people they ratio to 1100 simultaneous business
calls and you will need  6-9 Lintel servers, again depending on the
conferencing load and the transcoding. 

William Boehlke
Signate



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Denis Galvão -
iSolve
Sent: Thursday, July 07, 2005 2:24 PM
To: Asterisk Users
Subject: [Asterisk-Users] MeetMe hardware dimensioning

Hi all.

What is the best hardware configuration to handle this following scenario?

- 4 IVR menu with conference applications for each option;
- Only SIP/g711 user access
- 3500 simultaneous users(800 at the beginning)
- No ZAP channels

Where is the most important point of failure? CPU? Ethernet? RAM?

Im planning to separate in three servers:

Server01: 01 Xeon 3Ghz getting the 1st level of the 4 IVR options.
Server02: 01 Xeon 3Ghz with 2 IVR suboptions and 2 conference room
Server03: 01 Xeon 3Ghz with 2 IVR suboptions and 2 conference room

How it sounds to you?

Denis.
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Re: [Asterisk-Users] Unable to forward frame/voice

[Christian M. Watts]

A brief update for those who are interested.

This turned out to be my problem (DOH!). After doing a PRI debug on the 
span on

the telco side to a log file, I found that the 'Unable to forward voice'
message corresponded one for one with the following progress message:

< Message type: PROGRESS (3)
< [08 02 84 81]
< Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0) 0: 0   Location:
Public network serving the remote user (4)
<  Ext: 1  Cause: Unallocated (unassigned) number (1), class =
Normal Event (0) ]


In other words, it looks like the failed calls are failing because the numbers
are invalid. I verified this with a regular POTS phone on a few, and that was,
in fact, the case.

A suggestion, if anyone is interested, would be that 'Unable to forward voice'
is maybe not the best message for app_dial to put in the asterisk log 
when this

condition occurrs. It would have been nice (and easier to figure out) if what
went into the log was something more along the lines of the actual cause for
what happened.

Thanks,
Christian


Quoting "Christian M. Watts" <[EMAIL PROTECTED]>:


Hi,

We've exhausted our internal capabilities as well as Sangoma tech support and
were hoping someone with some expertise could help us with a pointer. 
Briefly,

our issue is as follows.

Periodically (several times an hour), we get either of the following error
messages in our asterisk messages log. These correspond with dropped outbound
calls on a one-to-one basis when the second error happens. The first error
sometimes causes a dropped call and sometimes does not:

Jun 30 16:40:27 WARNING[5395] app_dial.c: Unable to forward frame
Jun 30 16:45:07 WARNING[5455] app_dial.c: Unable to forward voice


Our hardware is as follows:

Compaq DL380 Dual PIII 1Ghz, 1.2 GB RAM, Onboard SmartArray for SCSI RAID
Sangoma A102U dual-port T1 card
Digi Datafire T1 fax/modem board


Our software is as follows:

Linux 2.4.30
Asterisk, Zaptel and Libpri from CVS HEAD as of 6/28/05
Sangoma wanpipe 2.3.3-beta11 (latest as of this post)
Patton electronic's latest drivers and firmware for our Digi Datafire board
(still no 2.6 Linux support, which is why we're on 2.4)
Hylafax 4.2.1 driving the Digi Datafire


The path (for the problem calls) looks like this:

Digi Datafire -> Sangoma Port B -> Sangoma Port A -> Telco

Basically, sending a fax over a PRI with asterisk doing TDM bridging in the
middle.


We have confirmed the following (based on similar posts to this list 
related to

the same problem with Digium boards as well as Sangoma tech support
assistance):

1. Sangoma Port A takes clocking from the telco
2. Sangoma Port B retransmits A's clocking and acts as master
3. Sangoma tech support says our configs are correct
4. Zaptel.conf is set up with Sangoma Port A as the primary clock source, and
Port B to not be used as a clock source
5. LBO, switch options, etc. are correct for the environment (since 98% of
outbound calls are fine, this seems fairly obvious)
6. ISDN Transfer Capability gets properly set to 3K1AUDIO for calls
7. No IRQ sharing on the system
8. IDE DMA mode is irrelevant, since there are no IDE disks in the 
system (other

than the CDROM)


We have tried the following:

1. Asterisk, libpri and zaptel versions from 6/1/2005, 6/15/2005 and 
6/28/2005 -

no change in behavior
2. Wanpipe drivers 2.3.3-beta8 and 2.3.3-beta11 - no change in behavior
3. Wanpipe configured both with and without the D-Channel hardware HDLC - no
change in behavior
4. Firmware versions 24 (shipped version) and 25 (latest version) on 
the Sangoma

card - no change in behavior
5. callprogress and busydetect both 'yes' and 'no' in zapata.conf (currently
'no') - no change in behavior
6. Added SetTransferCapability(3K1AUDIO) to our dialplan, just to be 
sure - no

change in behavior


General environment:

1. We run TDM only, no VoIP protocols are in use. SIP, IAX2, MGCP are 
all noload

in modules.conf.
2. This problem occurs with as few as one simultaneous channel active and as
many as 15 simultaneous channels active with equal frequency (i.e.: not load
related). The load on the box is negligible in any case, plenty of 
RAM is free,

etc.
3. Restarting asterisk does seem to cause the problem not to 
re-present itself

for 30 minutes to 2 hours. When asterisk is restarted, the Sangoma and Zaptel
kernel modules are also unloaded and reloaded.


Again, any pointers or help would be greatly appreciated.

Thanks,
Christian
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--
Christian Watts
EC Data Systems, Inc.
303.991.6020 - Voice
303.991.6021 - Fax
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[Asterisk-Users] MeetMe hardware dimensioning

[Denis Galvão - iSolve]

Hi all.

What is the best hardware configuration to handle this following  
scenario?


- 4 IVR menu with conference applications for each option;
- Only SIP/g711 user access
- 3500 simultaneous users(800 at the beginning)
- No ZAP channels

Where is the most important point of failure? CPU? Ethernet? RAM?

Im planning to separate in three servers:

Server01: 01 Xeon 3Ghz getting the 1st level of the 4 IVR options.
Server02: 01 Xeon 3Ghz with 2 IVR suboptions and 2 conference room
Server03: 01 Xeon 3Ghz with 2 IVR suboptions and 2 conference room

How it sounds to you?

Denis.
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[Asterisk-Users] IAX2 Trunking - CVS-Head

[Clive]

Hi

Is anyone successfully using iax2 trunking with CVS head ?

The reason I am asking is that I have heard there may be some audio 
problems, which I would like to know about before sending customer's 
calls over a iax2 trunked connection.

Thanks in advance.
Clive

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Re: [Asterisk-Users] phone comparison matrix

[GLNX]

I was just doing that compilation for a comparative matrix a couple 
weeks ago, to get a snapshot of models/features/price range


It's just a draft toy for personal use but it might be useful if I share 
it and more people help to fill it. If we setup a matrix table in 
voip-info wiki, i'll try my $0.01 releasing it


Patrick Fortin wrote:


Hi

Is there a phone comparison matrix I could consult

I have a series of features that I would like to evaluate on the most 
common phones on the market


example:

dual-ethernet
POE / direct power / both
number of lines
speed dials programmable buttons
BLF LEDS
Headset plug
conference call built in
hands free operation
display size
codecs
communication protocol (SIP, h.323)
price
availability
reliability
know bugs / limitations
asterisk compatibility

If someone has done this recently that would save me some time

Patrick



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Re: [Asterisk-Users] Retrieving number of messages in a mailbox by an application

[GLNX]

Several ways to check voicemail counters:

- with the CLI: "show voicemail users for "

- with dialplan applications: "HasVoiceMail" and "HasNewVoiceMail"  
(check "show application hasvoicemail" from Asterisk CLI for help)


- with AMI management commands: "Mailboxcount" and "MailboxStatus"

Regards

Ramin Nikaeen wrote:

 


Valued Colleagues,

 

Can anyone tell me how the asterisk keeps track of the number of 
existing old (read)


and new (unread) messages in a mailbox? Is there a database table or 
somewhere


else from which this data can be retrieved by an application?

 


Thanks

 


ramin



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Re: [Asterisk-Users] Re: Braodvoice - UK Non Geographic Numbers

[Gavin Hamill]

On Thursday 07 July 2005 20:42, Mark Phillips wrote:
> My take on this is that they are protecting themselves against fraud.
>
> Discounting the freefone numbers for a while, the "national rate"
> numbers are charged at variying rates and so how is a company to know
> just what they are gonna get charged.

