[asterisk-users] about the Dial application

[arun kumar]

Hi guys
I am working in Kanpur, India.
When someone calls to my server i forward the call to someone else by Dial 
command. After dialing it says Native bridging. And after that I am unable to 
detect whether the call was answered, the called number was busy or the call 
was not completed.
One more issue, I want to record the discussion going on between the two 
persons. I used the options "wW", but was unable to do it. 
Please if someone know how to solve these problems, please help me out.
Thanks in advance..

arun


  From Chandigarh to Chennai - find friends all over India.. Go to 
http://in.promos.yahoo.com/groups/citygroups/___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] Major problem with 1.4.21 asterisk

[Steve Totaro]

Just out of curiosity, why did you feel they needed an upgrade?

Thanks,
Steve

On Thu, Jun 26, 2008 at 12:01 AM, Michael J. Liberatore
<[EMAIL PROTECTED]> wrote:
> Hopefully the other guy with the problem can test it because this is a
> production server and the client is already upset about the problems
> this caused for a day or two till I realized what the issue is so I cant
> risk it.   Maybe I can off hours if he cant though.
>
> Mike
>
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Tilghman
> Lesher
> Sent: Wednesday, June 25, 2008 9:32 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Major problem with 1.4.21 asterisk
>
> On Tuesday 24 June 2008 23:56:22 Michael J. Liberatore wrote:
>> Hi, i upgraded the other ay to 1.4.21 from 1.4.19 and started having
>> major iax2 problems.  All of a sudden calls wouldnt come in on the
>> iax2 DID, and we couldnt make calls out even though everything looked
> ok.
>> Also there was usually a hung iax2 channel when this happened.
>> Stopping asterisk also wouldnt work, i would do a "Stop now" and it
>> would just go back to the cli prompt.  I would do a ? and it wouldnt
>> work.  I would have to kill asterisk via ps and then restart it via
>> init.d and then
>> iax2 would start working again for a short while (maybe a few hours)
>>
>> I reinstalled 1.4.19 and the problems went away.  There appears to be
>> a major bug in 1.4.21 but i am not sure.
>
> Please try the patch in bug number 12903:
> http://bugs.digium.com/view.php?id=12903
>
> --
> Tilghman
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now:
> http://www.astricon.net
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> 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.
>
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>

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

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] Major problem with 1.4.21 asterisk

[Michael J. Liberatore]

Hopefully the other guy with the problem can test it because this is a
production server and the client is already upset about the problems
this caused for a day or two till I realized what the issue is so I cant
risk it.   Maybe I can off hours if he cant though.

Mike
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tilghman
Lesher
Sent: Wednesday, June 25, 2008 9:32 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Major problem with 1.4.21 asterisk

On Tuesday 24 June 2008 23:56:22 Michael J. Liberatore wrote:
> Hi, i upgraded the other ay to 1.4.21 from 1.4.19 and started having 
> major iax2 problems.  All of a sudden calls wouldnt come in on the 
> iax2 DID, and we couldnt make calls out even though everything looked
ok.
> Also there was usually a hung iax2 channel when this happened.  
> Stopping asterisk also wouldnt work, i would do a "Stop now" and it 
> would just go back to the cli prompt.  I would do a ? and it wouldnt 
> work.  I would have to kill asterisk via ps and then restart it via 
> init.d and then
> iax2 would start working again for a short while (maybe a few hours)
>
> I reinstalled 1.4.19 and the problems went away.  There appears to be 
> a major bug in 1.4.21 but i am not sure.

Please try the patch in bug number 12903:
http://bugs.digium.com/view.php?id=12903

--
Tilghman

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

AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now:
http://www.astricon.net

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


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.


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

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] Major problem with 1.4.21 asterisk

[Michael J. Liberatore]

Yes I forgot to mention, I did need to do kill -9 to finally kill it.
We have the exact same bug.  Yes mine works for 10 - 20 minutes also.  I
am glad I am not alone on this.

Mike
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Thomas
Kenyon
Sent: Wednesday, June 25, 2008 6:17 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Major problem with 1.4.21 asterisk

Thomas Kenyon wrote:
> Michael J. Liberatore wrote:
>> Hi, i upgraded the other ay to 1.4.21 from 1.4.19 and started having 
>> major iax2 problems.  All of a sudden calls wouldnt come in on the 
>> iax2 DID, and we couldnt make calls out even though everything looked
ok.
>> Also there was usually a hung iax2 channel when this happened.  
>> Stopping asterisk also wouldnt work, i would do a "Stop now" and it 
>> would just go back to the cli prompt.  I would do a ? and it wouldnt 
>> work.  I would have to kill asterisk via ps and then restart it via 
>> init.d and then
>> iax2 would start working again for a short while (maybe a few hours)
>>  
>> I reinstalled 1.4.19 and the problems went away.  There appears to be

>> a major bug in 1.4.21 but i am not sure.
>>  
>> thanks
>>  
>> mike
>>  
> I seem to have exactly the same problem, have rolled back to 1.4.19.2
.
> 
> Although on my machine I needed to kill -9 the process before it 
> finally died. (process is launched by safe_asterisk).
> 
> 1.6.0b9 (running at home) doesn't suffer this.
> 
I forgot to mention that for the 10 to 20 minutes (at a time) asterisk
1.4.21 is working, chan_alsa also appears to have stopped working (well
produces chan_alsa.c:693 alsa_read: Read error: Resource temporarily
unavailable).

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

AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now:
http://www.astricon.net

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


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.


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

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] Cisco Presence

[Peder @ NetworkOblivion]

SIP.

Michiel van Baak wrote:
> On 14:59, Wed 25 Jun 08, Peder @ NetworkOblivion wrote:
>> Does anybody have the settings that you use on a Cisco 7970/79x1 to get 
>> presence?  I see the * side settings, but I can't find the Cisco side 
>> settings anywhere.
> 
> Sip or Skinny ?
>> ___
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
>> Register Now: http://www.astricon.net
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
> 

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

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] Building a Complex IVR

[Douglas Garstang]

I don't think anyone did, and I was hoping someone would. :)


- Original Message 
From: Steve Murphy <[EMAIL PROTECTED]>
To: Asterisk Users Mailing List - Non-Commercial Discussion 

Sent: Tuesday, June 24, 2008 3:57:48 PM
Subject: Re: [asterisk-users] Building a Complex IVR

On Mon, 2008-06-23 at 09:54 -0700, Douglas Garstang wrote:
> I'm about to build a complex IVR with Asterisk.
> 
> Having done it a few times with the dial plan, I know it's going to be
> pretty ugly. What are my other options? I guess I could do it in
> AGI/FastAGI. What about VxML (about which I know almost nothing...)?
> 
> Using Asterisk 1.2
> 
> Thanks,
> Doug.
> 

Sorry, I tried to peak thru all the stuff in this thread, but I may 
have missed it; has anyone suggested the externalIVR app? If not,
it might be worth consideration...?

murf

> 
-- 
Steve Murphy
Software Developer
Digium



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

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] does asterisk 1.4.20 run on a 486 sx

[Tilghman Lesher]

On Wednesday 25 June 2008 17:15:26 Stelios Koroneos wrote:
> Depending on the gcc version you use you need to set it to produce i486
> code.
> The illegal instructions are probably because the default makefile builds
> for a later arch.
> Also without an fpu and fp kernel emulation don't expect things like dtmf
> to work. Kernel fp emulation is very slow.

Or because it's using the RDTSC instruction, which if he upgraded to the
revision I posted earlier in the thread, would be avoided.

-- 
Tilghman

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

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] does asterisk 1.4.20 run on a 486 sx

[Stelios Koroneos]

Depending on the gcc version you use you need to set it to produce i486
code.
The illegal instructions are probably because the default makefile builds
for a later arch.
Also without an fpu and fp kernel emulation don't expect things like dtmf to
work. Kernel fp emulation is very slow.



Stelios S. Koroneos

Digital OPSiS - Embedded Intelligence
http://www.digital-opsis.com
 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> Jerry Geis
> Sent: Tuesday, June 24, 2008 8:40 PM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] does asterisk 1.4.20 run on a 486 sx
> 
> I have compiled asterisk 1.4.20 on a 486 (sx) machine. No 
> floating point but math emulation is used in the kernel.
> When I run asterisk -vc all I get is Illegal instruction.
> 
> I compiled as normally I do. Whats my next step. this is 
> download source and compiled.
> 
> Jerry
> 
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register 
> Now: http://www.astricon.net
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
> 


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

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] Google Apps IMAP

[Gavin Henry]

Google Apps version might.

2008/6/25 Marc Smith <[EMAIL PROTECTED]>:
> Hi,
>
> Anyone using Asterisk IMAP voicemail storage with Google Apps / GMail
> IMAP? If so, does their IMAP implementation support any kind of
> "master user" (Dovecot) abililty? Good? Bad?
>
> --Marc
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
http://www.suretecsystems.com/services/openldap/

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

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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

[asterisk-users] iax2_trunk_queue: Maximum trunk data space exceeded

[Edwin Lam]

hi folks.

one of the servers i setup recently start exhibiting
"iax2_trunk_queue: Maximum trunk data space exceeded"
errors. there was only 1 call going on at the time.
usually i have to reload chan_iax2.so or restart Asterisk.
but the errors came back within a few minutes. i did a
google search on that error. but nothing useful came up.
i'm kinda stuck. anyone have any ideas?

p.s.
hardware: HP proliant BL460c server
OS: Linux 2.6.18 (Debian Etch AMD64 customize kernel)
Asterisk: 1.4.18
i have several setup with the same version of everything
except the hardware slightly different. they never have
this kind of problems.

