Re: [asterisk-users] Transfer Call to Cell Phone

[John Faubion]

>We do have full features on our lines so both lines are free once the
>transfer is complete. We also have toll calls on our lines so it would
>not be a problem, so I do not have to worry about AT&T dropping the

The issue really isn't whether you have the ability to make toll calls on
your line. The concern here is in what the regulatory agencies call "toll
bridging" which is using a system to relay a call from one local calling are
to another local calling area to avoid a toll charge. This is one of those
gray areas that can become a problem if your not careful. The problem comes
up if you have customers that can call you as a local call and you are
forwarding them on to another party that is a local call for you but would
be a toll call for the customer. This is essentially what toll bridging is
about. Now your not likely to have to worry about the legal ramifications of
this since your merely connecting the customer with an extension of your
company, namely your salesman. Where this could become a problem for you
would be in transferring the customer using the same pots line. The reason
is that AT&T is handling the transfer. When you transfer the call, it
essentially becomes a new call. The main difference is that you have
provided the called number. So the software in the Class 5 (End office)
switch, takes the number you provide and runs the call through its routing
translations (similar to the Asterisk dialing plan) and if it determines
that the destination number is outside the originators Local Area Transport
Area or LATA, then it will either drop the originator to a message that
says, "You must first dial a 0 or 1 before calling this number" or it may
deny the transfer allowing you to stay connected to the customer. Neither
one looks very professional. The only way around this would be to provide
another line or trunk to pass the call down. Now if your not in an
overlapping LATA this probably isn't an issue.


>The only way I can get it to work is by have the call on the 1st
>line then transfer it out on the 2nd line. After that is complete both
>lines are free.

Are you saying that you are able to route a call from line 1 to line 2 and
have the call transfer, thus freeing the lines or that once the call
completes the lines are freed? I've never seen the first scenario. The
second scenario is the normal behavior.


>Can you give an example of creating an extension which points to a cell
>phone. Secondly how can you have if no one answers an extension it dials
>the cell number next. That maybe answered in the example.

In extensions.conf use something like this.
[global]
SIP-PROV = "sip.urprovider.com"
; Now set the call forward numbers
CFN21 => "551234"  ; These are normally set in an external file

[internal]
exten => 21,1,Macro(stdext,${SIP/21},${CFN${EXTEN}})

[macro-stdext];
; Standard extension macro:
;   ${ARG1} - Device(s) to ring
;   ${ARG2} - Our call forward number
exten => s,1,Dial(${ARG1},10)
exten => s,2,Goto(s-${DIALSTATUS},1)
exten => s-NOANSWER,1,GotoIf($[${LEN(${ARG2})}>0]?s-CFWD,1)
exten => s-NOANSWER,2,Voicemail(${MACRO_EXTEN},u)
exten => s-BUSY,1,Voicemail(${MACRO_EXTEN},b)
exten => s-CFWD,1,Dial(SIP/[EMAIL PROTECTED],20)
exten => s-CFWD,2,Goto(s-NOANSWER,2)
exten => _s-.,1,Goto(s-NOANSWER,2)
exten => a,1,VoicemailMain(${MACRO_EXTEN})


There is more to this but this should show the basics of what we use. I
store my Call Forward Numbers (CFN) in an external file. This allow me to
update the file externally (currently with a web interface but as soon as I
get the prompts recorded it will be done with an IVR) and then just reload
the extensions to activate the new numbers. Also I using SIP for pretty much
everything. Our TDM400 doesn't even have modules, it's just there for
timing. However you should be able to convert the SIP calls to ZAP calls for
you use. The internal context is included in our default context. Dialing
extension 21 calls the stdext macro. This dials the local extension first.
If not answered after 10 seconds, we check to make sure we have a phone
number to send the call out with. If not we send it on to voice mail.
Otherwise we send it to the s-CFWD. The check listed here is a very
rudimentary check but again I hope you get the idea. Next we try the call to
the CFN. If not answered in 20 seconds, then we send it to voice mail.
Finally if the user presses the star button during the attempt, we send them
on to Voicemail mail so they can check their messages.

Hopefully this helps.

John


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Re: [asterisk-users] Ring the second line when 1st line is busy

[Rob Townley]

Isn't this what you are looking for?

http://voipspeak.net/index.php?option=com_content&task=view&id=72&Itemid=28

On 6/26/07, Deepak Naidu <[EMAIL PROTECTED]> wrote:


Do any one any clue.

This is what I need.
I have a Polycom 501 phone, which support multiple lines ie on the LCD you
can see the extensions asssigned to a user as.
 555 --- Line 1 -- Extensions which registers with SIP(Asterisk) -- User A
8555 --- Line 2 -- Extensions which registers with SIP(Asterisk) -- User A
So when now someone calls one Extension 555 to User A, scenario as below.
1) If he is busy on the first line Ext 555, then ring Line 2 Ext 8555, if
no response send to voicemail
2) If he doesnt picks the line 1 Ext 555, then send to voicemail rather
then ringging on his second line ie Ext 8555.
This is what I need, if I can dow it with Follow me, then how, if through
ring group how.


*Deepak Naidu <[EMAIL PROTECTED]>* wrote:

Hi,
I ma using Asterisk 1.2.18 & FreePBX 2.2.1.  I have assigned every
users in office with Polycom with 2 extensions as below

 555
8555

I have configured Follow-me to ring when the users doesn't picks the phone
on line 1(555) after 10 seconds & then ring the line 2(8555).  But this is
not a standard telephony which I have been advised to change like below.

If someone calls Ext 555 & its busy(means on a phone with someone), then
only ring the second line ie 8555 & even if that is busy send on voicemail.

If the first line 555 is free & no one picks up then let it go on the
voicemail & not second line, bcos now no one has picked the phone nor busy.

I could find anyway to do in FreePBX, so was wondering how about doing
this.  Thanx for any input.

exten => 555,1,Macro(exten-vm,555,555)
exten => 555,n,Hangup
exten => 555,hint,SIP/555
exten => ${VM_PREFIX}555,1,Macro(vm,555,DIRECTDIAL)
exten => ${VM_PREFIX}555,n,Hangup


--
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Re: [asterisk-users] kore dump

[Luki]

> just an idea, but maybe qmail, samba, and bind have a smaller memory
> footprint than an in-use asterisk?

No, probably not. Asterisk's is about 20-40 MB depending on the number
of extensions, etc. Smbd's is similar, bind's is actually 90 MB (with
about 600 zones).


> can you take the hardware offline long enough for a memtest?

The machine has been retired (routine upgrade cycle). But I hardly
doubt that was the problem. My guess is it was somehow related to
limited CPU power (thread switching, interrupts, or whatnot). The old
hardware was single CPU and a lot slower.

--Luki

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Re: [asterisk-users] Asterisk GUI

[Paul Hales]

> In Paul's defense, it looked to me like his original post was simply  
> a joke that was misunderstood. (I thought it was funny, anyway)
> 

I have written a few jokes for this list over the years - it's nice to
know that some people find them funny.

PaulH



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Re: [asterisk-users] Transfer Call to Cell Phone

[Ryan Goldberg]

OCOSA ListAcc wrote:
> 
> Can you give an example of creating an extension which points to a cell 
> phone. Secondly how can you have if no one answers an extension it dials 
> the cell number next. That maybe answered in the example. I have the 
> system setup so it just dials out which ever line is not busy. Thanks!

I'm quite new to *, but I've got this in place in my first rendition, and 
I'm pretty sure it does what you want:

exten => 101,1,Dial(SIP/${EXTEN},15,t)
exten => 101,n,Dial(Zap/4/12185551212,30,tpm)
exten => 101,n,VoiceMail([EMAIL PROTECTED])
exten => 101,n,Playback(vm-goodbye)
exten => 101,n,Hangup

caller dials extension 101.  It first tries his desk for 15 seconds, then 
it tries his cell over a zap channel (the 'p' turns on call screening), 
then it finally hits voicemail.  In our actual dialplan, the cell phone 
call goes out over sip, so the line looks like this:

exten => 101,1,Dial(SIP/lesnet/12185551212,30,tpm)

Alternatively, the first line could be:

exten => 101,1,Dial(SIP/${EXTEN}&Zap/4/12185551212,30,tpm)

which would dial both the desk and the cell at the same time...

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

Hope that helps.

Ryan

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[asterisk-users] Module '***.so' did not register itself during load

[Igor Bonny]

Hi, 

I've experiencing this kind of problem. 

Actually, my asterisk is running perfectly. I've tested it, and I called some 
computer in my LAN. Then I enter the CLI and entered these commands
- show modules
- modules status (or so.. I forget)
- restart now

After I enter the last command, the CLI is exiting and nothing happened. Then I 
try to run the asterisk with command
- asterisk

But then nothing is happened. I check with "ps" command, and realize that 
asterisk isn't running at all. When I check the log, i got this kind of message 
:

[Jun 27 08:58:17] WARNING[1255] loader.c: Module 'format_pcm_alaw.so' did not 
register itself during load
[Jun 27 08:58:17] WARNING[1255] loader.c: Module 'chan_modem_aopen.so' did not 
register itself during load
[Jun 27 08:58:17] WARNING[1255] loader.c: Module 'app_setcidname.so' did not 
register itself during load
[Jun 27 08:58:17] WARNING[1255] loader.c: Error loading module 
'app_txtcidname.so': /usr/lib/asterisk/modules/app_txtcidname.so: undefined 
symbol: option_priority_jumping
[Jun 27 08:58:18] WARNING[1255] loader.c: Module 'app_cut.so' did not register 
itself during load
[Jun 27 08:58:18] WARNING[1255] loader.c: Module 'app_setcidnum.so' did not 
register itself during load
[Jun 27 08:58:18] WARNING[1255] loader.c: Error loading module 'chan_h323.so': 
libh323_linux_x86_r.so.1.17.1: cannot open shared object file: No such file or 
directory
[Jun 27 08:58:18] WARNING[1255] res_smdi.c: No SMDI interfaces are available to 
listen on, not starting SDMI listener.
[Jun 27 08:58:18] WARNING[1255] loader.c: Module 'chan_modem_i4l.so' did not 
register itself during load
[Jun 27 08:58:18] NOTICE[1255] cdr_radius.c: Cannot load radiusclient-ng 
configuration file /etc/radiusclient-ng/radiusclient.conf.

Is anybody know what happened? Because I really don't understand this. Please, 
any body, help me :( 



  

Park yourself in front of a world of choices in alternative vehicles. Visit the 
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Re: [asterisk-users] Nokia N95 + Dial Plan

[Kevin Withnall]

I just had a similar problem and solved it with..

[from-internal-intldial]
exten => _+61X,1,Goto(from-internal,0${EXTEN:-9},1)
exten => _X.,1,Goto(from-internal,${EXTEN},1)

And put the E65 into the new contect. The first line stops all +61 (my
default country in australia)
The second catches everything else (like local 1300, 1800 etc numbers
that are in the phone without a +)

I can't find anywhere that tells me that the + is a valid pattern in a
dialplan but it works :-)

Obviously you could add any country codes you wish. I couldn't think of
something that would work for all codes as they are differing lengths.

Regards
Kevin


--
Kevin Withnall http://kevin.withnall.com/
ILB Computing http://www.ilb.com.au
PH: 02 4227 0001 Mobile: 0412 453 846 FAX: 02 4227 0081
Please consider the environment before printing this e-mail
 
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nitesh
Divecha
Sent: Monday, 25 June 2007 1:56 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Nokia N95 + Dial Plan

Hello All,

Recently I added some Nokia N95 customers and it worked pretty good.
Now the customers are complaining about the dialing rules...
They are used to dialing +12486543210 and +4479XX for long distance 
calls.

Is there anyway to create a "+" sign dial plan which will allow them to 
dial a number with "+" sign.

Cheers,
Nitesh


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Re: [asterisk-users] access to asterisk server since internet

[Don Kelly]

See http://www.dyndns.com/services/dns/dyndns/

 

You can establish a name like "yassir.dyndns.org" that will point to the
dynamic ip address for your Asterisk server. You should be able to use this
for the domain for your VoIP service provider.

  --Don

Don Kelly
PCF Corp
Real Support for your Virtual Office
651 842-1000
888 Don Kell(y)
651 842-1001 fax



  _  

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of skalli yassir
Sent: Friday, June 22, 2007 11:54 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] access to asterisk server since internet

 

hi
i have configured an asterisk server which i have tested locally with x-lite
and that's ok but when i wanted to access to it since internet that hasent
taken place
knowing that my server has access to internet by a wifi router that has a
public ip address (e-g a.b.d.c) and asterisk server has a private ip
192.168.1.111 (the firewall is disabled)
can some one tellme  how i should configure the x-lite clients with this
configuration
and what should i change to access to my server since internet 

-- 
Scales of success are not easy to be ridden 

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Re: [asterisk-users] Need to increase call count

[Mike Diehl]

It turns out that we were/are having trouble with out uplink

On Tuesday 19 June 2007 05:39:27 am Dave Bour wrote:
> Have you tested the actual throughput on the link?  What's it max out...
> What kind of latency are you seeing as it gets loaded.
> Can you do a local call to your own internal network (softphone or
> hardphone) as the system gets loaded and play the same file.   Do you
> have quality issues.  This will help isolate if it's a network or server
> issue.  Shouldn't be a server issue at that load...but never hurts to
> look.
> D
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Mike Diehl
> Sent: Tuesday, June 19, 2007 12:27 AM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Need to increase call count
>
> Hi all.
>
> I've got a project where I need to make outbound calls and play a
> prerecorded .wav file to the called number.
>
> So far, I've only been able to make about 15 concurrent calls before the
> cound quality gets poor, and I really need to increase this.
>
> I've got QoS configured to prioritize IAX2 traffic above all and my
> connection to the Internet is a PtoP 100Mb ethernet link.
> (255.255.255.252 subnet mask)
>
> The server is an AMD Athlon(tm) 64 Processor 3400+ with 512Mb of RAM.
> The Nic isn't sharing an IRQ with anything else and the CPU never
> exceeds 15% utilization.
>
> Any ideas where I can look for improvement?
>
> TIA,
>
> --
> Mike Diehl
>
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-- 
Mike Diehl

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Re: [asterisk-users] Asterisk to Cisco 2600 GW DTMF Not Working

[Mark Phillips]

Sounds to me like inband vs rfc2833 issues.

I found that one has to use the same codec throughout in order to make
DTMF function and then use inband. This in turn forces you down the road
of alaw or ulaw codecs.



On Tue, 2007-06-26 at 18:01 -0500, JR Richardson wrote:
> Hi All,
> 
> I have Asterisk 1.2.9.1 sending SIP calls to a Cisco 2620XM Router
> with a PRI card in it, handing off to a PBX and vise verse.  Calls in
> and out are working fine except for DTMF from Asterisk to the 2600.
> DTMF from the 2600 to Asterisk is fine.
> 
> Here are the Asterisk console warnings I get when I send DTMF from
> Asterisk to the 2600:
> 
>   == Forcing Marker bit, because SSRC has changed
> Jun 26 17:53:52 WARNING[14248]: channel.c:2328 set_format: Unable to
> find a codec translation path from ilbc to ulaw
> Jun 26 17:53:52 WARNING[14248]: channel.c:2328 set_format: Unable to
> find a codec translation path from ilbc to ulaw
> Jun 26 17:53:52 WARNING[14248]: chan_sip.c:2555 sip_write: Asked to
> transmit frame type 1024, while native formats is 4 (read/write = 4/4)
> Jun 26 17:53:52 WARNING[14248]: channel.c:2693
> ast_channel_make_compatible: No path to translate from
> SIP/53061-92e0(4) to SIP/10.10.10.10-78fa(1024)
> Jun 26 17:53:52 WARNING[14248]: channel.c:3520 ast_channel_bridge:
> Can't make SIP/53061-92e0 and SIP/10.10.10.10-78fa compatible
> Jun 26 17:53:52 WARNING[14248]: res_features.c:1381 ast_bridge_call:
> Bridge failed on channels SIP/53061-92e0 and SIP/10.10.10.10-78fa
>   == Spawn extension (iaxtest, 2144466715, 3) exited non-zero on
> 'SIP/53061-92e0'
> 
> The call drops.
> 
> If I enable ILBC codec with Asterisk, here is what I get:
> 
>   == Forcing Marker bit, because SSRC has changed
> Jun 26 17:56:28 WARNING[14332]: codec_ilbc.c:175 ilbctolin_framein:
> Huh?  An ilbc frame that isn't a multiple of 50 bytes long from RTP
> (160)?
> Jun 26 17:56:28 NOTICE[14332]: rtp.c:575 ast_rtp_read: Unknown RTP
> codec 122 received
> Jun 26 17:56:28 NOTICE[14332]: rtp.c:575 ast_rtp_read: Unknown RTP
> codec 122 received
> Jun 26 17:56:28 NOTICE[14332]: rtp.c:575 ast_rtp_read: Unknown RTP
> codec 122 received
> Jun 26 17:56:28 NOTICE[14332]: rtp.c:575 ast_rtp_read: Unknown RTP
> codec 122 received
> Jun 26 17:56:28 NOTICE[14332]: rtp.c:575 ast_rtp_read: Unknown RTP
> codec 122 received
> Jun 26 17:56:28 NOTICE[14332]: rtp.c:575 ast_rtp_read: Unknown RTP
> codec 122 received
> Jun 26 17:56:28 NOTICE[14332]: rtp.c:575 ast_rtp_read: Unknown RTP
> codec 122 received
> Jun 26 17:56:28 NOTICE[14332]: rtp.c:575 ast_rtp_read: Unknown RTP
> codec 122 received
> Jun 26 17:56:28 NOTICE[14332]: rtp.c:575 ast_rtp_read: Unknown RTP
> codec 122 received
> 
> The call continues with this error until I hang up.
> 
> I have been adjusting the dial-peer dtmf settings in the 2600 and have
> tried all the dtmf settings in Asterisk.
> 
> Any guidance will be appreciated.
> 
> Thanks.
> 
> JR


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Re: [asterisk-users] access to asterisk server since internet

[Mojo with Horan & Company, LLC]

 From what you provided, I'm not sure that 'firewall is disabled' will 
help you.  Your firewall probably needs to be configured to forward some 
ports to the asterisk box's internal IP address.  I usually do the 
following:

For SSH connections to the box to manage it:
forward a.b.d.c's external port 22 to 192.168.1.111's port 22

if using SIP, also:
forward a.b.d.c's external port 5060 to 192.168.1.111's port 5060
forward a.b.d.c's external ports 1-2 to 192.168.1.111's ports 
1-2.

and, if using IAX, also the following:
forward a.b.d.c's external port 4569 to 192.168.1.111's port 4569

if you're running some manner of web management GUI, you might:
forward a.b.d.c's external port 80 to 192.168.1.111's port 80
BUT be wary that any GUI you choose is NOT NECESSARILY safe to point at 
the public.  Someone could compromise it, who knows.  Honestly I would 
use the SSH port, port 22, to create a tunnel to port 80 on the asterisk 
box.

Moj



skalli yassir wrote:
> hi
> i have configured an asterisk server which i have tested locally with 
> x-lite and that's ok but when i wanted to access to it since internet 
> that hasent taken place
> knowing that my server has access to internet by a wifi router that has 
> a public ip address (e-g a.b.d.c) and asterisk server has a private ip 
> 192.168.1.111  (the firewall is disabled)
> can some one tellme  how i should configure the x-lite clients with this 
> configuration
> and what should i change to access to my server since internet
> 
> -- 
> Scales of success are not easy to be ridden
> 
> 
> 
> 
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Re: [asterisk-users] Ring the second line when 1st line is busy

[Deepak Naidu]

Do any one any clue.

This is what I need.
I have a Polycom 501 phone, which support multiple lines ie on the LCD you can 
see the extensions asssigned to a user as.
  555 --- Line 1 -- Extensions which registers with SIP(Asterisk) -- User A
 8555 --- Line 2 -- Extensions which registers with SIP(Asterisk) -- User A
 So when now someone calls one Extension 555 to User A, scenario as below.
 1) If he is busy on the first line Ext 555, then ring Line 2 Ext 8555, if no 
response send to voicemail
 2) If he doesnt picks the line 1 Ext 555, then send to voicemail rather then 
ringging on his second line ie Ext 8555.
 This is what I need, if I can dow it with Follow me, then how, if through ring 
group how.