In the UK, the charging bands for each band of numbers is publically available 
information - the service provider can know accurately what they will be 
billed for each charge band.

i.e.
http://www.serviceview.bt.com/list/current/docs/Call_Charges.boo/16321.htm

http://www.serviceview.bt.com/list/current/docs/Call_Charges.boo/00165.htm

Of course these are BT retail rates but I fully expect wholesale rates based 
on call prefix will be available for carriers / ITSP

Cheers,
Gavin.
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Re: [Asterisk-Users] IAXphone -> ip address -> extension number.

[Time Bandit]

> >What about define those phones on the SIP.conf and use sip, instead of IAX.
> >That protocol use be more used to communicate Asterisk servers more than
> >phones.
That's not totally true. An IAX softphone will work easily behind a
NAT/Firewall. The same can't be said for a SIP one. I've tested IAX
succesfully working behind 3 NAT, and all the 4 softphones where able
to place/receive calls. I really don't think you could do the same
with SIP.

> Ah - ok - I understood from the docs that IAX was better and, as the
> phone was capable of both, I've been trying to get it going via IAX.
My opinion is to keep using IAX, because, like you concluded, it's a
better protocol.

hth
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Re: [Asterisk-Users] Long Distance

[Matt Riddell]

Don Brearley wrote:

Hello Everyone,

Pardon me if im sounding like a total idiot, but im new to this and have to ask.

Numerous people have been telling me that I will be able to somehow
do long distance calling for free when I roll out Asterisk.. and yet none
of them can explain to me how exactly that will be.

So.. I'll ask the community at large..  is this total BS or is there
actually a way to reduce my long distance charges by rolling this out?


Yes and no.

If you have free internet and have Asterisk servers at both ends (of the 
call) then yes, the calls between them will be free.


What I and some others do is group together so that we can offer each 
other free dialing in our respective cities.


This is in New Zealand where local calls are free.

YMMV.

--
Cheers,

Matt Riddell
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[Asterisk-Users] zapata.conf reload

[John Riek]

I am using asterisk stable version 1.0.8.  Is it
correct to say that the only way to reload
zapdata.conf is to restart asterisk?

Thanks,
   John Riek




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Re: [Asterisk-Users] Re: Braodvoice - UK Non Geographic Numbers

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

On Thu, 2005-07-07 at 20:17 +0100, Gavin Hamill wrote:
> On Thursday 07 July 2005 19:55, Russell Horn wrote:
> 
> > I can call non-geographic numbers from my land line in the US, my
> > mobile phone and from any calling card I have tried.  This isn't an
> > issue with BT but with broadvoice and those they contract to supply
> > connections to the UK PSTN.
> 
>  If BroadVoice don't let you call national rate numbers, then use a 
> second ITSP for those routes, or switch completely. They're cheap and nasty, 
> but they do use IAX... 
> 
> http://www.call1899.co.uk/voip.php
> http://www.call1899.co.uk/voiprates.php
> 
> No signup fee, no contract, cheap rates, lukewarm reliability :) Suck it and 
> see - better than no service at all.

and a requirement you have a UK phone :(  or at least they did a couple
months ago..

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


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RE: [Asterisk-Users] Long Distance

[Dean Collins]

Don, you can do free calling from point a to point b over the internet
for free using either sip or iax.

So lets say you have a campus in LA and in NY, the only cost you will
incur making calls between the two (or transiting the network and
connecting to a NY number etc) is the cost of your internet bandwidth.

However don't let this discourage you, there are any other reasons to
move to an IP pabx, and many additional specific reasons for using
Asterisk.

Can I make a suggestion however in that whilst you could pick all of
this information over time a 300 extension project could benefit from
someone's expertise. 

My suggestion is you post your location and then have consultants in
your area contact you with proposals.

Cheers,
Dean


> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Don Brearley
> Sent: Thursday, 7 July 2005 3:33 PM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] Long Distance
> 
> 
> Hello Everyone,
> 
> Pardon me if im sounding like a total idiot, but im new to this and
have
> to ask.
> 
> Numerous people have been telling me that I will be able to somehow
> do long distance calling for free when I roll out Asterisk.. and yet
none
> of them can explain to me how exactly that will be.
> 
> So.. I'll ask the community at large..  is this total BS or is there
> actually a way to reduce my long distance charges by rolling this out?
> 
> I appreciate any info provided..  Thanks!
> 
> -Don Brearley
> 
> 
> PS:  I'm planning on deploying asterisk on a 300-phone line campus,
and
> this is
> all in the planning stage at this point.
> 
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RE: [Asterisk-Users] Long Distance

[Bill McLaughlin]

There are actually a number of ways to reduce or eliminate long distance
charges using Asterisk(or any number of other similar solutions out there).

  1) Set up two Asterisk PBX's connected via network at different locations,
and configure calls to be routed between them across the network rather than
through the normal phone network, this would allow callers in either
location to call out to the other area without long distance, as Asterisk
could route the calls across the network then out to the public network at
the remote site

  2) Find a VOIP provider that provides voice lines through an internet
connection, and offers unlimited long distance to the areas you need to call

  3) Find a telephone provider that offers an unlimited plan through the
PSTN and interface it to that (Can be done with any phone system, not just
Asterisk, obviously)

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Don Brearley
Sent: Thursday, July 07, 2005 1:33 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Long Distance


Hello Everyone,

Pardon me if im sounding like a total idiot, but im new to this and have to
ask.

Numerous people have been telling me that I will be able to somehow
do long distance calling for free when I roll out Asterisk.. and yet none
of them can explain to me how exactly that will be.

So.. I'll ask the community at large..  is this total BS or is there
actually a way to reduce my long distance charges by rolling this out?

I appreciate any info provided..  Thanks!

-Don Brearley


PS:  I'm planning on deploying asterisk on a 300-phone line campus, and this
is
all in the planning stage at this point.

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[Asterisk-Users] Parial Hang with cvs-HEAD and queues/agentcallbacklogin

[Edward Eastman]

Title: Parial Hang with cvs-HEAD and queues/agentcallbacklogin






Hi

Last night I upgraded an asterisk install from cvs of early this year to current cvs head and all seemed to be working OK, but now I’m having several problems which seem to be related to queues.  First off queues don’t work, there’s no error message, the channel just seems to hang – cli output as follows:

    -- Executing Answer("Local/[EMAIL PROTECTED],2", "") in new stack

Jul  7 20:09:46 WARNING[27638]: channel.c:640 channel_find_locked: Avoided initial deadlock for '0x86e3948', 10 retries!

    -- Executing Playback("Local/[EMAIL PROTECTED],2", "support-welcome") in new stack

    -- Local/[EMAIL PROTECTED],1 answered SIP/ed-1-fc54

    -- Playing 'support-welcome' (language 'en')

  == Spawn extension (itg, 800, 2) exited non-zero on 'Local/[EMAIL PROTECTED],2'

    -- Executing Set("SIP/ed-1-fc54", "CALLERID(name)=Support") in new stack

    -- Executing Queue("SIP/ed-1-fc54", "support|t|||180") in new stack 

When I hang up the dialling phone there is no cli ouput and “show channels” shows the channel as still there:

SIP/ed-1-93ce  (macro-queueinbound s    4   )  Up Queue support|t|||180

Calling an agent produces the same result, and “show agents” on the CLI produces no output.  We’re using dynamic agents with agentcallbacklogin.


Other calls seem to proceed OK, although it does seem to be rather slow – for instance 4 goto’s and a set callerid takes approx 6 seconds. This is a low load system using no more than 3-4% cpu normally and asterisk isn’t using an abnormal amount of cpu or memory.

Does anyone have any ideas what’s causing this, or how to set about debugging it further?

Many thanks

Ed


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Re: [Asterisk-Users] Long Distance

[Darren Wiebe]

I don't have all the answers.  You should be able to save money on LD 
because you can (in my experience) pick up substantially better rates 
for voip termination than typical pstn LD.  You can get plans from some 
providers that allow "unlimited" long distance but it is all a balancing 
act.


