asterisk users - Mar 2006 - RE: Asterisk-Users Digest, Vol 20, Issue 13

On Thu, 2006-03-02 at 11:42 -0600, Jordan Novak wrote:
> Does anyone have a way to do wake calls?
> 
>  
> 
> Jordan Novak
> 
> Communications Technician
> 
> Logistics Health Inc.
You could use cron and /var/spool/asterisk/outgoing scripts to dial
numbers, etc...
> 
Can you elaborate, I am fairly new to Linux and a phone guy to boot. I
am looking for a way for the users to set a wake up call for themselves
from the phone...
Something like...

Dial an extension for wakeups
The caller is asked to set a time and the number of days for which they
want it set. The system then calls at those times, and every ten minutes
until it is answered.





------------------------------

Message: 3
Date: Thu, 2 Mar 2006 13:10:53 -0500
From: "Wojciech Tryc" <wojtek@VoIPMan.ORG>
Subject: Re: [Asterisk-Users] OT - Cisco IP Phone and PC in diferent
	VLANs(with 802.1x)
To: <joao.pereira@fccn.pt>,	"Asterisk Users Mailing List -
	Non-Commercial Discussion"
<asterisk-users@lists.digium.com>
Message-ID: <002501c63e24$a5f94780$4c45a8c0@kanatek.com>
Content-Type: text/plain; format=flowed; charset="iso-8859-1";
	reply-type=response

Your pc has to able to support tagged vlans. The switch on the phone
will 
pass through both tagged and untagged vlans.
W
----- Original Message ----- 
From: "Joao Pereira" <joao.pereira@fccn.pt>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
<asterisk-users@lists.digium.com>
Sent: Thursday, March 02, 2006 11:51 AM
Subject: Re: [Asterisk-Users] OT - Cisco IP Phone and PC in diferent 
VLANs(with 802.1x)

> And about the 802.1x ?
> The phones can work as passthrough and force the PC to use 802.1x ?
> What configuration do we put in the switches? Do we put the switch as 
> "access" (with 802.1x) or "trunk" (without 802.1x) ?
>
> Thanks
> Joao Pereira
>
>
>
> Greg Oliver wrote:
>
>>It actually depends on the switch model.  Some put the port into
>>trunking mode automatically with the sw voi command, and some do not.
>>
>>Hopefully one day Cisco will finally make their own products and
become
>>uniform instead of buying several companies and glue'ing them all
>>together to get an ethernet switch that works.  At least they got the
>>routers right :)
>>
>>On Thu, 2006-03-02 at 08:13 -0800, Gary Richardson wrote:
>>
>>>You don't need switchport mode trunk when using switchport voice
>>>vlan..
>>>On 3/1/06, Nicholas Kathmann
>>><nicholas.kathmann@kathmannconsulting.com> wrote:
>>>        Joao Pereira wrote:
>>>        > Hello to all > I would like to know If some of you
have
already 
>>> configured
>>>        an Cisco
>>>        > IP Phone (7940 or 7960) to work in a different VLAN
than
the
>>>        PC that
>>>        > is connected through the phone switch?
>>>        > I know that this can be done with the Skinny firmware,
but
I
>>>        dont if > it works with the SIP firmware.
>>>        >
>>>        > The Cisco technical staff told me that these phones
dont
>>>        support
>>>        > 802.1x but can work as pass-through. This way I can
still
>>>        use the PCs
>>>        > with 802.1x and the phones in the same Ethernet plug.
>
>>>        > Did someone made it with the Cisco IP phones? What
>>>        configuration do I
>>>        > need in the phones and in the switch?
>>>        > Thanks
>>>        > Joao Pereira
>>>        >
>>>        If configuring with Cisco switches, I'm pretty sure they
pull
>>>        the information for which VLAN to operate in from the
switch.
>>> You
>>>        have to
>>>        configure the switchports on the Cisco switch like so:
>>>        interface fastethernet 0/1
>>>           switchport trunk native vlan <your data vlan>
switchport
mode 
>>> trunk
>>>           switchport voice vlan <your voice vlan>
>>>           spanning-tree portfast trunk
>>>        etc.
>>>        Thanks,
>>>        Nicholas Kathmann, CISSP
>>>        Kathmann Consulting, LLC
>>>        _______________________________________________ --Bandwidth
and 
>>> Colocation provided by Easynews.com --
>>>        Asterisk-Users mailing list
>>>        To UNSUBSCRIBE or update options visit:
>>>           http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>>_______________________________________________
>>>--Bandwidth and Colocation provided by Easynews.com --
>>>
>>>Asterisk-Users mailing list
>>>To UNSUBSCRIBE or update options visit:
>>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>
>>_______________________________________________
>>--Bandwidth and Colocation provided by Easynews.com --
>>
>>Asterisk-Users mailing list
>>To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
> 


