[Asterisk-Users] PRI Errors

[Andrew McRory]

I've been running Asterisk CVS-02/29/04-12:09:10 for a couple of weeks 
with no real incidents...

LEC-PRI --- T400P - * - SIP/IAX
|
|- MicroCom ISPorte (faxserver)
|
|- Max4004 (dialup)

This configuration has added some flexibility we didn't have before and I
love it but two weeks of uptime the following errors appeared in the logs.
All connections were dropped. Is this an * problem or something on the
PRI?

It happened at 3AM so it wasn't a big deal this time but I hate to see it
happen in the middle of the day.


=
Mar 14 03:11:54 WARNING[11276]: PRI: Read on 108 failed: Unknown error 500
Mar 14 03:11:54 NOTICE[11276]: PRI got event: 6 on span 1
Mar 14 03:11:54 WARNING[998419]: PRI: Short write: -1/15 (Unknown error 500)
Mar 14 03:11:54 WARNING[998419]: Detected alarm on channel 1: Red Alarm
Mar 14 03:11:54 WARNING[15376]: Detected alarm on channel 2: Red Alarm
Mar 14 03:11:54 WARNING[15376]: Detected alarm on channel 4: Red Alarm
Mar 14 03:11:54 WARNING[15376]: Detected alarm on channel 5: Red Alarm
Mar 14 03:11:54 WARNING[15376]: Detected alarm on channel 6: Red Alarm
Mar 14 03:11:54 WARNING[15376]: Detected alarm on channel 7: Red Alarm
Mar 14 03:11:54 WARNING[15376]: Detected alarm on channel 8: Red Alarm
Mar 14 03:11:54 WARNING[15376]: Detected alarm on channel 9: Red Alarm
Mar 14 03:11:54 WARNING[15376]: Detected alarm on channel 10: Red Alarm
Mar 14 03:11:54 WARNING[15376]: Detected alarm on channel 11: Red Alarm
Mar 14 03:11:54 WARNING[15376]: Detected alarm on channel 12: Red Alarm
Mar 14 03:11:54 WARNING[15376]: Detected alarm on channel 13: Red Alarm
Mar 14 03:11:54 WARNING[15376]: Detected alarm on channel 14: Red Alarm
Mar 14 03:11:54 WARNING[15376]: Detected alarm on channel 15: Red Alarm
Mar 14 03:11:54 WARNING[15376]: Detected alarm on channel 16: Red Alarm
Mar 14 03:11:54 WARNING[15376]: Detected alarm on channel 17: Red Alarm
Mar 14 03:11:54 WARNING[15376]: Detected alarm on channel 18: Red Alarm
Mar 14 03:11:54 WARNING[15376]: Detected alarm on channel 19: Red Alarm
Mar 14 03:11:54 WARNING[15376]: Detected alarm on channel 20: Red Alarm
Mar 14 03:11:54 WARNING[15376]: Detected alarm on channel 21: Red Alarm
Mar 14 03:11:54 WARNING[15376]: Detected alarm on channel 22: Red Alarm
Mar 14 03:11:54 WARNING[15376]: Detected alarm on channel 23: Red Alarm
Mar 14 03:11:54 WARNING[836629]: PRI: Short write: -1/15 (Unknown error 500)
Mar 14 03:11:54 WARNING[836629]: Detected alarm on channel 3: Red Alarm
Mar 14 03:11:54 WARNING[11276]: PRI: Read on 108 failed: Unknown error 500
Mar 14 03:11:54 NOTICE[11276]: PRI got event: 4 on span 1
Mar 14 03:11:55 WARNING[12301]: PRI: !! Got reject for frame 105, retransmitting frame 
105 now, updating n_r!
Mar 14 03:11:55 WARNING[12301]: PRI: !! Got reject for frame 105, retransmitting frame 
106 now, updating n_r!
Mar 14 03:11:55 WARNING[11276]: PRI: !! Got reject for frame 41, retransmitting frame 
41 now, updating n_r!
Mar 14 03:11:55 WARNING[11276]: PRI: !! Got reject for frame 41, retransmitting frame 
42 now, updating n_r!
Mar 14 03:11:59 NOTICE[15376]: Alarm cleared on channel 1
Mar 14 03:11:59 NOTICE[15376]: Alarm cleared on channel 2
Mar 14 03:11:59 NOTICE[15376]: Alarm cleared on channel 3
Mar 14 03:11:59 NOTICE[15376]: Alarm cleared on channel 4
Mar 14 03:11:59 NOTICE[15376]: Alarm cleared on channel 5
Mar 14 03:11:59 NOTICE[15376]: Alarm cleared on channel 6
Mar 14 03:11:59 NOTICE[15376]: Alarm cleared on channel 7
Mar 14 03:11:59 NOTICE[15376]: Alarm cleared on channel 8
Mar 14 03:11:59 NOTICE[15376]: Alarm cleared on channel 9
Mar 14 03:11:59 NOTICE[15376]: Alarm cleared on channel 10
Mar 14 03:11:59 NOTICE[15376]: Alarm cleared on channel 11
Mar 14 03:11:59 NOTICE[15376]: Alarm cleared on channel 12
Mar 14 03:11:59 WARNING[11276]: PRI: Read on 108 failed: Unknown error 500
Mar 14 03:11:59 NOTICE[15376]: Alarm cleared on channel 13
Mar 14 03:11:59 NOTICE[11276]: PRI got event: 5 on span 1
Mar 14 03:11:59 NOTICE[15376]: Alarm cleared on channel 14
Mar 14 03:11:59 NOTICE[15376]: Alarm cleared on channel 15
Mar 14 03:11:59 NOTICE[15376]: Alarm cleared on channel 16
Mar 14 03:11:59 NOTICE[15376]: Alarm cleared on channel 17
Mar 14 03:11:59 NOTICE[15376]: Alarm cleared on channel 18
Mar 14 03:11:59 NOTICE[15376]: Alarm cleared on channel 19
Mar 14 03:11:59 NOTICE[15376]: Alarm cleared on channel 20
Mar 14 03:11:59 NOTICE[15376]: Alarm cleared on channel 21
Mar 14 03:11:59 NOTICE[15376]: Alarm cleared on channel 22
Mar 14 03:11:59 NOTICE[15376]: Alarm cleared on channel 23

/etc/zaptel.conf 
span=1,1,0,esf,b8zs
span=2,0,0,esf,b8zs
span=3,0,0,esf,b8zs
span=4,0,0,esf,b8zs
bchan=1-23
dchan=24
bchan=25-47
dchan=48
bchan=49-71
dchan=72
bchan=73-95
dchan=96

/etc/asterisk/zapata.conf ==

[Asterisk-Users] alias in h323.conf

[Asan Mambetaliev]

Kind day by all.
Whether there is an opportunity in a file h323.conf to specify group alias?
I would like to check only IP address for many users.
That that like it
[group]
type=user
host=192.168.0.18
host=192.168.0.19
host=192.168.0.20
context=incoming

I shall be very grateful for examples of configurations.
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[Asterisk-Users] Sipura click click bad quality

[Miguel Cavazos]

hello guys heres my setup i have 2 asterisk servers in 2 different
houses sipura ulaw --- asterisk ---> iax2 (ilbc) ---> asterisk -- sipura
ulaw, this is my setup but when i call the other sipura i can listen
like click click click click it doesnt seem a bandwidth issue because it
has dedicated dsl lines on both ends im using on both sipuras the
lastest firmware .31

my sip.conf looks like this

[101]
type=friend
secret=mysecrete
context=master-6186
callerid="Mike" <101>
host=dynamic
reinvite=no
canreinvite=no 
dtmfmode=inband
transfer=yes
nat=0
disallow=all
allow=ulaw

and iax.conf looks like this

[oficina]
type=friend
username=oficina
secret=mysecrete
auth=plaintext
context=asterisk
host=dynamic
disallow=all
allow=ilbc

this setup is for one of the servers the second server looks very alike
to this could you help me out??? ill appreciate a response even i know
your not consern about asterisk but i just get this problem with
sipuras.

Miguel Cavazos

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RE: [Asterisk-Users] OT: cisco 7960G powered by 3com 3CNJPSE

[Greg Boehnlein]

On Tue, 16 Mar 2004, Dean Collins wrote:

> Cisco have the terminals around the other way, this is a well known
> problem, do a search and you'll find what you need to do.
> 
> Cheers,
> Dean

Alright.. since I'm the one that posted about using the standard POE 
injectors wih the 7960, I think I might need to revise the Wiki page.

Am I to understand that the 7960G does -NOT- support the 802.3af Power 
Over Ethernet standard?

If that is the case, then I need to modify the Wiki page, which I am 
happy to do.

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



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RE: [Asterisk-Users] AgentCallBackLogin ??

[AstGrp]

It looks like the AgentCallBackLogin app is not working in the latest
CVS Asterisk CVS-03/15/04.  Can someone please verify this.  I had
this exact setup running on a different CVS load prior to running the
updates.

Thanks,

-gcc

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of AstGrp
Posted At: Monday, March 15, 2004 3:27 PM
Posted To: Asterisk User Group
Conversation: AgentCallBackLogin ??
Subject: [Asterisk-Users] AgentCallBackLogin ??


I could use a little assistance.. I am sure I am doing something
stupid. The problem I am having is when the call comes in and runs
the context [411].  The call is generated, but never makes the call.  It
rings back the user who is making the call.  It works fine if I dial
context [411] from the inside.  It sounds like I need to add some
context somewhere just not sure what where?

[agents]
agent => 4001,4001,Geoff Clark

[general]
[default]
[tech]
member => Agent/4001
strategy = roundrobin
timeout = 30
retry = 10


[411]
exten => 411,1,Answer
exten => 411,2,Wait,2
exten => 411,3,Background(auth-thankyou)
exten => 411,4,Queue(tech)
exten => *6,1,AgentCallbackLogin(@411)
exten => *4001,1,Dial(${TRUNK}/${GCELL:${TRUNKMSD}})

Thanks,

gcc
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RE: [Asterisk-Users] OT: cisco 7960G powered by 3com 3CNJPSE

[Dean Collins]

Cisco have the terminals around the other way, this is a well known
problem, do a search and you'll find what you need to do.

Cheers,
Dean


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Hermann
Wecke
Sent: Tuesday, 16 March 2004 1:44 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] OT: cisco 7960G powered by 3com 3CNJPSE

On Mon, 15 Mar 2004, stan wrote:
> Is anyone using a 3com 3CNJPSE to power a 7960G?

Forgot to mention that I also have a a 7960G and I tried to use a Compaq
PoE (http://www.compaq.ca/english/business/mobile/wireless/poe.asp) and
a
3com Network Jack NJ100
(http://www.3com.com/products/en_US/detail.jsp?tab=support&pathtype=supp
ort&sku=3CNJ100POE-CRM-20)
and none of them worked with my 7960G.

When I asked about the "native PoE support", I received these two
replies:

From: Peter Lei
Subject: Re: [FWD] Cisco 7960G (off topic)

7960's can be powered over ethernet... it is the Cisco POE spec though.

==

From: Mike Purdon
Subject: Re: [FWD] Cisco 7960G (off topic)

Any old -48V DC at 5 Watts will do. Cisco can supply the -48V thru the
RJ45 but your switchblade needs to support this.

Mike
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Re: [Asterisk-Users] OT: cisco 7960G powered by 3com 3CNJPSE

[Hermann Wecke]

On Mon, 15 Mar 2004, stan wrote:
> Is anyone using a 3com 3CNJPSE to power a 7960G?

Forgot to mention that I also have a a 7960G and I tried to use a Compaq
PoE (http://www.compaq.ca/english/business/mobile/wireless/poe.asp) and a
3com Network Jack NJ100
(http://www.3com.com/products/en_US/detail.jsp?tab=support&pathtype=support&sku=3CNJ100POE-CRM-20)
and none of them worked with my 7960G.

When I asked about the "native PoE support", I received these two replies:

From: Peter Lei
Subject: Re: [FWD] Cisco 7960G (off topic)

7960's can be powered over ethernet... it is the Cisco POE spec though.

==

From: Mike Purdon
Subject: Re: [FWD] Cisco 7960G (off topic)

Any old -48V DC at 5 Watts will do. Cisco can supply the -48V thru the
RJ45 but your switchblade needs to support this.

Mike
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Re: [Asterisk-Users] Need help to format asterisk MGCP packet.

[Duane Cox]

And this is the debug output from the normal chan_mgcp.c CVS from today.



MGCP read: 
rsip 14412 aaln/[EMAIL PROTECTED] MGCP 1.0
RM:forced
RD:0

from 63.252.240.2:2427MGCP read: 
rsip 14412 aaln/[EMAIL PROTECTED] MGCP 1.0
RM:forced
RD:0

from 63.252.240.2:2427Verb: 'rsip', Identifier: '14412', Endpoint: 'aaln/[EMAIL 
PROTECTED]', Version: 'MGCP 1.0'
3 headers, 0 lines
Handling request 'rsip' on aaln/[EMAIL PROTECTED]
-- Resetting interface aaln/[EMAIL PROTECTED]
Transmitting:
200 14412 OK

 to 63.252.240.2:2427
-- MGCP Asked to indicate tone:  on  aaln/[EMAIL PROTECTED] in cxmode: inactive
Posting Request:
RQNT 8 aaln/[EMAIL PROTECTED] MGCP 1.0
X: 14b0e00a
R: hd(N)
 to 63.252.240.2:2427
MGCP read: 
rsip 14413 aaln/[EMAIL PROTECTED] MGCP 1.0
RM:restart
RD:0

from 63.252.240.2:2427MGCP read: 
rsip 14413 aaln/[EMAIL PROTECTED] MGCP 1.0
RM:restart
RD:0

from 63.252.240.2:2427Verb: 'rsip', Identifier: '14413', Endpoint: 'aaln/[EMAIL 
PROTECTED]', Version: 'MGCP 1.0'
3 headers, 0 lines
Handling request 'rsip' on aaln/[EMAIL PROTECTED]
Mar 15 20:19:50 NOTICE[-1136878672]: chan_mgcp.c:398 dump_queue: Removing message from 
aaln/[EMAIL PROTECTED] tansaction 8
-- Resetting interface aaln/[EMAIL PROTECTED]
Transmitting:
200 14413 OK

 to 63.252.240.2:2427
-- MGCP Asked to indicate tone:  on  aaln/[EMAIL PROTECTED] in cxmode: inactive
Posting Request:
RQNT 9 aaln/[EMAIL PROTECTED] MGCP 1.0
X: 14b0e00a
R: hd(N)
 to 63.252.240.2:2427
MGCP read: 
500 8 Unknown endpoint

from 63.252.240.2:2427MGCP read: 
500 8 Unknown endpoint

from 63.252.240.2:2427Verb: '500', Identifier: '8', Endpoint: 'Unknown', Version: 
'endpoint'
1 headers, 0 lines
Mar 15 20:19:50 NOTICE[-1136878672]: chan_mgcp.c:1716 handle_response: Got response 
back on aaln/[EMAIL PROTECTED] for transaction 8 we aren't sending? (current = 9)

 
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Re: [Asterisk-Users] Need help to format asterisk MGCP packet.

[Duane Cox]

I disabled and enabled the voice port on the FTTH device.
This is the Asterisk debug output when I enabled it.