Hope this helps a little,

Darren Wiebe
[EMAIL PROTECTED]

Don Brearley wrote:


Hello Everyone,

Pardon me if im sounding like a total idiot, but im new to this and have to ask.

Numerous people have been telling me that I will be able to somehow
do long distance calling for free when I roll out Asterisk.. and yet none
of them can explain to me how exactly that will be.

So.. I'll ask the community at large..  is this total BS or is there
actually a way to reduce my long distance charges by rolling this out?

I appreciate any info provided..  Thanks!

-Don Brearley


PS:  I'm planning on deploying asterisk on a 300-phone line campus, and this is
all in the planning stage at this point.

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[Asterisk-Users] Asterisk Crashes after update

[sbrown]

After doing an update from SUSE 9.2 to 9.3 and Checking out the latest from 
CVS, Asterisk crashes on startup with an apparent MySQL 
(res_config_register) error: 


# asterisk -vvvgc > asterisk_startup_error1.log
asterisk: symbol lookup error:
/usr/lib/asterisk/modules/res_config_mysql.so: un
   defined symbol: ast_cust_config_register 


The log is shown below.  I've seen the posts from
1/25/05 and several more recent ones regarding this
same issue or a similar one with the 


ast_cust_config_register being undefined, however
reverting to that build of 1/24/05 does not solve the
problem in my case. 


Is there another issue with mySQL that may cause this
problem?  I'm using SUSE 9.3 on an Athlon 64 with 64
bit release 2.6 of Linux.  I've made sure that all the
ODBC and MySQL modules for SUSE 9.3 are installed. 

I'm a rank noob with * and would appreciate any help. 

Thanks!!! 

Log Pasted below for more info: 



  == Parsing
'/etc/asterisk/asterisk.conf': Found 


  == Parsing
'/etc/asterisk/extconfig.conf': Found 


  == Parsing
'/etc/asterisk/asterisk.conf': Found 

Asterisk CVS-HEAD, Copyright (C) 1999 - 2005 Digium. 

Written by Mark Spencer <[EMAIL PROTECTED]> 

= 


  == Parsing
'/etc/asterisk/logger.conf': Found 


Asterisk Event Logger Started
/var/log/asterisk/event_log 

Asterisk Dynamic Loader loading preload modules: 


  == Parsing
'/etc/asterisk/modules.conf': Found 


  == Manager registered action
Ping 


  == Manager registered action
Events 


  == Manager registered action
Logoff 


  == Manager registered action
Hangup 


  == Manager registered action
Status 


  == Manager registered action
Setvar 


  == Manager registered action
Getvar 


  == Manager registered action
Redirect 


  == Manager registered action
Originate 


  == Manager registered action
Command 


  == Manager registered action
ExtensionState 


  == Manager registered action
AbsoluteTimeout 


  == Manager registered action
MailboxStatus 


  == Manager registered action
MailboxCount 


  == Manager registered action
ListCommands 


  == Parsing
'/etc/asterisk/manager.conf': Found 


  == Parsing
'/etc/asterisk/cdr.conf': Not found (No such file or
directory)
Jul  6 21:32:24 NOTICE[8492]:
cdr.c:1162
do_reload: CDR simple logging
enabled. 


  == Parsing
'/etc/asterisk/rtp.conf': Found 


  == RTP Allocating from port
range 1 -> 2 

Asterisk PBX Core Initializing 

Registering builtin applications: 

 [AbsoluteTimeout] 


  == Registered application
'AbsoluteTimeout' 

 [Answer] 


  == Registered application
'Answer' 

 [BackGround] 


  == Registered application
'BackGround' 

 [Busy] 


  == Registered application
'Busy' 

 [Congestion] 


  == Registered application
'Congestion' 

 [DigitTimeout] 


  == Registered application
'DigitTimeout' 

 [Goto] 


  == Registered application
'Goto' 

 [GotoIf] 


  == Registered application
'GotoIf' 

 [GotoIfTime] 


  == Registered application
'GotoIfTime' 

 [ExecIfTime] 


  == Registered application
'ExecIfTime' 

 [Hangup] 


  == Registered application
'Hangup' 

 [NoOp] 


  == Registered application
'NoOp' 

 [Prefix] 


  == Registered application
'Prefix' 

 [Progress] 


  == Registered application
'Progress' 

 [ResetCDR] 


  == Registered application
'ResetCDR' 

 [ResponseTimeout] 


  == Registered application
'ResponseTimeout' 

 [

Re: [Asterisk-Users] Re: Braodvoice - UK Non Geographic Numbers

[Mark Phillips]

My take on this is that they are protecting themselves against fraud.

Discounting the freefone numbers for a while, the "national rate" 
numbers are charged at variying rates and so how is a company to know 
just what they are gonna get charged.




Gavin Hamill wrote:

On Thursday 07 July 2005 19:55, Russell Horn wrote:



I can call non-geographic numbers from my land line in the US, my
mobile phone and from any calling card I have tried.  This isn't an
issue with BT but with broadvoice and those they contract to supply
connections to the UK PSTN.



 If BroadVoice don't let you call national rate numbers, then use a 
second ITSP for those routes, or switch completely. They're cheap and nasty, 
but they do use IAX... 


http://www.call1899.co.uk/voip.php
http://www.call1899.co.uk/voiprates.php

No signup fee, no contract, cheap rates, lukewarm reliability :) Suck it and 
see - better than no service at all.


Cheers,
Gavin.
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--

Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
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[Asterisk-Users] Long Distance

[Don Brearley]

Hello Everyone,

Pardon me if im sounding like a total idiot, but im new to this and have to ask.

Numerous people have been telling me that I will be able to somehow
do long distance calling for free when I roll out Asterisk.. and yet none
of them can explain to me how exactly that will be.

So.. I'll ask the community at large..  is this total BS or is there
actually a way to reduce my long distance charges by rolling this out?

I appreciate any info provided..  Thanks!

-Don Brearley


PS:  I'm planning on deploying asterisk on a 300-phone line campus, and this is
all in the planning stage at this point.

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Re: [Asterisk-Users] Re: Braodvoice - UK Non Geographic Numbers

[Gavin Hamill]

On Thursday 07 July 2005 19:55, Russell Horn wrote:

> I can call non-geographic numbers from my land line in the US, my
> mobile phone and from any calling card I have tried.  This isn't an
> issue with BT but with broadvoice and those they contract to supply
> connections to the UK PSTN.

 If BroadVoice don't let you call national rate numbers, then use a 
second ITSP for those routes, or switch completely. They're cheap and nasty, 
but they do use IAX... 

http://www.call1899.co.uk/voip.php
http://www.call1899.co.uk/voiprates.php

No signup fee, no contract, cheap rates, lukewarm reliability :) Suck it and 
see - better than no service at all.

Cheers,
Gavin.
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[Asterisk-Users] changing "Nobody picked up in 30000 ms"

[wassim darwish]

how to edit the time "3 ms" for ringing  to "4
ms", i ve tried but i dindt know how,so please help me please.



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[Asterisk-Users] NOTICE[180235]: chan_zap.c:7358 pri_dchannel: PRI got event: 6 on Primary D-channel of span 1

[Lance Grover]

Does anyone have comment on this?


I am getting:
NOTICE[180235]: chan_zap.c:7358 pri_dchannel: PRI got event: 6 on
Primary D-channel of span 1

on my asterisk box and it seems to be causing a poping sound in the
phones, I am wondering if anyone can shed some light on this.  I have
scanned the archives and get possibilities ranging form motherboards,
to pri, to loaded module problems.  Can someone tell me the best way
to start tracking this down?

--
Thanks,

Lance Grover
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RE: [Asterisk-Users] URGENT: hardware spesifications needed

[Michael L Smith]

Who are you to decide what Information can and cannot be "legitimately be
sought here:?

Just curious.

--Mike


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rusty
Shackleford
Sent: Thursday, July 07, 2005 12:03 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] URGENT: hardware spesifications needed


> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Jason Frisch
> Sent: Wednesday, July 06, 2005 4:22 PM
> To: Jimmy Smith; Asterisk Users Mailing List - Non-Commercial 
> Discussion
> Subject: Re: [Asterisk-Users] URGENT: hardware spesifications needed
> 
> 
> 
> Come on now children. Is this not a place to share knowledge?

Well..., yes, and no. Information that isn't readily available elsewhere
may legitimately be sought here. However, when the question is of the
FAQ variety, and it is clear that the person asking it has not even
attempted to find the information for himself, then rude replies are not
out of line, IMO.

