Re: [Asterisk-Users] how do i get rid of this blasted echo !!!

[Herman Cremer]

On the echo...

I have 2 extensions, with different analog phones.
The one works fine, the other echos and scratches 
like mad !!

I have switched the ports, cables etc but its ALWAYS
the same phone...

Maybe this could be it ?

Is it ok from a SIP phone ?

Herman cremer





___
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

[Asterisk-Users] TE110P experiance

[Mario . Spoljar]

Hello to all,
I would like to ask some Digium TE110P users if they can share experiance
about this card. I put in service card yesterday but I noticed following
(strange) behaviar:
- if I have to reboot my computer my zaptel driver fail to start and
produce this error:
  ZT_SPANCONFIG failed on span 1: No such device or address (6)
- to solve this problem I have to power cycle my computer and in all cases
this brings up card!

- does anybody have any info about this hardware, example there are two LED
- what is the meaning of these LEDs. I bought this card and got anly card
without any papers (just bill :-( )
Regards,

[EMAIL PROTECTED]

___
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

RE: [Asterisk-Users] Comparison Charts

[Jim Van Meggelen]

[EMAIL PROTECTED] wrote:
 I couldnt agree with you more Jim. Im realdy using Asterisk
 and agree 100% with what you say... I was asking for a
 comparison list with other PBX's because for example, for a
 customer, they have heard of Avaya and Cisco and they all are
 selling IP now... So In order to get your customer to
 trust Asterisk over those guys, you need to show him the
 diff. Between the two and some lists of the features on the
 others compared to Asterisk..

It kind of reminds me of the challenge in selling the Internet to
management in the early 90s. The trick was getting them to think in a
whole different way. In many cases, they bought into it simply because
they had no choice. Businesses didn't fully get it until everyone was
using it.

You could compare Asterisk to other products, but that wouldn't show it
in its best light. It might be better to explore some of the things that
Asterisk can do that the other systems cannot.

The VoIP part is a total red-herring - we've had VoIP for over 10 years;
the real power is in the flexibility. Defining exactly what that
flexibility is will in large part depend on your audience. Find out what
excites them. Is it cost? Asterisk has a compelling story to tell. Is it
standards-compliance? Asterisk again scores points. Flexibility? Yep.
Open-source (or avoiding vendor lock-in)? You betcha!

I would almost want to see a list of features that those other products
had that could NOT be configured on Asterisk. 

Who really knows what the limits are? Ten years later we're still
finding out new uses for the Internet. I imagine that ten years from now
we'll still be adding features to open-source telephony . . . will we
even call it telephony then? I'm betting no.




 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of
 Jim Van Meggelen
 Sent: Jueves, 10 de Marzo de 2005 12:17 a.m.
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: RE: [Asterisk-Users] Comparison Charts
 
 [EMAIL PROTECTED] wrote:
 Guys.
 
 Anybody has a URL or some document with comparison charts with
 Asterisk's features against other PBXs?
 
 I would argue that what you ask is in some ways impossible.
 Asterisk is orders of magnitude more flexible than any other
 PBX you may have encountered, because it is more like a
 toolkit than a PBX. Whatever is missing can be built, so
 there's no list of features that can ever be considered complete.
 
 For people who are looking for a PBX that has a user-friendly
 interface and is easy to configure, Asterisk will tend to
 dissappoint. Where Asterisk shines is for those people who
 want to--need to--build their own PBX. People who are willing
 to do the work themselves; designing, testing, debugging,
 re-designing . . . 
 
 Many of us believe that Asterisk is going to transform the
 telecommunication industry, but it won't do it because it has
 more features, it'll do it because it puts the control of
 the features list where it belongs: in the customer's hands.
 
 I would suggest that the best way to approach Asterisk is to
 have a list of things that you need your telephone system to
 do. Then, one-by-one, figure out how to handle each of those
 in Asterisk. When you are done, you may have a few that you
 couldn't find a satisfactory solution to. Those can typically
 be custom developed, and surprisingly, you will still
 probably come in at a lower cost than a closed, so-called
 full-featured proprietary system.
 
 What's more, as your needs grow, Asterisk can grow with you.
 Five years from now you won't need to hear oh sorry but that
 system is no longer supported. Want new functionality?
 Install it. Is the hard drive wearing out? Replace it. Need
 more CPU power? Migrate to a new chassis.
 
 Asterisk changes all the rules. Therfore, to understand it,
 you have to adopt a new way of thinking about telecom systems.
 
 Welcome to Asterisk!

-- 
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.308 / Virus Database: 266.7.1 - Release Date: 09/03/2005
 

___
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

Re: [Asterisk-Users] Bandwidth

[Jean-Michel Hiver]

asterisk wrote:
Assuming I'm using a VOIP provider of some sort, what kind of 
bandwidth requirements / line should I expect to have in place?  I 
currently have 8 traditional voice lines, and a FAX line that doubles 
as my DSL source.  Ballpark, what do I need to have in place to move 
everything to asterisk?
- I recommend having a dedicated Linux box (I use debian + a couple of 
ethernet cards) which does Network Bridge + Asterisk + Traffic shaping.

- If your bandwith is short (for example 256kbit/s), install another 
asterisk box on a dedicated hosting facility with plenty of bandwith. 
Then buy some g.729 licenses so that you can use g.729 between your 
office behind DSL and your dedicated box.

- Keep your FAX line for faxes, emergencies, and failover.
Cheers,
Jean-Michel.
___
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

RE: [Asterisk-Users] Panasonic TDA200 E1 - E100P negotiation issues

[James Bean]

Is the E1 card an isdn card or something else? There are a several
signalling systems that can run over an E1.

When running cas you do not have a D channel for the signalling.
Instead each voice channel has a few dedicated 
bits in channel 16 (hence Channel Associated Signalling). This is used
for EM, loopstart etc and is incompatible 
with the ISDN signalling that you tried.

You need to tell us more about what card you have in the Panasonic PBX.

Ok not exactly sure what info to give you, I ordered an E1 card from
panasonic for the phone system and its what they sent me, it has an RJ45
interface and coax TX/RX connectors as well, I also have full access
with the techs version of the panasonic control/programming software
and know my way around if there is something specific I could get out of
the settings on the card for you to allow you to know which card it is.
I would assume (I know that's bad) that it is an ISDN card as it
should be the card that is used to connect to a telco directly.

I know that it is using hdb3, it only shows up with 30 channels on the
card in the E1 slot setup.

What happens to channel 16 which is usually set as the d-channel, or
should I be including channel 16 in with the rest and not using port 31
in the channels?

James
___
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

RE: [Asterisk-Users] No dialtone in a E1

[Peter Svensson]

On Fri, 11 Feb 2005, Marco Castillo wrote:

 Thank you Peter, how can I add the options to Dial to generate ringback???
 do you have an example???

run show application dial in the cli. It should explain the options, 
including the r option.

 By the way, it is a PRI E1, with 30 bchannels and 1 dchannel. For a little
 background, I'm intending to replace my actual PBX with Asterisk, and
 everything is just working fine, until yesterday when I realized that when a
 call was made from some external lines, this lines didn't receive a
 dialtone. For this reason, I began to make some exhaustive test cases, and
 began to make calls from distinct providers to my E1. In all this testing I
 received a dialtone, except for a GSM cellular phone from a specific Telco.
 I tested some others GSM cellulars from the same Telco, and got always the
 same functionality, they didn't receive a dialtone. I think that if Asterisk
 can generate a ringback, this is going to solve all my problems with this
 little issue.

The pstn has the option of either generating progress tones locally or 
allowing the remote end to send them. Since asterisk always claims In 
band progress tines available, it has to generate them if the destination 
does not. If the destination is a pbx it will usually fill the back 
channel with progress tones (i.e. ringback) which will pass streight 
through asterisk. For destinations that does not (such as most voip 
phones) asterisk has to fill in the back channel with progress tones. 

Using or not using the audio in this reverse channel is up to the 
originating telco.

Peter

___
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

RE: [Asterisk-Users] Panasonic TDA200 E1 - E100P negotiation issues

[Peter Svensson]

On Fri, 11 Mar 2005, James Bean wrote:

 You need to tell us more about what card you have in the Panasonic PBX.
 
 Ok not exactly sure what info to give you, I ordered an E1 card from
 panasonic for the phone system and its what they sent me, it has an RJ45
 interface and coax TX/RX connectors as well, I also have full access
 with the techs version of the panasonic control/programming software
 and know my way around if there is something specific I could get out of
 the settings on the card for you to allow you to know which card it is.
 I would assume (I know that's bad) that it is an ISDN card as it
 should be the card that is used to connect to a telco directly.

Well, you can connect to the telco using non-isdn signalling as well. In 
Europe isdn is by far the most common signalling form used on an E1. Can 
you find the model number for the E1 card?

 I know that it is using hdb3, it only shows up with 30 channels on the
 card in the E1 slot setup.

An E1 always has 30 voice channels, one signalling channel (running CAS or 
CCS) and one timing channel. (Well, you _can_ run voice over channel 16, 
but then you would not have any signalling as RBS is not normally used on 
an E1).

 What happens to channel 16 which is usually set as the d-channel, or
 should I be including channel 16 in with the rest and not using port 31
 in the channels?

Channles 16 on an E1 is always reserved for signalling. There are several 
signalling mechanisms which can be transported in that slot. Isdn uses 
CCS, but there are other non-isdn signalling systems that instead use a 
few bits per channel each frame, CAS.

Can you send me the result of a pri intense debug span X from asterisk? 
Have asterisk set to be the clock source (the timing set to 0 in the span 
line) and configured as pri_net.

Peter



___
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

[Asterisk-Users] from sip to asterisk to h323..how

[Altus Snyman]

Goo day all
This is our setup


Client phone--(SIP)--asterisk server---SIP/IAX---asterisk---
-- goes out to international server running sip/iax
But now I want to dial out to H323 server?
I.O.W I want asterisk to act as a H323 client that will rout some calls
out to a H323 server.How do I do this an can asterisk eve do this
I had a quick look on the net and only saw that asterisk can be a h323
server not client.
Please Help
Thanks
Altus

___
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

Re: [Asterisk-Users] how do i get rid of this blasted echo !!!

[David Wilson]

Is MMX support not enabled by default in the Zap drivers ?
So this is something we need to do if using any PII, PIII, P4 
AMDK6/Duron/Athlon and Celeron CPU ?

Kindest regards
David Wilson
___
D c D a t a
Tel +27 33 342 7003
Fax +27 33 345 4155
Cell +27 82 4147413
http://www.dcdata.co.za
[EMAIL PROTECTED]
Powered by Linux, driven by passion !
___
Computers are not intelligent. They only think they are.
- Original Message - 
From: Shidan [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Sent: Friday, March 11, 2005 8:24 AM
Subject: Re: [Asterisk-Users] how do i get rid of this blasted echo !!!


Yes they do, they've had it since the K6, and compiling with
mmx-support enabled will make a real difference with them as well.

Do AMD processors have MMX as well? I seem to recall that they did, but
can't remember for sure?
Thanks,
Adam
--
 --
Adam Goryachev
Website Managers
Ph:  +61 2 8304 [EMAIL PROTECTED]
Fax: +61 2 9345 4396www.websitemanagers.com.au
___
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
___
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 
___
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

RE: [Asterisk-Users] Panasonic TDA200 E1 - E100P negotiation issues

[James Bean]

Well, you can connect to the telco using non-isdn signalling as well.
In Europe isdn is by far 
the most common signalling form used on an E1. Can you find the model
number for the E1 card?

An E1 always has 30 voice channels, one signalling channel (running CAS
or
CCS) and one timing channel. (Well, you _can_ run voice over channel
16, but then you 
would not have any signalling as RBS is not normally used on an E1).

Channles 16 on an E1 is always reserved for signalling. There are
several signalling mechanisms 
which can be transported in that slot. Isdn uses CCS, but there are
other non-isdn signalling 
systems that instead use a few bits per channel each frame, CAS.

Can you send me the result of a pri intense debug span X from
asterisk? 
Have asterisk set to be the clock source (the timing set to 0 in the
span
line) and configured as pri_net.

Attached is the pri dump from asterisk just bringing the E1 into service
with the settings you suggested.

James
*CLI pri intense debug span 1
Enabled EXTENSIVE debugging on span 1
*CLI Sending Set Asynchronous Balanced Mode Extended

 [ 00 01 7f ]

 Unnumbered frame:
 SAPI: 00  C/R: 0 EA: 0
  TEI: 000EA: 1
   M3: 3   P/F: 1 M2: 3 11: 3  [ SABME (set asynchronous balanced mode 
 extended) ]
 0 bytes of data
Sending Set Asynchronous Balanced Mode Extended

 [ 00 01 7f ]

 Unnumbered frame:
 SAPI: 00  C/R: 0 EA: 0
  TEI: 000EA: 1
   M3: 3   P/F: 1 M2: 3 11: 3  [ SABME (set asynchronous balanced mode 
 extended) ]
 0 bytes of data
Sending Set Asynchronous Balanced Mode Extended

 [ 00 01 7f ]

 Unnumbered frame:
 SAPI: 00  C/R: 0 EA: 0
  TEI: 000EA: 1
   M3: 3   P/F: 1 M2: 3 11: 3  [ SABME (set asynchronous balanced mode 
 extended) ]
 0 bytes of data
Sending Set Asynchronous Balanced Mode Extended

 [ 00 01 7f ]

 Unnumbered frame:
 SAPI: 00  C/R: 0 EA: 0
  TEI: 000EA: 1
   M3: 3   P/F: 1 M2: 3 11: 3  [ SABME (set asynchronous balanced mode 
 extended) ]
 0 bytes of data
Sending Set Asynchronous Balanced Mode Extended

 [ 00 01 7f ]

 Unnumbered frame:
 SAPI: 00  C/R: 0 EA: 0
  TEI: 000EA: 1
   M3: 3   P/F: 1 M2: 3 11: 3  [ SABME (set asynchronous balanced mode 
 extended) ]
 0 bytes of data
Mar 11 19:50:19 WARNING[7051]: chan_zap.c:5653 handle_init_event: Detected 
alarm on channel 1: Red Alarm
Mar 11 19:50:19 WARNING[7051]: chan_zap.c:1254 zt_disable_ec: Unable to disable 
echo cancellation on channel 1
Mar 11 19:50:19 WARNING[7051]: chan_zap.c:5653 handle_init_event: Detected 
alarm on channel 2: Red Alarm
Mar 11 19:50:19 WARNING[7051]: chan_zap.c:1254 zt_disable_ec: Unable to disable 
echo cancellation on channel 2
Mar 11 19:50:19 WARNING[7051]: chan_zap.c:5653 handle_init_event: Detected 
alarm on channel 3: Red Alarm
Mar 11 19:50:19 WARNING[7051]: chan_zap.c:1254 zt_disable_ec: Unable to disable 
echo cancellation on channel 3
Mar 11 19:50:19 WARNING[7051]: chan_zap.c:5653 handle_init_event: Detected 
alarm on channel 4: Red Alarm
Mar 11 19:50:19 WARNING[7051]: chan_zap.c:1254 zt_disable_ec: Unable to disable 
echo cancellation on channel 4
Mar 11 19:50:19 WARNING[7051]: chan_zap.c:5653 handle_init_event: Detected 
alarm on channel 5: Red Alarm
Mar 11 19:50:19 WARNING[7051]: chan_zap.c:1254 zt_disable_ec: Unable to disable 
echo cancellation on channel 5
Mar 11 19:50:19 WARNING[7051]: chan_zap.c:5653 handle_init_event: Detected 
alarm on channel 6: Red Alarm
Mar 11 19:50:19 WARNING[7051]: chan_zap.c:1254 zt_disable_ec: Unable to disable 
echo cancellation on channel 6
Mar 11 19:50:19 WARNING[7051]: chan_zap.c:5653 handle_init_event: Detected 
alarm on channel 7: Red Alarm
Mar 11 19:50:19 WARNING[7051]: chan_zap.c:1254 zt_disable_ec: Unable to disable 
echo cancellation on channel 7
Mar 11 19:50:19 WARNING[7051]: chan_zap.c:5653 handle_init_event: Detected 
alarm on channel 8: Red Alarm
Mar 11 19:50:19 WARNING[7051]: chan_zap.c:1254 zt_disable_ec: Unable to disable 
echo cancellation on channel 8
Mar 11 19:50:19 WARNING[7051]: chan_zap.c:5653 handle_init_event: Detected 
alarm on channel 9: Red Alarm
Mar 11 19:50:19 WARNING[7051]: chan_zap.c:1254 zt_disable_ec: Unable to disable 
echo cancellation on channel 9
Mar 11 19:50:19 WARNING[7051]: chan_zap.c:5653 handle_init_event: Detected 
alarm on channel 10: Red Alarm
Mar 11 19:50:19 WARNING[7051]: chan_zap.c:1254 zt_disable_ec: Unable to disable 
echo cancellation on channel 10
Mar 11 19:50:19 WARNING[7051]: chan_zap.c:5653 handle_init_event: Detected 
alarm on channel 11: Red Alarm
Mar 11 19:50:19 WARNING[7051]: chan_zap.c:1254 zt_disable_ec: Unable to disable 
echo cancellation on channel 11
Mar 11 19:50:19 NOTICE[7051]: chan_zap.c:7395 pri_dchannel: PRI got event: 
Alarm (4) on Primary D-channel of span 1
Mar 11 19:50:19 WARNING[7051]: chan_zap.c:1931 pri_find_dchan: No D-channels 
available!  Using Primary on channel anyway 16!
Mar 11 19:50:19 WARNING[7051]: chan_zap.c:5653 handle_init_event: Detected 
alarm on channel 12: Red Alarm
Mar 11 19:50:19 WARNING[7051]: 

RE: [Asterisk-Users] Panasonic TDA200 E1 - E100P negotiation issues

[James Bean]

Whooppss had pri_cpe set, redid the debug as attached.