------------------------------

Message: 4
Date: Thu, 02 Mar 2006 18:15:28 +0000
From: Joao Pereira <joao.pereira@fccn.pt>
Subject: Re: [Asterisk-Users] OT - Cisco IP Phone and PC in diferent
	VLANs(with	802.1x)
To: Wojciech Tryc <wojtek@VoIPMan.ORG>,
	asterisk-users@lists.digium.com
Message-ID: <44073640.1020100@fccn.pt>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed

Ok, but the PC has an 802.1x client that configures the VLAN when he 
authenticates.
Is this going to pass through the phone?
And will the switch accept it?
Thanks
Joao Pereira


Wojciech Tryc wrote:
> Your pc has to able to support tagged vlans. The switch on the phone 
> will pass through both tagged and untagged vlans.
> W
> ----- Original Message ----- From: "Joao Pereira"
<joao.pereira@fccn.pt>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
> <asterisk-users@lists.digium.com>
> Sent: Thursday, March 02, 2006 11:51 AM
> Subject: Re: [Asterisk-Users] OT - Cisco IP Phone and PC in diferent 
> VLANs(with 802.1x)
>
>
>> And about the 802.1x ?
>> The phones can work as passthrough and force the PC to use 802.1x ?
>> What configuration do we put in the switches? Do we put the switch as
>> "access" (with 802.1x) or "trunk" (without 802.1x)
?
>>
>> Thanks
>> Joao Pereira
>>
>>
>>
>> Greg Oliver wrote:
>>
>>> It actually depends on the switch model.  Some put the port into
>>> trunking mode automatically with the sw voi command, and some do
not.
>>>
>>> Hopefully one day Cisco will finally make their own products and
become
>>> uniform instead of buying several companies and glue'ing them
all
>>> together to get an ethernet switch that works.  At least they got
the
>>> routers right :)
>>>
>>> On Thu, 2006-03-02 at 08:13 -0800, Gary Richardson wrote:
>>>
>>>> You don't need switchport mode trunk when using switchport
voice
>>>> vlan..
>>>> On 3/1/06, Nicholas Kathmann
>>>> <nicholas.kathmann@kathmannconsulting.com> wrote:
>>>>        Joao Pereira wrote:
>>>>        > Hello to all > I would like to know If some of
you have
>>>> already configured
>>>>        an Cisco
>>>>        > IP Phone (7940 or 7960) to work in a different VLAN
than
the
>>>>        PC that
>>>>        > is connected through the phone switch?
>>>>        > I know that this can be done with the Skinny
firmware, but
I
>>>>        dont if > it works with the SIP firmware.
>>>>        >
>>>>        > The Cisco technical staff told me that these phones
dont
>>>>        support
>>>>        > 802.1x but can work as pass-through. This way I can
still
>>>>        use the PCs
>>>>        > with 802.1x and the phones in the same Ethernet
plug. >
>>>>        > Did someone made it with the Cisco IP phones? What
>>>>        configuration do I
>>>>        > need in the phones and in the switch?
>>>>        > Thanks
>>>>        > Joao Pereira
>>>>        >
>>>>        If configuring with Cisco switches, I'm pretty sure
they
pull
>>>>        the information for which VLAN to operate in from the 
>>>> switch. You
>>>>        have to
>>>>        configure the switchports on the Cisco switch like so:
>>>>        interface fastethernet 0/1
>>>>           switchport trunk native vlan <your data vlan>
switchport
>>>> mode trunk
>>>>           switchport voice vlan <your voice vlan>
>>>>           spanning-tree portfast trunk
>>>>        etc.
>>>>        Thanks,
>>>>        Nicholas Kathmann, CISSP
>>>>        Kathmann Consulting, LLC
>>>>        _______________________________________________
--Bandwidth
>>>> and Colocation provided by Easynews.com --
>>>>        Asterisk-Users mailing list
>>>>        To UNSUBSCRIBE or update options visit:
>>>>          
http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>
>>>> _______________________________________________
>>>> --Bandwidth and Colocation provided by Easynews.com --
>>>>
>>>> Asterisk-Users mailing list
>>>> To UNSUBSCRIBE or update options visit:
>>>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>
>>>
>>> _______________________________________________
>>> --Bandwidth and Colocation provided by Easynews.com --
>>>
>>> Asterisk-Users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>
>> _______________________________________________
>> --Bandwidth and Colocation provided by Easynews.com --
>>
>> Asterisk-Users mailing list
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>


