[asterisk-users] chan_h323 build failure - `IPTOS_MINCOST' undeclared

[Bruce McAlister]

Hi All,

I am trying to build chan_h323 for use with asterisk 1.4.18 on Solaris 
10. When I compile asterisk, the build fails at chan_h323 with:

--
chan_h323.c: In function `reload_config':
chan_h323.c:2863: error: `IPTOS_MINCOST' undeclared (first use in this 
function)
chan_h323.c:2863: error: (Each undeclared identifier is reported only once
chan_h323.c:2863: error: for each function it appears in.)
gmake[1]: *** [chan_h323.o] Error 1
gmake: *** [channels] Error 2
--

I have downloaded PWLIB v1.10.0 and OpenH323 v1.18.0 and they are both 
built and installed properly. Has anyone come across this issue, or do I 
have to log a bug report at Digiums bug tracker?

Thanks
Bruce

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Re: [asterisk-users] Voted most stable and easy to use phone?

[randulo]

On Thu, Feb 21, 2008 at 7:32 PM, arkda <[EMAIL PROTECTED]> wrote:
> I'm a huge fan of the Linksys SPA-942s for users. They run around $125, are
> pretty straightforward to manage via TFTP, and work really well with
> Asterisk.

I agree, we've had zero trouble with these. Easy to install and they just work.

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Re: [asterisk-users] Asterisk, Zaptel and the Kernal Compatibility Matrix

[Matt Florell]

Hello,

I was never able to get the TE407P card running on a 2.4 Linux kernel.
Using a 2.6 kernel I was able to get it working.

Not really surprising since a lot of companies do not support or even
test on Linux 2.4 any more.

MATT---


On 2/21/08, Kevin P. Fleming <[EMAIL PROTECTED]> wrote:
> bilal ghayyad wrote:
>
>  > How can I know the needed Zaptel and Kernel versions
>  > for my Asterisk version? Where I can find the
>  > compatibility matrix for such thing?
>
>
> There is no such thing. If the version of Zaptel you have isn't
>  compatible with the version of Asterisk you are trying to build,
>  Asterisk won't use it. If the version of the kernel you have isn't
>  compatible with the version of Zaptel you are trying to build (which is
>  unlikely), Zaptel won't build against it. Asterisk does not care about
>  kernel versions.
>
>
>  --
>  Kevin P. Fleming
>  Director of Software Technologies
>  Digium, Inc. - "The Genuine Asterisk Experience" (TM)
>
>
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Re: [asterisk-users] Pager ("beeper") Emulation Script

[Darryl Dunkin]

I've done similar notifications in the dialplan.

It would probably look something like this:
exten => s,1,Read(PAGE,enter-phone-number10,10)
exten => s,2,System(/bin/echo "Page content: ${PAGE}" | /bin/mail -s
"Page subject" [EMAIL PROTECTED]) 


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andreas
van dem Helge
Sent: Thursday, February 21, 2008 21:53
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Pager ("beeper") Emulation Script

Does anyone have a script that will emulate a normal numeric pager but
send the number to an email address? Also anyone happen to have the
traditional tones used in North America?

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[asterisk-users] chan_woomera tries to connect to strange host

[Ganbold Tsagaankhuu]

Hi,

It is very strange that following chan_woomera code part gives IP address
44.215.5.41.

static int connect_woomera(int *new_socket, woomera_profile *profile, int
flags)
{
struct sockaddr_in localAddr, remoteAddr;
struct hostent *hp;
struct ast_hostent ahp;
int res = 0;

*new_socket=-1;

printf("WOOMERA HOST: %s\n",profile->woomera_host); // THIS PRINTS CORRECT
IP ADDRESS
if ((hp = ast_gethostbyname(profile->woomera_host, &ahp))) {
remoteAddr.sin_family = hp->h_addrtype; memcpy((char *)
&remoteAddr.sin_addr.s_addr, hp->h_addr_list[0], hp->h_length);

printf(" WOOMERA HOST: %s\n", inet_ntoa(
remoteAddr.sin_addr)); // THIS PRINTS 44.215.5.41
remoteAddr.sin_port = htons(profile->woomera_port);

And chan_woomera tries to connect to 44.215.5.41:42240, it is very strange,
because
I have already defined host 192.168.0.18 in woomera.conf.

I can hardcode it like:

// memcpy((char *) &remoteAddr.sin_addr.s_addr, hp->h_addr_list[0],
hp->h_length);
inet_aton("192.168.0.18", &remoteAddr.sin_addr);

But I would like to know the right solution.
Please let me know.

thanks,

Ganbold
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[asterisk-users] Friday 22 FEB 08 @ 12 Noon EST ISPBX COGOBLUE

[randulo]

Happy coming Spring,

Every week we try to get guests with ideas, products and services you
haven't had time to check out to come and talk about what they're
doing.

Friday, February 22 at 12:00 PM (Eastern US) 9AM PST, 5PM GMT

* Call (724) 444-7444   or   sip:[EMAIL PROTECTED]

After the call connects, enter the conf: 22622# and your_PIN# (or 1#
if you have no PIN)

http://VoIPUsersConference.org for how to listen and join.

COGOBLUE is the visual, drag & drop configuration tool for Asterisk
PBX systems.The concept is simple: use easily identifiable icons to
represent components and features of the PBX system, drag & drop the
icons onto a pane and arrange them to get the configuration you want.
Save your work and COGOBLUE automatically generates the Asterisk
configuration files. A simple concept with powerful results: that's
COGOBLUE. http://cogo.ispbx.com/

http://food4wine.ning.com is the VUC Community Site (archives of all
sessions are here)

IRC freenode.net #voip-users-conference is the channel to ask
questions if you can't call

Google Group/Mailing List: http://groups.google.com/group/voip-users-conference

How to set up asterisk to call in via SIP:
http://voipusersconference.org/asterisktalkshoecallinsetup.htm

/r

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[asterisk-users] Pager ("beeper") Emulation Script

[Andreas van dem Helge]

Does anyone have a script that will emulate a normal numeric pager but
send the number to an email address? Also anyone happen to have the
traditional tones used in North America?

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[asterisk-users] (no subject)

[sandeep]

hi,

how to write a advanced dial plan

for example:
dial to a extension(123).if the user didnot pick the call, caller should get a 
ivr script(Enter 1 to to dial operator  and 2 to go to voicemail)
If caller press 1 it should dial to the operator,else if he dials 2 it should 
go to the voicemail of calle's extension.

thanks
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Re: [asterisk-users] Voted most stable and easy to use phone?

[John Faubion]

>A while back i had asked about possible replacements for snom 360 phones
that were breaking and causing
>issues and we all discussed the problems we had with the 360s and some
suggestions were  made but the
>new polycom phones had just hit the market and not many people were able to
comment on them.

We just installed a dozen of the Polycom IP-330 phones. Initially out of the
box I wasn't real sure about the decision to use them. The phones are very
small and don't seem to have very many features. However in use they have
been great. They don't waste a lot of desk space, they don't overwhelm the
users and they seem to provide adequate information. They're easy to use and
Polycom reliable. The speaker phone is still really good though I'm not sure
it is as good as the 501/601 phones. I haven't really done a side by side
comparison of that but I think the 501/601 has a better speaker phone. I
can't see buying another GXP after using these. The difference in price just
isn't worth the aggravation.

John
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[asterisk-users] spandsp/tx_fax/rx_fax frustrations

[Edwin Lam]

hi

does any body know which version combination of
spandsp/tx_fax/rx_fax will work with * 1.2.24?

i tried different combo. they're either seg fault
during runtime or won't compile.

very frustrated :/

p.s. i know. hylafax/iaxmodem is far more stable. but i have
specific reasons to use rx_fax.

-- 
Edwin Lam <[EMAIL PROTECTED]>
Systems Engineer, Office General, Inc.
Ph: +1 415 439 4988 Fax: +1 415 283 3370
http://pgpkeys.mit.edu:11371/pks/lookup?op=get&search=0xD6506D20


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[asterisk-users] Chan_h323 isn`t dropping calls comming with wrong codecs

[Andre Luiz Martins Rodrigues]

I` using chan_h323 on my asterisk-1.4 to receive incomings calls. I need 
to set just two codecs to receive this call (g723 and g729), but I`m using

disallow=all
allow=g729
allow=g723.1

In h323.conf, but when I received a call using codec g711 for example, 
the call is answered, but doesn`t have audio. I made a test today using just

disallow=all

In h323.conf, but the call was answered too!!

the log of this test:


[Feb 21 23:44:56] DEBUG[4264]: chan_h323.c:2112 setup_incoming_call: 
Setting up incoming call for ip$189.0.24.69:4020/28391
-- Setting up Call
--  CLI>Call token:  [ip$189.0.24.69:4020/28391]
--  CLI>Calling party name:  [200]
--  CLI>Calling party number:  [200]
--  CLI>Called party name:  [30144588]
--  CLI>Called party number:  [30144588]
--  CLI>Calling party IP:  [189.0.24.69]
[Feb 21 23:44:56] DEBUG[4264]: chan_h323.c:1611 find_user: Could not 
find user by name 200 or address 189.0.24.69
[Feb 21 23:44:56] DEBUG[4264]: chan_h323.c:2177 setup_incoming_call: 
Sending [EMAIL PROTECTED] to context [ss7] extension 30144588
[Feb 21 23:44:56] DEBUG[4264]: chan_h323.c:2478 set_local_capabilities: 
Setting capabilities for connection ip$189.0.24.69:4020/28391
Setting capabilities to 0x0 (nothing)
Capabilities in preference order is ()
DTMF mode is 1
Allowed Codecs for ip$189.0.24.69:4020/28391 
(ip$201.7.99.242:1720): 
   
 

 Zap/33-1 answered H323/ip$189.0.24.69:4020/28391
[Feb 21 23:44:57] DEBUG[4264]: chan_h323.c:666 oh323_answer: Answering 
on H323/ip$189.0.24.69:4020/28391
Answering call 
ip$189.0.24.69:4020/28391   
 
 

Receiving RFC2833 on payload 101
Peer capability is G.711-uLaw-64k <1>
Found peer capability G.711-uLaw-64k <1>, Asterisk code is 4, frame size 
(in ms) is 20
Peer capability is G.711-ALaw-64k <2>
Found peer capability G.711-ALaw-64k <2>, Asterisk code is 8, frame size 
(in ms) is 20
Peer capabilities = 0xc (ulaw|alaw), ordered list is (ulaw|alaw)
[Feb 21 23:44:57] DEBUG[4264]: chan_h323.c:2448 set_peer_capabilities: 
Got remote capabilities from connection ip$189.0.24.69:4020/28391
[Feb 21 23:44:57] DEBUG[4264]: chan_h323.c:2462 set_peer_capabilities: 
prefs[0]=ulaw:20
[Feb 21 23:44:57] DEBUG[4264]: chan_h323.c:2462 set_peer_capabilities: 
prefs[1]=alaw:20
=-= In OnConnectionEstablished for call 28391
-- Connection Established with "200 [189.0.24.69]"
[Feb 21 23:44:57] DEBUG[4264]: chan_h323.c:2055 connection_made: Call 
ip$189.0.24.69:4020/28391 answered 


Some one knows why isn`t asterisk droping the call?

Andre Luiz Martins


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Re: [asterisk-users] How to get a clean, basic configuration?

[Mojo with Horan & Company, LLC]

Tzafrir Cohen wrote:
>> Delete extensions.ael too, unless you're using AEL instead of the dialplan
>> 
>
> extensions.ael is harmless on its own.
>   
It seemed that the default extensions.ael created some demo contexts and 
extensions that might befuddle a new user, I could be wrong


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[asterisk-users] FW: jabber

[clive.chan(atn)]

 
 
Hi all, 
Do some one experiencing running jabber applications (jabberstatus...) in
asterisk? I do experinced Asterisk 1.4.18 and wish to start it, however I
got such result.
IBM*CLI> help jabber
No such command 'jabber'.
IBM*CLI> help jabberstatus
No such command 'jabberstatus'.

 
Any one can help me on this, or may be I miss out somethings that cause
jabber applications did'nt install.
 
 
 
Best ragards
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Re: [asterisk-users] Allow INVITE for hold to pass through

[Kevin P. Fleming]

Mayur wrote:

>I would like to configure asterisk to allow INVITE for hold to pass
> through it and not provide music on hold by itself. Can anyone help me
> out here?

In Asterisk 1.4 this is a configuration option; check the sample
sip.conf file in the configs directory for documentation.

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

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Re: [asterisk-users] Asterisk, Zaptel and the Kernal Compatibility Matrix

[Kevin P. Fleming]

bilal ghayyad wrote:

> How can I know the needed Zaptel and Kernel versions
> for my Asterisk version? Where I can find the
> compatibility matrix for such thing?

There is no such thing. If the version of Zaptel you have isn't
compatible with the version of Asterisk you are trying to build,
Asterisk won't use it. If the version of the kernel you have isn't
compatible with the version of Zaptel you are trying to build (which is
unlikely), Zaptel won't build against it. Asterisk does not care about
kernel versions.

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

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[asterisk-users] Question regarding AGI

[sanjay . rajdev]

I have questions about AGI.