This debug is with the altered code (chan_mgcp.c)

---

MGCP read: 
rsip 14406 aaln/[EMAIL PROTECTED] MGCP 1.0
RM:forced
RD:0

from 63.252.240.2:2427MGCP read: 
rsip 14406 aaln/[EMAIL PROTECTED] MGCP 1.0
RM:forced
RD:0

from 63.252.240.2:2427Verb: 'rsip', Identifier: '14406', Endpoint: 'aaln/[EMAIL 
PROTECTED]', Version: 'MGCP 1.0'
3 headers, 0 lines
Handling request 'rsip' on aaln/[EMAIL PROTECTED]
-- Resetting interface aaln/[EMAIL PROTECTED]
Transmitting:
200 14406 OK

 to 63.252.240.2:2427
-- MGCP Asked to indicate tone:  on  aaln/[EMAIL PROTECTED] in cxmode: inactive
Posting Request:
RQNT 5 aaln/[EMAIL PROTECTED] MGCP 1.0
X: 00640f40
R: hd(N)
 to 63.252.240.2:2427
MGCP read: 
rsip 14407 aaln/[EMAIL PROTECTED] MGCP 1.0
RM:restart
RD:0

from 63.252.240.2:2427MGCP read: 
rsip 14407 aaln/[EMAIL PROTECTED] MGCP 1.0
RM:restart
RD:0

from 63.252.240.2:2427Verb: 'rsip', Identifier: '14407', Endpoint: 'aaln/[EMAIL 
PROTECTED]', Version: 'MGCP 1.0'
3 headers, 0 lines
Handling request 'rsip' on aaln/[EMAIL PROTECTED]
Mar 15 20:09:27 NOTICE[-1137189968]: chan_mgcp.c:398 dump_queue: Removing message from 
aaln/[EMAIL PROTECTED] tansaction 5
-- Resetting interface aaln/[EMAIL PROTECTED]
Transmitting:
200 14407 OK

 to 63.252.240.2:2427
-- MGCP Asked to indicate tone:  on  aaln/[EMAIL PROTECTED] in cxmode: inactive
Posting Request:
RQNT 6 aaln/[EMAIL PROTECTED] MGCP 1.0
X: 00640f40
R: hd(N)
 to 63.252.240.2:2427
MGCP read: 
200 5 OK

from 63.252.240.2:2427MGCP read: 
200 5 OK

from 63.252.240.2:2427Verb: '200', Identifier: '5', Endpoint: 'OK', Version: '(null)'
1 headers, 0 lines
Mar 15 20:09:27 NOTICE[-1137189968]: chan_mgcp.c:1716 handle_response: Got response 
back on aaln/[EMAIL PROTECTED] for transaction 5 we aren't sending? (current = 6)
MGCP read: 
200 6 OK

from 63.252.240.2:2427MGCP read: 
200 6 OK

from 63.252.240.2:2427Verb: '200', Identifier: '6', Endpoint: 'OK', Version: '(null)'
1 headers, 0 lines


 
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Re: [Asterisk-Users] Need help to format asterisk MGCP packet.

[Derek Bruce]

post the asterisk side of the conversation with this device with debugging
turned on... maybe there is a way to detect if the device have this
'anomaly' and deal with it automagically...


- Original Message -
From: "Duane Cox" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, March 15, 2004 5:46 PM
Subject: Re: [Asterisk-Users] Need help to format asterisk MGCP packet.


> YES, that change in the chan_mgcp.c worked for my setup.  Like I said
before, it may not be the best long term solution.
>
> What would need to be done to get this into the CVS?  Maybe not this exact
hack, but some sort of option to allow the change in the format of the
packet?


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Re[2]: [Asterisk-Users] Grandstream TFTP Config

[Masakazu Nakano]

On 16 Mar 2004 01:08:27 +
"Hermann Wecke" <[EMAIL PROTECTED]> wrote:

> On Mon, 15 Mar 2004, Matthew Marlowe wrote:
> > I can confirm  1.0.4.53 is bad as well. :) 1.0.4.50 has been working
> > fine for me.
> 
> I received the 1.0.4.54 firmware. So far, so good. No new problems.

How can I get this firm?

that isn't in here ;-)

http://www.grandstream.com/BETATEST/

mack_jpn

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RE: [Asterisk-Users] Grandstream TFTP Config

[Hermann Wecke]

On Mon, 15 Mar 2004, Matthew Marlowe wrote:
> I can confirm  1.0.4.53 is bad as well. :) 1.0.4.50 has been working
> fine for me.

I received the 1.0.4.54 firmware. So far, so good. No new problems.
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Re: [Asterisk-Users] ISDN BRI with DDI support

[Maciej Kietlinski]

> I want to build in my small company little PBX with asterisk.
> I have one ISDN BRI link with DDI preselection and a couple of analog
> phones.
> 
> I need somethink like this
> 
>+--+--- phone 1 (extension 1)
>| LINUX with ASTERISK  +--- phone 2 (extension 2)
>|  | 
> ISDN BRI DDI   |  +--- phone 3 (extension 3)
> ---+  |
>|  +--- phone 4 (extension 4)
>|  |
>+--+--- phone 5 (extension 5)
> 
> 
> So, my problem is, I'm lost in lots of abbreviations, and I have no idea
> which pci card to select.
> 
> I know, I need one ISDN BRI DDI card for incoming isdn line
> and some pci cards with RJ11 for analog phones.
> 
> 1) can anybody help me to select correct pci cards?
> 2) can i use more then one isdn cards (connected together) if I need
> more the 2 simultaneous phone calls?

For phones: Digium TDM?0B,
For line(s): Eicon Diva Server - with 1 or 4 BRI interfaces.

Maciej Kietlinski

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Re: [Asterisk-Users] Need help to format asterisk MGCP packet.

[Duane Cox]

YES, that change in the chan_mgcp.c worked for my setup.  Like I said before, it may 
not be the best long term solution.

What would need to be done to get this into the CVS?  Maybe not this exact hack, but 
some sort of option to allow the change in the format of the packet?

ODDLY, the device sends up to asterisk in that format, and asterisk could understand 
the device, it just couldn't communicate back (before the change) 

Apparently with this STACK used by this device, the manufactuer is dead set on NOT 
changing it. I am finding that there is no 'true MGCP standard' as I work with it 
between vendors


***LOG SHOWS IT IS NOW WORKING***

Aug  2 00:49:54 [4280831] MGCP DEBUG INFO at mgcp/common/transport/rvmgcpstack.c 1602:
SEND: rsip 14394 aaln/[EMAIL PROTECTED] MGCP 1.0
RM:forced
RD:0

Aug  2 00:49:54 [4280831] MGCP DEBUG INFO at mgcp/common/transport/rvmgcpstack.c 1602:
SEND: 200 6 OK

Aug  2 00:49:54 [4280832] MGCP DEBUG INFO at mgcp/common/transport/rvmgcpstack.c 1143:
RECEIVE: RQNT 6 aaln/[EMAIL PROTECTED] MGCP 1.0
X: 0704259f
R: hd(N)

Aug  2 00:49:54 [4280832] MGCP DEBUG INFO at mgcp/common/transport/rvmgcpstack.c 1481:
SEND: 200 5 OK

Aug  2 00:49:55 [4280832] MGCP DEBUG INFO at mgcp/common/transport/rvmgcpstack.c 1143:
RECEIVE: RQNT 6 aaln/[EMAIL PROTECTED] MGCP 1.0
X: 0704259f
R: hd(N)

Aug  2 00:49:55 [4280832] MGCP DEBUG INFO at mgcp/common/transport/rvmgcpstack.c 1143:
RECEIVE: RQNT 6 aaln/[EMAIL PROTECTED] MGCP 1.0
X: 0704259f
R: hd(N)

Aug  2 00:49:55 [4280832] MGCP DEBUG INFO at mgcp/common/transport/rvmgcpstack.c 1481:
SEND: 200 6 OK

Aug  2 00:49:55 [4280832] MGCP DEBUG INFO at mgcp/common/transport/rvmgcpstack.c 1143:
RECEIVE: 200 14394 OK


-- Original Message --
From: "Duane Cox" <[EMAIL PROTECTED]>
Reply-To: [EMAIL PROTECTED]
Date:  Mon, 15 Mar 2004 18:28:08 -0600

>I will apply this and recompile.  I was hoping I could do this with a format or a 
>flag (as I may need to send MGCP to other devices in the correct format).
>
>I will let you know how it works.
>
>MANY THANKS!
>
>Duane Cox
>
>
>
>
>-- Original Message --
>From: "Derek Bruce" <[EMAIL PROTECTED]>
>Reply-To: [EMAIL PROTECTED]
>Date:  Mon, 15 Mar 2004 17:14:59 -0700
>
>>I don't use MGCP, but after a quick look at the code I may have a solution
>>for you...
>>
>>in chan_mgcp.c, replace the snprintf line in init_req (line#1367 in march 1
>>cvs version)with:
>>
>>snprintf(req->header[req->headers], sizeof(req->data) - req->len, "%s %d
>>[EMAIL PROTECTED] MGCP 1.0\r\n", verb, oseq, p->name, p->parent->name);
>>
>>this should give you what you need...
>>
>>
>>- Original Message -
>>From: "Duane Cox" <[EMAIL PROTECTED]>
>>To: <[EMAIL PROTECTED]>
>>Sent: Monday, March 15, 2004 4:21 PM
>>Subject: [Asterisk-Users] Need help to format asterisk MGCP packet.
>>
>>
>>> Hello, I am trying to setup a 'gateway' fiber to the home device from
>>wave7optics (www.wave7optics.com)
>>>
>>> They use a MGCP stack from "RAD?"  and did not code it themselves.  My
>>MGCP name-convetion on this device is set to IP, but this stack expects the
>>received MGCP packets to be formated as 'aaln/[EMAIL PROTECTED]' (notice the
>>brackets)
>>>
>>> I have tried to adjust my extensions.conf and mgcp.conf to compensate, but
>>could not get asterisk to send in that format.
>>>
>>> Can anyone help?
>>>
>>> This is the debug log from the 'gateway' device.
>>> You can see the format that it is sending out, and it is expecting to get
>>the same format back.  You can see that asterisk is sending to the device in
>>the format (without the brackets) and this gateway device responds with
>>'endpoint unknown'
>>>
>>> ***LOG***
>>>
>>>
>>> Aug  1 23:19:27 [3195449] MGCP DEBUG INFO at
>>mgcp/common/transport/rvmgcpstack.c 1602:
>>> SEND: rsip 14389 aaln/[EMAIL PROTECTED] MGCP 1.0
>>> RM:restart
>>> RD:0
>>>
>>> Aug  1 23:19:27 [3195450] MGCP DEBUG INFO at
>>mgcp/common/transport/rvmgcpstack.c 1143:
>>> RECEIVE: RQNT 6 aaln/[EMAIL PROTECTED] MGCP 1.0
>>> X: 2467de76
>>> R: hd(N)
>>>
>>> Aug  1 23:19:27 [3195450] MGCP DEBUG INFO at
>>mgcp/common/transport/rvmgcpstack.c 1143:
>>> RECEIVE: RQNT 6 aaln/[EMAIL PROTECTED] MGCP 1.0
>>> X: 2467de76
>>> R: hd(N)
>>>
>>> Aug  1 23:19:28 [3195450] MGCP DEBUG INFO at
>>mgcp/common/transport/rvmgcpstack.c 1481:
>>> SEND: 500 5 Unknown endpoint
>>>
>>> Aug  1 23:19:28 [3195450] MGCP DEBUG INFO at
>>mgcp/common/transport/rvmgcpstack.c 1143:
>>> RECEIVE: RQNT 6 aaln/[EMAIL PROTECTED] MGCP 1.0
>>> X: 2467de76
>>> R: hd(N)
>>>
>>> Aug  1 23:19:28 [3195450] MGCP DEBUG INFO at
>>mgcp/common/transport/rvmgcpstack.c 1143:
>>> RECEIVE: RQNT 6 aaln/[EMAIL PROTECTED] MGCP 1.0
>>> X: 2467de76
>>> R: hd(N)
>>>
>>> Aug  1 23:19:28 [3195450] MGCP DEBUG INFO at
>>mgcp/common/transport/rvmgcpstack.c 1481:
>>> SEND: 500 5 Unknown endpoint
>>>
>>>
>>> ___
>>> Asterisk-Users mailing list
>>> [EMAIL PROTECTED]
>>> htt

RE: [Asterisk-Users] x100p CLI in the UK

[David J Carter]

Chris,

May be a bad card, or more likely Microfilter, I have had mine on the same
line as the ADSL for 3 months now and no problems.

As for UK CLI I will be glad when I can get CLI from either BT or Telewest.

Regards


Dave

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Chris Lee
Sent: 16 March 2004 00:26
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] x100p CLI in the UK


First, is the lack of UK CLI on the x100P hardware or software related?

Secondly, My US Robotics Voice modem does get UK CLI, so could I get UK
CLI and the same functionality as the x100p using a USR Modem with *?

Has anyone done this?

As an aside, has anyone experienced or solved the problem with the x100p
producing a loop condition on the PSTN line (it really mucks up my ADSL
connection something horrid when it is connected).
I think it is due to an impedance mismatch between the card and the
network, but have no way of testing these things. (Dont know enough to
just get out my meter and start probing without risk of killing my x100p
or the POTS Line)
I know the Loop condition is there as a kindly BT eng was monitoring the
line and asking me to plug things in, when the x100p was plugged in he
said something along the lines of: "theres your problem, what did you
just plug in? It is creating a 36 K Ohm Loop condition"

Now the router is not the most stable at the best of times but plug in
the x100p and the line bounces up and down like there is no tomorrow.

Regards

Chris.
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Re: [Asterisk-Users] x100p CLI in the UK

[Eric Wieling]

On Mon, 2004-03-15 at 18:25, Chris Lee wrote:
> First, is the lack of UK CLI on the x100P hardware or software related?

Check the extensive discussions in the mailing list archives.

> Secondly, My US Robotics Voice modem does get UK CLI, so could I get UK 
> CLI and the same functionality as the x100p using a USR Modem with *?

Also see the description of someone doing this in the mailing list
arcgives.

> As an aside, has anyone experienced or solved the problem with the x100p 
> producing a loop condition on the PSTN line (it really mucks up my ADSL 
> connection something horrid when it is connected).

Also see the mailing list archives for discussion about loop current.

Do you see a pattern here?  The archives are you friend.  

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Re: [Asterisk-Users] Need help to format asterisk MGCP packet.

[Duane Cox]

I will apply this and recompile.  I was hoping I could do this with a format or a flag 
(as I may need to send MGCP to other devices in the correct format).

I will let you know how it works.

MANY THANKS!

Duane Cox




-- Original Message --
From: "Derek Bruce" <[EMAIL PROTECTED]>
Reply-To: [EMAIL PROTECTED]
Date:  Mon, 15 Mar 2004 17:14:59 -0700

>I don't use MGCP, but after a quick look at the code I may have a solution
>for you...
>
>in chan_mgcp.c, replace the snprintf line in init_req (line#1367 in march 1
>cvs version)with:
>
>snprintf(req->header[req->headers], sizeof(req->data) - req->len, "%s %d
>[EMAIL PROTECTED] MGCP 1.0\r\n", verb, oseq, p->name, p->parent->name);
>
>this should give you what you need...
>
>
>- Original Message -
>From: "Duane Cox" <[EMAIL PROTECTED]>
>To: <[EMAIL PROTECTED]>
>Sent: Monday, March 15, 2004 4:21 PM
>Subject: [Asterisk-Users] Need help to format asterisk MGCP packet.
>
>
>> Hello, I am trying to setup a 'gateway' fiber to the home device from
>wave7optics (www.wave7optics.com)
>>
>> They use a MGCP stack from "RAD?"  and did not code it themselves.  My
>MGCP name-convetion on this device is set to IP, but this stack expects the
>received MGCP packets to be formated as 'aaln/[EMAIL PROTECTED]' (notice the
>brackets)
>>
>> I have tried to adjust my extensions.conf and mgcp.conf to compensate, but
>could not get asterisk to send in that format.
>>
>> Can anyone help?
>>
>> This is the debug log from the 'gateway' device.
>> You can see the format that it is sending out, and it is expecting to get
>the same format back.  You can see that asterisk is sending to the device in
>the format (without the brackets) and this gateway device responds with
>'endpoint unknown'
>>
>> ***LOG***
>>
>>
>> Aug  1 23:19:27 [3195449] MGCP DEBUG INFO at
>mgcp/common/transport/rvmgcpstack.c 1602:
>> SEND: rsip 14389 aaln/[EMAIL PROTECTED] MGCP 1.0
>> RM:restart
>> RD:0
>>
>> Aug  1 23:19:27 [3195450] MGCP DEBUG INFO at
>mgcp/common/transport/rvmgcpstack.c 1143:
>> RECEIVE: RQNT 6 aaln/[EMAIL PROTECTED] MGCP 1.0
>> X: 2467de76
>> R: hd(N)
>>
>> Aug  1 23:19:27 [3195450] MGCP DEBUG INFO at
>mgcp/common/transport/rvmgcpstack.c 1143:
>> RECEIVE: RQNT 6 aaln/[EMAIL PROTECTED] MGCP 1.0
>> X: 2467de76
>> R: hd(N)
>>
>> Aug  1 23:19:28 [3195450] MGCP DEBUG INFO at
>mgcp/common/transport/rvmgcpstack.c 1481:
>> SEND: 500 5 Unknown endpoint
>>
>> Aug  1 23:19:28 [3195450] MGCP DEBUG INFO at
>mgcp/common/transport/rvmgcpstack.c 1143:
>> RECEIVE: RQNT 6 aaln/[EMAIL PROTECTED] MGCP 1.0
>> X: 2467de76
>> R: hd(N)
>>
>> Aug  1 23:19:28 [3195450] MGCP DEBUG INFO at
>mgcp/common/transport/rvmgcpstack.c 1143:
>> RECEIVE: RQNT 6 aaln/[EMAIL PROTECTED] MGCP 1.0
>> X: 2467de76
>> R: hd(N)
>>
>> Aug  1 23:19:28 [3195450] MGCP DEBUG INFO at
>mgcp/common/transport/rvmgcpstack.c 1481:
>> SEND: 500 5 Unknown endpoint
>>
>>
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[Asterisk-Users] x100p CLI in the UK

[Chris Lee]

First, is the lack of UK CLI on the x100P hardware or software related?