------------------------------

Message: 5
Date: Thu, 2 Mar 2006 19:21:12 +0100
From: "ADEGOKE ARUNA" <goksie@gmail.com>
Subject: RE: [Asterisk-Users] my zap channel not ringing & source from
	internal to telco line
To: "'Asterisk Users Mailing List - Non-Commercial
Discussion'"
	<asterisk-users@lists.digium.com>
Message-ID: <440737b2.06b53a3d.4c69.ffffbccf@mx.gmail.com>
Content-Type: text/plain;	charset="us-ascii"

Yes, I think I made a progress,

I got this from my pri status, but my sync source is till saying
"internally
blocked" I have made several attempt at changing it to line but no
success
yet.

How can I change my clock source from internal to telco line

My debugs are as follows:

gnugk*CLI> pri show span 1
Primary D-channel: 16
Status: Provisioned, Up, Active
Switchtype: EuroISDN
Type: CPE
Window Length: 0/7
Sentrej: 0
SolicitFbit: 0
Retrans: 0
Busy: 0
Overlap Dial: 0
T200 Timer: 1000
T203 Timer: 10000
T305 Timer: 30000
T308 Timer: 4000
T313 Timer: 4000
N200 Counter: 3



gnugk*CLI> pri intense debug spanX
< [ 02 01 7f ]

< Unnumbered frame:
< SAPI: 00  C/R: 1 EA: 0
<  TEI: 000        EA: 1
<   M3: 3   P/F: 1 M2: 3 11: 3  [ SABME (set asynchronous balanced mode
extended) ]
< 0 bytes of data
-- Got SABME from network peer.
Sending Unnumbered Acknowledgement
> [ 02 01 73 ]
> Unnumbered frame:
> SAPI: 00  C/R: 1 EA: 0
>  TEI: 000        EA: 1
>   M3: 3   P/F: 1 M2: 0 11: 3  [ UA (unnumbered acknowledgement) ]
> 0 bytes of data
-- Restarting T203 counter
-- Restarting T203 counter
  == Primary D-Channel on span 1 up
    -- Accepting AUTHENTICATED call from 10.80.1.151:
       > requested format = ulaw,
       > requested prefs = (),
       > actual format = gsm,
       > host prefs = (),
       > priority = mine
    -- Executing Answer("IAX2/marko-3", "") in new stack
    -- Executing Dial("IAX2/marko-3",
"Zap/g1/6210006|60|Ttr") in new
stack
    -- Requested transfer capability: 0x00 - SPEECH
> [ 00 01 00 00 08 02 00 0a 05 04 03 80 90 a3 18 03 a9 83 81 6c 07 21 81
31
31 31 31 31 70 08 a1 36 32 31 30 30 30 36 a1 ]
> Informational frame:
> SAPI: 00  C/R: 0 EA: 0
>  TEI: 000        EA: 1
> N(S): 000   0: 0
> N(R): 000   P: 0
> 35 bytes of data
-- Restarting T203 counter
Stopping T_203 timer
Starting T_200 timer
> Protocol Discriminator: Q.931 (8)  len=35
> Call Ref: len= 2 (reference 10/0xA) (Originator)
> Message type: SETUP (5)
> [04 03 80 90 a3]
> Bearer Capability (len= 5) [ Ext: 1  Q.931 Std: 0  Info transfer
capability: Speech (0)
>                              Ext: 1  Trans mode/rate: 64kbps,
circuit-mode
(16)
>                              Ext: 1  User information layer 1: A-Law
(35)
> [18 03 a9 83 81]
> 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 ]
> [6c 07 21 81 31 31 31 31 31]
> Calling Number (len= 9) [ Ext: 0  TON: National Number (2)  NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1)
>                           Presentation: Presentation permitted, user
number passed network screening (1) '11111' ]
> [70 08 a1 36 32 31 30 30 30 36]
> Called Number (len=10) [ Ext: 1  TON: National Number (2)  NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1) '6210006'
]
> [a1]
> Sending Complete (len= 1)
    -- Called g1/6210006
gnugk*CLI> pri intense debug spanX
< [ 02 01 7f ]

