[asterisk-users] Disappearing B-Channels

2008-02-09 Thread Mark Greene 
In my efforts to solve a mystery of asterisk slowly loosing it's ability to
take incoming and outgoing calls I set asterisk to restart b-channels every
60 seconds hoping I would find something odd after some time.

So now I am looking at the CLI a few hours later and look what happens when
asterisk restarts the 23 b-channels I have.

pbx1*CLI>
-- B-channel 0/19 successfully restarted on span 1
-- B-channel 0/21 successfully restarted on span 1
  == Primary D-Channel on span 1 down
[Feb 10 01:41:23] WARNING[4102]: chan_zap.c:2401 pri_find_dchan: No
D-channels available!  Using Primary channel 24 as D-channel anyway!
[Feb 10 01:41:24] ERROR[4102]: chan_zap.c:8200 zt_pri_error: !! Got S-frame
while link down
  == Primary D-Channel on span 1 up
-- B-channel 0/19 successfully restarted on span 1
-- B-channel 0/21 successfully restarted on span 1
-- B-channel 0/23 successfully restarted on span 1
pbx1*CLI>


That's the output while I've been writing this email. Those are TWO restarts
of the b-channels. Notice I am missing a seizable amount of my 23
b-channels.

Where are they going?! How do I find out?

I've recompiled my asterisk, zaptel, and libpri to the most recent versions
but that's made no difference.

- Mark
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Dialing SIP server user extension... Dial string issue...

2008-02-09 Thread Rob Hillis 
Why are you specifying the password and server IP in the dial string
when it's included in sip.conf?  It's unnecessary.

I believe that Dial(SIP/gs102/1234) will achieve what you want.

ast guy wrote:
> Hi,
>
>  I'm trying to call a SIP server while providing the SIP server
> username/password in dial string but it's not working ...
>
> Dial(SIP/gs102:[EMAIL PROTECTED] );
>
> User on sip server (192.168.2.81 ):
>
> [gs102]
> disallow=all
> allow=ulaw
> allow=alaw
> type=friend
> username=gs102
> secret=test
> host=dynamic
> dtmfmode=inband
> defaultip=192.168.2.1 
> qualify=1000
> mailbox=102
> context=context-gs102
>
> Extensions.conf entry
>
> [context-gs102]
>
> exten => s,1, Answer();
> exten => s,n, Playback(demo-congrats);
> exten => s,n, Meetme(8600051);
>
> exten => 1234,1, Answer();
> exten => 1234,n, Playback(demo-congrats);
> exten => 1234,n, Meetme(8600051);
>
>
> When I dial I get following error on console
>
>-- Executing Dial("SIP/331-6263", "SIP/gs102:[EMAIL PROTECTED]
> ") in new stack
> -- Called gs102:[EMAIL PROTECTED] 
> -- SIP/192.168.2.81-0343 is circuit-busy
>   == Everyone is busy/congested at this time (1:0/1/0)
> -- Executing Hangup("SIP/331-6263", "") in new stack
>   == Spawn extension (default, 1234, 2) exited non-zero on 'SIP/331-6263'
>
>
> I want to call extension 1234 defined under gs102 defined
> context-gs102 context... what should be the exact Dialed SIP URL ?
>
>
> -ag
> 
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] voicemail to non-default context user does not work

2008-02-09 Thread Rob Hillis 
According to voip-info, the syntax for the VoiceMail command is as
follows...

VoiceMail([/flags/]/[EMAIL PROTECTED]&[EMAIL PROTECTED]&boxnumber3]/)


If you check the syntax for the VoiceMail command, it indicates that the
mailbox parameter is /not/ optional, so I'm surprised this works at
all.  Asterisk will default to the @default context if the context isn't
specified, so you /might/ try Voicemail(@03) otherwise I suspect you're
going to need an IVR to achieve what you want.

Zen Kato wrote:
> Hi,
>
> I input "0203#" after "mailbox?" voice prompt from Voicemail cmd
> on extensions.conf such as
>
> exten => 0021,1,Ringing
> exten => 0021,2,Wait(1)
> exten => 0021,3,Voicemail
> exten => 0021,4,Hangup
>
> *CLI> -- Executing [EMAIL PROTECTED]:1] Ringing("SIP/0103-09a308b0", "") 
> in new stack
> -- Executing [EMAIL PROTECTED]:2] Wait("SIP/0103-09a308b0", "1") in new 
> stack
> -- Executing [EMAIL PROTECTED]:3] VoiceMail("SIP/0103-09a308b0", "") in 
> new stack
> --  Playing 'vm-whichbox' (language 'en')
> [Feb  9 17:11:54] WARNING[3574]: app_voicemail.c:2850 leave_voicemail: No 
> entry in voicemail config file for '0203'
> -- Executing [EMAIL PROTECTED]:4] Hangup("SIP/0103-09a308b0", "") in new 
> stack
>   == Spawn extension (sip, 0021, 4) exited non-zero on 'SIP/0103-09a308b0'
> -- Executing [EMAIL PROTECTED]:1] Hangup("SIP/0103-09a308b0", "") in new 
> stack
>   == Spawn extension (sip, h, 1) exited non-zero on 'SIP/0103-09a308b0'
>
> I have an entry of 0203 at Context 03 on voicemail.conf as follows;
>
> *CLI> voicemail show users
> ContextMbox  User  Zone   NewMsg
> defaultgeneral New User  0
> 03 01030
> 03 02030
> 03 03030
>
> But, I could not enter into "[EMAIL PROTECTED]" mailbox.
> Any idea?
>
> asterisk-1.4.18
>
> --
> Zen
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>   
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] oneway audio with asterisk behind cisco pix 506

2008-02-09 Thread Wendell Hamilton 
Did you only open up the one port (1)?  You need to open up a range, if 
you're doing it this way, like 1-10020 and then set your rtp ports in 
asterisk to the same range. 