1.  When Using "CONTROL STREAM FILE" command with all the parameter, I could 
not find any way to "*" or "#" in the DTMF, it only returns if any digit is 
pressed, even if I set forward and rewind digits to BLANK ("")

2.  When I call out using ZAP, is their a way to find if the call went to the 
Person Called, or if their was a message played by his service provider, e.g. 
"Number does not exists", "Out of range", "Switched off", etc. Can we even 
achieve this using some Dialplan applications

Can someone please help.

Regards,
Sanjay.


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Re: [asterisk-users] USB ISDN interface

[Paul Hales]

The Xorcom stuff should be easy enough to find.

PaulH


On Thu, 2008-02-21 at 06:42 -0700, Michael Blood wrote:
> I have been looking for one of the many USB to ISDN products listed on
> this page,  
> http://www.nslu2-linux.org/wiki/OpenSlug/Asterisk
> But I have not been able to find a single item for sale anywhere in
> the US,  has anyone seen where I can find a couple of these?
>  
> Thanks
>  
> Michael
>  
>  
>  
>  
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Re: [asterisk-users] Answered Call marked as "NO ANSWER"

[Jorge Mendoza]

Raúl,

Callprogress is not reliable for call supervision. Sorry.
For maximum reliability with callprogress, the tones and cadences send
by the CO must match every well with the tones plan defined in your
asterisk box. Probably the tones of the other telephone company, where
the answer detection fail, are different or the cadences are different.

Jorge

Raúl Gómez C. wrote:
> Jorge,
>
> I think our telco doesn't provide disconnection supervision because I
> had to use "callprogress", "busydetect" and "busycount" in order to
> properly disconnect a terminated call (and to avoid the infamous long
> message in the voicemail), so I think I can't disable the
> "callprogress" option.
>
> I will try to contact the telco provider of these "numbers" in order
> to ask them what kind of answer supervision they provide.
>
> Any other ideas???
>
> Thanks again
>
>
> -- 
> Raul
> Linux Counter #156439
>
>
> On Fri, Feb 22, 2008 at 1:15 PM, Jorge Mendoza <[EMAIL PROTECTED]
> > wrote:
>
> Raúl,
>
> From your conf file I guess the CO provide reversal polarity for
> answer
> supervision. Verify if for those "numbers" the CO revert the line
> polarity when callee answer.
> callprogress=no is a good test too.
>
> Jorge
>
> Raúl Gómez C. wrote:
> > Hi list,
> >
> > I'm having problems transferring certain calls made by the attendant
> > between the PSTN and to an internal extension. Although, transfers
> > between the majority of the calls ends successfully.
> >
> > Debugging this, I've found that calls made to certain "numbers"
> > (Telephony Providers), aren't detected as ANSWERED in the CDR,
> so they
> > are not properly accounted (for billing), neither transferred to
> > internals extensions.
> >
> > How can I solve this??? Is this a incompatibility issue between
> > technologies??? Or just a config that I haven't made right???
> >
> > Thanks in advance...
> >
> >
> > My Setup:
> >
> > - Asterisk 1.4.17
> > - Sangoma Remora A400D HEC PCI Card (2 FXS / 10 FXO)
> > - Wanpipe 3.2.1
> > - Zaptel *MailScanner warning: numerical links are often
> malicious:* 1.4.7.1 
> > - Grandstream GXP-2000 Phones
> >
>
> 
>
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[asterisk-users] Asterisk, Zaptel and the Kernal Compatibility Matrix

[bilal ghayyad]

Hi List;

How can I know the needed Zaptel and Kernel versions
for my Asterisk version? Where I can find the
compatibility matrix for such thing?

Regards
Bilal


  

Looking for last minute shopping deals?  
Find them fast with Yahoo! Search.  
http://tools.search.yahoo.com/newsearch/category.php?category=shopping

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Re: [asterisk-users] High CPU load after upgrading to 1.4

[Jared Smith]

On Thu, 2008-02-21 at 21:12 +, [EMAIL PROTECTED] wrote:
> I currently have 1558 sip peers loaded in
> Asterisk and the current CPU load is 10% when no calls are being
> processed and no sip registrations.

At first glance, I would think that maybe you have "qualify=yes" in each
of your SIP peers, which is keeping Asterisk busy checking to see if the
peers are responding or not.

-- 
Jared Smith
Community Relations Manager
Digium, Inc.



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[asterisk-users] cid_rewrite.php -- Caller ID Name lookup

[Jay Milk]

For those folks who are still using it --

I updated the cid_rewrite script.  I noticed that two of the providers 
were "iffy" and one had changed format a little while ago.  It's working 
again.

http://muware.com/asterisk has the latest (1.2.0)

Enjoy,
-- JM

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Re: [asterisk-users] Dial+Macro and Queue

[Shaun R.]

What is it that you think is missing, call comes into incomming, call gets 
queued, member [EMAIL PROTECTED] is called and a macro is played to them, they 
hit 
option 3, MACRO_RESULT gets set to CONTINUE and the call hangs up on the 
member while the caller continues on... the caller now though gets put back 
into the queue rahter than continueing on in the dialplan like 
MACRO_RESULT=Continue is suppose to do.

~Shaun


"Paul Hales" <[EMAIL PROTECTED]> wrote in message 
news:[EMAIL PROTECTED]
>
> This really looks like we are missing a lot of the associated code.
>
> PaulH
>
>
> On Wed, 2008-02-20 at 00:28 -0800, Shaun R. wrote:
>> A call comes in and goes into the queue, the queue dials a sip channel 
>> using
>> a macro.  The macro plays a set of options to the callee and if the 
>> callee
>> presses 3 it sets MACRO_RESULT=CONTINUE and the macro ends.  For some 
>> reason
>> the caller goes back into the queue rather than continueing on in the 
>> dial
>> plan.  Why is this, i could have sworn in 1.2 if i set 
>> MACRO_RESULT=CONTINUE
>> that the caller exited the queue() and continued on in the dialplan...
>>
>> [incomming]
>> exten => 1,1,Queue(mainqueue,td)
>> exten => 1,2,voicemail([EMAIL PROTECTED])
>> exten => 1,3,hangup
>>
>> [screen]
>> exten => _3XX,1,ChanIsAvail(SIP/${EXTEN}&IAX2/${EXTEN})
>> exten => _3XX,2,GotoIf($["${AVAILCHAN}" = ""]?4)
>> exten => _3XX,3,Dial(${CUT(AVAILCHAN,-,1)},30,mgM(screencallee,s,1))
>> exten => _3XX,4,Hangup
>>
>> [macro-screencallee]
>> exten => s,1,read(SCREEN_OPT,screenoptions)
>> exten => s,2,GotoIf($["${SCREEN_OPT}" = "" ]?s,1)
>> exten => s,3,GotoIf($["${SCREEN_OPT}" = "3" ]?3,1)
>> exten => 3,1,Set(MACRO_RESULT=CONTINUE)
>> exten => t,1,Set(MACRO_RESULT=BUSY)
>> exten => h,1,Set(MACRO_RESULT=BUSY)
>>
>> queues.conf
>> [mainqueue]
>> musicclass = default
>> strategy = ringall
>> timeout = 600
>> joinempty = yes
>> member => local/[EMAIL PROTECTED]
>>
>> ~Shaun
>>
>>
>>
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Re: [asterisk-users] ata device but for a soundcard

[Kelvin Chen]

> 
> I am looking for an ATA like device but instead of VOIP to analog
phone
> I want VOIP to low level audio out. Something that looks like a sound
card
> output.
> 
> I know I can use cheap PC's but that then you have HD's to setup
etc...
> HD failures etc...
> 
> Anyone know of something like that?
> 

Not an expert on this one. And I didn't read too much into chan_alsa's
documentation.
So just out of curiosity, isn't chan_alsa built for this kind of
application?
 

kel

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Re: [asterisk-users] Answered Call marked as "NO ANSWER"

[Raúl Gómez C.]

Jorge,

I think our telco doesn't provide disconnection supervision because I had to
use "callprogress", "busydetect" and "busycount" in order to properly
disconnect a terminated call (and to avoid the infamous long message in the
voicemail), so I think I can't disable the "callprogress" option.

I will try to contact the telco provider of these "numbers" in order to ask
them what kind of answer supervision they provide.

Any other ideas???

Thanks again


-- 
Raul
Linux Counter #156439


On Fri, Feb 22, 2008 at 1:15 PM, Jorge Mendoza <[EMAIL PROTECTED]> wrote:

> Raúl,
>
> From your conf file I guess the CO provide reversal polarity for answer
> supervision. Verify if for those "numbers" the CO revert the line
> polarity when callee answer.
> callprogress=no is a good test too.
>
> Jorge
>
> Raúl Gómez C. wrote:
> > Hi list,
> >
> > I'm having problems transferring certain calls made by the attendant
> > between the PSTN and to an internal extension. Although, transfers
> > between the majority of the calls ends successfully.
> >
> > Debugging this, I've found that calls made to certain "numbers"
> > (Telephony Providers), aren't detected as ANSWERED in the CDR, so they
> > are not properly accounted (for billing), neither transferred to
> > internals extensions.
> >
> > How can I solve this??? Is this a incompatibility issue between
> > technologies??? Or just a config that I haven't made right???
> >
> > Thanks in advance...
> >
> >
> > My Setup:
> >
> > - Asterisk 1.4.17
> > - Sangoma Remora A400D HEC PCI Card (2 FXS / 10 FXO)
> > - Wanpipe 3.2.1
> > - Zaptel 1.4.7.1
> > - Grandstream GXP-2000 Phones
> >
>
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[asterisk-users] High CPU load after upgrading to 1.4

[xrem1x]

Hi,

Since I upgraded from Asterisk 1.2.18 to 1.4.17 I've been experiencing
high CPU utilization from the chan_sip module.  I've notice the more sip
peers I have loaded, the higher the CPU load goes when there are no
active calls.  I am currently using a Pentium 4 3.0Ghz with CentOS 4
Kernel 2.6.9-42.0.2.EL.   I currently have 1558 sip peers loaded in
Asterisk and the current CPU load is 10% when no calls are being
processed and no sip registrations.  When calls are being processed it
is also higher than normal aside from CPU load when computer is idle.
Before the upgrade I never had a CPU load of 10% when there were no
calls being processed, it was always close to 0%.

When I install  CentOS 5 on a different computer with latest 32bit
kernel I am still able to reproduce the high CPU load.  But when I
install the x86_64 Kernel the high CPU load problem disappears.  It
appears that there must be a kernel setting that is causing the CPU to
spike up higher than normal for Asterisk 1.4.  Does anyone have an idea
on what could be causing this high CPU load?

Thanks,


Remi Quezada

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[asterisk-users] Asterisk-addons 1.6.0-beta2 Released

[The Asterisk Development Team]

The Asterisk.org development team has released Asterisk-addons version 
1.6.0-beta2.

This release contains the following improvement, along with some other minor bug
fixes.
 - 11614, Updated app_fax to allow termination and origination of faxes over
  T.38

The full list of changes is available in the ChangeLog.  The release is
available for download from http://downloads.digium.com.

Thank you for your support!

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[asterisk-users] Asterisk 1.6.0-beta4 Released

[The Asterisk Development Team]

The Asterisk.org development team has released version 1.6.0-beta4.

Here are some highlights from the changes, with the associated issue numbers
from bugs.digium.com if an issue was associated with the change.

This release contains the following improvements:
 - 12020, a CLI formatting improvement
 - 11964, added the ability to get the original called number on SS7 calls
 - 11873, Added core API changes to handle T.38 origination and termination
   (The version of app_fax in Asterisk-addons now supports this.)
 - 11553, Added a status variable to the ChannelRedirect() application

The changes in this release include fixes for the following issues (trivial and
minor issues not included):
 - 11960, a crash in chan_sip
 - 12021, a crash related to invalid formats being specified for voicemail
 - 11779, fix enabling echo cancellation for incoming SS7 calls
 - 11740, DTMF handling fixes
 - 11864, Fixed device state reporting on incoming calls on FXO
 - 12012, a crash in chan_local
 - Fix a regression in codec handling that was introduced in 1.6.0-beta3

A full list of changes can be found in the ChangeLog.  This release is available
for immediate download from http://downloads.digium.com/.

Thank you for your support!

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Re: [asterisk-users] Pattern matching....

[Eric Wieling]

That will match the following as well

770
700
740
400
470
670
600
604
608
etc.

You example says:

The first digit can be 7 or 4 or 6.  The 2nd digit can be 7 or 0 or 4. 
The 3rd digit can be 0 or 4 or 8.