< Unnumbered frame:
< SAPI: 00  C/R: 1 EA: 0
<  TEI: 000        EA: 1
<   M3: 3   P/F: 1 M2: 3 11: 3  [ SABME (set asynchronous balanced mode
extended) ]
< 0 bytes of data
-- Got SABME from network peer.
Sending Unnumbered Acknowledgement
> [ 02 01 73 ]
> Unnumbered frame:
> SAPI: 00  C/R: 1 EA: 0
>  TEI: 000        EA: 1
>   M3: 3   P/F: 1 M2: 0 11: 3  [ UA (unnumbered acknowledgement) ]
> 0 bytes of data
-- Restarting T203 counter
  == Primary D-Channel on span 1 up
gnugk*CLI> pri intense debug spanX


gnugk*CLI>
> Informational frame:
> SAPI: 00  C/R: 0 EA: 0
>  TEI: 000        EA: 1
> N(S): 000   0: 0
> N(R): 000   P: 0
> 35 bytes of data
-- Restarting T203 counter
Stopping T_203 timer
Starting T_200 timer
> Protocol Discriminator: Q.931 (8)  len=35
> Call Ref: len= 2 (reference 12/0xC) (Originator)
> Message type: SETUP (5)
> [04 03 80 90 a3]
> Bearer Capability (len= 5) [ Ext: 1  Q.931 Std: 0  Info transfer
capability: Speech (0)
>                              Ext: 1  Trans mode/rate: 64kbps,
circuit-mode
(16)
>                              Ext: 1  User information layer 1: A-Law
(35)
> [18 03 a9 83 81]
> 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 ]
> [6c 07 21 81 31 31 31 31 31]
> Calling Number (len= 9) [ Ext: 0  TON: National Number (2)  NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1)
>                           Presentation: Presentation permitted, user
number passed network screening (1) '11111' ]
> [70 08 a1 36 32 31 30 30 30 34]
> Called Number (len=10) [ Ext: 1  TON: National Number (2)  NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1) '6210004'
]
> [a1]CLI>
> Sending Complete (len= 1)
    -- Called g1/6210004
gnugk*CLI>


I have changed the extension.conf as advised.

goksie






-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of yusuf
Sent: Thursday, March 02, 2006 6:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] my zap channel not ringing