- "Ravichandran Rajagopal" <[EMAIL PROTECTED]> wrote:
> I made the following changes and I am still facing one way audio with
> my call flow.
> 
> -Original Message-
> From: Wendell Hamilton [mailto:[EMAIL PROTECTED] 
> Sent: Saturday, February 09, 2008 1:58 PM
> To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
> Discussion
> Cc: Joris Cras; [EMAIL PROTECTED]; Asterisk Users Mailing List -
> Non-Commercial Discussion
> Subject: Re: [asterisk-users] oneway audio with asterisk behind cisco
> pix 506
> 
> try:
> access-list asterisk permit udp any host x.x.x.x eq 1
> 
> - "Ravichandran Rajagopal" <[EMAIL PROTECTED]>
> wrote:
> > I tried the following ACL command
> > 
> > "access-list asterisk permit udp 0.0.0.0 192.168.5.0  range 1
> > 2"
> > 
> > and I got the following response back
> > 
> > "[no] access-list  [line ] deny|permit icmp
> >   | interface  | object-group
> > 
> >   | interface  | object-group
> > 
> > [ | object-group ]
> > [log [disable|default] | [] [interval ]]
> > Restricted ACLs for route-map use:
> > [no] access-list  deny|permit {any |   | host
> > }
> > Command failed"
> > 
> > I don't know how to enter into the linux interface of the Cisco Pix
> > 506
> > firewall
> > 
> > 
> > 
> > -Original Message-
> > From: Joris Cras [mailto:[EMAIL PROTECTED] 
> > Sent: Saturday, February 09, 2008 3:23 AM
> > To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
> Non-Commercial
> > Discussion
> > Subject: Re: [asterisk-users] oneway audio with asterisk behind
> cisco
> > pix
> > 506
> > 
> > Ravi,
> > 
> > there is a easy way of creating all those commands in linux.
> > just run the following in a shell:
> > for x in $(seq 10001 10050); do echo 192.168.5.0 eq $x any conduit
> > permit udp host 192.168.5.0 eq $x any conduit permit udp host;done
> > 
> > This will create all your PIX rules at ones.
> >  
> > I think you could also use Cisco ACL's
> >  access-list [name] permit udp [source] [destination] range
> > This would be in your case something like:
> >  access-list asterisk permit udp 0.0.0.0 192.168.5.0  range 1
> > 10050
> > 
> > Good luck.
> > 
> > Joris
> > 
> > Ravichandran Rajagopal wrote:
> > > Otis,
> > > I wanted to clarify what you said and what I comprehended. 
> > >
> > > the SIP protocols are disabled in fixup. 
> > > 
> > > Having said that I guess all I have to do is just the following.
> > > the inside IP of asterisk server is 192.168.5.0
> > >
> > > On the cisco PIX firewall enter the following.
> > > 192.168.5.0 eq 1 any conduit permit udp host 192.168.5.0 eq
> > 10001 any
> > > conduit permit udp host
> > > 192.168.5.0 eq 10001 any conduit permit udp host 192.168.5.0 eq
> > 10002 any
> > > conduit permit udp host
> > > 
> > > ...
> > > .
> > > 192.168.5.0 eq 10049 any conduit permit udp host 192.168.5.0 eq
> > 10050 any
> > > conduit permit udp host
> > >
> > > in the rtp.conf in /etc/asterisk 
> > > change the ending port 2 (which is what it currently is) to
> > 10050 
> > >
> > > Is there an easier way to make the entries in Cisco PIX firewall
> ?
> > >
> > > Thx
> > > Ravi 
> > >
> > > -Original Message-
> > > From: ListAcct [mailto:[EMAIL PROTECTED] 
> > > Sent: Saturday, February 09, 2008 12:18 AM
> > > To: [EMAIL PROTECTED]
> > > Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> > > Subject: Re: [asterisk-users] oneway audio with asterisk behind
> > cisco pix
> > > 506
> > >
> > > No problem.  :-P  I thought it might wise to include everything
> you
> > 
> > > needed just in case!! LOL! You are welcome!!!
> > >
> > > --Otis 
> > >
> > > Ravichandran Rajagopal wrote:
> > >   
> > >> LOL I guess all I was asking for the changes to be made in the
> > Cisco PIX
> > >> 506. I think you gave me a short tutorial on VI as well. Thanks
> > once
> > again
> > >> for this help. Let me work on these changes and test the one-way
> > audio
> > >> problem and go from there.
> > >> Thx
> > >> Ravi
> > >>
> > >> -Original Message-
> > >> From: ListAcct [mailto:[EMAIL PROTECTED] 
> > >> Sent: Friday, February 08, 2008 11:55 PM
> > >> To: [EMAIL PROTECTED]
> > >> Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> > >> Subject: Re: [asterisk-users] oneway audio with asterisk behind
> > cisco pix
> > >> 506
> > >>
> > >> Ravi,
> > >>
> > >> I will explain changing the config in asterisk and the pix:
> > >>
> > >> Asterisk Box - vi to /etc/asterisk/rtp.conf and change the port
> > span to 
> > >> 1 to 10050 (to start, you will need to increase later as
> ports
> > fill
> > >> 
> > > up)
> > >   
> > >> (use insert to make a change in a f

[asterisk-users] Carrier SIP resource?