Deepak Naidu <[EMAIL PROTECTED]> wrote: Hi,
I ma using Asterisk 1.2.18 & FreePBX 2.2.1.  I have assigned every 
users in office with Polycom with 2 extensions as below

 555
8555

I have configured Follow-me to ring when the users doesn't picks the phone on 
line 1(555) after 10 seconds & then ring the line 2(8555).  But this is not a 
standard telephony which I have been advised to change like below.

If someone calls Ext 555 & its busy(means on a phone with someone), then only 
ring the second line ie 8555 & even if that is busy send on voicemail.

If the first line 555 is free & no one picks up then let it go on the  
voicemail & not second line, bcos now no one has picked the phone nor busy.

I could find anyway to do in FreePBX, so was wondering how about doing this.  
Thanx for any input.

exten => 555,1,Macro(exten-vm,555,555)
exten => 555,n,Hangup
exten =>  555,hint,SIP/555
exten => ${VM_PREFIX}555,1,Macro(vm,555,DIRECTDIAL)
exten => ${VM_PREFIX}555,n,Hangup


--
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Re: [asterisk-users] kore dump

[Mojo with Horan & Company, LLC]

just an idea, but maybe qmail, samba, and bind have a smaller memory 
footprint than an in-use asterisk? can you take the hardware offline 
long enough for a memtest?

Moj

Luki wrote:
> It's no unusual seeing uptime for say
> qmail, samba or bind of 200+ days. 

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Re: [asterisk-users] Modification of Caller ID based on context

[arkda]

Great examples Matthew, really appreciate it. This is exactly what I've been
searching for!

On 6/26/07, Matthew Brothers <[EMAIL PROTECTED]> wrote:


> Hi,
>
> I have been looking for an example of accomplishing this, but
> I've been unable to locate something similar to what I'm trying
> to do.
>
> Here's the scenario:
>
> Users caller ID is set to their internal extension (200-250).
> This is set in sip.conf for each user. Each user has a local DID
> as well (hosted through Vitelity, for example (555)111-). The
>  problem is that this extension was being passed to the outside
> world. I currently have a SetCallerID command changing the
> CallerID to our main office number, but some users want their DID
>  sent, not the general number.
>
> The problem is that if their caller ID is set to their DID, when
> users hit redial on their phones internally they dial out and
> back in. I corrected this by putting each DID in extensions.conf
> under their three digit extension, but that seems a bit like a
> kludge obviously.
>
> I'm looking for a method of sending the internal three digit
> extension only when a user is dialing another user internally,
> otherwise it will send their DID. Is their a method to do this in
>  the dial plan? Anyone have an example of how to accomplish this?
>
>
> Thanks in advance.


Mike,

I have a similar setup (I even use Vitel) and the easiest and
cleanest method that I have found to accomplish this is with the
AstDB. You can simply create a cross-reference of DIDs and Internal
extensions similar to extdid/200 => 555111 ... extdid/250 =>
5551112272 in the AstDB. Then you can change your outgoing dialplan
to change the caller id based upon this cross reference. Example:


exten => NXXNXX,n,Set(outgoingCID=)

exten => NXXNXX,n,
GotoIf($[ ${DB_EXISTS(extdid/${CALLERID(num)})} = 0 ]?makecall)

exten => NXXNXX,n,
Set(outgoingCID=${DB(extdid/${CALLERID(num)})})

exten => NXXNXX,n(makecall),Set(CALLERID(num)=${outgoingCID})

...

You could even simplify your incoming context by cross-referencing
in the other direction. That is didext/555111 => 200 ...
didext/5551112272 => 250.

exten => NXXNXX,n,
Goto(internal-extensions,${DB(didext/${EXTEN})},1)

OR you could do something similar with LOCAL channels or with a Dial
command.

-Matthew

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[asterisk-users] Asterisk to Cisco 2600 GW DTMF Not Working

[JR Richardson]

Hi All,

I have Asterisk 1.2.9.1 sending SIP calls to a Cisco 2620XM Router
with a PRI card in it, handing off to a PBX and vise verse.  Calls in
and out are working fine except for DTMF from Asterisk to the 2600.
DTMF from the 2600 to Asterisk is fine.

Here are the Asterisk console warnings I get when I send DTMF from
Asterisk to the 2600:

  == Forcing Marker bit, because SSRC has changed
Jun 26 17:53:52 WARNING[14248]: channel.c:2328 set_format: Unable to
find a codec translation path from ilbc to ulaw
Jun 26 17:53:52 WARNING[14248]: channel.c:2328 set_format: Unable to
find a codec translation path from ilbc to ulaw
Jun 26 17:53:52 WARNING[14248]: chan_sip.c:2555 sip_write: Asked to
transmit frame type 1024, while native formats is 4 (read/write = 4/4)
Jun 26 17:53:52 WARNING[14248]: channel.c:2693
ast_channel_make_compatible: No path to translate from
SIP/53061-92e0(4) to SIP/10.10.10.10-78fa(1024)
Jun 26 17:53:52 WARNING[14248]: channel.c:3520 ast_channel_bridge:
Can't make SIP/53061-92e0 and SIP/10.10.10.10-78fa compatible
Jun 26 17:53:52 WARNING[14248]: res_features.c:1381 ast_bridge_call:
Bridge failed on channels SIP/53061-92e0 and SIP/10.10.10.10-78fa
  == Spawn extension (iaxtest, 2144466715, 3) exited non-zero on
'SIP/53061-92e0'

The call drops.

If I enable ILBC codec with Asterisk, here is what I get:

  == Forcing Marker bit, because SSRC has changed
Jun 26 17:56:28 WARNING[14332]: codec_ilbc.c:175 ilbctolin_framein:
Huh?  An ilbc frame that isn't a multiple of 50 bytes long from RTP
(160)?
Jun 26 17:56:28 NOTICE[14332]: rtp.c:575 ast_rtp_read: Unknown RTP
codec 122 received
Jun 26 17:56:28 NOTICE[14332]: rtp.c:575 ast_rtp_read: Unknown RTP
codec 122 received
Jun 26 17:56:28 NOTICE[14332]: rtp.c:575 ast_rtp_read: Unknown RTP
codec 122 received
Jun 26 17:56:28 NOTICE[14332]: rtp.c:575 ast_rtp_read: Unknown RTP
codec 122 received
Jun 26 17:56:28 NOTICE[14332]: rtp.c:575 ast_rtp_read: Unknown RTP
codec 122 received
Jun 26 17:56:28 NOTICE[14332]: rtp.c:575 ast_rtp_read: Unknown RTP
codec 122 received
Jun 26 17:56:28 NOTICE[14332]: rtp.c:575 ast_rtp_read: Unknown RTP
codec 122 received
Jun 26 17:56:28 NOTICE[14332]: rtp.c:575 ast_rtp_read: Unknown RTP
codec 122 received
Jun 26 17:56:28 NOTICE[14332]: rtp.c:575 ast_rtp_read: Unknown RTP
codec 122 received

The call continues with this error until I hang up.

I have been adjusting the dial-peer dtmf settings in the 2600 and have
tried all the dtmf settings in Asterisk.

Any guidance will be appreciated.

Thanks.

JR
-- 
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Engineering for the Masses

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Re: [asterisk-users] AudioCodes Gateway and Asterisk

[Shanon Swafford]

When you see "[ERROR]" in the Message Log, either the MP firmware is buggy
or the far end is sending something out of spec in the SIP Message.

You'll need to upgrade to the latest MP firmware then report this to
whomever you bought it from.  Or fix the far end to send the message in spec
or form that doesn't cause the "[ERROR]".

Also, do your supporter a favor and don't paste those logs directly into
emails.  The wrap makes them horrible to read and they can't send them on to
Audiocodes like that.  Put them in a text file which preserves the line
length.

Regards,
Shanon
http://www.abptech.com/support/qa/


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dovid B
Sent: Sunday, June 24, 2007 2:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] AudioCodes Gateway and Asterisk



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

Sent: Thursday, June 21, 2007 6:27 PM
Subject: Re: [asterisk-users] AudioCodes Gateway and Asterisk


>
>>On 6/21/07, Dovid B <[EMAIL PROTECTED]> wrote:
>>> Hi List,
>>> I am trying to call from my asterisk box (1.2.18) to and audiocodes
> MP114. I
>>> keep getting an error from asterisk of -- Got SIP response 415
> "Unsupported
>>> Media Type" back from XXX.XXX.XX.XX. Both box's are set up to use G729.
>>> Anyone have a hint as to what it may be ?
>
>>Are you sure, your asterisk supports G729? It isn't supported by
>>default, you need additional modules or hardware cards for G729
>>support. If it is - what are you using for G729 - that might help to
>>identify the problem.
>
>>Regards,
>>Atis
>
> If the AudioCodes is sending back that 415, the "Message Log" in the
> AudioCodes is invaluable.  Set your debug level to 5/6 and watch it while
> you make test calls.  Once you learn how to interpret this output, you'll 
> be
> well on your way with AudioCodes.
>
> If G729 is active on the MP, but still giving back that error, G729 might
> not be in a profile if you are using them.
>
> Also, firmware that comes on the MPs is normally sorta buggy, ask your
> reseller for the latest version.
>
> http://www.abptech.com/support/faqs/
>
> Regards,
> Shanon
> ABP Technology
>

Shanon,
The audiocodes were preftctly with other providers using G729. It's just 
having an issue with asterisk. Here is the output from the AudioCodes:



Log is Activated



12d:23h:36m:17s ( lgr_flow)(828 )  Incoming SIP Message from 
XXX.XXX.XX.XXX:5060  [File: Line:-1]

12d:23h:36m:17s INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 
Record-Route:  Via: 
SIP/2.0/UDP XXX.XXX.XX.XXX;branch=z9hG4bK1db7.ecac6e86.0 Via: SIP/2.0/UDP 
XXX.XXX.XX.XXX:5060;branch=z9hG4bK1ef72aa9;rport=5060 From: "55560888" 
;tag=as4a537e63 To: 
 Contact: 
 Call-ID: 
[EMAIL PROTECTED] CSeq: 102 INVITE User-Agent:

Enswitch Max-Forwards: 16 Date: Wed, 20 Jun 2007 19:44:42 GMT Allow: INVITE,

ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: 
application/sdp Content-Length: 490 v=0 o=root 29170 29170 IN IP4 
XXX.XXX.XX.XXX s=session c=IN IP4 XXX.XXX.XX.XXX t=0 0 m=audio 14878 RTP/AVP

18 0 8 10 3 111 5 7 110 97 101 a=rtpmap:18 G729/8000 a=fmtp:18 
annexb=no;mode-change-period=0 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/80

12d:23h:36m:17s ( sip_stack)(830 ) ?? [WARNING] AcSIPParser: Unrecognized 
Header was detected at line: 12 [File: Line:-1]

12d:23h:36m:17s ( lgr_flow)(831 ) | | new GetNewSIPCall created - #8 [File: 
Line:-1]

12d:23h:36m:17s ( sip_stack)(832 ) new AcSIPCallAPI created - #5 [File: 
Line:-1]

12d:23h:36m:17s ( lgr_stk_mngr)(833 ) Resource StackSession <#5> 
Allocated [File: Line:-1]

12d:23h:36m:17s ( lgr_flow)(834 ) | |(SIPTU#8)INVITE State:Idle() [File: 
Line:-1]

12d:23h:36m:17s ( sip_stack)(835 ) SIPCall(#8) changes state from Idle to 
Invited [File: Line:-1]

12d:23h:36m:17s ( sip_stack)(836 ) AcSIPParser: Problem in 
AcSIPCallAPI::ParseSDP [File: Line:-1]

12d:23h:36m:17s ( sip_stack)(837 ) !! [ERROR] AcSIPParser: Parse Error. 
Unexpected symbol ' [File: Line:-1]

12d:23h:36m:17s ( sip_stack)(838 ) !! [ERROR] Message type: INVITE [File: 
Line:-1]

12d:23h:36m:17s ( sip_stack)(839 ) !! [ERROR] Source header: [File: Line:-1]

12d:23h:36m:17s ( sip_stack)(840 ) !! [ERROR] Line: 20. Column: 27 [File: 
Line:-1]

12d:23h:36m:17s ( lgr_flow)(841 ) | | | 
#5:SIP_SETUP_EV([EMAIL PROTECTED]) [File: 
Line:-1]

12d:23h:36m:17s ( lgr_stk_ses)(842 ) SIPStackSession::HandleStackSetupEV - 
NEWCALL: SrcPN=0 [File: Line:-1]

12d:23h:36m:17s ( lgr_stk_ses)(843 )  SendToCall - event: 
NEW_CALL_EV m_Call = 32173400 [File: Line:-1]

12d:23h:36m:17s ( lgr_flow)(844 ) | | 
#5:NEW_CALL_EV:([EMAIL PROTECTED]) [File: 
Line:-1]

12d:23h:36m:17s ( lgr_psbrdif)(845 ) AcBoard::GetTrunkGroupId - No entry 
found for: DstNum:55560918 SrcNum:5

Re: [asterisk-users] Modification of Caller ID based on context

[Matthew Brothers]

> Hi,
> 
> I have been looking for an example of accomplishing this, but 
> I've been unable to locate something similar to what I'm trying 
> to do.
> 
> Here's the scenario:
> 
> Users caller ID is set to their internal extension (200-250). 
> This is set in sip.conf for each user. Each user has a local DID 
> as well (hosted through Vitelity, for example (555)111-). The
>  problem is that this extension was being passed to the outside 
> world. I currently have a SetCallerID command changing the 
> CallerID to our main office number, but some users want their DID
>  sent, not the general number.
> 
> The problem is that if their caller ID is set to their DID, when 
> users hit redial on their phones internally they dial out and 
> back in. I corrected this by putting each DID in extensions.conf 
> under their three digit extension, but that seems a bit like a 
> kludge obviously.
> 
> I'm looking for a method of sending the internal three digit 
> extension only when a user is dialing another user internally, 
> otherwise it will send their DID. Is their a method to do this in
>  the dial plan? Anyone have an example of how to accomplish this?
> 
> 
> Thanks in advance.


Mike,

I have a similar setup (I even use Vitel) and the easiest and
cleanest method that I have found to accomplish this is with the
AstDB. You can simply create a cross-reference of DIDs and Internal
extensions similar to extdid/200 => 555111 ... extdid/250 =>
5551112272 in the AstDB. Then you can change your outgoing dialplan
to change the caller id based upon this cross reference. Example:


exten => NXXNXX,n,Set(outgoingCID=)

exten => NXXNXX,n,
GotoIf($[ ${DB_EXISTS(extdid/${CALLERID(num)})} = 0 ]?makecall)

exten => NXXNXX,n,
Set(outgoingCID=${DB(extdid/${CALLERID(num)})})

exten => NXXNXX,n(makecall),Set(CALLERID(num)=${outgoingCID})

...

You could even simplify your incoming context by cross-referencing
in the other direction. That is didext/555111 => 200 ...
didext/5551112272 => 250.

exten => NXXNXX,n,
Goto(internal-extensions,${DB(didext/${EXTEN})},1)

OR you could do something similar with LOCAL channels or with a Dial
command.

-Matthew

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Re: [asterisk-users] kore dump

[Luki]

> I am having a problem with Asterisk frequently crashing on me as
> well.  I just run it under supervise:

But that's just a band-aid. If it crashes, it takes all calls with it.
Hardly a good thing, unless you only have 1 call at a time -- then
it's probably no the end of the world.

I still don't know what's up with the crashes but here are two
observations I made:

1) I moved the same installation from one hardware to another. On the
former hardware it would crash every 2 weeks, on average. On the new
hardware, it has not yet crashed and it has 9 weeks of uptime. Same
call volume, same devices, same network. I'm running asterisk
chroot'ed so all libraries, binaries, config files, etc. are
identical. Only the hardware and kernel are different.

2) The same old hardware has been in service for 3 years and no other
programs crash on it. Ever. It's no unusual seeing uptime for say
qmail, samba or bind of 200+ days. I have therefore reasons to believe
that the hardware is OK.

So go figure. And BTW, the crashes (based on the core dumps) are
always at a different place. There is no consistency. Right now I'm
just glad it no longer core dump on me :).

--Luki

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Re: [asterisk-users] Provisioning Linksys WIP330 phones

[Shanon Swafford]

 
http://www.abptech.com/support/qa/index.php?target=linksy_remote_p
 
 


  _  

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michelle
Dupuis
Sent: Sunday, June 24, 2007 10:53 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Provisioning Linksys WIP330 phones


We're contemplating Linksys WIP 330 phones, but we're concerned about
configuration effort.  Does anyone have the file format for an XML file to
configure this phone?  We got auto-provisioning to D/L a file, but the XML
file format seems to be a secret...
 
Thanks,
Michelle
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Re: [asterisk-users] No such host error from SIP for non-peer configuration.

[Lucian Romi]

I figured it out.

srvlookup=no


On 6/26/07, Lucian Romi <[EMAIL PROTECTED]> wrote:


Is there a way to let chan_sip skip host lookup?
Problem is I have to have a peer host config for every sip message
outgoing. For example, I cann't have this

in extension.conf
exten => 500,n,Dial(SIP/[EMAIL PROTECTED])

It'll return,
chan_sip.c:2738 create_addr: No such host: 192.168.1.79

when call forwarding


I have to have a peer in SIP

[outgoing]
host=192.168.1.79
...

in sip.conf

and have to use this peer in extension.conf
exten => 500,n,Dial(SIP/[EMAIL PROTECTED])

Is there any configuration in sip.conf restrict me to do this?
Thanks!

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Re: [asterisk-users] Bridging two PSTN calls

[Jared Smith]

On 6/26/07, Doug Zingel <[EMAIL PROTECTED]> wrote:
> I'm wondering if its possible to receive a call from
> an external number (PSTN) say A. Then make a call to
> another external number (PSTN) say B - and then bridge
> the two calls so that A is talking to B? What hardware
> will I need to be able to do this.

Yes, that's one of the many things Asterisk can do.  Obviously, you'd
need a hardware card to connect Asterisk to the phone lines.  You'd
need FXO ports to connect Asterisk to analog phone *lines*, and FXS
ports to connect Asterisk to analog *phones*.  You could also use IP
phones, and/or a VoIP provider.

> Secondly, if I had x number of simultaneous calls (A
> talking to B) - how many PSTN lines would I need? I
> think 2x.
>
> I know this sounds ridiculously inefficient - but I
> think the experiment might open some avenues.

Yes, you would need 2x the number of calls for a straight analog
setup, unless you try to get fancy and setup flash transfers with your
telco.  Again, you could use a VoIP provider to get around the
hardware costs of analog ports if port cost is a concern to you.

-Jared

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[asterisk-users] No such host error from SIP for non-peer configuration.

[Lucian Romi]

Is there a way to let chan_sip skip host lookup?
Problem is I have to have a peer host config for every sip message outgoing.
For example, I cann't have this

in extension.conf
exten => 500,n,Dial(SIP/[EMAIL PROTECTED])

It'll return,
chan_sip.c:2738 create_addr: No such host: 192.168.1.79

when call forwarding


I have to have a peer in SIP

[outgoing]
host=192.168.1.79
...

in sip.conf

and have to use this peer in extension.conf
exten => 500,n,Dial(SIP/[EMAIL PROTECTED])

Is there any configuration in sip.conf restrict me to do this?
Thanks!
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Re: [asterisk-users] zaptel 1.2.18 and HPEC

[Kevin P. Fleming]

Paul Hales wrote:

> Any idea why I can't build HPEC for zaptel 1.2.18?
> 
> It builds fine with 1.4.3...

I don't know why this change just happened, but it has been fixed in
revision 2668 of SVN branch-1.2 of Zaptel.