ADEGOKE ARUNA wrote:
> 
> I need your help
> 
> I have a sangoma A104D on my dell server; I got card status ok with no
alarm
> If I dialed the extension 6210006, it shows the output as stated
below,
but
> there is no ringing from the pstn number nor the iax softphone am
using on
> my pc.
> 
> I will be glad if someone can give me a working config?
> 
> What I want to achieve is to send all my call to the pstn on A104D?
> 
> The pstn am talking to is alcatel S12 and the pri status on their
switch
is
> showing the channel is external blocked meaning that the channels are
> blocked from my asterisk box.
> .
> 
> 
> Output from asterisk cli
> 
> -- Accepting AUTHENTICATED call from 10.80.1.151:
>        > requested format = ulaw,
>        > requested prefs = (),
>        > actual format = gsm,
>        > host prefs = (),
>        > priority = mine
>     -- Executing Answer("IAX2/marko-3", "") in new
stack
>     -- Executing Dial("IAX2/marko-3",
"Zap/g1/6210006,60,r") in new
stack
>     -- Called g1/6210006,60,r
>     -- Zap/1-1 answered IAX2/marko-3
>     -- Hungup 'Zap/1-1'
>   == Spawn extension (default, 6210006, 2) exited non-zero on
'IAX2/marko-3'
>     -- Hungup 'IAX2/marko-3'
> 
> Extension.conf (extract)
> 
> exten => _621XXXX,1,Answer()
> exten => _621XXXX,n,Dial,Zap/g1/${EXTEN),60,r
> ;exten => _621XXXX,n,Voicemail(u${EXTEN})
> exten => _621XXXX,n,Hangup()
> 
> Zaptel.conf
> 
> span=1,1,0,ccs,hdb3,crc4
> span=1,2,0,ccs,hdb3,crc4
> span=1,3,0,ccs,hdb3,crc4
> span=1,4,0,ccs,hdb3,crc4
> bchan = 1-15, 17-31, 32-46, 48-62, 63-77, 79-93, 94-108, 110-124
> dchan = 16, 47, 78, 109
> 
> Zapata.conf
> 
> [channels]
> language=en
> context=default
> switchtype=qsig
> signalling=pri_cpe
> usecallerid=yes
> hidecallerid=no
> callwaiting=yes
> usecallingpres=yes
> callwaitingcallerid=yes
> threewaycalling=yes
> transfer=yes
> canpark=yes
> cancallforward=yes
> callreturn=yes
> echocancel=yes
> echocancelwhenbridged=yes
> rxgain=0.0
> txgain=0.0
> group=1
> channel =>1-15, 17-31
> ;callgroup=1
> ;pickupgroup=1
> immediate=no
> ;callerid=6216000
> ; signalling = pri_cpe
> group = 2
> channel => 32-46, 48-62
> 
> group = 3
> channel => 63-77, 79-93
> 
> group = 4
> channel => 94-108, 110-124
> 
> 
> 
> the channel status
> 
> *CLI> zap show status
> Description                              Alarms     IRQ        bpviol
> CRC4
> wanpipe1 card 0                          OK         0          0
0
> wanpipe2 card 1                          RED        0          0
0
> wanpipe3 card 2                          RED        0          0
0
> wanpipe4 card 3                          RED        0          0
0
> 
> 
> 
> 1 19:33:14 NOTICE[17314]: chan_zap.c:6342 handle_init_event: Alarm
cleared
> on channel 1
> Mar  1 19:33:14 NOTICE[17314]: chan_zap.c:6342 handle_init_event:
Alarm
> cleared on channel 2
> Mar  1 19:33:14 NOTICE[17314]: chan_zap.c:6342 handle_init_event:
Alarm
> cleared on channel 3
> Mar  1 19:33:14 NOTICE[17314]: chan_zap.c:6342 handle_init_event:
Alarm
> cleared on channel 4
> Mar  1 19:33:14 NOTICE[17314]: chan_zap.c:6342 handle_init_event:
Alarm
> cleared on channel 5
> Mar  1 19:33:14 NOTICE[17314]: chan_zap.c:6342 handle_init_event:
Alarm
> cleared on channel 6
> Mar  1 19:33:14 NOTICE[17314]: chan_zap.c:6342 handle_init_event:
Alarm
> cleared on channel 7
> Mar  1 19:33:14 NOTICE[17314]: chan_zap.c:6342 handle_init_event:
Alarm
> cleared on channel 8
> Mar  1 19:33:14 NOTICE[17314]: chan_zap.c:6342 handle_init_event:
Alarm
> cleared on channel 9
> Mar  1 19:33:14 NOTICE[17314]: chan_zap.c:6342 handle_init_event:
Alarm
> cleared on channel 10
> Mar  1 19:33:14 NOTICE[17314]: chan_zap.c:6342 handle_init_event:
Alarm
> cleared on channel 11
> For all the 31 channels
> 
> goksie
Hi,

maybe this is just been pedantic but why do you answer the channel 
first, you dont use IVR, so why answer it. and use dial like this:

exten => _621XXXX,1,Dial(Zap/g1/${EXTEN),60,r)

everything else looks good though, it should work.  I also have a
sangoma A104D.

also, should'nt be (could be wrong, but i have it like this):

  span=1,1,0,ccs,hdb3,crc4
  span=2,2,0,ccs,hdb3,crc4
  span=3,3,0,ccs,hdb3,crc4
  span=4,4,0,ccs,hdb3,crc4
  bchan = 1-15, 17-31, 32-46, 48-62, 63-77, 79-93, 94-108, 110-124
  dchan = 16, 47, 78, 109



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



------------------------------

Message: 6
Date: Thu, 2 Mar 2006 13:24:00 -0500
From: "Albert Chaffman" <achaffman@ml3group.com>
Subject: [Asterisk-Users] setmusiconhold doesn't work between 2 SIP
	phones
To: asterisk-users@lists.digium.com
Message-ID:
	