2008-02-09 Thread John 
Dear List:

Can anyone refer me to a resource to better understand how the SIP protocol
is used by carriers to provide voice circuits between * and the PSTN?

Thanks a bunch!
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] oneway audio with asterisk behind cisco pix 506

2008-02-09 Thread Ravichandran Rajagopal 
I made the following changes and I am still facing one way audio with my call 
flow.

-Original Message-
From: Wendell Hamilton [mailto:[EMAIL PROTECTED] 
Sent: Saturday, February 09, 2008 1:58 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Joris Cras; [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial 
Discussion
Subject: Re: [asterisk-users] oneway audio with asterisk behind cisco pix 506

try:
access-list asterisk permit udp any host x.x.x.x eq 1

- "Ravichandran Rajagopal" <[EMAIL PROTECTED]> wrote:
> I tried the following ACL command
> 
> "access-list asterisk permit udp 0.0.0.0 192.168.5.0  range 1
> 2"
> 
> and I got the following response back
> 
> "[no] access-list  [line ] deny|permit icmp
> | interface  | object-group
> 
> | interface  | object-group
> 
>   [ | object-group ]
>   [log [disable|default] | [] [interval ]]
> Restricted ACLs for route-map use:
> [no] access-list  deny|permit {any |   | host
> }
> Command failed"
> 
> I don't know how to enter into the linux interface of the Cisco Pix
> 506
> firewall
> 
> 
> 
> -Original Message-
> From: Joris Cras [mailto:[EMAIL PROTECTED] 
> Sent: Saturday, February 09, 2008 3:23 AM
> To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
> Discussion
> Subject: Re: [asterisk-users] oneway audio with asterisk behind cisco
> pix
> 506
> 
> Ravi,
> 
> there is a easy way of creating all those commands in linux.
> just run the following in a shell:
> for x in $(seq 10001 10050); do echo 192.168.5.0 eq $x any conduit
> permit udp host 192.168.5.0 eq $x any conduit permit udp host;done
> 
> This will create all your PIX rules at ones.
>  
> I think you could also use Cisco ACL's
>  access-list [name] permit udp [source] [destination] range
> This would be in your case something like:
>  access-list asterisk permit udp 0.0.0.0 192.168.5.0  range 1
> 10050
> 
> Good luck.
> 
> Joris
> 
> Ravichandran Rajagopal wrote:
> > Otis,
> > I wanted to clarify what you said and what I comprehended. 
> >
> > the SIP protocols are disabled in fixup. 
> > 
> > Having said that I guess all I have to do is just the following.
> > the inside IP of asterisk server is 192.168.5.0
> >
> > On the cisco PIX firewall enter the following.
> > 192.168.5.0 eq 1 any conduit permit udp host 192.168.5.0 eq
> 10001 any
> > conduit permit udp host
> > 192.168.5.0 eq 10001 any conduit permit udp host 192.168.5.0 eq
> 10002 any
> > conduit permit udp host
> > 
> > ...
> > .
> > 192.168.5.0 eq 10049 any conduit permit udp host 192.168.5.0 eq
> 10050 any
> > conduit permit udp host
> >
> > in the rtp.conf in /etc/asterisk 
> > change the ending port 2 (which is what it currently is) to
> 10050 
> >
> > Is there an easier way to make the entries in Cisco PIX firewall ?
> >
> > Thx
> > Ravi 
> >
> > -Original Message-
> > From: ListAcct [mailto:[EMAIL PROTECTED] 
> > Sent: Saturday, February 09, 2008 12:18 AM
> > To: [EMAIL PROTECTED]
> > Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> > Subject: Re: [asterisk-users] oneway audio with asterisk behind
> cisco pix
> > 506
> >
> > No problem.  :-P  I thought it might wise to include everything you
> 
> > needed just in case!! LOL! You are welcome!!!
> >
> > --Otis 
> >
> > Ravichandran Rajagopal wrote:
> >   
> >> LOL I guess all I was asking for the changes to be made in the
> Cisco PIX
> >> 506. I think you gave me a short tutorial on VI as well. Thanks
> once
> again
> >> for this help. Let me work on these changes and test the one-way
> audio
> >> problem and go from there.
> >> Thx
> >> Ravi
> >>
> >> -Original Message-
> >> From: ListAcct [mailto:[EMAIL PROTECTED] 
> >> Sent: Friday, February 08, 2008 11:55 PM
> >> To: [EMAIL PROTECTED]
> >> Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> >> Subject: Re: [asterisk-users] oneway audio with asterisk behind
> cisco pix
> >> 506
> >>
> >> Ravi,
> >>
> >> I will explain changing the config in asterisk and the pix:
> >>
> >> Asterisk Box - vi to /etc/asterisk/rtp.conf and change the port
> span to 
> >> 1 to 10050 (to start, you will need to increase later as ports
> fill
> >> 
> > up)
> >   
> >> (use insert to make a change in a file)
> >>
> >> to save:
> >>
> >>1. esc
> >>2. shift + colon
> >>3. wq (to save)
> >>
> >> If you made a mistake and do not want to save but you changed
> something 
> >> in the file:
> >>
> >>1. esc
> >>2. shift + colon
> >>3. q! (to exit)
> >>
> >>
> >> Cisco Pix - on my old Pix 520 UR I do not use the ACLs for this
> case the 
> >> static and conduit commands so this is a example from my setup.
> >>
> >> Theses are not usable IPs on the Internet or my IPs but just an
> >> 
> > example
> >   
> >> outside