-- 
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.323 / Virus Database: 267.8.10/43 - Release Date:
07/06/2005
 

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Re: [Asterisk-Users] Braodvoice - UK Non Geographic Numbers

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

its not you, its their false advertising that makes you think you can
dial these (after all their rates page *still* claims they provide
service and that its unlimited based on plan).

There are threads on voxilla.com in the broadvoice forums, which have
chat logs between me and the CTO nathan stratton, along with a slew of
phone numbers of people at broadvoice (managers and such).  

Basically Nathan said they found a UK provider to terminate these calls
and then nothing ever happened, he did give me a credit on my account
becuase I was unable to call, something I am going to have to get again
since I still havent been able to call and NCFA/LCFA/FREE were the
reasons I choose broadvoice vs someone else. 

There is also an interesting thread there on the ownership structure of
broadvoice, and how broadvoice is a registered trademark of broadcomm
being used without permission by broadvoice.com.


On Thu, 2005-07-07 at 14:38 -0400, Russell Horn wrote:
> Since May 05 I have been unable to call any non-geographic number in
> the UK via Broadvoice. Thse are numbers such as the 0800 range (free
> to call) 087xx (local / national rate calls). Broadvoice support have
> been unhelpful, and can't say if there's any intention to fix this. A
> case has been upen since May 24 without any updates.
> 
> Is anyone else having this problem? Has anyone else spoken to
> broadvoice about it? Did you get any further? Is there any indication
> it might be resolved?
> 
> The last customer rep I spoke to recommended I close my account if I
> need to dial these numbers - I'd prefer to keep my phone number, but
> if all else fails...

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


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Re: [Asterisk-Users] FXO hangup Problem.....

[Moises Silva]

i have similar problem, but the sip phone just rings 1 or 2 more
times, not until the timeout expires. what is your config in
zapata.conf

specifically callprogress an busydetect parameters can help

best regards

On 7/7/05, Nahid Hossain <[EMAIL PROTECTED]> wrote:
>  
>  
> 
> Hello, 
> 
>   
> 
> I am getting problem for delay call hang-up with the below scenario: 
> 
>   
> 
> PSTN User (calling Party)---àPSTN Line à FXO with Asterisk
> Box-àSIP IP Phone (called party) 
> 
>   
> 
>   
> 
> I am using X100P card with my Asterisk-1.0.7 box. I am also using
> Zaptel-1.0.7 version. 
> 
>   
> 
> When PSTN user makes call to my PSTN line and after getting IVR, PSTN user
> dial my SIP IP Phone extension, as soon as PSTN user gets one ring back
> tone, PSTN user cut off the current call. But SIP IP Phone rings till its
> timeout. 
> 
>   
> 
> I would appreciate if anyone give me solution for the above case. 
> 
>   
> 
> Regards 
> 
> Nahid 
> 
>   
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[Asterisk-Users] Re: Braodvoice - UK Non Geographic Numbers

[Russell Horn]

Broadvoice could connect to non geographic numbers without difficulty
until the fourth week of May 2005.

I can call non-geographic numbers from my land line in the US, my
mobile phone and from any calling card I have tried.  This isn't an
issue with BT but with broadvoice and those they contract to supply
connections to the UK PSTN.

On 7/7/05, Michael Welter <[EMAIL PROTECTED]> wrote:
> Russell Horn wrote:
> > Since May 05 I have been unable to call any non-geographic number in
> > the UK via Broadvoice. Thse are numbers such as the 0800 range (free
> > to call) 087xx (local / national rate calls). Broadvoice support have
> > been unhelpful, and can't say if there's any intention to fix this. A
> > case has been upen since May 24 without any updates.
> > 
> > Is anyone else having this problem? Has anyone else spoken to
> > broadvoice about it? Did you get any further? Is there any indication
> > it might be resolved?
> > 
> > The last customer rep I spoke to recommended I close my account if I
> > need to dial these numbers - I'd prefer to keep my phone number, but
> > if all else fails...
> > 
> > Russell.
> > ___
> > Asterisk-Users mailing list
> > Asterisk-Users@lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
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> >http://lists.digium.com/mailman/listinfo/asterisk-users
> > 
> > 
> I lost a client because of this.  BT will not allow premium numbers to 
> be called from outside the UK.  I even tried it from an ITSP in the 
> Netherlands, and the call didn't go through :-(
> 
> The AT&T monopoly is gone.  Hopefully, BT's time will come--the sooner 
> the better.
> 
> 
>
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Re: [Asterisk-Users] IAX Transfers

[Moises Silva]

what does asterisk says in the console when you try to transfer from
the "buggy" phones??
asterisk -vvr

On 7/7/05, Brent Davidson <[EMAIL PROTECTED]> wrote:
> I'm having a strange problem with transfers on IAX phones.  I have two
> IAX phones behind my firewall that are extensions from my office phone
> system.  Both phones can receive calls, but only one of the extensions
> can do blind transfers by pressing the # key.  I have a similar problem
> at the office.  Some of the phones can transfer calls, some of them
> can't.  And my Zap lines can always transfer.
> 
> I have all of my IAX extensions configured exactly the same way in
> iax.conf.  All handsets are configured the same way and runnign the same
> firmware.  I thought at first that it was a problem with NAT, but none
> of the office phones are behind firewalls.
> 
> Any ideas?
> 
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RE: [Asterisk-Users] Newbie Question: Type of card

[Bates, Curtis]

Here is what I use:  
http://www.digitnetworks.com/store/product_info.php?cPath=22&products_id=28

I have used it with Slack, but now I am running it with FC4.

-Original Message-
From: Dan Adams [mailto:[EMAIL PROTECTED]
Sent: Thursday, July 07, 2005 12:50 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Newbie Question: Type of card


Hi, I am sorta a newbie to the asterisk community at least in the realm of 
hardware types. I was wondering, what type of card is used to allow asterisk, 
on a slackware installation to talk to a standard phone line so that asterisk 
can call out?

Dan
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Re: [Asterisk-Users] Braodvoice - UK Non Geographic Numbers

[Michael Welter]

Russell Horn wrote:

Since May 05 I have been unable to call any non-geographic number in
the UK via Broadvoice. Thse are numbers such as the 0800 range (free
to call) 087xx (local / national rate calls). Broadvoice support have
been unhelpful, and can't say if there's any intention to fix this. A
case has been upen since May 24 without any updates.

Is anyone else having this problem? Has anyone else spoken to
broadvoice about it? Did you get any further? Is there any indication
it might be resolved?

The last customer rep I spoke to recommended I close my account if I
need to dial these numbers - I'd prefer to keep my phone number, but
if all else fails...

Russell.
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I lost a client because of this.  BT will not allow premium numbers to 
be called from outside the UK.  I even tried it from an ITSP in the 
Netherlands, and the call didn't go through :-(


The AT&T monopoly is gone.  Hopefully, BT's time will come--the sooner 
the better.



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[Asterisk-Users] Braodvoice - UK Non Geographic Numbers

[Russell Horn]

Since May 05 I have been unable to call any non-geographic number in
the UK via Broadvoice. Thse are numbers such as the 0800 range (free
to call) 087xx (local / national rate calls). Broadvoice support have
been unhelpful, and can't say if there's any intention to fix this. A
case has been upen since May 24 without any updates.

Is anyone else having this problem? Has anyone else spoken to
broadvoice about it? Did you get any further? Is there any indication
it might be resolved?

The last customer rep I spoke to recommended I close my account if I
need to dial these numbers - I'd prefer to keep my phone number, but
if all else fails...

Russell.
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Re: [Asterisk-Users] aah and astcc

[Erick Weber V.]

Darren:

Thanks for your interest

I would like that once you have been verified you can use aah dial plan so 
you can get all the reports for the astcc calls


Thanks for your help

Erick Weber

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


Sent: Wednesday, July 06, 2005 8:26 PM
Subject: Re: [Asterisk-Users] aah and astcc


How exactly are you thinking.  So that a certain aah extension points to 
it or so that once you have been verified you can call aah extensions?


Darren

Erick Weber V. wrote:


Hello:

Does anyone know how to incorporate astcc to aah so it will use amah 
extensions.


Any help will be appreciate

Thanks

Erick W.