You can fix it on your system by adding the following line:

.EXPORT_ALL_VARIABLES:

just above the 'FORCE:' line in the Makefile.

-- 
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)

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Re: [asterisk-users] kore dump

[J. Oquendo]

Vadim Berezniker wrote:


use the safe_asterisk script

it will restart asterisk if it crashes and it enables core dumps (your 
core size limit is probably set to 0 when you start asterisk).


*From:* [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] *On Behalf Of *Ed Nuñez

*Sent:* Tuesday, June 26, 2007 2:22 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion; 
[EMAIL PROTECTED]

*Subject:* [asterisk-users] kore dump

I am running Asterisk 1.4.5 and addons 1.4.1 in a CentOS 5 Server.

My PBX has experienced several core dumps the last couple of days and 
I am not sure if this is what’s causing it, but it always seems to 
happen when a particular extension on a grandstream phone uses ChanSpy 
SIP group.


I have not been able to locate where the core dump file is being 
saved. I can’t find it in my TMP directory.


I would also like to know if Asterisk can be setup to automatically re 
start if there is a core dump. I was thinking of setting up a cron job 
to launch Asterisk every minute. If it’s running, no harm done, and if 
it crashes, the cron job will make sure that it’s started every 60 
seconds.


Any suggestions?

Thank you

Ed Nuñez



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If that fails you could always try something like:
*/2 * * * * /bin/ps -C /usr/bin/asterisk || { /usr/bin/asterisk & }

or so...

--

J. Oquendo
http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x1383A743
echo infiltrated.net|sed 's/^/sil@/g' 


"Wise men talk because they have something to say;
fools, because they have to say something." -- Plato




smime.p7s
Description: S/MIME Cryptographic Signature
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Re: [asterisk-users] Cisco 7941 localized menus with SIP firmware

[Olivier]

2007/6/26, Steve Kennedy <[EMAIL PROTECTED]>:


On Tue, Jun 26, 2007 at 09:45:48PM +0200, Olivier wrote:

>Has anyone met any success, installing localized (ie non-english)
menus
>within SIP firmware enabled Cisco 7941 ?
>Those phones seem to be trying to download localized menus from Cisco
>Call Manager but as they are managed by an Asterisk server, I'm
looking
>for a workaround.
>Any advice ?

Has anyone written a configuration tool for Cisco's that generates the
correct XML files?

Wouldn't that be a useful thing?


Steve

--
NetTek Ltd  UK mob +44-(0)7775 755503
UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455
Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED]
Euro Tech News Blog http://eurotechnews.blogspot.com

Hi,


Do you mean all 7941 menus provisionning is about, is downloading a given
XML file ?
Is it easy to get a sample of this file ?

Regards
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Re: [asterisk-users] More FAX over T1

[David Gomillion]

On 6/26/07, Lacy Moore - Aspendora <[EMAIL PROTECTED]> wrote:


On 6/26/07, Joe acquisto <[EMAIL PROTECTED]> wrote:
>
> One idea is to utilize DID, and have Asterisk "forward" the calls to the
current FAX lines, preserving the DID as "Caller ID".  I am fairly sure
Asterisk itself can do this. (The call would appear to be "from" this
assigned ID).  If so, I could, apparently, massage Hylafax into dealing with
each FAX based on the "Caller ID".
>

That's definitely an idea.  If you don't need the Caller ID on the fax
(and in most cases, you probably don't), this might be your best
solution.  Assuming, of course, the faxmodems on Hylafax are picking
up the caller ID and you have Caller ID from the phone company.

That would take up 2 of your PRI channels, though, per fax reception.



I think I read that you have 4 fax lines. If this is correct, then the
calculation is thus:

4 lines * 30 per month = 120 per month.

channel bank = $200 new (for faxes, I have never had a problem with zhone
CBs found at http://www.digital-loggers.com/CB.html)
2-port T1 card = $900 new, for a total of $1100 in equipment (a one-time
cost)

So, if you keep the solution going for more than 9 months, you'll come out
ahead just buying the equipment. If you pay more than $30 per month per
phone line, your break-even will be much quicker. Also, if you ever needed
to add more lines, you already can have 24 faxes through Asterisk, and your
fax server would be the bottleneck.

This is why we run all of our fax lines off of our PRI, even though our
local dialtone provider tried to convince us to buy POTS lines for each one.
At around $30 per line (not including taxes), it just doesn't add up.
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Re: [asterisk-users] Fax Throughput

[Don Kelly]

Matt, thanks for pointing me to zaptel.conf; I'm new to Asterisk and can use
all the help I get!

Here are the non-comment lines from zaptel.conf (not set up by me):

span=1,1,0,esf,b8zs
span=2,1,0,esf,b8zs
bchan=1-23
dchan=24
bchan=25-47
dchan=48
loadzone = us
defaultzone=us

The first span is connected to the PSTN. The second is connected to a
Windows-based server using Dialogic hardware and custom software.

The second span has a clock priority equal to the first one. I'm guessing
that this has the effect of ignoring clock from the first span (same as '0')
and using clock from the second. Not good.

I've changed the clock priority for span 2 to '0'; if we lose the PSTN we'll
rely on the Digium card for clock.

Fax throughput seems fine with this change.

In zapata.conf I find:

; Network Side
signalling => pri_cpe
group => 1
context => pstn-inbound
channel => 1-23


; IVR Side
signalling=> pri_net
group => 2
context => ivr-inbound
channel => 25-47

The default would be switchtype=national, which is correct.

I see that for 'signalling', 'group' and 'context' "=>" has been used,
rather than the "=" that I see in documentation. Does this matter?

  --Don

Don Kelly
PCF Corp
Real Support for your Virtual Office
651 842-1000
888 Don Kell(y)
651 842-1001 fax

 

-Original Message-
From: Matthew Fredrickson [mailto:[EMAIL PROTECTED] 
Sent: Tuesday, June 26, 2007 9:22 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Fax Throughput

Can you post your zaptel.conf so we can verify your timing settings?

---
Matthew Fredrickson
Software Engineer
Digium, Inc.

On Jun 25, 2007, at 11:10 PM, Don Kelly wrote:

> I've tried timing faxes two ways:
>
> From a fax machine on a station port of an AltiGen PC/PBX served by  
> an MCI
> PRI calling back into the same PRI and reaching a RightFax server on a
> station port behind the AltiGen.
>
> From the same fax machine on the same station port of the AltiGen  
> PC/PBX
> served by the same MCI PRI calling a number on an XO PRI connected  
> to an
> Asterisk system (Digium TE410P), dialing out on another channel on  
> the same
> PRI back into the MCI PRI and reaching the RightFax server on the  
> station
> port behind the AltiGen.
>
> extensions.conf includes:
> exten => 6122353002,1,dial(zap/g1/6122590773)
>
> Sending a one-page fax with moderate density (no graphics) takes  
> almost five
> minutes longer going through the Asterisk server.
>
> Any suggestions?
>
>   --Don
>
> Don Kelly
> PCF Corp
> Real Support for your Virtual Office
> 651 842-1000
> 888 Don Kell(y)
> 651 842-1001 fax
>
>
>
>
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Re: [asterisk-users] Cisco 7941 localized menus with SIP firmware

[Steve Kennedy]

On Tue, Jun 26, 2007 at 09:45:48PM +0200, Olivier wrote:

>Has anyone met any success, installing localized (ie non-english) menus
>within SIP firmware enabled Cisco 7941 ?
>Those phones seem to be trying to download localized menus from Cisco
>Call Manager but as they are managed by an Asterisk server, I'm looking
>for a workaround.
>Any advice ?

Has anyone written a configuration tool for Cisco's that generates the
correct XML files?

Wouldn't that be a useful thing?


Steve

-- 
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Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED]
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Re: [asterisk-users] Ring/Off-hook in strange state 6

[Alex Mcdowell]

I don't have caller ID at all, not on the verizon side and
usecallerid=no in zapata.conf. I do, however have the DSL on this
line. I have a splitter and then I have a filter on the asterisk side.
I am guessing this is the root of the problem. Thanks for any
insight.-Alex

On 6/26/07, Eric ManxPower Wieling <[EMAIL PROTECTED]> wrote:
> Daniel already pointed you in the right direction.
>
> I have seen this error many times, but it never causes a problem.
>
> Alex Mcdowell wrote:
> > Can anybody at least point me in a direction??
> >
> > On 6/25/07, Alex Mcdowell <[EMAIL PROTECTED]> wrote:
> >> I don't think my cards are bad, but maybe there is a problem with the
> >> one. It has been two weeks since I put my ticket in with Digium...and
> >> still no word. I am starting to get frustrated.
> >>
> >> On 6/22/07, Daniel Hazelbaker <[EMAIL PROTECTED]> wrote:
> >>>
> >>> Alex,
> >>>
> >>>
> >>>   I had this problem with a new TDM2400 card that we purchased.  
> >>> Specifically I would get that message and then it would pick up the 
> >>> ringing line AND the line next to it.  Basically, lines 1 & 2 had been 
> >>> cross-linked somehow.  After a few weeks of trouble-shooting with Digium 
> >>> tech support they cross-shipped me a new card and the problem (and that 
> >>> message) went away.
> >>>
> >>>
> >>> Daniel Hazelbaker
> >>> High Desert Church
> >>>
> >>>
> >>>
> >>> On Jun 22, 2007, at 1:22 PM, Alex Mcdowell wrote:
> >>>
> >>>
> >>>
> >>> HI I have two servers both of which get this message on one of the lines.
> >>>
> >>> Ring/Off-hook in strange state 6. The one server seems to be ok with it, 
> >>> but
> >>>
> >>> the other one when an extension picks up there is no one there and the
> >>>
> >>> incoming call keeps ringing. I tried to adjust the levels in wcfxo.c like
> >>>
> >>> someone had suggested, but it didn't do anything. I also upgraded zaptel 
> >>> to
> >>>
> >>> the latest. 1.2.18 and asterisk is running at 1.2.14. callprogess is set 
> >>> to
> >>>
> >>> no, as well as busydetect=no. This is a major problem since this box only
> >>>
> >>> has 1 other line, but at least it works. I can't seem to find much info on
> >>>
> >>> this issue. I can't believe others haven't run into it.  I started a 
> >>> ticket
> >>>
> >>> with digium, but I guess they are pretty backed up. Here is what I am
> >>>
> >>> getting in the CLI:  Thanks for any help -Alex
> >>>
> >>> -- Starting simple switch on 'Zap/4-1'
> >>>
> >>> -- Executing Wait("Zap/4-1", "1") in new stack
> >>>
> >>> -- Executing Answer("Zap/4-1", "") in new stack
> >>>
> >>> -- Executing Dial("Zap/4-1", "SIP/4125|20|tr") in new stack
> >>>
> >>> -- Called 4125
> >>>
> >>> Jun 22 13:44:45 WARNING[24296]: chan_zap.c:3926 zt_handle_event:
> >>>
> >>> Ring/Off-hook in strange state 6 on channel 4
> >>>
> >>> -- SIP/4125-09559118 is ringing
> >>>
> >>> Jun 22 13:44:46 WARNING[24296]: chan_zap.c:3926 zt_handle_event:
> >>>
> >>> Ring/Off-hook in strange state 6 on channel 4
> >>>
> >>> -- SIP/4125-09559118 answered Zap/4-1
> >>>
> >>>   == Spawn extension (incoming, s, 3) exited non-zero on 'Zap/4-1'
> >>>
> >>> -- Hungup 'Zap/4-1'
> >>>
> >>> -- Starting simple switch on 'Zap/4-1'
> >>>
> >>> -- Executing Wait("Zap/4-1", "1") in new stack
> >>>
> >>> -- Executing Answer("Zap/4-1", "") in new stack
> >>>
> >>> -- Executing Dial("Zap/4-1", "SIP/4125|20|tr") in new stack
> >>>
> >>> -- Called 4125
> >>>
> >>> Jun 22 13:44:57 WARNING[24303]: chan_zap.c:3926 zt_handle_event:
> >>>
> >>> Ring/Off-hook in strange state 6 on channel 4
> >>>
> >>> -- SIP/4125-09559118 is ringing
> >>>
> >>> Jun 22 13:44:58 WARNING[24303]: chan_zap.c:3926 zt_handle_event:
> >>>
> >>> Ring/Off-hook in strange state 6 on channel 4
> >>>
> >>> -- SIP/4125-09559118 answered Zap/4-1
> >>>
> >>>   == Spawn extension (incoming, s, 3) exited non-zero on 'Zap/4-1'
> >>>
> >>> -- Hungup 'Zap/4-1'
> >>>
> >>> -- Starting simple switch on 'Zap/4-1'
> >>>
> >>> -- Executing Wait("Zap/4-1", "1") in new stack
> >>>
> >>> -- Executing Answer("Zap/4-1", "") in new stack
> >>>
> >>> -- Executing Dial("Zap/4-1", "SIP/4125|20|tr") in new stack
> >>>
> >>> -- Called 4125
> >>>
> >>> -- SIP/4125-09559118 is ringing
> >>>
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[asterisk-users] Cisco 7941 localized menus with SIP firmware

[Olivier]

Hi,

Has anyone met any success, installing localized (ie non-english) menus
within SIP firmware enabled Cisco 7941 ?

Those phones seem to be trying to download localized menus from Cisco Call
Manager but as they are managed by an Asterisk server, I'm looking for a
workaround.
Any advice ?

Regards
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Re: [asterisk-users] kore dump

[Eric Lubow]

Ed,

   I am having a problem with Asterisk frequently crashing on me as
well.  I just run it under supervise:
   http://cr.yp.to/daemontools/supervise.html

   This way it will be restarted if svc determines it isn't running.

Eric

On Tue, 2007-06-26 at 13:22 -0500, Ed Nuñez wrote:
> I am running Asterisk 1.4.5 and addons 1.4.1 in a CentOS 5 Server.
> 
>  
> 
> My PBX has experienced several core dumps the last couple of days and
> I am not sure if this is what’s causing it, but it always seems to
> happen when a particular extension on a grandstream phone uses ChanSpy
> SIP group.
> 
>  
> 
> I have not been able to locate where the core dump file is being
> saved.   I can’t find it in my TMP directory.
> 
>  
> 
> I would also like to know if Asterisk can be setup to automatically re
> start if there is a core dump.  I was thinking of setting up a cron
> job to launch Asterisk every minute.  If it’s running, no harm done,
> and if it crashes, the cron job will make sure that it’s started every
> 60 seconds.
> 
>  
> 
> Any suggestions?
> 
>  
> 
>  
> 
> Thank you
> 
>  
> 
> Ed Nuñez
> 
> 
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Systems Administrator
e: [EMAIL PROTECTED]
w: www.linkexperts.com
p: 212.542.5201


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Re: [asterisk-users] More FAX over T1

[Joe acquisto]

>>> On 6/26/2007 at 3:04 PM, "Lacy Moore - Aspendora" <[EMAIL PROTECTED]> wrote:
> On 6/26/07, Joe acquisto <[EMAIL PROTECTED]> wrote:
>>
>> One idea is to utilize DID, and have Asterisk "forward" the calls to the 
> current FAX lines, preserving the DID as "Caller ID".  I am fairly sure 
> Asterisk itself can do this. (The call would appear to be "from" this 
> assigned ID).  If so, I could, apparently, massage Hylafax into dealing with 
> each FAX based on the "Caller ID".
>>
> 
> That's definitely an idea.  

Wow, you mean I actually had one??? 

>If you don't need the Caller ID on the fax
> (and in most cases, you probably don't), 

If you mean (not) printed on the FAX, yes (no ?) it is *not* needed.

>this might be your best
> solution.  Assuming, of course, the faxmodems on Hylafax are picking
> up the caller ID and you have Caller ID from the phone company.

Worthy of investigation.
 
> That would take up 2 of your PRI channels, though, per fax reception.
> 

That should not be a problem, at this point. 

joe a.


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Re: [asterisk-users] More FAX over T1

[Lacy Moore - Aspendora]

On 6/26/07, Joe acquisto <[EMAIL PROTECTED]> wrote:
>
> One idea is to utilize DID, and have Asterisk "forward" the calls to the 
> current FAX lines, preserving the DID as "Caller ID".  I am fairly sure 
> Asterisk itself can do this. (The call would appear to be "from" this 
> assigned ID).  If so, I could, apparently, massage Hylafax into dealing with 
> each FAX based on the "Caller ID".
>

That's definitely an idea.  If you don't need the Caller ID on the fax
(and in most cases, you probably don't), this might be your best
solution.  Assuming, of course, the faxmodems on Hylafax are picking
up the caller ID and you have Caller ID from the phone company.

That would take up 2 of your PRI channels, though, per fax reception.

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Re: [asterisk-users] kore dump

[Jared Smith]

On 6/26/07, Ed Nuñez <[EMAIL PROTECTED]> wrote:
> I have not been able to locate where the core dump file is being saved.   I
> can't find it in my TMP directory.

Check the directory in which you're starting Asterisk.  It doesn't
sound like you're using the Red Hat initscript to start Asterisk, so
you're probably starting it manually.  I'd check there first.

> I would also like to know if Asterisk can be setup to automatically re start
> if there is a core dump.  I was thinking of setting up a cron job to launch
> Asterisk every minute.  If it's running, no harm done, and if it crashes,
> the cron job will make sure that it's started every 60 seconds.

Check out the safe_asterisk script that's installed with Asterisk.
Again, I recommend you check out the initscript, which calls
safe_asterisk and restarts Asterisk if it crashes.  It also ensures
that your core dumps end up in /tmp where you can find them more
easily.  On Red Hat and CentOS, you can install the initscript by
typing "make config" in the Asterisk source directory.

-Jared

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Re: [asterisk-users] Transfer Call to Cell Phone

[OCOSA ListAcc]

Well seems like I am already doing first method minus the extension. We 
do have full features on our lines so both lines are free once the 
transfer is complete. We also have toll calls on our lines so it would 
not be a problem, so I do not have to worry about AT&T dropping the 
call. I tried to create a separate extension for this but it did not 
work. The only way I can get it to work is by have the call on the 1st 
line then transfer it out on the 2nd line. After that is complete both 
lines are free. A worst case scenario would be where our sales volume 
picked and we needed to transfer a call and could not because of the 
slots are filled.

Can you give an example of creating an extension which points to a cell 
phone. Secondly how can you have if no one answers an extension it dials 
the cell number next. That maybe answered in the example. I have the 
system setup so it just dials out which ever line is not busy. Thanks!


Otis



John Faubion wrote:
>> Polycom Phones
>> 1. New call
>> 2. Press 9 access outside line
>> 3. Dial Cell Number
>> 4. Transfer the call that way.
>> 
>
> Once you initiate a new call you will tie up the second line. Your asterisk
> box will now be bridging the two lines. The lines will stay tied up until
> the salesman drops the call.
>
> One method you might be able to employ here would be to add a call transfer
> to the pots lines. Then you would need to send a hook flash to the pots
> line, and dial the salesman's number when you get the dial tone. Then,
> depending on how your local Telco supports the call transfer feature, you
> may be able to free up the line. Not all Telcos support this the same way as
> some consider it a method of toll avoidance and thus drop the call. This
> would be possible in an area where a call from party A to party B is a local
> call and the call from party B to party C is a local call but a call from
> party A to party C would be a toll call. Since the call from party A to
> party C is a toll call, the Telco may opt to drop the call. If the transfer
> part works, there may even be a way to setup the dial plan to intercept your
> phones call transfer feature and use a 1-2 digit code to select which phone
> number to send out. I have not done this but I think it is reasonable as
> I've heard of home users doing it.
>
> By the second option, are you talking about the TDMA/GSM gateway? If so, yes
> this is pretty slick. We considered it initially as well. Our decision not
> to use it was based on the fact that many of our agents are on different
> mobile plans. I think when we requested the info from the agents we had 6
> different wireless companies represented. Since Sprint/Nextel,
> Cingular/AT&T, T-Mobile don't like to play nicely with other, we didn't
> expect to see any real savings from the free mobile to mobile calls. This
> was mainly due to the fact that we don't pay for the agents phones and thus
> we can't really tell the agents which carriers to use. I do know of a couple
> of installations where the company does provide the phones and I understand
> the savings can be significant. I was told by friend that the box they
> installed paid for itself in just a couple of months. But their phone were
> already on the same plan.
>
> John
>
>
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Re: [asterisk-users] kore dump

[Vadim Berezniker]

use the safe_asterisk script

 

it will restart asterisk if it crashes and it enables core dumps (your core 
size limit is probably set to 0 when you start asterisk).