Mike Trest - Personal wrote:
> [746][704][048]
> 
> [At 01:21 PM 2/21/2008, you wrote:
>> On Thu, 2008-02-21 at 12:44 -0500, Michael Munger wrote:
>>> Will this work to match any number from the 770,404, or 678 area
>>> codes?
>>>
>>>
>>>
>>> _[404|770|678]NXX
>>>
>>>
>>>
>>> If this won’t work, is there a pattern that will do this?
>>>
>>>
>> No, it won't work, there's no '|' for alternative matches, and no parens
>> available for grouping, either. And your usage of char sets is off.
>> Try something like this:
>>
>> _404NXX
>>
>> _770NXX
>>
>> _678NXX
>>
>>
>> as three separate extensions.
>>
>> If you REALLY want to keep that as one extension, then you could:
>>
>> _NXXNXX  =>  {
>> Set(areacode=${EXTEN:0:3})
>> if ('${areacode}' = '404') {
>> 
>> } else if ('${areacode}' = '770') {
>> 
>> } else if ('${areacode}' = '678') {
>> 
>> }
>>
>>
>> OR, you could do it this way, also:
>>
>> _NXXNXX  =>  {
>> Set(areacode=${EXTEN:0:3})
>> switch(${areacode})
>> {
>> case 404:
>> 
>> break;
>> case 770:
>> 
>> break;
>> case 678:
>> 
>> break;
>> }
>>
>> This is, of course AEL code, and this stuff would be inside a context
>> construct...
>>
>> murf
>>
>>
>>
>>
>>
>>
>>>
>>> Yours,
>>>
>>> Michael Munger, dCAP
>>>
>>> 404-438-2128
>>>
>>> [EMAIL PROTECTED]
>>>
>>>
>>>
>>> Attachment encrypted? click here.
>>>
>>>
>>>
>>>
>>> ___
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>>> To UNSUBSCRIBE or update options visit:
>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>> --
>> Steve Murphy
>> Software Developer
>> Digium
>>
>>
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> 
> 
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> 

-- 
Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, 
T-1, PRI, Frame Relay, Linux, and network design.  Based near 
Birmingham, AL.  Now accepting clients worldwide.


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Re: [asterisk-users] Pattern matching....

[Eric Wieling]

No that will not work.  You would want three exten => lines, one for 
each area code.

Michael Munger wrote:
> Will this work to match any number from the 770,404, or 678 area codes? 
> 
>  
> 
> _[404|770|678]NXX
> 
>  
> 
> If this won't work, is there a pattern that will do this?
> 
>  
> 
> Yours,
> 
> Michael Munger, dCAP
> 
> 404-438-2128
> 
> [EMAIL PROTECTED]  
> 
>  
> 
> Attachment encrypted? click here
>  .
> 
>  
> 
> 
> 
> 
> 
> 
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-- 
Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS, 
T-1, PRI, Frame Relay, Linux, and network design.  Based near 
Birmingham, AL.  Now accepting clients worldwide.

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[asterisk-users] Asterisk-addons 1.4.6 Released

[The Asterisk Development Team]

The Asterisk.org development team has released Asterisk-addons version 1.4.6.

This releases includes a fix for a build related issue for the OOH323 channel
driver.  (issue #9643)

Thank you for your support!

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Re: [asterisk-users] Zaptel 1.4.8 breaks tor2 support on CentOS 5.1? (kernel panic)

[Tzafrir Cohen]

On Thu, Feb 21, 2008 at 02:17:20PM -0500, Nick Seraphin wrote:
> 
> 
> Thank you very much!!!
> 
> What was the one line fix?
> 
> Also, what file was the problem in?  Also, if you know the line number or
> function it was in, that would be nice too.  I'd do a diff, but I assume
> there has been other changes since 1.4.9 was released.

You can find the fix Kevin has just commited at:

  http://svn.digium.com/view/zaptel?view=rev&revision=3863

Generally http://svn.digium.com/view/zaptel/branches/1.4/ is a good
starting point to check for "recent changes".

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

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Re: [asterisk-users] Zaptel 1.4.8 breaks tor2 support on CentOS 5.1? (kernel panic)

[Nick Seraphin]

Thank you very much!!!

What was the one line fix?

Also, what file was the problem in?  Also, if you know the line number or
function it was in, that would be nice too.  I'd do a diff, but I assume
there has been other changes since 1.4.9 was released.

Thanks again!

-- Nick


On Thu, 21 Feb 2008, Kevin P. Fleming wrote:

> Kevin P. Fleming wrote:
> 
> > I've just located an E400P from our graveyard of old cards... if it
> > works, I'll be able to solve this problem in the morning.
> 
> This has been fixed in revision 3863 of the 1.4 branch; it's a one line
> fix that you should be able to easily apply to existing Zaptel source
> code, or you can wait for the next Zaptel release which should happen
> later today. Sorry for the breakage.
> 
> -- 
> Kevin P. Fleming
> Director of Software Technologies
> Digium, Inc. - "The Genuine Asterisk Experience" (TM)
> 
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Re: [asterisk-users] How to get a clean, basic configuration?

[Tzafrir Cohen]

On Thu, Feb 21, 2008 at 09:23:49AM -0900, Mojo with Horan & Company, LLC wrote:
> Mindaugas Kezys wrote:
> > We do:
> >
> > in modules.conf:
> >
> > noload => pbx_ael.so
> > noload => pbx_dundi.so
> > noload => res_config_pgsql.so
> > noload => res_smdi.so
>
> Delete extensions.ael too, unless you're using AEL instead of the dialplan

extensions.ael is harmless on its own.

> >
> > in extensions.conf delete every context [default], [demo], whatever
> >
> > in sip.conf, iax.conf delete all peer/users if any

Hmmm... try starting from an empty configuration directory and put in
only what you need?

The sample files are very handy, but are always available for you
separately.

For the brave: use modules.conf without 'autoload = yes'. This promises
you many hours of interesting dialplan debugging. Enjoy.

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

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Re: [asterisk-users] Pattern matching....

[Jared Smith]

On Thu, 2008-02-21 at 13:34 -0500, Mike Trest - Personal wrote:
> [746][704][048]

Nope, that's not going to do exactly what you want either... that
pattern would match a lot of area codes besides the ones you're looking
for.  (For example, you could have 7 as the first digit and 0 as the
second digit and 0 as the third digit.)

You really need to have three separate patters; one for each area code.

-- 
Jared Smith
Community Relations Manager
Digium, Inc.


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Re: [asterisk-users] Answered Call marked as "NO ANSWER"

[Raúl Gómez C.]

Thanks Jorge, I'll be checking that...

On Fri, Feb 22, 2008 at 1:15 PM, Jorge Mendoza <[EMAIL PROTECTED]> wrote:

> Raúl,
>
> From your conf file I guess the CO provide reversal polarity for answer
> supervision. Verify if for those "numbers" the CO revert the line
> polarity when callee answer.
> callprogress=no is a good test too.
>
> Jorge
>
> Raúl Gómez C. wrote:
> > Hi list,
> >
> > I'm having problems transferring certain calls made by the attendant
> > between the PSTN and to an internal extension. Although, transfers
> > between the majority of the calls ends successfully.
> >
> > Debugin this, I've found that calls made to certain "numbers"
> > (Telephony Providers), aren't detected as ANSWERED in the CDR, so they
> > are not properly accounted (for billing), neither transferred to
> > internals extensions.
> >
> > How can I solve this??? Is this a incompatibility issue between
> > technologies??? Or just a config that I haven't made right???
> >
> > Thanks in advance...
> >
> >
> > My Setup:
> >
> > - Asterisk 1.4.17
> > - Sangoma Remora A400D HEC PCI Card (2 FXS / 10 FXO)
> > - Wanpipe 3.2.1
> > - Zaptel *MailScanner warning: numerical links are often malicious:*
> > 1.4.7.1 
> > - Grandstream GXP-2000 Phones
> >
>


-- 
Nacho
Linux Counter #156439
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Re: [asterisk-users] Voted most stable and easy to use phone?

[arkda]

I'm a huge fan of the Linksys SPA-942s for users. They run around $125, are
pretty straightforward to manage via TFTP, and work really well with
Asterisk.

On Thu, Feb 21, 2008 at 4:07 AM, Michael J. Liberatore <
[EMAIL PROTECTED]> wrote:

>  A while back i had asked about possible replacements for snom 360 phones
> that were breaking and causing issues and we all discussed the problems we
> had with the 360s and some suggestions were made but the  new polycom phones
> had just hit the market and not many people were able to comment on them.
>
> Basically i am looking to get some new phones and in the process get rid
> of the countless number of problems i have had that has always been caused
> by phones (snom 360's and gxp-2000's).  I would like to get the feedback of
> the list on the phone voted best for stability, working with *, and ease of
> use for dumb non tech users.
>
> I was thinking of trying one of these new polycom phones that are about
> $150, but havent gotten any feedback on them yet.
>
> Basically i am interested in any phones but snom's, grandstreams, and
> sipura's/linksys.  mainly polycom's i guess.
>
> thanks
>
> mike
>
> This E-mail, including any attachments, may be intended solely for the
> personal and confidential use of the sender and recipient(s) named above.
> This message may include advisory, consultative and/or deliberative material
> and, as such, would be privileged and confidential and not a public
> document. Pursuant to 42 CFR, any information in this e-mail identifying a 
> former,
> present, or potential client of Straight & Narrow is confidential. If you
> have received this e-mail in error, you must not review, transmit, convert
> to hard copy, copy, use or disseminate this e-mail or any attachments to it
> and you must delete this message. You are requested to notify the sender by
> return e-mail.
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Re: [asterisk-users] Pattern matching....

[Mike Trest - Personal]

[746][704][048]

[At 01:21 PM 2/21/2008, you wrote:
>On Thu, 2008-02-21 at 12:44 -0500, Michael Munger wrote:
> > Will this work to match any number from the 770,404, or 678 area
> > codes?
> >
> >
> >
> > _[404|770|678]NXX
> >
> >
> >
> > If this won’t work, is there a pattern that will do this?
> >
> >
>
>No, it won't work, there's no '|' for alternative matches, and no parens
>available for grouping, either. And your usage of char sets is off.
>Try something like this:
>
>_404NXX
>
>_770NXX
>
>_678NXX
>
>
>as three separate extensions.
>
>If you REALLY want to keep that as one extension, then you could:
>
>_NXXNXX  =>  {
> Set(areacode=${EXTEN:0:3})
> if ('${areacode}' = '404') {
> 
> } else if ('${areacode}' = '770') {
> 
> } else if ('${areacode}' = '678') {
> 
> }
>
>
>OR, you could do it this way, also:
>
>_NXXNXX  =>  {
> Set(areacode=${EXTEN:0:3})
> switch(${areacode})
> {
> case 404:
> 
> break;
> case 770:
> 
> break;
> case 678:
> 
> break;
> }
>
>This is, of course AEL code, and this stuff would be inside a context
>construct...
>
>murf
>
>
>
>
>
>
> >
> >
> > Yours,
> >
> > Michael Munger, dCAP
> >
> > 404-438-2128
> >
> > [EMAIL PROTECTED]
> >
> >
> >
> > Attachment encrypted? click here.
> >
> >
> >
> >
> > ___
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> >
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> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
>--
>Steve Murphy
>Software Developer
>Digium
>
>
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Re: [asterisk-users] Sangoma FXO EC vs Rhino FXO EC

[Rod Montgomery]

Michael,

I'm very interested in helping you resolve any and all trouble you have with 
Digium products. The experience you describe with our products not typical, so 
I'd like to address them specifically.

My comments are inline:

- "Michael J. Liberatore" <[EMAIL PROTECTED]> wrote:
> 
> I have noticed overall sound quality has increased 10 fold with the
> sangoma echo cancellation card but I had never tried hpec with the
> digium card.  I did try mg2 after fxotune and spending lots of time
> working out the levels and they still are unhappy with the call
> quality.

Yes, any decent hardware echo canceler would provide a significant improvement 
over mg2 on difficult trunks. As noted already in this thread, Digium offers 
software-based High Performance Echo Canceler (HPEC) and hardware-based echo 
cancellation options. They're well worth the incremental expense when you need 
reliable, high-quality echo cancellation. 

Evaluating a Digium interface without hardware echo cancellation versus Sangoma 
with hardware echo cancellation isn't really an even comparison, is it?  I 
invite you to evaluate similarly equipped interfaces.  I'm confident that 
you'll change your opinion about Digium's product performance.



> No I I have the TDM-04B, I was going by voipsupply.com, that's why I
> thought I had the latest, they don't have the tdm410.  

The TDM400 does not support on-board echo cancellation; the TDM410 does. For 
low density analog, the TDM410 is an outstanding solution. Voipsupply's website 
catalog isn't exactly up to date, but there are plenty of other reputable 
resellers that would be eager to help you with the latest interface technology; 
just visit http://www.digium.com/en/wheretobuy/>.

However, it sounds like you're reselling these to end users: have you 
considered joining Digium's reseller program? You would enjoy many benefits 
that would smooth your path to success, including special pricing with an 
authorized Digium distributor. See 
http://www.digium.com/en/ecosystem/resellers/> for more information.



> My issue is mainly that I bought all digium cards previously and have
> nothing but nightmares. Multiple cards died for no reason, 2 of them
> digium wouldn't warranty cause they were like 13 months old.  

That's troubling... even before the recent announcement of an expanded warranty 
as part of our Exceptional Satisfaction Program, all Digium interface cards 
carried a two year warranty. There's no way we would have turned you away at 13 
months. Or 15 months, or 18.

I went to investigate this and couldn't find any record of you contacting 
Digium Technical Support. Did you use a different company name? Perhaps a 
different representative of your firm contacted us. Are you confusing your 
e-tailer's limited one-year return policy for Digium's warranty?  Even if your 
webstore refuses a return, Digium will honor the product warranty.