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[Asterisk-Users] TDMoE bandwidth and load

[mattf]

Hello,

We've just started using TDMoE(local T1s connecting between Asterisk servers
in the same building over the LAN) to connect a few of our high-availability
servers instead of using crossover T1 cables. The 3 servers we have
connected to each other over TDMoE are running just fine and we have no
audio quality issues or bandwidth issues, but I'm considering using TDMoE to
connect 8 other servers to a main server and was wondering if a single
ethernet interface on the Main server can handle the load of 8 dynamic spans
connecting to it from other Asterisk servers.

Does anyone have any experience with using TDMoE to run 8 virtual T1s on a
single Ethernet port?

Thanks,

MATT---
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Re: [Asterisk-Users] Newbie Question: Type of card

[MF Hulber]

Take a look here:

http://www.digium.com/index.php?menu=product_detail&category=hardware&product=TDM400P

MARK.

Dan Adams wrote:

Hi, I am sorta a newbie to the asterisk community at least in the realm of 
hardware types. I was wondering, what type of card is used to allow asterisk, 
on a slackware installation to talk to a standard phone line so that asterisk 
can call out?


Dan
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Re: [Asterisk-Users] Newbie Question: Type of card

[Robert Webb]

On Thu, 7 Jul 2005 10:49:32 -0700
 Dan Adams <[EMAIL PROTECTED]> wrote:
Hi, I am sorta a newbie to the asterisk community at 
least in the realm of 
hardware types. I was wondering, what type of card is 
used to allow asterisk, 
on a slackware installation to talk to a standard phone 
line so that asterisk 
can call out?


Dan


The link below gives you great information on the card you 
need. Look espicially close to the box with all the 
writing in it just below the URL to www.asterisk.org that 
starts with "For interconnection with digital and analog 
telephony equipment"


http://www.voip-info.org/wiki-Asterisk
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Re: [Asterisk-Users] IAXphone -> ip address -> extension number.

[Zoltan Szecsei]

Carlos Alperin wrote:


What about define those phones on the SIP.conf and use sip, instead of IAX.
That protocol use be more used to communicate Asterisk servers more than
phones.

Regards,

Carlos Alperin
 



Ah - ok - I understood from the docs that IAX was better and, as the 
phone was capable of both, I've been trying to get it going via IAX.


regards,
Zoltan

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[Asterisk-Users] Newbie Question: Type of card

[Dan Adams]

Hi, I am sorta a newbie to the asterisk community at least in the realm of 
hardware types. I was wondering, what type of card is used to allow asterisk, 
on a slackware installation to talk to a standard phone line so that asterisk 
can call out?

Dan
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Re: [Asterisk-Users] problem with iax2 and 2 peers behind nat

[SAT MADRID]

Hi carlos, the dialplan its the same, i have only
change the line dial[sip/peer] by dial[aix2/peer].

  - Original Message - 
  From:
  Carlos
  Alperin 
  To: 'Asterisk Users Mailing List -
  Non-Commercial Discussion' 
  Sent: Thursday, July 07, 2005 6:51
  PM
  Subject: RE: [Asterisk-Users] problem
  with iax2 and 2 peers behind nat
  
  
  Do you have different
  dialplan for IAX & SIP?, that shoudn’t depend on the protocol
  used.
   
  
  
  
  
  From: [EMAIL PROTECTED]
  [mailto:[EMAIL PROTECTED] On Behalf Of SAT MADRIDSent: Thursday, July 07, 2005 12:27
  PMTo: Asterisk Users Mailing List - Non-Commercial
  DiscussionSubject: Re: [Asterisk-Users] problem
  with iax2 and 2 peers behind nat
   
  
  HI all, thanks Carlos, now its all
  working, but i have other cuestion, how y transfer call to other peer, when i
  try sip y do it pressing the # key but with iax it is not
  working.
  

- Original Message -


From: Carlos
Alperin 

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

Sent:
Wednesday, July 06, 2005 7:06 PM

Subject: RE:
[Asterisk-Users] problem with iax2 and 2 peers behind
nat

 
Juan,

 
That is not going
to work. Asterisk shouldn’t be behind a NAT to get registration of boxes
behind NAT.
 
Put the asterisk on
DMZ zone of their router to make that happen.
 
Carlos
Alperin
[EMAIL PROTECTED]
 




From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of SAT MADRIDSent: Wednesday, July 06, 2005 12:52
PMTo:
asterisk-users@lists.digium.comSubject: [Asterisk-Users] problem with
iax2 and 2 peers behind nat
 

 

 

Hi
all,

 

i have a problem with 2 peers
conecting to an asterisk machine, both are conected behind nat without any
port mapping in the router, and the * is conected behind other nat with the
port 4569 mapped to it address, the problem
is:

 

when a peer register to the
asterisk the other cant register and viceversa, only gets registration the
first one, im using firefly and a hardphone from wuchuan, itried with 2
firefly and the error its the same, it could be because the 2 peers are
going to the internet with the same ip addres(both behind nat)? if i conect
both peers in the same lan there is no problem so i think it cpuld be a
problem with nat, i dont konw if i had to change some configuration in
iax.conf.

 

Thanks.

 

Juan
Lopez.

 [EMAIL PROTECTED]

  
  

  Mensaje
  analizado y protegido, tecnologia antivirus
  www.trendmicro.es
 

  
  

  _Mensaje
  analizado y protegido, tecnologia antivirus
  www.trendmicro.es



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  _Mensaje
analizado y protegido, tecnologia antivirus
  www.trendmicro.es
  


  _Mensaje
analizado y protegido, tecnologia antivirus
  www.trendmicro.es
  
  

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RE: [Asterisk-Users] URGENT: hardware spesifications needed

[Rusty Shackleford]

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Jason Frisch
> Sent: Wednesday, July 06, 2005 4:22 PM
> To: Jimmy Smith; Asterisk Users Mailing List - Non-Commercial 
> Discussion
> Subject: Re: [Asterisk-Users] URGENT: hardware spesifications needed
> 
> 
> 
> Come on now children. Is this not a place to share knowledge?

Well..., yes, and no. Information that isn't readily available elsewhere
may legitimately be sought here. However, when the question is of the
FAQ variety, and it is clear that the person asking it has not even
attempted to find the information for himself, then rude replies are not
out of line, IMO.

-- 
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.323 / Virus Database: 267.8.10/43 - Release Date:
07/06/2005
 

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RE: [Asterisk-Users] problem with iax2 and 2 peers behind nat

[Carlos Alperin]

Do you have different dialplan for IAX &
SIP?, that shoudn’t depend on the protocol used.

 









From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of SAT MADRID
Sent: Thursday, July 07, 2005
12:27 PM
To: Asterisk
 Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users]
problem with iax2 and 2 peers behind nat



 



HI all, thanks Carlos, now its all working, but i have other
cuestion, how y transfer call to other peer, when i try sip y do it pressing
the # key but with iax it is not working.







- Original Message - 





From: Carlos
Alperin 





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





Sent: Wednesday, July
06, 2005 7:06 PM





Subject: RE:
[Asterisk-Users] problem with iax2 and 2 peers behind nat





 



Juan, 

 

That is not going to work. Asterisk
shouldn’t be behind a NAT to get registration of boxes behind NAT.

 

Put the asterisk on DMZ zone of their
router to make that happen.

 

Carlos Alperin

[EMAIL PROTECTED]

 









From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of SAT MADRID
Sent: Wednesday, July 06, 2005
12:52 PM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] problem
with iax2 and 2 peers behind nat



 



 





 





Hi all,





 





i have a problem with 2 peers conecting to an asterisk
machine, both are conected behind nat without any port mapping in the router,
and the * is conected behind other nat with the port 4569 mapped to it address,
the problem is:





 





when a peer register to the asterisk the other cant register
and viceversa, only gets registration the first one, im using firefly and a
hardphone from wuchuan, itried with 2 firefly and the error its the same, it
could be because the 2 peers are going to the internet with the same ip
addres(both behind nat)? if i conect both peers in the same lan there is no
problem so i think it cpuld be a problem with nat, i dont konw if i had to
change some configuration in iax.conf.





 





Thanks.





 





Juan Lopez.