 

From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ed Nuñez
Sent: Tuesday, June 26, 2007 2:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion; [EMAIL PROTECTED]
Subject: [asterisk-users] kore dump

 

I am running Asterisk 1.4.5 and addons 1.4.1 in a CentOS 5 Server.

 

My PBX has experienced several core dumps the last couple of days and I am not 
sure if this is what's causing it, but it always seems to happen when a 
particular extension on a grandstream phone uses ChanSpy SIP group.

 

I have not been able to locate where the core dump file is being saved.   I 
can't find it in my TMP directory.

 

I would also like to know if Asterisk can be setup to automatically re start if 
there is a core dump.  I was thinking of setting up a cron job to launch 
Asterisk every minute.  If it's running, no harm done, and if it crashes, the 
cron job will make sure that it's started every 60 seconds.

 

Any suggestions?

 

 

Thank you

 

Ed Nuñez

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Re: [asterisk-users] kore dump

[John Faubion]

 > I would also like to know if Asterisk can be setup to automatically re
start if there is a core dump.

Sure! You should already have the required script. Just run it from
safe_asterisk. Here is a link with more info:
http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current/docs
-html/x389.html


John
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[asterisk-users] TE412 / HPDL380G5 / * 1.4 / CentOS 4.5 Experience

[James FitzGibbon]

Has anyone successfully run * 1.4 with the following configuration (or
something very similar)?

HP DL380 G5 (3Ghz Xeon)
CentOS 4.5 (kernel 2.6.9-55)
Asterisk 1.4.5 (or 1.4.4)
Zaptel 1.4.3 (or 1.4.2.1)
TE412P
TDM400B (2x FXO and 2x FXS modules)

I've had this rig running * 1.2.18 with Zaptel 1.2.17.1 for several months
without any issues.  Upon trying to upgrade to * 1.4.4 and Zaptel 1.4.2.1 a
few weeks ago, I saw several kernel panics, easily reproduceable with a load
test suite that bridged calls from the PSTN to SIP desk phones.  The one
time I exposed it to real world traffic (a small office, 30 extensions), I
saw five panics in the space of three hours.

Interestingly, when the load test only walked through the IVR and queues, I
couldn't get it to panic, even if I filled the TE412P and had 90
simultaneous calls going through the system.  Only bridged calls seem to
cause problems.

The panics didn't seem to follow any particular pattern.  I saw NMI traps a
few times, then failures in the wct4xxx interrupt handler, a few "swap
tainted" errors, etc.

To rule out an interaction between the TE412P and the HP motherboard, I
moved the installation to an Intel server board (model SE7320) with a 3 Ghz
Pentium D.  The hard drive was a direct clone from the HP.  Running the same
load test against 1.4.4/1.4.2.1 - no panics.  Running the load test against
1.4.5/1.4.3 - no panics.  I'm now nearly 6 hours into having the system
exposed to real-world traffic - no panics.

At this point, I'm pretty much ready to just put this Intel board into a
rackmount server and be done with it, but I am interested to see if anyone
else has seen similar problems, or if anyone has run this configuration
without any problems.  Part of me is saying "I can't be the only person to
run * 1.4 on a current HP server", which in turn leads me to wonder if this
is an incompatibility with the specific board I have, or if I've got a
faulty server (though I have run the full HP diagnostic suite several
times).

Thanks for any feedback you can provide.

--
j.
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[asterisk-users] kore dump

[Ed Nuñez]

I am running Asterisk 1.4.5 and addons 1.4.1 in a CentOS 5 Server.

 

My PBX has experienced several core dumps the last couple of days and I am not 
sure if this is what's causing it, but it always seems to happen when a 
particular extension on a grandstream phone uses ChanSpy SIP group.

 

I have not been able to locate where the core dump file is being saved.   I 
can't find it in my TMP directory.

 

I would also like to know if Asterisk can be setup to automatically re start if 
there is a core dump.  I was thinking of setting up a cron job to launch 
Asterisk every minute.  If it's running, no harm done, and if it crashes, the 
cron job will make sure that it's started every 60 seconds.

 

Any suggestions?

 

 

Thank you

 

Ed Nuñez

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Re: [asterisk-users] Inexpensive Layer 3 Switch?

[David Gomillion]

On 6/26/07, Marty Mastera <[EMAIL PROTECTED]> wrote:


  The only reason to route the voice VLAN is if you need the phones to
access the Internet and/or vice-versa. If you only need to worry about the
computers on the data VLAN accessing Trixbox's web interface, I would
suggest using the Ethernet VLAN capabilities of Linux. You can create
eth0.vlan1 for data on Trixbox, and have the "default" vlan for the port
on the switch be voice. Then, the voice VLAN goes nowhere but to your PBX
and the phones.

The other option is to put in another NIC, one for the voice VLAN, the
other for the data VLAN.

I've been pretty happy with the Linksys 24-port layer 2 switches
(SRW224P). They're running around $400 right now. If you really need layer3
support, I would steer clear of the Netgear. I've had a lot of problems with
them, and the support was disappointing. But then again, I got a bunch that
don't work that I could sell you ;)





Ahh, interesting idea…if I understood correctly, you're basically using a
layer 2 switch and trunking the voice and data VLAN to the asterisk box and
doing the routing and ACL work there?  Advantage is lower cost because you
don't need a layer 3 switch anymore and don't have to learn a new CLI or
other config method.?

Here's a bit more information…the client is a building owner who occupies
the first floor and is renting out the rest of the building.  In addition to
his own voice/data network (which would be on separate VLANs) they want to
offer the building tenants the ability to use their PBX and internet
connection.  Due to a quirk in the service providers SIP ALG all  IP phones
in the building must be on the same network (VLAN) which I don't see a
problem with, but each tenant's data will be in a separate VLAN.  I'm
thinking I could trunk the voice VLAN and all of the individual tenant data
VLANs to the Trixbox to allow them access to the web interface?

Any other ideas out there based on this scenario?



We do something somewhat similar. Each switch has 2 data VLANs, and also is
part of the Voice VLAN. Each VLAN for data is routed, but the voice VLAN
only carries voice traffic. Our Asterisk server does not route packets
between the networks. So, aside from some nasty attacks that sniff and
replicate VLAN headers, our voice network is pretty secure.

So our network has 20 different data VLANs (again, 2 per edge switch), 1
server VLAN, 1 voice VLAN, 1 wireless VLAN, and one DMZ VLAN. The data and
server VLANs are all routed, and everything else is not. They have to go
through some type of bridge between the networks. For wireless, that's our
wireless switch. For the DMZ, it's our firewall. The voice VLAN can only
reach our Asterisk box.

If you use a SIP provider, you may have to either take another approach, or
realize that all SIP traffic will have to remain on the host (i.e. reinvites
are bad when you don't have a network path from A to B). But we're strictly
IAX between offices, and PSTN thru PRI.
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Re: [asterisk-users] More FAX over T1

[Joe acquisto]

. . .
> It looks to me like you have two choices.  The first you probably
> can't do.  That is, get a two port board in the Asterisk system with
> the second T1 going into an Eicon Board in a Hylafax system.  Then,
> you can assign DIDs with whatever web interface you have on this
> Asterisk system to go to the second T1 port.
> 
> The second alternative is to get a second T1 and an Eicon card going
> into a Hylafax Server.  This solution has a big monthly expense to it,
> especially if you aren't fully utilizing all channels on your existing
> T1.
> 
> I'm moving towards the first solution being we send out (and receive)
> large faxes and the IAXModem solution, because of patent issues, is
> not able to send at the faster speeds.  I've received complaints about
> our slow fax machine.  The Eicon card can support the faster
> transmission.
> 

No solution, thus far, seems very "cost effective" for this client.   The Eicon 
and other T1/fax board
are in the 4-7K$ range.  

As this venture is still in it's infancy, it would not be acceptable to shell 
out such, at the moment.

One idea is to utilize DID, and have Asterisk "forward" the calls to the 
current FAX lines, preserving the DID as "Caller ID".  I am fairly sure 
Asterisk itself can do this. (The call would appear to be "from" this assigned 
ID).  If so, I could, apparently, massage Hylafax into dealing with each FAX 
based on the "Caller ID".

joe a.


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Re: [asterisk-users] More FAX over T1

[David Gomillion]

On 6/26/07, Joe acquisto <[EMAIL PROTECTED]> wrote:


This install uses a Sangoma card.

Could you expand on "redirect to a channel bank"?  Could you illuminate
the connectivity for me?

A single T1 connects to???   Is the Digium card "smart" as in, can it
"break out" DS0 line(s) on a second port (to go to the channel bank)?



What we did is have 1 PRI (over T1 in the US) coming in from the telco, into
a 4-port T1 card. Then, we have 2 channel banks coming off of it: 1 has an
external echo can, and goes to our in-house phone extensions. Cordless
phones, wall-hanging phones, and anywhere that we couldn't get 2 pair into.
Basically any required analog that a person would be on. The second channel
bank has no echo can on it, and connects to our modems, fax machines, etc.

Assume for a moment that your incoming lines are in zaptel group 0, your
voice channel bank lines are in group 1, and your other channel bank is in
group 2

exten => 55,1,Goto(default,1000,1) ;go into the internal context to
route the call
exten => 56,1,Dial(Zap/25) ;ring one phone
exten => 57,1,Dial(Zap/G2/${EXTEN}) ;go out group 2, starting at
highest channel number, since the incoming calls probably start at the
lowest channel numbers, and best not to have any contention, in my opinion

In this way, Asterisk will establish a new call going out Group 2 and dial
your number. The box receiving the faxes will get the extension, and as long
as you've left your DID in there, that's what will get passed. So, it will
appear to your fax box as if it were sitting on the PSTN. Asterisk just has
to know which DIDs should have the calls passed along.

Now, in practice, we do a lot more than the above snippet, and use macros
extensively. But this should get you pushed in the right direction.

Hope that helps,
David
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Re: [asterisk-users] Inexpensive Layer 3 Switch?

[Marty Mastera]

The only reason to route the voice VLAN is if you need the phones to access the 
Internet and/or vice-versa. If you only need to worry about the computers on 
the data VLAN accessing Trixbox's web interface, I would suggest using the 
Ethernet VLAN capabilities of Linux. You can create eth0.vlan1 for data on 
Trixbox, and have the "default" vlan for the port on the switch be voice. Then, 
the voice VLAN goes nowhere but to your PBX and the phones.

The other option is to put in another NIC, one for the voice VLAN, the other 
for the data VLAN. 

I've been pretty happy with the Linksys 24-port layer 2 switches (SRW224P). 
They're running around $400 right now. If you really need layer3 support, I 
would steer clear of the Netgear. I've had a lot of problems with them, and the 
support was disappointing. But then again, I got a bunch that don't work that I 
could sell you ;) 

 

 

Ahh, interesting idea…if I understood correctly, you’re basically using a layer 
2 switch and trunking the voice and data VLAN to the asterisk box and doing the 
routing and ACL work there?  Advantage is lower cost because you don’t need a 
layer 3 switch anymore and don’t have to learn a new CLI or other config 
method.?

Here’s a bit more information…the client is a building owner who occupies the 
first floor and is renting out the rest of the building.  In addition to his 
own voice/data network (which would be on separate VLANs) they want to offer 
the building tenants the ability to use their PBX and internet connection.  Due 
to a quirk in the service providers SIP ALG all  IP phones in the building must 
be on the same network (VLAN) which I don’t see a problem with, but each 
tenant’s data will be in a separate VLAN.  I’m thinking I could trunk the voice 
VLAN and all of the individual tenant data VLANs to the Trixbox to allow them 
access to the web interface?

Any other ideas out there based on this scenario?

 

Thanks again


No virus found in this outgoing message.
Checked by AVG Free Edition. 
Version: 7.5.476 / Virus Database: 269.9.9/870 - Release Date: 6/26/2007 10:07 
AM
 
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Re: [asterisk-users] DTMF doesn't work between Asterisk and Cisco SIP Proxy

[Jason Ma]

Buddies,
Thanks for  you response.
I have resolved the issue,it was not the DTMF mismatch between
Asterisk and Cisco proxy.
In fact,there is a Convedia media box behind Cisco proxy as conference
bridge,after checked the whole trace through the patch,I found that my
asterisk send video codec information in the SDP of invite,but the
Convedia media box doesn't support video,it got the request but did
not reject,just add a blank IP address 0.0.0.0 in SDP of 200 OK,so
there were two sections in 200 OK SDP,one is audio section with audio
IP address and port,the other is video section with a 0.0.0.0 IP
address.
When Asterisk got 200 OK,it was strange that it treated the video IP
0.0.0.0 as audio address,so the call was established but could not go
on,that was why I input anything it did not work,I think.
When I disabled the video support in Asterisk,it worked.

On 6/26/07, Ed Nuñez <[EMAIL PROTECTED]> wrote:
> To configure the Cisco for RFC 2833 add the following line to the desired
> dial-peer
>
> dtmf-relay rtp-nte
>
> Hope this helps.
>
> Ed Nuñez
>
>
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Eric
> "ManxPower" Wieling
> Sent: Tuesday, June 26, 2007 11:41 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] DTMF doesn't work between Asterisk and Cisco
> SIP Proxy
>
> This is usually a Cisco issue.
>
> You need to set the Cisco to use RFC2833 DTMF.  Check the Cisco docs.
>
> tracinet wrote:
> > Jason,
> > I am at least having similar issues with rfc2833 DTMF:
> >
> > http://bugs.digium.com/view.php?id=10058
> >
> >
> > On 6/20/07, Jason Ma <[EMAIL PROTECTED]> wrote:
> >>
> >> Hi buddies,
> >> I encountered DTMF issue when I tried to place call from x-lite to a
> >> sip conference serice,here is the diagram.
> >> X-lite>Asterisk--->Cisco SIP proxy>SIP Conference service
> >>
> >> The Call can be established,and I can hear from x-lite the prompt of
> >> the conference,but when I input any digits,nothing happened,the
> >> conference service did not recognize my input.At the same time,in the
> >> log of asterisk,I can find that asterisk recognized all the
> >> digitsI tried "rfc2833","inband","info" in the "dtmfmode"
> >> parameter,but did not work ,I'm not sure whether asterisk send the
> >> right dtmf to cisco proxy,how can I track that?
> >>
> >> I made another test,dialing from x-lite registered with Cisco proxy to
> >> voicemail service of Asterisk.
> >> x-lite>Cisco SIP proxy>Asterisk--->Voicemail service
> >>
> >> Both the call and dtmf worked fine,I can input my mailbox number and
> >> password and listen my  voicemail.both "rfc2933" and "inband" worked
> >> in this situation,but not "info".
> >>
> >> My Asterisk is 1.4.4 with asterisk now,I did not configure dtmfmode in
> >> the section of  xlite and the trunk to cisco proxy,just configure the
> >> dtmfmode in sip.conf.
> >>
> >> When I used "rfc2833",I can see the log in asterisk as :
> >>
> >> [2007-06-19 16:01:40] DTMF[8925] channel.c: DTMF begin '2' received on
> >> SIP/-08269470
> >> [2007-06-19 16:01:41] DTMF[8925] channel.c: DTMF end '2' received on
> >> SIP/-08269470, duration 160 ms
> >> [2007-06-19 16:01:42] DTMF[8925] channel.c: DTMF begin '1' received on
> >> SIP/-08269470
> >> [2007-06-19 16:01:42] DTMF[8925] channel.c: DTMF end '1' received on
> >> SIP/-08269470, duration 140 ms
> >>
> >> and when I used "inband",I can see :
> >>
> >> [2007-06-19 15:55:21] DTMF[8852] channel.c: DTMF end '2' received on
> >> SIP/-09d916c0, duration 0 ms
> >> [2007-06-19 15:55:22] DTMF[8852] channel.c: DTMF end '1' received on
> >> SIP/-09d916c0, duration 0 ms
> >>
> >> Is that right?Can I check what digits that asterisk sent out ?
> >>
> >> How can I track where is wrong with the dtmf?Did asterisk send dtmf to
> >> Cisco proxy correctly?
> >> I really have no idea about that.Please advise.Thank you very much
> >>
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> >>http://lists.digium.com/mailman/listinfo/asterisk-users
> >>
> >
> >
> > 
> >
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Re: [asterisk-users] Ring/Off-hook in strange state 6

[Eric \"ManxPower\" Wieling]

Daniel already pointed you in the right direction.

I have seen this error many times, but it never causes a problem.

Alex Mcdowell wrote:
> Can anybody at least point me in a direction??
> 
> On 6/25/07, Alex Mcdowell <[EMAIL PROTECTED]> wrote:
>> I don't think my cards are bad, but maybe there is a problem with the
>> one. It has been two weeks since I put my ticket in with Digium...and
>> still no word. I am starting to get frustrated.
>>
>> On 6/22/07, Daniel Hazelbaker <[EMAIL PROTECTED]> wrote:
>>>
>>> Alex,
>>>
>>>
>>>   I had this problem with a new TDM2400 card that we purchased.  
>>> Specifically I would get that message and then it would pick up the ringing 
>>> line AND the line next to it.  Basically, lines 1 & 2 had been cross-linked 
>>> somehow.  After a few weeks of trouble-shooting with Digium tech support 
>>> they cross-shipped me a new card and the problem (and that message) went 
>>> away.
>>>
>>>
>>> Daniel Hazelbaker
>>> High Desert Church
>>>
>>>
>>>
>>> On Jun 22, 2007, at 1:22 PM, Alex Mcdowell wrote:
>>>
>>>
>>>
>>> HI I have two servers both of which get this message on one of the lines.
>>>
>>> Ring/Off-hook in strange state 6. The one server seems to be ok with it, but
>>>
>>> the other one when an extension picks up there is no one there and the
>>>
>>> incoming call keeps ringing. I tried to adjust the levels in wcfxo.c like
>>>
>>> someone had suggested, but it didn't do anything. I also upgraded zaptel to
>>>
>>> the latest. 1.2.18 and asterisk is running at 1.2.14. callprogess is set to
>>>
>>> no, as well as busydetect=no. This is a major problem since this box only
>>>
>>> has 1 other line, but at least it works. I can't seem to find much info on
>>>
>>> this issue. I can't believe others haven't run into it.  I started a ticket
>>>
>>> with digium, but I guess they are pretty backed up. Here is what I am
>>>
>>> getting in the CLI:  Thanks for any help -Alex
>>>
>>> -- Starting simple switch on 'Zap/4-1'
>>>
>>> -- Executing Wait("Zap/4-1", "1") in new stack
>>>
>>> -- Executing Answer("Zap/4-1", "") in new stack
>>>
>>> -- Executing Dial("Zap/4-1", "SIP/4125|20|tr") in new stack
>>>
>>> -- Called 4125
>>>
>>> Jun 22 13:44:45 WARNING[24296]: chan_zap.c:3926 zt_handle_event:
>>>
>>> Ring/Off-hook in strange state 6 on channel 4
>>>
>>> -- SIP/4125-09559118 is ringing
>>>
>>> Jun 22 13:44:46 WARNING[24296]: chan_zap.c:3926 zt_handle_event:
>>>
>>> Ring/Off-hook in strange state 6 on channel 4
>>>
>>> -- SIP/4125-09559118 answered Zap/4-1
>>>
>>>   == Spawn extension (incoming, s, 3) exited non-zero on 'Zap/4-1'
>>>
>>> -- Hungup 'Zap/4-1'
>>>
>>> -- Starting simple switch on 'Zap/4-1'
>>>
>>> -- Executing Wait("Zap/4-1", "1") in new stack
>>>
>>> -- Executing Answer("Zap/4-1", "") in new stack
>>>
>>> -- Executing Dial("Zap/4-1", "SIP/4125|20|tr") in new stack
>>>
>>> -- Called 4125
>>>
>>> Jun 22 13:44:57 WARNING[24303]: chan_zap.c:3926 zt_handle_event:
>>>
>>> Ring/Off-hook in strange state 6 on channel 4
>>>
>>> -- SIP/4125-09559118 is ringing
>>>
>>> Jun 22 13:44:58 WARNING[24303]: chan_zap.c:3926 zt_handle_event:
>>>
>>> Ring/Off-hook in strange state 6 on channel 4
>>>
>>> -- SIP/4125-09559118 answered Zap/4-1
>>>
>>>   == Spawn extension (incoming, s, 3) exited non-zero on 'Zap/4-1'
>>>
>>> -- Hungup 'Zap/4-1'
>>>
>>> -- Starting simple switch on 'Zap/4-1'
>>>
>>> -- Executing Wait("Zap/4-1", "1") in new stack
>>>
>>> -- Executing Answer("Zap/4-1", "") in new stack
>>>
>>> -- Executing Dial("Zap/4-1", "SIP/4125|20|tr") in new stack
>>>
>>> -- Called 4125
>>>
>>> -- SIP/4125-09559118 is ringing
>>>
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Re: [asterisk-users] Bad Echo between SIP calls

[Mojo with Horan & Company, LLC]

First of all, Alex, sorry for not seeing your reply.  Nearly two weeks 
ago now :(

Honestly, with canreinvite=yes, I'm not sure what is meant by "the 
signalling still travels through asterisk"... I would ASSUME that 
includes out-of-band dtmf as well.  Sorry!