-- 
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


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

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] Warning: CDRfix branches about to be merged into 1.4, 1.6.0, trunk!

[Grey Man]

On Tue, Jun 24, 2008 at 4:28 PM, Steve Murphy <[EMAIL PROTECTED]> wrote:
> This is just a note that the fixes in the CDRfix4 and CDRfix6 branches
> are getting closer to being merged into 1.4, trunk, and 1.6.x.
>
> If CDR's are important to you, and you ignore this notice, then
> you deserve what you get!
>

Hi murf,

>From some preliminary testing on the CDRfix4 branch it looks like the
CDRs for attended transfers are now correct which is fantastic. For
blind transfers the CDR for the first call leg is still incorrect with
the duration only being recorded up until the point the transfer
occurs.

For people on the list following this bug my company got stung by this
in the last week so there now appear to be some people out there
actively looking for Asterisk systems to exploit. The incident for us
was a user using attended transfers to place free calls through a 1.2
system. In the past we have had normal users stumble across the
problem but in this case it was a directed attempt. So if like us you
are a provider and use Asterisk and are required to support transfers
it would be highly advisable to keep a close eye on things!

Regards,

Greyman.

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

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] SIP vs. SKINNY

[Michiel van Baak]

On 14:16, Wed 25 Jun 08, Joe Carroll wrote:
> Can anyone comment on the performance benefits when comparing sip to skinny ?

Most cisco phones work better with the skinny firmware.

That is not true when connecting to asterisk though.

It all depends on the version of asterisk you are running.
I have a setup with over 20 skinny phones on asterisk -trunk and that
works great. Specially after today, now that chan_skinny supports
transfers.

If you are running 1.4 I'm not sure what is best. It basically depends
on what you are doing with the phones.
In my home setup it worked great, but in my business I have to run trunk
for the phones to be as workable as the sip variant.

The skinny firmware has some neat stuff like XML push etc.
Dont know how the current SIP firmware is doing, as I have not run it in
over 2 years now.

YMMV
-- 

Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x71C946BD

"Why is it drug addicts and computer aficionados are both called users?"


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

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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

[asterisk-users] SIP vs. SKINNY

[Joe Carroll]

Can anyone comment on the performance benefits when comparing sip to skinny ?
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] Cisco Presence

[Michiel van Baak]

On 14:59, Wed 25 Jun 08, Peder @ NetworkOblivion wrote:
> Does anybody have the settings that you use on a Cisco 7970/79x1 to get 
> presence?  I see the * side settings, but I can't find the Cisco side 
> settings anywhere.

Sip or Skinny ?
> 
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> 
> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> Register Now: http://www.astricon.net
> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users

-- 

Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x71C946BD

"Why is it drug addicts and computer aficionados are both called users?"


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

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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

[asterisk-users] Cisco Presence

[Peder @ NetworkOblivion]

Does anybody have the settings that you use on a Cisco 7970/79x1 to get 
presence?  I see the * side settings, but I can't find the Cisco side 
settings anywhere.

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

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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

[asterisk-users] Res: Asterisk with Nextone using H323

[Everton Goularth]

Thank`s all,

Chris Ziomkowski wrote:
 > If you only want to use H.323 with Asterisk, you should configure it 
as an H.323 gateway.
 > Why are you trying to set "softswitch"?

I was asked by a costumer, because he could not use a asterisk as a 
softswitch in the Nextone configuration, so I`m looking for the 
difference in asterisk configs files.

Thank`s a lot for you help...

Everton Goularth



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

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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

[asterisk-users] Cisco 7960 Promiscuous Redirect?

[Brent Torrenga]

List,

A Cisco 7960 is registered to servers A and B, where B is the backup server,
only used by the 7960 if A is unreachable.  That is the behavior of these
phones.

A call comes from server B to the 7960, which is successful.  The 7960 then
tries to park the call via an attended transfer, so the 7960 calls
[EMAIL PROTECTED]  However, the transfer fails, though I am not sure why.  I 
have
promiscredir=yes set in all the sip.confs.  So I am thinking that the 7960
has a problem handing the incoming call from server B to server A.

Does anyone know that this is the problem, or else know of a solution?  I
can route calls from server B through server A to get to the 7960, but I'd
rather just call the 7960 from server B directly.

--Brent




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

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] play sound on a specific channel

[nik600]

i've seen that there is the PlayDTMF command.

Bye

On Tue, Jun 24, 2008 at 8:37 AM, nik600 <[EMAIL PROTECTED]> wrote:
> any idea?
>
> On Sat, Jun 14, 2008 at 9:50 AM, nik600 <[EMAIL PROTECTED]> wrote:
>> Hi to all
>>
>> can i play a sound or a dtmf tone on a specific channel using AMI?
>>
>> Thanks to all
>>
>> --
>> /*/
>> nik600
>> https://sourceforge.net/projects/ccmanager
>> https://sourceforge.net/projects/reportmaker
>> https://sourceforge.net/projects/nikstresser
>>
>
>
>
> --
> /*/
> nik600
> https://sourceforge.net/projects/ccmanager
> https://sourceforge.net/projects/reportmaker
> https://sourceforge.net/projects/nikstresser
>



-- 
/*/
nik600
https://sourceforge.net/projects/ccmanager
https://sourceforge.net/projects/reportmaker
https://sourceforge.net/projects/nikstresser

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

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] included context not being priori tized properly

[Tilghman Lesher]

On Wednesday 25 June 2008 11:39:33 Brian J. Murrell wrote:
> On Wed, 2008-06-25 at 11:25 -0500, Tilghman Lesher wrote:
> > That's only true within the same context.  ONLY if a match is not found
> > in the current context will it go into an included context.
>
> Ahhh.  Well, then that explains it.  Any thoughts on how to achieve my
> goal, without having to encode all of what would be in the override
> context in the outbound-ld context?

Reverse the contexts.  Put all of your exact matches in the main context
and include another context with the pattern match.

-- 
Tilghman

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

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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

[asterisk-users] [Fwd: Bridging an existing PBX in with Asterisk]

[Matthew Ratliff]

Correctionit's a Fujitsu 9600 PBX



Have any of you worked with a Fujitsu J5600 PBX before?  How about connecting 
it to a Asterisk server?  Currently this customer has two pri's connected to 
the Fujitsu pbx  box.  I'll be introducing Asterisk into their environment, and 
will slowly migrate all users to Asterisk over a period of several months.  
What is the best approach to this?  Should the PRI's remain connected to the 
Fujitsu box or move them to the Asterisk server?  Should the PRI's be split 
between the two?  How about connecting up Asterisk to the old PBX?  Any help on 
this is great appreciated!!

_
This email was transferred using an Office free edition
of AXIGEN Mail Server.


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

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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

[asterisk-users] Bridging an existing PBX in with Asterisk

[Matthew Ratliff]

Have any of you worked with a Fujitsu J5600 PBX before?  How about connecting 
it to a Asterisk server?  Currently this customer has two pri's connected to 
the Fujitsu pbx  box.  I'll be introducing Asterisk into their environment, and 
will slowly migrate all users to Asterisk over a period of several months.  
What is the best approach to this?  Should the PRI's remain connected to the 
Fujitsu box or move them to the Asterisk server?  Should the PRI's be split 
between the two?  How about connecting up Asterisk to the old PBX?  Any help on 
this is great appreciated!!

_
This email was transferred using an Office free edition
of AXIGEN Mail Server.


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

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] Can asterisk support using different ip for rtp?

[Jun Yin]

some vendors(like alcatel-lucent) developed a kind of sip proxy which
includes two parts: one sip signaling module and one or more voice
modules. voice modules are responsible for receiving/sending voice
traffic(RTP). each voice module has its own IP. so , when the sip
signaling part sends out "invite" packet, it has sip ip in its sip
content and different RTP ip in SDP content. (also for 200OK)
Now I'm trying to do a test to simulate that product with asterisk. I
hope asterisk can sends out different rtp address based on user or
domain name. Based on network side, there are many ways to do it: we
can configure the network card with multiple IPs, one for SIP and
others for RTP.  or , we can setup multiple network cards for the
asterisk server, one card is for sip signaling and other cards for rtp
traffic connecting to different carriers.   I think this diagram is
reasonable but I was surprised that asterisk does not support it.
Maybe asterisk can do this by special configuration? or, there is
other free sip proxy software can do this?

Thanks.

> Message: 10
> Date: Wed, 25 Jun 2008 05:15:29 -0400
> From: "Raj Jain" <[EMAIL PROTECTED]>
> Subject: Re: [asterisk-users] Can asterisk support using different ip
>for rtp?
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
> Message-ID:
><[EMAIL PROTECTED]>
> Content-Type: text/plain; charset="iso-8859-1"
>
> On Tue, Jun 24, 2008 at 9:26 PM, Jun Yin <[EMAIL PROTECTED]> wrote:
>
>> Currently, RTP IP have to be the same as SIP IP. But, SIP RFC allows
>> RTP to use different IP as SIP ip.
>>
>> Is there any way to configure it? GUI or CLI? or , will we support it in
>> future?
>>
>
> SIP is decoupled from RTP, so they can emanate from different IP addresses.
> Can you present a scenario where this will make sense (in the context where
> Asterisk is anchoring the media) ?
>
> --
> Raj Jain

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

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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

[asterisk-users] Google Apps IMAP

[Marc Smith]

Hi,

Anyone using Asterisk IMAP voicemail storage with Google Apps / GMail
IMAP? If so, does their IMAP implementation support any kind of
"master user" (Dovecot) abililty? Good? Bad?