<B727B1BBE6D96F40B469B325256709BC1A2DE1@sourceonemail.sourceone.dom>
Content-Type: text/plain;	charset="us-ascii"

Here is my scenario:

Sip phone number 1 and 2 are defined in sip.conf, and both have
musiconhold=<class> set to the same outbound class that I want.  This
works fine for outbound calls (out to the pstn)

Also, in extensions.conf for each extension that is setup to dial each
of those sip phones, the first priority is SetMusicOnHold(<class>)

So this works when a call comes in from the PSTN to either SIP phone,
and the SIP phone puts the call on hold - The PSTN side hears the
correct music

What doesn't work is when SIP 1 calls SIP 2.  When Sip 1 calls Sip 2, If
SIP 2 puts the call on hold, SIP 1 hears the correct music, BUT if SIP 1
puts the call on hold, SIP 2 hears the default music.

The same goes in reverse - SIP 2 calls SIP 1.  SIP 1 puts the call on
hold, and SIP 2 hears the correct music, but if SIP2 puts the call on
hold, SIP1 hears the default music.

Any ideas?


Albert Chaffman
ML3Group, LLC
6031 University Blvd. Suite 180
Ellicott City, MD 21043
Main: 410-750-1780
Direct: 410-750-1016
Fax: 410-750-1781
achaffman@ml3group.com



------------------------------

Message: 7
Date: Thu, 2 Mar 2006 10:26:11 -0800 (PST)
From: asterisk@anime.net
Subject: Re: [Asterisk-Users] Re: sipura 841 mass provisioning
To: Asterisk Users Mailing List - Non-Commercial Discussion
	<asterisk-users@lists.digium.com>
Message-ID: <Pine.LNX.4.63.0603021025520.12962@sasami.anime.net>
Content-Type: TEXT/PLAIN; charset=US-ASCII; format=flowed

On Thu, 2 Mar 2006, Matt wrote:
> I am guessing you need to escape the <'s.  Possibly with a \ but
I'm
> not sure.  So
> |\<9;\>
No. Use &gt; and &lt;

-Dan


------------------------------

Message: 8
Date: Thu, 2 Mar 2006 10:30:02 -0800 (PST)
From: asterisk@anime.net
Subject: Re: [Asterisk-Users] Sipura SPA-3000 vs Linksys SPA3000
To: Asterisk Users Mailing List - Non-Commercial Discussion
	<asterisk-users@lists.digium.com>
Message-ID: <Pine.LNX.4.63.0603021029360.12962@sasami.anime.net>
Content-Type: TEXT/PLAIN; charset=US-ASCII; format=flowed

On Thu, 2 Mar 2006, John Jensen wrote:
> You might want to get hold of the SPA3102 if you can.
... where?

-Dan


------------------------------

Message: 9
Date: Thu, 2 Mar 2006 19:44:52 +0100
From: "Joash Herbrink" <Joash.Herbrink@Kahuna.nl>
Subject: RE: [Asterisk-Users] OT - Cisco IP Phone and PC in diferent
	VLANs(with 802.1x)
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
	<asterisk-users@lists.digium.com>
Message-ID: <819464631e0f321166ba6007cfc155de44073908@kahuna.nl>
Content-Type: text/plain;	charset="us-ascii"

Cisco phones act a as a switch.
If you do not use the CDP protocol to "tell" the phone it needs to be
in
a special VLAN (802.1q) then it will just use the access port settings
on the switch, and, also allow the PC connected to the 2nd Ethernet port
to have access to the network.

However, if you have an all cisco powered network, with all cisco
phones, I could advise you to use the CDP protocol to allow the phone to
use a special voice vlan.

A config somewhat like this will do that for you.

Make sure the * server has access to the vlan.
This can be done by configuring an access port into the voice vlan, or
to enable 802.1q on the * server.

Anyway, this config will detect (with CDP) that a phone is connected,
and the switchport will go into trunk mode, allow 2 vlan's (802.1q) to
pass through it.

If no phone is detected (or at least no CDP capable device) the switch
will automatically make it an access port, allowing only access to the
native vlan, so, the switch port can be used very dynamically.

Of course you need to define the vlan first, before you can create
configs like this.