Moj

Alex Crow wrote:
> Moj,
> 
> Does this mean that even out-of-band DTMF still gets sent
> SIP-phone<-->SIP-phone without Asterisk hearing them? (eg RFC DTMF,
> can't remember the number right now)
> 
> Forgive me for butting into this thread but this is interesting...
> 
> Cheers
> 
> Alex
> 
> 
> On Tue, 2007-06-12 at 09:21 -0800, Mojo with Horan & Company, LLC wrote:
>> theoretically, with canreinvite=yes, it's phone <-> phone.  with 
>> canreinvite=no, it's phone <-> asterisk <-> phone.   BUT there are a few 
>> reasons which canreinvite=yes will not be this way.  If for example you 
>> have a T or a t in the Dial string, asterisk will _remain_ in the media 
>> path so it can still detect the DTMF requests for transfer.
>>
>> Moj
>>
>> Deepak Naidu wrote:
>>> Sounds crazy right? even was I, more over support guy logged in unloaded 
>>> the zap modules to test them, still an echo.
>>>
>>> Ya, I was clear saying that we have SIP--- SIP issue ie internal 
>>> extension echo problem.  It seems the echo with SIP--SIP has many 
>>> factors.  I am just curios to eliminate any possibility of Asterisk 
>>> failing to cancel the echo.
>>>
>>> OK, one question here howz the call flow when a SIP---SIP call is 
>>> established ie.  is the connection between 2 phones when an Internal 
>>> call is made or does the SIP call goes via Asterisk once the SIP--SIP 
>>> call is establised.
>>>
>>> --
>>> Deepak
>>>
>>> */Matthew Fredrickson <[EMAIL PROTECTED]>/* wrote:
>>>
>>>
>>> On Jun 8, 2007, at 6:00 PM, Deepak Naidu wrote:
>>>
>>>  > Hi,
>>>  >   We have a PRI connection & when its was on test
>>> networks we
>>>  > had echo problems withoutside line. 
>>>  >
>>>  > So I bought a TE212P card resolve the echo problem.  Which did to an
>>>  > extent. Its using asterisk 1.2.18 & RHEL4-Update 4.
>>>  >
>>>  >
>>>  > But now when we are live, there is a terrible echo between 2 SIP
>>>  > calls. If I call the same extension from outside the voice is clear.
>>>  >
>>>  > I am not sure whats the problem.  Also there's slight echo when
>>>  > calling Digium support.
>>>  >
>>>  > Totally lost Digium says we need to remove the echo module to
>>> resolve
>>>  > SIP echo problems. Then ? the heck we pay for..
>>>
>>> Are you sure that they understood that you were having this problem
>>> between 2 SIP endpoints? That advice only makes sense to test if one
>>> side is Zap and the other side is SIP.
>>>
>>>
>>> ---
>>> Matthew Fredrickson
>>> Software Engineer
>>> Digium, Inc.
>>>
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>>>
>>> 
>>> Yahoo! Answers - Get better answers from someone who knows. Try it now 
>>> .
>>>  
>>>
>>>
>>>
>>> 
>>>
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Re: [asterisk-users] More FAX over T1

[Lacy Moore - Aspendora]

On 6/26/07, Joe acquisto <[EMAIL PROTECTED]> wrote:
> Thoughts vary to second T1, with channel bank, "breaking out" some DS0's into 
> a channel bank, or finding a T1/fax board (do they exist?), to go directly 
> into the FAX server (PC/linux based)
>

It looks to me like you have two choices.  The first you probably
can't do.  That is, get a two port board in the Asterisk system with
the second T1 going into an Eicon Board in a Hylafax system.  Then,
you can assign DIDs with whatever web interface you have on this
Asterisk system to go to the second T1 port.

The second alternative is to get a second T1 and an Eicon card going
into a Hylafax Server.  This solution has a big monthly expense to it,
especially if you aren't fully utilizing all channels on your existing
T1.

I'm moving towards the first solution being we send out (and receive)
large faxes and the IAXModem solution, because of patent issues, is
not able to send at the faster speeds.  I've received complaints about
our slow fax machine.  The Eicon card can support the faster
transmission.

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Re: [asterisk-users] Ring/Off-hook in strange state 6

[Bruce Reeves]

I have seen this on cards waiting for the callerID and there being a problem
with the callerid signal. Is callerid working on theses lines?

On 6/26/07, Alex Mcdowell <[EMAIL PROTECTED]> wrote:


Can anybody at least point me in a direction??

On 6/25/07, Alex Mcdowell <[EMAIL PROTECTED]> wrote:
> I don't think my cards are bad, but maybe there is a problem with the
> one. It has been two weeks since I put my ticket in with Digium...and
> still no word. I am starting to get frustrated.
>
> On 6/22/07, Daniel Hazelbaker <[EMAIL PROTECTED]> wrote:
> >
> >
> > Alex,
> >
> >
> >   I had this problem with a new TDM2400 card that we
purchased.  Specifically I would get that message and then it would pick up
the ringing line AND the line next to it.  Basically, lines 1 & 2 had been
cross-linked somehow.  After a few weeks of trouble-shooting with Digium
tech support they cross-shipped me a new card and the problem (and that
message) went away.
> >
> >
> > Daniel Hazelbaker
> > High Desert Church
> >
> >
> >
> > On Jun 22, 2007, at 1:22 PM, Alex Mcdowell wrote:
> >
> >
> >
> > HI I have two servers both of which get this message on one of the
lines.
> >
> > Ring/Off-hook in strange state 6. The one server seems to be ok with
it, but
> >
> > the other one when an extension picks up there is no one there and the
> >
> > incoming call keeps ringing. I tried to adjust the levels in wcfxo.clike
> >
> > someone had suggested, but it didn't do anything. I also upgraded
zaptel to
> >
> > the latest. 1.2.18 and asterisk is running at 1.2.14. callprogess is
set to
> >
> > no, as well as busydetect=no. This is a major problem since this box
only
> >
> > has 1 other line, but at least it works. I can't seem to find much
info on
> >
> > this issue. I can't believe others haven't run into it.  I started a
ticket
> >
> > with digium, but I guess they are pretty backed up. Here is what I am
> >
> > getting in the CLI:  Thanks for any help -Alex
> >
> > -- Starting simple switch on 'Zap/4-1'
> >
> > -- Executing Wait("Zap/4-1", "1") in new stack
> >
> > -- Executing Answer("Zap/4-1", "") in new stack
> >
> > -- Executing Dial("Zap/4-1", "SIP/4125|20|tr") in new stack
> >
> > -- Called 4125
> >
> > Jun 22 13:44:45 WARNING[24296]: chan_zap.c:3926 zt_handle_event:
> >
> > Ring/Off-hook in strange state 6 on channel 4
> >
> > -- SIP/4125-09559118 is ringing
> >
> > Jun 22 13:44:46 WARNING[24296]: chan_zap.c:3926 zt_handle_event:
> >
> > Ring/Off-hook in strange state 6 on channel 4
> >
> > -- SIP/4125-09559118 answered Zap/4-1
> >
> >   == Spawn extension (incoming, s, 3) exited non-zero on 'Zap/4-1'
> >
> > -- Hungup 'Zap/4-1'
> >
> > -- Starting simple switch on 'Zap/4-1'
> >
> > -- Executing Wait("Zap/4-1", "1") in new stack
> >
> > -- Executing Answer("Zap/4-1", "") in new stack
> >
> > -- Executing Dial("Zap/4-1", "SIP/4125|20|tr") in new stack
> >
> > -- Called 4125
> >
> > Jun 22 13:44:57 WARNING[24303]: chan_zap.c:3926 zt_handle_event:
> >
> > Ring/Off-hook in strange state 6 on channel 4
> >
> > -- SIP/4125-09559118 is ringing
> >
> > Jun 22 13:44:58 WARNING[24303]: chan_zap.c:3926 zt_handle_event:
> >
> > Ring/Off-hook in strange state 6 on channel 4
> >
> > -- SIP/4125-09559118 answered Zap/4-1
> >
> >   == Spawn extension (incoming, s, 3) exited non-zero on 'Zap/4-1'
> >
> > -- Hungup 'Zap/4-1'
> >
> > -- Starting simple switch on 'Zap/4-1'
> >
> > -- Executing Wait("Zap/4-1", "1") in new stack
> >
> > -- Executing Answer("Zap/4-1", "") in new stack
> >
> > -- Executing Dial("Zap/4-1", "SIP/4125|20|tr") in new stack
> >
> > -- Called 4125
> >
> > -- SIP/4125-09559118 is ringing
> >
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--
Bruce Reeves
Nortex Networks
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[asterisk-users] Slip Events

[Jon Weisman]

All,

I'm using  a Digium TE410P w/ Asterisk 1.2.18. Trying to connect it to our NACT 
STX switch via PRI, d channel is up, T1 shows normal, but I'm getting crazy 
errors. I rewired this thing three times, then I connected the same cable from 
the STX to a Cisco AS5300 (same pri settings as asterisk), and all slip events 
and frame sync errors went away, so the cable is good. 

STX --> DSX Panel -->Asterisk

I've confrimed all wiring is good, not sure what the issue is, tried playing 
with the clock source as well, but no dice. Any ideas? 

Errors:

CRC Errors
Frame Sync Errors
Slip Events

1 Error every second

Zaptel.conf
loadzone= us
defaultzone = us
span=1,1,0,esf,b8zs
span=2,2,0,esf,b8zs
span=3,3,0,esf,b8zs
span=4,4,0,esf,b8zs
bchan=1-23
dchan=24
bchan=25-47
dchan=48
bchan=49-71
dchan=72
bchan=73-95
dchan=96


TIA,

Jon Weisman | Sales Engineer
International Bell Communications
www.ibell.net
www.ibellhost.com
www.aeronhelpdesk.com
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Re: [asterisk-users] More FAX over T1

[Joe acquisto]

. . .
> With all due respect, this project should be handed over to whomever has
> authorization to administer the Asterisk box. We can tell you how to do it
> in Asterisk, but if you can't take our advice, our ability to help you will
> be severely limited.

Thanks.  Point taken.  I'm, unfortunately, playing a form of "monkey in the 
middle".  Seems the Vendor is
unwilling to, unable to, or is outrageously priced.  I am not privy to any of 
those discussions.

I view this as a learning experience for me.

> Now, we have many, many fax machines. We have our incoming through PRI, and
> then redirect to a channel bank. We have no problems with fax reception.
> When we used a Sangoma card, we did, but now that we're back on Digium
> hardware, we've been doing well, thus far. Probably had to do with the echo
> cancellation, but without infinite time to troubleshoot, we just had to get
> it working.

This install uses a Sangoma card.

Could you expand on "redirect to a channel bank"?  Could you illuminate the 
connectivity for me?

A single T1 connects to???   Is the Digium card "smart" as in, can it "break 
out" DS0 line(s) on a second port (to go to the channel bank)?

I am not that familiar with that technology.  As may be evident.

joe a.


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Re: [asterisk-users] Inexpensive Layer 3 Switch?

[John Faubion]

> If you really need layer3 support, I would steer clear of the Netgear.
I've had a lot of problems
> with them, and the support was disappointing.

What model did you use? I've been very happy with the FS728TP as I mentioned
earlier. I haven't had any problems so far. Granted I haven't had to call
Netgear so I don't have anything on which to judge their service. I will say
that unless your dealing with a very small system, you should probably steer
away from the FS726TP. It only supports PoE on the first 12 ports and
doesn't have anywhere near the features of the FS728TP.

John

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Re: [asterisk-users] Ring/Off-hook in strange state 6

[Alex Mcdowell]

Can anybody at least point me in a direction??

On 6/25/07, Alex Mcdowell <[EMAIL PROTECTED]> wrote:
> I don't think my cards are bad, but maybe there is a problem with the
> one. It has been two weeks since I put my ticket in with Digium...and
> still no word. I am starting to get frustrated.
>
> On 6/22/07, Daniel Hazelbaker <[EMAIL PROTECTED]> wrote:
> >
> >
> > Alex,
> >
> >
> >   I had this problem with a new TDM2400 card that we purchased.  
> > Specifically I would get that message and then it would pick up the ringing 
> > line AND the line next to it.  Basically, lines 1 & 2 had been cross-linked 
> > somehow.  After a few weeks of trouble-shooting with Digium tech support 
> > they cross-shipped me a new card and the problem (and that message) went 
> > away.
> >
> >
> > Daniel Hazelbaker
> > High Desert Church
> >
> >
> >
> > On Jun 22, 2007, at 1:22 PM, Alex Mcdowell wrote:
> >
> >
> >
> > HI I have two servers both of which get this message on one of the lines.
> >
> > Ring/Off-hook in strange state 6. The one server seems to be ok with it, but
> >
> > the other one when an extension picks up there is no one there and the
> >
> > incoming call keeps ringing. I tried to adjust the levels in wcfxo.c like
> >
> > someone had suggested, but it didn't do anything. I also upgraded zaptel to
> >
> > the latest. 1.2.18 and asterisk is running at 1.2.14. callprogess is set to
> >
> > no, as well as busydetect=no. This is a major problem since this box only
> >
> > has 1 other line, but at least it works. I can't seem to find much info on
> >
> > this issue. I can't believe others haven't run into it.  I started a ticket
> >
> > with digium, but I guess they are pretty backed up. Here is what I am
> >
> > getting in the CLI:  Thanks for any help -Alex
> >
> > -- Starting simple switch on 'Zap/4-1'
> >
> > -- Executing Wait("Zap/4-1", "1") in new stack
> >
> > -- Executing Answer("Zap/4-1", "") in new stack
> >
> > -- Executing Dial("Zap/4-1", "SIP/4125|20|tr") in new stack
> >
> > -- Called 4125
> >
> > Jun 22 13:44:45 WARNING[24296]: chan_zap.c:3926 zt_handle_event:
> >
> > Ring/Off-hook in strange state 6 on channel 4
> >
> > -- SIP/4125-09559118 is ringing
> >
> > Jun 22 13:44:46 WARNING[24296]: chan_zap.c:3926 zt_handle_event:
> >
> > Ring/Off-hook in strange state 6 on channel 4
> >
> > -- SIP/4125-09559118 answered Zap/4-1
> >
> >   == Spawn extension (incoming, s, 3) exited non-zero on 'Zap/4-1'
> >
> > -- Hungup 'Zap/4-1'
> >
> > -- Starting simple switch on 'Zap/4-1'
> >
> > -- Executing Wait("Zap/4-1", "1") in new stack
> >
> > -- Executing Answer("Zap/4-1", "") in new stack
> >
> > -- Executing Dial("Zap/4-1", "SIP/4125|20|tr") in new stack
> >
> > -- Called 4125
> >
> > Jun 22 13:44:57 WARNING[24303]: chan_zap.c:3926 zt_handle_event:
> >
> > Ring/Off-hook in strange state 6 on channel 4
> >
> > -- SIP/4125-09559118 is ringing
> >
> > Jun 22 13:44:58 WARNING[24303]: chan_zap.c:3926 zt_handle_event:
> >
> > Ring/Off-hook in strange state 6 on channel 4
> >
> > -- SIP/4125-09559118 answered Zap/4-1
> >
> >   == Spawn extension (incoming, s, 3) exited non-zero on 'Zap/4-1'
> >
> > -- Hungup 'Zap/4-1'
> >
> > -- Starting simple switch on 'Zap/4-1'
> >
> > -- Executing Wait("Zap/4-1", "1") in new stack
> >
> > -- Executing Answer("Zap/4-1", "") in new stack
> >
> > -- Executing Dial("Zap/4-1", "SIP/4125|20|tr") in new stack
> >
> > -- Called 4125
> >
> > -- SIP/4125-09559118 is ringing
> >
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> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>

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Re: [asterisk-users] Inexpensive Layer 3 Switch?

[Steve Totaro]

Marty Mastera wrote:
>
> Any recommendations on an economical layer 3 switch? Preferably 
> something that you have hands on experience with connecting to IP 
> phones with attached PCs? Specifically I need the ability to set the 
> VLAN in the phone to tag voice packets and to set a native VLAN on a 
> per port basis on the switch to put the untagged packets from the 
> attached PC into a separate VLAN.
>
> POE is not a requirement but if you have suggestions for an economical 
> layer 3 switch with POE I’d be glad to hear them…so far I’m looking at 
> the SFE2000 from Linksys.
>
> thanks
>
>
I have been eying the Dell webmanaged Gigabit switches. They are 
inexpensive, managed, support port mirroring, VLAN, QoS, and much more. 
I am probably going to get the 16 or 24 port for testing. The forwarding 
rates are an often overlooked spec. $202 (currently on sale) for 16 
Gigabit ports is almost too good to be true for a managed switch.

I am in no way recommending this switch since I have yet to test it but 
I certainly will give it a try.

http://www.dell.com/content/products/productdetails.aspx/pwcnt_2716?c=us&cs=04&l=en&s=bsd
16 10/100/1000 BASE-T ports
Auto-negotiation for speed, duplex mode and flow control
Auto MDI/MDIX mode and flow control
Integrated Port LEDs
Individual port controls

Performance 
Switching Capacity 32.0 Gbps
Forwarding Rate 23.7 Mpps

Management  
Web-based management interface
BootP/DHCP IP address management or Static IP address assignment
RMON statistics

Class of Service
Four priority queues per port
Adjustable WRR and strict priority
Layer 2 IEEE 802.1p tagging and port-based priority
Layer 3-aware prioritization using DSCP values

Security
Switch access password protection (read-only and read-write access)
Restricted IP address

VLAN
IEEE 802.1Q port-based tagging up to 64 VLANs
Honors all 4096 VLAN tags

Switching Features  
Link Aggregation, up to six groups and up to four aggregated links per 
group (IEEE 802.3ad)
Port mirroring (up to four source ports)
Jumbo frame support up to 9000 Bytes (2716 & 2724 only)

Availability
Firmware Uploads to the Switch
Broadcast Storm Control
Virtual Cable Tester by Marvell^®
Optical Transceiver Diagnostics

Chassis 
*Dimensions:* 1.70 in (H) x 12.99 in (W) x 9.07 in (D)
*Height:* 1U rack
*Weight:* 6.16 lbs
*Voltage:* 100-240VAC, 50-60Hz

Standards Supported 
IEEE802.3 CSMA/CD
IEEE802.3u 100BaseTx
IEEE802.3z/ab 1000BaseT
IEEE802.3x Flow Control
IEEE 802.1p

Environmental   
Operating Temperature: 0º C to 45º C (32º F to 113º F)
Storage temperature: -20º C to 70º C (-4º F to 158º F)
Operating Humidity: 10% to 90% Relative Humidity
Storage Humidity: 10% to 95% Relate Humidity

Power   
Maximum Power: 1.0A @ 100V



Once upon a time I would say "Go Cisco", but I think Dell has come a 
long way with their servers and switches.