Re: [asterisk-users] BLF and Asterisk 1.6.0b2

2008-02-09 Thread Thomas Kenyon 
Russell Bryant wrote:
> Thomas Kenyon wrote:
>> Is anyone else having trouble with Asterisk 1.6.0b2 not sending busy 
>> hints to phones?
>>
>> I'm not reporting this a s a bug because (although I have it working 
>> with Asterisk 1.4.17, the hardware involved is different.
> 
> What type of device are you subscribing to, is it another SIP phone?  If so, 
> what is the associated configuration in sip.conf?  Do you have call-limit set 
> to 
> some value, or the combination of callcounter and busylevel?  If so, what are 
> they set to?  (You must have these options set for it to work)
> 
I have enough kit around to set the machine I'm testing 1.6.0b2 to use 
the same configuration as the working machines.

I have got call-limits set, but it did occur to me that there's no 
reason asterisk would know that there is only one extension on 
SIP/.

The stranger thing is, on the machine that it's all working on, there is 
a call-limit=4 set on every extension (from what I remember it prevented 
a bug that got fixed ages ago and I didn't get round to lowering it again).

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] HP proliant and hpasm

2008-02-09 Thread Steven 
Is anyone successfully running asterisk on an HP proliant while using 
their management software, hpasm?

I have two DL360's and two TE220B's.  The cards have their own IRQ's.  
No matter what combination of settings I use, the cards fail the 
patlooptest if hpasm (ver 7.9.1) is running.  If I stop it the cards 
pass the test.

I really want to run the management software, so I'd like to know if 
anyone has it working.

Thanks.

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Upgrade 1.2 -> 1.4 voice files

2008-02-09 Thread Russell Bryant 
Adrian Marsh wrote:
> In the Make menuselect, I noticed theres no .SLN voicefile selection for
> the basic audiofiles - has SLN been depreciated?

No, the sln format is still supported.  We have just never distributed any 
files 
in that raw format.  Previously, we only had gsm recordings.  For Asterisk 1.4, 
we got all of the prompts re-recorded so that we could distribute them in a 
number of higher-quality codecs, as well as in 3 languages.

The actually scripts of the files has not changed much, as far as I remember. 
The sounds.txt file in 1.2, and the 1.4 sounds packages should say exactly what 
they are.  You can always compare them with diff.

-- 
Russell Bryant
Senior Software Engineer
Open Source Team Lead
Digium, Inc.

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] BLF and Asterisk 1.6.0b2

2008-02-09 Thread Russell Bryant 
Thomas Kenyon wrote:
> Is anyone else having trouble with Asterisk 1.6.0b2 not sending busy 
> hints to phones?
> 
> I'm not reporting this a s a bug because (although I have it working 
> with Asterisk 1.4.17, the hardware involved is different.

What type of device are you subscribing to, is it another SIP phone?  If so, 
what is the associated configuration in sip.conf?  Do you have call-limit set 
to 
some value, or the combination of callcounter and busylevel?  If so, what are 
they set to?  (You must have these options set for it to work)

-- 
Russell Bryant
Senior Software Engineer
Open Source Team Lead
Digium, Inc.

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] oneway audio with asterisk behind cisco pix 506

2008-02-09 Thread Wendell Hamilton 
try:
access-list asterisk permit udp any host x.x.x.x eq 1

