Re: [asterisk-users] conferencing help

[Matt Riddell]

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Tzafrir Cohen wrote:
> (This tests both cases right away: gives different error messages in the
> different cases)

Sweet :)

- --
Kind Regards,

Matt Riddell
Director
___

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Re: [asterisk-users] Distorted audio over Eicon Diva Server BRI

[Matt Riddell]

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CSB wrote:
> Asterisk is sending it's data packets (160 bytes each, i.e. 20ms) in too
> large intervals. This causes the transmitter of the Diva Server card to
> underrun and thus to fill with idle samples in regular intervals. It's
> almost between any two packets where we have to insert samples.
> 
> 0:00:29.710 CAPI20_PUT(030)
> 
>  0:00:29.730 CAPI20_PUT(030)
> 
>  0:00:29.751 CAPI20_PUT(030)
> 
>  0:00:29.771 CAPI20_PUT(030)
> 
>  0:00:29.791 CAPI20_PUT(030)
> 
>  0:00:29.812 CAPI20_PUT(030)
> 
>  0:00:29.832 CAPI20_PUT(030)
> 
>  0:00:29.853 CAPI20_PUT(030)
> 
>  0:00:29.873 CAPI20_PUT(030)
> 
>  0:00:29.894 CAPI20_PUT(030)
> I wonder if anyone could provide any advice on how to continue
> troubleshooting this issue? 

Sounds very similar to an issue I was having.

Are you using mISDN?

- --
Kind Regards,

Matt Riddell
Director
___

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http://www.venturevoip.com/news.php (Daily Asterisk News - html)
http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss)
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=ASrD
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Re: [asterisk-users] app_rxfax.c and app_txxfax.c where?

[Tzafrir Cohen]

On Tue, Jan 08, 2008 at 04:19:38PM +1100, Klaverstyn, David C wrote:
> Hi All,
> 
>  
> 
> Where can I find copies of the app_rxfax.c, app_txfax.c and
> apps_Makefile.patch.  They don't seem to be located at soft-switch.org
> anymore.

http://sourceforge.net/projects/agx-ast-addons

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Re: [asterisk-users] conferencing help

[Tzafrir Cohen]

On Tue, Jan 08, 2008 at 06:45:14PM +1300, Matt Riddell wrote:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
> 
> Nhadie wrote:
> > Hi Matt,
> > 
> > it seems i don't have that command.
> 
> :)
> 
> You'll need to make sure that:
> 
> 1. You have zaptel compiled
> 2. You compile Asterisk *after* zaptel is compiled and installed
> 3. You have either modprobed zaptel + ztdummy or made the service and
> started it.

In other words, what is the output of the following command from the
Asteris CLI:

  module load chan_zap.so

(This tests both cases right away: gives different error messages in the
different cases)

-- 
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Re: [asterisk-users] Remote hold on PRI

[Gaëtan Minet]

Hi Shane

yes, with verbose activated, I get something like this:

starting Music on hold for Channel SIP/XXX (the local caller sip side)  
with the class specified in sip.conf.


With zap span debugging enabled, I see the bridged zap channel  
receives "REMOTE HOLD" notification just before the moh starts.


That sounds like a nice feature between enterprise servers, but not so  
good between different companies of course.


I wonder if this is not a side effect of the (relatively new ?)  
"mohinterpret=passtrough" option. That would be nice to have a  
"mohinterpret=disable"  in fact:).


Thank you

Regards,
Gaetan


On 07/01/2008, at 21:28, Shane Spencer wrote:


So, watching the asterisk console with full debug on shows something
about "Starting Music On Hold for Channel xx/yy-zz"?

Shane

On Jan 7, 2008 11:00 AM, Gaëtan Minet <[EMAIL PROTECTED]> wrote:


Hi

Nobody has an Idea ? Should I try and fill a bug report (or feature  
request

?) at Digium  ?
The only solution I personally see is a patch in the source.

Regards

Gaetan



On 04/01/2008, at 23:26, Gaëtan Minet wrote:
Hi everybody

We have a strange problem with several asterisk servers (Version
1.4.11) using PRI cards (tied to telco here in Belgium).