Secondly, My US Robotics Voice modem does get UK CLI, so could I get UK 
CLI and the same functionality as the x100p using a USR Modem with *?

Has anyone done this?

As an aside, has anyone experienced or solved the problem with the x100p 
producing a loop condition on the PSTN line (it really mucks up my ADSL 
connection something horrid when it is connected).
I think it is due to an impedance mismatch between the card and the 
network, but have no way of testing these things. (Dont know enough to 
just get out my meter and start probing without risk of killing my x100p 
or the POTS Line)
I know the Loop condition is there as a kindly BT eng was monitoring the 
line and asking me to plug things in, when the x100p was plugged in he 
said something along the lines of: "theres your problem, what did you 
just plug in? It is creating a 36 K Ohm Loop condition"

Now the router is not the most stable at the best of times but plug in 
the x100p and the line bounces up and down like there is no tomorrow.

Regards

Chris.
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[Asterisk-Users] Re: incoming fax x100p

[Francois]

I don't know, but can SIP devices handle fax transmissions?  Why are
you trying to send a fax to a SIP phone?

Francois


On Mon, 15 Mar 2004 14:31:06 +, Simon Chappell
<[EMAIL PROTECTED]> wrote:

>thanks for the help
>
>I have added exten => fax,1,Dial(SIP/2004) in my extensions.conf
>What i can see from the CLI is that asterisk is reporting fax detected 
>but no fax extension , have i got the syntax wrong or do i need to add 
>it elseware
>
>thanks again
>
>Simon
>
>Jim Sneeringer wrote:
>
>>There is a special extension, called "fax," that you use to specify where
>>you want incoming fax calls to go.  It detects faxes by the CNG tone and
>>sends them where you specify.  Here is an example:
>>
>> exten => fax,1,Dial(${Fax}) ; Use Zap channel for 2004 instead of ${Fax}
>>
>>There are some faxes that do not send CNG tones, so this will occasionally
>>fail for that reason. You might also send calls that time out to the fax
>>extension, or you can use distinctive ringing, as someone else suggested.
>>
>>Let me know how this works for you. I am still having trouble with garbled
>>fax transmissions once I get a connection to the fax, and I would be
>>interested in your experience sending and receiving faxes.
>>
>>Jim
>>

Thanks,

Francois

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Re: [Asterisk-Users] MySQL Dynamic Extensions

[Andy Powell]

You could take a look at 

http://andreasotto.net/asterisk/

and modify that to suit

Andy

*** REPLY SEPARATOR  ***

On 15/03/2004 at 16:46 Tony Wasson wrote:

>Darren Nay wrote:
>> Hello All,
>> 
>>  
>> 
>> I am just looking into Asterisk as a viable voicemail solution for our
>phone
>> service.  In order to use it though I will need to make extensions.conf
>> dynamic (ie. Via MySQL).  Is this possible?
>> 
>
>Sure..
>
>Set up the database as you see documented. You can schedule this up to 
>once a minutes using crontab.
>
>Setting something like this in your /etc/crontab should do it nicely
>
>*/5 * * * * root /usr/local/sbin/update-voicemail 2>&1 > /var/log/vm.log
>
>NOTE: You can increase the frequency by using */3 or */1. */5 means 
>every 5 minutes. */3 means every 3 minutes.
>
>Then make /usr/local/sbin/update-voicemail look like:
>
>   #!/bin/bash
>   /path/to/retrieve_extensions_from_mysql.pl
>   /usr/sbin/asterisk -rx "extensions reload"
>
>Next, make the script executable
>#chmod +x /usr/local/sbin/update-voicemail
>
>This is not totally dynamic, but it ought to be close enough. You could 
>make this completely dynamic using a trigger.
>
>Tony Wasson
>
>
>> 
>> I've found the following information on this subject:
>> 
>> http://www.voip-info.org/wiki-Asterisk+extensions+from+mysql
>>  
>> 
>>  
>> 
>> However, this is not a fully dynamic function.  It requires me to pull
>the
>> mysql database every so often (presumably via cron) and then restart
>> asterisk after updating extensions.conf.
>> 
>>  
>> 
>> Is it possible to setup asterisks so that extensions.conf is fully
>dynamic
>> via a MySQL database?
>> 
>>  
>> 
>> Thanks for the help!! 
>> 
>>  
>> 
>> Regards,
>> 
>>  
>> 
>> Darren Nay
>> 
>> [EMAIL PROTECTED]  
>> 
>>  
>> 
>> 
>
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Re: [Asterisk-Users] Need help to format asterisk MGCP packet.

[Derek Bruce]

I don't use MGCP, but after a quick look at the code I may have a solution
for you...

in chan_mgcp.c, replace the snprintf line in init_req (line#1367 in march 1
cvs version)with:

snprintf(req->header[req->headers], sizeof(req->data) - req->len, "%s %d
[EMAIL PROTECTED] MGCP 1.0\r\n", verb, oseq, p->name, p->parent->name);

this should give you what you need...


- Original Message -
From: "Duane Cox" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, March 15, 2004 4:21 PM
Subject: [Asterisk-Users] Need help to format asterisk MGCP packet.


> Hello, I am trying to setup a 'gateway' fiber to the home device from
wave7optics (www.wave7optics.com)
>
> They use a MGCP stack from "RAD?"  and did not code it themselves.  My
MGCP name-convetion on this device is set to IP, but this stack expects the
received MGCP packets to be formated as 'aaln/[EMAIL PROTECTED]' (notice the
brackets)
>
> I have tried to adjust my extensions.conf and mgcp.conf to compensate, but
could not get asterisk to send in that format.
>
> Can anyone help?
>
> This is the debug log from the 'gateway' device.
> You can see the format that it is sending out, and it is expecting to get
the same format back.  You can see that asterisk is sending to the device in
the format (without the brackets) and this gateway device responds with
'endpoint unknown'
>
> ***LOG***
>
>
> Aug  1 23:19:27 [3195449] MGCP DEBUG INFO at
mgcp/common/transport/rvmgcpstack.c 1602:
> SEND: rsip 14389 aaln/[EMAIL PROTECTED] MGCP 1.0
> RM:restart
> RD:0
>
> Aug  1 23:19:27 [3195450] MGCP DEBUG INFO at
mgcp/common/transport/rvmgcpstack.c 1143:
> RECEIVE: RQNT 6 aaln/[EMAIL PROTECTED] MGCP 1.0
> X: 2467de76
> R: hd(N)
>
> Aug  1 23:19:27 [3195450] MGCP DEBUG INFO at
mgcp/common/transport/rvmgcpstack.c 1143:
> RECEIVE: RQNT 6 aaln/[EMAIL PROTECTED] MGCP 1.0
> X: 2467de76
> R: hd(N)
>
> Aug  1 23:19:28 [3195450] MGCP DEBUG INFO at
mgcp/common/transport/rvmgcpstack.c 1481:
> SEND: 500 5 Unknown endpoint
>
> Aug  1 23:19:28 [3195450] MGCP DEBUG INFO at
mgcp/common/transport/rvmgcpstack.c 1143:
> RECEIVE: RQNT 6 aaln/[EMAIL PROTECTED] MGCP 1.0
> X: 2467de76
> R: hd(N)
>
> Aug  1 23:19:28 [3195450] MGCP DEBUG INFO at
mgcp/common/transport/rvmgcpstack.c 1143:
> RECEIVE: RQNT 6 aaln/[EMAIL PROTECTED] MGCP 1.0
> X: 2467de76
> R: hd(N)
>
> Aug  1 23:19:28 [3195450] MGCP DEBUG INFO at
mgcp/common/transport/rvmgcpstack.c 1481:
> SEND: 500 5 Unknown endpoint
>
>
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Re: [Asterisk-Users] MySQL Dynamic Extensions

[Tony Wasson]

Darren Nay wrote:
Hello All,

 

I am just looking into Asterisk as a viable voicemail solution for our phone
service.  In order to use it though I will need to make extensions.conf
dynamic (ie. Via MySQL).  Is this possible?
Sure..

Set up the database as you see documented. You can schedule this up to 
once a minutes using crontab.

Setting something like this in your /etc/crontab should do it nicely

*/5 * * * * root /usr/local/sbin/update-voicemail 2>&1 > /var/log/vm.log

NOTE: You can increase the frequency by using */3 or */1. */5 means 
every 5 minutes. */3 means every 3 minutes.

Then make /usr/local/sbin/update-voicemail look like:

  #!/bin/bash
  /path/to/retrieve_extensions_from_mysql.pl
  /usr/sbin/asterisk -rx "extensions reload"
Next, make the script executable
#chmod +x /usr/local/sbin/update-voicemail
This is not totally dynamic, but it ought to be close enough. You could 
make this completely dynamic using a trigger.

Tony Wasson


I've found the following information on this subject:

http://www.voip-info.org/wiki-Asterisk+extensions+from+mysql
 

 

However, this is not a fully dynamic function.  It requires me to pull the
mysql database every so often (presumably via cron) and then restart
asterisk after updating extensions.conf.
 

Is it possible to setup asterisks so that extensions.conf is fully dynamic
via a MySQL database?
 

Thanks for the help!! 

 

Regards,

 

Darren Nay

[EMAIL PROTECTED]  

 


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[Asterisk-Users] New Firefly Beta - with SIP and G.729

[Adam Hart]

I've been sitting on this release for a week so I thought I'd better 
just release it :) Firefly now has SIP but it's still in a beta state. 
If you manage to crash it, send me the hex address of the crash. If you 
find it doesn't work with another SIP phone, let me know and I'll happy 
get it working for you. I'll be interested to hear people's experiences 
behind NATs.

To download the beta version of Firefly: 
http://www.virbiage.com/firefly/download/firefly-dev.exe
(the current stable version of firefly will not have sip or g.729)

G729 support via dll - basically as we all know, G.729 ain't free but 
you can get a free development version from Voiceage (Sipro), so I've 
added support for using that. Download 
http://www.virbiage.com/firefly/download/g729.zip and follow the 
instructions in the Readme. You'll need to agree to their license and 
download their library.

Firefly's Protocol Support now is:

Voip Protocols: SIP, IAX
Codecs: ulaw, alaw, iLBC, GSM, G.729 (via DLL)
Next major feature will be conferencing.

feel free to email me,

   Adam Hart
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[Asterisk-Users] MySQL Dynamic Extensions

[Darren Nay]

Hello All,

 

I am just looking into Asterisk as a viable voicemail
solution for our phone service.  In order to use it though I will need to
make extensions.conf dynamic (ie. Via MySQL).  Is this possible?

 

I've found the following information on this subject:

http://www.voip-info.org/wiki-Asterisk+extensions+from+mysql

 

However, this is not a fully dynamic function.  It
requires me to pull the mysql database every so often (presumably via cron) and
then restart asterisk after updating extensions.conf.

 

Is it possible to setup asterisks so that extensions.conf is
fully dynamic via a MySQL database?

 

Thanks for the help!! 

 

Regards,

 

Darren Nay

[EMAIL PROTECTED]

 

[Asterisk-Users] Need help to format asterisk MGCP packet.

[Duane Cox]

Hello, I am trying to setup a 'gateway' fiber to the home device from wave7optics 
(www.wave7optics.com)

They use a MGCP stack from "RAD?"  and did not code it themselves.  My MGCP 
name-convetion on this device is set to IP, but this stack expects the received MGCP 
packets to be formated as 'aaln/[EMAIL PROTECTED]' (notice the brackets)

I have tried to adjust my extensions.conf and mgcp.conf to compensate, but could not 
get asterisk to send in that format.

Can anyone help?

This is the debug log from the 'gateway' device.
You can see the format that it is sending out, and it is expecting to get the same 
format back.  You can see that asterisk is sending to the device in the format 
(without the brackets) and this gateway device responds with 'endpoint unknown'

***LOG***


Aug  1 23:19:27 [3195449] MGCP DEBUG INFO at mgcp/common/transport/rvmgcpstack.c 1602:
SEND: rsip 14389 aaln/[EMAIL PROTECTED] MGCP 1.0
RM:restart
RD:0

Aug  1 23:19:27 [3195450] MGCP DEBUG INFO at mgcp/common/transport/rvmgcpstack.c 1143:
RECEIVE: RQNT 6 aaln/[EMAIL PROTECTED] MGCP 1.0
X: 2467de76
R: hd(N)

Aug  1 23:19:27 [3195450] MGCP DEBUG INFO at mgcp/common/transport/rvmgcpstack.c 1143:
RECEIVE: RQNT 6 aaln/[EMAIL PROTECTED] MGCP 1.0
X: 2467de76
R: hd(N)

Aug  1 23:19:28 [3195450] MGCP DEBUG INFO at mgcp/common/transport/rvmgcpstack.c 1481:
SEND: 500 5 Unknown endpoint

Aug  1 23:19:28 [3195450] MGCP DEBUG INFO at mgcp/common/transport/rvmgcpstack.c 1143:
RECEIVE: RQNT 6 aaln/[EMAIL PROTECTED] MGCP 1.0
X: 2467de76
R: hd(N)

Aug  1 23:19:28 [3195450] MGCP DEBUG INFO at mgcp/common/transport/rvmgcpstack.c 1143:
RECEIVE: RQNT 6 aaln/[EMAIL PROTECTED] MGCP 1.0
X: 2467de76
R: hd(N)

Aug  1 23:19:28 [3195450] MGCP DEBUG INFO at mgcp/common/transport/rvmgcpstack.c 1481:
SEND: 500 5 Unknown endpoint

 
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[Asterisk-Users] Transparent Switch - PRI / IAX2 / PRI

[Daniel Bichara]

Hi,

I wish * switch calls "transparent" from one port PRI to another * using IAX. If I have a line at extension.conf like this:

_X.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})

I get a connection PRI Q931 message before ringing other side:

> Protocol Discriminator: Q.931 (8)  len=14
> Call Ref: len= 2 (reference 32770/0x8002) (Terminator)
> Message type: CONNECT (7)
> Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0, Exclusive Dchan:
0
>ChanSel: Reserved
>   Ext: 1  Coding: 0   Number Specified   Channel Type: 3
>   Ext: 1  Channel: 1 ]
> Progress Indicator (len= 2) [ Ext: 1  Coding: CCITT (ITU) standard (0) 0: 0
Location: Private network serving the local user (1)
>   Ext: 1  Progress Description: Called equipment i
s non-ISDN. (2) ]
-- IAX2[192.168.110.2:4569]/1 stopped sounds
-- IAX2[192.168.110.2:4569]/1 is ringing

How could I deal with this signalization? I wish to receive CONNECT after Answered.

-- Zap/1-1 answered IAX2[[EMAIL PROTECTED]:4569]/1

Thanks in advance.

Daniel




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Re: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call

[James Sizemore]

Some firewalls when doing nat will alter the return address (need to 
make nat work)
but not recalculate the header checksum,  (Sonic walls come to mind.), 
Linux  will
proply delete any tcp/udp packet that fails its checksum at the kernel 
level, and send
an error to the app.  If this is happening to you Asterisk should log 
some kind of error.

AstGrp wrote:

Update...

I did some more testing today.. And with the same setup but one box
behind a Linksys router and another box behind a Pix firewall.. The
linksys works with no problems... So it appears to be how the PIX is
handling NAT & SIP...  If any one has any thoughts on this , it would be
greatly appreciated.
And thank you James for the support you have given today.

Thanks,

gcc

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of AstGrp
Posted At: Friday, March 12, 2004 4:29 PM
Posted To: Asterisk User Group
Conversation: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum
retries exceeded on call
Subject: RE: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum
retries exceeded on call
Do I need to associate the outside interface of the PIX with the phone
on the inside.. I don't remember doing this before...
Setup 

* Server ---> PIX FW ---> WWW CLOUD > PIX FW ---> IP Phone

Again the only difference than before is the First PIX FW Old setup
was (Different server though)
* Server > Linksys Router > WWW CLOUD > PIX FW > IP
Phone
Any thoughts?

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of James
Sizemore Posted At: Friday, March 12, 2004 2:58 PM Posted To: Asterisk
User Group
Conversation: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum
retries exceeded on call
Subject: Re: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum
retries exceeded on call
The pings are pinging the out side port on the nat device,  You don't 
have a
rule in your nat table to associate it with a device on the inside.  You

should
reset the phone and then see if the qualify shows a return time.  You
will need to make the phone register every time you change you config
till the qualify shows a time. A good way to do this is to reboot the
phone. Your nat device will have a default time that it keep nat rules
in its 
table.
Your qualify time will need to be lower then this value.