- "Ravichandran Rajagopal" <[EMAIL PROTECTED]> wrote:
> I tried the following ACL command
> 
> "access-list asterisk permit udp 0.0.0.0 192.168.5.0  range 1
> 2"
> 
> and I got the following response back
> 
> "[no] access-list  [line ] deny|permit icmp
> | interface  | object-group
> 
> | interface  | object-group
> 
>   [ | object-group ]
>   [log [disable|default] | [] [interval ]]
> Restricted ACLs for route-map use:
> [no] access-list  deny|permit {any |   | host
> }
> Command failed"
> 
> I don't know how to enter into the linux interface of the Cisco Pix
> 506
> firewall
> 
> 
> 
> -Original Message-
> From: Joris Cras [mailto:[EMAIL PROTECTED] 
> Sent: Saturday, February 09, 2008 3:23 AM
> To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
> Discussion
> Subject: Re: [asterisk-users] oneway audio with asterisk behind cisco
> pix
> 506
> 
> Ravi,
> 
> there is a easy way of creating all those commands in linux.
> just run the following in a shell:
> for x in $(seq 10001 10050); do echo 192.168.5.0 eq $x any conduit
> permit udp host 192.168.5.0 eq $x any conduit permit udp host;done
> 
> This will create all your PIX rules at ones.
>  
> I think you could also use Cisco ACL's
>  access-list [name] permit udp [source] [destination] range
> This would be in your case something like:
>  access-list asterisk permit udp 0.0.0.0 192.168.5.0  range 1
> 10050
> 
> Good luck.
> 
> Joris
> 
> Ravichandran Rajagopal wrote:
> > Otis,
> > I wanted to clarify what you said and what I comprehended. 
> >
> > the SIP protocols are disabled in fixup. 
> > 
> > Having said that I guess all I have to do is just the following.
> > the inside IP of asterisk server is 192.168.5.0
> >
> > On the cisco PIX firewall enter the following.
> > 192.168.5.0 eq 1 any conduit permit udp host 192.168.5.0 eq
> 10001 any
> > conduit permit udp host
> > 192.168.5.0 eq 10001 any conduit permit udp host 192.168.5.0 eq
> 10002 any
> > conduit permit udp host
> > 
> > ...
> > .
> > 192.168.5.0 eq 10049 any conduit permit udp host 192.168.5.0 eq
> 10050 any
> > conduit permit udp host
> >
> > in the rtp.conf in /etc/asterisk 
> > change the ending port 2 (which is what it currently is) to
> 10050 
> >
> > Is there an easier way to make the entries in Cisco PIX firewall ?
> >
> > Thx
> > Ravi 
> >
> > -Original Message-
> > From: ListAcct [mailto:[EMAIL PROTECTED] 
> > Sent: Saturday, February 09, 2008 12:18 AM
> > To: [EMAIL PROTECTED]
> > Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> > Subject: Re: [asterisk-users] oneway audio with asterisk behind
> cisco pix
> > 506
> >
> > No problem.  :-P  I thought it might wise to include everything you
> 
> > needed just in case!! LOL! You are welcome!!!
> >
> > --Otis 
> >
> > Ravichandran Rajagopal wrote:
> >   
> >> LOL I guess all I was asking for the changes to be made in the
> Cisco PIX
> >> 506. I think you gave me a short tutorial on VI as well. Thanks
> once
> again
> >> for this help. Let me work on these changes and test the one-way
> audio
> >> problem and go from there.
> >> Thx
> >> Ravi
> >>
> >> -Original Message-
> >> From: ListAcct [mailto:[EMAIL PROTECTED] 
> >> Sent: Friday, February 08, 2008 11:55 PM
> >> To: [EMAIL PROTECTED]
> >> Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> >> Subject: Re: [asterisk-users] oneway audio with asterisk behind
> cisco pix
> >> 506
> >>
> >> Ravi,
> >>
> >> I will explain changing the config in asterisk and the pix:
> >>
> >> Asterisk Box - vi to /etc/asterisk/rtp.conf and change the port
> span to 
> >> 1 to 10050 (to start, you will need to increase later as ports
> fill
> >> 
> > up)
> >   
> >> (use insert to make a change in a file)
> >>
> >> to save:
> >>
> >>1. esc
> >>2. shift + colon
> >>3. wq (to save)
> >>
> >> If you made a mistake and do not want to save but you changed
> something 
> >> in the file:
> >>
> >>1. esc
> >>2. shift + colon
> >>3. q! (to exit)
> >>
> >>
> >> Cisco Pix - on my old Pix 520 UR I do not use the ACLs for this
> case the 
> >> static and conduit commands so this is a example from my setup.
> >>
> >> Theses are not usable IPs on the Internet or my IPs but just an
> >> 
> > example
> >   
> >> outside (interface) - 192.168.1.0/24 (192.168.1.1-192.168.1.254)
> >> dmz (interface) - 192.168.254.0/24 (192.168.254.1-192.168.254.254)
> >>
> >> interface ethernet0 100full (sets the duplex and turns on
> interface)
> >> interface ethernet1 100full (sets the duplex and turns on
> interface)
> >>
> >> nameif ethernet0 outside security0 ( lower security)
> >> nameif ethernet1 dmz security50 (higher security)
> >>
> >> no fixup protocol sip 5060
> >> no fixup

Re: [asterisk-users] oneway audio with asterisk behind cisco pix 506

2008-02-09 Thread Ravichandran Rajagopal 
I tried the following ACL command

"access-list asterisk permit udp 0.0.0.0 192.168.5.0  range 1 2"

and I got the following response back

"[no] access-list  [line ] deny|permit icmp
  | interface  | object-group

  | interface  | object-group

[ | object-group ]
[log [disable|default] | [] [interval ]]
Restricted ACLs for route-map use:
[no] access-list  deny|permit {any |   | host }
Command failed"

I don't know how to enter into the linux interface of the Cisco Pix 506
firewall



-Original Message-
From: Joris Cras [mailto:[EMAIL PROTECTED] 
Sent: Saturday, February 09, 2008 3:23 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [asterisk-users] oneway audio with asterisk behind cisco pix
506

Ravi,

there is a easy way of creating all those commands in linux.
just run the following in a shell:
for x in $(seq 10001 10050); do echo 192.168.5.0 eq $x any conduit
permit udp host 192.168.5.0 eq $x any conduit permit udp host;done

This will create all your PIX rules at ones.
 
I think you could also use Cisco ACL's
 access-list [name] permit udp [source] [destination] range
This would be in your case something like:
 access-list asterisk permit udp 0.0.0.0 192.168.5.0  range 1 10050

Good luck.