 [EMAIL PROTECTED]




 
  
  
  Mensaje analizado y protegido, tecnologia antivirus www.trendmicro.es
  
 


 


 
  
  _
  Mensaje analizado y protegido, tecnologia antivirus www.trendmicro.es
  
 








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RE: [Asterisk-Users] IAXphone -> ip address -> extension number.

[Carlos Alperin]

What about define those phones on the SIP.conf and use sip, instead of IAX.
That protocol use be more used to communicate Asterisk servers more than
phones.

Regards,

Carlos Alperin

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Zoltan Szecsei
Sent: Thursday, July 07, 2005 12:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] IAXphone -> ip address -> extension number.

Hi,

I'm trying to set up two ACT SIP/IAX capable phones to communicate with 
each other on the same internal network, using asterisk 1.0.9 on SuSE 
9.3 (because I intend to grow the situation after this basic setup is 
functioning)

The phone IPs are set to 192.168.0.201 and 202 respectively.

I've had a look at iax.conf and extensions.conf but cannot see how to 
tie these IPs to an extension number, let alone how to dial that extension.

The "Getting Started with Asterisk" and the "Asterisk Doc Proj - Vol 1" 
that I have been using just have far too much info to work out what can 
be ignored in order to get such a simple setup working.

I'd be happy for any help or pointers to steps that I should have followed.

TIA,
Zoltan

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Re: [Asterisk-Users] res_config_mysql.so in CVS asterisk-addons broken?

[Matthew Boehm]

Sahil Gupta wrote:

Hi,
I spent quite a few days with this and in the end I find that the 1.07 
release is by far the most stable.


I had a lot of trouble with the CVS release.

Ofcourse, thats just in my case, what do the others feel on this?

Regards,


Sahil Gupta
VoiceValley


Been using CVS-HEAD in production env with 80 SIP UA's, and Digium T1 
card for several months now. No crashes. No problems. Love it. Use 
RealTime for SIP registration, Extensions and Voicemail with 
res_config_mysql. No problems here.


-Matthew

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Re: [Asterisk-Users] How to slow down dialing

[John Novack]

Randy MacKay wrote:


I would like to know if it is possible to slow down the dialing process in 
asterisk.

I have 4 of my 8 phone lines that are VoDSL.  When we try and dial out these 4 
VoDSL Lines, the number is often miss dialed, or incomplete.  I added a  wait 
before Asterisk tries to dial the whole number, but that has not solved my 
problem.  If I use a regular phone and dial out these lines, they work fine.

My assumption is that asterisk dial tones are too fast and I would like to try 
slowing them down, or spacing out each digit, to see if this helps.

I am using two TDM04B cards to connect the 4 pots and 4 VoDSL Lines.

Any help or ideas would be appreciated.

Randy


Assume you are dialing DTMF -

The DURATION of  Asterisk generated tones can be one source of  the problem.
Those smarter with the code can be more specific, but MANY telco related 
systems generate tones that are too short , 75-80 Ms  should work, but 
frequently they are as short as 50 Ms.

Interdigit time is another possibility.
Both probably can be adjusted in the source and  recompiled, but  the 
smarter code guys need to address that.
Inserting multiple "w" in the dial string will mask any slow dialtone 
issue, as Asterisk doesn't detect dialtone either.
As an aside, it has been found that the DETECTION of dial pulses ( 
remember pulse dialing? ) inbound on a TDM FXS interface is also too 
restrictive, and can be corrected in the driver source.

Anyone interested, E-mail me off list.

John Novack



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Re: [Asterisk-Users] Re: Remote SIP Connections

[dbruce]

Ok... You will need to give us more information...

What type of SIP Phones are you using?? (Make and Model)

What model of WRT54G are you using? What firmware do you have on the WRT54G?

Regards,
Derek


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

Sent: Wednesday, July 06, 2005 9:28 PM
Subject: [Asterisk-Users] Re: Remote SIP Connections


forgot to include the list

-- Forwarded message --
From: Blake Krone <[EMAIL PROTECTED]>
Date: Jul 6, 2005 9:07 PM
Subject: Re: [Asterisk-Users] Re: Remote SIP Connections
To: dbruce <[EMAIL PROTECTED]>


Just had my brother connect from his time warner cable in minnesota to
my adelphia in colorado springs, both NAT'd and I have my DMZ on,
still nothing :(

Any other ideas???
I wanted to setup an asterisk server so I could have VoIP in the house
but then send SIP phones to my parents in Minnesota to save on long
distance costs and cell minute usage.

Thanks!

On 7/5/05, Blake Krone <[EMAIL PROTECTED]> wrote:
> Well I had it setup with DMZ and port forwarding, removed the port
> forwards and still no luck :(
>
> Might end up going back to @home seen as other things like music on
> hold won't work properly, maybe something is just messed up in my
> gentoo install of asterisk.
>
> -Blake
>
> On 7/5/05, dbruce <[EMAIL PROTECTED]> wrote:
> > You have forgotten that the WRT54G is a NAT router.
> >
> > The phones that are trying to connect to your server are also very
likely to
> > be behind a NAT router. This make it almost impossible to tell what
ports
> > are actually going to be used for inbound or outbound traffic... many
NAT
> > routers do not attach any significance to SIP protocol messages. Add to
that
> > the fact that many IP phones do not use the same port range for RTP that
> > asterisk uses by default, and you have a VERY difficult time determining
> > which port ranges need to be forwarded.
> >
> > Your easiest solution is to remove the forwarding rules, give your
asterisk
> > server a static IP address on your local network, and configure that IP
> > address as the DMZ. All unsolicited requests to the router are sent to
the
> > IP address configured as the DMZ.
> >
> > The DMZ settings are found under the "Applications & Gaming" tab on the
> > WRT54G.
> >
> > You could also play with port triggering settings, but that is also a
very
> > dificult process.
> >
> > Regards,
> > Derek Bruce
> >
> >
> > - Original Message -
> > From: "Blake Krone" <[EMAIL PROTECTED]>
> > To: 
> > Sent: Tuesday, July 05, 2005 7:10 PM
> > Subject: [Asterisk-Users] Re: Remote SIP Connections
> >
> >
> > I have gotten them to be able to connect but I am unable to hear the
> > other person and they can't hear me either.
> >
> > What else am I missing?
> >
> > On 7/5/05, Blake Krone <[EMAIL PROTECTED]> wrote:
> > > Hello all, I have my * server setup behind a Linksys WRT54G on
> > > Adelphia cable. I have forwarded 5060,1-10020, and another port
> > > set can't remember off the top of my head but I can't seem to connect
> > > to the * server from any locations that are direct connects to the
> > > Internet. Am I missing a portset for forwarding?
> > >
> > > If I use the name service (voip.*.com) from my home connection on
> > > the same LAN as the * server it will connect fine.
> > >
> > > Any ideas?
> > >
> > > TIA!
> > > -blake
> > >
> > ___
> > 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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Re: [Asterisk-Users] Re: app_conference and AGI

[Jean-Hugues ROBERT]

At 15:31 07/07/2005 +, Tony Mountifield wrote:

In article <[EMAIL PROTECTED]>,
Jean-Hugues ROBERT <[EMAIL PROTECTED]> wrote:
> But... what is wrong with MEETME_AGI_BACKGROUND ? (besides the fact
> that unfortunately it does not work for SIP channels due to the mixing
> not being done in the zaptel driver but app_meetme itself, sort of, AFAIK).

It's the other way round. The mixing is always done in the zaptel driver.
For non-Zap channels, MeetMe creates a Zap pseudo channel, and in its main
loop it copies frames of audio in both directions between the non-Zap
channel and the associated pseudo channel. The Zaptel driver mixes the
audio in the pseudo channels and the real Zaptel channels.

When using MEETME_AGI_BACKGROUND, the main loop that does the pseudo
channel copying is not invoked, so only the hardware channels get mixed in
the Zaptel driver.

Cheers
Tony


Thanks for the clarification Tony.

Any idea on how to make it so that MEETME_AGI_BACKGROUND would work on
SIP channels (well, I suspect the issue is there with IAX too or any
VoIP channel for that matter...) ?

Maybe there could be a thread that would do what the main loop does.

But... there might be an issue if the two threads (the one dealing with
AGI and the "main loop" one) both try to read the frames...

If this is not possible, then maybe the copying should occur earlier, before
frame is delivered to the AGI ? This may require an additional data
member in the channel structure. Well... this is kind of beyond my
current needs/knowledge.