Truly, the tired old reasons to avoid Digium -- IRQ-related headaches, 
"motherboard incompatibilities," lack of hardware echo cancellation on analog 
and single-span T1, etc. -- just don't stand up any more. Digium's Exceptional 
Satisfaction Program signals a new standard of quality and reliability in the 
telephony market. And our Technical Support team has a great track record, 
often mentioned on this list. We're ready to work with you to make your Digium 
Asterisk deployments a resounding success.

Sincerely,
rm
--
Rod Montgomery <[EMAIL PROTECTED]>
Director of Services, Digium, Inc.


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Re: [asterisk-users] How to get a clean, basic configuration?

[Mojo with Horan & Company, LLC]

Delete extensions.ael too, unless you're using AEL instead of the dialplan
Mindaugas Kezys wrote:
> We do:
>
> in modules.conf:
>
> noload => pbx_ael.so
> noload => pbx_dundi.so
> noload => res_config_pgsql.so
> noload => res_smdi.so
>
> in extensions.conf delete every context [default], [demo], whatever
>
> in sip.conf, iax.conf delete all peer/users if any
>
> Regards,
> Mindaugas Kezys
> http://www.kolmisoft.com
> MOR PRO - Advanced Billing for Asterisk PBX
>
>
>
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vincent
> Sent: Thursday, February 21, 2008 4:31 AM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] How to get a clean, basic configuration?
>
> Hello
>
> I'm using a standard Asterisk install with default settings, and when
> I run "reload", I see that Asterisk fetches configuration information
> from a lot more sources than just my extensions.conf and sip.conf.
>
> For instance:
>
> -- Registered indication country 've'
> -- Registered indication country 'za'
> -- Setting default indication country to 'us'
>   == Parsing '/etc/asterisk/features.conf': Found
>   == Parsing '/etc/asterisk/adsi.conf': Found
>   == Parsing '/etc/asterisk/dundi.conf': Found
>   == Parsing '/etc/asterisk/extensions.conf': Found
>   == Parsing '/etc/asterisk/users.conf': Found
> [Feb 21 03:29:15] NOTICE[2563]: pbx_ael.c:4094 pbx_load_module:
> Starting AEL load process.
> [Feb 21 03:29:15] NOTICE[2563]: pbx_ael.c:4101 pbx_load_module: AEL
> load process: calculated config file name
> '/etc/asterisk/extensions.ael'.
> etc.
>
> How can I go and trim things down?
>
> Thank you.
>
>
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Re: [asterisk-users] Pattern matching....

[Steve Murphy]

On Thu, 2008-02-21 at 12:44 -0500, Michael Munger wrote:
> Will this work to match any number from the 770,404, or 678 area
> codes? 
> 
>  
> 
> _[404|770|678]NXX
> 
>  
> 
> If this won’t work, is there a pattern that will do this?
> 
>  

No, it won't work, there's no '|' for alternative matches, and no parens
available for grouping, either. And your usage of char sets is off.
Try something like this:

_404NXX

_770NXX

_678NXX


as three separate extensions.

If you REALLY want to keep that as one extension, then you could:

_NXXNXX  =>  {
Set(areacode=${EXTEN:0:3})
if ('${areacode}' = '404') {

} else if ('${areacode}' = '770') {

} else if ('${areacode}' = '678') {

}


OR, you could do it this way, also:

_NXXNXX  =>  {
Set(areacode=${EXTEN:0:3})
switch(${areacode})
{
case 404:

break;
case 770:

break;
case 678:

break;
}

This is, of course AEL code, and this stuff would be inside a context
construct...

murf






> 
> 
> Yours,
> 
> Michael Munger, dCAP
> 
> 404-438-2128
> 
> [EMAIL PROTECTED]
> 
>  
> 
> Attachment encrypted? click here.
> 
>  
> 
> 
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-- 
Steve Murphy
Software Developer
Digium


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Re: [asterisk-users] IVR No sound on other provider

[Ron]

, "0?continue") in new stack
-- Executing Set("SIP/23456789-0821de80", "__TTL=64") in new stack
-- Executing GotoIf("SIP/23456789-0821de80", "1?continue") in new stack
-- Goto (macro-user-callerid,s,23)
-- Executing NoOp("SIP/23456789-0821de80", "Using CallerID "98765432" 
<98765432>") in new stack
-- Executing Set("SIP/23456789-0821de80", "FROMCONTEXT=exten-vm") in new 
stack
-- Executing Set("SIP/23456789-0821de80", "VMBOX=novm") in new stack
-- Executing Set("SIP/23456789-0821de80", "EXTTOCALL=300") in new stack
-- Executing Set("SIP/23456789-0821de80", "CFUEXT=") in new stack
-- Executing Set("SIP/23456789-0821de80", "CFBEXT=") in new stack
-- Executing Set("SIP/23456789-0821de80", "RT=") in new stack
-- Executing Macro("SIP/23456789-0821de80", "record-enable|300|IN") in new 
stack
-- Executing GotoIf("SIP/23456789-0821de80", "0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing AGI("SIP/23456789-0821de80", 
"recordingcheck|20080221-175654|1203645383.3") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
  recordingcheck|20080221-175654|1203645383.3: Inbound recording not enabled
-- AGI Script recordingcheck completed, returning 0
-- Executing NoOp("SIP/23456789-0821de80", "No recording needed") in new 
stack
-- Executing Macro("SIP/23456789-0821de80", "dial||tr|300") in new stack
-- Executing GotoIf("SIP/23456789-0821de80", "1?dial") in new stack
-- Goto (macro-dial,s,3)
-- Executing AGI("SIP/23456789-0821de80", "dialparties.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
  dialparties.agi: Starting New Dialparties.agi
  == Manager 'admin' logged on from 127.0.0.1
  dialparties.agi: Caller ID name is '98765432' number is '98765432'
  dialparties.agi: Methodology of ring is  'none'
--  dialparties.agi: Added extension 300 to extension map
--  dialparties.agi: Extension 300 cf is disabled
--  dialparties.agi: Extension 300 do not disturb is disabled
--  dialparties.agi: dbset CALLTRACE/300 to 98765432
  == Manager 'admin' logged off from 127.0.0.1
-- AGI Script dialparties.agi completed, returning 0
-- Executing Dial("SIP/23456789-0821de80", "SIP/300||tr") in new stack
Destroying call '[EMAIL PROTECTED]'
  == Everyone is busy/congested at this time (1:0/0/1)
-- Executing Set("SIP/23456789-0821de80", "DIALSTATUS=CHANUNAVAIL") in new 
stack
-- Executing Set("SIP/23456789-0821de80", "SV_DIALSTATUS=CHANUNAVAIL") in 
new stack
-- Executing GosubIf("SIP/23456789-0821de80", "0?docfu|1") in new stack
-- Executing GosubIf("SIP/23456789-0821de80", "0?docfb|1") in new stack
-- Executing Set("SIP/23456789-0821de80", "DIALSTATUS=CHANUNAVAIL") in new 
stack
-- Executing NoOp("SIP/23456789-0821de80", "Voicemail is novm") in new stack
-- Executing GotoIf("SIP/23456789-0821de80", "1?s-CHANUNAVAIL|1") in new 
stack
-- Goto (macro-exten-vm,s-CHANUNAVAIL,1)
-- Executing PlayTones("SIP/23456789-0821de80", "congestion") in new stack
-- Executing Congestion("SIP/23456789-0821de80", "10") in new stack
Retransmitting #4 (NAT) to 10.987.654.321:1024:
OPTIONS sip:[EMAIL PROTECTED]:5067 SIP/2.0
Via: SIP/2.0/UDP 123.456.789.10:5060;branch=z9hG4bK0124efd6;rport
From: "Unknown" ;tag=as6740d2f8
To: 
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 22 Feb 2008 01:56:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
Destroying call '[EMAIL PROTECTED]'
<-- SIP read from 111.222.163.170:5060:

---
Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms
12 headers, 0 lines
Reliably Transmitting (NAT) to 10.987.654.321:1024:
OPTIONS sip:[EMAIL PROTECTED]:5067 SIP/2.0
Via: SIP/2.0/UDP 123.456.789.10:5060;branch=z9hG4bK1034e03d;rport
From: "Unknown" ;tag=as53f2521d
To: 
Contact: 
Call-ID: [EMAIL PROTECTED]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 22 Feb 2008 01:57:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0

---
  == Spawn extension (macro-exten-vm, s-CHANUNAVAIL, 2) exited non-zero on 
'SIP/23456789-0821de80' in macro 'exten-vm'
  == Spawn extension (macro-exten-vm, s-CHANUNAVAIL, 2) exited non-zero on 
'SIP/23456789-0821de80'
Scheduling destruction of call '[EMAIL PROTECTED]' in 32000 ms
set_destination: Parsing  for 
address/port to send to
set_destination: set destination to 111.222.31.243, port 5060
Reliably Transmitting (no NAT) to 111.222.31.243:5060:
BYE sip:[EMAIL PROTECTED]:6889 SIP/2.0
Via: SIP/2.0/UDP 123.456.789.10:5060;branch=z9hG4bK49a2be58;rport
Route: 
From: ;tag=as21820660
To: ;tag=1808342796
Call-ID: [EMAIL PROTECTED]
CSeq: 102 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
sipc*CLI>
<-- SIP read from 111.222.31.243:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 123.456.789.10:5060;branch=z9hG4bK49a2be58;rport=5060
From: ;tag=as21820660
To: ;tag=1808342796
Call-ID: [EMAIL PROTECTED]
CSeq: 102 BYE
User-Agent: sip-gw V 3.1.2 070924
Content-Length: 0


--- (8 headers 0 lines) ---
sipc*CLI>
<-- SIP read from 111.222.31.243:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 123.456.789.10:5060;branch=z9hG4bK49a2be58;rport=5060
From: ;tag=as21820660
To: ;tag=1808342796
Call-ID: [EMAIL PROTECTED]
CSeq: 102 BYE
User-Agent: sip-gw V 3.1.2 070924
Content-Length: 0


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Re: [asterisk-users] Which echo-can for Digium B410P ?

[Darren Wright]

The HWEC, not software.
 
-Darren
 



From: [EMAIL PROTECTED] on behalf of Olivier
Sent: Thu 2/21/2008 12:37 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Which echo-can for Digium B410P ?


Hi,

Which echo-canceler shall I pick for Digium B410P ? 

Is HPEC relevant ? Reading from its datasheet, it seems related to analog cards.
Regards


This message was sent from D2 Technology, INC.

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Re: [asterisk-users] Coppercom and Asterisk

[Mike Hammett]

I put that in, but it appears that it is trying to contact the private IP 
address of their SIP server.  I have successfully registered to this server 
from over the public Internet using an Innomedia ATA.

[Feb 21 11:49:18] NOTICE[4608]: chan_sip.c:7364 sip_reg_timeout:--  
Registration for '[EMAIL PROTECTED]' timed out, trying again 
(Attempt #1)
REGISTER 12 headers, 0 lines
Reliably Transmitting (no NAT) to 10.1.3.2:5060:
REGISTER sip:sip.essex1.com SIP/2.0
Via: SIP/2.0/UDP 10.1.5.5:5060;branch=z9hG4bK5ddcc06a;rport
From: ;tag=as58a684a6
To: 
Call-ID: [EMAIL PROTECTED]
CSeq: 103 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Expires: 120
Contact: 
Event: registration
Content-Length: 0


---
Really destroying SIP dialog '[EMAIL PROTECTED]' 
Method: REGISTER
Retransmitting #1 (no NAT) to 10.1.3.2:5060:
REGISTER sip:sip.essex1.com SIP/2.0
Via: SIP/2.0/UDP 10.1.5.5:5060;branch=z9hG4bK5ddcc06a;rport
From: ;tag=as58a684a6
To: 
Call-ID: [EMAIL PROTECTED]
CSeq: 103 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Expires: 120
Contact: 
Event: registration
Content-Length: 0


---
Retransmitting #2 (no NAT) to 10.1.3.2:5060:
REGISTER sip:sip.essex1.com SIP/2.0
Via: SIP/2.0/UDP 10.1.5.5:5060;branch=z9hG4bK5ddcc06a;rport
From: ;tag=as58a684a6
To: 
Call-ID: [EMAIL PROTECTED]
CSeq: 103 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Expires: 120
Contact: 
Event: registration
Content-Length: 0


---
Retransmitting #3 (no NAT) to 10.1.3.2:5060:
REGISTER sip:sip.essex1.com SIP/2.0
Via: SIP/2.0/UDP 10.1.5.5:5060;branch=z9hG4bK5ddcc06a;rport
From: ;tag=as58a684a6
To: 
Call-ID: [EMAIL PROTECTED]
CSeq: 103 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Expires: 120
Contact: 
Event: registration
Content-Length: 0