Thanks,
Steve Totaro


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Re: [asterisk-users] More FAX over T1

[David Gomillion]

On 6/26/07, Joe acquisto <[EMAIL PROTECTED]> wrote:


This is a follow up to an earlier post.

Looking for a means to "individualize" incoming FAX, so as to distribute
them to the intended recipient.

While the PBX is based on Asterisk, it is not possible for me to "enter"
the box to modify things, to any great degree.  I thank those who mentioned
IAXMODEM, earlier, but that seems a no go.



With all due respect, this project should be handed over to whomever has
authorization to administer the Asterisk box. We can tell you how to do it
in Asterisk, but if you can't take our advice, our ability to help you will
be severely limited.

Now, we have many, many fax machines. We have our incoming through PRI, and
then redirect to a channel bank. We have no problems with fax reception.
When we used a Sangoma card, we did, but now that we're back on Digium
hardware, we've been doing well, thus far. Probably had to do with the echo
cancellation, but without infinite time to troubleshoot, we just had to get
it working.

I would not recommend passing fax data across the PCI bus between cards. I'm
probably just superstitious, but I wouldn't do it. But it would be very
simple to do with just a Dial statement.

Basically, just go out and try it. Your business requirements and what
you're allowed to do obviously drive where your decision is going to go. If
you get stuck, and can't find answers through Google or the Wiki, then ask
this list. But you can't expect us to tell you what's going to work in your
business when you aren't empowered to follow our advice.
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Re: [asterisk-users] DTMF doesn't work between Asterisk and Cisco SIP Proxy

[Ed Nuñez]

To configure the Cisco for RFC 2833 add the following line to the desired
dial-peer

dtmf-relay rtp-nte

Hope this helps.

Ed Nuñez



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric
"ManxPower" Wieling
Sent: Tuesday, June 26, 2007 11:41 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DTMF doesn't work between Asterisk and Cisco
SIP Proxy

This is usually a Cisco issue.

You need to set the Cisco to use RFC2833 DTMF.  Check the Cisco docs.

tracinet wrote:
> Jason,
> I am at least having similar issues with rfc2833 DTMF:
> 
> http://bugs.digium.com/view.php?id=10058
> 
> 
> On 6/20/07, Jason Ma <[EMAIL PROTECTED]> wrote:
>>
>> Hi buddies,
>> I encountered DTMF issue when I tried to place call from x-lite to a
>> sip conference serice,here is the diagram.
>> X-lite>Asterisk--->Cisco SIP proxy>SIP Conference service
>>
>> The Call can be established,and I can hear from x-lite the prompt of
>> the conference,but when I input any digits,nothing happened,the
>> conference service did not recognize my input.At the same time,in the
>> log of asterisk,I can find that asterisk recognized all the
>> digitsI tried "rfc2833","inband","info" in the "dtmfmode"
>> parameter,but did not work ,I'm not sure whether asterisk send the
>> right dtmf to cisco proxy,how can I track that?
>>
>> I made another test,dialing from x-lite registered with Cisco proxy to
>> voicemail service of Asterisk.
>> x-lite>Cisco SIP proxy>Asterisk--->Voicemail service
>>
>> Both the call and dtmf worked fine,I can input my mailbox number and
>> password and listen my  voicemail.both "rfc2933" and "inband" worked
>> in this situation,but not "info".
>>
>> My Asterisk is 1.4.4 with asterisk now,I did not configure dtmfmode in
>> the section of  xlite and the trunk to cisco proxy,just configure the
>> dtmfmode in sip.conf.
>>
>> When I used "rfc2833",I can see the log in asterisk as :
>>
>> [2007-06-19 16:01:40] DTMF[8925] channel.c: DTMF begin '2' received on
>> SIP/-08269470
>> [2007-06-19 16:01:41] DTMF[8925] channel.c: DTMF end '2' received on
>> SIP/-08269470, duration 160 ms
>> [2007-06-19 16:01:42] DTMF[8925] channel.c: DTMF begin '1' received on
>> SIP/-08269470
>> [2007-06-19 16:01:42] DTMF[8925] channel.c: DTMF end '1' received on
>> SIP/-08269470, duration 140 ms
>>
>> and when I used "inband",I can see :
>>
>> [2007-06-19 15:55:21] DTMF[8852] channel.c: DTMF end '2' received on
>> SIP/-09d916c0, duration 0 ms
>> [2007-06-19 15:55:22] DTMF[8852] channel.c: DTMF end '1' received on
>> SIP/-09d916c0, duration 0 ms
>>
>> Is that right?Can I check what digits that asterisk sent out ?
>>
>> How can I track where is wrong with the dtmf?Did asterisk send dtmf to
>> Cisco proxy correctly?
>> I really have no idea about that.Please advise.Thank you very much
>>
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> 
> 
> 
> 
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Re: [asterisk-users] Problems with ChanIsAvail always return status 0

[Alvaro Parres]

Jared:

  As you see i have the s option. That works fine on Version 1.2. Let me
see config the call limit con sip channels it works.

Thanks.


On 6/25/07, Jared Smith <[EMAIL PROTECTED]> wrote:


On 6/25/07, Alvaro Parres <[EMAIL PROTECTED]> wrote:
> I'm having the next problem, it appear that the application
ChanIsAvail
> is not working on Asterisk 1.4.5 always return me 0 in AVAILSTATUS.
> I add my dialplan and the output to the cli.

This isn't really a problem with ChanIsAvail... it's more of a
misunderstanding of what's going on.  In your case, it appears that
your SIP device will accept multiple calls at the same time from
Asterisk.  So even if your phone is on a call, Asterisk will come
along, try to make another call to it, and the phone says "Hey, go
ahead! I don't mind!"

You've got quite a few options to solve your problem.  While none of
them are exactly perfect, it's good to have lots of options:

o  Try using the 's' option to ChanIsAvail().  (You might have to
turn on call limits in sip.conf to get this to work correctly.  Last
time I played with this, it seems that the limitonpeers setting had to
be set to yes as well.)
o  Use the GROUP() dialplan function to assign calls to call groups,
and then use the GROUP_COUNT() function to check to see if that phone
is already on any calls.
o  Turn off call waiting on your IP phone, so that it'll only accept
one call at a time
o  Simply get call limits in sip.conf working correctly.  (This is
probably the hardest to do, unfortunately.)

Hopefully, one of those options will help you out.  (I've placed them
in the order I'd try... but your mileage may vary.)

-Jared

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--
Alvaro I. Parres Peredo
Director de IT
Grupo Xmarts SA de CV
Tel: +52 (33) 35 63 6261 Ext. 112
 01 800  087 2260
Cel: +52 (33) 33 68 1087
[EMAIL PROTECTED]
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[asterisk-users] SpectraLink SVP protocol support in asterisk

[Michelle Dupuis]

Does anyone know if Asterisk can natively support the SVP protocol from
SpectraLink?  
 
Thanks,
MD
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Re: [asterisk-users] DTMF doesn't work between Asterisk and Cisco SIP Proxy

[Eric \"ManxPower\" Wieling]

This is usually a Cisco issue.

You need to set the Cisco to use RFC2833 DTMF.  Check the Cisco docs.

tracinet wrote:
> Jason,
> I am at least having similar issues with rfc2833 DTMF:
> 
> http://bugs.digium.com/view.php?id=10058
> 
> 
> On 6/20/07, Jason Ma <[EMAIL PROTECTED]> wrote:
>>
>> Hi buddies,
>> I encountered DTMF issue when I tried to place call from x-lite to a
>> sip conference serice,here is the diagram.
>> X-lite>Asterisk--->Cisco SIP proxy>SIP Conference service
>>
>> The Call can be established,and I can hear from x-lite the prompt of
>> the conference,but when I input any digits,nothing happened,the
>> conference service did not recognize my input.At the same time,in the
>> log of asterisk,I can find that asterisk recognized all the
>> digitsI tried "rfc2833","inband","info" in the "dtmfmode"
>> parameter,but did not work ,I'm not sure whether asterisk send the
>> right dtmf to cisco proxy,how can I track that?
>>
>> I made another test,dialing from x-lite registered with Cisco proxy to
>> voicemail service of Asterisk.
>> x-lite>Cisco SIP proxy>Asterisk--->Voicemail service
>>
>> Both the call and dtmf worked fine,I can input my mailbox number and
>> password and listen my  voicemail.both "rfc2933" and "inband" worked
>> in this situation,but not "info".
>>
>> My Asterisk is 1.4.4 with asterisk now,I did not configure dtmfmode in
>> the section of  xlite and the trunk to cisco proxy,just configure the
>> dtmfmode in sip.conf.
>>
>> When I used "rfc2833",I can see the log in asterisk as :
>>
>> [2007-06-19 16:01:40] DTMF[8925] channel.c: DTMF begin '2' received on
>> SIP/-08269470
>> [2007-06-19 16:01:41] DTMF[8925] channel.c: DTMF end '2' received on
>> SIP/-08269470, duration 160 ms
>> [2007-06-19 16:01:42] DTMF[8925] channel.c: DTMF begin '1' received on
>> SIP/-08269470
>> [2007-06-19 16:01:42] DTMF[8925] channel.c: DTMF end '1' received on
>> SIP/-08269470, duration 140 ms
>>
>> and when I used "inband",I can see :
>>
>> [2007-06-19 15:55:21] DTMF[8852] channel.c: DTMF end '2' received on
>> SIP/-09d916c0, duration 0 ms
>> [2007-06-19 15:55:22] DTMF[8852] channel.c: DTMF end '1' received on
>> SIP/-09d916c0, duration 0 ms
>>
>> Is that right?Can I check what digits that asterisk sent out ?
>>
>> How can I track where is wrong with the dtmf?Did asterisk send dtmf to
>> Cisco proxy correctly?
>> I really have no idea about that.Please advise.Thank you very much
>>
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> 
> 
> 
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Re: [asterisk-users] CDR Records "s" as dst

[Eric \"ManxPower\" Wieling]

Does doing it this way give you the correct DST?

[macro-dialout]

exten = s,1,Goto(${MACRO_EXTEN},1)

exten = _X.,1,Set(TOUCH_MONITOR=${TIMESTAMP}_${CALLERID(num)}-${ARG1})
exten = _X.,n,NoOp(CID_NAME  : ${CID_NAME})
exten = _X.,n,NoOp(CID_NUMBER: ${CID_NUMBER})
exten = _X.,n,NoOp(CID_CLIR  : ${CID_CLIR})
exten = _X.,n,NoOp(TRUNK : ${TRUNK})

   [rest of your macro]


Jaswinder Singh wrote:
> This is due to changes in cdr in asterisk 1.4.5 so in all outgoing
> dial from macro it shows 's' in cdr . Could this be a bug ? I doubt if
> it was intended to be that way .
> 
> On 25/06/07, Troy - Purple Oranges <[EMAIL PROTECTED]> wrote:
>> I am using VoiceOne http://voiceone.it/ as my management interface.
>>
>> I am not 100% sure when it started, but my CDR is now full of "s" as
>> the DST instead of the actual dialed number.
>>
>> As I understand it - it is because it is being recorded in the CDR
>> while in a macro (as below).
>>
>> Is there any work around so that I can record the actual dialed number?
>>
>> [macro-dialout]
>> exten = s,1,Set(TOUCH_MONITOR=${TIMESTAMP}_${CALLERID(num)}-${ARG1})
>> exten = s,n,NoOp(CID_NAME  : ${CID_NAME})
>> exten = s,n,NoOp(CID_NUMBER: ${CID_NUMBER})
>> exten = s,n,NoOp(CID_CLIR  : ${CID_CLIR})
>> exten = s,n,NoOp(TRUNK : ${TRUNK})
>> exten = s,n,Set(CALLERID(name)=${CID_NAME})
>> exten = s,n,Set(CALLERID(num)=${CID_NUMBER})
>> exten = 
>> s,n,Set(PRESENTATION=${IF($["${CID_CLIR}"="1"]?prohib_not_screened:allowed_not_screened)})
>> exten = s,n,SetCallerPres(${PRESENTATION})
>> exten = s,n,GotoIf(${ISNULL(${TRUNK})}?s-CONGESTION,1)
>> exten = s,n,Dial(${TRUNK}/${ARG1}${TRUNKOPTIONS}||gTW)   ;Ring the interface
>> exten = s,n,NoOp(DIALSTATUS = ${DIALSTATUS})
>> exten = s,n,Goto(s-${DIALSTATUS},1)  ;Jump based on status
>> (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
>> exten = s-BUSY,1,Playtones(busy)
>> exten = s-CONGESTION,1,Playtones(congestion)
>> exten = _s-.,1,Goto(s-CONGESTION,1)  ;Treat anything else as no answer
>>
>> --
>> Regards,
>> Troy Kelly
>> Director
>> Purple Oranges Pty Ltd
>> http://purpleoranges.com/
>> --
>> Brisbane (07) 3018 2840
>> Fax (07)  3105 5987
>> 
>> Disclaimer - This email and any files transmitted with it are
>> confidential and contain privileged or copyright information. You must
>> not present this message to another party without gaining permission
>> from the sender. If you are not the intended recipient you must not
>> copy, distribute or use this email or the information contained in it
>> for any purpose other than to notify us.
>>
>> Any views expressed in this message are those of the individual
>> sender, except where the sender specifically states them to be the
>> views of Purple Oranges Pty Ltd.
>>
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Re: [asterisk-users] Inexpensive Layer 3 Switch?

[David Gomillion]

On 6/26/07, Marty Mastera <[EMAIL PROTECTED]> wrote:


Sorry, I forgot to mention that I want to route between VLANs without an
external router and do some simple ACLs to allow PCs on the data VLANs to
access the web interface of the Trixbox on the voice VLAN.

thanks



The only reason to route the voice VLAN is if you need the phones to access
the Internet and/or vice-versa. If you only need to worry about the
computers on the data VLAN accessing Trixbox's web interface, I would
suggest using the Ethernet VLAN capabilities of Linux. You can create
eth0.vlan1 for data on Trixbox, and have the "default" vlan for the port on
the switch be voice. Then, the voice VLAN goes nowhere but to your PBX and
the phones.

The other option is to put in another NIC, one for the voice VLAN, the other
for the data VLAN.

I've been pretty happy with the Linksys 24-port layer 2 switches (SRW224P).
They're running around $400 right now. If you really need layer3 support, I
would steer clear of the Netgear. I've had a lot of problems with them, and
the support was disappointing. But then again, I got a bunch that don't work
that I could sell you ;)
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Re: [asterisk-users] Bridging two PSTN calls

[Don Pobanz]

> Doug Zingel wrote on Tuesday, June 26, 2007 4:39 AM
>
> I'm wondering if its possible to receive a call from
> an external number (PSTN) say A. Then make a call to
> another external number (PSTN) say B - and then bridge
> the two calls so that A is talking to B? 

Yes, look at blind transfer or attended transfer. 

> What hardware will I need to be able to do this.

The same hardware as to set up a system to take calls. The hardware will
depend on what interface you will be using to the outside world. IP
(SIP)? POTS? DID/DOD trunks over T1? 

> Secondly, if I had x number of simultaneous calls (A
> talking to B) - how many PSTN lines would I need? I
> think 2x.

That is correct. 

Don Pobanz

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Re: [asterisk-users] AstPligg

[lenz]

Niothing, it just serves a different purpouse. A site like voip-info,  
astrecipes, asterisk book wiki is for adding content; a site like digg is  
more for "pointing out" things you find on the various sites and to share  
them with other people.
l.

In data Tue, 26 Jun 2007 15:12:38 +0200, Jon Weisman <[EMAIL PROTECTED]>  
ha scritto:

> Whats wrong w/ voip-info.org?
>
>
> Jon Weisman | Sales Engineer
> International Bell Communications
> www.ibell.net
>
>   - Original Message -
>   From: Alex Robar
>   To: Asterisk Users Mailing List - Non-Commercial Discussion
>   Sent: Tuesday, June 26, 2007 8:38 AM
>   Subject: Re: [asterisk-users] AstPligg
>
>
>   Give the new site a break. I think it's a good idea. Sure there are  
> lots of news sites for VoIP, but many of them are poorly designed, and I  
> can't recall any that are very good at letting the users provide the  
> news content. I agree that the name could be better, but after having  
> just tried it out, I really like AstPligg.
>
>   AR
>
>
>   On 6/25/07, Mark Phillips <[EMAIL PROTECTED]> wrote:
> Great! Another one. With such a catchy name too!
>
> On Tue, 2007-06-26 at 01:42 +0200, lenz wrote:
> > Hello list,
> > AstPligg is a new Digg-like website devoted to * and VoIP news.
> >
> > At the moment, it's in beta stage and very basic - no fancy custom
> > templates. It allows posting new stories, comments on stories, RSS  
> feeds
> > and tags. Still, it can be very useful, as the number of * sites  
> and blogs
> > grows every day, and keeping track of what is hot in the * world is
> > increasingly difficult. Yes, I know, it's not much; but at least  
> it's
> > there and can be used immediately.
> >
> > You can find it at http://oinko.net/astpligg
> >
> > I'm looking forward to your comments (and stories) to make it a  
> useful
> > tool for the * community!
> > l.
> >
>
>
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>
>
>
>   --
>   Alex Robar
>   [EMAIL PROTECTED]
>
>
> --
>
>
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[asterisk-users] More FAX over T1

[Joe acquisto]

This is a follow up to an earlier post.

Looking for a means to "individualize" incoming FAX, so as to distribute them 
to the intended recipient.

While the PBX is based on Asterisk, it is not possible for me to "enter" the 
box to modify things, to any great degree.  I thank those who mentioned 
IAXMODEM, earlier, but that seems a no go.

Currently, there is a dedicated T1 into the Asterisk box.  There is a separate 
bank of 4 POTS lines going into a FAX server.

Looking for a way to assign numbers as "incoming FAX" lines and have them 
"received" with the incoming number intact.   Having these forwarded to one of 
the analog numbers is a thought, but I am concerned about various issues, data 
corruption, etc, going that route.

Thoughts vary to second T1, with channel bank, "breaking out" some DS0's into a 
channel bank, or finding a T1/fax board (do they exist?), to go directly into 
the FAX server (PC/linux based)

joe a.



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Re: [asterisk-users] Inexpensive Layer 3 Switch?

[Marty Mastera]

Sorry, I forgot to mention that I want to route between VLANs without an 
external router and do some simple ACLs to allow PCs on the data VLANs to 
access the web interface of the Trixbox on the voice VLAN.

thanks

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Jones
Sent: Tuesday, June 26, 2007 8:21 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Inexpensive Layer 3 Switch?

You do not need an L3 switch for this, just any managed switch which  
does vlans
Unless there is something else?

On Jun 26, 2007, at 12:07 AM, Marty Mastera wrote:

> Any recommendations on an economical layer 3 switch?  Preferably  
> something that you have hands on experience with connecting to IP  
> phones with attached PCs? Specifically I need the ability to set  
> the VLAN in the phone to tag voice packets and to set a native VLAN  
> on a per port basis on the switch to put the untagged packets from  
> the attached PC into a separate VLAN.
>
>
> POE is not a requirement but if you have suggestions for an  
> economical layer 3 switch with POE I’d be glad to hear them…so far  
> I’m looking at the SFE2000 from Linksys.
>
>
> thanks
>
>
> No virus found in this outgoing message.
> Checked by AVG Free Edition.
> Version: 7.5.476 / Virus Database: 269.9.7/868 - Release Date:  
> 6/25/2007 12:20 PM
>
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No virus found in this incoming message.
Checked by AVG Free Edition. 
Version: 7.5.476 / Virus Database: 269.9.7/868 - Release Date: 6/25/2007 12:20 
PM
 

No virus found in this outgoing message.
Checked by AVG Free Edition. 
Version: 7.5.476 / Virus Database: 269.9.9/870 - Release Date: 6/26/2007 10:07 
AM
 

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Re: [asterisk-users] Transfer Call to Cell Phone

[John Faubion]

>Polycom Phones
>1. New call
>2. Press 9 access outside line
>3. Dial Cell Number
>4. Transfer the call that way.