Joris

Ravichandran Rajagopal wrote:
> Otis,
> I wanted to clarify what you said and what I comprehended. 
>
> the SIP protocols are disabled in fixup. 
> 
> Having said that I guess all I have to do is just the following.
> the inside IP of asterisk server is 192.168.5.0
>
> On the cisco PIX firewall enter the following.
> 192.168.5.0 eq 1 any conduit permit udp host 192.168.5.0 eq 10001 any
> conduit permit udp host
> 192.168.5.0 eq 10001 any conduit permit udp host 192.168.5.0 eq 10002 any
> conduit permit udp host
> 
> ...
> .
> 192.168.5.0 eq 10049 any conduit permit udp host 192.168.5.0 eq 10050 any
> conduit permit udp host
>
> in the rtp.conf in /etc/asterisk 
> change the ending port 2 (which is what it currently is) to 10050 
>
> Is there an easier way to make the entries in Cisco PIX firewall ?
>
> Thx
> Ravi 
>
> -Original Message-
> From: ListAcct [mailto:[EMAIL PROTECTED] 
> Sent: Saturday, February 09, 2008 12:18 AM
> To: [EMAIL PROTECTED]
> Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: Re: [asterisk-users] oneway audio with asterisk behind cisco pix
> 506
>
> No problem.  :-P  I thought it might wise to include everything you 
> needed just in case!! LOL! You are welcome!!!
>
> --Otis 
>
> Ravichandran Rajagopal wrote:
>   
>> LOL I guess all I was asking for the changes to be made in the Cisco PIX
>> 506. I think you gave me a short tutorial on VI as well. Thanks once
again
>> for this help. Let me work on these changes and test the one-way audio
>> problem and go from there.
>> Thx
>> Ravi
>>
>> -Original Message-
>> From: ListAcct [mailto:[EMAIL PROTECTED] 
>> Sent: Friday, February 08, 2008 11:55 PM
>> To: [EMAIL PROTECTED]
>> Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion'
>> Subject: Re: [asterisk-users] oneway audio with asterisk behind cisco pix
>> 506
>>
>> Ravi,
>>
>> I will explain changing the config in asterisk and the pix:
>>
>> Asterisk Box - vi to /etc/asterisk/rtp.conf and change the port span to 
>> 1 to 10050 (to start, you will need to increase later as ports fill
>> 
> up)
>   
>> (use insert to make a change in a file)
>>
>> to save:
>>
>>1. esc
>>2. shift + colon
>>3. wq (to save)
>>
>> If you made a mistake and do not want to save but you changed something 
>> in the file:
>>
>>1. esc
>>2. shift + colon
>>3. q! (to exit)
>>
>>
>> Cisco Pix - on my old Pix 520 UR I do not use the ACLs for this case the 
>> static and conduit commands so this is a example from my setup.
>>
>> Theses are not usable IPs on the Internet or my IPs but just an
>> 
> example
>   
>> outside (interface) - 192.168.1.0/24 (192.168.1.1-192.168.1.254)
>> dmz (interface) - 192.168.254.0/24 (192.168.254.1-192.168.254.254)
>>
>> interface ethernet0 100full (sets the duplex and turns on interface)
>> interface ethernet1 100full (sets the duplex and turns on interface)
>>
>> nameif ethernet0 outside security0 ( lower security)
>> nameif ethernet1 dmz security50 (higher security)
>>
>> no fixup protocol sip 5060
>> no fixup protocol sip udp 5060
>>
>> ! - this makes things easier so now the pix knows the IP of the asterisk 
>> box and maps the ip to the name just for configuration purposes only so 
>> if you had 20 servers or devices you wanted public access to it's just 
>> easier to remember their names versus IPs.
>> name 192.168.254.11 dns
>> name 192.168.254.10 asterisk
>>
>> ! - the static command is used as a permanent mapper from one inside, 
>> dmz, or other to

Re: [asterisk-users] [asterisk-dev] Monitor Asterisk using C

2008-02-09 Thread Soumya Kat 
>Soumya Kat wrote:
> What I would like to know is how to get information such as SIP users,
> number of SIP connections and traffic associated with those from asterisk
> using a C Code.

>Russell Bryant
> There is actually no good way to do this inside of Asterisk right now.
 It's
> certainly all possible ... it's just software ... but there is no
> straightforward common API to accomplish what you are looking for.


Well then how can I monitor asterisk using net-SNMP. There should be someway
in which I can monitor asterisk.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Cisco phone 79xx get database information

2008-02-09 Thread Doug Lytle 
Javier Temponi wrote:
> Hi, may be this question is a bit silly, but I couldn’t find any 
> document or post or anything that say that if this is possible or not.
> I want to show information on my phones cisco 7960/40 when a call 
> arrive. May be a bit more than a caller ID, show more detail level, if 
> is that possible.
> I already have an asterisk and the phones registered there, and I need 
> to show on the phone display, when the call is ringing, the customer 
> information..
>
> Something like this:
> source number: xx
> Customer Number: x
> Name of Customer: x

You may want to check out this bug:

http://bugs.digium.com/view.php?id=8824

Doug

-- 
Ben Franklin quote:

"Those who would give up Essential Liberty to purchase a little Temporary 
Safety, deserve neither Liberty nor Safety."