They seem the same but just in case.

James
Enabled EXTENSIVE debugging on span 1
*CLI Sending Set Asynchronous Balanced Mode Extended

 [ 02 01 7f ]

 Unnumbered frame:
 SAPI: 00  C/R: 1 EA: 0
  TEI: 000EA: 1
   M3: 3   P/F: 1 M2: 3 11: 3  [ SABME (set asynchronous balanced mode 
 extended) ]
 0 bytes of data
Sending Set Asynchronous Balanced Mode Extended

 [ 02 01 7f ]

 Unnumbered frame:
 SAPI: 00  C/R: 1 EA: 0
  TEI: 000EA: 1
   M3: 3   P/F: 1 M2: 3 11: 3  [ SABME (set asynchronous balanced mode 
 extended) ]
 0 bytes of data
Sending Set Asynchronous Balanced Mode Extended

 [ 02 01 7f ]

 Unnumbered frame:
 SAPI: 00  C/R: 1 EA: 0
  TEI: 000EA: 1
   M3: 3   P/F: 1 M2: 3 11: 3  [ SABME (set asynchronous balanced mode 
 extended) ]
 0 bytes of data
Sending Set Asynchronous Balanced Mode Extended

 [ 02 01 7f ]

 Unnumbered frame:
 SAPI: 00  C/R: 1 EA: 0
  TEI: 000EA: 1
   M3: 3   P/F: 1 M2: 3 11: 3  [ SABME (set asynchronous balanced mode 
 extended) ]
 0 bytes of data
Sending Set Asynchronous Balanced Mode Extended

 [ 02 01 7f ]

 Unnumbered frame:
 SAPI: 00  C/R: 1 EA: 0
  TEI: 000EA: 1
   M3: 3   P/F: 1 M2: 3 11: 3  [ SABME (set asynchronous balanced mode 
 extended) ]
 0 bytes of data
Mar 11 19:58:13 WARNING[7151]: chan_zap.c:5653 handle_init_event: Detected 
alarm on channel 1: Red Alarm
Mar 11 19:58:13 WARNING[7151]: chan_zap.c:1254 zt_disable_ec: Unable to disable 
echo cancellation on channel 1
Mar 11 19:58:13 WARNING[7151]: chan_zap.c:5653 handle_init_event: Detected 
alarm on channel 2: Red Alarm
Mar 11 19:58:13 WARNING[7151]: chan_zap.c:1254 zt_disable_ec: Unable to disable 
echo cancellation on channel 2
Mar 11 19:58:13 WARNING[7151]: chan_zap.c:5653 handle_init_event: Detected 
alarm on channel 3: Red Alarm
Mar 11 19:58:13 WARNING[7151]: chan_zap.c:1254 zt_disable_ec: Unable to disable 
echo cancellation on channel 3
Mar 11 19:58:13 WARNING[7151]: chan_zap.c:5653 handle_init_event: Detected 
alarm on channel 4: Red Alarm
Mar 11 19:58:13 WARNING[7151]: chan_zap.c:1254 zt_disable_ec: Unable to disable 
echo cancellation on channel 4
Mar 11 19:58:13 WARNING[7151]: chan_zap.c:5653 handle_init_event: Detected 
alarm on channel 5: Red Alarm
Mar 11 19:58:13 WARNING[7151]: chan_zap.c:1254 zt_disable_ec: Unable to disable 
echo cancellation on channel 5
Mar 11 19:58:13 WARNING[7151]: chan_zap.c:5653 handle_init_event: Detected 
alarm on channel 6: Red Alarm
Mar 11 19:58:13 WARNING[7151]: chan_zap.c:1254 zt_disable_ec: Unable to disable 
echo cancellation on channel 6
Mar 11 19:58:13 WARNING[7151]: chan_zap.c:5653 handle_init_event: Detected 
alarm on channel 7: Red Alarm
Mar 11 19:58:13 WARNING[7151]: chan_zap.c:1254 zt_disable_ec: Unable to disable 
echo cancellation on channel 7
Mar 11 19:58:13 WARNING[7151]: chan_zap.c:5653 handle_init_event: Detected 
alarm on channel 8: Red Alarm
Mar 11 19:58:13 WARNING[7151]: chan_zap.c:1254 zt_disable_ec: Unable to disable 
echo cancellation on channel 8
Mar 11 19:58:13 WARNING[7151]: chan_zap.c:5653 handle_init_event: Detected 
alarm on channel 9: Red Alarm
Mar 11 19:58:13 WARNING[7151]: chan_zap.c:1254 zt_disable_ec: Unable to disable 
echo cancellation on channel 9
Mar 11 19:58:13 WARNING[7151]: chan_zap.c:5653 handle_init_event: Detected 
alarm on channel 10: Red Alarm
Mar 11 19:58:13 WARNING[7151]: chan_zap.c:1254 zt_disable_ec: Unable to disable 
echo cancellation on channel 10
Mar 11 19:58:13 WARNING[7151]: chan_zap.c:5653 handle_init_event: Detected 
alarm on channel 11: Red Alarm
Mar 11 19:58:13 WARNING[7151]: chan_zap.c:1254 zt_disable_ec: Unable to disable 
echo cancellation on channel 11
Mar 11 19:58:13 NOTICE[7151]: chan_zap.c:7395 pri_dchannel: PRI got event: 
Alarm (4) on Primary D-channel of span 1
Mar 11 19:58:13 WARNING[7151]: chan_zap.c:1931 pri_find_dchan: No D-channels 
available!  Using Primary on channel anyway 16!
Mar 11 19:58:13 WARNING[7151]: chan_zap.c:5653 handle_init_event: Detected 
alarm on channel 12: Red Alarm
Mar 11 19:58:13 WARNING[7151]: chan_zap.c:1254 zt_disable_ec: Unable to disable 
echo cancellation on channel 12
Mar 11 19:58:13 WARNING[7151]: chan_zap.c:5653 handle_init_event: Detected 
alarm on channel 13: Red Alarm
Mar 11 19:58:13 WARNING[7151]: chan_zap.c:1254 zt_disable_ec: Unable to disable 
echo cancellation on channel 13
Mar 11 19:58:13 WARNING[7151]: chan_zap.c:5653 handle_init_event: Detected 
alarm on channel 14: Red Alarm
Mar 11 19:58:13 WARNING[7151]: chan_zap.c:1254 zt_disable_ec: Unable to disable 
echo cancellation on channel 14
Mar 11 19:58:13 WARNING[7151]: chan_zap.c:5653 handle_init_event: Detected 
alarm on channel 15: Red Alarm
Mar 11 19:58:13 WARNING[7151]: chan_zap.c:1254 zt_disable_ec: Unable to disable 
echo cancellation on channel 15
Mar 11 19:58:13 WARNING[7151]: chan_zap.c:5653 handle_init_event: Detected 
alarm on channel 17: Red Alarm
Mar 11 19:58:13 WARNING[7151]: chan_zap.c:1254 

RE: [Asterisk-Users] TE110P experiance

[Rob Scott]

I have noticed the following:

   - the PCI ID of the card seems to change over time which means that
loading the module does not always recognise the card, only way to reset
this is to power cycle the machine

   - you cannot unload the module once it is loaded, it hangs the
machine, which also means if you have automatic shutdown scripts for
restarting the machine then the machine will hang on reboot

One of the LEDs shows the status of the connection.
If it is off, then it is not active, i.e. zaptel drivers not loaded.
If red then bad connection i.e. it is not talked to the other end,
usually a wiring problem
If green then everything OK.
Could also be a yellow state but I haven't seen that.

Once you get it working, leave the thing up is my only advice.
It is a shame that it is not bug free, neither the hardware nor the
software so far.
I don't know what Digium want to do about the hardware.
I hope there is a firmware fix rather than having to mess with the
actual physical hardware.

___
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

[Asterisk-Users] NuFone Configuration [problem]

[Edward Banfa]

Hello,
I am trying to configure the my asterisk box here with the following

**iax.conf***
[NuFone]
type=peer
host=switch-1.nufone.net
secret=xx

***extensions.conf:***

exten = _1NXXNXX,1,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN}
exten = _011N.,1,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN}

I have a couple of Xlite softphones and 2 analogue phones connected to a
mediatrix 1102 connected to our lan. The mediatrix talks sip to the asterisk
box on the lan. We are running asterisk  on FC3 .

SOFTPHONES[XLITE] ---SIP-- ASTERISKIAX---NUFONE[ASTERISK]

ANALOGPHONES---MEDIATRIX_1102---SIP---ASTERISK---IAX---NUFONE[ASTERISK]

Well the problem goes something like this.
1) I can dial a number form the softphones and when the call is answered I
can hear the user on the other end but the user can't hear me
2) I can dial a number from the analog phones (via mediatrix tru to
asterisk)(the mediatrix is properly registered with our asterisk box) and
when the call is answered both ends can't hear a word, its just silent.

I think I am having a codec problem here. What am I doing wrong. We would
sincerely appreciate any help/pointers.

Thank you all
Edward Banfa

**EXTENSION.CONF***
[general]
static=yes

[from-sip]
exten = 100,1,Dial(SIP/edward,20)
exten = 100,2,Hangup

exten = 101,1,Dial(SIP/phone1,20)
exten = 101,2,Hangup

exten = 102,1,Dial(SIP/phone2,20)
exten = 102,2,Hangup

exten = _1NXXNXX,1,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN}
exten = _011N.,1,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN}


*IAX.CONF*
[general]
port=5036
bind=0.0.0.0
bandwidth=low
disallow=lpc10

[NuFone]
type=peer
host=switch-1.nufone.net
secret=xx
disallow=all
allow=ilbc
allow=gsm
allow=ulaw


**SIP.CONF*
[general]
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes

[edward] ;My Xlite softphone
type=friend
host=dynamic
secret=pass-da-word
context=from-sip
callerid=edward 100
mailbox=100
disallow=all
allow=gsm
allow=ulaw
allow=alaw
allow=ilbc
allow=g726

[phone1] ;First analog phone connected to mediatrix
type=friend
host=dynamic
secret=pass-da-word
context=from-sip
callerid=phone1 101
mailbox=101
disallow=all
allow=gsm
allow=ulaw
allow=alaw
allow=ilbc
allow=g726

[phone2] ;Second analog phone connected to mediatrix
type=friend
host=dynamic
secret=pass-da-word
context=from-sip
callerid=phone2 102
mailbox=102
disallow=all
allow=gsm
allow=ulaw
allow=alaw
allow=ilbc
allow=g726












___
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

[Asterisk-Users] Load Balancing b/w 2 asterisk servers using SIP load balancer

[Jagan Mohan]

Hi,

  I'm trying to do load balancing between 2 asterisk servers using SIP 
load balancer, provided by http://www.vovida.org

  I used the following options on lbproxy, but I get the below message 
continuously. 

./lbProxy -name seneca -reqPort 5060 -respPort 5061 -proxy A1 -proxy A2

No proxies are up - can not send message to anyone

Xlite is not able to register to the asterisk server.

Is there anything which needs to be tweaked on Asterisk side to get this
working? Please help.

Thanks,
Jagan
___
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

RE: [Asterisk-Users] Asterisk and USB ISDN controllers ...

[Vledder, Hans]

Hi Steve,

Since you don't mention what USB ISDN adapter specifically you are
thinking about, what do you think we will be able to tell you.

All I know about the adapter is what I've told you. It's a USB Colognechip
based ISDN controller - probably HCF-USB based. It's supported by Linux, but
there's no info on access to B and D channels.

Regards,
Hans
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Steven
Critchfield
Sent: Thursday, March 10, 2005 6:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk and USB ISDN controllers ...


On Thu, 2005-03-10 at 18:13 +0100, Vledder, Hans wrote:
 Guys,
 
 I am planning on building a small SIP PBX with a single ISDN line.
Currently
 I am looking into the specs of a very tiny barebone system that has an
 option Colognechip base ISDN controller. The only thing is that the ISDN
 module that comes with this barebone hooks up to the motherboard using
USB.
 My intention is to allow incoming and outgoing calls from SIP to ISDN. Is
 this setup in any way supported by *?

Since you don't mention what USB ISDN adapter specifically you are
thinking about, what do you think we will be able to tell you.

The first step would really be to ask if your specific ISDN adapter can
be used under linux. After that, can that specific ISDN adapter give
access to voice channels. What method is used to get access to the audio
and the signaling.

It may well be usable if the drivers for it implements the same API as
the current ISDN cards in use support.
-- 
Steven Critchfield [EMAIL PROTECTED]

___
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


-- 
The contents of this e-mail are intended for the named addressee only. It
contains information that may be confidential. Unless you are the named
addressee or an authorized designee, you may not copy or use it, or disclose
it to anyone else. If you received it in error please notify us immediately
and then destroy it. 

___
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

[Asterisk-Users] FW: IAX Settings

[Dennie Verstrepen]

Title: FW: IAX Settings






Hello,


Has anyone a complete overview of all the settings you can use in the iax.conf file and also where those settings can belong (e.g. in the general section, in a context of type=peer or type=user)?

Thank you in advance


Dennie




__This mail has been scanned for all known viruses by AXSWeb powered by SecuTeam NV.
_
This mail has been scanned for all known viruses by AXS Mail powered by SecuTeam NV.
Register for AXS Mail at http://www.secuteam.com!

___
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

[Asterisk-Users] Intermittent volume deterioration in conferences

[Tony Mountifield]

I wonder if anyone can suggest ways to diagnose an infuriating problem
being experienced by customers of a company I did a large Asterisk
project for.

First some background:

The system is a conferencing system using a modified MeetMe. There are
seven Asterisk boxes (we call them bridges) each with four T1 PRIs into a
TE405P. No VoIP is involved. A conference is always local to a single
bridge.  The conference leader has a control screen and may dial into the
bridge, or may instruct the bridge to dial him/her. Once the leader is in
the conference, they instruct the bridge to dial each other participant.
Each conference is recorded locally in the Asterisk system. The bridges
are in Oklahoma and all the leaders and most of the participants are all
over Texas.

The problem:

For the first three or four months of operation everything went very
well, but from early February the customer started reporting problems
with the volume of audio. Initially the reports seemed to be localized
to a particular area of Texas, and to be small in number. Over time,
they have increased in frequency and been reported from different areas.

Sometimes one participant can't be heard very well by the others, and is
also faint on the recording. Other times a participant has trouble
hearing the others, but the others are ok on the recording. There does
not seem to be any significant distortion, just faint volume.

It sounds to me like a phone network issue, but proving that is turning
out to be a nightmare. The fact that it is not confined to one bridge
but is randomly spread across them would seem to suggest it is not a
bridge hardware problem, because it is unlikely to happen in them all.
No changes were made to the hardware, Zaptel drivers or Asterisk on the
bridges since installation.

A day or so ago we disabled echo cancellation on the zap channels, to see
if that would make a difference, but it doesn't seem to have. It still
wouldn't explain why the problem did not previously exist, and started
happening spontaneously.