---
Really destroying SIP dialog '[EMAIL PROTECTED]' 
Method: REGISTER
Retransmitting #4 (no NAT) to 10.1.3.2:5060:
REGISTER sip:sip.essex1.com SIP/2.0
Via: SIP/2.0/UDP 10.1.5.5:5060;branch=z9hG4bK5ddcc06a;rport
From: ;tag=as58a684a6
To: 
Call-ID: [EMAIL PROTECTED]
CSeq: 103 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Expires: 120
Contact: 
Event: registration
Content-Length: 0


---
Aiur*CLI>



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


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

Sent: Thursday, February 21, 2008 1:53 AM
Subject: Re: [asterisk-users] Coppercom and Asterisk


> In the [general] section, put:
>
> register => 8159093010:[EMAIL PROTECTED]
>
> Then add a SIP peer for the outbound proxy.  Something like:
>
> [essex1_outbound]
>
> fromdomain=proxy.essex1.com
> host=proxy.essex1.com
> port=5060
> insecure=very
> username=8159093010
> secret=X
> type=peer
> qualify=no
> canreinvite=no
> dtmfmode=rfc2833
> disallow=all
> allow=ulaw
>
> The first one is needed for the registrations, and the second one is
> needed to answer 407 proxy challenges.
>
> Mike Hammett wrote:
>> My provider has a Coppercom switch.  I have included the authentication
>> information they gave me.  How would I structure this in Asterisk to the
>> registration and the entry in sip.conf?
>>
>> User Name - 8159093010
>> Password - X
>> No Pin
>> Proxy - sip.essex1.com (10.1.3.2)
>> Outbound Proxy - proxy.essex1.com (63.164.210.14)
>> Change setting to use "outbound Proxy"
>>
>>
>> --
>> Mike Hammett
>> Intelligent Computing Solutions
>> http://www.ics-il.com
>>
>>
>>
>>
>> 
>>
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>>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> -- 
> Alex Balashov
> Evariste Systems
> Web: http://www.evaristesys.com/
> Tel: (+1) (678) 954-0670
> Direct : (+1) (678) 954-0671
> Mobile : (+1) (706) 338-8599
>
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Re: [asterisk-users] Zaptel 1.4.8 breaks tor2 support on CentOS 5.1? (kernel panic)

[Kevin P. Fleming]

Kevin P. Fleming wrote:

> I've just located an E400P from our graveyard of old cards... if it
> works, I'll be able to solve this problem in the morning.

This has been fixed in revision 3863 of the 1.4 branch; it's a one line
fix that you should be able to easily apply to existing Zaptel source
code, or you can wait for the next Zaptel release which should happen
later today. Sorry for the breakage.

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

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Re: [asterisk-users] Answered Call marked as "NO ANSWER"

[Jorge Mendoza]

Raúl,

From your conf file I guess the CO provide reversal polarity for answer
supervision. Verify if for those "numbers" the CO revert the line
polarity when callee answer.
callprogress=no is a good test too.

Jorge

Raúl Gómez C. wrote:
> Hi list,
>
> I'm having problems transferring certain calls made by the attendant
> between the PSTN and to an internal extension. Although, transfers
> between the majority of the calls ends successfully.
>
> Debugin this, I've found that calls made to certain "numbers"
> (Telephony Providers), aren't detected as ANSWERED in the CDR, so they
> are not properly accounted (for billing), neither transferred to
> internals extensions.
>
> How can I solve this??? Is this a incompatibility issue between
> technologies??? Or just a config that I haven't made right???
>
> Thanks in advance...
>
>
> My Setup:
>
> - Asterisk 1.4.17
> - Sangoma Remora A400D HEC PCI Card (2 FXS / 10 FXO)
> - Wanpipe 3.2.1
> - Zaptel *MailScanner warning: numerical links are often malicious:*
> 1.4.7.1 
> - Grandstream GXP-2000 Phones
>
>
> =
> *zaptel.conf*
> /# Autogenerated by /usr/local/sbin/sangoma/setup-sangoma -- do not
> hand edit
> # Zaptel Channels Configurations (zaptel.conf)
> #
> loadzone=us
> defaultzone=us
>
> #Sangoma A400 [slot:4 bus:16 span:1]
> fxoks=1
> fxoks=2
> fxsks=3
> fxsks=4
> fxsks=5
> fxsks=6
> fxsks=7
> fxsks=8
> fxsks=9
> fxsks=10
> fxsks=11
> fxsks=12/
>
>
> =
> *zapata.conf*
> /;autogenerated by /usr/local/sbin/config-zaptel  do not hand edit
> ;Zaptel Channels Configurations (zapata.conf)
> ;
> ;For detailed zapata options, view /etc/asterisk/zapata.conf.orig
>
> [trunkgroups]
>
> [channels]
> context=default
> ;usecallerid=yes
> ;hidecallerid=no
> callwaiting=yes
> usecallingpres=yes
> ;callwaitingcallerid=yes
> threewaycalling=yes
> transfer=yes
> canpark=yes
> cancallforward=yes
> callreturn=yes
> echocancel=yes
> echocancelwhenbridged=yes
> relaxdtmf=yes
> rxgain=0.0
> txgain=0.0
> group=0
> callgroup=0
> pickupgroup=1
>
> callerid="Llamada Externa"
> busydetect=yes
> busycount=4
> callprogress=yes
> progzone=us
> hanguponpolarityswitch=yes
>
> immediate=no
>
> ;Sangoma A400 [slot:4 bus:16 span:1]
> context=watch
> group=1
> signalling = fxo_ks
> channel => 1
>
> context=fax
> group=1
> signalling = fxo_ks
> channel => 2
>
> context=from-zaptel
> group=0
> signalling = fxs_ks
> channel => 3
>
> context=from-zaptel
> group=0
> signalling = fxs_ks
> channel => 4
>
> context=from-zaptel
> group=0
> signalling = fxs_ks
> channel => 5
>
> context=from-zaptel
> group=0
> signalling = fxs_ks
> channel => 6
>
> context=from-zaptel
> group=2
> signalling = fxs_ks
> channel => 7
>
> context=from-zaptel
> group=2
> signalling = fxs_ks
> channel => 8
>
> context=from-zaptel
> group=3
> signalling = fxs_ks
> channel => 9
>
> context=from-zaptel
> group=4
> signalling = fxs_ks
> channel => 10
>
> context=from-zaptel
> group=5
> signalling = fxs_ks
> channel => 11
>
> context=from-zaptel
> group=6
> signalling = fxs_ks
> channel => 12/
>
>
> -- 
> Nacho
> Linux Counter #156439
> 
>
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Re: [asterisk-users] IVR No sound on other provider

[Steve Totaro]

On Thu, Feb 21, 2008 at 12:40 PM, Ron <[EMAIL PROTECTED]> wrote:
> Hi All,
>
>  I have setup 2 trunks using 2 different voip providers using sip.
>
>  the first one i have no problem calling inbound then redirected to an
>  IVR, i can hear the IVR.
>
>  the second one has issues, inbound works going to IVR as i can see it on
>  the CLI, but i don't hear anything. i tried redirecting it to an
>  extension not an IVR just to see if inbound really works, and it rings
>  the extensions directly and i can here myself talking both ways. is the
>  issue maybe incompatible audio format?
>
>  i am using freepbx, and i recorded the ivr using the IP phone (by
>  dialing *77), my provider says they are only using codecs iLBC and
>  g711a. is that the reason why i cant hear the ivr? where do i start
>  checking it?
>
>  thank you
>
>  regards,
>
>  Ron

Post your SIP debug and verbose 3 for the call that does not have IVR.

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[asterisk-users] Pattern matching....

[Michael Munger]

Will this work to match any number from the 770,404, or 678 area codes? 

 

_[404|770|678]NXX

 

If this won't work, is there a pattern that will do this?

 

Yours,

Michael Munger, dCAP

404-438-2128

[EMAIL PROTECTED]  

 

Attachment encrypted? click here
 .

 

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[asterisk-users] IVR No sound on other provider

[Ron]

Hi All,

I have setup 2 trunks using 2 different voip providers using sip.

the first one i have no problem calling inbound then redirected to an 
IVR, i can hear the IVR.

the second one has issues, inbound works going to IVR as i can see it on 
the CLI, but i don't hear anything. i tried redirecting it to an 
extension not an IVR just to see if inbound really works, and it rings 
the extensions directly and i can here myself talking both ways. is the 
issue maybe incompatible audio format?

i am using freepbx, and i recorded the ivr using the IP phone (by 
dialing *77), my provider says they are only using codecs iLBC and 
g711a. is that the reason why i cant hear the ivr? where do i start 
checking it?

thank you

regards,

Ron

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[asterisk-users] Which echo-can for Digium B410P ?

[Olivier]

Hi,

Which echo-canceler shall I pick for Digium B410P ?

Is HPEC relevant ? Reading from its datasheet, it seems related to analog
cards.
Regards
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Re: [asterisk-users] interactive menu with DTMF tones

[Tilghman Lesher]

On Thursday 21 February 2008 08:44:11 John Von Essen wrote:
> This may be a dumb question, but I have never done menus, how do I link
> the below up to my phone number? For example, right now I route calls
> to the SIP phone like so:
>
> [ipcomms]
> include = default
> exten => 211212, 1, Dial(SIP/6000,20,tr)
>
> where ipcomms is my context from sip.conf for origination, and 6000 is
> my sip phone. So how do I connect that 215 number to the below menu
> system?

Use a Goto:

Goto(s,1) or Goto(menucontext,s,1)

-- 
Tilghman

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Re: [asterisk-users] Best ATA. Period.

[lists]

Any Linksys ATA is best value for money. I have used SPA AND PAP 2 series.
 I have also CISCO ATA in office and still no problems for almost 2 years.

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anthony Francis
Sent: Thursday, February 21, 2008 5:19 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Best ATA. Period.

SIP wrote:
> Adam Moffett wrote:
>   
>> In all seriousness, my requirements were a little silly.  A Cisco router 
>> can fail just as a netgear router can.  But I think we would find Cisco 
>> failures to be statistically less likely.
>>
>> I also think we can agree that not all devices of a certain type are 
>> created equal.  Do you have any opinions on which VoIP products are more 
>> likely to be consistent and reliable?
>>
>>   
>> 
> Realistically, I've had issues with every ATA I've used to SOME degree. 
> The Leadtek BVA series has numerous issues. I've had bizarre things 
> occur in all of my Linksys/Sipura adapters(2000,3000,3201) (issues with 
> timeouts on a lost connection, NAT traversal, etc).  My Grandstream 
> HT486 and 488s have intermittent dialing failures. I've had a lot of 
> issues with the Audiocodes MPs.
>
> The only ATA I've NOT actually had any issues with has been my 
> Grandstream HT386. Granted, I have issues with its capabilities overall, 
> but on the whole, it's the only one that's not simply had some weird 
> random failure as the others have.
>
> Does this mean I'd recommend an HT386 as solid testing piece? Heavens 
> no. I'd probably recommend the Linksys SPA3102.  But be aware that there 
> ARE issues with just about all of them, and I doubt there have even been 
> enough variants sold/used by everyone to merit statistical analyses. 
> What you're going to get for recommendations will be, at best, anecdotal.
>
> N.
>
>   
I have had good luck with Cisco's ATA's.

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[asterisk-users] Maybe OT: SIP - Missing 407 messages

[Kristian Kielhofner]

Hello everyone,

  I have many Asterisk clients registered to an OpenSER proxy.
Sometimes (for reasons unknown) the 407 Proxy Authentication Required
sent by OpenSER to Asterisk is not received by Asterisk on the client,
causing the call to fail.  No other SIP messages or other IP traffic
seems to be lost (i.e. no packet loss).  We can confirm (via a SPAN
session on the switch) that the 407 makes it to the wire.  We can
confirm (via ngrep/tcpdump) that the 407 is never received by
Asterisk.  At this point I don't think it's an Asterisk or OpenSER
problem but it's so strange I can't be certain.  I've already checked
everything out on OpenSER with Bogdan and he says it looks good.  I
believe him :).

  If anything, I'd like to try the SIP TCP support in Asterisk 1.6 and
see if that makes a difference.  I hope so...

  I've done a full writeup on my blog (took advantage of HTML in a
couple of places) with a description of the problem.  If you think
something is relevant, feel free to copy and paste it into your reply
for completeness in the archives:

ttph://blog.krisk.org/2008/02/missing-sip-traffic.html

Thanks!

-- 
Kristian Kielhofner

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[asterisk-users] Answered Call marked as "NO ANSWER"

[Raúl Gómez C.]

Hi list,

I'm having problems transferring certain calls made by the attendant between
the PSTN and to an internal extension. Although, transfers between the
majority of the calls ends successfully.

Debugin this, I've found that calls made to certain "numbers" (Telephony
Providers), aren't detected as ANSWERED in the CDR, so they are not properly
accounted (for billing), neither transferred to internals extensions.

How can I solve this??? Is this a incompatibility issue between
technologies??? Or just a config that I haven't made right???

Thanks in advance...