--Marc

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

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] included context not being prioritized properly

[Brian J. Murrell]

On Wed, 2008-06-25 at 11:25 -0500, Tilghman Lesher wrote:
> That's only true within the same context.  ONLY if a match is not found in the
> current context will it go into an included context.

Ahhh.  Well, then that explains it.  Any thoughts on how to achieve my
goal, without having to encode all of what would be in the override
context in the outbound-ld context?

b.




signature.asc
Description: This is a digitally signed message part
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] included context not being prioritized properly

[Tilghman Lesher]

On Wednesday 25 June 2008 10:54:19 Brian J. Murrell wrote:
> I have an "outbound-ld" context as follows:
>
> [ Context 'outbound-ld' created by 'pbx_config' ]
>   '_1NXXNXX' => 1. Macro(enumdial|${EXTEN})  
> [pbx_config] 102. Wait(1)  [pbx_config]
> 103. Set(LINE=${IF($[${LINE}=pots]?link2voip:${LINE})}) [pbx_config] 104.
> Macro(dial${LINE}|${EXTEN})  [pbx_config] 105. Hangup()
> [pbx_config] Include =>'toll-free-override'
>  [pbx_config]
>
> and "toll-free-override" has:
>
>   '18002687096' =>  1. Macro(dialpots|${EXTEN})  
> [pbx_config] '18009598281' =>  1. Macro(dialpots|${EXTEN}) 
>  [pbx_config] '18664277451' =>  1. Macro(dialpots|${EXTEN})
>   [pbx_config] '18668797179' =>  1. Macro(dialpots|${EXTEN})   
>[pbx_config] '18884189338' =>  1. Macro(dialpots|${EXTEN})  
> [pbx_config] '18884716070' =>  1. Macro(dialpots|${EXTEN}) 
>  [pbx_config] '18886684636' =>  1. Macro(dialpots|${EXTEN})
>   [pbx_config]
>
> But when I dial a number in the toll-free-override context (i.e.
> 18664277451) it still gets processed by the "enumdial" macro.  I thought
> that an exact match would take precedence over a "_" match.  What am I
> missing?

That's only true within the same context.  ONLY if a match is not found in the
current context will it go into an included context.

-- 
Tilghman

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

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] Any SLA alternatives?

[Adam Moffett]

> The simplest way to do this is to use a different prefix for dialling out.
> For example, if they dial out with a prefix of 9, use the shared number, and
> if they dial out with a prefix of 8, use the private number.
>
>   
Took the words right out of my mouth.

Basically if you have something like this:

exten => 1NXXNXX,1,Dial(outgoingtrunk/${EXTEN})

Add something like this:

exten => 91NXXNXX,1,Set(CALLERID(number)=sharedoutgoingnumber)
exten => 91NXXNXX,2,Dial(outgoingtrunk/${EXTEN})



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

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] Chef-secretary scenario

[Grygoriy Dobrovolskyy]

You have 2 choices to pickup someone's phone with snom's

1: imagine yourself prefix of pickup, let's say 4
exten=>4XX,1,Pickup([EMAIL PROTECTED])

so if u call 4 + phone number you will pickup that one.

Second you can add pickupgroup=number for each phone you want to be in the
group, and add a dtmf button on snom with string set in features.conf
(pickup)

To answer you next question: Yes i would be nice to pickup a phone by
pressing blinking button on snom,and use that button fo call when ext is out
of use, but i dont know the way to make asterisk doing that.

2008/6/25 Vazquez David <[EMAIL PROTECTED]>:

> Though I wonder...
>
> The scenario is as follows:
>
> I have 4 phones with the following extensions:
> 11 (SIP/11)
> 12 (SIP/12)
> 13 (SIP/13)
> 15 (SIP/15)
>
> Whenever SIP/11 receives a call, it hints the other phones. Is it
> possible to pick up that call from one of them?
>
> The relevant part of my extensions.conf looks as this now:
>
> [macro-stdexten];
> ;
> ; Standard extension macro:
> ;   ${ARG1} - Extension  (we could have used ${MACRO_EXTEN} here as well
> ;   ${ARG2} - Device(s) to ring
> ;
> exten => s,1,Dial(${ARG2},20)  ; Ring the interface, 20 seconds maximum
> exten => s,2,Goto(s-${DIALSTATUS},1)  ; Jump based on status
> (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
> exten => s,hint,SIP/11
> exten => 11,hint,SIP/12&SIP/13&SIP/15
> exten => s-NOANSWER,1,Voicemail(${ARG1},u)  ; If unavailable, send to
> voicemail w/ unavail announce
> exten => s-NOANSWER,2,Goto(default,s,1)  ; If they press #, return to start
> exten => s-BUSY,1,Voicemail(${ARG1},b)  ; If busy, send to voicemail w/
> busy announce
> exten => s-BUSY,2,Goto(default,s,1)  ; If they press #, return to start
> exten => _s-.,1,Goto(s-NOANSWER,1)  ; Treat anything else as no answer
> exten => a,1,VoicemailMain(${ARG1})  ; If they press *, send the user
> into VoicemailMain
>
> Thanks :-)
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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

[asterisk-users] included context not being prioritized properly

[Brian J. Murrell]

I have an "outbound-ld" context as follows:

[ Context 'outbound-ld' created by 'pbx_config' ]
  '_1NXXNXX' => 1. Macro(enumdial|${EXTEN})   [pbx_config]
102. Wait(1)  [pbx_config]
103. Set(LINE=${IF($[${LINE}=pots]?link2voip:${LINE})}) 
[pbx_config]
104. Macro(dial${LINE}|${EXTEN})  [pbx_config]
105. Hangup() [pbx_config]
  Include =>'toll-free-override'  [pbx_config]

and "toll-free-override" has:

  '18002687096' =>  1. Macro(dialpots|${EXTEN})   [pbx_config]
  '18009598281' =>  1. Macro(dialpots|${EXTEN})   [pbx_config]
  '18664277451' =>  1. Macro(dialpots|${EXTEN})   [pbx_config]
  '18668797179' =>  1. Macro(dialpots|${EXTEN})   [pbx_config]
  '18884189338' =>  1. Macro(dialpots|${EXTEN})   [pbx_config]
  '18884716070' =>  1. Macro(dialpots|${EXTEN})   [pbx_config]
  '18886684636' =>  1. Macro(dialpots|${EXTEN})   [pbx_config]

But when I dial a number in the toll-free-override context (i.e.
18664277451) it still gets processed by the "enumdial" macro.  I thought
that an exact match would take precedence over a "_" match.  What am I
missing?

b.



signature.asc
Description: This is a digitally signed message part
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] Any SLA alternatives?

[Tilghman Lesher]

On Wednesday 25 June 2008 10:20:14 Scott Moseman wrote:
> I have a group of people who have distinct phone numbers plus a shared
> number.  The shared number is actually a group that rings through to
> all of their direct numbers.  I want them to: 1) be able to make
> outgoing calls as the shared number and 2) be able to make outgoing
> calls as their direct number.

The simplest way to do this is to use a different prefix for dialling out.
For example, if they dial out with a prefix of 9, use the shared number, and
if they dial out with a prefix of 8, use the private number.

-- 
Tilghman

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

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] Chef-secretary scenario

[Vazquez David]

Though I wonder...

The scenario is as follows:

I have 4 phones with the following extensions:
11 (SIP/11)
12 (SIP/12)
13 (SIP/13)
15 (SIP/15)

Whenever SIP/11 receives a call, it hints the other phones. Is it
possible to pick up that call from one of them?

The relevant part of my extensions.conf looks as this now:

[macro-stdexten];
;
; Standard extension macro:
;   ${ARG1} - Extension  (we could have used ${MACRO_EXTEN} here as well
;   ${ARG2} - Device(s) to ring
;
exten => s,1,Dial(${ARG2},20)  ; Ring the interface, 20 seconds maximum
exten => s,2,Goto(s-${DIALSTATUS},1)  ; Jump based on status
(NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten => s,hint,SIP/11
exten => 11,hint,SIP/12&SIP/13&SIP/15
exten => s-NOANSWER,1,Voicemail(${ARG1},u)  ; If unavailable, send to
voicemail w/ unavail announce
exten => s-NOANSWER,2,Goto(default,s,1)  ; If they press #, return to start
exten => s-BUSY,1,Voicemail(${ARG1},b)  ; If busy, send to voicemail w/
busy announce
exten => s-BUSY,2,Goto(default,s,1)  ; If they press #, return to start
exten => _s-.,1,Goto(s-NOANSWER,1)  ; Treat anything else as no answer
exten => a,1,VoicemailMain(${ARG1})  ; If they press *, send the user
into VoicemailMain

Thanks :-)

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

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] Number portability in other parts of the world.

[randulo]

On Wed, Jun 25, 2008 at 5:28 PM, Dean Collins <[EMAIL PROTECTED]> wrote:
> Number portability exists in Australia but mobile numbers only across mobile
> carriers and 'pstn' numbers only across pstn carriers.