Indeed we noticed that whenever a local user places an outgoing call
through the PRI (and telco) to another IPBX (tied to telco using BRI
or PRI), if the remote party places the call on hold, the caller  
hears

the _local_ music on hold instead of the remote one.  In fact we can
briefly hear the remote music on hold start, then it is replaced by
the local one.

More precisely:

Company 1 uses an asterisk server with a PRI card tied to the telco.
Company 2 uses any PBX that ca place calls on hold and is tied to the
telco using a digital interface (tested with BRIs and PRIs)

A (company 1) calls B (company 2)
B answers and park or places the call on hold
A hears the MOH of company 1.

The same happens when calling a mobile: when the mobile user puts the
call on hold, instead of hearing the mobile operator's own moh, the
calling user hears the moh of his own company asterisk.

I think this has something to do with REMOTE_HOLD notifications on  
PRI
lines that gets reported back to the calling asterisk server, which  
in

turn somehow puts the bridged (SIP) channel on hold, but I can't find
much more information about this.
Is this the expected behavior ? A feature or a bug ? Do you know if
this can be tuned/tweaked/disabled (i.e. filter or ignore this
signaling on the zap channel(s) ?)

Kind regards
Thanks

NB: Oddly enough, when the local user hears the music on hold, his  
own

channel (a local SIP phone in this case) isn't reported as "On Hold"
when issuing "sip show channels" in cli,  and no AMI Hold/Unhold
events are generated. I double checked, the MOH that gets played is
the one specified in sip.conf, NOT zapata.conf.


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Re: [asterisk-users] app_rxfax.c and app_txxfax.c where?

[Doug]

At 23:19 1/7/2008, Klaverstyn, David C wrote:
>Content-Type: multipart/alternative;
> boundary="_=_NextPart_001_01C851B6.10468958"
>Content-class: urn:content-classes:message
>
>Hi All,
>
>Where can I find copies of the app_rxfax.c, app_txfax.c and 
>apps_Makefile.patch.  They don't seem to be located at soft-switch.org anymore.
>
>I am currently trying to compile Asterisk 1.2.26.1 and need the fax 
>components.
>
>Thanks.

Is this what you need?

 


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[asterisk-users] Early media support for Asterisk behind NAT

[Mayur]

Hi,

   I have asterisk 1.4.16 behind a NAT-FW which is using a hosted SIP trunk
for PSTN calling. Asterisk is configured to support nat with nat=yes in
sip.conf. Now the hosted PSTN Gateway supports symmetric RTP and early media
using 183 Session Progress. So If I call a PSTN number which has IVR message
played before the call is connected (via 183), those media RTP packets do
not reach the asterisk inside till asterisk sends out media packet to the
PSTN gateway. I have used rtpkeepalive option and set it to 1 sec. But it
seems that I drop rtp voice packets in the initial instructions played by
the IVR.

 

How do I make sure that asterisk sends RTP packets (null rtp) to the PSTN
gateway just after receiving the media details in 183 SDP to open the
firewall port from inside?

 

Regards,

Mayur   

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Re: [asterisk-users] pickup application failed

[dave cantera]

rilawich,
do you have the pickup group defined?
http://www.voip-info.org/wiki/view/Asterisk+callgroups+and+pickupgroups
daveC

Rilawich Ango wrote:
> Below is what I got from CLI
> [Jan  7 23:02:46] NOTICE[3450]: app_directed_pickup.c:159 pickup_exec:
> No target channel found for 111.
>
> On Jan 7, 2008 11:48 PM, Rilawich Ango <[EMAIL PROTECTED]> wrote:
>   
>> I have a TDM400 in the server.  I want to press **1XX to pickup a
>> call.  It is ok if I pickup a call dialled from 1XX to 1YY (internal
>> network call).  However, it is failed to pick up a call from PSTN
>> thro' TDM400 card.  It seems I can't guess the correct context of it.
>> How can I know the context of  the call using CLI?  The default
>> context of the TDM400 is from-pstn but pickup still failed if I add
>> exten => _**1XX,n,Pickup(${EXTEN:[EMAIL PROTECTED]) in the end of context
>> BLF_group_pickup.
>>
>> [BLF_group_pickup]
>> exten => _**1XX,1,Pickup(${EXTEN:[EMAIL PROTECTED])
>> exten => _**1XX,n,Pickup(${EXTEN:[EMAIL PROTECTED])
>>
>> 
>
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>   