Once you initiate a new call you will tie up the second line. Your asterisk
box will now be bridging the two lines. The lines will stay tied up until
the salesman drops the call.

One method you might be able to employ here would be to add a call transfer
to the pots lines. Then you would need to send a hook flash to the pots
line, and dial the salesman's number when you get the dial tone. Then,
depending on how your local Telco supports the call transfer feature, you
may be able to free up the line. Not all Telcos support this the same way as
some consider it a method of toll avoidance and thus drop the call. This
would be possible in an area where a call from party A to party B is a local
call and the call from party B to party C is a local call but a call from
party A to party C would be a toll call. Since the call from party A to
party C is a toll call, the Telco may opt to drop the call. If the transfer
part works, there may even be a way to setup the dial plan to intercept your
phones call transfer feature and use a 1-2 digit code to select which phone
number to send out. I have not done this but I think it is reasonable as
I've heard of home users doing it.

By the second option, are you talking about the TDMA/GSM gateway? If so, yes
this is pretty slick. We considered it initially as well. Our decision not
to use it was based on the fact that many of our agents are on different
mobile plans. I think when we requested the info from the agents we had 6
different wireless companies represented. Since Sprint/Nextel,
Cingular/AT&T, T-Mobile don't like to play nicely with other, we didn't
expect to see any real savings from the free mobile to mobile calls. This
was mainly due to the fact that we don't pay for the agents phones and thus
we can't really tell the agents which carriers to use. I do know of a couple
of installations where the company does provide the phones and I understand
the savings can be significant. I was told by friend that the box they
installed paid for itself in just a couple of months. But their phone were
already on the same plan.

John


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Re: [asterisk-users] chan_features.so / asterisk 1.4.5

[Jack]

Joshua Colp schrieb:
> Jack wrote:
>   
>> Thanks for your answer and sorry for my late response.
>>
>> So what does this exactly mean to me? Can I keep chan_features.so from 
>> 1.4.4? What consequences does it have when chan_features.so is disabled 
>> und why has this been done? Is chan_features.so related to features.conf?
>>
>> 
>
> chan_features.so doesn't provide anything useful, it's not used anywhere 
> because it's not really finished. As for why it was changed to be 
> disabled by default oej probably thought that since we aren't using it 
> why do we have it enabled by default.
>
> It's fine to live in a world without chan_features.so :)
>
>   
Thank you very much.

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Re: [asterisk-users] DTMF doesn't work between Asterisk and Cisco SIP Proxy

[tracinet]

Jason,
I am at least having similar issues with rfc2833 DTMF:

http://bugs.digium.com/view.php?id=10058


On 6/20/07, Jason Ma <[EMAIL PROTECTED]> wrote:


Hi buddies,
I encountered DTMF issue when I tried to place call from x-lite to a
sip conference serice,here is the diagram.
X-lite>Asterisk--->Cisco SIP proxy>SIP Conference service

The Call can be established,and I can hear from x-lite the prompt of
the conference,but when I input any digits,nothing happened,the
conference service did not recognize my input.At the same time,in the
log of asterisk,I can find that asterisk recognized all the
digitsI tried "rfc2833","inband","info" in the "dtmfmode"
parameter,but did not work ,I'm not sure whether asterisk send the
right dtmf to cisco proxy,how can I track that?

I made another test,dialing from x-lite registered with Cisco proxy to
voicemail service of Asterisk.
x-lite>Cisco SIP proxy>Asterisk--->Voicemail service

Both the call and dtmf worked fine,I can input my mailbox number and
password and listen my  voicemail.both "rfc2933" and "inband" worked
in this situation,but not "info".

My Asterisk is 1.4.4 with asterisk now,I did not configure dtmfmode in
the section of  xlite and the trunk to cisco proxy,just configure the
dtmfmode in sip.conf.

When I used "rfc2833",I can see the log in asterisk as :

[2007-06-19 16:01:40] DTMF[8925] channel.c: DTMF begin '2' received on
SIP/-08269470
[2007-06-19 16:01:41] DTMF[8925] channel.c: DTMF end '2' received on
SIP/-08269470, duration 160 ms
[2007-06-19 16:01:42] DTMF[8925] channel.c: DTMF begin '1' received on
SIP/-08269470
[2007-06-19 16:01:42] DTMF[8925] channel.c: DTMF end '1' received on
SIP/-08269470, duration 140 ms

and when I used "inband",I can see :

[2007-06-19 15:55:21] DTMF[8852] channel.c: DTMF end '2' received on
SIP/-09d916c0, duration 0 ms
[2007-06-19 15:55:22] DTMF[8852] channel.c: DTMF end '1' received on
SIP/-09d916c0, duration 0 ms

Is that right?Can I check what digits that asterisk sent out ?

How can I track where is wrong with the dtmf?Did asterisk send dtmf to
Cisco proxy correctly?
I really have no idea about that.Please advise.Thank you very much

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Re: [asterisk-users] Fax Throughput

[Matthew Fredrickson]

Can you post your zaptel.conf so we can verify your timing settings?

---
Matthew Fredrickson
Software Engineer
Digium, Inc.

On Jun 25, 2007, at 11:10 PM, Don Kelly wrote:

> I've tried timing faxes two ways:
>
> From a fax machine on a station port of an AltiGen PC/PBX served by  
> an MCI
> PRI calling back into the same PRI and reaching a RightFax server on a
> station port behind the AltiGen.
>
> From the same fax machine on the same station port of the AltiGen  
> PC/PBX
> served by the same MCI PRI calling a number on an XO PRI connected  
> to an
> Asterisk system (Digium TE410P), dialing out on another channel on  
> the same
> PRI back into the MCI PRI and reaching the RightFax server on the  
> station
> port behind the AltiGen.
>
> extensions.conf includes:
> exten => 6122353002,1,dial(zap/g1/6122590773)
>
> Sending a one-page fax with moderate density (no graphics) takes  
> almost five
> minutes longer going through the Asterisk server.
>
> Any suggestions?
>
>   --Don
>
> Don Kelly
> PCF Corp
> Real Support for your Virtual Office
> 651 842-1000
> 888 Don Kell(y)
> 651 842-1001 fax
>
>
>
>
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[asterisk-users] rcf2833 DTMF broken in asterisk SIP channel?

[tracinet]

I posted this bug yesterday:

http://bugs.digium.com/view.php?id=10058

but really was hoping that one of you would be willing to try something
simple for me and reply back with your results.

Basically - I have run into a problem where Asterisk RFC2833 DTMF does not
seem to be compatible with large SIP providers such as Level 3 and Global
Crossing.  Can someone who is using rfc2833 DTMF with a non-asterisk SIP
provider try inserting this in their dial plan to see if it works (trying to
see if the DTMF tones are heard on the PSTN side of the equation).  What
happens to me is the first digit gets heard but then silence as the next 8
digits are sent.

extensions.conf:
exten => 55,1,Dial(SIP/sip_provider/55,20,D(123456789))

For your info - here is what I have in my sip.conf:

[general]
disallow = all
allow=ulaw
port = 5060
context = incoming
maxexpirey=180
defaultexpirey=160
canreinvite=no
srvlookup=yes
videosupport=no
nat=no
tos=reliability
dtmfmode=rfc2833

[sip_provider]
type=friend
username=123456789
secret=password
host=10.0.0.1
disallow=all
allow=ulaw
maxexpirey=15
relaxdtmf=yes
dtmfmode=rfc2833
nat=no
insecure=very
canreinvite=no
promiscredir=yes

Thanks in advance for your help!
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Re: [asterisk-users] Inexpensive Layer 3 Switch?

[Jerry Jones]

You do not need an L3 switch for this, just any managed switch which  
does vlans
Unless there is something else?

On Jun 26, 2007, at 12:07 AM, Marty Mastera wrote:

> Any recommendations on an economical layer 3 switch?  Preferably  
> something that you have hands on experience with connecting to IP  
> phones with attached PCs? Specifically I need the ability to set  
> the VLAN in the phone to tag voice packets and to set a native VLAN  
> on a per port basis on the switch to put the untagged packets from  
> the attached PC into a separate VLAN.
>
>
> POE is not a requirement but if you have suggestions for an  
> economical layer 3 switch with POE I’d be glad to hear them…so far  
> I’m looking at the SFE2000 from Linksys.
>
>
> thanks
>
>
> No virus found in this outgoing message.
> Checked by AVG Free Edition.
> Version: 7.5.476 / Virus Database: 269.9.7/868 - Release Date:  
> 6/25/2007 12:20 PM
>
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[asterisk-users] Multi port IAX Gateway

[Mike Hammett]

I am looking for a gateway that has several FXS ports and uses IAX.  I have
a need for 16 ports, but will accept 6 or 8 port gateways as well. 

 

 

 

-
Mike Hammett
Intelligent Computing Solutions
  http://www.ics-il.com

 

 

 

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Re: [asterisk-users] AstPligg

[Alex Robar]

I don't think anything is _wrong_ with VoIP-Info at all, I just think the
sites serve different purposes. This is all just personal preference, but to
me VoIP-Info does not work that well as a "social" news site, as all
stories/headlines, good or bad, have equal weight. With the Pligg system,
the stories that are better are usually voted up so they have higher
exposure. VoIP-Info is a great site for sharing Asterisk recipes and HowTos,
but I'm not a fan of it as a news site.

AR

On 6/26/07, Jon Weisman <[EMAIL PROTECTED]> wrote:


 Whats wrong w/ voip-info.org?


Jon Weisman | Sales Engineer
International Bell Communications
www.ibell.net

- Original Message -
*From:* Alex Robar <[EMAIL PROTECTED]>
*To:* Asterisk Users Mailing List - Non-Commercial 
Discussion
*Sent:* Tuesday, June 26, 2007 8:38 AM
*Subject:* Re: [asterisk-users] AstPligg

Give the new site a break. I think it's a good idea. Sure there are lots
of news sites for VoIP, but many of them are poorly designed, and I can't
recall any that are very good at letting the users provide the news content.
I agree that the name could be better, but after having just tried it out, I
really like AstPligg.

AR

On 6/25/07, Mark Phillips <[EMAIL PROTECTED]> wrote:
>
> Great! Another one. With such a catchy name too!
>




--
Alex Robar
[EMAIL PROTECTED]
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Re: [asterisk-users] Transfer Call to Cell Phone

[OCOSA ListAcct]

John Faubion wrote:
>> by the way selling does not depend on the amount of lines you have and
>> we are very productive trust me
>> 
>
>
> True, very true. There are lots of very productive sales people that don't
> need a phone at all. From the paper boy to car dealers, lots of sales don't
> require many phone lines. Of course at the same time, a typical call center
> wouldn't be very productive with only two lines.
>
>
>   
>> I have seen a million dollar corp work off four lines so your statement
>> is quite vague...
>> 
>
>
> We have a few agents that have million dollar months and even a couple that
> have had million dollar weeks! But that isn't the point, is it Otis? The
> problem is you got your feelings hurt because instead of reading my reply,
> you assumed that I was putting your company down. My first paragraph was
> kind of a open thought process so that you and others might comprehend the
> basis of my reply. What I was trying to wrap my head around, was just how
> productive a system with only two lines could be if a single call came in
> and was then routed back out the other line to the outside sales guy. Now if
> you were using a digital line, perhaps we could consider using the signaling
> to redirect the call from the originating source directly to the salesman's
> phone and thus free up the lines for the next call. But no, you said pots
> lines as in "Plain Old Telephone Service" (POTS) which means we don't have
> the option of using some fancy, out of band signaling to redirect the call.
> So my thinking was, as I said before, "Surely you have more than two lines."
> In my twenty-two years of telephony experience, dealing with everything from
> single line phones to key systems to PBX systems to Nortel DMS-500 switches,
> I only remember one sales "office" that only had two lines and that office
> was literally an 8 foot by 8 foot closet with two phones and all calls were
> outgoing.
>   

You are right but finish reading thisTo be honest I did not get my 
feelings hurt, so assumptions are not needed I was simply stating one 
scenario where a local company here was very product to better your 
understanding of how some companies work off 2 to 10 lines and still 
produce. If you have read my first statement you would have understood 
that I did read and I did appreciate your reply. As the other methods 
has no interest to me at this time. I do agree maybe I should have sent 
a paragraph with details but I felt like knowing only two lines for the 
sales office was plenty. Now do not confuse us with a call center. Trust 
me by no means could we be as productive as others but then again we are 
not a giant but a small business and do not sign up thousands a day so 
we do not have a need for more lines yet. But there again you still made 
a judgment call about the two lines. If I tell this is the setup then 
there is nothing to question. Sometimes all tech guys have a problem 
with assuming sometimes there are more to a situation than what was 
presented. I am guilty as well. LOL!!
> Yes, my answer was a little vague. So was the information you provided. Now
> had you bothered to read the 2nd and 3rd paragraph, you might have noticed
> that I provided a few methods that you could consider. My intention for
> doing this was simple. Maybe one of the ways mentioned would spark a
> response from you that would help to clarify the "right" way. Now suppose
> for a moment that you had actually read the reply. Let's also pretend that
> in reading it you realized that, yes, you have two pots lines, but what you
> had meant to say was that you had two "unused" pots lines along with some
> other form of incoming trunks. Then maybe you would have responded with an
> email to clarify that, to which I could have suggested that maybe you could
> look into a two port cell phone gateway to keep the incoming lines free and
> still keep connected to your sales guy. Can you see how we could have used
> that information to consider the "right" option?
>   

Here again had you read my first statement you would have understood 
that I appreciated your reply.

John

thanks for the input.

forget about my "right" way ok!

I made a mistake on putting this in...this is what I was really 
looking for: explained later down..

> Considering that this list is for non commercial discussion, our only form
> of payment here is in the repayment of our debt to others that have gone
> before us and helped us out. Next time please appreciate the fact that
> someone else took time out of their busy day to consider and to reply your
> request for information. Now if you would like to provide a little more
> detail with your request, I'm fairly sure that someone here will likely
> respond to it.
>
>   

There again making assumptions are not right because I did try your 
first option. But John do not get me wrong as I have been thinking about 
the second so before you can say please appreciate th

Re: [asterisk-users] AstPligg

[Jon Weisman]

Whats wrong w/ voip-info.org? 


Jon Weisman | Sales Engineer
International Bell Communications
www.ibell.net

  - Original Message - 
  From: Alex Robar 
  To: Asterisk Users Mailing List - Non-Commercial Discussion 
  Sent: Tuesday, June 26, 2007 8:38 AM
  Subject: Re: [asterisk-users] AstPligg


  Give the new site a break. I think it's a good idea. Sure there are lots of 
news sites for VoIP, but many of them are poorly designed, and I can't recall 
any that are very good at letting the users provide the news content. I agree 
that the name could be better, but after having just tried it out, I really 
like AstPligg. 

  AR


  On 6/25/07, Mark Phillips <[EMAIL PROTECTED]> wrote:
Great! Another one. With such a catchy name too!

On Tue, 2007-06-26 at 01:42 +0200, lenz wrote:
> Hello list,
> AstPligg is a new Digg-like website devoted to * and VoIP news.
>
> At the moment, it's in beta stage and very basic - no fancy custom 
> templates. It allows posting new stories, comments on stories, RSS feeds
> and tags. Still, it can be very useful, as the number of * sites and blogs
> grows every day, and keeping track of what is hot in the * world is 
> increasingly difficult. Yes, I know, it's not much; but at least it's
> there and can be used immediately.
>
> You can find it at http://oinko.net/astpligg 
>
> I'm looking forward to your comments (and stories) to make it a useful
> tool for the * community!
> l.
>


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  -- 
  Alex Robar
  [EMAIL PROTECTED] 


--


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Re: [asterisk-users] chan_features.so / asterisk 1.4.5

[Joshua Colp]

Jack wrote:
> Thanks for your answer and sorry for my late response.
> 
> So what does this exactly mean to me? Can I keep chan_features.so from 
> 1.4.4? What consequences does it have when chan_features.so is disabled 
> und why has this been done? Is chan_features.so related to features.conf?
> 

chan_features.so doesn't provide anything useful, it's not used anywhere 
because it's not really finished. As for why it was changed to be 
disabled by default oej probably thought that since we aren't using it 
why do we have it enabled by default.

It's fine to live in a world without chan_features.so :)

-- 
Joshua Colp
Software Developer
Digium, Inc. - "The Genuine Asterisk Experience" (TM)


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[asterisk-users] No CID on Zaps - TDM400

[Eric Estes]

I'm running Trixbox 1.2.3 with 2 TDM400s (FXOs).

With Trixbox out of the mix and a regular phone connected I get the CID
fine yet Trixbox shows 'unknown':

dialparties.agi: Caller ID name is 'unknown' number is 'unknown'
dialparties.agi: Methodology of ring is  'ringall'

Here is my Zapata.conf if it helps:

#
;
; Zapata telephony interface
;
; Configuration file

[trunkgroups]

[channels]

language=en
context=from-zaptel
signalling=fxs_ks
rxwink=300  ; Atlas seems to use long (250ms) winks
;
; Whether or not to do distinctive ring detection on FXO lines
;
;usedistinctiveringdetection=yes

usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=no
echotraining=800
rxgain=0.0
txgain=0.0
group=0
callgroup=1
pickupgroup=1
immediate=no

;faxdetect=both
faxdetect=incoming
;faxdetect=outgoing
;faxdetect=no

;Include genzaptelconf configs
#include zapata-auto.conf

group=1

;Include AMP configs
#include zapata_additional.conf


#

Thank You,

Eric

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Re: [asterisk-users] Best wifi IP phone for asterisk (LINKSYS SUPPORT QUALITY)

[Michelle Dupuis]

I haven't been too impressed with the WIP330 - but my experience with
Linksys tech support has been disastrous!

I spent approximately 50 minutes on hold, I was transferred between 4
different people (all of whom had a poor grasp of the English language),
none of them understood the features of the phone.  All they could do was
read the user manual I already had in front of me.  None of them could
explain the "auto-provisioning" feature mentioned on the Linksys website, or
on the phone menus, etc.  None of them even understood what SIP and RTP
protocols were.  They were just there to read the manual to me.

Here's a warning for the group...watch our for Linksys!

-MD-

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nick Seraphin
Sent: Tuesday, June 26, 2007 4:58 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Best wifi IP phone for asterisk


On Tue, 26 Jun 2007, Hendrik Visage wrote:

> On 6/25/07, Michelle Dupuis <[EMAIL PROTECTED]> wrote:
> >
> >
> > We're looking at a large wifi phone deployment, and we're looking 
> > for wifi phones that:
>
> HAve a look at the Linksys WIP 300 (or something) Can be charged from 
> the USB port


I have a WIP330 and it doesn't work.  Maybe it needs a firmware upgrade.
Maybe it's a defective unit and all the others work fine.  I haven't called
support yet because I haven't had the chance.

Audio in one direction cuts out completely for about 4 seconds every 10
seconds during the call.  For 10 seconds it works fine... audio in both
directions... then 4 seconds of silence in one direction... then 10 seconds
of normal, etc etc, repeating forever.

Completely unusable as-is.

All my other Linksys IP Phones work great, though.  I only have one WIP330.
I don't have any WIP300's.

My recommendation:  Whatever you go with, buy ONE first for testing to make
sure you're happy with it BEFORE you buy a boatload of them.

-- Nick




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Re: [asterisk-users] AstPligg

[Alex Robar]

Give the new site a break. I think it's a good idea. Sure there are lots of
news sites for VoIP, but many of them are poorly designed, and I can't
recall any that are very good at letting the users provide the news content.
I agree that the name could be better, but after having just tried it out, I
really like AstPligg.

AR

On 6/25/07, Mark Phillips <[EMAIL PROTECTED]> wrote:


Great! Another one. With such a catchy name too!