Incidentally, in France my Internet provider (Neuf) will pay for your
cell number to be ported. I'd be suspicious of anyone who wanted me
that badly :)

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

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] Number portability in other parts of the world.

[randulo]

On Wed, Jun 25, 2008 at 4:49 PM, Alexander Lopez <[EMAIL PROTECTED]> wrote:
> How does a person in Europe go fully VoIP and still keep the main number?
> Do they use call forwarding?

I shopuld have mentioned too, that if you run your own asterisk you
can of course just be connected to your number(s) and deal with it all
in the dialplan. This is exactly where I am today. Tomorrow, we'll
have ported the number to a service provider in order not to have to
leave an asterisk box running in an empty office, which is the case
now.

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

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] Chef-secretary scenario

[Vazquez David]

Thankyou all! I think I've got it working :-D

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

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] Number portability in other parts of the world.

[randulo]

On Wed, Jun 25, 2008 at 4:49 PM, Alexander Lopez <[EMAIL PROTECTED]> wrote:
> Are phone numbers portable in other countries?

Recent legislation in France (and perhaps this means the EU?) have made it so.

> Are the same rules and conditions that exist here in the States mirrored
> elsewhere?
>
>
>
> How does a person in Europe go fully VoIP and still keep the main number?

This is exactly what we are in the process of doing. We have asked our
provider to port our main phone number to their ipbx. I have used the
same people for IAX origination/termination so I presume that if we
don't like the ipbx we can go back to asterisk here or hosted and they
will be handling those calls.

It can't be said enough how important it is for a business to control
the destiny of its phone numbers. I feel these should be possible to
handle like domain names where you can buy one for life by renewing
it. Not quite the case today, but we get the same effect by paying a
fee each time we need to port the number.

r

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

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] Number portability in other parts of the world.

[Dean Collins]

Number portability exists in Australia but mobile numbers only across
mobile carriers and 'pstn' numbers only across pstn carriers.

 

Some of the earlier Voip providers (yeh I'm talking about you Faktortel
  )
don't allow portability as they assigned numbers that belonged to isdn
100 unit 'pstn blocks', but apart from that it's a pretty good system.

 

 

Regards,

Dean Collins
[EMAIL PROTECTED] 

+1-212-203-4357 (Direct) 
+1-917-207-3420 (Mobile)
+61-2-9016-5642 (Sydney in-dial)
http://www.Cognation.net  



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alexander
Lopez
Sent: Wednesday, 25 June 2008 10:49 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Number portability in other parts of the
world.

 

Are phone numbers portable in other countries?

 

Are the same rules and conditions that exist here in the States mirrored
elsewhere?

 

How does a person in Europe go fully VoIP and still keep the main
number? 

 

Do they use call forwarding? 

 

Is their another way to use an origination carrier without loosing your
number?

 

Alex

 

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

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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

[asterisk-users] Any SLA alternatives?

[Scott Moseman]

I have a group of people who have distinct phone numbers plus a shared
number.  The shared number is actually a group that rings through to
all of their direct numbers.  I want them to: 1) be able to make
outgoing calls as the shared number and 2) be able to make outgoing
calls as their direct number.

Currently it's working in a non-ideal state.  Each user has a phone
with two lines.  One line has the shared number and the other has
their distinct number.  Only one phone has the registration for the
shared number at any point in time.  The outgoing calls from either
line works great from all of the phones.

The problem is when they need to forward the shared line somewhere
else.  Only the phone that's currently registered as the shared line
can do the forwarding.  The users have no way to know, without logging
into all of the phone GUIs, to know which one actually is holding the
registration.

I researched using shared line appearance (SLA), but several people
may need to be able to make outgoing calls using the shared number at
the same time.  From what I read about SLA, it makes the line appear
busy on the other phones.  This is not what I'm looking for.  Is there
a solution for what I'm wanting to do?

[phone1]
line1=5000
line2=5001

[phone2]
line1=5000
line2=5002

[phone3]
line1=5000
line2=5003

That is a brief overview of the phone configs.  I want any/all of them
to be able to make outgoing calls from the 5000 extension, but also
maintain the ability to make calls from their 500x extensions.  I
thought about overwriting all of their Caller ID values, but that
defeats the purpose of having distinct lines per user.

Thanks,
Scott

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

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] Number portability in other parts of the world.

[Steve Kennedy]

On Wed, Jun 25, 2008 at 10:49:18AM -0400, Alexander Lopez wrote:

>Are phone numbers portable in other countries?

Depends what country

>Are the same rules and conditions that exist here in the States
>mirrored elsewhere?
>How does a person in Europe go fully VoIP and still keep the main
>number?

In the UK numbers are portable, though the telco wanting the number must
have a porting agreement with the telco that has the number. Not all
telcos have porting agreements.

>Do they use call forwarding?

Can do.


Steve

-- 
NetTek Ltd  UK mob +44 7775 755503
UK +44 20 7993 2612  /  US +1 310 857 7715  /  Fax +44 20 7483 2455
Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED]
Euro Tech News Blog http://eurotechnews.blogspot.com

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

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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

[asterisk-users] Number portability in other parts of the world.

[Alexander Lopez]

Are phone numbers portable in other countries?

 

Are the same rules and conditions that exist here in the States mirrored
elsewhere?

 

How does a person in Europe go fully VoIP and still keep the main
number? 

 

Do they use call forwarding? 

 

Is their another way to use an origination carrier without loosing your
number?

 

Alex

 

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

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] Chef-secretary scenario

[Rob Hillis]

Vazquez David wrote:
> Yes, I'm using 1.4. And I don't really use sip.conf, but have all my
> phones on users.conf. Should I put limitonpeers and call-limit on the
> general section of sip.conf? or on each entry in users.conf
[general] should be sufficient, so long as having them set as default 
for all your connections isn't a problem.

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

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] Major problem with 1.4.21 asterisk

[Tilghman Lesher]

On Tuesday 24 June 2008 23:56:22 Michael J. Liberatore wrote:
> Hi, i upgraded the other ay to 1.4.21 from 1.4.19 and started having
> major iax2 problems.  All of a sudden calls wouldnt come in on the iax2
> DID, and we couldnt make calls out even though everything looked ok.
> Also there was usually a hung iax2 channel when this happened.  Stopping
> asterisk also wouldnt work, i would do a "Stop now" and it would just go
> back to the cli prompt.  I would do a ? and it wouldnt work.  I would
> have to kill asterisk via ps and then restart it via init.d and then
> iax2 would start working again for a short while (maybe a few hours)
>
> I reinstalled 1.4.19 and the problems went away.  There appears to be a
> major bug in 1.4.21 but i am not sure.

Please try the patch in bug number 12903:
http://bugs.digium.com/view.php?id=12903

-- 
Tilghman

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

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] asterisk seg fault

[Tilghman Lesher]

On Wednesday 25 June 2008 07:42:21 Jerry Geis wrote:
> I am running asterisk from svn check out from yesterday Jun 24.
> I started with 1.4.20, then 1.4.21 then svn.
>
> I am getting:
> pcm_local.h:389 snd_pcm_channel_area_addr assertion bitsofs %8 = 0 failed
> segment fault.
>
> I am running debian i386, on a 486 sx machine.
> I am connecting to the Console/DSP and then I get the seg fault.
> Only thing in asterisk I changed from the default was turning off
> codec_lpc10.
> Which I am not using anyway.
>
> What should I do with this error?

That error is not from Asterisk code, but from libalsa.  You'll need to query
that project as to how to proceed.

-- 
Tilghman

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

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] asterisk seg fault

[Jerry Geis]

I managed to catch the whole trace from the seg fault. What is my next step?

Program received signal SIGABRT, Aborted.
[Switching to Thread -1224033360 (LWP 2440)]
0xb7d77947 in raise () from /lib/tls/libc.so.6
(gdb) where
#0  0xb7d77947 in raise () from /lib/tls/libc.so.6
#1  0xb7d790c9 in abort () from /lib/tls/libc.so.6
#2  0xb7d7105f in __assert_fail () from /lib/tls/libc.so.6
#3  0xb741861f in snd_pcm_area_copy (dst_area=0x81cc62c, dst_offset=0, 
src_area=0x81cb7ec, src_offset=5223, samples=816, 
format=SND_PCM_FORMAT_S16_LE) at pcm_local.h:499
#4  0xb744b7b7 in snd_pcm_dsnoop_sync_ptr (pcm=0x81cb930) at pcm_dsnoop.c:77
#5  0xb7410b11 in snd_pcm_hwsync (pcm=0x81cb930) at pcm.c:932
#6  0xb7422b55 in snd1_pcm_generic_hwsync (pcm=0x81cc1c8) at pcm_generic.c:143
#7  0xb7410b11 in snd_pcm_hwsync (pcm=0x81cc1c8) at pcm.c:932
#8  0xb742fb4d in snd_pcm_rate_hwsync (pcm=0x81cc3f8) at pcm_rate.c:624
#9  0xb7410b11 in snd_pcm_hwsync (pcm=0x81cc3f8) at pcm.c:932
#10 0xb7417817 in snd1_pcm_read_areas (pcm=0x81cc3f8, areas=0xb70a5b70, 
offset=0, size=160, func=0xb7421e10 )
at pcm.c:6376
#11 0xb742111f in snd_pcm_mmap_readi (pcm=0x81cc3f8, buffer=0xb74a5a40, 
size=160) at pcm_mmap.c:236
#12 0xb740fff4 in snd_pcm_readi (pcm=0x81cc038, buffer=0xb74a5a40, size=2440)
at pcm_local.h:521
#13 0xb749579b in alsa_read (chan=0x8213f18) at chan_alsa.c:683
#14 0x080855b1 in __ast_read (chan=0x8213f18, dropaudio=0) at channel.c:2100
#15 0x08087725 in ast_channel_bridge (c0=0x8212858, c1=0x817cf68, 
config=0xb70a6eac, fo=0xb70a5fe8, rc=0xb70a5fe4) at channel.c:2380
#16 0xb7c75592 in ast_bridge_call (chan=0x8212858, peer=0x8213f18, 
config=0xb70a6eac) at res_features.c:1490
#17 0xb75c38fd in dial_exec_full (chan=0x8212858, data=0xb70a8fd8, 
peerflags=0xb70a6f84, continue_exec=0x0) at app_dial.c:1773
#18 0xb75c519d in dial_exec (chan=0x8212858, data=0xb70a8fd8)
at app_dial.c:1827
#19 0x080c6e3f in pbx_extension_helper (c=0x8212858, con=0x0, 
context=0x82129d8 "smvoice-mediaport-firepanel", exten=0x8212a28 "s", 
priority=3, label=0x0, callerid=0x8215450 "404", action=E_SPAWN)
at pbx.c:537
#20 0x080c9185 in __ast_pbx_run (c=0x8212858) at pbx.c:2317
#21 0x080c9ebe in pbx_thread (data=0x8212858) at pbx.c:2636
#22 0x080f51d2 in dummy_start (data=0x8213178) at utils.c:909
#23 0xb7eff0bd in start_thread () from /lib/tls/libpthread.so.0
#24 0xb7e1b01e in clone () from /lib/tls/libc.so.6
(gdb) quit