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[asterisk-users] Distorted audio over Eicon Diva Server BRI

[CSB]

We are experiencing slightly distorted audio with playing of recordings on
our Asterisk server when the call comes in over our Eicon Diva Server BRI
card. An example is an incoming call to IVR and playing some of the standard
Asterisk voice prompts. Note that there is no audio problem with internal
access to the same recording. Neither is there a problem with calls not
involving the playing of recordings. The problem occurs consistently and is
not related to system load. According to Eicon support:

Asterisk is sending it's data packets (160 bytes each, i.e. 20ms) in too
large intervals. This causes the transmitter of the Diva Server card to
underrun and thus to fill with idle samples in regular intervals. It's
almost between any two packets where we have to insert samples.

0:00:29.710 CAPI20_PUT(030)

 0:00:29.730 CAPI20_PUT(030)

 0:00:29.751 CAPI20_PUT(030)

 0:00:29.771 CAPI20_PUT(030)

 0:00:29.791 CAPI20_PUT(030)

 0:00:29.812 CAPI20_PUT(030)

 0:00:29.832 CAPI20_PUT(030)

 0:00:29.853 CAPI20_PUT(030)

 0:00:29.873 CAPI20_PUT(030)

 0:00:29.894 CAPI20_PUT(030)

 

I wonder if anyone could provide any advice on how to continue
troubleshooting this issue? 

 

Regards

 

Cameron

 

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Re: [asterisk-users] help need

[Matt Riddell]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

pgck nirukshitha wrote:
> Hi All
> 
> We received  following  error  .Please help us to  sort  out. 
> 
> WARNING[3281]: frame.c:1426 speex_samples: Had error while reading wideband 
> frames for speex samples.

Um well for starters Asterisk kinda doesn't do wideband so you'd need to
do narrowband (unless speex downsamples it).

Russell and Kevin have been playing with bits of res_resample and I
noticed a couple of 16khz bits going in, but I'm not 100% sure where
that's going :)

First off I'd change the device you are using to use narrowband speex,
GSM, g711.Alaw or g.Ulaw etc.

- --
Kind Regards,

Matt Riddell
Director
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Re: [asterisk-users] conferencing help

[Paul Hales]