Sometimes if it's really difficult for people to hear, the leader closes
the conference and reverts to their older conferencing system (that our
system replaced), and reports that the volume is then fine. I don't know
where the older system is located, but I believe it is more local. This is
obviously a worrying scenario.

If anyone can suggest any ideas of ways to tackle the problem, and to
determine whether it really is the Asterisk bridges or the phone systems,
I would be very, very grateful, as it is turning into a nightmare!

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
___
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

[Asterisk-Users] Asterisk + Call hangup

[Giovanni Miano]

Scenario

 PSTN - ZAP CHANNEL - ASTERISK - SIP

When i recive call i fwd it to SIP Phone 
-  SIP PHONE ringing 

If From External Line PSTN hungup call SIP Phone Ringing too, why ?
___
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

Re: [Asterisk-Users] Asterisk + Call hangup

[Isamar Maia]

Giovani,

Are you using a X100P ?
In my case here for a similar situation, the same happens because
the Zaptel takes sometime to understand the call was hangup.
Try to play with Busydetect/busycount option in zapata.conf


Isamar


On Fri, 11 Mar 2005, Giovanni Miano wrote:

 Scenario

  PSTN - ZAP CHANNEL - ASTERISK - SIP

 When i recive call i fwd it to SIP Phone
 -  SIP PHONE ringing

 If From External Line PSTN hungup call SIP Phone Ringing too, why ?
 ___
 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



___
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

[Asterisk-Users] QuadBRI ,TDM400 and SuSE9.2 (Sencond try)

[Manuel Antonio Casal Hernández]

Hi all, this time with the complete configuration files...
We need help with our SuSe9.2 asterisk box
We have one QuadBRI and one TDM40B in an ASUS pundit R-2 barebone.
We have downloaded the bristuff (0.2.0-RC7j) and installed it without
problems.
once we downloaded and compiled asterisk, zaptel and all other stuff,
the module installation succed in this order:
modprobe zaptel
modprobe qozap
modprobe wcfsx
then the ztcfg output this:
Zaptel Configuration
==
SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1)
SPAN 2: CCS/ AMI Build-out: 399-533 feet (DSX-1)
SPAN 3: CCS/ AMI Build-out: 399-533 feet (DSX-1)
SPAN 4: 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)
Channel 04: Individual Clear channel (Default) (Slaves: 04)
Channel 05: Individual Clear channel (Default) (Slaves: 05)
Channel 06: D-channel (Default) (Slaves: 06)
Channel 07: Individual Clear channel (Default) (Slaves: 07)
Channel 08: Individual Clear channel (Default) (Slaves: 08)
Channel 09: D-channel (Default) (Slaves: 09)
Channel 10: Individual Clear channel (Default) (Slaves: 10)
Channel 11: Individual Clear channel (Default) (Slaves: 11)
Channel 12: D-channel (Default) (Slaves: 12)
Channel 13: FXO Kewlstart (Default) (Slaves: 13)
Channel 14: FXO Kewlstart (Default) (Slaves: 14)
Channel 15: FXO Kewlstart (Default) (Slaves: 15)
Channel 16: FXO Kewlstart (Default) (Slaves: 16)
16 channels configured.
but when we start asterisk it stops at zapata.conf parsing:
[chan_zap.so] = (Zapata Telephony w/PRI)
  == Parsing '/etc/asterisk/zapata.conf': Found
Mar 11 20:29:37 ERROR[5512]: chan_zap.c:6467 mkintf: Unable to get
parameters
Mar 11 20:29:37 ERROR[5512]: chan_zap.c:10247 setup_zap: Unable to
register channel '1-2'
Mar 11 20:29:37 WARNING[5512]: loader.c:345 ast_load_resource:
chan_zap.so: load_module failed, returning -1
  == Unregistered channel type 'Tor'
  == Unregistered channel type 'Zap'
Mar 11 20:29:37 WARNING[5512]: loader.c:440 load_modules: Loading module
chan_zap.so failed!
Any kind of suggestions or help will be welcome, We are stucked and
dessesperated with this issue...:(
Thanks very much in advance for your help, again.
-- zaptel.conf
#loadzone=no
#defaultzone=us
loadzone=nl
defaultzone=nl
# qozap span definitions
# most of the values should be bogus because we are not really zaptel
span=1,1,3,ccs,ami
span=2,0,3,ccs,ami
span=3,0,3,ccs,ami
span=4,0,3,ccs,ami
bchan=1,2,4,5,7,8,10,11
dchan=3,6,9,12
fxoks=13-16
-- zapata.conf
;
; Zapata telephony interface
;
; Configuration file
[channels]
;
; Default language
;
;language=en
;
; Default context
;
;
switchtype = euroisdn
; p2mp TE mode (for connecting ISDN lines in point-to-multipoint mode)
;signalling = bri_cpe_ptmp
; p2p TE mode (for connecting ISDN lines in point-to-point mode)
signalling = bri_cpe
; p2mp NT mode (for connecting ISDN phones in point-to-multipoint mode)
;signalling = bri_net_ptmp
pridialplan = local
prilocaldialplan = local
nationalprefix = 0
internationalprefix = 00
echocancel = yes
context=RDSI
group = 1
; S/T port 1
channel = 1-2
; S/T port 2
channel = 4-5
; S/T port 3
channel = 7-8
; S/T port 4
channel = 10-11
switchtype =fxo_ks
context=EXTENSIONES
group = 2
channel =13-16
-- extensions.conf
[RDSI]
exten = _X.,1,Dial(ZAP/g2/${EXTEN},60)
exten = _X.,2,Hangup
[EXTENSIONES]
exten = _X.,1,Dial(ZAP/g1/${EXTEN},60)
exten = _X.,2,Hangup
; if the called party is busy
exten = _X.,102,Playtones(busy)
exten = _X.,103,Wait(10)
exten = _X.,104,Hangup
; if all zaptel channels in that group are in use
; or the D channels are down
exten = _X.,202,Playtones(congestion)
exten = _X.,203,Wait(10)
exten = _X.,204,Hangup
-- asterisk output
[chan_zap.so] = (Zapata Telephony w/PRI)
  == Parsing '/etc/asterisk/zapata.conf': Found
Mar 11 20:29:37 ERROR[5512]: chan_zap.c:6467 mkintf: Unable to get
parameters
Mar 11 20:29:37 ERROR[5512]: chan_zap.c:10247 setup_zap: Unable to
register channel '1-2'
Mar 11 20:29:37 WARNING[5512]: loader.c:345 ast_load_resource:
chan_zap.so: load_module failed, returning -1
  == Unregistered channel type 'Tor'
  == Unregistered channel type 'Zap'
Mar 11 20:29:37 WARNING[5512]: loader.c:440 load_modules: Loading module
chan_zap.so failed!
--
Manuel Casal Hernández
[EMAIL PROTECTED]
-
Dep. Desarrollo y Consultoría
[EMAIL PROTECTED]
http://www.e-sistemas.net
[EMAIL PROTECTED]
(T) 902 678 006



smime.p7s
Description: S/MIME Cryptographic Signature
___
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

[Asterisk-Users] Incomplete incoming fax using spandsp 0.0.2pre10

[Steve FH]

Hi,

I have successfully compiled spandsp 0.0.2pre10 with * 1.05 which can accept
inbound fax calls. However, all fax received are incomplete  (the first 10%
of an A4 page is fine, the remaining is either missing or garbled). I
suspect this is due to 'training error' (see below) which, according to
Steve Underwood's postings, cannot be resolved further. I wonder if it would
help to upgrade my test machine which uses Asus m/b + Celeron 2.4GHz cpu +
512MB Ram + 2 Clone X100P cards (MD3200 chipset based) + RH 7.3. Secondly,
will the use of a Digium card fix the problem?

Thank you for sharing your inputs.

Steve FH


-- Executing RxFAX(Zap/2-1,
/var/spool/asterisk/incoming/1110539705.0.tif) in new stack
Mar 11 19:15:27 NOTICE[22614]: chan_zap.c:4160 zt_read: Fax detected, but no
fax extension
DCS with final frame tag
In state 9
Coarse carrier frequency 1694.12 (54)
Training failed (sequence failed)
Coarse carrier frequency 1699.59 (66)
Training error 0.687966
Training succeeded (constellation mismatch 0.834685)
Start rx document
Start rx page - compression 2
Training failed (sequence failed)
Coarse carrier frequency 1699.59 (66)
Training error 0.544494
Training succeeded (constellation mismatch 0.591839)
EOP with final frame tag
In state 5
DCN with final frame tag
In state 8
  == Auto fallthrough, channel 'Zap/2-1' status is 'UNKNOWN'


___
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

[Asterisk-Users] QuadBRI ,TDM400 and SuSE9.2 (Sencond try)

[Manuel Antonio Casal Hernández]

Hi all, this time with the complete configuration files...
We need help with our SuSe9.2 asterisk box
We have one QuadBRI and one TDM40B in an ASUS pundit R-2 barebone.
We have downloaded the bristuff (0.2.0-RC7j) and installed it without
problems.
once we downloaded and compiled asterisk, zaptel and all other stuff,
the module installation succed in this order:
modprobe zaptel
modprobe qozap
modprobe wcfsx
then the ztcfg output this:
Zaptel Configuration
==
SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1)
SPAN 2: CCS/ AMI Build-out: 399-533 feet (DSX-1)
SPAN 3: CCS/ AMI Build-out: 399-533 feet (DSX-1)
SPAN 4: 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)
Channel 04: Individual Clear channel (Default) (Slaves: 04)
Channel 05: Individual Clear channel (Default) (Slaves: 05)
Channel 06: D-channel (Default) (Slaves: 06)
Channel 07: Individual Clear channel (Default) (Slaves: 07)
Channel 08: Individual Clear channel (Default) (Slaves: 08)
Channel 09: D-channel (Default) (Slaves: 09)
Channel 10: Individual Clear channel (Default) (Slaves: 10)
Channel 11: Individual Clear channel (Default) (Slaves: 11)
Channel 12: D-channel (Default) (Slaves: 12)
Channel 13: FXO Kewlstart (Default) (Slaves: 13)
Channel 14: FXO Kewlstart (Default) (Slaves: 14)
Channel 15: FXO Kewlstart (Default) (Slaves: 15)
Channel 16: FXO Kewlstart (Default) (Slaves: 16)
16 channels configured.
but when we start asterisk it stops at zapata.conf parsing:
[chan_zap.so] = (Zapata Telephony w/PRI)
   == Parsing '/etc/asterisk/zapata.conf': Found
Mar 11 20:29:37 ERROR[5512]: chan_zap.c:6467 mkintf: Unable to get
parameters
Mar 11 20:29:37 ERROR[5512]: chan_zap.c:10247 setup_zap: Unable to
register channel '1-2'
Mar 11 20:29:37 WARNING[5512]: loader.c:345 ast_load_resource:
chan_zap.so: load_module failed, returning -1
   == Unregistered channel type 'Tor'
   == Unregistered channel type 'Zap'
Mar 11 20:29:37 WARNING[5512]: loader.c:440 load_modules: Loading module
chan_zap.so failed!
Any kind of suggestions or help will be welcome, We are stucked and
dessesperated with this issue...:(
Thanks very much in advance for your help, again.
-- zaptel.conf
#loadzone=no
#defaultzone=us
loadzone=nl
defaultzone=nl
# qozap span definitions
# most of the values should be bogus because we are not really zaptel
span=1,1,3,ccs,ami
span=2,0,3,ccs,ami
span=3,0,3,ccs,ami
span=4,0,3,ccs,ami
bchan=1,2,4,5,7,8,10,11
dchan=3,6,9,12
fxoks=13-16
-- zapata.conf
;
; Zapata telephony interface
;
; Configuration file
[channels]
;
; Default language
;
;language=en
;
; Default context
;
;
switchtype = euroisdn
; p2mp TE mode (for connecting ISDN lines in point-to-multipoint mode)
;signalling = bri_cpe_ptmp
; p2p TE mode (for connecting ISDN lines in point-to-point mode)
signalling = bri_cpe
; p2mp NT mode (for connecting ISDN phones in point-to-multipoint mode)
;signalling = bri_net_ptmp
pridialplan = local
prilocaldialplan = local
nationalprefix = 0
internationalprefix = 00
echocancel = yes
context=RDSI
group = 1
; S/T port 1
channel = 1-2
; S/T port 2
channel = 4-5
; S/T port 3
channel = 7-8
; S/T port 4
channel = 10-11
switchtype =fxo_ks
context=EXTENSIONES
group = 2
channel =13-16
-- extensions.conf
[RDSI]
exten = _X.,1,Dial(ZAP/g2/${EXTEN},60)
exten = _X.,2,Hangup
[EXTENSIONES]
exten = _X.,1,Dial(ZAP/g1/${EXTEN},60)
exten = _X.,2,Hangup
; if the called party is busy
exten = _X.,102,Playtones(busy)
exten = _X.,103,Wait(10)
exten = _X.,104,Hangup
; if all zaptel channels in that group are in use
; or the D channels are down
exten = _X.,202,Playtones(congestion)
exten = _X.,203,Wait(10)
exten = _X.,204,Hangup
-- asterisk output
[chan_zap.so] = (Zapata Telephony w/PRI)
   == Parsing '/etc/asterisk/zapata.conf': Found
Mar 11 20:29:37 ERROR[5512]: chan_zap.c:6467 mkintf: Unable to get
parameters
Mar 11 20:29:37 ERROR[5512]: chan_zap.c:10247 setup_zap: Unable to
register channel '1-2'
Mar 11 20:29:37 WARNING[5512]: loader.c:345 ast_load_resource:
chan_zap.so: load_module failed, returning -1
   == Unregistered channel type 'Tor'
   == Unregistered channel type 'Zap'
Mar 11 20:29:37 WARNING[5512]: loader.c:440 load_modules: Loading module
chan_zap.so failed!
--
Manuel Casal Hernández
[EMAIL PROTECTED]
-
Dep. Desarrollo y Consultoría
[EMAIL PROTECTED]
http://www.e-sistemas.net
[EMAIL PROTECTED]
(T) 902 678 006

___
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

Re: [Asterisk-Users] how do i get rid of this blasted echo !!!

[support]

Hi Herman,

Look at the bottom of your phones and compare the REN values of both. Do
they both value of REN 1.0?  I think the one with the problem might have
an REN value other than one.  You tell me!

Errol Samuels
Don't let SIP Drive you crazy, use IAX2



 On the echo...

 I have 2 extensions, with different analog phones.
 The one works fine, the other echos and scratches
 like mad !!

 I have switched the ports, cables etc but its ALWAYS
 the same phone...

 Maybe this could be it ?

 Is it ok from a SIP phone ?

 Herman cremer





 ___
 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



___
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

RE: [Asterisk-Users] AAH 0.06 - IAX Connection Over NAT Firewall

[C. Tomlinson]

Title: [Asterisk-Users] SIP to H.323 no audio








As I understand it if you use that deny
statement, all calls will be disallowed, hence why you couldnt get any
incoming calls.

If you add an allow line with the VOIP
providers IP that it send the call from, you can then use that line to disallow
everything else.



It is just a security feature really.



C











From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wiley Siler
Sent: 11 March 2005 07:23
To: Asterisk
 Users Mailing List - Non-Commercial Discussion; Asterisk Users Mailing List - Non-Commercial Discussion;
[EMAIL PROTECTED] Digium. Com
Subject: RE: [Asterisk-Users] AAH
0.06 - IAX Connection Over NAT Firewall









OK. I removed the deny statement
they have me using and now I can get incoming calls.











Do I need the deny 0.0.0.0/0.0.0.0 statement?











Thanks,





Wiley















From:
[EMAIL PROTECTED] on behalf of Wiley Siler
Sent: Thu 3/10/2005 11:59 PM
To: Asterisk
 Users Mailing List - Non-Commercial Discussion;
[EMAIL PROTECTED] Digium. Com
Subject: [Asterisk-Users] AAH 0.06
- IAX Connection Over NAT Firewall





Hello
all,

I
am having trouble getting my IAX based Voip provider setup. Any pointers
are welcome.

So
here is the deal. I am registered up and I can make outgoing calls but
incoming calls fail.
Configs all look good I thought.

My PBX is behind our firewall with a direct NAT of one to one for an external
IP.
IAX port is forwarded UDP and TCP to the internal IP.

*
shows good registration and Ips and ports show solid.