My Setup:

- Asterisk 1.4.17
- Sangoma Remora A400D HEC PCI Card (2 FXS / 10 FXO)
- Wanpipe 3.2.1
- Zaptel 1.4.7.1
- Grandstream GXP-2000 Phones


=
*zaptel.conf*
*# Autogenerated by /usr/local/sbin/sangoma/setup-sangoma -- do not hand
edit
# Zaptel Channels Configurations (zaptel.conf)
#
loadzone=us
defaultzone=us

#Sangoma A400 [slot:4 bus:16 span:1]
fxoks=1
fxoks=2
fxsks=3
fxsks=4
fxsks=5
fxsks=6
fxsks=7
fxsks=8
fxsks=9
fxsks=10
fxsks=11
fxsks=12*


=
*zapata.conf*
*;autogenerated by /usr/local/sbin/config-zaptel  do not hand edit
;Zaptel Channels Configurations (zapata.conf)
;
;For detailed zapata options, view /etc/asterisk/zapata.conf.orig

[trunkgroups]

[channels]
context=default
;usecallerid=yes
;hidecallerid=no
callwaiting=yes
usecallingpres=yes
;callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
relaxdtmf=yes
rxgain=0.0
txgain=0.0
group=0
callgroup=0
pickupgroup=1

callerid="Llamada Externa"
busydetect=yes
busycount=4
callprogress=yes
progzone=us
hanguponpolarityswitch=yes

immediate=no

;Sangoma A400 [slot:4 bus:16 span:1]
context=watch
group=1
signalling = fxo_ks
channel => 1

context=fax
group=1
signalling = fxo_ks
channel => 2

context=from-zaptel
group=0
signalling = fxs_ks
channel => 3

context=from-zaptel
group=0
signalling = fxs_ks
channel => 4

context=from-zaptel
group=0
signalling = fxs_ks
channel => 5

context=from-zaptel
group=0
signalling = fxs_ks
channel => 6

context=from-zaptel
group=2
signalling = fxs_ks
channel => 7

context=from-zaptel
group=2
signalling = fxs_ks
channel => 8

context=from-zaptel
group=3
signalling = fxs_ks
channel => 9

context=from-zaptel
group=4
signalling = fxs_ks
channel => 10

context=from-zaptel
group=5
signalling = fxs_ks
channel => 11

context=from-zaptel
group=6
signalling = fxs_ks
channel => 12*


-- 
Nacho
Linux Counter #156439
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[asterisk-users] SendDTMF not Working - Possible Echo Cancelling Issues

[Jake Wicke]

I am having issues with SendDTMF on an Asterisk box using Asterisk 1.4.18 and 
Zaptel 1.4.8.  As far as I can see, the issues seem to be related to echo 
cancellation.  The box has a TDM2400p installed and is using the HPEC echo 
canceller.

The problems occur when I attempt to do an outcall.  The outcall dials a pager 
number, then connects the dialed call to a context in my extensions.conf which 
sends a number of DTMF digits to the pager number.

I have changed the pager number to my telephone number and can hear a large 
amount of variance in the DTMF digits. I believe this variance is what is 
causing the problem when the pager is dialed.

I have attempted to change the toneduration in zaptel.conf - this setting seems 
to not work at all - any changes made to this setting make no difference in the 
duration of the tone sent over the channel.  The output of the script seems to 
be better (the DTMF is more clear) when echocancelwhenbridged is set to yes.

How do I change the DTMF duration on a zap channel using SendDTMF() as 
toneduration does nothing?  Is there any way to disable echo cancellation for a 
single call?

Jake Wicke





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Re: [asterisk-users] Best ATA. Period.

[Anthony Francis]

SIP wrote:
> Adam Moffett wrote:
>   
>> In all seriousness, my requirements were a little silly.  A Cisco router 
>> can fail just as a netgear router can.  But I think we would find Cisco 
>> failures to be statistically less likely.
>>
>> I also think we can agree that not all devices of a certain type are 
>> created equal.  Do you have any opinions on which VoIP products are more 
>> likely to be consistent and reliable?
>>
>>   
>> 
> Realistically, I've had issues with every ATA I've used to SOME degree. 
> The Leadtek BVA series has numerous issues. I've had bizarre things 
> occur in all of my Linksys/Sipura adapters(2000,3000,3201) (issues with 
> timeouts on a lost connection, NAT traversal, etc).  My Grandstream 
> HT486 and 488s have intermittent dialing failures. I've had a lot of 
> issues with the Audiocodes MPs.
>
> The only ATA I've NOT actually had any issues with has been my 
> Grandstream HT386. Granted, I have issues with its capabilities overall, 
> but on the whole, it's the only one that's not simply had some weird 
> random failure as the others have.
>
> Does this mean I'd recommend an HT386 as solid testing piece? Heavens 
> no. I'd probably recommend the Linksys SPA3102.  But be aware that there 
> ARE issues with just about all of them, and I doubt there have even been 
> enough variants sold/used by everyone to merit statistical analyses. 
> What you're going to get for recommendations will be, at best, anecdotal.
>
> N.
>
>   
I have had good luck with Cisco's ATA's.

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[asterisk-users] HoldMusic Beep

[Forrest Beck]

Does anyone have a audio file they would be willing to share for on hold
music?
I am looking for something like the old norstar beep every few seconds.

I tried 3 seconds silence, beep.wav, beep.wav.  But it just didn't sound
right.  I need one that has a "softer" beep.

Thanks!

-- 
***
Forrest Beck
IAXTEL: 17002871718
[EMAIL PROTECTED]
http://www.shift8.biz
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Re: [asterisk-users] T1 Timing Troubleshooting

[Mark Greene]

Jon, did you ever discover a solution to your problem. I'm in the same boat.

On Sun, Dec 2, 2007 at 1:22 PM, Jonathan C. Bailey <
[EMAIL PROTECTED]> wrote:

> I'm having (I think) timing issues in relation to bridged T1-T1 calls via
> dynamic spans. Fax calls are intermittently working, but voice is fine. My
> box has a Sangoma A400 inside it as the primary Zaptel timing source. My T1
> PRIs that are hooked to the box come in via a foneBRIDGE2 (dynamic TDMoE
> spans). PRI #1 is the telco and PRI #2 is an existing Comdial FX-II. For
> some reason, bridged TDM calls (when it comes to faxing) must be having
> timing issues since they intermittently fail.
>
> I found what seems to be an issue in zaptel.conf (timing source for the
> Comdial side was 2 - changed to 0), but I don't know if that's it. I've also
> turned off echo cancellation. Any other thoughts on why I may be having what
> seem to be timing issues? Also, is timing passed through on dynamic spans &
> bridged calls? And is there a way to verify this? Thanks!
>
>
> -
>
> /etc/zaptel.conf (16 channels on each PRI):
> loadzone=us
> defaultzone=us
>
> #Sangoma A400 [slot:7 bus:1 span:1]
> fxsks=1
> fxsks=2
> fxsks=3
> fxsks=4
> fxsks=5
> fxsks=6
> fxsks=7
> fxsks=8
> fxoks=11
> fxoks=12
>
> dynamic=eth,eth1/00:50:c2:65:d0:3c/0,24,1
> dynamic=eth,eth1/00:50:c2:65:d0:3c/1,24,0
> # bchan=25-47
> bchan=25-40
> dchan=48
> # bchan=49-71
> bchan=49-64
> dchan=72
>
>
> -
> /etc/asterisk/zapata.conf:
>
> [trunkgroups]
>
>
> [channels]
> context=default
> usecallerid=yes
> hidecallerid=no
> callwaiting=yes
> usecallingpres=yes
> callwaitingcallerid=yes
> threewaycalling=yes
> transfer=yes
> canpark=yes
> cancallforward=yes
> callreturn=yes
> ; Turned echo cancellation off 11-15-2007 due to possible fax issues on
> bridged calls.
> echocancel=no
> faxdetect=no
> echocancelwhenbridged=no
> rxgain=0.0
> txgain=0.0
> callgroup=1
> pickupgroup=1
> immediate=no
> overlapdial=yes
>
> ;Sangoma A400 [slot:7 bus:1 span:1]
> context=from-zaptel
> group=0
> signalling = fxs_ks
> channel => 1-8
>
> context=from-internal
> group=1
> signalling = fxo_ks
> channel => 11-12
>
> ; First port on foneBRIDGE2 - This is the PSTN side
> group=2
> signalling = pri_cpe
> context=from-pstn
> ;channel => 25-47 (for a full PRI)
> ; Channels 25-40 are for a partial PRI (16 channels)
> channel => 25-40
>
> ; Second port on foneBRIDGE2 - This is the Comdial side
> group=3
> context=from-comdial
> signalling = pri_net
> ;channel => 49-71 (for a full PRI)
> ; Channels 49-64 are for a partial PRI (16 channels)
> channel => 49-64
>
>
>
>
>
> -Jon
>
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Re: [asterisk-users] chan_h323 requirements

[Vlasis Hatzistavrou (KTI)]

Hello Bruce,

Bruce McAlister wrote:
> 
> Did your patch for building with OpenH323+ make it into the 1.4 edition 
> of Asterisk?
> 

No, it didn't as it was considered a new feature and by Digium's policy 
new features can only be added in the trunk versions.

The strange thing is that I added it in trunk version, too, but it 
didn't make it in the upcoming 1.6 version either.

Best regards,
Vlasis Hatzistavrou


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Re: [asterisk-users] Contents of asterisk-users digest

[garry liu]

Re: Contents of asterisk-users digest
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Re: [asterisk-users] Converence/Meetme with Manager API

[Lee Jenkins]

Mitchell Jackson wrote:
> Hello!  I am having problems figuring out how to do something, and any 
> help would be much appreciated.
> 
> I would like to use the manager API to take an existing call on a 
> specific SIP extension, dial and conference in a third party.
> 
>  From what I can tell, the way to do this would be to take the two 
> original parties on the call and stick them in a meetme room using 
> Redirect with ExtraChannel, then dial the new party and also dump them 
> into the meetme room.
> 
> The problem I am having is this:  I know the extension of the SIP phone 
> that is on the call, but I don't know it's channel, or the channel of 
> the other party.  I need to figure both of these out to be able to use 
> the Manager API and dump those callers into the meetme room.
> 
> Can anybody tell me how to determine the channels on an active call?
> 
> Kind Regards,
> 

You need to track those calls somehow, Mitch.

Someone can correct me where I'm wrong, but I see you can do this in a couple 
of 
ways.

1. Track the status of peers.  My application performs a sippeers manager (and 
zapshowchannels) command to get the status of each device I'm watching at start 
up.  As events are sent from AMI, I match each device with that event, 
specifically, the "LINK" event (changed to "Bridge" event in AMI 1.1).  This 
way, when the user goes to click on or drag and drop a device on screen, we 
already know its information such as its channel info and linked channel 
information.

2. Another way I can think of would be to use the CLI command "show channels" 
from AMI and parse the output for your device.  After figuring out which one is 
the device you're interested in, you can use the "Status" manager AMI command 
to 
get the info (including linked channel on the device).  As you probably figured 
out, the "Status" command requires the channel of the device and not just its 
name/ident such as "sip/114" so you have to go through the "Show Channels" hoop 
first, I imagine.

As you say, its the easiest to just "redirect" both parties to an extension 
already setup in your extensions.conf.  I also "push" channel variables from my 
application to Asterisk channel vars for use in the dialplan.  This way I can 
have a bit of dynamic operations.  If my user want to create a new conference 
by 
dragging a "live" sip phone to the conference view of my application, I just 
prompt the user for conference number, send it as a var along with my redirect 
request to AMI and use dialplan logic from there.

As I said, I'm still learning (although learning a lot!) about AMI operations 
as 
I build my own application for AMI so take my info with a minuscule portion of 
sodium. ;)




-- 
Warm Regards,

Lee

"Everything I needed to learn in life, I learned selling encyclopedias door to 
door."

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Re: [asterisk-users] interactive menu with DTMF tones

[John Von Essen]

This may be a dumb question, but I have never done menus, how do I link 
the below up to my phone number? For example, right now I route calls 
to the SIP phone like so:

[ipcomms]
include = default
exten => 211212, 1, Dial(SIP/6000,20,tr)

where ipcomms is my context from sip.conf for origination, and 6000 is 
my sip phone. So how do I connect that 215 number to the below menu 
system?

Thanks
John

On Feb 19, 2008, at 1:25 AM, Tilghman Lesher wrote:
>
> extensions.conf:
> exten => s,1,Answer
> exten => s,n,Read(account,account-prompt)
> exten => s,n,Read(passcode,passcode-prompt)
> exten => s,n,GotoIf(${ODBC_LOOKUP(${account},${passcode})}?granted)
> exten => s,n,Playback(denied)
> exten => s,n,Hangup
> exten => s,n(granted),SendDTMF(123456789)
> exten => s,n,Hangup
>
> func_odbc.conf:
> [LOOKUP]
> dsn=mysql-asterisk
> read=SELECT COUNT(*) FROM passcode_table WHERE account='${ARG1}' AND
> passcode='${ARG2}'
>
> -- 
> Tilghman
>
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Re: [asterisk-users] Best ATA. Period.