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

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] Chef-secretary scenario

[Vazquez David]

Rob Hillis wrote:
> Can I assume you're using Asterisk 1.4 and that you've configured your 
> phones as peers?
>
> If this is the case, then you need to set limitonpeers to yes and 
> call-limit to some value in sip.conf.  Once this has been done, you 
> should find that BLF behaves as you expect.
>
>
> Vazquez David wrote:
>   
>> Hi all,
>>
>> I'm trying to implement such a scenario where the "Chef" picks up his
>> phone and his "secretary" can see that he is busy. Something like blf, I
>> guess. But so far I've only managed to notify the "secretary" that the
>> "chef" is receiving a call. I want to do it the other way around
>> though.  I'd like for her to see in her phone, the light corresponding
>> to the "chef"'s extension light up whenever he uses the phone (also when
>> he picks it up if that's possible).  So she should always know when he's
>> busy.
>>
>> Is there a way to do that?
>>
>> Thanks,
>> David
>>
>> ___
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
>> Register Now: http://www.astricon.net
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>> !DSPAM:4860fae540252026166755!
>>
>>
>>   
>> 
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>   
Yes, I'm using 1.4. And I don't really use sip.conf, but have all my
phones on users.conf. Should I put limitonpeers and call-limit on the
general section of sip.conf? or on each entry in users.conf?

Thanks :-D

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

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] Chef-secretary scenario

[Rob Hillis]

Can I assume you're using Asterisk 1.4 and that you've configured your 
phones as peers?

If this is the case, then you need to set limitonpeers to yes and 
call-limit to some value in sip.conf.  Once this has been done, you 
should find that BLF behaves as you expect.


Vazquez David wrote:
> Hi all,
>
> I'm trying to implement such a scenario where the "Chef" picks up his
> phone and his "secretary" can see that he is busy. Something like blf, I
> guess. But so far I've only managed to notify the "secretary" that the
> "chef" is receiving a call. I want to do it the other way around
> though.  I'd like for her to see in her phone, the light corresponding
> to the "chef"'s extension light up whenever he uses the phone (also when
> he picks it up if that's possible).  So she should always know when he's
> busy.
>
> Is there a way to do that?
>
> Thanks,
> David
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
> !DSPAM:4860fae540252026166755!
>
>
>   

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

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] Chef-secretary scenario

[Vazquez David]

Grygoriy Dobrovolskyy wrote:
> ok, so do you configured you snom phone tu subscribe to monito
> extension in asterisk ?
>
> It is simple to verify:
>
> core show hints
>
> Snom's behave exactly like you want when blf enabled.
>
> 2008/6/25 Vazquez David <[EMAIL PROTECTED]
> >:
>
> Gordon Henderson wrote:
> > On Tue, 24 Jun 2008, Vazquez David wrote:
> >
> >
> >> Hi all,
> >>
> >> I'm trying to implement such a scenario where the "Chef" picks
> up his
> >> phone and his "secretary" can see that he is busy. Something
> like blf, I
> >> guess.
> >>
> >
> > It's like BLF because that's exactly what it's for..
> >
> >
> >> But so far I've only managed to notify the "secretary" that the
> >> "chef" is receiving a call. I want to do it the other way around
> >> though.  I'd like for her to see in her phone, the light
> corresponding
> >> to the "chef"'s extension light up whenever he uses the phone
> (also when
> >> he picks it up if that's possible).  So she should always know
> when he's
> >> busy.
> >>
> >
> >
> >> Is there a way to do that?
> >>
> >
> > Yes. Configure hints in asterisk and BLF in the secretarys phone.
> >
> > I don't understand how you get notifications one way, but not
> the other
> > though (unless you're doing it by not using BLF)
> >
> > What phones have you got? What do your hints look like in the
> > extensions.conf file?
> >
> > You can't get the status of lifting the handset on a SIP phone
> though
> > (well, not that I'm aware of)
> >
> > However, if it's an analogue phone on a TDM400 card (or
> equivalent, I
> > guess) then it does work and the BLF LED on my Grandstream phone
> turns Red
> > as soon as I take the analogue phone off-hook...
> >
> > So BLF is what you want, and optionally an analogue phone +TDM
> card for
> > the boss...
> >
> > Gordon
> >
> > ___
> > -- Bandwidth and Colocation Provided by
> http://www.api-digital.com --
> >
> > AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> > Register Now: http://www.astricon.net
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> My hints on extensions.conf:
>
> » [exten15]
> » include = numberplan-custom-1
> » exten => 11,hint,SIP/12&SIP/11
>
> Where numberplan-custom-1 defines the general rules to dial any
> extension. And the secretary's phone (SIP/15) is subscribed to the
> context [exten15]. Now, I don't know if that's the right way of
> doing it
> (I'm a total n00b, you can tell, huh?).
>
> Oh, and I'm using Snoms. All Snoms. If there's not the possibility of
> knowing when a phone is lifted up, no problem...
>
> Thanks :)
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> 
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-us

You mean like, from the web interface (Snom's) configure a function key
to an extension? I guess so. I configured a function key to be mapped to
an identity and then, instead of line, I told it to be an extension.
Finally I wrote the extension I wanted to monitor...

Anyways, in a couple of hours I'll try it again... Thanks for the help :-)

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

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] Chef-secretary scenario

[Vazquez David]

Dr. Michael J. Chudobiak wrote:
>> I use Snoms. I know there's the feature. I just don't know how to use
>> it, and there's so little documentation on the web.. Anyway, with "see"
>> I meant that the "secretary"'s phone would have one of the function keys
>> "on" whenever the "chef" is on the phone (also when he picks it up,
>> right before dialing). Until now I've only managed to make both phones
>> blink on incoming calls. But that's not what I want and I could've done
>> that with "extension => 11,1,Dial(SIP/11&SIP/12&SIP/13...)".
>> 
>
> This should be very easy. Use something like:
>
> exten => 602,1,Dial(SIP/boss_office&SIP/boss_home,20,trj)
> exten => 602,2,Voicemail([EMAIL PROTECTED])
> exten => 602,102,Voicemail([EMAIL PROTECTED])
> exten => 602,hint,SIP/boss_office&SIP/boss_home
>
> exten => 603,1,Dial(SIP/secretary,20,trj)
> exten => 603,2,Voicemail([EMAIL PROTECTED])
> exten => 603,102,Voicemail([EMAIL PROTECTED])
> exten => 603,hint,SIP/secretary
>
> and set snom function keys to extensions 602 and 603. (Some firmware 
> versions say "Destination" instead of extension, I think.)
>
> - Mike
>
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>   
Thanks, I'll try that in a few hours... :-)

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

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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

[asterisk-users] asterisk seg fault

[Jerry Geis]

I am running asterisk from svn check out from yesterday Jun 24.
I started with 1.4.20, then 1.4.21 then svn.

I am getting:
pcm_local.h:389 snd_pcm_channel_area_addr assertion bitsofs %8 = 0 failed
segment fault.

I am running debian i386, on a 486 sx machine.
I am connecting to the Console/DSP and then I get the seg fault.
Only thing in asterisk I changed from the default was turning off 
codec_lpc10.
Which I am not using anyway.

What should I do with this error?

Jerry

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

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] Chef-secretary scenario

[Dr. Michael J. Chudobiak]

> I use Snoms. I know there's the feature. I just don't know how to use
> it, and there's so little documentation on the web.. Anyway, with "see"
> I meant that the "secretary"'s phone would have one of the function keys
> "on" whenever the "chef" is on the phone (also when he picks it up,
> right before dialing). Until now I've only managed to make both phones
> blink on incoming calls. But that's not what I want and I could've done
> that with "extension => 11,1,Dial(SIP/11&SIP/12&SIP/13...)".