Within
my AAH I have the registration like the provier said to do. I get
absolutely nothing on the incoming. IAX2 debug shows nothing on
incoming. Just a fast busy. Outgoing works perfectly however.

I
have a defined DID in the AMP interface and verified it is written to confs and
have reloaded.

Can
anyone tell me another way to verify that something is coming in? Or did
I just miss something on the whole IAX over NAT?



Thanks
all,

Wiley














___
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

[Asterisk-Users] Unable to create Zap channel when dialing using a bri cellular gateway

[David Masure]

Hi 
all,


I have an asterisk 
box set up with a bri card (using zaphfc). I have a bri cellular gateway 
connected to it beacuse I'd like to route all my cellular calls through that 
gateway.

The probel I 
encounter is that when trying to dial a phone number, I've the message : unable 
to create a zap channel.

My card is normally 
well configured because when connected to the NT, It works perfectly... 
The gateway is configured as a NT as well so no worry...

Has anyone an idea 
of what I should look for ?

Thank 
you

David 
Masure

___
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

Re: [Asterisk-Users] multiple enum results

[Duane]

On Fri, March 11, 2005 14:20, Jon Lewis said:

 Before I hack this into enumlookup.agi or write a new one, I'm just
 curious, have others done this, or are there other better ways to do what
 I'm looking to do?

There was talk on the dev list on fixing this, not sure how far things
went. I got tired of not having proper enum routing in asterisk I hacked
up a php script ages ago to handle it...

http://www.e164.org/enum.phps

-- 
Best regards,
 Duane

http://www.cacert.org - Free Security Certificates
http://www.nodedb.com - Think globally, network locally
http://www.sydneywireless.com - Telecommunications Freedom
http://happysnapper.com.au - Sell your photos over the net!
http://e164.org - Using Enum.164 to interconnect asterisk servers

In the long run the pessimist may be proved right,
but the optimist has a better time on the trip.

___
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

[Asterisk-Users] Asterisk@home 0.6 + Modem.conf

[Giovanni Miano]

How i use Modem.conf with AMP? It allow only ZAP IAX or SIP Trunk 

is there a patch to manage it ? 


And how configure extension.conf to use Modem/g1 channels when Zap/g0
channels are busy ?
___
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

[Asterisk-Users] Incoming echo cancel

[Roberto Vargas]

Hi, 


  I'm having the same problems in echo cancellations that are mentioning
in this mail of the list
http://lists.digium.com/pipermail/asterisk-users/2003-July/016073.html ,
but I haven't found some reply to this mail. 

I haven't echo problem on outcoming calls but echo cancellation is
disabled in zaptel channels in incoming calls. Status of zaptel channel
is the next:

localhost*CLI zap show channel 32
Channel: 32LI
File Descriptor: 49
Span: 2
Extension: 958238500
Dialing: no
Context: incoming
Caller ID string: 685975350
Destroy: 0
InAlarm: 0
Signalling Type: PRI Signalling
Owner: Zap/32-1
Real: Zap/32-1
Callwait: None
Threeway: None
Confno: -1
Propagated Conference: -1
Real in conference: 0
DSP: yes
Relax DTMF: no
Dialing/CallwaitCAS: 0/0
Default law: alaw
Fax Handled: no
Pulse phone: no
Echo Cancellation: 256 taps, currently OFF
PRI Flags: Call
PRI Logical Span: Implicit
Actual Confinfo: Num/0, Mode/0x
Actual Confmute: No
Actual Hookstate: Onhook



I don't know because Asterisk doesn't enable echo cancelation.




Roberto Vargas.

___
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

Re: [Asterisk-Users] how do i get rid of this blasted echo !!!

[Herman Cremer]

Thanks Error.

I have switched to IAX looong agomuch better !
Just battle when doing double NAT :)

I dont have the phones here with me,
but lets say its different...is there away
to adjust the channel to fix the err ?

-herman



On Fri, 2005-03-11 at 13:24, [EMAIL PROTECTED] wrote:
 Hi Herman,
 
 Look at the bottom of your phones and compare the REN values of both. Do
 they both value of REN 1.0?  I think the one with the problem might have
 an REN value other than one.  You tell me!
 
 Errol Samuels
 Don't let SIP Drive you crazy, use IAX2
 
 
 
  On the echo...
 
  I have 2 extensions, with different analog phones.
  The one works fine, the other echos and scratches
  like mad !!
 
  I have switched the ports, cables etc but its ALWAYS
  the same phone...
 
  Maybe this could be it ?
 
  Is it ok from a SIP phone ?
 
  Herman cremer
 
 
 
 
 
  ___
  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
 
 
 
 ___
 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

___
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

Re: [Asterisk-Users] Unable to create Zap channel when dialing using a bri cellular gateway

[Niksa Baldun]

Obviously, your ISDN gateway is misconfigured somehow. I would suggest
that you configure the gateway to dial some extension on your * box and
see if incoming calls work. If they don't, then there is a problem with
configuration of gateway's ISDN interface. If incoming calls work, then
it is possible that the gateway is rejecting outgoing calls based on
number called (I had that problem once), or perhaps you just forgot to
pay the bill to your mobile operator :)).

Niksa


David Masure wrote:

  
  
 Hi all,
  
  
 I have an asterisk box set up with a bri card (using zaphfc).  I have
 a bri cellular gateway connected to it beacuse I'd like to route all
 my cellular calls through that gateway.
  
 The probel I encounter is that when trying to dial a phone number,
 I've the message : unable to create a zap channel.
  
 My card is normally well configured because when connected to the NT,
 It works perfectly...  The gateway is configured as a NT as well so no
 worry...
  
 Has anyone an idea of what I should look for ?
  
 Thank you
  
 David Masure
  



___
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

___
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

[Asterisk-Users] CDR database

[Ronald Wiplinger]

I am looking at AMP and read All the graphic  reports are based over 
the CDR database.
How do I get the CDRs into a database?

bye
Ronald
___
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

[Asterisk-Users] SIP signalling and RTP to different servers

[Marc Storck]

Hello,
we're in process of testing an interconnection with a trans-european 
carrier. But the carrier wants the SIP signalling to server 1 and the 
RTP stream to server 2. How do I configure asterisk to work with that 
type of installation. It seems they are using NexTone as SIP Signaling 
and RTP servers. Can someone help me???

Regards,
Marc
--
CTOMarc Storck
MS Networks SA [EMAIL PROTECTED]
IT Service Providerhttp://www.msnetworks.lu
15, route d'Esch   Phone: +352 2727 3030
L-4450 Belvaux Fax:   +352 2727 3060
--- MS Networks powered service ---
http://www.LuxAdmin.com   Hosting and housing solutions
---
___
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

RE: [Asterisk-Users] TE110P experiance

[Robbie Hughes]

   - the PCI ID of the card seems to change over time which means that
loading the module does not always recognise the card, only way to reset
this is to power cycle the machine
I noticed this behaviour as well.
i thought it was my motherboard wrongly assigning irq values -
the symptoms i noticed were:
irq value set by me - machine starts, module loads, all functional
reboot
irq value reset to another /shared/ irq - machine starts, module fails
reboot
irq value set by me again -  machine starts, module fails
check irq - it has been reset to something else
irq value set by me - machine starts, module loads, all functional
unfortunately my bios doesn't allow manual assigning of irqs - i have to 
swap them arond based on the ones it gives me...
i ended up disabling my usb bus as i don't need it...

i can't find any consistency to it and am living in fear of the reboot..
very odd..
___
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

[Asterisk-Users] SIP Phone Unreachable

[Renato Mintz]

Hi Folks,

I found a strange problem trying to install a system on a customer. I
have the following network configuration:

 Asterisk - Router (NAT) - Internet - Router (NAT) - Grandstream Phone

The routers are low end D-Link router + broadband access. The router
near asterisk has 5060 and 1-10009 ports opened and assigned to
Asterisk server. The router near the phone has default configuration:
outgoing ok, incoming blocked.

I have Qualify = 1000. As soon as * is restarted I get a message
telling the phone is unreachable. Looking at SIP debug I see *
transmitting OPTIONS and receiving OK but it seems that * discards the
OKs, because it always transmits OPTIONS 4 times (and receives 4 OKs),
stop a little and begin transmitting OPTIONS again.

Looking at the SIP messages I found that the Call-ID in the OPTIONS
message uses the Asterisk EXTERNAL IP address but the OK coming from
the GS Phone has its Call-ID with the Asterisk INTERNAL IP address.

I run ethereal near the phone and the OK it sends has Asterisk
EXTERNAL IP address! Somebody is translating the EXTERNAL IP into the
INTERNAL IP at the Call-Id header.

I also run tcpdump at the Asterisk Server and the result is the same
as the sip debug.

My simple conclusion is: the router is opening the SIP message and
translating the Call-Id header IP, but I don't believe in that.

Any clue?

Thanks?

Renato
___
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

Re: [Asterisk-Users] CDR database

[Yair Hakak]

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


On Fri, 11 Mar 2005 19:58:37 +0800, Ronald Wiplinger [EMAIL PROTECTED] wrote:
 I am looking at AMP and read All the graphic  reports are based over
 the CDR database.
 How do I get the CDRs into a database?
 
 bye
 
 Ronald
 
 ___
 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

___
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

Re: [Asterisk-Users] Asterisk@home silly problem, please help!

[JunkMail]

Solved!
The problem was that capiinit start can only be done by user root and
asterisk is started as user asterisk.
Once I edited sudo (visudo) and gave permission, the problem was solved.

Regards

M.G.

- Original Message - 
From: Junk Mail [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Wednesday, March 09, 2005 11:12 PM
Subject: [Asterisk-Users] [EMAIL PROTECTED] silly problem, please help!


 Hi all!

 After much struggling I got my [EMAIL PROTECTED] working fine AND making use 
 of two
 AVMFritz!PCI cards. Really nice !  (kernel 2.4.2x)

 There's however a silly glitch that's getting on my nerves, and, kind of a
 newbie that I am to linux, it should be easy to get help :

 -- capiinit start MUST BE run before Asterisk. (any other way makes *
not
 to start because chan_capi doesn't find CAPI support)

 You must find this an easy thing, as I did. So I entered /etc/rc.d/ and
 inserted capiinit start to start as early as possible. Also added some
 lines of junk text so to see them going by as the system boots...

 What's making me desperate is that the lines go by, capiinit is, in fact,
 runned, and Asterisk still fails in the end.
 I login and type my very first command asterisk -vvvc and it then starts
 with no trouble.

 Is this strange or what ?

 Thanks in advance for your help.

 M.G.

 ___
 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


___
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

[Asterisk-Users] IAX, double NAT

[Herman Cremer]

has anyone managed to get IAX client (firefly 3rd party version)
to work,
where the *Server is behind single NAT,
with port forwarding enabled on the NAT router, and 
the client behind double NAT ?

clients behind single nat to * work fine.

hermancremer

___
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

[Asterisk-Users] One single record file for a meetme monitor?

[taintedham-mailinglists]

I'm trying to figure out the best way to record a
conference.

Many people suggest something like this:
exten = 2060,1,Answer
exten = 2060,2,Wait(1)
exten = 2060,3,Monitor(wav,myfilename)
exten = 2060,4,Meetme(1,ps) 

However, this creates two files for each user that
connects to the meetme.  (I know they can be mux'd
together to make one with sox..I've done that too) 
However, you still get 10 files if 10 users enter the
meetme.

I'd really like to be able to simple record a single
file with all the channels mux'd together.

Someone suggested executing a script and having the
monitor application join the meetme.  However, I have
yet to see this work correctly and it isn't the
best solution because I've got to have some logic to
add the local listener when the first person enters...
and exit when the last person exits.

Anyways, just wanted to see if any of you have this
worked out already.  I really think there should be an
option on the meetme.

Thanks,
Dave
___
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

Re: [Asterisk-Users] SetCallerID({$NEWCALLERID})

[Iqbal]

that would do it, the $ is in the wrong place

Iqbal

On 3/11/2005, beonice [EMAIL PROTECTED] wrote:


--- Steven Frazier [EMAIL PROTECTED] wrote:
 I am trying to SetCallerID to a variable I have
 defined.  This obviously is
 wrong.  It actually sets the caller ID to
 $NEWCALLERID.  I have search
 through the examples on wiki but wasn't able to find
 something similar to
 see what I was doing wrong.  Could someone tell me
 the correct way to
 SetCallerID to a defined variable?

  exten = 2125551212,5,SetCallerID({$NEWCALLERID})

  --- snipped the rest ---

Off-hand, not having actually tested this, I'd guess
that you have the $ in the wrong place. Move it one
character to the left.

Cheers,
Maya




__
Do you Yahoo!?
Yahoo! Small Business - Try our new resources site!
http://smallbusiness.yahoo.com/resources/
___
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


___
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

[Asterisk-Users] Asterisk@home 0.6 + bristuff

[Giovanni Miano]

How install bristuff in [EMAIL PROTECTED] ? 
 
i tried version 0.2.Rca to last RC7k and when try to compile zaptel
(after patched it) i've this error:
 
make: *** [zaptel.o] Error 1
___
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

[Asterisk-Users] Voicemail - No Audio Output!

[Julius Kidubuka]

Hi all,

I am able to receive voicemail in my mail box but when I try to play the
audio file attachment, I hear nothing at all (yet the caller on the other
end does leave a voicemail message)!

Anyone had a similar problem before? Ideas are welcome!

Note: I am using [EMAIL PROTECTED] 0.6

Thanks in advance,
-- 
Rgds,
Julius Kidubuka.
My advice to you is get married: if you find a good wife you'll be happy;
if not, you'll become a philosopher.




___
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

RE: [Asterisk-Users] Panasonic TDA200 E1 - E100P negotiation issues

[Peter Svensson]

On Fri, 11 Mar 2005, James Bean wrote:

 Whooppss had pri_cpe set, redid the debug as attached.
 
 They seem the same but just in case.

Asterisk does not see anything coming in on the D channel. What does 
zttool say about the state of the link?

As I said before, if the card is an isdn card you need to use ccs 
signalling. Cas signalling is unusual, but possible, over an E1. Can you 
find out the model number of the E1 card in the Panasonic pbx?

Peter

___
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

Re: [Asterisk-Users] IAX, double NAT

[Time Bandit]

 has anyone managed to get IAX client (firefly 3rd party version)
 to work,
 where the *Server is behind single NAT,
 with port forwarding enabled on the NAT router, and
 the client behind double NAT ?
 
 clients behind single nat to * work fine.
Strange, I tested with iaxcomm and this was the setup :

Asterisk - NAT - Internet - NAT - NAT - NAT - iaxcomm (x3)

and everything was working.

I would suggest that you try with a different IAX client. 
Take iaxcomm (http://iaxclient.sourceforge.net/iaxcomm/index.html) 
or MediaX (http://www.marccharbonneau.com/asterisk/mediaxphone.php) 
and try your test again.

hth
___
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

[Asterisk-Users] Some Hardware Advice

[Brett, Gary]

Hi there

Just a quick post to ask you guys if you've had any bad (even good)
experiences using current model HP or Dell servers ?? specifically the HP
proliant ML110 and the Dell Poweredge 1800 SATA, (but I will welcome your
recommendations on any current Models) . I will be rolling out some small to
medium systems with a max 100 Sip extensions and 60 outbound (2 x e1) for
the larger rollouts and as little as  5-10 users for the smaller systems (
zap channels on TDM400P's) . 

Any advice would be greatly appreciated

Gary
___
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

[Asterisk-Users] What is that area code?

[Ronald Wiplinger]

Can anybody help me  and explain me the following area codes:
1-340   US-USVI
1-670   US-CNMI
1-710   US-Governement
1-787   US-Poerto Rico
1-802 ~ 1-808 ???
1-939   US-Poerto Rico
1-600   Canada
Are the above codes are USA Continental tarrif (NuFone / Broadvoice ... 
)

What are the codes for mobile phones in USA?
bye
Ronald
___
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

RE: [Asterisk-Users] how do i get rid of this blasted echo !!!

[Brett, Gary]

So is it accepted as standard that compiling with MMX will help improve echo
type issues ?


-Original Message-
From: Herman Cremer [mailto:[EMAIL PROTECTED] 
Sent: 11 March 2005 11:56
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] how do i get rid of this blasted echo !!!

Thanks Error.

I have switched to IAX looong agomuch better !
Just battle when doing double NAT :)

I dont have the phones here with me,
but lets say its different...is there away
to adjust the channel to fix the err ?