Hope this helps,

joash

interface FastEthernet3/1
 switchport access vlan 200
 switchport trunk encapsulation dot1q
 switchport trunk native vlan 100
 switchport mode trunk
 switchport voice vlan 101
 qos trust dscp
 qos trust extend
 spanning-tree portfast trunk




-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Greg
Oliver
Sent: Thursday, March 02, 2006 6:24 PM
To: joao.pereira@fccn.pt; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [Asterisk-Users] OT - Cisco IP Phone and PC in diferent
VLANs(with 802.1x)

I have never used a switchport for .1x to a PC connected through a
phone.  I would say it probably will not work since it bypasses the idea
of .1x entirely if it does.

You maybe could use it in 802.11 mode, but the phone would probably not
have access until the PC auths (if it would work at all)..


On Thu, 2006-03-02 at 16:51 +0000, Joao Pereira wrote:
> And about the 802.1x ?
> The phones can work as passthrough and force the PC to use 802.1x ?
> What configuration do we put in the switches? Do we put the switch as 
> "access" (with 802.1x) or "trunk" (without 802.1x) ?
> 
> Thanks
> Joao Pereira
> 
> 
> 
> Greg Oliver wrote:
> 
> >It actually depends on the switch model.  Some put the port into
> >trunking mode automatically with the sw voi command, and some do not.
> >
> >Hopefully one day Cisco will finally make their own products and
become
> >uniform instead of buying several companies and glue'ing them all
> >together to get an ethernet switch that works.  At least they got the
> >routers right :)
> >
> >On Thu, 2006-03-02 at 08:13 -0800, Gary Richardson wrote:
> >  
> >
> >>You don't need switchport mode trunk when using switchport
voice
> >>vlan.. 
> >>
> >>On 3/1/06, Nicholas Kathmann
> >><nicholas.kathmann@kathmannconsulting.com> wrote:
> >>        Joao Pereira wrote:
> >>        > Hello to all 
> >>        > I would like to know If some of you have already
configured
> >>        an Cisco
> >>        > IP Phone (7940 or 7960) to work in a different VLAN
than
the
> >>        PC that
> >>        > is connected through the phone switch?
> >>        > I know that this can be done with the Skinny firmware,
but
I
> >>        dont if 
> >>        > it works with the SIP firmware.
> >>        >
> >>        > The Cisco technical staff told me that these phones
dont
> >>        support
> >>        > 802.1x but can work as pass-through. This way I can
still
> >>        use the PCs
> >>        > with 802.1x and the phones in the same Ethernet plug. 
> >>        >
> >>        > Did someone made it with the Cisco IP phones? What
> >>        configuration do I
> >>        > need in the phones and in the switch?
> >>        > Thanks
> >>        > Joao Pereira
> >>        >
> >>        If configuring with Cisco switches, I'm pretty sure
they
pull
> >>        the 
> >>        information for which VLAN to operate in from the switch.
You
> >>        have to
> >>        configure the switchports on the Cisco switch like so:
> >>        
> >>        interface fastethernet 0/1
> >>           switchport trunk native vlan <your data vlan> 
> >>           switchport mode trunk
> >>           switchport voice vlan <your voice vlan>
> >>           spanning-tree portfast trunk
> >>        
> >>        etc.
> >>        
> >>        Thanks,
> >>        Nicholas Kathmann, CISSP
> >>        Kathmann Consulting, LLC
> >>        
> >>        _______________________________________________ 
> >>        --Bandwidth and Colocation provided by Easynews.com --
> >>        
> >>        Asterisk-Users mailing list
> >>        To UNSUBSCRIBE or update options visit:
> >>           http://lists.digium.com/mailman/listinfo/asterisk-users
> >>
> >>_______________________________________________
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> >>    
> >>
> >
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> >  
> >
> 
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------------------------------

Message: 10
Date: Thu, 2 Mar 2006 13:46:57 -0500
From: "Wojciech Tryc" <Wojciech.Tryc@pikatech.com>
Subject: RE: [Asterisk-Users] OT - Cisco IP Phone and PC in different
	VLANs(with802.1x)
To: <joao.pereira@fccn.pt>, "Wojciech Tryc"
<wojtek@VoIPMan.ORG>,
	<asterisk-users@lists.digium.com>
Message-ID:
	<C27FDFC2C3916348AD20F6B44605A949037A55D3@srv00020.kanatek.com>
Content-Type: text/plain;	charset="us-ascii"

Switch is only tagging the vlan packets. Once the PC loads the vlan
aware driver ("client") it will be able to read tagged packet for the
vlan which PC has been configured to use. Nothing to be done on the
switch. 
W