This should be very easy. Use something like:

exten => 602,1,Dial(SIP/boss_office&SIP/boss_home,20,trj)
exten => 602,2,Voicemail([EMAIL PROTECTED])
exten => 602,102,Voicemail([EMAIL PROTECTED])
exten => 602,hint,SIP/boss_office&SIP/boss_home

exten => 603,1,Dial(SIP/secretary,20,trj)
exten => 603,2,Voicemail([EMAIL PROTECTED])
exten => 603,102,Voicemail([EMAIL PROTECTED])
exten => 603,hint,SIP/secretary

and set snom function keys to extensions 602 and 603. (Some firmware 
versions say "Destination" instead of extension, I think.)

- Mike


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

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] Chef-secretary scenario

[Grygoriy Dobrovolskyy]

ok, so do you configured you snom phone tu subscribe to monito extension in
asterisk ?

It is simple to verify:

core show hints

Snom's behave exactly like you want when blf enabled.

2008/6/25 Vazquez David <[EMAIL PROTECTED]>:

> Gordon Henderson wrote:
> > On Tue, 24 Jun 2008, Vazquez David wrote:
> >
> >
> >> Hi all,
> >>
> >> I'm trying to implement such a scenario where the "Chef" picks up his
> >> phone and his "secretary" can see that he is busy. Something like blf, I
> >> guess.
> >>
> >
> > It's like BLF because that's exactly what it's for..
> >
> >
> >> But so far I've only managed to notify the "secretary" that the
> >> "chef" is receiving a call. I want to do it the other way around
> >> though.  I'd like for her to see in her phone, the light corresponding
> >> to the "chef"'s extension light up whenever he uses the phone (also when
> >> he picks it up if that's possible).  So she should always know when he's
> >> busy.
> >>
> >
> >
> >> Is there a way to do that?
> >>
> >
> > Yes. Configure hints in asterisk and BLF in the secretarys phone.
> >
> > I don't understand how you get notifications one way, but not the other
> > though (unless you're doing it by not using BLF)
> >
> > What phones have you got? What do your hints look like in the
> > extensions.conf file?
> >
> > You can't get the status of lifting the handset on a SIP phone though
> > (well, not that I'm aware of)
> >
> > However, if it's an analogue phone on a TDM400 card (or equivalent, I
> > guess) then it does work and the BLF LED on my Grandstream phone turns
> Red
> > as soon as I take the analogue phone off-hook...
> >
> > So BLF is what you want, and optionally an analogue phone +TDM card for
> > the boss...
> >
> > Gordon
> >
> > ___
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >
> > AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> > Register Now: http://www.astricon.net
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> My hints on extensions.conf:
>
> » [exten15]
> » include = numberplan-custom-1
> » exten => 11,hint,SIP/12&SIP/11
>
> Where numberplan-custom-1 defines the general rules to dial any
> extension. And the secretary's phone (SIP/15) is subscribed to the
> context [exten15]. Now, I don't know if that's the right way of doing it
> (I'm a total n00b, you can tell, huh?).
>
> Oh, and I'm using Snoms. All Snoms. If there's not the possibility of
> knowing when a phone is lifted up, no problem...
>
> Thanks :)
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] Centile ipbx, anyone heard of this?

[Grygoriy Dobrovolskyy]

Dont worry i did a lot worse.

2008/6/25 randulo <[EMAIL PROTECTED]>:

> Nothing that embarrassing,just didn't want to mention the even more OT
> stuff. Everyone already knows I do not too bright things like turning
> a phone off and then complaining it doesn't work :)
>
> On Wed, Jun 25, 2008 at 12:39 PM, Grygoriy Dobrovolskyy
> <[EMAIL PROTECTED]> wrote:
> > Private messagind:)
> >
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] Chef-secretary scenario

[Vazquez David]

Gordon Henderson wrote:
> On Tue, 24 Jun 2008, Vazquez David wrote:
>
>   
>> Hi all,
>>
>> I'm trying to implement such a scenario where the "Chef" picks up his
>> phone and his "secretary" can see that he is busy. Something like blf, I
>> guess.
>> 
>
> It's like BLF because that's exactly what it's for..
>
>   
>> But so far I've only managed to notify the "secretary" that the
>> "chef" is receiving a call. I want to do it the other way around
>> though.  I'd like for her to see in her phone, the light corresponding
>> to the "chef"'s extension light up whenever he uses the phone (also when
>> he picks it up if that's possible).  So she should always know when he's
>> busy.
>> 
>
>   
>> Is there a way to do that?
>> 
>
> Yes. Configure hints in asterisk and BLF in the secretarys phone.
>
> I don't understand how you get notifications one way, but not the other 
> though (unless you're doing it by not using BLF)
>
> What phones have you got? What do your hints look like in the 
> extensions.conf file?
>
> You can't get the status of lifting the handset on a SIP phone though 
> (well, not that I'm aware of)
>
> However, if it's an analogue phone on a TDM400 card (or equivalent, I 
> guess) then it does work and the BLF LED on my Grandstream phone turns Red 
> as soon as I take the analogue phone off-hook...
>
> So BLF is what you want, and optionally an analogue phone +TDM card for 
> the boss...
>
> Gordon
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>   
My hints on extensions.conf:

» [exten15]
» include = numberplan-custom-1
» exten => 11,hint,SIP/12&SIP/11

Where numberplan-custom-1 defines the general rules to dial any
extension. And the secretary's phone (SIP/15) is subscribed to the
context [exten15]. Now, I don't know if that's the right way of doing it
(I'm a total n00b, you can tell, huh?).

Oh, and I'm using Snoms. All Snoms. If there's not the possibility of
knowing when a phone is lifted up, no problem...

Thanks :)

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

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] Chef-secretary scenario

[Vazquez David]

Grygoriy Dobrovolskyy wrote:
> To see? how? what phone do you use?
>
> Snoms imprement that, you got BLINKING and ON state
> BLINKING=calling or being called
> ON=on the phone
> 2008/6/24 Vazquez David <[EMAIL PROTECTED]
> >:
>
> Hi all,
>
> I'm trying to implement such a scenario where the "Chef" picks up his
> phone and his "secretary" can see that he is busy. Something like
> blf, I
> guess. But so far I've only managed to notify the "secretary" that the
> "chef" is receiving a call. I want to do it the other way around
> though.  I'd like for her to see in her phone, the light corresponding
> to the "chef"'s extension light up whenever he uses the phone
> (also when
> he picks it up if that's possible).  So she should always know
> when he's
> busy.
>
> Is there a way to do that?
>
> Thanks,
> David
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> 
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
I use Snoms. I know there's the feature. I just don't know how to use
it, and there's so little documentation on the web.. Anyway, with "see"
I meant that the "secretary"'s phone would have one of the function keys
"on" whenever the "chef" is on the phone (also when he picks it up,
right before dialing). Until now I've only managed to make both phones
blink on incoming calls. But that's not what I want and I could've done
that with "extension => 11,1,Dial(SIP/11&SIP/12&SIP/13...)".

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

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] Centile ipbx, anyone heard of this?

[randulo]

Nothing that embarrassing,just didn't want to mention the even more OT
stuff. Everyone already knows I do not too bright things like turning
a phone off and then complaining it doesn't work :)

On Wed, Jun 25, 2008 at 12:39 PM, Grygoriy Dobrovolskyy
<[EMAIL PROTECTED]> wrote:
> Private messagind:)
>

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

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] Centile ipbx, anyone heard of this?

[Grygoriy Dobrovolskyy]

Private messagind:)

2008/6/25 randulo <[EMAIL PROTECTED]>:

> On Tue, Jun 24, 2008 at 4:50 PM, Michael Graves <[EMAIL PROTECTED]> wrote:
> > Randy,
> >
> > This is exactly what was happening when I used an Aastra 480i CT with
> > OnSIP. According to OnSIP it's not a supported phone, although the
> > newer 57i CT does work with OnSIP.
> >
> > It seemed that the phone was losing registration with the provider. I
> > was not able to overcome this in the phone or provider settings.
>
> Hi Michael,
>
> I'm writing you privately for a reason you will immediately
> understand. In a moment of frustration, I set up the AA50 asterisk
> appliance, thinking, hey, I can have the best of all possible worlds
> here, including all my dialplans and kludges and still benefit from
> the centrex. Because I only have a single DSL at the moment, I messed
> with this but didn't get it running completely. I'm pretty sure it
> will work if I want to go that way, once the AA50 is acting as a
> router and not behind one.
>
> However, eason the Siemens wasn't accepting calls is simpler than I
> thought:
>
> I had "accept calls" unchecked for that provider on both handsets!!!
>
> I guess I must have done this while testing another phone on the same
> account parameters.
>
> D'oh!
>
> So you saw that JR Richardson is talking anbout scaling on the
> conference Friday. I decided to do one after all, from the office at
> the usual time.
>
> Also, got a guest post in voipsupply blog. You're next!
>
> Best,
>
> Randy
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] Can asterisk support using different ip for rtp?

[Johansson Olle E]

25 jun 2008 kl. 11.15 skrev Raj Jain:

> On Tue, Jun 24, 2008 at 9:26 PM, Jun Yin <[EMAIL PROTECTED]> wrote:
> Currently, RTP IP have to be the same as SIP IP. But, SIP RFC allows
> RTP to use different IP as SIP ip.
>
> Is there any way to configure it? GUI or CLI? or , will we support  
> it in future?
>
> SIP is decoupled from RTP, so they can emanate from different IP  
> addresses. Can you present a scenario where this will make sense (in  
> the context where Asterisk is anchoring the media) ?