-herman



On Fri, 2005-03-11 at 13:24, [EMAIL PROTECTED] wrote:
 Hi Herman,
 
 Look at the bottom of your phones and compare the REN values of both. Do
 they both value of REN 1.0?  I think the one with the problem might have
 an REN value other than one.  You tell me!
 
 Errol Samuels
 Don't let SIP Drive you crazy, use IAX2
 
 
 
  On the echo...
 
  I have 2 extensions, with different analog phones.
  The one works fine, the other echos and scratches
  like mad !!
 
  I have switched the ports, cables etc but its ALWAYS
  the same phone...
 
  Maybe this could be it ?
 
  Is it ok from a SIP phone ?
 
  Herman cremer
 
 
 
 
 
  ___
  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
 
 
 
 ___
 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

___
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
___
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

Re: [Asterisk-Users] Some Hardware Advice

[Dennis Webb]

I know the g4 370's(I think that's the model) have issues with zaptel cards and the new intel chipset. I've also seen people have issues with the 3ware SATA raid cards if that is what ships in the machine. The last thing I saw recently was a dell server having trouble with NMI, and disabling the USB seemed to fix it. Google the list for the above things to find more info.

It seems the main issue with hardware is irq and pci latency. I've also seen people mention hyperthreading causing some issues.

On Fri, 2005-03-11 at 07:47, Brett, Gary wrote:

Hi there

Just a quick post to ask you guys if you've had any bad (even good)
experiences using current model HP or Dell servers ?? specifically the HP
proliant ML110 and the Dell Poweredge 1800 SATA, (but I will welcome your
recommendations on any current Models) . I will be rolling out some small to
medium systems with a max 100 Sip extensions and 60 outbound (2 x e1) for
the larger rollouts and as little as  5-10 users for the smaller systems (
zap channels on TDM400P's) . 

Any advice would be greatly appreciated

Gary
___
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



___
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

RE: [Asterisk-Users] how do i get rid of this blasted echo !!!

[Dennis Webb]

I've added MMX and it didn't help. I also did the CFLAGS+=-march=pentium4 with no help there either. The more I search, the more I found and I'm down to disabling any hardware not used in the box such as USB and recompiling the kernel with a fresh copy from kernel.org. It seems there were a lot of problems solved when 2.6.9 came out.

If I ever get mine fixed, I will try to post everything I did.

On Fri, 2005-03-11 at 07:50, Brett, Gary wrote:

So is it accepted as standard that compiling with MMX will help improve echo
type issues ?


-Original Message-
From: Herman Cremer [mailto:[EMAIL PROTECTED] 
Sent: 11 March 2005 11:56
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] how do i get rid of this blasted echo !!!

Thanks Error.

I have switched to IAX looong agomuch better !
Just battle when doing double NAT :)

I dont have the phones here with me,
but lets say its different...is there away
to adjust the channel to fix the err ?

-herman



On Fri, 2005-03-11 at 13:24, [EMAIL PROTECTED] wrote:
 Hi Herman,
 
 Look at the bottom of your phones and compare the REN values of both. Do
 they both value of REN 1.0?  I think the one with the problem might have
 an REN value other than one.  You tell me!
 
 Errol Samuels
 Don't let SIP Drive you crazy, use IAX2
 
 
 
  On the echo...
 
  I have 2 extensions, with different analog phones.
  The one works fine, the other echos and scratches
  like mad !!
 
  I have switched the ports, cables etc but its ALWAYS
  the same phone...
 
  Maybe this could be it ?
 
  Is it ok from a SIP phone ?
 
  Herman cremer
 
 
 
 
 
  ___
  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
 
 
 
 ___
 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

___
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
___
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



___
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

[Asterisk-Users] Quescom AS/400 GSM Gateway + Asterisk

[Jean-Michel Hiver]

Hi List,
I'm wondering if anybody on the list managed to get one of these beasts 
working with asterisk?

FYI They're Windows NT embedded (yuk!) based H.323 / SIP compliant 
devices with a *very* complicated admin interface. Can't figure out how 
to get it working... yet :)

Cheers,
Jean-Michel.
___
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

Re: [Asterisk-Users] IAX2 800 Termination

[Michael Graves]

I have used www.clearpath1.com for a year. Very reliable. Nice people
too.

Michael

On Thu, 10 Mar 2005 15:07:27 -0600, Linn Boyd wrote:

I am looking for a good provider for IAX2/800 termination. I am 
currently using FreeWorldTel and wanted to use NuFone but it seems that 
both of them don't provide customer service. FreeWorld has terrible 
voice quality and NuFone never answers their phone or responds to messages.

Thanks,

Linn

___
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



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

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



___
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

Re: [Asterisk-Users] What is that area code?

[Eric Wieling]

Ronald Wiplinger wrote:
Can anybody help me  and explain me the following area codes:
1-340   US-USVI
1-670   US-CNMI
1-710   US-Governement
1-787   US-Poerto Rico
1-802 ~ 1-808 ???
1-939   US-Poerto Rico
1-600   Canada
Are the above codes are USA Continental tarrif (NuFone / Broadvoice ... 
Puerto Rico is not part of the Continental USA.  USVI may be US Virgin 
 Islands, which is not part of the Continental USA.  For telcom 
Continental USA usually means the 48 USA states (which is all USA 
states, except Alaska and Hawaii).

What are the codes for mobile phones in USA?
The USA does not have mobile phone area codes.  Mobile phones use the 
same area codes as other phones.  In the USA it does not cost extra to 
call a mobile phone.
___
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

[Asterisk-Users] Manager (5038)

[Anderson Alves de Albuquerque]

 I am using this site 
(http://www.digium.com/asterisk_handbook/manager.html) to access port 5038 
(manager port). But I have problem below.
 Is there another step that I need to do?

---
Trying 127.0.0.1...
Connected to localhost.
Escape character is '^]'.
Asterisk Call Manager/1.0
Action:Login
Username:theuser
Secret:somepassword

Response: Error
Message: Missing action in request

Action: Login
Username: theuser
Secret: somepassword

Response: Success
Message: Authentication accepted

---


___
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

Re: [Asterisk-Users] IAX2 800 Termination

[Gary Reuter]

What date was this?
I've been waiting since Jan 24 on my 'pending' US50CA number -- I
think you got VERY lucky!


On Thu, 10 Mar 2005 23:59:00 -0700, Paul Fielding [EMAIL PROTECTED] wrote:
 Mine was up with LiveVoip within 30 minutes of ordering via the online
 website.  And that was at midnight on a Saturday night.  Of course, they
 don't guarantee that, I think I just got lucky... :)

___
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

Re: [Asterisk-Users] Application SetVarCDR

[Matthew Boehm]

I know this isn't the best way but I stopped using 's' and I use _X. on
everything now. It is really stupid to say That person dialed the number
's' or That phone call's final destination was 's'. That doesn't help
anything; debugging nor billing.

My $0.02
Matthew

William M. Sandiford wrote:
 Hello:

 I found a reference to the application SetVarCDR in the following
 post but I don't seem to have this available to me in my version of
 *.

 HYPERLINK

http://lists.digium.com/pipermail/asterisk-cvs/2005-February/005337.htmlht
tp://lists.digium.com/pipermail/asterisk-cvs/2005-February/005337.html

 My version of * is CVS-HEAD-03/10/05-18:42:35

 I would like to change the value of the src and dst variables in the
 CDR as I sometimes find that they don't have entirely accurate
 information.  For example my dst field quite often has a value of s
 because I do my call processing in the s extension.  This is no good
 to me.

 Does anyone know How I can get access to the functions mentioned
 above.  FYI I have tried doing a set var just before I dial like
 this:

 exten = s,13,SetVar(CDR(dst)=12345)
 exten = s,14,Dial(HYPERLINK
 mailto:SIP/[EMAIL PROTECTED]SIP/[EMAIL PROTECTED])

 but that doesn't seem to work...I still get s in the dst field of my
 CDR

 Regards,
 Bill



 ___
 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

___
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

Re: [Asterisk-Users] OT: Best DB

[Matthew Boehm]

 If you're a VoIP provider, and are trying to
 provide a near carrier-grade service, postgres shines.

I'm not disagreeing with you, but we are a CLEC and we do provide
'carrier-grade' service and we use MySQL everywhere.

IMHO, MySQL is just so much more easy to use, install and maintain.
phpMyAdmin makes it even easier.

 -Matthew

Mohit Muthanna wrote:
 On Thu, 10 Mar 2005 19:14:36 -0500, Giudice, Salvatore
 [EMAIL PROTECTED] wrote:
 I vote for MySQL. PostgreSQL is fine, but MySQL handles much better
 under extreme load. MySQL is also usually touted as being generally

 I'd have to (respectfully) disagree with that... MySQL just cannot
 handle high load or large datasets... it's inherent design does not
 allow it to scale too well...

 I lost countless hours trying to optimize disk / filesystem
 distribution, SQL queries, kernel parameters etc. etc. to get MySQL to
 _not crawl_. After many failed attempts, I switched to Postgres and
 haven't looked back.

 I personally believe there is a right tool for the right job. MySQL
 works great for small datasets and (relatively) lighter load. Infact,
 it shines there. But don't expect it to perform as your database grows
 in orders of magnitude.

 Postgres is certainly a database that is recommended (IMHO) for
 production environments. If you're a VoIP provider, and are trying to
 provide a near carrier-grade service, postgres shines.

 Moht.

___
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

Re: [Asterisk-Users] Load Balancing b/w 2 asterisk servers using SIPload balancer

[Matthew Boehm]

How do you plan on supporting call queues, parking and agents with 2 *
servers? This is something that has blocked us from being able to do our own
SER-based load balancing.

-Matthew

Jagan Mohan wrote:
 Hi,

   I'm trying to do load balancing between 2 asterisk servers using SIP
 load balancer, provided by http://www.vovida.org

   I used the following options on lbproxy, but I get the below message
 continuously.

 ./lbProxy -name seneca -reqPort 5060 -respPort 5061 -proxy A1 -proxy
 A2

 No proxies are up - can not send message to anyone

 Xlite is not able to register to the asterisk server.

 Is there anything which needs to be tweaked on Asterisk side to get
 this working? Please help.

 Thanks,
 Jagan
 ___
 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

___
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

[Asterisk-Users] Am i right by Asterisk?

[Stefan Stolz]

Hello,
i tryed to read the Wiki, but i am not sure if i am right with Asterisk.
Until now i made my phone calls with ant-phone over my ISDN Fritz Card. Now i 
tryed to search a way to phone from other computers in the internal net over 
the Fritz Card on the Server. Someone told me Asterisk can do this.
I read in the Wiki that Asterisk is in special for Voip, but it looked like 
that it can also make ISDN calls.
Can Asterisk do this? What do i need to phone with Asterisk over ISDN into the 
phone net? Or where can i read about this things? I think all i need stands 
in the Wiki, but it was to much for me to find the right thing out for me...
I think i need ISDN4Linux, because ant-phone used this and it worked. I read 
that i need a special Plugin for this because Asterisk per default cant do 
this?
Can you help me to get order in my confusion? ;-)
Thank you very much!
-- 
Grüsse
Stolzi


pgpoTWj5NdQEU.pgp
Description: PGP signature
___
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

[Asterisk-Users] Multiple IAX Phones Behind NAT

[Will Fletcher]

Hi folks,

Ok, I've seen this question go unanswered on the mailing list, and I
assume it's because no one had the heart to break the bad news to the
guy asking, but be honest with me, I can take it. At this time it's
flat impossible
to have multiple IAX phones behind a NAT without using an * gateway
because there's no way to have a client listen on a port besides 4569.
Is my only option to learn about SIP and attempt to forward that
through my NAT?

Thanks,
Will Fletcher
-- 
Auburn University
Department of Computer Science
107 Dunstan Hall
Auburn, AL 36849
334-332-9544
[EMAIL PROTECTED]



___
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

Re: [Asterisk-Users] Sipura-841 Problems

[Wolfgang S. Rupprecht]

[EMAIL PROTECTED] (LES.NET 1996 INC.) writes:
 Yes, I upgraded some prior to the problem.  it seems to affect both
 versions of the firmware.

 But you cannot upgrade them after they lock up.

I don't know if this is related, but I couldn't get my sipura spa-841
working using any of the half-dozen store-bought cat-5 patch cables I
had laying around.  It just refused to register.  Tcpdump confirmed
that packets were coming from it, and we answered, but it never
heard us.  Just out of randomness I tried the shorter enclosed cable
that came with the spa-841 and would you believe that it started
working?

As far as I can tell, the rj-45 socket on the phone is just a bit
non-standard and the wires just don't make reliable contact to the
spades on the cable.  It isn't a case of some of the wires in the
socket being bent, they are all straight and look normal.  All I can
think is that the contact wires have a slightly higher than normal
angle and end up hitting the plastic lip of the rj-45 plug instead of
resting on the gold spade contacts.

-wolfgang
-- 
Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/
___
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

[Asterisk-Users] memory consumption

[Matias G.]

Hi I'm using Asterisk CVS-HEAD-03/09/05-18:25:28, Debian 3.0 rc3 and a
Pentium IV 2.4 Ghz 512 Mb.

When I boot my computer, top reads:
Mem:515824K total,33852K used,   481972K free, 1292K buffers
Swap:   979924K total,0K used,   979924K free,17052K cached

after two days running I have only 9000K free (less than 9 Mb) physical
memory available... the only way I have found to recover the lost memory is
to reboot the computer...

any help will be greatly appreciated.

bye,
Matias
___
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

RE: [Asterisk-Users] Application SetVarCDR

[William M. Sandiford]

Makes senseespecially since I used _X. to jump to s...(duh...slaps self in 
forehead)...do you get the correct dst field in your CDR's?



-Original Message-
From: Matthew Boehm [mailto:[EMAIL PROTECTED] 
Sent: Friday, March 11, 2005 9:43 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Application SetVarCDR


I know this isn't the best way but I stopped using 's' and I use _X. on 
everything now. It is really stupid to say That person dialed the number 's' 
or That phone call's final destination was 's'. That doesn't help anything; 
debugging nor billing.

My $0.02
Matthew

William M. Sandiford wrote:
 Hello:

 I found a reference to the application SetVarCDR in the following post 
 but I don't seem to have this available to me in my version of *.

 HYPERLINK

http://lists.digium.com/pipermail/asterisk-cvs/2005-February/005337.htmlht
tp://lists.digium.com/pipermail/asterisk-cvs/2005-February/005337.html

 My version of * is CVS-HEAD-03/10/05-18:42:35

 I would like to change the value of the src and dst variables in the 
 CDR as I sometimes find that they don't have entirely accurate 
 information.  For example my dst field quite often has a value of s 
 because I do my call processing in the s extension.  This is no good 
 to me.

 Does anyone know How I can get access to the functions mentioned 
 above.  FYI I have tried doing a set var just before I dial like
 this:

 exten = s,13,SetVar(CDR(dst)=12345)
 exten = s,14,Dial(HYPERLINK
 mailto:SIP/[EMAIL PROTECTED]SIP/[EMAIL PROTECTED])

 but that doesn't seem to work...I still get s in the dst field of my 
 CDR

 Regards,
 Bill



 ___
 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

___
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

-- 
No virus found in this incoming message.
Checked by AVG Anti-Virus.
Version: 7.0.308 / Virus Database: 266.6.4 - Release Date: 3/7/2005
 

-- 
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.308 / Virus Database: 266.6.4 - Release Date: 3/7/2005
 
___
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

[Asterisk-Users] EADS6550 and asterisk - echo on PSTN call

[administrator tootai]

Hi list,
would like to know if some of you have tested asterisk connected to an 
EADS 6550 analogique PBX (also know as Nexpan50).

Our set up is a Dell Optiplex with 1 TDM400 4 FXO, 1 TDM400 4 FXS, (no 
other card, each of them have their own IRQ) all ports connected to the 
EADS. We have GS ATA286 as EP. Calling from ATA to ATA give a perfect 
sound. Calling to PSTN numbers or reverse side, give echo.

We can do what we want in zconfig.h (STEVE2, MARK2, MMX, 
AGGRESSIV_SUPPRESSOR, NOECHOCAN_DISABLE) or zapata.conf (tx gain=-10.0, 
echocancel=32 ... 256,), test with differents POTS phone, it change 
nothing. We even didn't notice changes between our various changes in 
those files (and yes modules where unloaded between each test). Always 
the same echo.