On Tue, 2007-06-26 at 01:42 +0200, lenz wrote:
> Hello list,
> AstPligg is a new Digg-like website devoted to * and VoIP news.
>
> At the moment, it's in beta stage and very basic - no fancy custom
> templates. It allows posting new stories, comments on stories, RSS feeds
> and tags. Still, it can be very useful, as the number of * sites and
blogs
> grows every day, and keeping track of what is hot in the * world is
> increasingly difficult. Yes, I know, it's not much; but at least it's
> there and can be used immediately.
>
> You can find it at http://oinko.net/astpligg
>
> I'm looking forward to your comments (and stories) to make it a useful
> tool for the * community!
> l.
>


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--
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[EMAIL PROTECTED]
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Re: [asterisk-users] chan_features.so / asterisk 1.4.5

[Jack]

Carsten Bock schrieb:
> José Luis Ledesma schrieb:
>   
>> In my asterisk 1.4.5 chan_features.so has been installed properly... 
>> check in your asterisk-source if /channels/chan_features.so is present
>>
>>   regards,
>>
>> Jack escribió:
>> 
>
>   
>>> Is chan_features.so deprecated for asterisk 1.4.5 or why is this
>>> module not installed by asterisk 1.4.5?
>>>
>>>   
> See the 1.4.5 Changelog:
> (http://ftp.digium.com/pub/asterisk/ChangeLog-1.4.5)
>
> 2007-06-07 19:46 + [r68196]  Olle Johansson <[EMAIL PROTECTED]>
>
>   * channels/chan_features.c: Disable chan_features by default in
> menuselect
>
>
>
>   
Thanks for your answer and sorry for my late response.

So what does this exactly mean to me? Can I keep chan_features.so from 
1.4.4? What consequences does it have when chan_features.so is disabled 
und why has this been done? Is chan_features.so related to features.conf?

Regards, Jens
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>   


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Re: [asterisk-users] Fax Throughput

[Lee Howard]

Don Kelly wrote:

>I've tried timing faxes two ways:
>
>>From a fax machine on a station port of an AltiGen PC/PBX served by an MCI
>PRI calling back into the same PRI and reaching a RightFax server on a
>station port behind the AltiGen.
>
>>From the same fax machine on the same station port of the AltiGen PC/PBX
>served by the same MCI PRI calling a number on an XO PRI connected to an
>Asterisk system (Digium TE410P), dialing out on another channel on the same
>PRI back into the MCI PRI and reaching the RightFax server on the station
>port behind the AltiGen.
>
>extensions.conf includes:
>exten => 6122353002,1,dial(zap/g1/6122590773)
>
>Sending a one-page fax with moderate density (no graphics) takes almost five
>minutes longer going through the Asterisk server.
>

The longer transmit time is probably a result either (or both)...

1) retransmissions due to the audio being consistently corrupted, and 
ECM retransmissions to correct the corruption

2) training failure (probably due to corrupt audio) resulting in a 
slower transmission rate (e.g. 9600 bps vs 14400 bps)

As to how to fix it... it's almost certainly audio degredation occurring 
in your Asterisk configuration or linkage... so debug your Asterisk setup.

Lee.

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[asterisk-users] call fail from audiocode to sip trunk

[satish patel]

Dear ALL

  I have audiocode MP -124 with configure in asterisk Endpoint 
configuration means every analog phone register in asterisk now thing is that i 
have one more SIP trunk with mediant 2000 

[auodiocode-mp-124]-[ * ]--[mediant 2000]-E1


When i call from audiocode MP -124 phone i got this error 

   -- Executing Dial("SIP/20-0889c4d8", "SIP/mediant/1") in new stack
-- Called mediant/1
-- SIP/mediant-088a1a18 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
-- Executing Congestion("SIP/20-0889c4d8", "") in new stack
  == Spawn extension (mysip, 111, 2) exited non-zero on 'SIP/20-0889c4d8'
-- Executing Dial("SIP/24-0889c4d8", "SIP/mediant/0") in new stack
-- Called mediant/0

my extension.conf file is 

exten => 43,1,Answer
exten => 43,2,Dial(SIP/43)
exten => 43,3,Hangup
exten => 777,1,Answer()
exten => 777,2,Dial(SIP/777)
exten => 777,3,Hangup()
exten => 888,1,Answer()
exten => 888,2,Dial(SIP/888)
exten => 55,1,Dial(SIP/55)
exten => 66,1,Dial(SIP/66)

exten => _11.,1,Dial(SIP/mediant/${EXTEN:2})
exten => _11.,2,Congestion

what is the problem

 
-
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[asterisk-users] Digium TE120P setup problem [was: Re: Query]

[Tzafrir Cohen]

Setting subject to a more descriptive one.

On Tue, Jun 26, 2007 at 03:35:12PM +0530, [EMAIL PROTECTED] wrote:
> Hi,
>  I have put Digium TE120P card in PCI slot.  So, lspci command gives the 
> information in followimg format.
> 02:0a.0 Ethernet controller: Unknown device d161:0120 
> (rev 11)
>   Following modules are running when seen through lspci

lsmod, right?

>  wcte11xp   22304  -
>  ztdynamic   9804  -
>  ztdummy 3468  -
>  ip_conntrack_irc6640  -
>  ip_conntrack_ftp7312  -
>  ipt_state   1864  - 
>  iptable_mangle  2696  -
>  ipt_REJECT  5160  -
>  ipt_LOG 6280  -
> ipt_multiport   2376  -
> ip_conntrack   47524  -
> iptable_filter  2856  -
>ipt_limit   2280  -
> ip_tables  18168  - 
>wcte12xp   44352  -

This one is probably the one you actually need.

>zaptel180036  -
> 
>   but on running asterisk -vvvgc it stops by printing the following errrors
> 
> '###' 
> at line 41 of /etc/asterisk/zapata.conf

Comments in asterisk config files begin with a ';', not a '#'.
(/etc/zaptel.conf is not an asterisk config file and comments there do
begin with '#').

> Jun 26 15:19:38 WARNING[1691]: config.c:497 process_text_line: Unknown 
> directive 
> '###' 
> at line 43 of /etc/asterisk/zapata.conf
> Jun 26 15:19:38 NOTICE[1691]: chan_zap.c:10388 setup_zap: Ignoring unknown
> keyword 'group' in trunkgroups

Should this be in [channels]?

Anyway,

> Jun 26 15:19:38 WARNING[1691]: chan_zap.c:1072 zt_open: Unable to specify 
> channel 1: No such device or address
> Jun 26 15:19:38 ERROR[1691]: chan_zap.c:7050 mkintf: Unable to open channel 
> 1: No such device or address
> here = 0, tmp->channel = 1, channel = 1
> Jun 26 15:19:38 ERROR[1691]: chan_zap.c:10484 setup_zap: Unable to register 
> channel '1-31'
> Jun 26 15:19:38 WARNING[1691]: loader.c:415 __load_resource: chan_zap.so: 
> load_module failed, returning -1
> Jun 26 15:19:38 WARNING[1691]: loader.c:555 load_modules: Loading module 
> chan_zap.so failed!
>  What is the problem actually can anybody tell me.

What is the output of:

  cat /proc/zaptel/*

What do you have in /etc/zaptel.conf ? In /etc/asterisk/zapata.conf ?

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir

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[asterisk-users] Test Message

[Gary]

Sorry to clutter up the mailiing list, but I've been unable to post to this
list for the past 2 WEEKS!
My ISP's blocking SMPT from other than his own servers.
I think I've worked around it. - But if I see this message in the digest
then I know I'm okay.
Again. - Sorry for any inconvenience.
Gary Guthary



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[asterisk-users] Query

[sanchal . singh]

Hi,
 I have put Digium TE120P card in PCI slot.  So, lspci command gives the 
information in followimg format.
02:0a.0 Ethernet controller: Unknown device d161:0120 (rev 
11)
  Following modules are running when seen through lspci
 wcte11xp   22304  -
 ztdynamic   9804  -
 ztdummy 3468  -
 ip_conntrack_irc6640  -
 ip_conntrack_ftp7312  -
 ipt_state   1864  - 
 iptable_mangle  2696  -
 ipt_REJECT  5160  -
 ipt_LOG 6280  -
ipt_multiport   2376  -
ip_conntrack   47524  -
iptable_filter  2856  -
   ipt_limit   2280  -
ip_tables  18168  - 
   wcte12xp   44352  -
   zaptel180036  -

  but on running asterisk -vvvgc it stops by printing the following errrors

'###' 
at line 41 of /etc/asterisk/zapata.conf
Jun 26 15:19:38 WARNING[1691]: config.c:497 process_text_line: Unknown 
directive 
'###' 
at line 43 of /etc/asterisk/zapata.conf
Jun 26 15:19:38 NOTICE[1691]: chan_zap.c:10388 setup_zap: Ignoring unknown
keyword 'group' in trunkgroups
Jun 26 15:19:38 WARNING[1691]: chan_zap.c:1072 zt_open: Unable to specify 
channel 1: No such device or address
Jun 26 15:19:38 ERROR[1691]: chan_zap.c:7050 mkintf: Unable to open channel 1: 
No such device or address
here = 0, tmp->channel = 1, channel = 1
Jun 26 15:19:38 ERROR[1691]: chan_zap.c:10484 setup_zap: Unable to register 
channel '1-31'
Jun 26 15:19:38 WARNING[1691]: loader.c:415 __load_resource: chan_zap.so: 
load_module failed, returning -1
Jun 26 15:19:38 WARNING[1691]: loader.c:555 load_modules: Loading module 
chan_zap.so failed!
 What is the problem actually can anybody tell me.

Thanx and regards
sanchal 

  


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[asterisk-users] Bridging two PSTN calls

[Doug Zingel]

Hi,
I'm a novice trying out an experiment with Asterisk
and was unsure of the hardware needs for it.

I'm wondering if its possible to receive a call from
an external number (PSTN) say A. Then make a call to
another external number (PSTN) say B - and then bridge
the two calls so that A is talking to B? What hardware
will I need to be able to do this.

Secondly, if I had x number of simultaneous calls (A
talking to B) - how many PSTN lines would I need? I
think 2x.

I know this sounds ridiculously inefficient - but I
think the experiment might open some avenues.

Thanks,
Doug


   

Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now (it's updated for 
today's economy) at Yahoo! Games.
http://get.games.yahoo.com/proddesc?gamekey=monopolyherenow  

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Re: [asterisk-users] asterisk 1.4.1 app_addon_sql_mysql

[didier]

It looks like you haven't install some mysql packages BEFORE make clean, make, 
make install.

just install:

libmysqlclient15-dev
mysql-client
mysql-server

D


  - Original Message - 
  From: Khaled Chehab 
  To: 'Asterisk Users Mailing List - Non-Commercial Discussion' 
  Cc: [EMAIL PROTECTED] 
  Sent: Thursday, June 21, 2007 7:24 AM
  Subject: [asterisk-users] asterisk 1.4.1 app_addon_sql_mysql


  when I enter asterisk-addons-1.4.1 and make menuselect 

  *

Asterisk-addons Module 
Selection

  
*

   

   Press 'h' for help.

   

  XXX 1.  
app_addon_sql_mysql

  [*] 2.  app_saycountpl

  XXX 3.  
cdr_addon_mysql

  [ ] 4.  chan_ooh323

  [*] 5.  format_mp3

  XXX 6.  
res_config_mysql

   

  Cannot install app_addon_sql_mysql ….

  Any dependencies required ?

   

   

  Regards

   

   

   




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[asterisk-users] VPN technology for snom 370

[Hirosh Dabui]

Hello,

as from now on the snom 370 a special firmware exist to build secure 
VoIP-Infrastructures via OpenVPN -Technology.
For further information go to http://snom.com/wiki/index.php/Networking/VPN

Note: *That is a pre-release, probably the software is still unstable*


cheers,

Hirosh Dabui

-- 
Hirosh Dabui
computer engineering
http://snom.com



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Re: [asterisk-users] asterisk-users Digest, Vol 35, Issue 92

[jr]

Greetings!

Due to high workload, I am currently checking and responding to e-mail twice 
daily at 12:00 PM EST and 9:00PM EST.

If you require urgent assistance (please ensure it is urgent) that cannot wait 
until either 12:00 PM or 9:00 PM, please contact me via phone at: 305-338-3867.

Thank you for understanding this move to more efficiency and effectiveness. It 
helps me accomplish more to serve you better.

Sincerely,

Harold Riley



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Re: [asterisk-users] Best wifi IP phone for asterisk

[Nick Seraphin]

On Tue, 26 Jun 2007, Hendrik Visage wrote:

> On 6/25/07, Michelle Dupuis <[EMAIL PROTECTED]> wrote:
> >
> >
> > We're looking at a large wifi phone deployment, and we're looking for wifi
> > phones that:
> 
> HAve a look at the Linksys WIP 300 (or something)
> Can be charged from the USB port


I have a WIP330 and it doesn't work.  Maybe it needs a firmware upgrade.
Maybe it's a defective unit and all the others work fine.  I haven't
called support yet because I haven't had the chance.

Audio in one direction cuts out completely for about 4 seconds every 10
seconds during the call.  For 10 seconds it works fine... audio in both
directions... then 4 seconds of silence in one direction... then 10
seconds of normal, etc etc, repeating forever.

Completely unusable as-is.

All my other Linksys IP Phones work great, though.  I only have one
WIP330.  I don't have any WIP300's.

My recommendation:  Whatever you go with, buy ONE first for testing to
make sure you're happy with it BEFORE you buy a boatload of them.

-- Nick




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[asterisk-users] realtime_extensions

[Pezhman Lali]

Hi 
now, I am using, realtime connection(mysql) for
dialplan,

but the following line must be added ,manualy to
extensions.conf, before reloading.for each new
context.

[NEW_CONTEXT]
switch => Realtime/@extensions

is there any idea, to add this line to dbase too?

thanks in advance
Best
MAni 


 

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Re: [asterisk-users] CDR Records "s" as dst

[Troy - Purple Oranges]

It can be "fixed" with the patch from

http://lists.digium.com/pipermail/asterisk-dev/2007-June/028093.html

Cheers, Troy

On 26/06/07, Jaswinder Singh <[EMAIL PROTECTED]> wrote:
> This is due to changes in cdr in asterisk 1.4.5 so in all outgoing
> dial from macro it shows 's' in cdr . Could this be a bug ? I doubt if
> it was intended to be that way .
>
> On 25/06/07, Troy - Purple Oranges <[EMAIL PROTECTED]> wrote:
> > I am using VoiceOne http://voiceone.it/ as my management interface.
> >
> > I am not 100% sure when it started, but my CDR is now full of "s" as
> > the DST instead of the actual dialed number.
> >
> > As I understand it - it is because it is being recorded in the CDR
> > while in a macro (as below).
> >
> > Is there any work around so that I can record the actual dialed number?
> >
> > [macro-dialout]
> > exten = s,1,Set(TOUCH_MONITOR=${TIMESTAMP}_${CALLERID(num)}-${ARG1})
> > exten = s,n,NoOp(CID_NAME  : ${CID_NAME})
> > exten = s,n,NoOp(CID_NUMBER: ${CID_NUMBER})
> > exten = s,n,NoOp(CID_CLIR  : ${CID_CLIR})
> > exten = s,n,NoOp(TRUNK : ${TRUNK})
> > exten = s,n,Set(CALLERID(name)=${CID_NAME})
> > exten = s,n,Set(CALLERID(num)=${CID_NUMBER})
> > exten = 
> > s,n,Set(PRESENTATION=${IF($["${CID_CLIR}"="1"]?prohib_not_screened:allowed_not_screened)})
> > exten = s,n,SetCallerPres(${PRESENTATION})
> > exten = s,n,GotoIf(${ISNULL(${TRUNK})}?s-CONGESTION,1)
> > exten = s,n,Dial(${TRUNK}/${ARG1}${TRUNKOPTIONS}||gTW)   ;Ring the interface
> > exten = s,n,NoOp(DIALSTATUS = ${DIALSTATUS})
> > exten = s,n,Goto(s-${DIALSTATUS},1)  ;Jump based on status
> > (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
> > exten = s-BUSY,1,Playtones(busy)
> > exten = s-CONGESTION,1,Playtones(congestion)
> > exten = _s-.,1,Goto(s-CONGESTION,1)  ;Treat anything else as no answer
> >
> > --
> > Regards,
> > Troy Kelly
> > Director
> > Purple Oranges Pty Ltd
> > http://purpleoranges.com/
> > --
> > Brisbane (07) 3018 2840
> > Fax (07)  3105 5987
> > 
> > Disclaimer - This email and any files transmitted with it are
> > confidential and contain privileged or copyright information. You must
> > not present this message to another party without gaining permission
> > from the sender. If you are not the intended recipient you must not
> > copy, distribute or use this email or the information contained in it
> > for any purpose other than to notify us.
> >
> > Any views expressed in this message are those of the individual
> > sender, except where the sender specifically states them to be the
> > views of Purple Oranges Pty Ltd.
> >
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[asterisk-users] Asterisk + Legacy PBX

[Marc Patino Gómez]

Hi all,

I have a isue with a Siemens Hicom conected to my asterisk, here is the 
scheme:

Telco    Asterisk  ---  Legacy PBX   --- Legacy phones

The asterisk box has a TE210 (one PRI conected to Telco another PRI 
conected to Siemens)

Everything works ok, but when I make an international call from legacy 
phones to the telco, for example: 0034934452740, the Siemens only sends 
to Asterisk the three first numbers 003.

Here is my config in extensions.conf:

[incoming-siemens]
exten => _X.,1,NoOP
exten => _X.,n,Dial(Zap/g2/${EXTEN})
exten => _X.,n,Hangup

[incoming-telco]
exten => _X.,1,NoOP
exten => _X.,n,Dial(Zap/g1/${EXTEN})
exten => _X.,n,Hangup


The other calls works great, incoming calls and outgoing calls. Any help 
will be very apreciated, I'm a newbie doing this kind of asterisk 
config, so any advice will be helpful.

Best regards,

Marc

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[asterisk-users] zaptel 1.2.18 and HPEC

[Paul Hales]

Any idea why I can't build HPEC for zaptel 1.2.18?

It builds fine with 1.4.3...

PaulH


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Re: [asterisk-users] Inexpensive Layer 3 Switch?

[John Faubion]

 > Any recommendations on an economical layer 3 switch?



I've been quite happy with the Netgear FS728TP ProSafe switches. These are
24 port 10/100 switches with 4 GigE ports and PoE. They also support 1000-SX
GBIC through an optional module. The total PoE budget for all 24 port is 195
watts. We run 38 GXP2000 phones on a pair of them. We have one with 19
phones and a DLink DWL-3200AP wireless access point drawing power and the
current load is under 75 watts average and has peaked at 97 watts. The other
one has 19 phones, two Ethernet cameras and draws even less. All for less
than $375 shipped. They even have a $25 rebate on them until the end of the
month. Plus they even have a Lifetime Warranty. One of the cool features I
discovered after installation was the built in Time Domain Reflectometer.
The TDR is great for testing out the cables right after installation. We
were able to use it to locate the screw that the drywallers intentionally
ran through one of our cables. It was so obvious that the contractor paid to
replace the cable. The only negative comment I have about them is the drone
of the fans. Our wall mount rack is in the break room and the switches are
easily the loudest item in the rack. They are not as loud as the Dell server
we originally bought so if your using a 1U server from Compaq, Dell, HP,
etc... you'll be fine.



John


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Re: [asterisk-users] Rining 180 and 183

[Andrew Joakimsen]

Replace with below. Actually Asterisk should only generate ringback when the
SIP phone is ringing.

On 6/25/07, satish patel <[EMAIL PROTECTED]> wrote:



exten => 222,1,Dial(SIP/222,r)
exten => 333,1,Dial(SIP/333,r)
exten => 555,1,Dial(SIP/555,r)
exten => 100,1,Dial(SIP/100,r)
exten => 112,1,Dial(SIP/112,r)
exten => 115,1,Dial(SIP/115,r)


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