In general, it's quite frequent in larger setups with remote RTP  
proxys or media servers. However, as I already said, Asterisk can't  
handle this today.

/O

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

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] Major problem with 1.4.21 asterisk

[Thomas Kenyon]

Thomas Kenyon wrote:
> Michael J. Liberatore wrote:
>> Hi, i upgraded the other ay to 1.4.21 from 1.4.19 and started having 
>> major iax2 problems.  All of a sudden calls wouldnt come in on the iax2 
>> DID, and we couldnt make calls out even though everything looked ok.  
>> Also there was usually a hung iax2 channel when this happened.  Stopping 
>> asterisk also wouldnt work, i would do a "Stop now" and it would just go 
>> back to the cli prompt.  I would do a ? and it wouldnt work.  I would 
>> have to kill asterisk via ps and then restart it via init.d and then 
>> iax2 would start working again for a short while (maybe a few hours)
>>  
>> I reinstalled 1.4.19 and the problems went away.  There appears to be a 
>> major bug in 1.4.21 but i am not sure. 
>>  
>> thanks
>>  
>> mike
>>  
> I seem to have exactly the same problem, have rolled back to 1.4.19.2 .
> 
> Although on my machine I needed to kill -9 the process before it finally 
> died. (process is launched by safe_asterisk).
> 
> 1.6.0b9 (running at home) doesn't suffer this.
> 
I forgot to mention that for the 10 to 20 minutes (at a time) asterisk 
1.4.21 is working, chan_alsa also appears to have stopped working (well 
produces chan_alsa.c:693 alsa_read: Read error: Resource temporarily 
unavailable).

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

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] Major problem with 1.4.21 asterisk

[Thomas Kenyon]

Michael J. Liberatore wrote:
> Hi, i upgraded the other ay to 1.4.21 from 1.4.19 and started having 
> major iax2 problems.  All of a sudden calls wouldnt come in on the iax2 
> DID, and we couldnt make calls out even though everything looked ok.  
> Also there was usually a hung iax2 channel when this happened.  Stopping 
> asterisk also wouldnt work, i would do a "Stop now" and it would just go 
> back to the cli prompt.  I would do a ? and it wouldnt work.  I would 
> have to kill asterisk via ps and then restart it via init.d and then 
> iax2 would start working again for a short while (maybe a few hours)
>  
> I reinstalled 1.4.19 and the problems went away.  There appears to be a 
> major bug in 1.4.21 but i am not sure. 
>  
> thanks
>  
> mike
>  
I seem to have exactly the same problem, have rolled back to 1.4.19.2 .

Although on my machine I needed to kill -9 the process before it finally 
died. (process is launched by safe_asterisk).

1.6.0b9 (running at home) doesn't suffer this.

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

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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

[asterisk-users] misdn issues

[Zaine Pretorius]

Hi,

I'm having issues where people call one of my ISDN numbers, and sometime, when 
we answer, the call is dead. Its only sometimes happens, like one in 15 
calls. I have analogue phones pluged into a mediatrix and a couple of pap's, 
and these connect to the asterisk server. Please could someone help me with 
this issue.

## extensions.conf 
[from-pstn]
; Incoming ISDN Calls
exten => _X.,1,Answer()
exten => _X.,n,NoOp("DNID = " ${DNID})
exten => _X.,n,NoOp("CALLERID = " ${CALLERID(all)})
exten => _X.,n,NoOp("EXTEN = " ${EXTEN})
exten => _X.,n,Dial(SIP/${EXTEN})


## /etc/asterisk/misdn.conf
echocancel=yes

[TEports]
ports=1,2,3,4
context=from-pstn
;msns=*
msns=5718.etc

## /etc/misdn-init.conf
card=1,0x4
poll=128
dsp_poll=128
dsp_options=0
dtmfthreshold=100
debug=0


## sip.conf
[5713]
type = friend
username = 5713
secret = 
regexten = 5713
host = dynamic
context = lofts
qualify=yes
nat=yes
disallow=all
allow=ulaw
allow=g729
allow=alaw
allow=g723.1
callerid=5713

Regards
Zaine

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

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] Can asterisk support using different ip for rtp?

[Raj Jain]

On Tue, Jun 24, 2008 at 9:26 PM, Jun Yin <[EMAIL PROTECTED]> wrote:

> Currently, RTP IP have to be the same as SIP IP. But, SIP RFC allows
> RTP to use different IP as SIP ip.
>
> Is there any way to configure it? GUI or CLI? or , will we support it in
> future?
>

SIP is decoupled from RTP, so they can emanate from different IP addresses.
Can you present a scenario where this will make sense (in the context where
Asterisk is anchoring the media) ?

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

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] Centile ipbx, anyone heard of this?

[randulo]

On Tue, Jun 24, 2008 at 4:50 PM, Michael Graves <[EMAIL PROTECTED]> wrote:
> This is very similar to another idea that I once had but never actually
> implemented. That is, using a small embedded Asterisk device as a
> SIP<>IAX2 protocol translator to facilitate complex NAT traversal. I
> thought that Astlinux on Gumstix hardware would be ideal for such a
> task.

Interesting thought, that. Use an embedded asterisk to act as a
"concentrator" on the NAT side. Why not?

/r

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

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] Centile ipbx, anyone heard of this?

[randulo]

On Tue, Jun 24, 2008 at 4:50 PM, Michael Graves <[EMAIL PROTECTED]> wrote:
> Randy,
>
> This is exactly what was happening when I used an Aastra 480i CT with
> OnSIP. According to OnSIP it's not a supported phone, although the
> newer 57i CT does work with OnSIP.
>
> It seemed that the phone was losing registration with the provider. I
> was not able to overcome this in the phone or provider settings.

Hi Michael,

I'm writing you privately for a reason you will immediately
understand. In a moment of frustration, I set up the AA50 asterisk
appliance, thinking, hey, I can have the best of all possible worlds
here, including all my dialplans and kludges and still benefit from
the centrex. Because I only have a single DSL at the moment, I messed
with this but didn't get it running completely. I'm pretty sure it
will work if I want to go that way, once the AA50 is acting as a
router and not behind one.

However, eason the Siemens wasn't accepting calls is simpler than I thought:

I had "accept calls" unchecked for that provider on both handsets!!!

I guess I must have done this while testing another phone on the same
account parameters.

D'oh!

So you saw that JR Richardson is talking anbout scaling on the
conference Friday. I decided to do one after all, from the office at
the usual time.

Also, got a guest post in voipsupply blog. You're next!

Best,

Randy

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

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] Can asterisk support using different ip for rtp?

[Johansson Olle E]

25 jun 2008 kl. 03.26 skrev Jun Yin:

> Currently, RTP IP have to be the same as SIP IP. But, SIP RFC allows
> RTP to use different IP as SIP ip.
>
> Is there any way to configure it? GUI or CLI? or , will we support  
> it in future?

There's currently no support for that in Asterisk.

/O

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

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] AS5400 E1 SS7

[Alex Balashov]

Alex Balashov wrote:

> Nhadie Ramos wrote:
> 
>> Would just like to inquire if anyone here has a setup of asterisk to 
>> send traffic to AS5400 connected to an SS7-PRI.  this is more of a 
>> AS54 question, as i've been reading and i always stumble upon PGW2200 
>> as a requirement to handle SS7 signaling on the AS54. Has anyone able 
>> to send calls from asterisk to an as 54 with SS7-PRI without PGW2200?
> 
> What, exactly, is an SS7 "PRI"?

I'm just going to assume you're referring to a T1-based SS7 IMT, which 
you have, for some reason, decided to term a PRI.

It is not, in fact, a PRI;  a PRI is an ISDN connection that uses ITU-T 
Q.931/921 signaling, not SS7.

If it is, in fact, an SS7 IMT, the answer is that no, the Cisco AS* 
series does not provide ISUP or any other SS7 features except certain 
stubs.  You need a real SS7 gateway.


-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

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

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] AS5400 E1 SS7

[Alex Balashov]

Nhadie Ramos wrote:

> Would just like to inquire if anyone here has a setup of asterisk to 
> send traffic to AS5400 connected to an SS7-PRI.  this is more of a AS54 
> question, as i've been reading and i always stumble upon PGW2200 as a 
> requirement to handle SS7 signaling on the AS54. Has anyone able to send 
> calls from asterisk to an as 54 with SS7-PRI without PGW2200?

What, exactly, is an SS7 "PRI"?

-- 
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599

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

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] Calls drop + "Didn't get a frame from channel" log message

[gincantalupo]

Hi Jean,

I have an Asterisk 1.12.18 with about 30 pc each with a Doro SIP phone 
on an unknown LAN.
I think google is useless in cases like this.
Many of the system we are working on are in production and we cannot 
make tests with them so the only hope is to gather infos from people 
experiencing the same problems and trying to understand something from 
the code, maybe asking the developers for info...that's why I'd like to 
understand what kind of frames Asterisk is waiting for in order to find 
what is the cause of this problem (maybe this is not the right 
mailing-list ?)
The strangest thing is problems like this are still happening and remain 
unsolved.but a PBX dropping calls (or not dropping at all, as 
sometime happens) is like a car without seats.yes, you can drive, 
but it is not a comfortable experience!