So know we start to doubt that this echo problem is asterisk related but 
perhaps more to the PBX. That's why we ask if some of you have/had 
similar setup with this PBX and if there is a solution.

Thanks for any hint.
--
Daniel
___
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

[Asterisk-Users] CAPI- 2 Cards

[adria vidal]

Some suggestion about how detect busy channels in a installation with 2 cards (AVM Fritz)? 
Can't find info about groups in capi channels.  Need to dial out trought some of the 4 avalaible channels.
Better try it with zaphfc ?

Adrià Vidal 
___
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

Re: [Asterisk-Users] Am i right by Asterisk?

[Bruno Hertz]

On Fri, 2005-03-11 at 15:32 +0100, Stefan Stolz wrote:
 Hello,
 i tryed to read the Wiki, but i am not sure if i am right with Asterisk.
 Until now i made my phone calls with ant-phone over my ISDN Fritz Card. Now i 
 tryed to search a way to phone from other computers in the internal net over 
 the Fritz Card on the Server. Someone told me Asterisk can do this.
 I read in the Wiki that Asterisk is in special for Voip, but it looked like 
 that it can also make ISDN calls.
 Can Asterisk do this? What do i need to phone with Asterisk over ISDN into 
 the 
 phone net? Or where can i read about this things? I think all i need stands 
 in the Wiki, but it was to much for me to find the right thing out for me...
 I think i need ISDN4Linux, because ant-phone used this and it worked. I read 
 that i need a special Plugin for this because Asterisk per default cant do 
 this?
 Can you help me to get order in my confusion? ;-)
 Thank you very much!

Stefan

to clarify what you want to achieve: you have an Fritz ISDN card and
what to issue calls from several computers on your LAN to the ISDN line,
right?

If that's your question, the answer is yes, asterisk can do this, and
I have exactly that setup

  LAN

|  |
 Host1 -   |- NAT- Internet
 Host2 - Asterisk Server 
 Host3 -   |- Fritz  - ISDN

For the communication between your computers and asterisk, you'll use
some VoIP protocol, like SIP, IAX or H323 and a corresponding client
(SJPhone, Iaxcomm, Gnomemeeting).

Regarding asterisk interfacing the Fritz card you might either use
chan_modem and isdn4linux, which I didn't test myself but it seems
it's not very recommended, or chan_capi and the AVM capi drivers,
which I have running myself and work OK. Another alternative is mISDN.

Finally, you'll need to setup a proper asterisk dial plan to link all
that together. It's not trivial at the beginning, but doing some
reading especially on the Wiki and in mailing list archives will help
you a lot, so it's not too hard either.

Good luck
Bruno



___
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

[Asterisk-Users] TDM04B lock up

[Goutam Shaw]

Hi
I have a strange situation. Once in a while (non-deterministic) the 2 TDM04B
cards lock up at the same time and stop processing incoming and outgoing
calls even though * shows that it is trying to communicate to ZAP channels
(at least on the outgoing). The only cure is to reboot the system when it
happens. It makes me very apprehensive of the system

Has anyone seen this problem. Could this be something to do with the IRQ
sharing. Here is the output of lspci -v.

I see that one of the cards shares IRQ # with VGA controller and the other
one with ICH4 IDE.

Any help would be appreciated.


00:00.0 Host bridge: Intel Corp. 82845G/GL [Brookdale-G] Chipset Host Bridge
(rev 01)
Subsystem: Dell Computer Corporation: Unknown device 0160
Flags: bus master, fast devsel, latency 0
Memory at f000 (32-bit, prefetchable) [size=128M]
Capabilities: [e4] #09 [1105]

00:02.0 VGA compatible controller: Intel Corp. 82845G/GL [Brookdale-G]
Chipset Integrated Graphics Device (rev 01) (prog-if 00 [VGA])
Subsystem: Dell Computer Corporation: Unknown device 0160
Flags: fast devsel, IRQ 11
Memory at e800 (32-bit, prefetchable) [size=128M]
Memory at feb8 (32-bit, non-prefetchable) [size=512K]
Capabilities: [d0] Power Management version 1

00:1e.0 PCI bridge: Intel Corp. 82801BA/CA/DB PCI Bridge (rev 81) (prog-if
00 [Normal decode])
Flags: bus master, fast devsel, latency 0
Bus: primary=00, secondary=01, subordinate=01, sec-latency=32
I/O behind bridge: d000-dfff
Memory behind bridge: fe90-feaf

00:1f.0 ISA bridge: Intel Corp. 82801DB ISA Bridge (LPC) (rev 01)
Flags: bus master, medium devsel, latency 0

00:1f.1 IDE interface: Intel Corp. 82801DB ICH4 IDE (rev 01) (prog-if 8a
[Master SecP PriP])
Subsystem: Dell Computer Corporation: Unknown device 0160
Flags: bus master, medium devsel, latency 0, IRQ 9
I/O ports at ignored
I/O ports at ignored
I/O ports at ignored
I/O ports at ignored
I/O ports at ffa0 [size=16]
Memory at feb7fc00 (32-bit, non-prefetchable) [size=1K]

00:1f.3 SMBus: Intel Corp. 82801DB SMBus (rev 01)
Subsystem: Dell Computer Corporation: Unknown device 0160
Flags: medium devsel, IRQ 3
I/O ports at efe0 [size=32]

01:04.0 Communication controller: Tiger Jet Network Inc. Intel 537
Subsystem: Unknown device b100:0003
Flags: bus master, medium devsel, latency 64, IRQ 11
I/O ports at dc00 [size=256]
Memory at fe9fc000 (32-bit, non-prefetchable) [size=4K]
Capabilities: [40] Power Management version 2

01:05.0 Ethernet controller: Realtek Semiconductor Co., Ltd.
RTL-8139/8139C/8139C+ (rev 10)
Subsystem: Realtek Semiconductor Co., Ltd. RT8139
Flags: bus master, medium devsel, latency 64, IRQ 3
I/O ports at dd00 [size=256]
Memory at fe9fbf00 (32-bit, non-prefetchable) [size=256]
Capabilities: [50] Power Management version 2

01:06.0 Communication controller: Tiger Jet Network Inc. Intel 537
Subsystem: Unknown device b100:0003
Flags: bus master, medium devsel, latency 64, IRQ 9
I/O ports at de00 [size=256]
Memory at fe9fd000 (32-bit, non-prefetchable) [size=4K]
Capabilities: [40] Power Management version 2

Regards
Goutam Shaw



___
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

RE: [Asterisk-Users] What is that area code?

[Jay Milk]

802 US Vermont
803 US South Carolina
804 US Virginia
805 US California
806 US Texas
807 Canada - Ontario
808 US Hawaii

You can look these up using http://voiprates.us/rateengine

US doesn't have cellular area codes, because cellular users are charged
for incoming minutes.  Calls to cellular phones are charged as if they
were landlines.

 -Original Message-
 From: Ronald Wiplinger [mailto:[EMAIL PROTECTED] 
 Sent: Friday, March 11, 2005 7:50 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [Asterisk-Users] What is that area code?
 
 
 Can anybody help me  and explain me the following area codes:
 
 1-340   US-USVI
 1-670   US-CNMI
 1-710   US-Governement
 1-787   US-Poerto Rico
 1-802 ~ 1-808 ???
 1-939   US-Poerto Rico
 1-600   Canada
 
 Are the above codes are USA Continental tarrif (NuFone / 
 Broadvoice ... 
 )
 
 
 What are the codes for mobile phones in USA?
 
 
 bye
 
 Ronald
 
 ___
 Asterisk-Users mailing list
 Asterisk-Users@lists.digium.com 
 http://lists.digium.com/mailman/listinfo/aster isk-users
 To 
 UNSUBSCRIBE or update options visit:

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



___
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

Re: [Asterisk-Users] memory consumption

[Matias G.]

More info, when in top sorted by Mem usage what I get is:

  PID USER PRI  NI  SIZE  RSS SHARE STAT %CPU %MEM   TIME COMMAND
  257 root   9   0  7584 7520  3604 S 0.0  1.4   0:00 asterisk
  258 root  14   0  7584 7520  3604 S 0.0  1.4   0:00 asterisk
  260 root   9   0  7584 7520  3604 S 0.0  1.4   0:00 asterisk
  261 root   9   0  7584 7520  3604 S 0.0  1.4   0:00 asterisk
  262 root   9   0  7584 7520  3604 S 0.0  1.4   0:00 asterisk
  263 root   9   0  7584 7520  3604 S 0.0  1.4   0:00 asterisk
  266 root   9   0  7584 7520  3604 S 0.0  1.4   0:00 asterisk
  267 root   9   0  7584 7520  3604 S 0.0  1.4   0:00 asterisk
  268 root  10   0  7584 7520  3604 S 0.0  1.4   0:00 asterisk
  269 root   9   0  7584 7520  3604 S 0.0  1.4   0:00 asterisk
  270 root  10   0  7584 7520  3604 S 0.0  1.4   0:01 asterisk
  271 root   9   0  7584 7520  3604 S 0.0  1.4   0:00 asterisk
  274 root   9   0  7584 7520  3604 S 0.0  1.4   0:00 asterisk
  275 root   9   0  7584 7520  3604 S 0.0  1.4   0:00 asterisk
  276 root   9   0  7584 7520  3604 S 0.0  1.4   0:00 asterisk
  277 root   9   0  7584 7520  3604 S 0.0  1.4   0:00 asterisk
  552 root  17   0  7584 7520  3604 R 0.5  1.4   0:00 asterisk
  353 root   9   0  1960 1960  1684 R 0.0  0.3   0:00 sshd

followed by some other processes...

thanks.
Matías

- Original Message - 
From: Matias G. [EMAIL PROTECTED]
To: UsersList * asterisk-users@lists.digium.com
Sent: Friday, March 11, 2005 12:22 PM
Subject: [Asterisk-Users] memory consumption


 Hi I'm using Asterisk CVS-HEAD-03/09/05-18:25:28, Debian 3.0 rc3 and a
 Pentium IV 2.4 Ghz 512 Mb.

 When I boot my computer, top reads:
 Mem:515824K total,33852K used,   481972K free, 1292K buffers
 Swap:   979924K total,0K used,   979924K free,17052K cached

 after two days running I have only 9000K free (less than 9 Mb) physical
 memory available... the only way I have found to recover the lost memory
is
 to reboot the computer...

 any help will be greatly appreciated.

 bye,
 Matias
 ___
 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


___
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

[Asterisk-Users] 1.0.6 music on hold bug ?!

[Calin Serbanescu]

hello list,

last night i upgraded my asterisk box from 1.0.5 to 1.0.6 and my music
on hold did not work anymore.

my setup is ISDN (wct1xxp)-SIP (Audiocodes mp124) and reverse. 
the system refuses to activate music on hold resource... i returned to
1.0.5 and it works fine again...

i'm i missing something? i can really make use of 1.0.6 bug-fixes and
i'm sorry i can't use it :((

Thanks.
Calin.

___
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

[Asterisk-Users] Phone suggestions

[James Murray]

  Can anyone offer any suggestions for quality hardware sip phones 
under $150. Preferable one with a 2 line caller id screen and the 
ability to disable call waiting.  It would also be very useful if it had 
a good voice echo cancellation built into the phone.

___
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

Re: [Asterisk-Users] TDM04B lock up

[Dennis Webb]

I had an issue with the same setup except only channel 1 on each card would work for incoming. All would work for outgoing, but asterisk never saw the other channels ringing. Restarting asterisk didn't help either. I panicked and just rebooted and the problem went away. I wish I had taken the time to have tried unloading and reloading the drivers. The system had been up for 22 days when this happened so I now just restart every sunday morning to be safe.

On Fri, 2005-03-11 at 09:36, Goutam Shaw wrote:

Hi
I have a strange situation. Once in a while (non-deterministic) the 2 TDM04B
cards lock up at the same time and stop processing incoming and outgoing
calls even though * shows that it is trying to communicate to ZAP channels
(at least on the outgoing). The only cure is to reboot the system when it
happens. It makes me very apprehensive of the system

Has anyone seen this problem. Could this be something to do with the IRQ
sharing. Here is the output of lspci -v.

I see that one of the cards shares IRQ # with VGA controller and the other
one with ICH4 IDE.

Any help would be appreciated.


00:00.0 Host bridge: Intel Corp. 82845G/GL [Brookdale-G] Chipset Host Bridge
(rev 01)
Subsystem: Dell Computer Corporation: Unknown device 0160
Flags: bus master, fast devsel, latency 0
Memory at f000 (32-bit, prefetchable) [size=128M]
Capabilities: [e4] #09 [1105]

00:02.0 VGA compatible controller: Intel Corp. 82845G/GL [Brookdale-G]
Chipset Integrated Graphics Device (rev 01) (prog-if 00 [VGA])
Subsystem: Dell Computer Corporation: Unknown device 0160
Flags: fast devsel, IRQ 11
Memory at e800 (32-bit, prefetchable) [size=128M]
Memory at feb8 (32-bit, non-prefetchable) [size=512K]
Capabilities: [d0] Power Management version 1

00:1e.0 PCI bridge: Intel Corp. 82801BA/CA/DB PCI Bridge (rev 81) (prog-if
00 [Normal decode])
Flags: bus master, fast devsel, latency 0
Bus: primary=00, secondary=01, subordinate=01, sec-latency=32
I/O behind bridge: d000-dfff
Memory behind bridge: fe90-feaf

00:1f.0 ISA bridge: Intel Corp. 82801DB ISA Bridge (LPC) (rev 01)
Flags: bus master, medium devsel, latency 0

00:1f.1 IDE interface: Intel Corp. 82801DB ICH4 IDE (rev 01) (prog-if 8a
[Master SecP PriP])
Subsystem: Dell Computer Corporation: Unknown device 0160
Flags: bus master, medium devsel, latency 0, IRQ 9
I/O ports at ignored
I/O ports at ignored
I/O ports at ignored
I/O ports at ignored
I/O ports at ffa0 [size=16]
Memory at feb7fc00 (32-bit, non-prefetchable) [size=1K]

00:1f.3 SMBus: Intel Corp. 82801DB SMBus (rev 01)
Subsystem: Dell Computer Corporation: Unknown device 0160
Flags: medium devsel, IRQ 3
I/O ports at efe0 [size=32]

01:04.0 Communication controller: Tiger Jet Network Inc. Intel 537
Subsystem: Unknown device b100:0003
Flags: bus master, medium devsel, latency 64, IRQ 11
I/O ports at dc00 [size=256]
Memory at fe9fc000 (32-bit, non-prefetchable) [size=4K]
Capabilities: [40] Power Management version 2

01:05.0 Ethernet controller: Realtek Semiconductor Co., Ltd.
RTL-8139/8139C/8139C+ (rev 10)
Subsystem: Realtek Semiconductor Co., Ltd. RT8139
Flags: bus master, medium devsel, latency 64, IRQ 3
I/O ports at dd00 [size=256]
Memory at fe9fbf00 (32-bit, non-prefetchable) [size=256]
Capabilities: [50] Power Management version 2

01:06.0 Communication controller: Tiger Jet Network Inc. Intel 537
Subsystem: Unknown device b100:0003
Flags: bus master, medium devsel, latency 64, IRQ 9
I/O ports at de00 [size=256]
Memory at fe9fd000 (32-bit, non-prefetchable) [size=4K]
Capabilities: [40] Power Management version 2

Regards
Goutam Shaw



___
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



___
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

Re: [Asterisk-Users] Phone suggestions

[Dennis Webb]

Polycom SIP300 works good with all the features except the echo cancellation. It says in the manual it has an echo can, but other sources say otherwise.

Not to advertise, but voipsupply.com lists their sip phones by price and might make your search a little easier.

I am not affiliated with the above site, but just used it for reference. Sorry if I am breaking the rules.

On Fri, 2005-03-11 at 09:58, James Murray wrote:

   Can anyone offer any suggestions for quality hardware sip phones 
under $150. Preferable one with a 2 line caller id screen and the 
ability to disable call waiting.  It would also be very useful if it had 
a good voice echo cancellation built into the phone.


___
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



___
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

RE: [Asterisk-Users] Asterisk and USB ISDN controllers ...

[Steven Critchfield]

On Fri, 2005-03-11 at 11:31 +0100, Vledder, Hans wrote:
 Hi Steve,
 
 Since you don't mention what USB ISDN adapter specifically you are
 thinking about, what do you think we will be able to tell you.
 