OTOH, isn't recording done in a distinct thread ? If so, then the same
kind of solution might be feasible.

Thanks again for the clarification.

Yours,

JeanHuguesRobert

-
Web:  http://hdl.handle.net/1030.37/1.1
Phone: +33 (0) 4 92 27 74 17

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Re: [Asterisk-Users] asterisk and wireless on site personal paging system

[Patrick]

On Thu, 2005-07-07 at 17:04 +0200, Frank Sautter wrote:
> hi patrick,
> 
> Patrick schrieb:
> > Did you try contacting the vendor of the base stations to see if they
> > have a EuroISDN firmware update? My Eicon Diva Server BRI card supports
> > the 1TR6 protocol. The firmware can be found here:
> > ftp://ftp.isdn4linux.org/pub/isdn4linux/utils/eicon/firmware/
> > Perhaps AVM supports 1TR6 too.
> 
> yes, eicon diva server supports (we also have one here) but i was not 
> able to load the capi drivers upon the 1TR6 stack?!?

Here is what I did to make it work with ETSI/TE on my box:

If you use Fedora Core or another distro that uses udev than you
probably have to add the following file in /etc/udev/rules.d *before*
modprobing the modules:

<-- start file 10-capi.rules -->
SYSFS{dev}="68:0",  NAME="capi20"
SYSFS{dev}="191:[0-9]*",NAME="capi/%n"
<-- end file 10-capi.rules  -->

Here is the order in which I load the kernel capi modules
from /etc/rc.d/rc.local:

# Start the Eicon card
/sbin/modprobe -v divas
sleep 5
/sbin/modprobe -v diva_idi
sleep 5
/sbin/modprobe -v kernelcapi
sleep 5
/sbin/modprobe capi
sleep 5
/sbin/modprobe divacapi
sleep 5
/sbin/divactrl load -c 1 -f ETSI -s 1 -vd6
sleep 5

The "sleep 5" is needed to give udev some time to generate the proper
devices. I don't know exactly which module triggers it so I put a "sleep
5" after each modprobe.

After you have manually activated the modules & divactrl above,
check /var/log/messages for any erros and the correct activation of the
card with /usr/bin/capiinfo. My output is something like:

[EMAIL PROTECTED] ~]# capiinfo
Number of Controllers : 1
Controller 1:
Manufacturer: Eicon Networks
CAPI Version: 2.0
Manufacturer Version: 2.0
Serial Number: 1884
BChannels: 2
[snip rest of output]

> the next problem would be, that we need a isdn interface in NT mode, 
> which is (to my knowledge) only possible with the cologne chip cards 
> (junghanns / beronet).

Yes those cards support NT mode. Loading the 1TR6 protocol on the Eicon
card would be done by first putting the 1TR6 firmware files (see url
previously mentioned) in /urs/share/eicon (for divactrl-2.1) and then
do:

/sbin/divactrl load -c 1 -f 1TR6

For NT mode I think you need to specify -s 2 too although the help
output from /sbin/divactrl ctrl mentions "PRI" and not "BRI".

Hope this helps.

Regards,
Patrick
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[Asterisk-Users] zaphfc: empty HDLC frame or bad CRC received

[Yousef Herzallah]

I have this problem 
zaphfc: empty HDLC frame or bad CRC received
My configurations are 
cat /proc/zaptel/1
Span 1: ZTHFC1 "HFC-S PCI A ISDN card 0 [NT] layer 1 ACTIVATED (G3)"
AMI/CCS

   1 ZTHFC1/0/1 Clear
   2 ZTHFC1/0/2 Clear
   3 ZTHFC1/0/3 HDLCFCS

cat /etc/zaptel.conf
# hfc-s pci a span definition
# most of the values should be bogus because we are not really zaptel
loadzone=it
defaultzone=it
span=1,1,3,ccs,ami
bchan=1-2
dchan=3

cat /etc/asterisk/zapata.conf
;
; Zapata telephony interface
;
; Configuration file

[channels]
language=it
switchtype=euroisdn

; p2mp TE mode
;signalling=bri_cpe_ptmp
; p2p TE mode
;signalling=bri_cpe
; p2mp NT mode
;signalling=bri_net_ptmp
; p2p NT mode
signalling=bri_net

pridialplan=dynamic
prilocaldialplan=local
nationalprefix=0
internationalprefix=00

echocancel=yes
echotraining=100
echocancelwhenbridged=yes

immediate=yes
group=1
context=default
channel => 1
channel => 2

ztcfg -vv

Zaptel Configuration
==

SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1)

Channel map:

Channel 01: Individual Clear channel (Default) (Slaves: 01)
Channel 02: Individual Clear channel (Default) (Slaves: 02)
Channel 03: D-channel (Default) (Slaves: 03)

3 channels configured.


Help 

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[Asterisk-Users] IAXphone -> ip address -> extension number.

[Zoltan Szecsei]

Hi,

I'm trying to set up two ACT SIP/IAX capable phones to communicate with 
each other on the same internal network, using asterisk 1.0.9 on SuSE 
9.3 (because I intend to grow the situation after this basic setup is 
functioning)


The phone IPs are set to 192.168.0.201 and 202 respectively.

I've had a look at iax.conf and extensions.conf but cannot see how to 
tie these IPs to an extension number, let alone how to dial that extension.


The "Getting Started with Asterisk" and the "Asterisk Doc Proj - Vol 1" 
that I have been using just have far too much info to work out what can 
be ignored in order to get such a simple setup working.


I'd be happy for any help or pointers to steps that I should have followed.

TIA,
Zoltan

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[Asterisk-Users] IAX Transfers

[Brent Davidson]

I'm having a strange problem with transfers on IAX phones.  I have two
IAX phones behind my firewall that are extensions from my office phone
system.  Both phones can receive calls, but only one of the extensions
can do blind transfers by pressing the # key.  I have a similar problem
at the office.  Some of the phones can transfer calls, some of them
can't.  And my Zap lines can always transfer.

I have all of my IAX extensions configured exactly the same way in
iax.conf.  All handsets are configured the same way and runnign the same
firmware.  I thought at first that it was a problem with NAT, but none
of the office phones are behind firewalls.

Any ideas?

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Re: [Asterisk-Users] problem with iax2 and 2 peers behind nat

[SAT MADRID]

HI all, thanks Carlos, now its all working, but i
have other cuestion, how y transfer call to other peer, when i try sip y do it
pressing the # key but with iax it is not working.

  - Original Message - 
  From:
  Carlos
  Alperin 
  To: 'Asterisk Users Mailing List -
  Non-Commercial Discussion' 
  Sent: Wednesday, July 06, 2005 7:06
  PM
  Subject: RE: [Asterisk-Users] problem
  with iax2 and 2 peers behind nat
  
  
  Juan,
  
   
  That is not going to
  work. Asterisk shouldn’t be behind a NAT to get registration of boxes behind
  NAT.
   
  Put the asterisk on
  DMZ zone of their router to make that happen.
   
  Carlos
  Alperin
  [EMAIL PROTECTED]
   
  
  
  
  
  From: [EMAIL PROTECTED]
  [mailto:[EMAIL PROTECTED] On Behalf Of SAT MADRIDSent: Wednesday, July 06, 2005 12:52
  PMTo:
  asterisk-users@lists.digium.comSubject: [Asterisk-Users] problem with
  iax2 and 2 peers behind nat
   
  
   
  
   
  
  Hi
  all,
  
   
  
  i have a problem with 2 peers
  conecting to an asterisk machine, both are conected behind nat without any
  port mapping in the router, and the * is conected behind other nat with the
  port 4569 mapped to it address, the problem
  is:
  
   
  
  when a peer register to the
  asterisk the other cant register and viceversa, only gets registration the
  first one, im using firefly and a hardphone from wuchuan, itried with 2
  firefly and the error its the same, it could be because the 2 peers are going
  to the internet with the same ip addres(both behind nat)? if i conect both
  peers in the same lan there is no problem so i think it cpuld be a problem
  with nat, i dont konw if i had to change some configuration in
  iax.conf.
  
   
  
  Thanks.
  
   
  
  Juan
  Lopez.
  
   [EMAIL PROTECTED]
  


  Mensaje
analizado y protegido, tecnologia antivirus
  www.trendmicro.es
  


  _Mensaje
analizado y protegido, tecnologia antivirus
  www.trendmicro.es
  
  

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Re: [Asterisk-Users] res_config_mysql.so in CVS asterisk-addons broken?