[Tim Johnson]

Quoting SIP <[EMAIL PROTECTED]>:

> Adam Moffett wrote:
>> In all seriousness, my requirements were a little silly.  A Cisco router
>> can fail just as a netgear router can.  But I think we would find Cisco
>> failures to be statistically less likely.
>>
>> I also think we can agree that not all devices of a certain type are
>> created equal.  Do you have any opinions on which VoIP products are more
>> likely to be consistent and reliable?
>>
>>
> Realistically, I've had issues with every ATA I've used to SOME degree.
> The Leadtek BVA series has numerous issues. I've had bizarre things
> occur in all of my Linksys/Sipura adapters(2000,3000,3201) (issues with
> timeouts on a lost connection, NAT traversal, etc).  My Grandstream
> HT486 and 488s have intermittent dialing failures. I've had a lot of
> issues with the Audiocodes MPs.
>
> The only ATA I've NOT actually had any issues with has been my
> Grandstream HT386. Granted, I have issues with its capabilities overall,
> but on the whole, it's the only one that's not simply had some weird
> random failure as the others have.
>
> Does this mean I'd recommend an HT386 as solid testing piece? Heavens
> no. I'd probably recommend the Linksys SPA3102.  But be aware that there
> ARE issues with just about all of them, and I doubt there have even been
> enough variants sold/used by everyone to merit statistical analyses.
> What you're going to get for recommendations will be, at best, anecdotal.
>
> N.
>

I have two SPA3102s, and two PAP2-NAs. All of them have worked  
flawlessly. I have my SPA3102s on a real IP, and have tested the  
PAP2's from real IPs and behind nat. They do seem to "just work" after  
being configured. My only complaint is with Linksys/Cisco's support,  
or lack of it.

Tim

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Re: [asterisk-users] Asterisk Nagios

[lordfuknowsyou]

Al lists wrote:
> Has anyone checked asterisk with check_udp plug in?
>
> 
>
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No, but I do use the check_sip .

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[asterisk-users] USB ISDN interface

[Michael Blood]

I have been looking for one of the many USB to ISDN products listed on this
page,  
http://www.nslu2-linux.org/wiki/OpenSlug/Asterisk
But I have not been able to find a single item for sale anywhere in the US,
has anyone seen where I can find a couple of these?
 
Thanks
 
Michael
 
 
 
 
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[asterisk-users] Allow INVITE for hold to pass through

[Mayur]

Hi,

   I would like to configure asterisk to allow INVITE for hold to pass
through it and not provide music on hold by itself. Can anyone help me out
here?

Regards,

Mayur

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Re: [asterisk-users] Voted most stable and easy to use phone?

[C F]

since if this email might contain confidential info, in which case I
must not review it. I can't really reaed it since it implies that I
read the disclaimer before the body of the email, and as such as long
as the disclaimer is in it I must NEVER Aread the email.

On 2/21/08, Michael J. Liberatore <[EMAIL PROTECTED]> wrote:
> A while back i had asked about possible replacements for snom 360 phones
> that were breaking and causing issues and we all discussed the problems
> we had with the 360s and some suggestions were made but the  new polycom
> phones had just hit the market and not many people were able to comment
> on them.
>
> Basically i am looking to get some new phones and in the process get rid
> of the countless number of problems i have had that has always been
> caused by phones (snom 360's and gxp-2000's).  I would like to get the
> feedback of the list on the phone voted best for stability, working with
> *, and ease of use for dumb non tech users.
>
> I was thinking of trying one of these new polycom phones that are about
> $150, but havent gotten any feedback on them yet.
>
> Basically i am interested in any phones but snom's, grandstreams, and
> sipura's/linksys.  mainly polycom's i guess.
>
> thanks
>
> mike
>
>
> This E-mail, including any attachments, may be intended solely for
> the personal and confidential use of the sender and recipient(s) named
> above. This message may include advisory, consultative and/or
> deliberative material and, as such, would be privileged and confidential
> and not a public document. Pursuant to 42 CFR, any information in this
> e-mail identifying a former, present, or potential client of Straight &
> Narrow is confidential. If you have received this e-mail in error, you must
> not review, transmit, convert to hard copy, copy, use or disseminate this
> e-mail or any attachments to it and you must delete this message. You are
> requested to notify the sender by return e-mail.
>
>

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Re: [asterisk-users] How to get a clean, basic configuration?

[C F]

first off I anwered you to use vi and you complained showing me cat.
then for your next question about ehat each module does. show module
in asterisk in combination with show application as well as a peak at
the source should give you a clue.
also the module names are quite descriptive.

On 2/21/08, Vincent <[EMAIL PROTECTED]> wrote:
> On Thu, 21 Feb 2008 15:00:15 +1100, Paul Hales
> <[EMAIL PROTECTED]> wrote:
> >Head off into /etc/asterisk/modules.conf and add some 'noload' lines.
>
> Ah, makes sense. Asterisk loads everything, and must be told
> explicitely _not_ to load something :-)
>
> Is there a comprehensive list that explains what each and every module
> does, so that I know what I can safely not load?
>
> Thanks.
>
>
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Re: [asterisk-users] How to get a clean, basic configuration?

[Peder @ NetworkOblivion]

"autoload=yes" says to load everything, so you either need to change it 
to no and then add load statements for every module you need, or leave 
it as yes and then add noload for everything you don't need.


Vincent wrote:
> On Wed, 20 Feb 2008 21:44:30 -0500, "C F" <[EMAIL PROTECTED]> wrote:
>> vi /etc/asterisk/modules.conf
> 
> Thanks, but this file doesn't hold much that's uncommented by default:
> 
> # cat /etc/asterisk/modules.conf
> [modules]
> autoload=yes
> noload => pbx_gtkconsole.so
> noload => pbx_kdeconsole.so
> load => res_musiconhold.so
> noload => chan_alsa.so
> 
> Is this really the only file that Asterisk reads to know what to load?
> 
> 
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Re: [asterisk-users] How to get a clean, basic configuration?

[Vincent]

On Thu, 21 Feb 2008 15:00:15 +1100, Paul Hales
<[EMAIL PROTECTED]> wrote:
>Head off into /etc/asterisk/modules.conf and add some 'noload' lines.

Ah, makes sense. Asterisk loads everything, and must be told
explicitely _not_ to load something :-)

Is there a comprehensive list that explains what each and every module
does, so that I know what I can safely not load?

Thanks.


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Re: [asterisk-users] Best ATA. Period.

[SIP]

Adam Moffett wrote:
> In all seriousness, my requirements were a little silly.  A Cisco router 
> can fail just as a netgear router can.  But I think we would find Cisco 
> failures to be statistically less likely.
>
> I also think we can agree that not all devices of a certain type are 
> created equal.  Do you have any opinions on which VoIP products are more 
> likely to be consistent and reliable?
>
>   
Realistically, I've had issues with every ATA I've used to SOME degree. 
The Leadtek BVA series has numerous issues. I've had bizarre things 
occur in all of my Linksys/Sipura adapters(2000,3000,3201) (issues with 
timeouts on a lost connection, NAT traversal, etc).  My Grandstream 
HT486 and 488s have intermittent dialing failures. I've had a lot of 
issues with the Audiocodes MPs.

The only ATA I've NOT actually had any issues with has been my 
Grandstream HT386. Granted, I have issues with its capabilities overall, 
but on the whole, it's the only one that's not simply had some weird 
random failure as the others have.

Does this mean I'd recommend an HT386 as solid testing piece? Heavens 
no. I'd probably recommend the Linksys SPA3102.  But be aware that there 
ARE issues with just about all of them, and I doubt there have even been 
enough variants sold/used by everyone to merit statistical analyses. 
What you're going to get for recommendations will be, at best, anecdotal.

N.

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[asterisk-users] UCS-2 Problem

[Nasir Iqbal]

Hi List,

Recently I tried sending sms using app_sms (hardware  TDM400P) in Singapore 
with land line telco provider singtel

it worked fine and can send sms in Latin characters 7-bits/8-bits

but I am unable to send Unicode (UCS-2 or 16-bits) sms in Arabic or Chinese. 
the problem is that my mobile show the message with invalid character

I have managed to capture the outgoing sms data as following hex values

91 31 01 00 0C 91 29 33 63 81 06 69 00 08 24 D9 85 D8 B4 D8 B1 D9 81 20 D8 AC 
D8 A7 D8 A6 DB 92 D8 8C 20 D8 AC D8 B3 D9 B9 D8 B3 20 D8 A2 D8 A6 DB 92 0A

and for your quick reference I have segmented the sms data as following 

?91 1001 0001 = ?
len  31   = 48
mix  01  0001 = SMS-SUBMIT
mr   00   = 0
da len   0C   = 12
da type  91 1001 0001 = International number + ISDN/telephone
numbering plan
da   293363810669 = +92333618
pid  00   = 0
dcs  08  1000 = UCS-2
ud len   24   = 36
ud
D985D8B4D8B1D98120D8ACD8A7D8A6DB92D88C20D8ACD8B3D9B9D8B320D8A2D8A6DB920A

as you can see that app_sms sending this message with dsc set to UCS-2 ud HEX 
string is also in correct format (I have tested it with third party web2sms 
service). but it can not be shown on my mobile corectly. so I am unable to 
determince why I am getting invlide charecters instead of a chines Message.

Can you please help me

Thanks


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Re: [asterisk-users] chan_h323 requirements

[Bruce McAlister]

Hi,

Thank you for the details of which versions to get. I will be building 
these two versions on Solaris to test chan_h323.

Did your patch for building with OpenH323+ make it into the 1.4 edition 
of Asterisk?

Thanks
Bruce

Vlasis Hatzistavrou (KTI) wrote:
> Hello,
> 
> To compile chan_h323 as is distributed you need to download OpenH323 
> v1.18.0 and PwLib v1.10.0 from:
> 
> http://www.voxgratia.org
> 
> Some months ago I had made a patch to compile the 1.4.x version and the 
> trunk version (which evolved to 1.6.x) with H323+.
> 
> Sadly, the patch was not included in the 1.6.x version which is being 
> released soon.
> 
> So, for the time being you need to use the above versions from Voxgratia.
> 
> Best regards,
> Vlasis Hatzistavrou.
> 
> Bruce McAlister wrote:
>> Hi All,
>>
>> I would just like to clarify the requirements of the h323 channel within 
>> asterisk.
>>
>> Can I use a recent edition of PTLib and OpenH323, for example, the 
>> editions located at OpenH323+:
>>
>> http://www.h323plus.org/source/
>>
>> OpenH323+ v1.20.2
>> PTLib v2.0.1
>>
>> Or do I need to use the versions at the original, now defunct, OpenH323 
>> website:
>>
>> http://www.openh323.org/
>>
>> OpenH323 v1.12.2
>> PWLib v1.5.2
>>
>> I am hoping to build this for Asterisk 1.4.18 running on Solaris 10.
>>
>> Thanks
>> Bruce
>>
>>
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> 
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> 

-- 
+---+
| Bruce McAlister  Blueface Ltd |
| <[EMAIL PROTECTED]>  http://www.blueface.ie |
+---+

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Re: [asterisk-users] chan_h323 requirements

[Bruce McAlister]

Hi,

Thanks for the information, I will keep this for reference.

Thanks
Bruce

Mindaugas Kezys wrote:
> This can help (script for Debian):
> 
> 
> apt-get install flex bison
> 
> #dirty hack to prevent error from missing file
> cd /usr/include/linux
> touch compiler.h
> 
> #PWLIB
> cd /usr/src
> wget 
> http://kent.dl.sourceforge.net/sourceforge/openh323/pwlib-v1_10_0-src-tar.gz
> tar zxvf pwlib-v1_10_0-src-tar.gz
> cd pwlib_v1_10_0/
> ./configure
> make
> make install
> make opt
> PWLIBDIR=/usr/src/pwlib_v1_10_0
> export PWLIBDIR
> 
> #OpenH323
> cd /usr/src
> wget 
> http://ovh.dl.sourceforge.net/sourceforge/openh323/openh323-v1_18_0-src-tar.gz
> tar zxvf openh323-v1_18_0-src-tar.gz
> cd openh323_v1_18_0/
> ./configure
> make
> make opt
> make install
> OPENH323DIR=/usr/src/openh323_v1_18_0/
> export OPENH323DIR
> 
> cd /usr/src/asterisk/channels/h323/
> make
> make opt
> cd /usr/src/asterisk
> ./configure
> make
> make install
> 
> echo "/usr/local/lib" >> /etc/ld.so.conf
> ldconfig
> 
> #or similar way 
> #cp /usr/local/lib/* /usr/lib
> 
> 
> 
> Regards,
> Mindaugas Kezys
> http://www.kolmisoft.com
> MOR PRO - Advanced Billing for Asterisk PBX
> 
> 
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce McAlister
> Sent: Thursday, February 21, 2008 10:58 AM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] chan_h323 requirements
> 
> Hi All,
> 
> I would just like to clarify the requirements of the h323 channel within 
> asterisk.
> 
> Can I use a recent edition of PTLib and OpenH323, for example, the 
> editions located at OpenH323+:
> 
> http://www.h323plus.org/source/
> 
> OpenH323+ v1.20.2
> PTLib v2.0.1
> 
> Or do I need to use the versions at the original, now defunct, OpenH323 
> website:
> 
> http://www.openh323.org/
> 
> OpenH323 v1.12.2
> PWLib v1.5.2
> 
> I am hoping to build this for Asterisk 1.4.18 running on Solaris 10.
> 
> Thanks
> Bruce
> 
> 
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> 

-- 
+---+
| Bruce McAlister  Blueface Ltd |
| <[EMAIL PROTECTED]>  http://www.blueface.ie |
+---+

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Re: [asterisk-users] chan_h323 requirements

[Vlasis Hatzistavrou (KTI)]

Hello,

To compile chan_h323 as is distributed you need to download OpenH323 
v1.18.0 and PwLib v1.10.0 from:

http://www.voxgratia.org

Some months ago I had made a patch to compile the 1.4.x version and the 
trunk version (which evolved to 1.6.x) with H323+.