AstGrp wrote:

 



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RE: [Asterisk-Users] Asterisk & Hylafax

[David Goldfein]

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rémi Letot
Sent: Monday, March 15, 2004 2:48 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Asterisk & Hylafax

Ignace CARIA <[EMAIL PROTECTED]> writes:

> Hi,
> I'd like to install ON THE SAME Machine, Asterisk and Hylafax.
>
> On http://www.voip-info.org/, I've found this:
>
> If you have an ISDN card supporting CAPI, there is a driver that
> connects the Hylafax  fax server directly to an Asterisk
> ISDN connection.
> (http://www.voip-info.org/wiki-Asterisk+and+faxes)
>
> Is somebody can help me to deploy both on the same machine (Asterisk
> is already running...impressive ;-) )

Don't know if it still applies, anyway here it is.

I have * and hylafax running on the same machine. One extension is
reserved for fax, the others are voice. I just configured chan_capi to
manage all voice extensions, and capi4hylafax on the fax extension.

Everything is fine, the capi layer isolates all those applications.



-- 
Rémi aka hobbes
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[Asterisk-Users] Merlin Legend trunk ports

[Steve Creel]

Our telco provided dialtone to the building by way of an Adtran Total
Access 624:

[Telco]--T1--[Adtran 624]--Loop Start w/ disconnect--[Legend]

We have been slowly integrating asterisk, and are currently using this
setup:
[Telco]--T1--[Asterisk]--T1--[Adtran]--Kewl Start--[Legend]

So now we can do a fair amount of upstream switching before the calls go
into the legend (or out to the telco).  For the past several days, I've
been fighting disconnect problems (or more specifically, the lack of
disconnect).  Asterisk would receive the upstream disconnect from the
telco, drop the channel on the telco T1, drop the channel on the Adtran
T1, but the Legend would bring the channel right back up.  It looked like
the Legend was just not honoring the disconnect request.  After a little
bit of rummaging, it turns out that the Legend wants to see a 900ms
disconnect signal.  Making the change in zaptel.h took care of the
problem.

(Found in the Merlin Legend "Pocket Reference" at
http://www.nyteldirect.com/pdf/MLPocketRef.pdf)

Not that I expect anyone else to run into the same thing, but if you do,
hopefully the information is useful to you.

Steve
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Re: [Asterisk-Users] Truncated Tcp Options?

[Rich Adamson]

Ops, wrong list... sorry for the noise...


> Can someone help me understand the following alert? (snort v2.1, current rules)
> 
> snort: [116:55:1] (snort_decoder): Truncated Tcp Options {TCP} 215.144.24.31:80 -> 
> 10.10.91.29:1354
> 
> This is from outside -> inside (a response packet). Seems thus far to only
> be associated with one workstation (or mabe a single remoe site).
> 
> Not sure how to deal with this. Suggestions? 
> 
> Rich
> 
> 
> 
> 
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---End of Original Message-


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Re: [Asterisk-Users] Asterisk & Hylafax

[Rémi Letot]

Ignace CARIA <[EMAIL PROTECTED]> writes:

> Hi,
> I'd like to install ON THE SAME Machine, Asterisk and Hylafax.
>
> On http://www.voip-info.org/, I've found this:
>
> If you have an ISDN card supporting CAPI, there is a driver that
> connects the Hylafax  fax server directly to an Asterisk
> ISDN connection.
> (http://www.voip-info.org/wiki-Asterisk+and+faxes)
>
> Is somebody can help me to deploy both on the same machine (Asterisk
> is already running...impressive ;-) )

Don't know if it still applies, anyway here it is.

I have * and hylafax running on the same machine. One extension is
reserved for fax, the others are voice. I just configured chan_capi to
manage all voice extensions, and capi4hylafax on the fax extension.

Everything is fine, the capi layer isolates all those applications.



-- 
Rémi aka hobbes
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Re: [Asterisk-Users] Hylafax integration

[Rémi Letot]

Alessio Focardi <[EMAIL PROTECTED]> writes:

> Hi,
>
> I have a working install of asterisk that I would like to integrate
> with Hylafax: an asterisk extention must be transfered to Hyla for fax
> receiving.

Sorry for the late answer, real life got in the way lately.

As you talk about isdn, I guess you are using kapejod's capi channel
with asterisk. If the fax extension is distinct from voice extensions,
just use capi4hylafax on the fax extension and chan_capi on voice
extensions. I use that at work and it works like a charm.



-- 
Rémi aka hobbes
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[Asterisk-Users] oh323 sending tech-prefix??

[Anthony Law]

Hi,

> exten => _1613XXX,1,Dial,OH323/[EMAIL PROTECTED]

>This should work. Provide a more detailed Asterisk log to trace the
>problem.

Sorry. It was my mistake. I believe it works now

Could * at the same time take h323 packets from let say a cisco gateway and
pass it through another softswitch?
I have tried setting up cisco gateway (using h323)to voip packets to * but I
instantely got disconnected and gives out below error. Can you please tell
me what I am doing wrong.

Mar 15 16:22:35 WARNING[18451]: channel.c:1856 ast_channel_make_compatible:
No path to translate from H323:1629(256) to H323:19928(1)
Mar 15 16:22:35 WARNING[18451]: channel.c:2245 ast_channel_bridge: Can't
make H323:1629 and H323:19928 compatible
Mar 15 16:22:35 WARNING[18451]: res_parking.c:226 ast_bridge_call: Bridge
failed on channels H323:1629 and H323:19928

Should the setup be done within oh323.conf? Is there documentation on how to
do it?





Regards,



Anthony

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Re: [Asterisk-Users] WiSip SIP settings locked?

[Andrew Kohlsmith]

> I *thought* I had read that some of you were using it with asterisk, but
> so far I can't figure out how to get the SIP settings changed. . .
> changing them causes the phone to reboot, and when it does the settings
> are back pointing at FWD. . .

Yup -- The trick is to set its TFTP server OFF of FWD's using the builtin 
web client.  Once I changed that my settings stopped getting overwritten.  
Not very cool on the part of Pulver Innovations.

I have since pointed it off at my own TFTP server and tried using the config 
file that Pulver was sending over, just with my own settings.  The phone 
contacts my TFTP server just fine but the settings are silently ignored.  
Not sure what's going on there.

Regards,
Andrew
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[Asterisk-Users] Truncated Tcp Options?

[Rich Adamson]

Can someone help me understand the following alert? (snort v2.1, current rules)

snort: [116:55:1] (snort_decoder): Truncated Tcp Options {TCP} 215.144.24.31:80 -> 
10.10.91.29:1354

This is from outside -> inside (a response packet). Seems thus far to only
be associated with one workstation (or mabe a single remoe site).

Not sure how to deal with this. Suggestions? 

Rich




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RE: [Asterisk-Users] ZapRAS over IAX anyone?

[Eric Wieling]

One would assume that ZapRAS only works on Zap channels.  ZapRAS only
works with ISDN DATA calls, not modem calls.

On Mon, 2004-03-15 at 15:13, Bisker, Scott (7805) wrote:
> Make that could not turn up in google.
> 
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Bisker, Scott
> (7805)
> Sent: Monday, March 15, 2004 4:07 PM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] ZapRAS over IAX anyone?
> 
> 
> I'm just pinging the list for some quick info that I could turn up in google.  Has 
> anyone played with doing ZapRAS over an IAX channel?  i.e. call comes in T-1 to 
> server 1.  Server 1 sends call to server 2 via IAX.  Server 2 picksup call with 
> ZapRAS, runs ppp... etc.   I don't see why this would be a major issue, just 
> checking to see if it's been done before.  
> 
> Thanks,
> 
> -sb
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-- 
For Asterisk PBX related documentation go to
http://www.digium.com/index.php?menu=documentation and look at the
"Unofficial Links" section also see
http://www.voip-info.org/wiki-Asterisk also see my site at
http://www.fnords.org/~eric/asterisk/

BTEL Consulting

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[Asterisk-Users] WiSip SIP settings locked?

[Brian Capouch]

I got my new WiSip, and it works for the most part with FWD.

I *thought* I had read that some of you were using it with asterisk, but 
so far I can't figure out how to get the SIP settings changed. . . 
changing them causes the phone to reboot, and when it does the settings 
are back pointing at FWD. . .

Could someone illuminate me?

Thx.

B.
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Re: [Asterisk-Users] extensions problem

[Jon Lawrence]

On Monday 15 March 2004 16:26, Asterisk DEV. Mailing List wrote:
> Your phone supports call waiting, so isn't giving out busy.  I had the
> same problem with a budgetone 102, you can't turn this off on the phone
> but you can work round it by adding
>
> Incominglimit=1
>
> Into the sip.conf entry for the phone

I can imagine situations where call waiting might be useful, but only if I can 
acknowledge the call with the phone either rejecting it to a queue or 
ditching the current call and picking up the incoming one - something to play 
with in the future (once I've found a way of getting UK callerID working).
I've added the Incominglimit=1 and that's fixed my immediate problem.

Thanks everyone.
Jon

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Re: [Asterisk-Users] extensions problem (SIP)

[Walker Haddock]

On Mon, Mar 15, 2004 at 09:56:04PM +0100, Olle E. Johansson wrote:
> Walker Haddock wrote:
> >On Mon, Mar 15, 2004 at 09:28:00PM +0100, Olle E. Johansson wrote:
> >
> >>The incominglimit limits how many simultaneous calls a UA may place to 
> >>Asterisk.
> >
> >I'm pretty sure that the incominglimit specifies how many calls that * can 
> >send to the SIP device.  If you set incominglimit=1 and then do a SIP SHOW 
> >INUSE from the *CLI then you will see the limit set.  The behavior of * 
> >then will consider the device busy if there is a call in progress and the 
> >inuse count is incremented.
> >
> >Paul Lieu did some work on this a few months ago and I've been using it on 
> >my Cisco 7960 and Grandstream BT-102 phones.

See bug (closed),
http://bugs.digium.com/bug_view_page.php?bug_id=408

This is the one that Paul Liew opened and provide a fix for.

> 
> The incominglimit= config is read in build_user, users place calls to 
> asterisk.
> Peers have no incominglimit.
> 
> Funny enough outgoinglimit= was also coded in build_user for users, even 
> though
> chan_sip place calls to peers.
> 
> So maybe it just happens to work as you say for "friends" that is both user 
> and peer.

Yes, I have my SIP UAs defined as `friends`.  Also, like Paul said in bug 1064, you 
have to put the username=xxx in the stanza as well.

> The find_user routine just checks users, not peers.
> 
> Would be greatful if someone cleaned up this part of chan_sip and added 
> support
> for outgoinglimit for peers.
> 
> Also, see bug
> http://bugs.digium.com/bug_view_page.php?bug_id=0001064
I see that Paul is commenting on this, so I don't need to get involved except to say 
that I am using the feature to limit the number of calls that * can make to the UA.  
If there is a disagreement in definition/perspective here, I guess we can change all 
of our sip.conf entries from incoming to outgoing easily enough.

Walker
-- 
   DataCrest, Inc. -- Technically Superior   **
Walker Haddock   http://www.datacrest.com
DataCrest, Inc.e-mail:  [EMAIL PROTECTED]
1634A Montgomery Hwy.phone:  1-888-941-3282, 1-205-335-8589
Birmingham, AL 35216  fax:  1-205-823-7838
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RE: [Asterisk-Users] ZapRAS over IAX anyone?

[Bisker, Scott (7805)]

Make that could not turn up in google.


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Bisker, Scott
(7805)
Sent: Monday, March 15, 2004 4:07 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] ZapRAS over IAX anyone?


I'm just pinging the list for some quick info that I could turn up in google.  Has 
anyone played with doing ZapRAS over an IAX channel?  i.e. call comes in T-1 to server 
1.  Server 1 sends call to server 2 via IAX.  Server 2 picksup call with ZapRAS, runs 
ppp... etc.   I don't see why this would be a major issue, just checking to see if 
it's been done before.  

Thanks,

-sb
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Re: [Asterisk-Users] extensions problem (SIP)

[Walker Haddock]

On Mon, Mar 15, 2004 at 08:56:17PM +, Jon Lawrence wrote:
> 
> The interface to my handytone is identical to a BT-102 so it may also work 
> with the handytone :). Where did you specify incominglimit=1 - is it in the 
> sip.conf for that UA ?
Yes, put it in the stanza for the devicd.  As Olle just pointed out, make sure the 
device is a `friend`

Walker
-- 
   DataCrest, Inc. -- Technically Superior   **
Walker Haddock   http://www.datacrest.com
DataCrest, Inc.e-mail:  [EMAIL PROTECTED]
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Birmingham, AL 35216  fax:  1-205-823-7838
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[Asterisk-Users] ZapRAS over IAX anyone?

[Bisker, Scott (7805)]

I'm just pinging the list for some quick info that I could turn up in google.  Has 
anyone played with doing ZapRAS over an IAX channel?  i.e. call comes in T-1 to server 
1.  Server 1 sends call to server 2 via IAX.  Server 2 picksup call with ZapRAS, runs 
ppp... etc.   I don't see why this would be a major issue, just checking to see if 
it's been done before.  

Thanks,

-sb
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Re: [Asterisk-Users] extensions problem (SIP)

[Jon Lawrence]

On Monday 15 March 2004 20:35, Walker Haddock wrote:
> On Mon, Mar 15, 2004 at 09:28:00PM +0100, Olle E. Johansson wrote:
> > The incominglimit limits how many simultaneous calls a UA may place to
> > Asterisk.
>
> I'm pretty sure that the incominglimit specifies how many calls that * can
> send to the SIP device.  If you set incominglimit=1 and then do a SIP SHOW
> INUSE from the *CLI then you will see the limit set.  The behavior of *
> then will consider the device busy if there is a call in progress and the
> inuse count is incremented.
>
> Paul Lieu did some work on this a few months ago and I've been using it on
> my Cisco 7960 and Grandstream BT-102 phones.

The interface to my handytone is identical to a BT-102 so it may also work 
with the handytone :). Where did you specify incominglimit=1 - is it in the 
sip.conf for that UA ?

Jon

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Re: [Asterisk-Users] extensions problem (SIP)

[Olle E. Johansson]

Walker Haddock wrote:
On Mon, Mar 15, 2004 at 09:28:00PM +0100, Olle E. Johansson wrote:

The incominglimit limits how many simultaneous calls a UA may place to 
Asterisk.
I'm pretty sure that the incominglimit specifies how many calls that * can send to the SIP device.  If you set incominglimit=1 and then do a SIP SHOW INUSE from the *CLI then you will see the limit set.  The behavior of * then will consider the device busy if there is a call in progress and the inuse count is incremented.

Paul Lieu did some work on this a few months ago and I've been using it on my Cisco 7960 and Grandstream BT-102 phones.
The incominglimit= config is read in build_user, users place calls to asterisk.
Peers have no incominglimit.
Funny enough outgoinglimit= was also coded in build_user for users, even though
chan_sip place calls to peers.
So maybe it just happens to work as you say for "friends" that is both user and peer.
The find_user routine just checks users, not peers.
Would be greatful if someone cleaned up this part of chan_sip and added support
for outgoinglimit for peers.
Also, see bug
http://bugs.digium.com/bug_view_page.php?bug_id=0001064
/O
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[Asterisk-Users] "Click to Call" Perl CGI script - TACI

[Tony Wasson]

I've hacked up a little "click to call" web CGI in Perl using the 
Asterisk Manager Interface and Net::Telnet. I've called it TACI - 
Trivial Asterisk Call-generator Interface.

http://www.azxws.com/asterisk/

This is far from perfect or finished, but it should be a start for 
anyone looking to make "clickable" URLs that spawn phone calls. It has a 
rudementary HTML interface to make a call. You can also make calls in a 
URL (providing a real click to call). Finally, you can use it from the 
command line for debugging and other purposes.

You'll need to specify a context, SIP account, and extension to call. 
This seems to work well with the 100 or so calls I've tested.

This should be easy to extend for anyone using IAX also.(It is hardcoded 
for SIP on 1 line). The README should get anyone interested in doing 
this going.

Hope this of use to you!
Tony Wasson
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Re: [Asterisk-Users] extensions problem (SIP)

[Walker Haddock]

On Mon, Mar 15, 2004 at 09:28:00PM +0100, Olle E. Johansson wrote:
> The incominglimit limits how many simultaneous calls a UA may place to 
> Asterisk.
I'm pretty sure that the incominglimit specifies how many calls that * can send to the 
SIP device.  If you set incominglimit=1 and then do a SIP SHOW INUSE from the *CLI 
then you will see the limit set.  The behavior of * then will consider the device busy 
if there is a call in progress and the inuse count is incremented.