___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Cisco phone 79xx get database information

2008-02-09 Thread Javier Temponi 
Hi, may be this question is a bit silly, but I couldn¹t find any document or
post or anything that say that if this is possible or not.
I want to show information on my phones cisco 7960/40 when a call arrive.
May be a bit more than a caller ID, show more detail level, if is that
possible.
I already have an asterisk and the phones registered there, and I need to
show on the phone display, when the call is ringing, the customer
information..

Something like this:
source number: xx
Customer Number: x
Name of Customer: x

I have that information on an external database, and I know for the source
number witch customer is calling.

The thing is how can I show on my cisco phone the information that I have on
my database when i receive a call?
Do I have to configure the directory services on the phones to look for the
information on the database?
Can I show those lines on the phone? or I have a limit of lines or info to
show?

Any help would be really appreciated
Thanks in advance!!
cheers
Javito
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] BLF and Asterisk 1.6.0b2

2008-02-09 Thread Thomas Kenyon 
Is anyone else having trouble with Asterisk 1.6.0b2 not sending busy 
hints to phones?

I'm not reporting this a s a bug because (although I have it working 
with Asterisk 1.4.17, the hardware involved is different.

Thanks.

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Monitor Asterisk using C

2008-02-09 Thread Soumya Kat 
Thank you for replying. The probleam is how do I use the
Asterisk_manager-API and implement them in my C code. Like how do I call a
API in my C program. It will be of great help if I can have an example.

By traffic I mean how much bandwidth or data transferring is taking place in
a call that is network traffic. I want to monitor the asterisk server.

Thank you.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] SIP user registration and Asterisk Realtime

2008-02-09 Thread ast guy 
Hi,

 I have installed asterisk real time and sip buddies information is being
stored in DB. Now I have a question,

Asterisk Realtime Server -A
Third party SIP server-B

Question: Is there any configuration in * RT that it can register with
defined sip user on Server-B
I was only able to find sip users information in DB not about user
registration on other server.

-ag
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] External MWI question for Asterisk

2008-02-09 Thread Grey Man 

> - Original Message 

> From: Olivier <[EMAIL PROTECTED]>

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

> Sent: Saturday, 9 February, 2008 6:55:15 AM

> Subject: Re: [asterisk-users] External MWI question for Asterisk



> Do you mean your script does send a NOTIFY messages to hardphones ? Then, how 
> did you write such SIP-aware script (language, ...) ?

If not, how external script and Asterisk do communicate ?



Our MWI program is written in C# but to send SIP NOTIFY requests is very 
straight forward and could be done in anything. Our program is 250 lines of 
code and that includes polling the database to decide when to send the NOTIFY 
requests. In the NOTIFY request the body consists of:

Messages-Waiting: yes|no

That's all you need to turn on and off the MWI indicators on every device we've 
come across. There's nothing tricky to MWI at all.

Regards,

Greyman.




  Get the name you always wanted with the new y7mail email address.
www.yahoo7.com.au/y7mail



___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] How to detect if SIP extension BUSY?

2008-02-09 Thread Csibra Gergo 
Hi,

My problem is in subject. As I read in documentations and
voip-info.org I can't user ChanIsAvalil because it not detects BUSY
information on SIP channel. I've tried to use SIPPEER function, but it
gives "OK (9 ms)" back on BUSY SIP channel. I use Asterisk 1.2.15, SIP
extensions are Linksys PAP2. Have any other idea?

-- 
Best regards,
 Csibra Gergomailto:[EMAIL PROTECTED]


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] oneway audio with asterisk behind cisco pix 506

2008-02-09 Thread Joris Cras 
Ravi,

there is a easy way of creating all those commands in linux.
just run the following in a shell:
for x in $(seq 10001 10050); do echo 192.168.5.0 eq $x any conduit
permit udp host 192.168.5.0 eq $x any conduit permit udp host;done

This will create all your PIX rules at ones.
 
I think you could also use Cisco ACL's
 access-list [name] permit udp [source] [destination] range
This would be in your case something like:
 access-list asterisk permit udp 0.0.0.0 192.168.5.0  range 1 10050

Good luck.