 All I know about the adapter is what I've told you. It's a USB Colognechip
 based ISDN controller - probably HCF-USB based. It's supported by Linux, but
 there's no info on access to B and D channels.

Okay, this shows where you should do some research. Just to point out
how easy it is to find out the answer, watch the steps.

1. use google to look up cologne isdn usb linux
2. follow link that points to the actual manufacturer
3. Notice that the usb driver is the hisax driver
4. use google to look up hisax site:lists.digium.com
5. notice how many people are already discussing the use of it.
6. follow a couple of posts.
7. conclude that it is possible to use.

optional steps
8. think about how little time and effort it took to follow the above
pattern to quickly answer questions on your own.

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] Behalf Of Steven
 Critchfield
 Sent: Thursday, March 10, 2005 6:42 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Asterisk and USB ISDN controllers ...
 
 
 On Thu, 2005-03-10 at 18:13 +0100, Vledder, Hans wrote:
  Guys,
  
  I am planning on building a small SIP PBX with a single ISDN line.
 Currently
  I am looking into the specs of a very tiny barebone system that has an
  option Colognechip base ISDN controller. The only thing is that the ISDN
  module that comes with this barebone hooks up to the motherboard using
 USB.
  My intention is to allow incoming and outgoing calls from SIP to ISDN. Is
  this setup in any way supported by *?
 
 Since you don't mention what USB ISDN adapter specifically you are
 thinking about, what do you think we will be able to tell you.
 
 The first step would really be to ask if your specific ISDN adapter can
 be used under linux. After that, can that specific ISDN adapter give
 access to voice channels. What method is used to get access to the audio
 and the signaling.
 
 It may well be usable if the drivers for it implements the same API as
 the current ISDN cards in use support.

-- 
Steven Critchfield [EMAIL PROTECTED]

___
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

Re: [Asterisk-Users] Application SetVarCDR

[Matthew Boehm]

So far I am. My only gripe is that when I use a Goto statement, for example:

exten = 888747,1,Goto(internal-phones,3044,1)

it shows dst as 3044 which is programatically correct, but again not usefull
in billing.

-Matthew

William M. Sandiford wrote:
 Makes senseespecially since I used _X. to jump to
 s...(duh...slaps self in forehead)...do you get the correct dst field
 in your CDR's?



 -Original Message-
 From: Matthew Boehm [mailto:[EMAIL PROTECTED]
 Sent: Friday, March 11, 2005 9:43 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] Application SetVarCDR


 I know this isn't the best way but I stopped using 's' and I use _X.
 on everything now. It is really stupid to say That person dialed the
 number 's' or That phone call's final destination was 's'. That
 doesn't help anything; debugging nor billing.

 My $0.02
 Matthew

 William M. Sandiford wrote:
 Hello:

 I found a reference to the application SetVarCDR in the following
 post but I don't seem to have this available to me in my version of
 *.

 HYPERLINK


http://lists.digium.com/pipermail/asterisk-cvs/2005-February/005337.htmlht
 tp://lists.digium.com/pipermail/asterisk-cvs/2005-February/005337.html

 My version of * is CVS-HEAD-03/10/05-18:42:35

 I would like to change the value of the src and dst variables in the
 CDR as I sometimes find that they don't have entirely accurate
 information.  For example my dst field quite often has a value of s
 because I do my call processing in the s extension.  This is no good
 to me.

 Does anyone know How I can get access to the functions mentioned
 above.  FYI I have tried doing a set var just before I dial like
 this:

 exten = s,13,SetVar(CDR(dst)=12345)
 exten = s,14,Dial(HYPERLINK
 mailto:SIP/[EMAIL PROTECTED]SIP/[EMAIL PROTECTED])

 but that doesn't seem to work...I still get s in the dst field of my
 CDR

 Regards,
 Bill



 ___
 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

 ___
 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

 --
 No virus found in this incoming message.
 Checked by AVG Anti-Virus.
 Version: 7.0.308 / Virus Database: 266.6.4 - Release Date: 3/7/2005

___
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

RE: [Asterisk-Users] how do i get rid of this blasted echo !!!

[Brett, Gary]

Dennis, Thanks, I am currently using
kernel 2.4, are you saying there are fixes for this sort of thing in kernel 2.6.9 



-Original
Message-
From: Dennis Webb
[mailto:[EMAIL PROTECTED] 
Sent: 11 March 2005 13:57
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: RE: [Asterisk-Users] how
do i get rid of this blasted echo !!!



I've added MMX and it
didn't help. I also did the CFLAGS+=-march=pentium4 with no help there
either. The more I search, the more I found and I'm down to disabling any
hardware not used in the box such as USB and recompiling the kernel with a fresh
copy from kernel.org. It seems there were a lot of problems solved when
2.6.9 came out.

If I ever get mine fixed, I will try to post everything I did.

On Fri, 2005-03-11 at 07:50, Brett, Gary wrote: 

So is it accepted as standard that compiling with MMX will help improve echotype issues ?-Original Message-From: Herman Cremer [mailto:[EMAIL PROTECTED] Sent: 11 March 2005 11:56To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] how do i get rid of this blasted echo !!!Thanks Error.I have switched to IAX looong agomuch better !Just battle when doing double NAT :)I dont have the phones here with me,but lets say its different...is there awayto adjust the channel to fix the err ?-hermanOn Fri, 2005-03-11 at 13:24, [EMAIL PROTECTED] wrote: Hi Herman,  Look at the bottom of your phones and compare the REN values of both. Do they both value of REN 1.0? I think the one with the problem might have an REN value other than one. You tell me!  Errol Samuels Don't let SIP Drive you crazy, use IAX2 On the echo...   I have 2 extensions, with different analog phones.  The one works fine, the other echos and scratches  like mad !!   I have switched the ports, cables etc but its ALWAYS  the same phone...   Maybe this could be it ?   Is it ok from a SIP phone ?   Herman cremer   ___  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___ 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___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users




___
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

Re: [Asterisk-Users] OT: Best DB

[Steven Critchfield]

On Fri, 2005-03-11 at 08:51 -0600, Matthew Boehm wrote:
  If you're a VoIP provider, and are trying to
  provide a near carrier-grade service, postgres shines.
 
 I'm not disagreeing with you, but we are a CLEC and we do provide
 'carrier-grade' service and we use MySQL everywhere.
 
 IMHO, MySQL is just so much more easy to use, install and maintain.
 phpMyAdmin makes it even easier.

If that is a deciding reason, you should check out phppgadmin sometime.
Very similar interface but for postgres. 
-- 
Steven Critchfield [EMAIL PROTECTED]

___
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

Re: [Asterisk-Users] CDR database

[Ronald Wiplinger]

Yair Hakak wrote:
http://www.voip-info.org/wiki-Asterisk+billing
 

Thanks! Found it!
Is there a easy way / tool available to import all (privious) Master.cvs 
into the database?

bye
Ronald
___
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

[Asterisk-Users] ASTCC and NuFone billing is different!!

[Ronald Wiplinger]

I have ASTCC installed, and compare it with NuFone, however, I find that 
the billing of NuFone is always a few secondes more (6 to 24 seconds)

Does anybody has an explanation / solution for it?
bye
Ronald
___
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

Re: [Asterisk-Users] AAH 0.06 - IAX Connection Over NAT Firewall

[Doug Millsaps]

Which provider, I have my AAH 0.6 box set up with VoicePulse using
IAX2.
At 12:59 AM 3/11/2005, you wrote:
Hello
all,

I am having trouble getting my IAX based Voip
provider setup. Any pointers are welcome.

So here is the deal. I am registered up
and I can make outgoing calls but incoming calls fail.
Configs all look good I thought.
My PBX is behind our firewall with a direct NAT of one to one for an
external IP. 
IAX port is forwarded UDP and TCP to the internal IP.

* shows good registration and Ips and ports
show solid.

Within my AAH I have the registration like the
provier said to do. I get absolutely nothing on the incoming.
IAX2 debug shows nothing on incoming. Just a fast busy.
Outgoing works perfectly however.

I have a defined DID in the AMP interface and
verified it is written to confs and have reloaded.

Can anyone tell me another way to verify that
something is coming in? Or did I just miss something on the whole
IAX over NAT?

Thanks all,

Wiley

___
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

[Asterisk-Users] PAP2-NA point to poitn calls ??...(Direct IP Dialing)

[Diego S. Cicero]

Hello, I need to know if there is an option in the PAP2-NA Web Configurator
like Enable IP dialing: yes/no

I need to make point to point calls with two PAP2-NA by IP address (The
PAP2-NA are in the same LAN, no Internet access). Is it possible ?

Thank you !!





___
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

Re: [Asterisk-Users] 1.0.6 music on hold bug ?!

[Jason Williams]

On Fri, 11 Mar 2005 17:45:55 +0200, Calin Serbanescu [EMAIL PROTECTED] wrote:
 hello list,
 
 last night i upgraded my asterisk box from 1.0.5 to 1.0.6 and my music
 on hold did not work anymore.
 

Download version 1.0.7 from Cvs this has the fixes in it
___
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

Re: [Asterisk-Users] OT: Active channels bridging with SNOM190

[David Wilson]

Hi Matt,
Thanks for all your help. Things have gone well today.
No bridged Zap channels so far ! Thank you so much for all your help.

Kindest regards
David Wilson
___
D c D a t a
Tel +27 33 342 7003
Fax +27 33 345 4155
Cell +27 82 4147413
http://www.dcdata.co.za
[EMAIL PROTECTED]
Powered by Linux, driven by passion !
___
Computers are not intelligent. They only think they are.
- Original Message - 
From: David Wilson [EMAIL PROTECTED]
To: Matt Kemner [EMAIL PROTECTED]
Cc: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com
Sent: Thursday, March 10, 2005 1:08 PM
Subject: Re: [Asterisk-Users] OT: Active channels bridging with SNOM190


Yea, True. No sweat.
Should be better now ? :-)
Kindest regards
David Wilson
___
D c D a t a
Tel +27 33 342 7003
Fax +27 33 345 4155
Cell +27 82 4147413
http://www.dcdata.co.za
[EMAIL PROTECTED]
Powered by Linux, driven by passion !
___
Computers are not intelligent. They only think they are.
- Original Message - 
From: Matt Kemner [EMAIL PROTECTED]
To: David Wilson [EMAIL PROTECTED]
Sent: Thursday, March 10, 2005 12:57 PM
Subject: Re: [Asterisk-Users] OT: Zap channels intermittently bridging 
withSNOM190


On Thu, 10 Mar 2005, quoth David Wilson:
Sorry about the misleading subject :) I started a couple days ago being 
very
unclear about how things were going wrong and thought it could be 
something
in Asterisk that was causing it.
Yeah I know what you mean.. I specifically didn't contact SNOM about this
bug because I also had this nagging feeling that it could be an asterisk
config problem, and I didn't want to hassle them about it if it was.
I only made the comment about the subject in case someone in the future
comes across this problem and looks in the archives, just so they're not
put off thinking it's a different bug.
- Matt
___
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 
___
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

[Asterisk-Users] Realtime does not work yet, ...

[Ronald Wiplinger]

I try to get Realtime to work, ... the debug looks like below.
Mar 12 00:56:56 DEBUG[25640]: MySQL RealTime: Retrieve SQL: SELECT * 
FROM sip_buddies WHERE name = '621'
Mar 12 00:56:56 DEBUG[25640]: MySQL RealTime: Everything is fine.
Mar 12 00:56:56 DEBUG[25640]: Unable to find key '621' in family 
'SIP/Registry'
Mar 12 00:56:56 DEBUG[25640]: Setting NAT on RTP to 524288
Mar 12 00:56:56 DEBUG[25640]: Exiting with DIALSTATUS=CONGESTION.
Mar 12 00:56:56 DEBUG[25640]: 
/var/spool/asterisk/voicemail/other/621/unavail doesn't exist, doing 
what we can

There are two things:
1. Unable to find key '621' in family 'SIP/Registry'
where have I forgotten to set that?
2. /var/spool/asterisk/voicemail/other/621/unavail doesn't exist, doing 
what we can
it is not there, because it is in /var/spool/asterisk/vm/621/
Where to correct that?

bye
Ronald
___
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

[Asterisk-Users] FC3 Dual Xeon Zaptel PANIC

[Dan Davis]

Hello:

My TE41P causes a PANIC on FC3. Any 
suggestions?

Thanks
Dan

___
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

Re: [Asterisk-Users] Multiple IAX Phones Behind NAT

[Steve Kann]

Will Fletcher wrote:
Hi folks,
Ok, I've seen this question go unanswered on the mailing list, and I 
assume it's because no one had the heart to break the bad news to the 
guy asking, but be honest with me, I can take it.  At this time it's 
flat impossible to have multiple IAX phones behind a NAT without using 
an * gateway because there's no way to have a client listen on a port 
besides 4569.  Is my only option to learn about SIP and attempt to 
forward that through my NAT?

Not true;
You just need the iax phones to register, and then they will work fine 
through the NAT; whatever port they're on on local machines, and 
whatever port they get NATted to won't matter.

-SteveK
___
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

[Asterisk-Users] Is it an AGI bug in 1.06? IAX Calls going to wrong extension with AGI.

[Kanuri, Seshu (Company IT)]

I am using PBXware for configuring users and 
extensions.
Pbxware uses Internal script called init.sh to process 
the calls 
based on its own version of extensions.conf defined in 
the GUI.

I have IAX2 Extensions 56 and 101 and SIP extensions 50 
and 51. 

I have used IAX2 extension 101 and dialed SIP Extension 
51

But the PBXWare's 
Init.shAGI command identifies the DNIS 
as another IAX Extension - extension 56, instead of SIP 
Extension 51 
and sends the call there.

I tried the same with Extension 50 and the result is 
the same? 
is this an AGI Bug or a bug in the GUI Software. 


Has anyone tried this before and had such 
problem?


VAR: 
agi_request: init.sh 
;( 
Init.sh is sent from PBXware)VAR: agi_channel: IAX2/[EMAIL PROTECTED]/2 VAR: agi_language: en 
VAR: agi_type: IAX2 VAR: agi_uniqueid: 
asterisk-28947-1110463619.0 VAR: agi_callerid: Seshu 
Kanuri 101 VAR: agi_dnid: 
56 
; Actual number dialed was 51VAR: agi_rdnis: 
unknown VAR: agi_context: default 
VAR: agi_extension: 56 VAR: agi_priority: 
1 VAR: agi_enhanced: 0.0 
VAR: agi_accountcode:  Detected 
protocol 'iax2' ... 200 result=1  Detected caller 
'101' ... 200 result=1  Set limit - 24 200 
result=1  Limit not exceeded (1  24) for 
localextensions 200 result=1  Set limit - 2 
200 result=1  Limit not exceeded (1  2) for 
101_out 200 result=1  Detecting destination for '56' 
... 200 result=1  Found Destination localextensions 
(range 56 - 56) 200 result=1  Setting destination 
'localextensions' ... 200 result=1  This is local 
extension, skipping Time Based Dialing/miniLCR ... 200 result=1 
 Set limit - 24 200 result=1  Limit 
not exceeded (2  24) for localextensions 200 result=1 
 Detecting Vertical Services ... 200 result=1 
 Set limit - 2 200 result=1  Limit 
not exceeded (1  2) for 56_in 200 result=1  
Checking for channel IAX2/56/56 ... 200 result=1 APP: exec 
ChanIsAvail IAX2/56/56 200 result=-1  Channel is not 
available ... 200 result=1  Dialing Voicemail 56 
... 200 result=1 APP: exec Voicemail u56 200 
result=-1 APP: answer 200 result=0  
Playing macro 'vm-goodbye' ... 200 result=1 APP: stream 
file vm-goodbye 200 result=-1 endpos=6880

Any clues or 
pointers?

Seshu




From: 
[EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Dennis 
WebbSent: Thursday, March 10, 2005 4:32 PMTo: Asterisk 
Users Mailing List - Non-Commercial DiscussionSubject: Re: 
[Asterisk-Users] Polycom phones do not talk to each other andcannot answer when 
we pickup
Never used pbxware, but the context the sip phones dial out using 
specified in sip.conf needs to include the dialplan context of the phones in 
extensions.conf.On Thu, 2005-03-10 at 15:08, Kanuri, Seshu (Company IT) 
wrote: 
We have bought PBXware GUI from Bicom systems and configured extensions
with Polycom Phones as UAs.

The Polycom Phones can dial out and make calls but I cannot make
extension to extension calling.