Paul Lieu did some work on this a few months ago and I've been using it on my Cisco 
7960 and Grandstream BT-102 phones.

Walker
-- 
   DataCrest, Inc. -- Technically Superior   **
Walker Haddock   http://www.datacrest.com
DataCrest, Inc.e-mail:  [EMAIL PROTECTED]
1634A Montgomery Hwy.phone:  1-888-941-3282, 1-205-335-8589
Birmingham, AL 35216  fax:  1-205-823-7838
***
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RE: [Asterisk-Users] Conference call?

[Justin Carlson]

thanks, I will look into the phones further :-)

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Greg
Retkowski
Sent: Monday, March 15, 2004 2:15 PM
To: [EMAIL PROTECTED]
Cc: 'Greg Retkowski'
Subject: RE: [Asterisk-Users] Conference call?


Not familiar with the snom's, but most IP phones implement 'conference'
internally.. It's a function of the phone independent of the PBX. However
if you set up conferencing on asterisk you can transfer someone into the
conference then dial it up yourself.. transfer is '#' followed by your
conference extension.

-- Greg

Greg Retkowski / I.T. Infrastructure Consultant   /)/|//`
[EMAIL PROTECTED]  http://www.rage.net/~greg/ C:408-455-3913 /|/ /_/


On Mon, 15 Mar 2004, Justin Carlson wrote:

> do you know if there is a way to get the conference button on the snom
> 200's to work?
>
> -Original Message-
> From: Greg Retkowski [mailto:[EMAIL PROTECTED]
> Sent: Monday, March 15, 2004 1:31 PM
> To: [EMAIL PROTECTED]; Justin Carlson
> Subject: Re: [Asterisk-Users] Conference call?
>
>
>
> On Mon, 15 Mar 2004, Justin Carlson wrote:
>
> > how do I setup cal conferencing? and get three-way calling going?  I am
> just
> > looking for some direction as I am having difficulty deciding where to
> > start.
>
> Look at the meetme application, you have to configure meetme.conf, and
> have some meetme() commands in your extensions.conf. You will also need a
> timing device, such as a digium card. If you don't have any hardware
> device you'll want to install a software timing device, such as
> kernel module 'ztdummy'. Hope this helps!!
>
> -- Greg
>
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Re: [Asterisk-Users] extensions problem (SIP)

[Olle E. Johansson]

Jon Lawrence wrote:

Surely * should know if a phone is in use ? After all it initiated/took part 
in the call in the first place ;)
Again, the SIP device is not a "slave" device. It could receive a call from
somewhere else and be busy without Asterisk having a clue. A lot of SIP UAs,
like Xten software and the  SNOM 200 phone support multiple SIP accounts.
So Asterisk could keep track of how many simultaneous calls it can place to
the UA, but not decide if the UA is busy or not. The outgoinglimit code
is used for this, but it's disabled and non-functional right now.
The incominglimit limits how many simultaneous calls a UA may place to Asterisk.

/Olle
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[Asterisk-Users] AgentCallBackLogin ??

[AstGrp]

I could use a little assistance.. I am sure I am doing something
stupid. The problem I am having is when the call comes in and runs
the context [411].  The call is generated, but never makes the call.  It
rings back the user who is making the call.  It works fine if I dial
context [411] from the inside.  It sounds like I need to add some
context somewhere just not sure what where?

[agents]
agent => 4001,4001,Geoff Clark

[general]
[default]
[tech]
member => Agent/4001
strategy = roundrobin
timeout = 30
retry = 10


[411]
exten => 411,1,Answer
exten => 411,2,Wait,2
exten => 411,3,Background(auth-thankyou)
exten => 411,4,Queue(tech)
exten => *6,1,AgentCallbackLogin(@411)
exten => *4001,1,Dial(${TRUNK}/${GCELL:${TRUNKMSD}})

Thanks,

gcc
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RE: [Asterisk-Users] Conference call?

[Greg Retkowski]

Not familiar with the snom's, but most IP phones implement 'conference'
internally.. It's a function of the phone independent of the PBX. However
if you set up conferencing on asterisk you can transfer someone into the
conference then dial it up yourself.. transfer is '#' followed by your
conference extension.

-- Greg

Greg Retkowski / I.T. Infrastructure Consultant   /)/|//`
[EMAIL PROTECTED]  http://www.rage.net/~greg/ C:408-455-3913 /|/ /_/


On Mon, 15 Mar 2004, Justin Carlson wrote:

> do you know if there is a way to get the conference button on the snom
> 200's to work?
>
> -Original Message-
> From: Greg Retkowski [mailto:[EMAIL PROTECTED]
> Sent: Monday, March 15, 2004 1:31 PM
> To: [EMAIL PROTECTED]; Justin Carlson
> Subject: Re: [Asterisk-Users] Conference call?
>
>
>
> On Mon, 15 Mar 2004, Justin Carlson wrote:
>
> > how do I setup cal conferencing? and get three-way calling going?  I am
> just
> > looking for some direction as I am having difficulty deciding where to
> > start.
>
> Look at the meetme application, you have to configure meetme.conf, and
> have some meetme() commands in your extensions.conf. You will also need a
> timing device, such as a digium card. If you don't have any hardware
> device you'll want to install a software timing device, such as
> kernel module 'ztdummy'. Hope this helps!!
>
> -- Greg
>
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RE: [Asterisk-Users] Pri Errors, Hanging up Owner

[Bisker, Scott (7805)]

Title: Pri Errors, Hanging up Owner



I had 
the same problem a few weeks ago.  I updated to latest zaptel and libpri, 
and the problem went away.  My date is 3/8/04
 
-sb
 
 

  -Original Message-From: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED]On Behalf Of Matthew 
  BrantonSent: Monday, March 15, 2004 2:48 PMTo: 
  Asterisk-Users (E-mail)Subject: [Asterisk-Users] Pri Errors, 
  Hanging up Owner
  Hey guys, 
  Every so often my pri channels degenerate into a 
  non stop series of Mar 15 06:51:50 
  WARNING[131081]: chan_zap.c:6263 pri_dchannel: Ring requested on channel 1 
  already in use on span 1.  Hanging up owner.
  Errors. Anyone else having this problem? I see an 
  old reference to updating your cvs, I am using a fairly updated version, as of 
  say a week ago. Anyone have any experience with this / knows what the problem 
  is?
  Matt 

Re: [Asterisk-Users] Low cost VOIP phone with headset possibility

[George Pajari]

ZyXEL's product looks like a relabelled Pulver WiSIP phone
(http://www.pulverinnovations.com/wisip.html) which sells in the US for
US$250.

g.
- Original Message - 
From: "Peer Oliver schmidt" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Sunday, March 14, 2004 7:50 AM
Subject: Re: [Asterisk-Users] Low cost VOIP phone with headset possibility


> Adam Goryachev wrote:
>
> >>Following Zyxel phone is VERY nice and you may attach a headset to it
> >>and walk around to your hearts contend, as look as you are
> >>anywhere near a WIFI AP.
>
>>http://www.zyxel.com/product/model.php?indexcate=1075688089&indexFlagvalue
=1075687935
>
> > Sounds interesting... just need to find out the cost next... :)
>
> Acording to a press release about 350 EURO.
>
> rgds
> pos

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RE: [Asterisk-Users] Conference call?

[Justin Carlson]

do you know if there is a way to get the conference button on the snom 200's
to work?

-Original Message-
From: Greg Retkowski [mailto:[EMAIL PROTECTED]
Sent: Monday, March 15, 2004 1:31 PM
To: [EMAIL PROTECTED]; Justin Carlson
Subject: Re: [Asterisk-Users] Conference call?



On Mon, 15 Mar 2004, Justin Carlson wrote:

> how do I setup cal conferencing? and get three-way calling going?  I am
just
> looking for some direction as I am having difficulty deciding where to
> start.

Look at the meetme application, you have to configure meetme.conf, and
have some meetme() commands in your extensions.conf. You will also need a
timing device, such as a digium card. If you don't have any hardware
device you'll want to install a software timing device, such as
kernel module 'ztdummy'. Hope this helps!!

-- Greg

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Re: [Asterisk-Users] asterisk MySQL

[WipeOut]

Joao Carlos Moura wrote:

I need to develop an web interface to include clients automatically in
Asterisk. So, to make this possible I need
that all my peers and exten being at a database (Mysql).
Where do I find doc´s regarded for it?
Thank you very much,
J Moura
 

I think MySQL friends is probably what you are after.. take a look at 
www.voip-info.org

Later..

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[Asterisk-Users] Pri Errors, Hanging up Owner

[Matthew Branton]

Title: Pri Errors, Hanging up Owner





Hey guys,


Every so often my pri channels degenerate into a non stop series of
Mar 15 06:51:50 WARNING[131081]: chan_zap.c:6263 pri_dchannel: Ring requested on channel 1 already in use on span 1.  Hanging up owner.

Errors. Anyone else having this problem? I see an old reference to updating your cvs, I am using a fairly updated version, as of say a week ago. Anyone have any experience with this / knows what the problem is?



Matt

Re: [Asterisk-Users] Panther OS X Installation

[Vijay Vaidyanathan]

I am playing with it on my G4/Jaguar (10.2, I still havent upgraded to 10.3)
and have gotten it to compile and tried some basic things with it,
but havent spent much time on it. However, my install is a VOIP-only
configuration, and I would guess that getting the drivers for the
cards to work on OS X may be something more of a challenge.

Contact me directly if you have any questions on getting to to work
in a VOIP-only configuration - I dont recall what I had to change to
get it to compile, but it was pretty straightforward.

- Vijay
==

On Saturday, March 13, 2004, at 10:09  AM, Phillip Jackson wrote:

I am interested In running Asterisk on an Apple G5 w/ Panther 10.3.  Is this possible, and are there any *good* how-to’s regarding such an installation?  I’ve installed successfully Asterisk on many Linux machines in the past, but am having difficulty with my Apple.

 

Cheers,

Phillip

Re: [Asterisk-Users] extensions problem (SIP)

[Jon Lawrence]

On Monday 15 March 2004 16:00, Olle E. Johansson wrote:
>
> It depends on your SIP device. Asterisk places the call to your SIP device
> regardless, since by SIP protocol design the UA is not a "slave", it is
> free. So the SIP ua must answer "busy" for Asterisk to understand that
> you're busy. If not, the call is placed to you and Asterisk has no
> knowledge that you are busy. Check you SIp phone if you can limit the
> number of concurrent calls.

So does anyone know if the Grandstream handytone-286 sends this "busy" answer 
?
I'm guessing it doesn't. In that case, what other ways are there of connecting 
my dect phones to a voip * system ? - can I just connect them into the 
x100p's phone socket (how do I send calls to that port) or do I need to get a 
fxs card and run wire's everywhere  - not an option :(
How does everyone else connect up DECT phones to a * based system.

Surely * should know if a phone is in use ? After all it initiated/took part 
in the call in the first place ;)

Jon

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Re: [Asterisk-Users] dbinit error message

[Tilghman Lesher]

On Monday 15 March 2004 13:07, Derek Barber wrote:
> I've been noticing the following error message on the asterisk
> console:
>
> Mar 15 10:49:13 WARNING[163851]: db.c:46 dbinit: Unable to open
> Asterisk database
>
> Any clues on what this could be?

You aren't running Asterisk as root and the directory where Asterisk
stores its database isn't writable.

-Tilghman

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Re: [Asterisk-Users] Conference call?

[Greg Retkowski]

On Mon, 15 Mar 2004, Justin Carlson wrote:

> how do I setup cal conferencing? and get three-way calling going?  I am just
> looking for some direction as I am having difficulty deciding where to
> start.

Look at the meetme application, you have to configure meetme.conf, and
have some meetme() commands in your extensions.conf. You will also need a
timing device, such as a digium card. If you don't have any hardware
device you'll want to install a software timing device, such as
kernel module 'ztdummy'. Hope this helps!!

-- Greg
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[Asterisk-Users] Conference call?

[Justin Carlson]

how do I setup cal conferencing? and get three-way calling going?  I am just
looking for some direction as I am having difficulty deciding where to
start.

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[Asterisk-Users] dbinit error message

[Derek Barber]

I've been noticing the following error message on the asterisk console:

Mar 15 10:49:13 WARNING[163851]: db.c:46 dbinit: Unable to open Asterisk
database

Any clues on what this could be?

derek

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RE: [Asterisk-Users] Cisco Call Manager and Asterisk and fastbusy?

[Kyle Stone]

I got it fixed.. I was routing the calls from * to CCM out the PTSN and
back into the gateway to the CCM.. It was kind freaking CCM out.  :)

Kyle


-Original Message-
From: Dave Packham [mailto:[EMAIL PROTECTED] 
Sent: Monday, March 15, 2004 1:25 PM
To: [EMAIL PROTECTED]; Kyle Stone
Cc: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Cisco Call Manager and Asterisk and
fastbusy?

can we get a copy of your saved configs?

Dave P



>>> [EMAIL PROTECTED] 3/15/2004 10:12:10 AM >>>
I gave my test * server away, so I can no longer test it, but I do
have
copies of
my configs.  I did not make extensive changes to get it to work.  I
was
using a
slightly older oh323 release (0.5.6), but I don't know that would make
a

difference.

I'd be glad to try an help further, but I'd need more information
about
your setup.
Are the * server and CCM on the same subnet?
What CSS is assigned to the CCM gateway?
What frame size is CCM using for G711?  Is * using the same?
Did you make any changes to FastStart/h245 tunneling options in
oh323.conf ?

We can also take the discussion off list if no-one else would be
interested.

Dan

-Original Message-
From: Kyle Stone [mailto:[EMAIL PROTECTED] 
Sent: Monday, March 15, 2004 7:17 AM
To: Dan Austin
Cc: [EMAIL PROTECTED] 
Subject: RE: [Asterisk-Users] Cisco Call Manager and Asterisk and fast
busy?

I tried that, to no avail.  It does the same thing.. Places the call,
and then the CCM hangsup 1.5 rings into it.

.


-Original Message-
From: Dan Austin [mailto:[EMAIL PROTECTED] 
Sent: Friday, March 12, 2004 8:18 PM
To: Kyle Stone
Cc: [EMAIL PROTECTED] 
Subject: RE: [Asterisk-Users] Cisco Call Manager and Asterisk and fast
busy?

Check the codecs allowed.  Cisco supports most, but in my tests I
limited
myself to G.711U.

Another possibility, and one I seem to remember having is that the IP 
address of the gateway did not match the * server, and as such did not
have access to the correct Calling Search Space and Media Resource
Groups.

Still codec choice seems the most likely.

Dan

-Original Message-
From: Kyle Stone [mailto:[EMAIL PROTECTED] 
Sent: Friday, March 12, 2004 6:40 AM
To: Dan Austin
Cc: [EMAIL PROTECTED] 
Subject: RE: [Asterisk-Users] Cisco Call Manager and Asterisk and fast
busy?



I got the latest oh323, and the libraries.  I got everything installed
and working.. somewhat.   I setup netmeeting and can call IAXcomm
extensions I setup. I also can place calls in the reverse.  When I use
either to route a call to CCM, setup as per you described.. The
CCM/skinny phone will ring one and a half times and hangup.  If
answered
before the end of the ringing I get a fast busy.   Any ideas?


Kyle Stone


-Original Message-
From: Dan Austin [mailto:[EMAIL PROTECTED] 
Sent: Monday, March 08, 2004 12:04 PM
To: [EMAIL PROTECTED] 
Subject: RE: [Asterisk-Users] Cisco Call Manager and Asterisk?

In CCM add a Gateway.  Use H.323 with H.225 as the device protocol.
Next add a route pattern to identify which calls to direct to *.

Lastly use chan_oh323 instead of chan_h323, as the former works
with CCM and the later does not (one way audio).

The setup is extremely easy and works just fine, with the correct
channel that is.

Dan






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RE: [Asterisk-Users] extensions problem (SIP)

[Bisker, Scott (7805)]

In your SIP.conf set callwaiting = no.  This will work for single registrations.  If 
you have multiple call appearance on you phone, then it will just ring to the second 
line (e.g. Cisco 7960).  If you only have a single registration, then you should be 
fine.