Joris

Ravichandran Rajagopal wrote:
> Otis,
> I wanted to clarify what you said and what I comprehended. 
>
> the SIP protocols are disabled in fixup. 
> 
> Having said that I guess all I have to do is just the following.
> the inside IP of asterisk server is 192.168.5.0
>
> On the cisco PIX firewall enter the following.
> 192.168.5.0 eq 1 any conduit permit udp host 192.168.5.0 eq 10001 any
> conduit permit udp host
> 192.168.5.0 eq 10001 any conduit permit udp host 192.168.5.0 eq 10002 any
> conduit permit udp host
> 
> ...
> .
> 192.168.5.0 eq 10049 any conduit permit udp host 192.168.5.0 eq 10050 any
> conduit permit udp host
>
> in the rtp.conf in /etc/asterisk 
> change the ending port 2 (which is what it currently is) to 10050 
>
> Is there an easier way to make the entries in Cisco PIX firewall ?
>
> Thx
> Ravi 
>
> -Original Message-
> From: ListAcct [mailto:[EMAIL PROTECTED] 
> Sent: Saturday, February 09, 2008 12:18 AM
> To: [EMAIL PROTECTED]
> Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: Re: [asterisk-users] oneway audio with asterisk behind cisco pix
> 506
>
> No problem.  :-P  I thought it might wise to include everything you 
> needed just in case!! LOL! You are welcome!!!
>
> --Otis 
>
> Ravichandran Rajagopal wrote:
>   
>> LOL I guess all I was asking for the changes to be made in the Cisco PIX
>> 506. I think you gave me a short tutorial on VI as well. Thanks once again
>> for this help. Let me work on these changes and test the one-way audio
>> problem and go from there.
>> Thx
>> Ravi
>>
>> -Original Message-
>> From: ListAcct [mailto:[EMAIL PROTECTED] 
>> Sent: Friday, February 08, 2008 11:55 PM
>> To: [EMAIL PROTECTED]
>> Cc: 'Asterisk Users Mailing List - Non-Commercial Discussion'
>> Subject: Re: [asterisk-users] oneway audio with asterisk behind cisco pix
>> 506
>>
>> Ravi,
>>
>> I will explain changing the config in asterisk and the pix:
>>
>> Asterisk Box - vi to /etc/asterisk/rtp.conf and change the port span to 
>> 1 to 10050 (to start, you will need to increase later as ports fill
>> 
> up)
>   
>> (use insert to make a change in a file)
>>
>> to save:
>>
>>1. esc
>>2. shift + colon
>>3. wq (to save)
>>
>> If you made a mistake and do not want to save but you changed something 
>> in the file:
>>
>>1. esc
>>2. shift + colon
>>3. q! (to exit)
>>
>>
>> Cisco Pix - on my old Pix 520 UR I do not use the ACLs for this case the 
>> static and conduit commands so this is a example from my setup.
>>
>> Theses are not usable IPs on the Internet or my IPs but just an
>> 
> example
>   
>> outside (interface) - 192.168.1.0/24 (192.168.1.1-192.168.1.254)
>> dmz (interface) - 192.168.254.0/24 (192.168.254.1-192.168.254.254)
>>
>> interface ethernet0 100full (sets the duplex and turns on interface)
>> interface ethernet1 100full (sets the duplex and turns on interface)
>>
>> nameif ethernet0 outside security0 ( lower security)
>> nameif ethernet1 dmz security50 (higher security)
>>
>> no fixup protocol sip 5060
>> no fixup protocol sip udp 5060
>>
>> ! - this makes things easier so now the pix knows the IP of the asterisk 
>> box and maps the ip to the name just for configuration purposes only so 
>> if you had 20 servers or devices you wanted public access to it's just 
>> easier to remember their names versus IPs.
>> name 192.168.254.11 dns
>> name 192.168.254.10 asterisk
>>
>> ! - the static command is used as a permanent mapper from one inside, 
>> dmz, or other to the global ip vice versa. (Rule of thumb if you map 
>> using static make sure you have a conduit command)
>> static (dmz,outside) 192.168.1.22 asterisk netmask 255.255.255.255 0 0
>>
>> ! - here is where you open the ports on the global side to the asterisk 
>> box. (the conduit command allows connections from lower security 
>> interfaces to higher security interfaces)
>> conduit permit udp host 192.168.1.22 eq 1 any
>> conduit permit udp host 192.168.1.22 eq 10001 any
>> conduit permit udp host 192.168.1.22 eq 10002 any
>> conduit permit udp host 192.168.1.22 eq 10003 any
>> conduit permit udp host 192.168.1.22 eq 10004 any
>> conduit permit udp host 192.168.1.22 eq 10005 any
>>
>> Hope this helps!
>>
>> --Otis
>>
>>
>> Ravichandran Rajagopal wrote:
>>   
>> 
>>> Otis,
>>> I am n

[asterisk-users] voicemail to non-default context user does not work

2008-02-09 Thread Zen Kato 
Hi,

I input "0203#" after "mailbox?" voice prompt from Voicemail cmd
on extensions.conf such as

exten => 0021,1,Ringing
exten => 0021,2,Wait(1)
exten => 0021,3,Voicemail
exten => 0021,4,Hangup

*CLI> -- Executing [EMAIL PROTECTED]:1] Ringing("SIP/0103-09a308b0", "") in 
new stack
-- Executing [EMAIL PROTECTED]:2] Wait("SIP/0103-09a308b0", "1") in new 
stack
-- Executing [EMAIL PROTECTED]:3] VoiceMail("SIP/0103-09a308b0", "") in new 
stack
--  Playing 'vm-whichbox' (language 'en')
[Feb  9 17:11:54] WARNING[3574]: app_voicemail.c:2850 leave_voicemail: No entry 
in voicemail config file for '0203'
-- Executing [EMAIL PROTECTED]:4] Hangup("SIP/0103-09a308b0", "") in new 
stack
  == Spawn extension (sip, 0021, 4) exited non-zero on 'SIP/0103-09a308b0'
-- Executing [EMAIL PROTECTED]:1] Hangup("SIP/0103-09a308b0", "") in new 
stack
  == Spawn extension (sip, h, 1) exited non-zero on 'SIP/0103-09a308b0'

I have an entry of 0203 at Context 03 on voicemail.conf as follows;

*CLI> voicemail show users
ContextMbox  User  Zone   NewMsg
defaultgeneral New User  0
03 01030
03 02030
03 03030

But, I could not enter into "[EMAIL PROTECTED]" mailbox.
Any idea?

asterisk-1.4.18

--
Zen

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users