Giorgio


Jean-Louis curty wrote:
> I have googled a lot to find solution to the same exact problem 
> described in your message but no real solution yet.
>
> here is my config
>
> 1 physical network
> 25 pc windows
> 25 phones IP330 & IP550 SIP 2.1.2 no vlan CDP disabled some with dhcp 
> some with fixed ip to see if there is a diff
>
> 3 switchs connected to each others
> 1 cisco switch 35xx for pcs
> 2 linksys 24P P OE for phones
>
> 1 patton PRI gateway to isdn
>
> 1 asterisk server 1.12.18 talking sip to Patton , for each phone, type 
> friend can re-invite no, nat no
>
> symptoms:
>
> call drop randomly , can be after 10 s or 2000 seconds ! same log 
> "didn't get frame etc
> fews drops per phones per day but very irritating for the customer :-(
>
>
> tried to power phones with adapters to avoid power pbs from the switch 
> , same result
>
> if someone met this problem before get an idea to fix it , I wd 
> appreciate !
>
> thanks
> jl
>
> On Tue, Jun 24, 2008 at 5:37 PM, gincantalupo 
> <[EMAIL PROTECTED] > 
> wrote:
>
> Hi,
>
> sometimes Asterisk drops calls and shows "Didn't get a frame from
> channel" in its log file. Unfortunately Google gives no answers
> even if
> a lot of people ask for help.
> A fast look into the code shows Asterisk entering a loop where
> voice is
> been transferred and every loop Asterisk waits for a frame,
> exiting the
> loop if no frame has arrived. It seems to be a problem not
> depending on
> the kind of channel...happens with ISDN and PRI lines.
> What is stopping the frames, making Asterisk exiting that loop and
> dropping the calls?
>
> Thank you.
>
> Giorgio.
>
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> 
>
> ___
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users


-- 

_
Giorgio Incantalupo, mailto:[EMAIL PROTECTED]
FG&A srl - http://www.fgasoftware.com -
[EMAIL PROTECTED] - The Agile PBX http://www.voiceatwork.eu
Tel: 02997663.14, Fax: 0291390172  


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

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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

[asterisk-users] AS5400 E1 SS7

[Nhadie Ramos]

Hi,

Would just like to inquire if anyone here has a setup of asterisk to send 
traffic to AS5400 connected to an SS7-PRI.  this is more of a AS54 question, as 
i've been reading and i always stumble upon PGW2200 as a requirement to handle 
SS7 signaling on the AS54. Has anyone able to send calls from asterisk to an as 
54 with SS7-PRI without PGW2200?

TIA
Regards,
Nhadie



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

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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

Re: [asterisk-users] mpg123 problem

[Tzafrir Cohen]

On Tue, Jun 24, 2008 at 10:06:00PM +0200, Stefan Tichy wrote:
> On Sun, Jun 22, 2008 at 12:24:22AM -0700, fateme fatah wrote:
> > I want to install mpg123-0.59r on my asterisk server.I downloaded it in
> > /usr/src then untared it and I typed these command :
> 
> Just have a look at www.mpg123.org and fetch the up to date version.
> 
> 0.59r is probably available with your distribution but it is known
> to cause some problems.

Actually 0.59r is probably older than the one available from your
distribution and has known security holes. It is recommended by the
Asterisk community by some strange voodoo reasons.

In most cases you don't need mpg123 anyway. The most common remaining
use case for mpg123 is streaming remote mp3 streams. But then you should
consider the known security holes that version has.

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

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

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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

[asterisk-users] unable to send a fax to a given FAX number

[reitenbach_pub]

Hi all,

I have some problem to send a FAX to a given number. I use asterisk 1.2.18, on 
a openSUSE 10.2, i586 host.

The FAX is sent out via an ISDN PRI interface, I'm in Germany, and the 
destination FAX devices are in Germany too, but in different areas, so I have 
to use a city prefix.

I did set the pri device in debug mode, below are two calls, to two different 
FAX numbers, the first is the one with the problem, the second is working 
well:

==WORKING=

==

==

 -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
 > fixlocalprefix: Using pattern 1081+112
 > fixlocalprefix: Using pattern 1081+110
 > fixlocalprefix: Using pattern 1081+11833
 > fixlocalprefix: Using pattern 1081+11880
 > fixlocalprefix: Using pattern 0|.
 == fixlocalprefix: Dialpattern 0|. matched. 01267383226 -> 
 1267383226
 -- AGI Script fixlocalprefix completed, returning 0
 -- Executing Set("SIP/233-b4e0a7f8", "OUTNUM=1267383226") in new 
 stack
 -- Executing Set("SIP/233-b4e0a7f8", "custom=ZAP/g1") in new stack
 -- Executing GotoIf("SIP/233-b4e0a7f8", "0?customtrunk") in new 
 stack
 -- Executing Dial("SIP/233-b4e0a7f8", "ZAP/g1/1267383226|120|r") 
 in new stack
 -- Making new call for cr 42167
 -- Requested transfer capability: 0x00 - SPEECH
 > Protocol Discriminator: Q.931 (8) len=43
 > Call Ref: len= 2 (reference 9399/0x24B7) (Originator)
 > Message type: SETUP (5)
 > [04 03 80 90 a3]
 > Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer 
 capability: Speech (0)
 > Ext: 1 Trans mode/rate: 64kbps, 
 circuit-mode (16)
 > Ext: 1 User information layer 1: A-Law 
 (35)
 > [18 03 a9 83 81]
 > Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, 
 Exclusive Dchan: 0
 > ChanSel: Reserved
 > Ext: 1 Coding: 0 Number Specified Channel 
 Type: 3
 > Ext: 1 Channel: 1 ]
 > [6c 0d 21 80 33 33 38 31 38 39 30 34 34 35 31]
 > Calling Number (len=15) [ Ext: 0 TON: National Number (2) NPI: 
 ISDN/Telephony Numbering Plan (E.164/E.163) (1)
 > Presentation: Presentation permitted, user 
 number not screened (0) '4482904233' ]
 > [70 0b a1 33 34 36 35 33 38 33 32 32 36]
 > Called Number (len=13) [ Ext: 1 TON: National Number (2) NPI: 
 ISDN/Telephony Numbering Plan (E.164/E.163) (1) '1267383226' ]
 -- Called g1/1267383226
 < Protocol Discriminator: Q.931 (8) len=10
 < Call Ref: len= 2 (reference 9399/0x24B7) (Terminator)
 < Message type: SETUP ACKNOWLEDGE (13)
 < [18 03 a9 83 81]
 < Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, 
 Exclusive Dchan: 0
 < ChanSel: Reserved
 < Ext: 1 Coding: 0 Number Specified Channel 
 Type: 3
 < Ext: 1 Channel: 1 ]
 -- Processing IE 24 (cs0, Channel Identification)
 < Protocol Discriminator: Q.931 (8) len=5
 < Call Ref: len= 2 (reference 9399/0x24B7) (Terminator)
 < Message type: CALL PROCEEDING (2)
 -- Zap/1-1 is proceeding passing it to SIP/233-b4e0a7f8
 < Protocol Discriminator: Q.931 (8) len=9
 < Call Ref: len= 2 (reference 9399/0x24B7) (Terminator)
 < Message type: ALERTING (1)
 < [1e 02 82 88]
 < Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard 
 (0) 0: 0 Location: Public network serving the local user (2)
 < Ext: 1 Progress Description: Inband 
 information or appropriate pattern now available. (8) ]
 -- Processing IE 30 (cs0, Progress Indicator)
 -- Zap/1-1 is ringing
 -- Remote UNIX connection
 -- Remote UNIX connection disconnected
 -- Remote UNIX connection
 -- Remote UNIX connection disconnected
 < Protocol Discriminator: Q.931 (8) len=12
 < Call Ref: len= 2 (reference 9399/0x24B7) (Terminator)
 < Message type: CONNECT (7)
 < [29 05 08 06 18 10 00]
 < Time Date (len= 7) [ 08-06-24 16:00 ]
 -- Processing IE 41 (cs0, Date/Time)
 > Protocol Discriminator: Q.931 (8) len=5
 > Call Ref: len= 2 (reference 9399/0x24B7) (Originator)
 > Message type: CONNECT ACKNOWLEDGE (15)
 < Protocol Discriminator: Q.931 (8) len=29
 < Call Ref: len= 2 (reference 9399/0x24B7) (Terminator)
 < Message type: FACILITY (98)
 < [1c 16 91 a1 13 02 02 56 72 02 01 22 30 0a a1 05 30 03 02 01 02 82 
 01 00]
 < Facility (len=24, codeset=0) [ 0x91, 0xa1, 0x13, 0x02, 0x02, 'Vr', 
 0x02, 0x01, 0x22, '0', 0x0a, 0xa1, 0x05, '0', 0x03, 0x02, 0x01, 0x02, 
 0x82, 0x01, 0x00 ]
 -- Processing IE 28 (cs0, Facility)
 Handle Q.932 ROSE Invoke component
 -- Zap/1-1 answered SIP/233-b4e0a7f8
 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Active, peerstate 
 Connect Request
 > Protocol Discriminator: Q.931 (8) len=9
 > Call Ref: len= 2 (reference 9399/0x24B7) (Originator)
 > Message type: DISCONNECT (69)
 > [08 02 81 90]
 > Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 
 Location: Private network serving the local user (1)
 > Ext: 1 Cause: Normal