-sb


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Olle E.
Johansson
Sent: Monday, March 15, 2004 11:01 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] extensions problem (SIP)


Jon Lawrence wrote:
> Hi,
> I've got 2 x100p's installed in my system.
> Both execute the same incoming contexts as follows:
> [inboundA]
> include => dialjon
> [inboundB]
> include => dialjon|09:00-16:30|Mon-Fri|*|*
> 
> [dialjon]
> exten => s,1,answer
> exten => s,2,Dial(SIP/2000,15)
> exten => s,3,Playback(noone)
> exten => s,103,Goto(onphone,s,1)
> 
> 
> Am I right in saying:
> if no one answers at ext 2000 then s,3 is executed.
> if ext 2000 is busy  then 103 is executed.
> 
> If so then sometihng is wrong. If I'm already on a call, I want 103 to be 
> executed however, this isn't happening. If a new call comes in (whilst I'm 
> talking on ext 2000) I here it ringing on my handset.
> 

It depends on your SIP device. Asterisk places the call to your SIP device regardless,
since by SIP protocol design the UA is not a "slave", it is free. So the SIP ua must
answer "busy" for Asterisk to understand that you're busy. If not, the call is placed
to you and Asterisk has no knowledge that you are busy. Check you SIp phone if you can
limit the number of concurrent calls.

There's some code in Asterisk chan_sip.c to limit the number of calls placed to
a SIP phone, but right now it's not working at all.

/Olle
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[Asterisk-Users] Bluetooth

[Serge Oleinikov]

Where i can download bluetooth support for * ?

RE: [Asterisk-Users] Cisco Call Manager and Asterisk and fast busy?

[Dave Packham]

can we get a copy of your saved configs?

Dave P



>>> [EMAIL PROTECTED] 3/15/2004 10:12:10 AM >>>
I gave my test * server away, so I can no longer test it, but I do
have
copies of
my configs.  I did not make extensive changes to get it to work.  I
was
using a
slightly older oh323 release (0.5.6), but I don't know that would make
a

difference.

I'd be glad to try an help further, but I'd need more information
about
your setup.
Are the * server and CCM on the same subnet?
What CSS is assigned to the CCM gateway?
What frame size is CCM using for G711?  Is * using the same?
Did you make any changes to FastStart/h245 tunneling options in
oh323.conf ?

We can also take the discussion off list if no-one else would be
interested.

Dan

-Original Message-
From: Kyle Stone [mailto:[EMAIL PROTECTED] 
Sent: Monday, March 15, 2004 7:17 AM
To: Dan Austin
Cc: [EMAIL PROTECTED] 
Subject: RE: [Asterisk-Users] Cisco Call Manager and Asterisk and fast
busy?

I tried that, to no avail.  It does the same thing.. Places the call,
and then the CCM hangsup 1.5 rings into it.

.


-Original Message-
From: Dan Austin [mailto:[EMAIL PROTECTED] 
Sent: Friday, March 12, 2004 8:18 PM
To: Kyle Stone
Cc: [EMAIL PROTECTED] 
Subject: RE: [Asterisk-Users] Cisco Call Manager and Asterisk and fast
busy?

Check the codecs allowed.  Cisco supports most, but in my tests I
limited
myself to G.711U.

Another possibility, and one I seem to remember having is that the IP 
address of the gateway did not match the * server, and as such did not
have access to the correct Calling Search Space and Media Resource
Groups.

Still codec choice seems the most likely.

Dan

-Original Message-
From: Kyle Stone [mailto:[EMAIL PROTECTED] 
Sent: Friday, March 12, 2004 6:40 AM
To: Dan Austin
Cc: [EMAIL PROTECTED] 
Subject: RE: [Asterisk-Users] Cisco Call Manager and Asterisk and fast
busy?



I got the latest oh323, and the libraries.  I got everything installed
and working.. somewhat.   I setup netmeeting and can call IAXcomm
extensions I setup. I also can place calls in the reverse.  When I use
either to route a call to CCM, setup as per you described.. The
CCM/skinny phone will ring one and a half times and hangup.  If
answered
before the end of the ringing I get a fast busy.   Any ideas?


Kyle Stone


-Original Message-
From: Dan Austin [mailto:[EMAIL PROTECTED] 
Sent: Monday, March 08, 2004 12:04 PM
To: [EMAIL PROTECTED] 
Subject: RE: [Asterisk-Users] Cisco Call Manager and Asterisk?

In CCM add a Gateway.  Use H.323 with H.225 as the device protocol.
Next add a route pattern to identify which calls to direct to *.

Lastly use chan_oh323 instead of chan_h323, as the former works
with CCM and the later does not (one way audio).

The setup is extremely easy and works just fine, with the correct
channel that is.

Dan






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RE: [Asterisk-Users] asterisk MySQL

[Michael Shuler]

http://bugs.digium.com/bug_view_page.php?bug_id=0001086 is what I use to
dynamically put in SIP entries.  It works really well.  

The extensions don't seem to be available in MySQL "officially" but I did
find
http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20DynExtenDB
which I have not tried yet.  It seems to be a bit old though and may cause
problems with the current Asterisk CVS/releases.



Michael Shuler, C.E.O.
BitWise Systems, Inc.
1301 W. Pioneer Parkway
Peoria, IL 61615
Office: (217) 585-0357
Cell: (309) 657-6365
Fax: (309) 213-3500
E-Mail: [EMAIL PROTECTED]
Customer Service: (877) 976-0711 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Joao Carlos Moura
> Sent: Monday, March 15, 2004 11:53 AM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] asterisk MySQL
> 
> 
> I need to develop an web interface to include clients automatically in
> Asterisk. So, to make this possible I need
> that all my peers and exten being at a database (Mysql).
> Where do I find doc´s regarded for it?
> 
> Thank you very much,
> J Moura
> 
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[Asterisk-Users] asterisk MySQL

[Joao Carlos Moura]

I need to develop an web interface to include clients automatically in
Asterisk. So, to make this possible I need
that all my peers and exten being at a database (Mysql).
Where do I find doc´s regarded for it?

Thank you very much,
J Moura

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RE: [Asterisk-Users] Consultants

[Vasyl Rublyov]

Don,

Just mark another number 703 395 0238... we might help
Feel free to call
--
Regards,
  Vasyl Rublyov
  IonIdea, Inc.
  [EMAIL PROTECTED]

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Don Feuer
Sent: Saturday, March 13, 2004 10:05 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Consultants
We would like to look at the feasibility of utilizing * as a network
infrastructure for a unified communications platform.  We would like a
list of consultants that work with * and have either developed a
platform which is easily usable in a true telco environment. =20
The system needs to have the following:  Billing, voice and fax unified
messaging, integration with h323, sip, aix to produce a Vonage type of
service.
Please forward your information to [EMAIL PROTECTED]

Sincerely,

Don Feuer
(949) 279-5290


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[Asterisk-Users] SIP Calls with * and Cisco AS5300

[Brian Rathman]

I am attempting to send calls from various Ip phones (Snom 105,200 and SIP
Express ATA) to a default SIP gateway (Cisco AS5300) and for some reason my
calls are failing 2 seconds after the called party picks up. This is what
the Asterisk console is displaying during the call attempt:

-- Executing Dial("SIP/7708183797-28a7",
"SIP/[EMAIL PROTECTED]|30") in new stack
-- Called [EMAIL PROTECTED]
-- SIP/66.102.15.212-bfdc is making progress passing it to
SIP/7708183797-28a7
-- SIP/66.102.15.212-bfdc answered SIP/7708183797-28a7
-- Attempting native bridge of SIP/7708183797-28a7 and
SIP/66.102.15.212-bfdc
-- Got SIP response 481 "Invalid CSeq Number" back from 66.102.15.212
  == Spawn extension (default, 6783528833, 1) exited non-zero on
'SIP/7708183797-28a7'

At this point I am not sure if it is a problem with the config on my cisco
box, or with the setup in my extension or sip.conf files. The cisco box is
handling SIP calls from another registrar now, so I doubt that it is the
problem.

Has anyone been able to get this setup working?

Thanks,
Brian

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Re[2]: [Asterisk-Users] IPC5000 (WIP-5000 from hitachi cable)

[Masakazu Nakano]

That is fahdtel's product?
http://www.dairiten.com/modules/mylinks/visit.php?lid=217

It looks very cool.

mack_jpn

On Mon, 15 Mar 2004 06:23:22 +
Miguel Cavazos <[EMAIL PROTECTED]> wrote:

> thanx for the review michael, could you send some pictures of the phone?
> can you tell how long does the battery lives? signaling what do the
> menus have how do you configure it etc? maybe after you do a full
> testing we can do a Wisip vs. IPC5000 working futures.
> 
> Miguel Cavazos

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Re: [Asterisk-Users] ISDN BRI with DDI support

[Diego Ercolani]

Il 17:36, lunedì 15 marzo 2004, Maros RAJNOCH ha scritto:
> Hi all,
>
> I want to build in my small company little PBX with asterisk.
> I have one ISDN BRI link with DDI preselection and a couple of analog
> phones.
>
> I need somethink like this
>
>+--+--- phone 1 (extension 1)
>
>| LINUX with ASTERISK  +--- phone 2 (extension 2)
>
> ISDN BRI DDI   |  +--- phone 3 (extension 3)
> ---+  |
>
>|  +--- phone 4 (extension 4)
>
>+--+--- phone 5 (extension 5)
>
>
> So, my problem is, I'm lost in lots of abbreviations, and I have no idea
> which pci card to select.
>
> I know, I need one ISDN BRI DDI card for incoming isdn line
> and some pci cards with RJ11 for analog phones.
>
> 1) can anybody help me to select correct pci cards?
> 2) can i use more then one isdn cards (connected together) if I need
> more the 2 simultaneous phone calls?
>
> THANK you very much.
This is what i've understood
1) Digium TDM40B offer 4 FXS to connect 4 analog phones two TDM cards are 
allowed on the samemachine but not more because of bus
2) No, althought ISDN is a bus, BRI connection can allow only two calls at the 
same time. You have to move to PRI if you want more usable lines.

I'm not totally sure but this would be true enought.
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Re: [Asterisk-Users] Guru's help with * and AVM C2 ISDN - Newbie going mad!!

[Jakob Strebel]

Nick,

you give us no info what you have tested so far and what are the results. I 
guess you have installed chan_capi.

Two things. May be you have checked that
if you start asterisk so you see anything about CAPI?
what do you get in the asterisk CLI when you enter: capi info?

If all this is ok I would make the following change in extensions.conf

( I I had to do it this way in my extensions.conf, even the context name in 
capi.conf was different. I do not understand why but it works)

hope this helps

Jakob

[default]
include -> demo
[demo]
;
; We start with what to do when a call first comes in.
;
exten => s,1,Wait,1 ; Wait a second, just for fun
exten => s,2,Answer ; Answer the line
exten => s,3,DigitTimeout,5 ; Set Digit Timeout to 5 seconds
exten => s,4,ResponseTimeout,10 ; Set Response Timeout to 10 seconds
exten => s,5,BackGround(demo-congrats)  ; Play a congratulatory message
exten => s,6,BackGround(demo-instruct)  ; Play some instructions
 

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RE: [Asterisk-Users] Cisco Call Manager and Asterisk and fast busy?

[Dan Austin]

I gave my test * server away, so I can no longer test it, but I do have
copies of
my configs.  I did not make extensive changes to get it to work.  I was
using a
slightly older oh323 release (0.5.6), but I don't know that would make a

difference.

I'd be glad to try an help further, but I'd need more information about
your setup.
Are the * server and CCM on the same subnet?
What CSS is assigned to the CCM gateway?
What frame size is CCM using for G711?  Is * using the same?
Did you make any changes to FastStart/h245 tunneling options in
oh323.conf ?

We can also take the discussion off list if no-one else would be
interested.

Dan

-Original Message-
From: Kyle Stone [mailto:[EMAIL PROTECTED] 
Sent: Monday, March 15, 2004 7:17 AM
To: Dan Austin
Cc: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Cisco Call Manager and Asterisk and fast
busy?

I tried that, to no avail.  It does the same thing.. Places the call,
and then the CCM hangsup 1.5 rings into it.

.


-Original Message-
From: Dan Austin [mailto:[EMAIL PROTECTED] 
Sent: Friday, March 12, 2004 8:18 PM
To: Kyle Stone
Cc: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Cisco Call Manager and Asterisk and fast
busy?

Check the codecs allowed.  Cisco supports most, but in my tests I
limited
myself to G.711U.

Another possibility, and one I seem to remember having is that the IP 
address of the gateway did not match the * server, and as such did not
have access to the correct Calling Search Space and Media Resource
Groups.

Still codec choice seems the most likely.

Dan

-Original Message-
From: Kyle Stone [mailto:[EMAIL PROTECTED] 
Sent: Friday, March 12, 2004 6:40 AM
To: Dan Austin
Cc: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Cisco Call Manager and Asterisk and fast
busy?



I got the latest oh323, and the libraries.  I got everything installed
and working.. somewhat.   I setup netmeeting and can call IAXcomm
extensions I setup. I also can place calls in the reverse.  When I use
either to route a call to CCM, setup as per you described.. The
CCM/skinny phone will ring one and a half times and hangup.  If answered
before the end of the ringing I get a fast busy.   Any ideas?


Kyle Stone


-Original Message-
From: Dan Austin [mailto:[EMAIL PROTECTED] 
Sent: Monday, March 08, 2004 12:04 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Cisco Call Manager and Asterisk?

In CCM add a Gateway.  Use H.323 with H.225 as the device protocol.
Next add a route pattern to identify which calls to direct to *.

Lastly use chan_oh323 instead of chan_h323, as the former works
with CCM and the later does not (one way audio).

The setup is extremely easy and works just fine, with the correct
channel that is.

Dan






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[Asterisk-Users] asterisk speech in italy

[Andrea Fino]

Hi guys,

I will do a speech of about 50 minutes at Webbit, Padua, Italy in may on 
asterisk.

Webbit it is an important event for italy regarding IT technology, it 
happens once at year, and this year for the first time it will happens 
in three different cities namely Padua, Milan and Bari.

For now I will be in Padua.

So I will be glad to hear any suggestion or hints or any thing you guys 
think could be important in such an event.

Please contact me in any way you like, on list or off list.

Thank you all,
Andrea Fino
--
Andrea Fino 8-) - "Sistemi su misura di qualita' industriale"
 "Handcrafted systems with industrial quality"
[Phone: +39 071 2070104]+[Fax: +39 071 2077919]+[iaxtel:17002876594]
[Web: http://www.faino.org]+[Email: [EMAIL PROTECTED]
-BEGIN GEEK CODE BLOCK-
Version: 3.1
GCS: d- s: a+ C+++ UL P++ L E--- W++ N++ o? K w--- O M V
PS++ PE+ Y+ PGP t+ 5? X-- R* tv- b-- DI+++ D G++ e* h r y+
--END GEEK CODE BLOCK--
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RE: [Asterisk-Users] MusicOnHold

[Brian Mulligan]

Mmm, sorry me being stupid. When I substituted

exten => 6000,1,Answer
exten => 6000,2,MusicOnHold()

instead of;

exten => 6000,1,Answer
exten => 6000,2,MusicOnHold

it worked. In my defense, the former syntax is the one in the Wiki for the
MusicOnHold command.
Brian

> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Tilghman
> Lesher
> Sent: 15 March 2004 16:28
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] MusicOnHold
>
>
> On Monday 15 March 2004 09:29, Brian Mulligan wrote:
> > Despite my best efforts I am unable to get this to work. I have
> > checked that mpg123 is correctly installed and that the symbolic
> > link exists for the binary. I have also made sure that the REdhat9
> > does not have the older version mpg321 installed. When I invoke
> > MusicOnHold from my dialplan i get the following message;
> >
> > Mar 15 15:01:10 WARNING[1225991360]: res_musiconhold.c:416
> > moh_alloc: No class: 30
> > Mar 15 15:01:10 WARNING[1225991360]: res_musiconhold.c:305
> > moh0_exec: Unable to start
> >  music on hold (class '30') on channel Zap/3-1
>
> Please post the contents of your musiconhold.conf and the relevant
> line(s) from extensions.conf.
>
> -Tilghman
>
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RE: [Asterisk-Users] Asterisk & Analog Modems

[Jim Sneeringer]

When I connect a modem to and FXS card and use it to send and receive faxes,
it doesn't work reliably. The faxes are mangled -- stretched or with parts
missing.

I haven't tried transmitting data with the modem, so I cannot address that,
but you should still be aware of the potential for a problem.

In trying to remedy the situation, I have checked for other CPU load on the
2.4 GHz P4 (there was none), tried different channels (in case it might be a
hardware sample defect), turned off echocancel, and boosted rxgain and
txgain, all to no avail. I have also confirmed that everything works fine
when the faxes are not routed through Asterisk. It doesn't matter whether
the fax is being sent or received.