[Sahil Gupta]

Hi,
I spent quite a few days with this and in the end I find that the 1.07 
release is by far the most stable.


I had a lot of trouble with the CVS release.

Ofcourse, thats just in my case, what do the others feel on this?

Regards,


Sahil Gupta
VoiceValley

On Thu, 7 Jul 2005, Christoph wrote:


Hi!

I would like to use the realtime extension of Asterisk and got the
latest asterisk-addons from CVS. Upon compiling things, I got a couple
of error messages from app_addon_mysql... is it me, or are the files in
the CVS broken?

Thanks,
Christoph

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Re: [Asterisk-Users] res_config_mysql.so in CVS asterisk-addons broken?

[Matthew Boehm]

Christoph wrote:

Hi!

I would like to use the realtime extension of Asterisk and got the
latest asterisk-addons from CVS. Upon compiling things, I got a couple
of error messages from app_addon_mysql... is it me, or are the files in
the CVS broken?

Thanks,
Christoph


Please explain why your email subject referes to res_config_mysql but 
your email says absolutly nothing about it?


The files in CVS are not broken. I'm using them right now in a prod 
environment.


What errors are you getting?

-Matthew

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Re: [Asterisk-Users] Incoming 800-number over IAX - first few words are cut-off

[Brian West]

Ok can you tell me if you get any errors on a "short" free call? :P   
You forgot to tell us what version of asterisk on both ends... wen  
can only guess at this point what the problem might be.


/b
---
Anakin: “You’re either with me, or you’re my enemy.”
Obi-Wan: “Only a Sith could be an absolutist.”

On Jul 6, 2005, at 2:04 PM, Joseph wrote:

I have an incoming 800-number over IAX from Teliax and I'm  
experiencing

the large packet loss on connection.
When a call comes in there is no ring tone and the first few words of
the welcome message are cut off, regardless of the delay I set.
Standard call (not 800-number) coming over IAX with the same provider
works just fine only the tall free number.

So it seems there are some packet loss only at the beginning, as the
call quality sounds just fine, even when I compile something and  
CPU is

at 99% use, there is no packet drop during conversation only on
connection of tall free number.

--
#Joseph
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[Asterisk-Users] Re: app_conference and AGI

[Tony Mountifield]

In article <[EMAIL PROTECTED]>,
Jean-Hugues ROBERT <[EMAIL PROTECTED]> wrote:
> But... what is wrong with MEETME_AGI_BACKGROUND ? (besides the fact
> that unfortunately it does not work for SIP channels due to the mixing
> not being done in the zaptel driver but app_meetme itself, sort of, AFAIK).

It's the other way round. The mixing is always done in the zaptel driver.
For non-Zap channels, MeetMe creates a Zap pseudo channel, and in its main
loop it copies frames of audio in both directions between the non-Zap
channel and the associated pseudo channel. The Zaptel driver mixes the
audio in the pseudo channels and the real Zaptel channels.

When using MEETME_AGI_BACKGROUND, the main loop that does the pseudo
channel copying is not invoked, so only the hardware channels get mixed in
the Zaptel driver.

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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[Asterisk-Users] FXO hangup Problem.....

[Nahid Hossain]

Hello,

 

I am getting problem for delay call hang-up with the below
scenario:

 

PSTN User (calling Party)---àPSTN Line
à FXO with Asterisk Box-àSIP IP Phone
(called party)

 

 

I am using X100P card with my Asterisk-1.0.7 box. I am also
using Zaptel-1.0.7 version.

 

When PSTN user makes call to my PSTN line and after getting
IVR, PSTN user dial my SIP IP Phone extension, as soon as PSTN user gets one
ring back tone, PSTN user cut off the current call. But SIP IP Phone rings till
its timeout. 

 

I would appreciate if anyone give me solution for the above
case.

 

Regards

Nahid

 






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Re: [Asterisk-Users] Snom phones - any advice

[Randy Williams]

Oops.

I forgot to add the recommendation to use the most current stable 
release of all firmware/boot loader/OS for the SNOM's as that can make a 
significant difference.  Also, it may be better to see if you can 
purchase them from a vendor that will also support you, if possible.


Sorry about that.

RandyW

Randy Williams wrote:


Greetings,

We are just finishing a roll-out of 25 of the SNOM 190s with a SNOM 
220 w/sidecar.


The only "gotcha" that I found is that the SNOM 190s use "rfc2833" for 
a default dtfm mode and not "inband" which is the default for the 
asterisk server.


I haven't ironed out the Mass deployment functionality yet, but will 
do so.  So with a tftp server running you should be fine.


Generally speaking, of course.

RandyW

Patrick Fortin wrote:


Hi

We are about to buy several Snom phones.

Does anyone have warnings or advices against these phones ?

Our finalists were Cisco, Polycom and Snom.

We will be using only the SIP protocol.

Thanks

Patrick


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[Asterisk-Users] res_config_mysql.so in CVS asterisk-addons broken?

[Christoph]

Hi!

I would like to use the realtime extension of Asterisk and got the
latest asterisk-addons from CVS. Upon compiling things, I got a couple
of error messages from app_addon_mysql... is it me, or are the files in
the CVS broken?

Thanks,
Christoph

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Re: [Asterisk-Users] ISDN PRI No Audio

[Matt Fredrickson]

On Wed, Jul 06, 2005 at 05:24:06PM -0500, Andy Brezinsky wrote:
> [Span 3 D-Channel 0]< Channel ID (len= 6) [ Ext: 1  IntID: Explicit, PRI 
> Spare: 0, Exclusive Dchan: 0
> [Span 3 D-Channel 0] [Span 3 D-Channel 0]<   Ext: 1  DS1 Identifier: 2
> [Span 3 D-Channel 0]<   Ext: 1  Coding: 0   Number 
> Specified   Channel Type: 3
> [Span 3 D-Channel 0]<   Ext: 1  Channel: 24 ]
> < [1e 02 81 83]

Make sure that your span map is correctly done.  It looks like the destination
b channel is channel 24 on span 2.  Make sure that you have your DS1s plugged in
in the correct order and it's using the right DS1 for this.  The channel that 
chan_zap
picked for that was 48, so make sure also that they are not numbering the DS1 
identifier
beginning with 0.  You might want to see if you need to adjust your spanmap and 
related
config in zapata.conf for all of this.

-- 
Matthew Fredrickson
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Re: [Asterisk-Users] asterisk and wireless on site personal paging system

[Frank Sautter]

hi patrick,

Patrick schrieb:

Did you try contacting the vendor of the base stations to see if they
have a EuroISDN firmware update? My Eicon Diva Server BRI card supports
the 1TR6 protocol. The firmware can be found here:
ftp://ftp.isdn4linux.org/pub/isdn4linux/utils/eicon/firmware/
Perhaps AVM supports 1TR6 too.


yes, eicon diva server supports (we also have one here) but i was not 
able to load the capi drivers upon the 1TR6 stack?!?
the next problem would be, that we need a isdn interface in NT mode, 
which is (to my knowledge) only possible with the cologne chip cards 
(junghanns / beronet).

so i think we need an new solutions with the old wireless pagers.

is there anybody who has experience with http://www.ascom.com/ws ?
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Re: [Asterisk-Users] app_rxfax does not receive

[David Romero]

are you sharing IRQ on yuor zap device?what version of libtiff you have?be sure you not are sharing IRQ whit your zap and other devices andbe sure you have the more recent version of libtiff.
On 7/6/05, Bohuslav Coufal <[EMAIL PROTECTED]> wrote:













Hi all,

 

I try to use app_rxfax. Aplication app_rxfax start
O.K., fax trying to send, but it will stop at the beginning of page and after
few seconds it stop with error 400.

 

Does anybody has any suggestions?

 

Thanks,

 

Bob.







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[Asterisk-Users] Logging SIP response codes

[Pedro]

Is there a way to log SIP response codes without enabling verbose
logging?  Reason being is that from time to time I see a call fail on
our primary provider and roll-over to our backup providers.  If I
happen to catch it on the console I can see the code "484" or similar.
 It would really help in troubleshooting with our primary provider if
I could log those types of codes.  Verbose just saves way to much
stuff in the log files.
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