Sadly, the patch was not included in the 1.6.x version which is being 
released soon.

So, for the time being you need to use the above versions from Voxgratia.

Best regards,
Vlasis Hatzistavrou.

Bruce McAlister wrote:
> Hi All,
> 
> I would just like to clarify the requirements of the h323 channel within 
> asterisk.
> 
> Can I use a recent edition of PTLib and OpenH323, for example, the 
> editions located at OpenH323+:
> 
> http://www.h323plus.org/source/
> 
> OpenH323+ v1.20.2
> PTLib v2.0.1
> 
> Or do I need to use the versions at the original, now defunct, OpenH323 
> website:
> 
> http://www.openh323.org/
> 
> OpenH323 v1.12.2
> PWLib v1.5.2
> 
> I am hoping to build this for Asterisk 1.4.18 running on Solaris 10.
> 
> Thanks
> Bruce
> 
> 
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Re: [asterisk-users] IAX2 trunks unreliable becoming UNREACHABLE after a time

[bilal ghayyad]

I am personally Waiting u :) - 

Thanks in advance.

Regards
Bilal
---
I may have found a solution to why this problem is
happening to me. All
 my
IAX trunks are up and working and have been for over a
day now. If
 there are
still up and running with no problems in a week I will
post again and
 let
everyone know.

At this point in time it seems the problem was caused
by a poorly
constructed initrd file.

My servers all run RAID-1 and my /var/ is mounted on
it's own RAID-1. I
 hand
crafted a new initrd to ensure that RAID-1 started
properly and that
 /var/
was mounted before init runs, it was not before. As of
now I am very
 hopeful
that the problem is gone, all indicators are that this
was the solution
 I
needed.


On Sun, Feb 10, 2008 at 12:33 PM, Royce Souther
 <[EMAIL PROTECTED]>
 wrote:

> I have a network of offices using Asterisk that are
connected via
 IAX2
> trunks. The trunks work great for a day or two then
for no reason
 at
 all one
> end of the trunk will become UNREACHABLE while the
other end is
 still
> connected. The only way to fix the problem is to
shutdown Asterisk
 completly
> then start it backup again. The end that dies is not
always the
 same,
 some
> times it is server A and some times it is server B.
Never have I
 seen
 that
> both ends die, just one. The side that is still
connected can make
 calls to
> the end that died but not the other way. If you call
from the
 server
 with
> the dead IAX2 trunk you here "All circuts are busy
now." All
 networks
 have
> static IP addresses and their firewalls are setup to
allow UDP
 4569
 to come
> in to the Asterisk systems.
>
> I have been doing a lot of research into this
problem. I found
 this
 bug
> tracker http://bugs.digium.com/view.php?id=5912 that
talks about
 it
 being
> an old problem with  version 1.2.1 using rand() and
it not being
 thread
> safe. This I can understand. The thread proposed
using rand_r() or
> ast_random() in place of rand(), that sounds like a
good idea. So
 when I
> look at my newer 1.2.18 version I find that it is
still using
 rand()
 and
> the bug tracker continues to be opened and closed
and reopened
 again
 and
> again.
>
> Do I dare ask if anyone has a reliable IAX2 trunk?
If so how?
 Should
 I
> avoid using IAX2 all together? I know SIP trunking
is an option
 but
 it
> becomes a real management problem with trying to
deal with all the
 many
> ports that need to be open through the firewalls,
IAX2 seems like
 a
 better
> way to go if only it was reliable.
>
> --
> Open Source: To innovate then create
> Proprietary: To imitate then litigate





  

Be a better friend, newshound, and 
know-it-all with Yahoo! Mobile.  Try it now.  
http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ 


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Re: [asterisk-users] How to get a clean, basic configuration?

[Mindaugas Kezys]

We do:

in modules.conf:

noload => pbx_ael.so
noload => pbx_dundi.so
noload => res_config_pgsql.so
noload => res_smdi.so

in extensions.conf delete every context [default], [demo], whatever

in sip.conf, iax.conf delete all peer/users if any

Regards,
Mindaugas Kezys
http://www.kolmisoft.com
MOR PRO - Advanced Billing for Asterisk PBX



-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vincent
Sent: Thursday, February 21, 2008 4:31 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] How to get a clean, basic configuration?

Hello

I'm using a standard Asterisk install with default settings, and when
I run "reload", I see that Asterisk fetches configuration information
from a lot more sources than just my extensions.conf and sip.conf.

For instance:

-- Registered indication country 've'
-- Registered indication country 'za'
-- Setting default indication country to 'us'
  == Parsing '/etc/asterisk/features.conf': Found
  == Parsing '/etc/asterisk/adsi.conf': Found
  == Parsing '/etc/asterisk/dundi.conf': Found
  == Parsing '/etc/asterisk/extensions.conf': Found
  == Parsing '/etc/asterisk/users.conf': Found
[Feb 21 03:29:15] NOTICE[2563]: pbx_ael.c:4094 pbx_load_module:
Starting AEL load process.
[Feb 21 03:29:15] NOTICE[2563]: pbx_ael.c:4101 pbx_load_module: AEL
load process: calculated config file name
'/etc/asterisk/extensions.ael'.
etc.

How can I go and trim things down?

Thank you.


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Re: [asterisk-users] chan_h323 requirements

[Mindaugas Kezys]

This can help (script for Debian):


apt-get install flex bison

#dirty hack to prevent error from missing file
cd /usr/include/linux
touch compiler.h

#PWLIB
cd /usr/src
wget 
http://kent.dl.sourceforge.net/sourceforge/openh323/pwlib-v1_10_0-src-tar.gz
tar zxvf pwlib-v1_10_0-src-tar.gz
cd pwlib_v1_10_0/
./configure
make
make install
make opt
PWLIBDIR=/usr/src/pwlib_v1_10_0
export PWLIBDIR

#OpenH323
cd /usr/src
wget 
http://ovh.dl.sourceforge.net/sourceforge/openh323/openh323-v1_18_0-src-tar.gz
tar zxvf openh323-v1_18_0-src-tar.gz
cd openh323_v1_18_0/
./configure
make
make opt
make install
OPENH323DIR=/usr/src/openh323_v1_18_0/
export OPENH323DIR

cd /usr/src/asterisk/channels/h323/
make
make opt
cd /usr/src/asterisk
./configure
make
make install

echo "/usr/local/lib" >> /etc/ld.so.conf
ldconfig

#or similar way 
#cp /usr/local/lib/* /usr/lib



Regards,
Mindaugas Kezys
http://www.kolmisoft.com
MOR PRO - Advanced Billing for Asterisk PBX


-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce McAlister
Sent: Thursday, February 21, 2008 10:58 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] chan_h323 requirements

Hi All,

I would just like to clarify the requirements of the h323 channel within 
asterisk.

Can I use a recent edition of PTLib and OpenH323, for example, the 
editions located at OpenH323+:

http://www.h323plus.org/source/

OpenH323+ v1.20.2
PTLib v2.0.1

Or do I need to use the versions at the original, now defunct, OpenH323 
website:

http://www.openh323.org/

OpenH323 v1.12.2
PWLib v1.5.2

I am hoping to build this for Asterisk 1.4.18 running on Solaris 10.

Thanks
Bruce


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[asterisk-users] Third Party Call Control - SIP to Iax Gateway

[Cavalera Claudio Luigi]

Hello,
can Asterisk be used in a 3PCC scenario as described in RFC:
ftp://ftp.rfc-editor.org/in-notes/rfc3725.txt

I'm not meaning using Asterisk as the controller, I mean Asterisk be
controlled by a 3rd party Back to Back User Agent.
In this case can Asterisk translate Sip into iax and hiding this from
the controller?

Regards,
Claudio


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Re: [asterisk-users] Best ATA. Period.

[Rob Hillis]

Of the three ATAs I've got (Linksys PAP2-NA, Sipura SPA-2000 and
SPA-3000) the Linksys PAP2-NA is the best of the bunch, even though the
SPA-2000 is supposedly cut from the same mould.

For the most part, you set 'em and forget 'em.  Most of the time when I
have a problem with a phone connected to the PAP2, I discover that the
phone itself has locked up.  The company I work for also sells them -
though I've had my PAP2 for a lot longer than I've worked there.


Adam Moffett wrote:
> Any opinions on the best ATA?
>
> For example, if someone was having a problem and I wanted to rule out 
> any ATA glitches or firmware issues, what device could I give them that 
> I could count on to always be a trouble free top performer that just 
> plain works?
>
>
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>http://lists.digium.com/mailman/listinfo/asterisk-users
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[asterisk-users] Voted most stable and easy to use phone?

[Michael J. Liberatore]

A while back i had asked about possible replacements for snom 360 phones
that were breaking and causing issues and we all discussed the problems
we had with the 360s and some suggestions were made but the  new polycom
phones had just hit the market and not many people were able to comment
on them.
 
Basically i am looking to get some new phones and in the process get rid
of the countless number of problems i have had that has always been
caused by phones (snom 360's and gxp-2000's).  I would like to get the
feedback of the list on the phone voted best for stability, working with
*, and ease of use for dumb non tech users.
 
I was thinking of trying one of these new polycom phones that are about
$150, but havent gotten any feedback on them yet.
 
Basically i am interested in any phones but snom's, grandstreams, and
sipura's/linksys.  mainly polycom's i guess.
 
thanks
 
mike


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[asterisk-users] chan_h323 requirements

[Bruce McAlister]

Hi All,

I would just like to clarify the requirements of the h323 channel within 
asterisk.

Can I use a recent edition of PTLib and OpenH323, for example, the 
editions located at OpenH323+:

http://www.h323plus.org/source/

OpenH323+ v1.20.2
PTLib v2.0.1

Or do I need to use the versions at the original, now defunct, OpenH323 
website:

http://www.openh323.org/

OpenH323 v1.12.2
PWLib v1.5.2

I am hoping to build this for Asterisk 1.4.18 running on Solaris 10.

Thanks
Bruce


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Re: [asterisk-users] SIP <> GSM

[Mindaugas Kezys]

Cyber-Telecom's CT-V372 is same box as PorTech MV-372 but with more advanced 
firmware. It supports more functions, such as SMS sending.

Regards,
Mindaugas Kezys
http://www.kolmisoft.com
MOR PRO - Advanced Billing for Asterisk PBX


-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ben Willcox
Sent: Wednesday, February 20, 2008 11:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SIP <> GSM

> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Kev S
> Sent: Tuesday, January 29, 2008 9:53 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] SIP <> GSM
> 
> With that sort of set up, If for example i get a 8 channel GSM gateway 
> and the X100P can i make more than 1 concurrent call though the gateway 
> with the X100P or does it only support 1 call at a time?
> 
> What im looking to do is get a multi channel GSM gateway, and have the 
> ability to make more than 1 call at once through it.

The PorTech MV-372 works nicely with asterisk and is multichannel (2, if 
that counts!)

Cheers,
Ben

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Re: [asterisk-users] Best ATA. Period.

[Mindaugas Kezys]

Linksys SPA 2102. No issues at all. Period.

Regards,
Mindaugas Kezys
http://www.kolmisoft.com
MOR PRO - Advanced Billing for Asterisk PBX


-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adam Moffett
Sent: Wednesday, February 20, 2008 11:26 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Best ATA. Period.

Any opinions on the best ATA?

For example, if someone was having a problem and I wanted to rule out 
any ATA glitches or firmware issues, what device could I give them that 
I could count on to always be a trouble free top performer that just 
plain works?


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Re: [asterisk-users] Coppercom and Asterisk

[Alex Balashov]

In the [general] section, put:

register => 8159093010:[EMAIL PROTECTED]

Then add a SIP peer for the outbound proxy.  Something like:

[essex1_outbound]

fromdomain=proxy.essex1.com
host=proxy.essex1.com
port=5060
insecure=very
username=8159093010
secret=X
type=peer
qualify=no
canreinvite=no
dtmfmode=rfc2833
disallow=all
allow=ulaw

The first one is needed for the registrations, and the second one is 
needed to answer 407 proxy challenges.

Mike Hammett wrote:
> My provider has a Coppercom switch.  I have included the authentication 
> information they gave me.  How would I structure this in Asterisk to the 
> registration and the entry in sip.conf?
>  
> User Name - 8159093010
> Password - X
> No Pin
> Proxy - sip.essex1.com (10.1.3.2)
> Outbound Proxy - proxy.essex1.com (63.164.210.14)
> Change setting to use "outbound Proxy"
>  
>  
> --
> Mike Hammett
> Intelligent Computing Solutions
> http://www.ics-il.com
>  
>  
> 
> 
> 
> 
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-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

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