I have been advised to use different fax software (I am using WinFax), but I
haven't gotten around to it yet.

Jim

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tilghman Lesher
Sent: Monday, March 15, 2004 10:25 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Asterisk & Analog Modems

On Monday 15 March 2004 05:09, Ignace CARIA wrote:
> Does it be possible to cnnect an External Analog Modem US Robotics
> Sportster to Asterisk?
>
> If it is, how? Through a FXS card? Serial cable? modem_chan ?

FXS card.

-Tilghman


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RE: [Asterisk-Users] Guru's help with * and AVM C2 ISDN - Newbie going mad!!

[nathan]

Hi Nick,

Unless I mistaken you don't seem to have a default context in
extensions.conf so if the CAPI side of things is working it 
will have no where to send the calls. Try renaming the demo
context to 'default', or change the CAPI config to deliver
calls to the 'demo' context

Also, what does /usr/sbin/asterisk -vc report when you
attempt to call in via one of the ISDN MSNs ?

-Nathan

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[Asterisk-Users] ISDN BRI with DDI support

[Maros RAJNOCH]

Hi all,

I want to build in my small company little PBX with asterisk.
I have one ISDN BRI link with DDI preselection and a couple of analog
phones.

I need somethink like this

   +--+--- phone 1 (extension 1)
   | LINUX with ASTERISK  +--- phone 2 (extension 2)
   |  | 
ISDN BRI DDI   |  +--- phone 3 (extension 3)
---+  |
   |  +--- phone 4 (extension 4)
   |  |
   +--+--- phone 5 (extension 5)


So, my problem is, I'm lost in lots of abbreviations, and I have no idea
which pci card to select.

I know, I need one ISDN BRI DDI card for incoming isdn line
and some pci cards with RJ11 for analog phones.

1) can anybody help me to select correct pci cards?
2) can i use more then one isdn cards (connected together) if I need
more the 2 simultaneous phone calls?

THANK you very much. 
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Re: [Asterisk-Users] MusicOnHold

[Tilghman Lesher]

On Monday 15 March 2004 09:29, Brian Mulligan wrote:
> Despite my best efforts I am unable to get this to work. I have
> checked that mpg123 is correctly installed and that the symbolic
> link exists for the binary. I have also made sure that the REdhat9
> does not have the older version mpg321 installed. When I invoke
> MusicOnHold from my dialplan i get the following message;
>
> Mar 15 15:01:10 WARNING[1225991360]: res_musiconhold.c:416
> moh_alloc: No class: 30
> Mar 15 15:01:10 WARNING[1225991360]: res_musiconhold.c:305
> moh0_exec: Unable to start
>  music on hold (class '30') on channel Zap/3-1

Please post the contents of your musiconhold.conf and the relevant
line(s) from extensions.conf.

-Tilghman

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RE: [Asterisk-Users] extensions problem

[Asterisk DEV. Mailing List]

Your phone supports call waiting, so isn't giving out busy.  I had the
same problem with a budgetone 102, you can't turn this off on the phone
but you can work round it by adding

Incominglimit=1

Into the sip.conf entry for the phone


>From: Jon Lawrence <[EMAIL PROTECTED]>
>To: [EMAIL PROTECTED]
>Date: Mon, 15 Mar 2004 15:29:01 +
>Subject: [Asterisk-Users] extensions problem
>Reply-To: [EMAIL PROTECTED]

>Hi,
>I've got 2 x100p's installed in my system.
>Both execute the same incoming contexts as follows:
>[inboundA]
>include => dialjon
>[inboundB]
>include => dialjon|09:00-16:30|Mon-Fri|*|*
>
>[dialjon]
>exten => s,1,answer
>exten => s,2,Dial(SIP/2000,15)
>exten => s,3,Playback(noone)
>exten => s,103,Goto(onphone,s,1)
>

>Am I right in saying:
>if no one answers at ext 2000 then s,3 is executed.
>if ext 2000 is busy  then 103 is executed.

>If so then sometihng is wrong. If I'm already on a call, I want 103 to
be 
>executed however, this isn't happening. If a new call comes in (whilst
I'm 
>talking on ext 2000) I here it ringing on my handset.

>Can anyone point out where I've gone wrong ?

>TIA
>Jon

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Re: [Asterisk-Users] Asterisk & Analog Modems

[Tilghman Lesher]

On Monday 15 March 2004 05:09, Ignace CARIA wrote:
> Does it be possible to cnnect an External Analog Modem US Robotics
> Sportster to Asterisk?
>
> If it is, how? Through a FXS card? Serial cable? modem_chan ?

FXS card.

-Tilghman

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Re: [Asterisk-Users] DTMF debugs

[Walker Haddock]

On Mon, Mar 15, 2004 at 09:56:31AM -0600, Bruce Marler wrote:
> Is their a debug that will show what DTMF is being detected or what is being
> sent by asterisk. I am having some problems with initial fax tones possibly
> being detected as fax and once I change my DTMF passing type I can then send
> faxes BUT get double DTMF digits sent at times.
> 
> Any debug info would be appreciated.
in /etc/asterisk/logger.conf, either uncomment the line
debug => debug

to have debug messages written to /var/log/asterisk/debug

or add `debug` to the line
messages => notice,warning,error

to include the debug messages in the /var/log/asterisk/messages file

at the * CLI prompt, enter `logger reload`

Now you will see the dtmf decoded in the debug messages.

-- 
   DataCrest, Inc. -- Technically Superior   **
Walker Haddock   http://www.datacrest.com
DataCrest, Inc.e-mail:  [EMAIL PROTECTED]
1634A Montgomery Hwy.phone:  1-888-941-3282, 1-205-335-8589
Birmingham, AL 35216  fax:  1-205-823-7838
***
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[Asterisk-Users] Guru's help with * and AVM C2 ISDN - Newbie going mad!!

[Nick Grindley]

Dear Guru’s,

Hi, I have now built a Suse 8.2 box for *

My AVM C2 ISDN card is recognised with no problem. I can call between
extensions, leave voicemail and retrieve same voicemail.

However,

I cannot configure my AVM C2 to either receive calls or make calls –
HELP

I am in the UK if that helps or gives a pointer about our ISDN?

Please find below my config: -

; CAPI config
;
[general]
mode=immediate
isdnmode=multipoint
;nationalprefix=0
;internationalprefix=00
rxgain=0.8
txgain=0.8

[interfaces]

msn=01723343950
incomingmsn=01723343950
controller=1
;softdtmf=1
;accountcode=
;context=demo
context=default
;echosquelch=1
;echocancel=yes
echocancel=1
echotail=64
;callgroup=1
;deflect=12345678
devices=1

msn=01723343951
incomingmsn=01723343951
controller=1
;softdtmf=1
;accountcode=
;context=demo
context=default
;echosquelch=1
;echocancel=yes
echocancel=1
echotail=64
;callgroup=1
;deflect=12345678
devices=1

msn=01723343952
incomingmsn=01723343952
controller=1
;softdtmf=1
;accountcode=
;context=demo
context=default
;echosquelch=1
;echocancel=yes
echocancel=1
echotail=64
;callgroup=1
;deflect=12345678
devices=1

msn=01723343953
incomingmsn=01723343953
controller=1
;softdtmf=1
;accountcode=
;context=demo
context=default
;echosquelch=1
;echocancel=yes
echocancel=1
echotail=64
;callgroup=1
;deflect=12345678
devices=1

msn=01723343954
incomingmsn=01723343954
controller=1
;softdtmf=1
;accountcode=
;context=demo
context=default
;echosquelch=1
;echocancel=yes
echocancel=1
echotail=64
;callgroup=1
;deflect=12345678
devices=1

msn=01723343955
incomingmsn=01723343955
controller=1
;softdtmf=1
;accountcode=
;context=demo
context=default
;echosquelch=1
;echocancel=yes
echocancel=1
echotail=64
;callgroup=1
;deflect=12345678
devices=1

msn=01723343956
incomingmsn=01723343956
controller=1
;softdtmf=1
;accountcode=
;context=demo
context=default
;echosquelch=1
;echocancel=yes
echocancel=1
echotail=64
;callgroup=1
;deflect=12345678
devices=1

msn=01723343957
incomingmsn=01723343957
controller=1
;softdtmf=1
;accountcode=
;context=demo
context=default
;echosquelch=1
;echocancel=yes
echocancel=1
echotail=64
;callgroup=1
;deflect=12345678
devices=1

msn=01723343958
incomingmsn=01723343958
controller=1
;softdtmf=1
;accountcode=
;context=demo
context=default
;echosquelch=1
;echocancel=yes
echocancel=1
echotail=64
;callgroup=1
;deflect=12345678
devices=1

msn=01723343959
incomingmsn=01723343959
controller=1
;softdtmf=1
;accountcode=
;context=demo
context=default
;echosquelch=1
;echocancel=yes
echocancel=1
echotail=64
;callgroup=1
;deflect=12345678
devices=1

;PointToPoint (55512-0)
;for outgoing calls use example 5551212
;and in dialplan you can use callerid like
;exten => _0XXX.,1,StripMSD,1
;exten => _XXX.,2,Dial,CAPI/55512${CALLERIDNUM}:bBYEXTENSION
;
;mode=immediate
;isdnmode=ptp
;msn=55512
;controller=2
;devices=30


extensions.conf
;
; Static extension configuration files, used by
; the pbx_config module.
;
; The "General" category is for certain variables.
;
[general]
;
; If static is set to no, or omitted, then the pbx_config will rewrite
; this file when extensions are modified.  Remember that all comments
; made in the file will be lost when that happens.
;
; XXX Not yet implemented XXX
;
static=yes
;
; if static=yes and writeprotect=no, you can save dialplan by
; CLI command 'save dialplan' too
;
writeprotect=no

; You can include other config files, use the #include command (without the
';')
; Note that this is different from the "include" command that includes
contexts within
; other contexts. The #include command works in all asterisk configuration
files.
;#include "filename.conf"

; The "Globals" category contains global variables that can be referenced
; in the dialplan with ${VARIABLE} or ${ENV(VARIABLE)} for Environmental
variable
; ${${VARIABLE}} or ${text${VARIABLE}} or any hybrid
;
[globals]
CONSOLE=Console/dsp ; Console interface for demo
;CONSOLE=Zap/1
;CONSOLE=Phone/phone0
;IAXINFO=guest  ; IAXtel username/password
;IAXINFO=myuser:mypass
;TRUNK=Zap/g2   ; Trunk interface
;TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0)
;TRUNK=IAX2/user:[EMAIL PROTECTED]
TRUNK=CAPI
[trunklocal]
exten => _01723.,1,Dial(CAPI/@343955:${EXTEN}|30|r)
[demo]
;
; We start with what to do when a call first comes in.
;
exten => s,1,Wait,1 ; Wait a second, just for fun
exten => s,2,Answer ; Answer the line
exten => s,3,DigitTimeout,5 ; Set Digit Timeout to 5 seconds
exten => s,4,ResponseTimeout,10 ; Set Response Timeout to 10 seconds
exten => s,5,BackGround(demo-congrats)  ; Play a congratulatory message
exten => s,6,BackGround(demo-instruct)  ; Play some instructions

exten => 2,1,BackGround(demo-moreinfo)  ; Give some more information.
exten => 2,2,Goto(s,6)

exten => 3,1,SetLanguage(fr

Re: [Asterisk-Users] extensions problem (SIP)

[Olle E. Johansson]

Jon Lawrence wrote:
Hi,
I've got 2 x100p's installed in my system.
Both execute the same incoming contexts as follows:
[inboundA]
include => dialjon
[inboundB]
include => dialjon|09:00-16:30|Mon-Fri|*|*
[dialjon]
exten => s,1,answer
exten => s,2,Dial(SIP/2000,15)
exten => s,3,Playback(noone)
exten => s,103,Goto(onphone,s,1)

Am I right in saying:
if no one answers at ext 2000 then s,3 is executed.
if ext 2000 is busy  then 103 is executed.
If so then sometihng is wrong. If I'm already on a call, I want 103 to be 
executed however, this isn't happening. If a new call comes in (whilst I'm 
talking on ext 2000) I here it ringing on my handset.

It depends on your SIP device. Asterisk places the call to your SIP device 
regardless,
since by SIP protocol design the UA is not a "slave", it is free. So the SIP ua must
answer "busy" for Asterisk to understand that you're busy. If not, the call is placed
to you and Asterisk has no knowledge that you are busy. Check you SIp phone if you can
limit the number of concurrent calls.
There's some code in Asterisk chan_sip.c to limit the number of calls placed to
a SIP phone, but right now it's not working at all.
/Olle
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[Asterisk-Users] ChanCAPI and withheld / unavailable callerid indication

[Patrick Lidstone (Personal E-mail)]

My ISDN phone is able to display "No. Suppressed" and "No. Unavailable",
depending on whether a caller wittheld their identity or is simply out
of area. I'd like to be able to make the same distinction with asterisk,
ideally within extensions.conf. I've had a look at the chan_capi source
code, but am none the wiser - I don't really know enough about ISDN
signalling, so I don't know if this feature is already available,
missing or if it's not possible to do it for some other reason. Any
thoughts?

Patrick







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[Asterisk-Users] DTMF debugs

[Bruce Marler]

Is their a debug that will show what DTMF is being detected or what is being
sent by asterisk. I am having some problems with initial fax tones possibly
being detected as fax and once I change my DTMF passing type I can then send
faxes BUT get double DTMF digits sent at times.

Any debug info would be appreciated.

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Re: Asterisk-Users digest, Vol 1 #3101 - 14 msgs Subject: Re: [Asterisk-Users] Asterisk on KNOPPIX, I have it working, somewhat.

[hank smith]

let me know if you get the noppix done
I would be interested!
- Original Message -
From: <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Sunday, March 14, 2004 7:18 PM
Subject: Asterisk-Users digest, Vol 1 #3101 - 14 msgs