Googling did not help much.

As you may be aware PBXware is a closed source software GUI from Bicom
Systems for configuring extensions. It is a good tool to configure and
manage users and phones but it does not allow to do any of the
customization tasks that are possible by directly editing the .conf
files, which may be required in for Polycom.

However if this is an issue of configuration on the Phone itself, we
want to be able to make changes and fix this problem.

Any tips?

Seshu 

 
NOTICE: If received in error, please destroy and notify sender.  Sender does not waive confidentiality or privilege, and use is prohibited. 
 
___
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




NOTICE: If received in error, please destroy and notify sender.  Sender does not waive confidentiality or privilege, and use is prohibited.

___
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

Re: [Asterisk-Users] PAP2-NA point to poitn calls ??...(Direct IP Dialing)

[Jer]

At 11:40 AM 3/11/2005, you wrote:
the only way I found to do this
was have them register with a * server and have * connect them

Hello, I need to know if there is an option in the PAP2-NA Web Configurator
like Enable IP dialing: yes/no
I need to make point to point calls with two PAP2-NA by IP address (The
PAP2-NA are in the same LAN, no Internet access). Is it possible ?
Thank you !!


___
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
___
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

[Asterisk-Users] VoIPJet and g.711

[Wojciech Tryc]

I am experiencing problems connecting to VoIPjet with g.711. It works with 
g.729 and ilbc. It used to work...
Anyone?
Regards,
Wojtek 

___
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

[Asterisk-Users] Vonage a provider?

[Frank Abernathy]

I am new to the mailing list, but I am very interested in
running my small home business office phone system using Asterisk. However,
Broadvoice, a VoIP provider of choice based on my research, is not available in
my area.



I currently use Vonage VoIP. Their website mentions nothing
about being able to link to Asterisk. I was wondering if any US subscribers have
been able to configure Vonage with Asterisk. Or if anyone has found Vonage to
be a non-compatible provider.



TIA!



Frank

[EMAIL PROTECTED] 














--
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.308 / Virus Database: 266.7.1 - Release Date: 3/9/2005
 
___
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

RE: [Asterisk-Users] AAH 0.06 - IAX Connection Over NAT Firewall

[Wiley Siler]

Title: [Asterisk-Users] SIP to H.323 no audio



Yep. Of course, problem is the provider gave the 
settings and the deny statement was part of it. Ooops to them i 
guess.

Thanks,
Wiley



From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of C. 
TomlinsonSent: Friday, March 11, 2005 4:32 AMTo: 'Asterisk 
Users Mailing List - Non-Commercial Discussion'Subject: RE: 
[Asterisk-Users] AAH 0.06 - IAX Connection Over NAT 
Firewall


As I understand it if 
you use that deny statement, all calls will be disallowed, hence why you 
couldnt get any incoming calls.
If you add an allow 
line with the VOIP providers IP that it send the call from, you can then use 
that line to disallow everything else.

It is just a security 
feature really.

C





From: 
[EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Wiley SilerSent: 11 March 2005 07:23To: Asterisk Users Mailing List - Non-Commercial 
Discussion; Asterisk Users Mailing List - Non-Commercial 
Discussion; [EMAIL PROTECTED] Digium. ComSubject: RE: [Asterisk-Users] AAH 0.06 - 
IAX Connection Over NAT Firewall



OK. I removed 
the deny statement they have me using and now I can get incoming 
calls.



Do I need the deny 0.0.0.0/0.0.0.0 
statement?



Thanks,

Wiley





From: 
[EMAIL PROTECTED] on behalf of Wiley SilerSent: Thu 3/10/2005 11:59 PMTo: Asterisk Users Mailing List - Non-Commercial 
Discussion; [EMAIL PROTECTED] Digium. ComSubject: [Asterisk-Users] AAH 0.06 - IAX 
Connection Over NAT Firewall

Hello 
all,
I 
am having trouble getting my IAX based Voip provider setup. Any pointers 
are welcome.
So 
here is the deal. I am registered up and I can make outgoing calls but 
incoming calls fail.Configs all look good I thought.My PBX is behind 
our firewall with a direct NAT of one to one for an external IP.IAX 
port is forwarded UDP and TCP to the internal IP.
* 
shows good registration and Ips and ports show 
solid.
Within my AAH I have the 
registration like the provier said to do. I get absolutely nothing on the 
incoming. IAX2 debug shows nothing on incoming. Just a fast 
busy. Outgoing works perfectly however.
I 
have a defined DID in the AMP interface and verified it is written to confs and 
have reloaded.
Can 
anyone tell me another way to verify that something is coming in? Or did I 
just miss something on the whole IAX over NAT?

Thanks 
all,
Wiley



___
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

Re: [Asterisk-Users] Transfering calls or using any feature

[Martin Renschler]

Try taking out the capital T, having both causes problems in some 
configurations.
Note that there are no defaults for features, you need to uncomment the 
entries in features.conf to activate the them.
/M

Anton Krall wrote:
Guys, this is puzzling.
Seems I cant use any of the feautes (call transfer, record call, etc)
defined in features.conf when a call comes in thru zap and I answer it on
hardphones... Although I CAN use them when Im the one that originates the
call, when received I just cant. 

My dialplan includes wtWT on all Dial cmds just to be sure but it doesn't
seem to be working.
Any pointers?
___
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
 

___
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

[Asterisk-Users] diffrent area codes for diffrent phones in dialplan

[Jer]

I have 3 sets of SIP phones all in diff area codes that need to access the PSTN
I need to it so that a 7 digit number is converted to a 10 digit with the 
correct ara code

eg a call coming from sip-phone1 needs aera code AAA and a call coming fom 
sip-phone2 needs BBB
how can this be setup in the dialplan
is there someway to set a var on a per sip group basis?
I thought of the accountcode...since i will not be using it for CDR

thoughts
Thanks
___
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

Re: [Asterisk-Users] Realtime does not work yet, ...

[Ronald Wiplinger]

Matthew Boehm wrote:
Just because it doesn't work for you doesn't mean it doesn't work for
everyone else.
 

If I would believe that, I would not bother with it anymore ;-)
1. Do you have a record in your database for user 621?
 

I put the record into the database and I can see the record with phpMyAdmin.
2. Run the query inside MySQL cli. How many rows where returned? If none,
then its your fault it failed.
 

How do I do that inside of CLI?
3. You have set the VM context for 621 to be other but it seems that 'you'
created the directory (in the wrong place i might add) as opposed to letting
VM create it for you.
 

I used the script in ../contrib/scripts/addmailbox  and it works fine, 
just in debug it mentions it is in the wrong directory.

bye
Ronald
-Matthew
Ronald Wiplinger wrote:
 

I try to get Realtime to work, ... the debug looks like below.
Mar 12 00:56:56 DEBUG[25640]: MySQL RealTime: Retrieve SQL: SELECT *
FROM sip_buddies WHERE name = '621'
Mar 12 00:56:56 DEBUG[25640]: MySQL RealTime: Everything is fine.
Mar 12 00:56:56 DEBUG[25640]: Unable to find key '621' in family
'SIP/Registry'
Mar 12 00:56:56 DEBUG[25640]: Setting NAT on RTP to 524288
Mar 12 00:56:56 DEBUG[25640]: Exiting with DIALSTATUS=CONGESTION.
Mar 12 00:56:56 DEBUG[25640]:
/var/spool/asterisk/voicemail/other/621/unavail doesn't exist, doing
what we can
There are two things:
1. Unable to find key '621' in family 'SIP/Registry'
where have I forgotten to set that?
2. /var/spool/asterisk/voicemail/other/621/unavail doesn't exist,
doing what we can
it is not there, because it is in /var/spool/asterisk/vm/621/
Where to correct that?
bye
Ronald
___
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
   

___
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

 


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

___
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

Re: [Asterisk-Users] Is it an AGI bug in 1.06? IAX Calls going to wrong extension with AGI.

[Steven Critchfield]

On Fri, 2005-03-11 at 12:26 -0500, Kanuri, Seshu (Company IT) wrote:
 I am using PBXware for configuring users and extensions.
 Pbxware uses Internal script called init.sh to process the calls 
 based on its own version of extensions.conf defined in the GUI.
  
 I have IAX2 Extensions 56 and 101 and SIP extensions 50 and 51. 
  
 I have used IAX2 extension 101 and dialed SIP Extension 51
  
 But the PBXWare's Init.sh  AGI command identifies the DNIS 
 as another IAX Extension - extension 56, instead of SIP Extension 51 
 and sends the call there.

Just a quick thought here, as the vast majority doesn't have access or
at the minimal don't use the software you are using to do config and as
it is an agi script outside of asterisk, you should go to the vendor of
PBXWare and see what they say.
-- 
Steven Critchfield [EMAIL PROTECTED]

___
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

Re: [Asterisk-Users] PAP2-NA point to poitn calls ??...(Direct IP Dialing)

[Luki]

Didn't try it, but quick Google search for sipura IP dialing gives:
http://www.sipura.com/Documents/faq/Section_2.html

3:   How do I call by IP address? 
A: This example illustrate calling via IP address from Line1 to Line2,
but can be generalized from one SPA to another SPA
- Go to line 1, assign UserID to be 1001. Go to line 2, assign UserID
to be 1002
- Set Enable IP dialing to yes 
- Set Make Call w/o Reg and Ans Call w/o Reg to yes 
- Assuming you're calling from line 1 to line 2, you'd press  
1002*IP_ADDRESS*5061# 
Similarly, if calling another SPA on it's line1, press 
uid_remote*ip_addr_remote*5060# 

--Luki
___
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

Re: [Asterisk-Users] Multiple IAX Phones Behind NAT

[Harold Fletcher]

Steve:

But how will that work for incoming calls?  Assume that three phones
have registered with an Asterisk box from inside a NAT, then * knows
that these three users can be found at x.x.x.x port 4569.  When *
receives an incoming call intended for one of those users, all it can do
is forward those calls to x.x.x.x:4569, right?  In that case I don't
understand how the NAT can know to which user the incoming call is
referring.

Will Fletcher



Auburn University
Department of Computer Science
107 Dunstan Hall
Auburn, AL 36849
334-332-9544
[EMAIL PROTECTED]
 [EMAIL PROTECTED] 03/11/05 11:22 AM 
Will Fletcher wrote:

 Hi folks,

 Ok, I've seen this question go unanswered on the mailing list, and I 
 assume it's because no one had the heart to break the bad news to the 
 guy asking, but be honest with me, I can take it.  At this time it's 
 flat impossible to have multiple IAX phones behind a NAT without using

 an * gateway because there's no way to have a client listen on a port 
 besides 4569.  Is my only option to learn about SIP and attempt to 
 forward that through my NAT?


Not true;

You just need the iax phones to register, and then they will work fine 
through the NAT; whatever port they're on on local machines, and 
whatever port they get NATted to won't matter.

-SteveK

___
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

___
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

Re: [Asterisk-Users] Broadvoice Multiple lines {Scanned}

[David Shaw]

I also have multiple line with Broadvoice. I would like to have each
incoming line ring a different extension and configure an internal user
to use his or her own broadvoice line..

Here is my sip.conf

register =
[EMAIL PROTECTED]:password:[EMAIL PROTECTED]


register =
[EMAIL PROTECTED]:password:[EMAIL PROTECTED]

[broadvoice1]
type=peer
username=XX
fromuser=XX
authuser=XX
secret=password
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
context=from-broadvoice1
dtmfmode=inband
disallow=all
allow=ulaw
canreinvite=no
nat=no

[bv-in-1]
type=friend
host=sip.broadvoice.com
context=from-broadvoice1
dtmfmode=inband
canreinvite=yes
nat=no
allow=ulaw

[broadvoice2]
type=peer
username=AA
fromuser=AA
authuser=AA
secret=password
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
context=from-broadvoice2
dtmfmode=inband
disallow=all
allow=ulaw
canreinvite=no
nat=no

[bv-in-2]
type=friend
host=sip.broadvoice.com
context=from-broadvoice2
dtmfmode=inband
canreinvite=yes
nat=no
allow=ulaw

so on so on

But all incoming calls on Broadvoice uses extensions.conf [broadvoice4]
or what evers the last line for broadvoice.


On Wed, 2005-03-09 at 18:53 -0600, James Taylor wrote:
 I configured this once now I forgot what I did.
 
 Two Broadvoice accounts.
 Incoming is simple - just use the phone numbers.
 
 Outgoing:
 
 Dial out on a specific line
 and/or
 set up the groups and select the other line if the first one is busy?
 
 
 -- 
 James Taylor
 MetroTel
 3505 Summerihll Road
 Suite 11
 Texarkana, Texas  75503
 903-793-1956
 ___
 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
 

___
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

Re: [Asterisk-Users] Vonage a provider?

[Greg Hill]

On Fri, 11 Mar 2005, Frank Abernathy wrote:

 I am new to the mailing list, but I am very interested in running my small
 home business office phone system using Asterisk.  However, Broadvoice, a
 VoIP provider of choice based on my research, is not available in my area.

 I currently use Vonage VoIP.  Their website mentions nothing about being
 able to link to Asterisk.  I was wondering if any US subscribers have been
 able to configure Vonage with Asterisk.  Or if anyone has found Vonage to be
 a non-compatible provider.

The only known (to me) way to connect Asterisk and vonage is to buy their
normal service using their provided terminal adapter, and then connect
that to a card in the Asterisk box, or to add on a softphone account and
feed those credentials to Asterisk so that it can connect with their
servers directly.

If you go to google and search for asterisk vonage site:lists.digium.com
you'll find references to several sample configurations.

Greg


___
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

RE: [Asterisk-Users] Multiple IAX Phones Behind NAT

[Wiley Siler]

The port for sip is 5060.
Why no just map an ext to an internal and the problem us solved.
Assuming you have FW access and enough Ips.

W


 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Harold
Fletcher
Sent: Friday, March 11, 2005 10:55 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Multiple IAX Phones Behind NAT

Steve:

But how will that work for incoming calls?  Assume that three phones
have registered with an Asterisk box from inside a NAT, then * knows
that these three users can be found at x.x.x.x port 4569.  When *
receives an incoming call intended for one of those users, all it can do
is forward those calls to x.x.x.x:4569, right?  In that case I don't
understand how the NAT can know to which user the incoming call is
referring.

Will Fletcher



Auburn University
Department of Computer Science
107 Dunstan Hall
Auburn, AL 36849
334-332-9544
[EMAIL PROTECTED]
 [EMAIL PROTECTED] 03/11/05 11:22 AM 
Will Fletcher wrote:

 Hi folks,

 Ok, I've seen this question go unanswered on the mailing list, and I 
 assume it's because no one had the heart to break the bad news to the 
 guy asking, but be honest with me, I can take it.  At this time it's 
 flat impossible to have multiple IAX phones behind a NAT without using

 an * gateway because there's no way to have a client listen on a port 
 besides 4569.  Is my only option to learn about SIP and attempt to 
 forward that through my NAT?


Not true;

You just need the iax phones to register, and then they will work fine
through the NAT; whatever port they're on on local machines, and
whatever port they get NATted to won't matter.

-SteveK

___
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

___
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
___
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

[Asterisk-Users] Asterisk security problem: authorized SIP users can fake any callerid!

[Deti Fliegl]

Hi there,
all that started by investigating what happens if SIP clients are 
calling anonymously.
The problem: Every client who is registered as a regular user with 
username and secret can fake any callerid in subsequent INVITEs. 
Asterisk does not apply an accountcode or callerid from sip.conf. Those 
calls end up unbilled and untraceable.

Is there any way to fix this problem - did I misunderstand something, 
what am I doing wrong?

Deti
___
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

[Asterisk-Users] Re: diffrent area codes for diffrent phones in dialplan

[Doug Meredith]

Jer [EMAIL PROTECTED] wrote:

I have 3 sets of SIP phones all in diff area codes that need to access the PSTN

I need to it so that a 7 digit number is converted to a 10 digit with the 
correct ara code

eg a call coming from sip-phone1 needs aera code AAA and a call coming fom 
sip-phone2 needs BBB
how can this be setup in the dialplan
is there someway to set a var on a per sip group basis?
I thought of the accountcode...since i will not be using it for CDR

How about a different initial context for each area code?

Doug
-- 
Doug Meredith ([EMAIL PROTECTED])
SystemGuard - Oracle remote support
877-974-8273 (87-SYSGUARD)
506-854-7997
www.systemguard.com

___
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