-----Original Message-----
From: Joao Pereira [mailto:joao.pereira@fccn.pt] 
Sent: Thursday, March 02, 2006 1:15 PM
To: Wojciech Tryc; asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] OT - Cisco IP Phone and PC in diferent
VLANs(with802.1x)

Ok, but the PC has an 802.1x client that configures the VLAN when he 
authenticates.
Is this going to pass through the phone?
And will the switch accept it?
Thanks
Joao Pereira


Wojciech Tryc wrote:
> Your pc has to able to support tagged vlans. The switch on the phone 
> will pass through both tagged and untagged vlans.
> W
> ----- Original Message ----- From: "Joao Pereira"
<joao.pereira@fccn.pt>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
> <asterisk-users@lists.digium.com>
> Sent: Thursday, March 02, 2006 11:51 AM
> Subject: Re: [Asterisk-Users] OT - Cisco IP Phone and PC in diferent 
> VLANs(with 802.1x)
>
>
>> And about the 802.1x ?
>> The phones can work as passthrough and force the PC to use 802.1x ?
>> What configuration do we put in the switches? Do we put the switch as
>> "access" (with 802.1x) or "trunk" (without 802.1x)
?
>>
>> Thanks
>> Joao Pereira
>>
>>
>>
>> Greg Oliver wrote:
>>
>>> It actually depends on the switch model.  Some put the port into
>>> trunking mode automatically with the sw voi command, and some do
not.
>>>
>>> Hopefully one day Cisco will finally make their own products and
become
>>> uniform instead of buying several companies and glue'ing them
all
>>> together to get an ethernet switch that works.  At least they got
the
>>> routers right :)
>>>
>>> On Thu, 2006-03-02 at 08:13 -0800, Gary Richardson wrote:
>>>
>>>> You don't need switchport mode trunk when using switchport
voice
>>>> vlan..
>>>> On 3/1/06, Nicholas Kathmann
>>>> <nicholas.kathmann@kathmannconsulting.com> wrote:
>>>>        Joao Pereira wrote:
>>>>        > Hello to all > I would like to know If some of
you have
>>>> already configured
>>>>        an Cisco
>>>>        > IP Phone (7940 or 7960) to work in a different VLAN
than
the
>>>>        PC that
>>>>        > is connected through the phone switch?
>>>>        > I know that this can be done with the Skinny
firmware, but
I
>>>>        dont if > it works with the SIP firmware.
>>>>        >
>>>>        > The Cisco technical staff told me that these phones
dont
>>>>        support
>>>>        > 802.1x but can work as pass-through. This way I can
still
>>>>        use the PCs
>>>>        > with 802.1x and the phones in the same Ethernet
plug. >
>>>>        > Did someone made it with the Cisco IP phones? What
>>>>        configuration do I
>>>>        > need in the phones and in the switch?
>>>>        > Thanks
>>>>        > Joao Pereira
>>>>        >
>>>>        If configuring with Cisco switches, I'm pretty sure
they
pull
>>>>        the information for which VLAN to operate in from the 
>>>> switch. You
>>>>        have to
>>>>        configure the switchports on the Cisco switch like so:
>>>>        interface fastethernet 0/1
>>>>           switchport trunk native vlan <your data vlan>
switchport
>>>> mode trunk
>>>>           switchport voice vlan <your voice vlan>
>>>>           spanning-tree portfast trunk
>>>>        etc.
>>>>        Thanks,
>>>>        Nicholas Kathmann, CISSP
>>>>        Kathmann Consulting, LLC
>>>>        _______________________________________________
--Bandwidth
>>>> and Colocation provided by Easynews.com --
>>>>        Asterisk-Users mailing list
>>>>        To UNSUBSCRIBE or update options visit:
>>>>          
http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>
>>>> _______________________________________________
>>>> --Bandwidth and Colocation provided by Easynews.com --
>>>>
>>>> Asterisk-Users mailing list
>>>> To UNSUBSCRIBE or update options visit:
>>>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>
>>>
>>> _______________________________________________
>>> --Bandwidth and Colocation provided by Easynews.com --
>>>
>>> Asterisk-Users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>
>> _______________________________________________
>> --Bandwidth and Colocation provided by Easynews.com --
>>
>> Asterisk-Users mailing list
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>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
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