> Send Asterisk-Users mailing list submissions to
> [EMAIL PROTECTED]
>
> To subscribe or unsubscribe via the World Wide Web, visit
> http://lists.digium.com/mailman/listinfo/asterisk-users
> or, via email, send a message with subject or body 'help' to
> [EMAIL PROTECTED]
>
> You can reach the person managing the list at
> [EMAIL PROTECTED]
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of Asterisk-Users digest..."
>
>
> Today's Topics:
>
>1. RE: How to send CallerID trough CAPI ? (Jakob Strebel)
>2. RE: ast_rtp_raw_write errors distorting sound on G729 passthrough
(Senad Jordanovic)
>3. Re: Asterisk on KNOPPIX, I have it working, somewhat. (Greg
Boehnlein)
>4. Re: Radius (Greg Boehnlein)
>5. RE: ast_rtp_raw_write errors distorting sound on G729 passthrough
(Senad Jordanovic)
>6. RE: VXML_URL and Cisco 7960 Phones? (Low, Adam)
>7. ISDN PRI A and B, cry for help. (Matthew Branton)
>8. Re: Radius (Derek Bruce)
>9. Asterisk NAT Gateway Setup (Kevin)
>   10. Re: Cisco SIP license (Matthew Enger)
>   11. Re: ISDN PRI A and B, cry for help. (Steve Underwood)
>   12. VoYP.Net: voip directory and ENUM registry (Greg Retkowski)
>   13. Re: VoYP.Net: voip directory and ENUM registry (Matt Riddell)
>   14. EchoCan (Matt Riddell)
>
> --__--__--
>
> Message: 1
> Date: Sun, 14 Mar 2004 17:49:37 +0100
> To: [EMAIL PROTECTED]
> From: Jakob Strebel <[EMAIL PROTECTED]>
> Subject: RE: [Asterisk-Users] How to send CallerID trough CAPI ?
> Reply-To: [EMAIL PROTECTED]
>
> Florian,
>
> Thanks. Now it works.
> jakob
>
>
> >The correct dial syntax for CAPI channels is like this:
> >
> >CAPI/12345678:b${EXTEN}
> >
> >where: 12345678 is your outgoing MSN (you would choose 0627775171) and
> >${EXTEN} is the number to dial. My mistake was I moved the 'b' too when I
> >switched the two numbers around.
> >
> >Please try Dial(CAPI/0627775171:b123456) or Dial(CAPI/627775171:b123456)
> >
> This are the relevant sections in the config:
>
> In extensions.conf
>
> 
>
> [globals]
> ;
> ; globals f=FCr ISDN
> CLID=3D0627775171
> 
> [outst]
> exten =3D> _0.,1,SetCIDNum(${CLID})
> exten =3D> _0.,2,Dial(CAPI/0627775171:b${EXTEN})  ; ok CID is sent correct
> 
>
> In capi.conf
> msn=3D0627775171
>
>
>
> --__--__--
>
> Message: 2
> From: "Senad Jordanovic" <[EMAIL PROTECTED]>
> To: <[EMAIL PROTECTED]>
> Subject: RE: [Asterisk-Users] ast_rtp_raw_write errors distorting sound on
G729 passthrough
> Date: Sun, 14 Mar 2004 17:07:26 -
> Reply-To: [EMAIL PROTECTED]
>
> Olle E. Johansson wrote:
> > Check out the latest CVS, Mark applied changes to the code in this
> > area tonight. The rtp.c is changed, so the old patch in
> > bugs.digium.com may not be necessary any more.
> >
> Yes, it is done..
> BUT
> Now I get MUCH higher values is the debug messages and can not
> understand a word from other party during the conversation.
>
> Here it is:
> Mar 14 17:04:46 DEBUG[262161]: rtp.c:980 ast_rtp_raw_write: Difference
> is 1386825128, ms is -1247094945
> Mar 14 17:04:46 DEBUG[262161]: rtp.c:980 ast_rtp_raw_write: Difference
> is 1386825608, ms is -1247095005
> Mar 14 17:04:46 DEBUG[262161]: rtp.c:980 ast_rtp_raw_write: Difference
> is 1386826088, ms is -1247095065
> Mar 14 17:04:46 DEBUG[262161]: rtp.c:980 ast_rtp_raw_write: Difference
> is 1386826560, ms is -1247095124
>
>
> --__--__--
>
> Message: 3
> Date: Sun, 14 Mar 2004 12:51:37 -0500 (EST)
> From: Greg Boehnlein <[EMAIL PROTECTED]>
> To: [EMAIL PROTECTED]
> Cc: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] Asterisk on KNOPPIX, I have it working,
somewhat.
> Reply-To: [EMAIL PROTECTED]
>
> I've got Asterisk running on a minimal install of Debian on a P133 w/ 16
> megs of ram. I can help with the Damn Small Linux side of things, and
> perhaps get you out of Dependency hell.
>
> Do you want the system to be self hosting? I.E. the distribution where
> Asterisk lives contains the appropriate compilers, source and includes to
> build the system?
>
> Or do you want a "Development" distribution and a "Target" platform? I.E.
> you build on the developmen distro and then run a few scripts to generate
> a target ISO w/ the binaries?
>
> In either case, I've got plenty of resources to offer. I've been toying
> with the idea of creating a "Knapterisk" installation for quite some time.
>
> --
> Vice President of N2Net, a New Age Consulting Service, Inc. Company
>  http://www.n2net.net Where everything clicks into place!
>  KP-216-121-ST
>
>
>
>
> --__--__--
>
> Message: 4
> Date: Sun, 14 Mar 2004 13:01:26 -0500 (EST)
> From: Greg Boehnlein <[EMAIL PROTECTED]>
> To: [EMAIL PROTECTED]
> Subject: Re: [Asterisk-Users] Radius
> Reply-To: [EMAIL PROTECTED]
>
> On 

RE: [Asterisk-Users] extensions problem

[Eric_Doiron]

Maybe try,

[dialjon]
exten => s,1,answer
exten => s,2,Dial(SIP/2000,15)
exten => s,3,congestion
exten => s,4,Playback(noone)
exten => s,103,Goto(onphone,s,1)

Not sure if it will work.. just thinking,

-Eric
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jon Lawrence
Sent: Monday, March 15, 2004 10:29 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] extensions problem

Hi,
I've got 2 x100p's installed in my system.
Both execute the same incoming contexts as follows:
[inboundA]
include => dialjon
[inboundB]
include => dialjon|09:00-16:30|Mon-Fri|*|*

[dialjon]
exten => s,1,answer
exten => s,2,Dial(SIP/2000,15)
exten => s,3,Playback(noone)
exten => s,103,Goto(onphone,s,1)


Am I right in saying:
if no one answers at ext 2000 then s,3 is executed.
if ext 2000 is busy  then 103 is executed.

If so then sometihng is wrong. If I'm already on a call, I want 103 to be 
executed however, this isn't happening. If a new call comes in (whilst I'm 
talking on ext 2000) I here it ringing on my handset.

Can anyone point out where I've gone wrong ?

TIA
Jon

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[Asterisk-Users] MusicOnHold

[Brian Mulligan]

Despite my best efforts I am unable to get this to work. I have checked that
mpg123 is correctly installed and that the symbolic link exists for the
binary. I have also made sure that the REdhat9 does not have the older
version mpg321 installed. When I invoke MusicOnHold from my dialplan i get
the following message;

Mar 15 15:01:10 WARNING[1225991360]: res_musiconhold.c:416 moh_alloc: No
class: 30
Mar 15 15:01:10 WARNING[1225991360]: res_musiconhold.c:305 moh0_exec: Unable
to start
 music on hold (class '30') on channel Zap/3-1

I have digium fxo and fxs cards installed and running properly. What is
class 30? I cannot see it under classes in the musiconhold.conf file. Is
there something trivial I am missing?
Any help would be warmly appreciated.
Brian

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[Asterisk-Users] post

[Matthew Marlowe]

[Asterisk-Users] extensions problem

[Jon Lawrence]

Hi,
I've got 2 x100p's installed in my system.
Both execute the same incoming contexts as follows:
[inboundA]
include => dialjon
[inboundB]
include => dialjon|09:00-16:30|Mon-Fri|*|*

[dialjon]
exten => s,1,answer
exten => s,2,Dial(SIP/2000,15)
exten => s,3,Playback(noone)
exten => s,103,Goto(onphone,s,1)


Am I right in saying:
if no one answers at ext 2000 then s,3 is executed.
if ext 2000 is busy  then 103 is executed.

If so then sometihng is wrong. If I'm already on a call, I want 103 to be 
executed however, this isn't happening. If a new call comes in (whilst I'm 
talking on ext 2000) I here it ringing on my handset.

Can anyone point out where I've gone wrong ?

TIA
Jon

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Re: [Asterisk-Users] incoming fax x100p

[Simon Chappell]

i have made some headway..

Asterisk detects the fax.. and sends it to 2004 :-)
however the handshaking is failing, i get
rtp.c:418 ast_rtp_read: Unknown RTP Codec 100 received
Any ideas ?
Thanks in advance
Simon

Jim Sneeringer wrote:

There is a special extension, called "fax," that you use to specify where
you want incoming fax calls to go.  It detects faxes by the CNG tone and
sends them where you specify.  Here is an example:
exten => fax,1,Dial(${Fax}) ; Use Zap channel for 2004 instead of ${Fax}

There are some faxes that do not send CNG tones, so this will occasionally
fail for that reason. You might also send calls that time out to the fax
extension, or you can use distinctive ringing, as someone else suggested.
Let me know how this works for you. I am still having trouble with garbled
fax transmissions once I get a connection to the fax, and I would be
interested in your experience sending and receiving faxes.
Jim

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Simon Chappell
Sent: Saturday, March 13, 2004 1:31 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] incoming fax x100p
Hello all

I notice there have been many mails flying about regarding fax softare 
and hylafax..
I have what i thought would be a simpler solution but i am still a bit 
lost with it.
I have a normal fax machine installed as extension 2004 on a sipura.
I also have a X100P.
I think i am right in saying that the X100P can tell if a fax is coming 
in , does that mean i can redirect an incoming fax to ext 2004? of 
course i could be missing the plot completely and the X100P does can not 
do this.
Any ideas greatly appreciated.

 

--
Kind Regards
Simon Chappell
url : www.isnsuk.com
email : [EMAIL PROTECTED]
PH: 01403 268474
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RE: [Asterisk-Users] Cisco Call Manager and Asterisk and fast busy?

[Kyle Stone]

I tried that, to no avail.  It does the same thing.. Places the call,
and then the CCM hangsup 1.5 rings into it.

.


-Original Message-
From: Dan Austin [mailto:[EMAIL PROTECTED] 
Sent: Friday, March 12, 2004 8:18 PM
To: Kyle Stone
Cc: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Cisco Call Manager and Asterisk and fast
busy?

Check the codecs allowed.  Cisco supports most, but in my tests I
limited
myself to G.711U.

Another possibility, and one I seem to remember having is that the IP 
address of the gateway did not match the * server, and as such did not
have access to the correct Calling Search Space and Media Resource
Groups.

Still codec choice seems the most likely.

Dan

-Original Message-
From: Kyle Stone [mailto:[EMAIL PROTECTED] 
Sent: Friday, March 12, 2004 6:40 AM
To: Dan Austin
Cc: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Cisco Call Manager and Asterisk and fast
busy?



I got the latest oh323, and the libraries.  I got everything installed
and working.. somewhat.   I setup netmeeting and can call IAXcomm
extensions I setup. I also can place calls in the reverse.  When I use
either to route a call to CCM, setup as per you described.. The
CCM/skinny phone will ring one and a half times and hangup.  If answered
before the end of the ringing I get a fast busy.   Any ideas?


Kyle Stone


-Original Message-
From: Dan Austin [mailto:[EMAIL PROTECTED] 
Sent: Monday, March 08, 2004 12:04 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Cisco Call Manager and Asterisk?

In CCM add a Gateway.  Use H.323 with H.225 as the device protocol.
Next add a route pattern to identify which calls to direct to *.

Lastly use chan_oh323 instead of chan_h323, as the former works
with CCM and the later does not (one way audio).

The setup is extremely easy and works just fine, with the correct
channel that is.

Dan




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RE: [Asterisk-Users] Grandstream TFTP Config

[Matthew Marlowe]

I can confirm  1.0.4.53 is bad as well. :) 1.0.4.50 has been working
fine for me.

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Greg Boehnlein
> Sent: Monday, March 15, 2004 9:28 AM
> To: Asterisk Users
> Subject: Re: [Asterisk-Users] Grandstream TFTP Config
> 
> On 15 Mar 2004, Hermann Wecke wrote:
> 
> > On Mon, 15 Mar 2004, Joshua McAdam wrote:
> > > So far I have managed to upgrade the firmware, but I am not sure 
> > > what the cfg.txt should contain as I have tried a few I 
> found from 
> > > searches of the list and also on the wiki.
> > 
> > I found this:
> > 
> http://lists.digium.com/pipermail/asterisk-users/2004-January/034125.h
> > tml
> > but I'm not using. I don't even try.
> > 
> > > A tcpdump reveals that the phone is downloading the files, has 
> > > anyone managed to get this working?
> > 
> > What I found is that the latest firmware (b14p4.53.zip) 
> crashed my phone.
> > I was able to receive calls but every call made returned a 
> "4" error 
> > (don't know if it was a 404, 484, 4XX - only a 4 was displayed).
> 
> I can confirm that 1.0.4.53 is bad. ;) I tried upgrading to 
> it last night, couldn't make calls, and downgraded to 1.0.4.50.
> 
> -- 
> Vice President of N2Net, a New Age Consulting Service, 
> Inc. Company
>  http://www.n2net.net Where everything clicks into place!
>  KP-216-121-ST
> 
> 
> 
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Re: [Asterisk-Users] OT: cisco 7960G powered by 3com 3CNJPSE

[Hermann Wecke]

On Mon, 15 Mar 2004, stan wrote:
> Is anyone using a 3com 3CNJPSE to power a 7960G?
> I have a couple of 7960Gs and 3CNJPSEs but no combination appears to
> work.  Both phones work fine with a cisco power cube.  I get a 47.6V
> reading across pins 5 and 8 on the patch cable coming from the 3CNJPSE.

I don't believe that the message below will apply to you, but it may worth
a reading for non-G users:

http://lists.digium.com/pipermail/asterisk-users/2004-February/038223.html
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Re: [Asterisk-Users] OT: cisco 7960G powered by 3com 3CNJPSE

[John Fraizer]

You need to make sure the polarity is right.  If I'm not mistaken, I think I 
saw someone post that the Cisco cubes had the polarity reversed.

John

stan wrote:
Is anyone using a 3com 3CNJPSE to power a 7960G?  
I have a couple of 7960Gs and 3CNJPSEs but no combination appears to
work.  Both phones work fine with a cisco power cube.  I get a 47.6V
reading across pins 5 and 8 on the patch cable coming from the 3CNJPSE.
The network still works through the 3CNJPSE, just no power to the phone
until I connect the power cube.  Using two straight through wired patch
cables, 7960G running sip firmware version 6.3.  I've tried different
patch cables with the same result.  Note this is a G version of the
phone which I understand is enable to work with 802.3af devices natively
and hence I believe doesn't require a specially wired patch cable.  I've
looked for a setting to tell the phone to get power via the network
without success, is there one?  Any ideas?

TIA
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Re: [Asterisk-Users] incoming fax x100p

[Simon Chappell]

thanks for the help

I have added exten => fax,1,Dial(SIP/2004) in my extensions.conf
What i can see from the CLI is that asterisk is reporting fax detected 
but no fax extension , have i got the syntax wrong or do i need to add 
it elseware

thanks again

Simon

Jim Sneeringer wrote:

There is a special extension, called "fax," that you use to specify where
you want incoming fax calls to go.  It detects faxes by the CNG tone and
sends them where you specify.  Here is an example:
exten => fax,1,Dial(${Fax}) ; Use Zap channel for 2004 instead of ${Fax}

There are some faxes that do not send CNG tones, so this will occasionally
fail for that reason. You might also send calls that time out to the fax
extension, or you can use distinctive ringing, as someone else suggested.
Let me know how this works for you. I am still having trouble with garbled
fax transmissions once I get a connection to the fax, and I would be
interested in your experience sending and receiving faxes.
Jim

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Simon Chappell
Sent: Saturday, March 13, 2004 1:31 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] incoming fax x100p
Hello all

I notice there have been many mails flying about regarding fax softare 
and hylafax..
I have what i thought would be a simpler solution but i am still a bit 
lost with it.
I have a normal fax machine installed as extension 2004 on a sipura.
I also have a X100P.
I think i am right in saying that the X100P can tell if a fax is coming 
in , does that mean i can redirect an incoming fax to ext 2004? of 
course i could be missing the plot completely and the X100P does can not 
do this.
Any ideas greatly appreciated.

 

--
Kind Regards
Simon Chappell
url : www.isnsuk.com
email : [EMAIL PROTECTED]
PH: 01403 268474
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[Asterisk-Users] OT: cisco 7960G powered by 3com 3CNJPSE

[stan]

Is anyone using a 3com 3CNJPSE to power a 7960G?  
I have a couple of 7960Gs and 3CNJPSEs but no combination appears to
work.  Both phones work fine with a cisco power cube.  I get a 47.6V
reading across pins 5 and 8 on the patch cable coming from the 3CNJPSE.
The network still works through the 3CNJPSE, just no power to the phone
until I connect the power cube.  Using two straight through wired patch
cables, 7960G running sip firmware version 6.3.  I've tried different
patch cables with the same result.  Note this is a G version of the
phone which I understand is enable to work with 802.3af devices natively
and hence I believe doesn't require a specially wired patch cable.  I've
looked for a setting to tell the phone to get power via the network
without success, is there one?  Any ideas?

TIA
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Re: [Asterisk-Users] Resetting Grandstream HT-286 to factory default settings?

[Hermann Wecke]

On Sat, 13 Mar 2004, Brian Buhrow wrote:
> address which contains letters -- do I pretend I'm dialing a name and use
> the numbers associated with the letters of the MAC address?  When I try to
> do this, it doesn't reset, and tells me my numbers are invalid.
>   Any suggestions on how to restore this box to factory freshness?

Yes... I found a manual (http://www.ovislink.ca/OVMNL/HandyTone.pdf) and I
believe it will work the same way as the GS BT phones: to enter a letter
"A", you will dial 11, for B, 111 and so on...
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Re: [Asterisk-Users] Grandstream TFTP Config

[Greg Boehnlein]

On 15 Mar 2004, Hermann Wecke wrote:

> On Mon, 15 Mar 2004, Joshua McAdam wrote:
> > So far I have managed to upgrade the firmware, but I am not sure what the
> > cfg.txt should contain as I have tried a few I found from searches of the
> > list and also on the wiki.
> 
> I found this:
> http://lists.digium.com/pipermail/asterisk-users/2004-January/034125.html
> but I'm not using. I don't even try.
> 
> > A tcpdump reveals that the phone is downloading the files, has anyone
> > managed to get this working?
> 
> What I found is that the latest firmware (b14p4.53.zip) crashed my phone.
> I was able to receive calls but every call made returned a "4" error
> (don't know if it was a 404, 484, 4XX - only a 4 was displayed).

I can confirm that 1.0.4.53 is bad. ;) I tried upgrading to it last night, 
couldn't make calls, and downgraded to 1.0.4.50.

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



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