Then it's time to build zaptel, then rebuild asterisk

later,

PaulH


On Tue, 2008-01-08 at 13:16 +0800, Nhadie wrote:
> Hi Matt,
> 
> it seems i don't have that command.
> 
> *CLI> zap show channels
> No such command 'zap' (type 'help' for help)
> *CLI>
> !   abort   add ael agent   agi 
> cdr databasedebug   dnsmgr  dontdump 
> dundi
> extensions  feature group   helpiax2include 
> indication  initloadlocal   logger  meetme 
> mgcp
> mixmonitor  moh no  realtimereload  remove 
> restart rtp set showsip skinny 
> soft
> stopunload
> 
> *CLI> show channeltypes
> TypeDescriptionDevicestate  Indications 
> Transfer
> --  ------  --- 
> 
> Feature Feature Proxy Channel Driver   no   yes  no 
> 
> Agent   Call Agent Proxy Channel   yes  yes  no 
> 
> Local   Local Proxy Channel Driver no   yes  no 
> 
> Skinny  Skinny Client Control Protocol no   yes  no 
> 
> Phone   Standard Linux Telephony API D no   no   no 
> 
> SIP Session Initiation Protocol (S yes  yes  yes 
> 
> IAX2Inter Asterisk eXchange Driver yes  yes  yes 
> 
> MGCPMedia Gateway Control Protocol no   yes  no 
> 
> 
> *CLI> show channeltypes
> TypeDescriptionDevicestate  Indications 
> Transfer
> --  ------  --- 
> 
> Feature Feature Proxy Channel Driver   no   yes  no 
> 
> Agent   Call Agent Proxy Channel   yes  yes  no 
> 
> Local   Local Proxy Channel Driver no   yes  no 
> 
> Skinny  Skinny Client Control Protocol no   yes  no 
> 
> Phone   Standard Linux Telephony API D no   no   no 
> 
> SIP Session Initiation Protocol (S yes  yes  yes 
> 
> IAX2Inter Asterisk eXchange Driver yes  yes  yes 
> 
> MGCPMedia Gateway Control Protocol no   yes  no 
> 
> 
>  -- Executing NoOp("SIP/104-58ae", "Using CallerID "104" <104>") in 
> new stack
>  -- Executing Set("SIP/104-58ae", "MEETME_ROOMNUM=6000") in new stack
>  -- Executing GotoIf("SIP/104-58ae", "0?USER") in new stack
>  -- Executing Answer("SIP/104-58ae", "") in new stack
>  -- Executing Wait("SIP/104-58ae", "1") in new stack
>  -- Executing Set("SIP/104-58ae", "MEETME_OPTS=") in new stack
>  -- Executing Goto("SIP/104-58ae", "STARTMEETME|1") in new stack
>  -- Goto (from-internal,STARTMEETME,1)
>  -- Executing MeetMe("SIP/104-58ae", "6000||") in new stack
> 
> 
> 
> Matt Riddell wrote:
> > -BEGIN PGP SIGNED MESSAGE-
> > Hash: SHA1
> > 
> > Nhadie wrote:
> >> hi shane,
> >>
> >> thanks for your reply. i actually tried 3 phones dialled to the 
> >> conference, but cant here anything from those phones. i also enabled the 
> >> usercount so i can hear something at least. but still no sound.
> >> i'm using ztdummy, as i dont have a card yet.
> > 
> > Can you do a "zap show channels" in the Asterisk console (without the ")
> > 
> > - --
> > Kind Regards,
> > 
> > Matt Riddell
> > Director
> > ___
> > 
> > http://www.venturevoip.com (Great new VoIP end to end solution)
> > http://www.venturevoip.com/news.php (Daily Asterisk News - html)
> > http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss)
> > -BEGIN PGP SIGNATURE-
> > Version: GnuPG v1.4.7 (MingW32)
> > Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org
> > 
> > iD8DBQFHgtckDQNt8rg0Kp4RAkw0AJ0R/xZowCQ1FGVNpblcUrdwAi5niACfQ5jh
> > JEjcAt3QDqV3aN0rAZGNq9g=
> > =Zqs+
> > -END PGP SIGNATURE-
> > 
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> > 
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Re: [asterisk-users] conferencing help

[Matt Riddell]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Nhadie wrote:
> Hi Matt,
> 
> it seems i don't have that command.

:)

You'll need to make sure that:

1. You have zaptel compiled
2. You compile Asterisk *after* zaptel is compiled and installed
3. You have either modprobed zaptel + ztdummy or made the service and
started it.

You didn't say, is this a straight Asterisk machine or trixbox/freepbx?

If those are done and it still doesn't work then you can report the
errors you get when you type (in the console):

module load chan_zap.so

- --
Kind Regards,

Matt Riddell
Director
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Re: [asterisk-users] help need

[Alex Balashov]

Well, what would you have one say?

It is caused by the return of a failure value of the function 
speex_get_wb_sz_at() in frame.c, which attempts to extract bits
from speex frames.  Presumably from some sort of data corruption or
invalid format.

On Mon, 7 Jan 2008, pgck nirukshitha wrote:

> Hi All
>
> We received  following  error  .Please help us to  sort  out.
>
> WARNING[3281]: frame.c:1426 speex_samples: Had error while reading wideband 
> frames for speex samples.
>
>
> Regards
> Nirukshitha
>
>
>
>
>  
> 
> Looking for last minute shopping deals?
> Find them fast with Yahoo! Search.  
> http://tools.search.yahoo.com/newsearch/category.php?category=shopping
>
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[asterisk-users] app_rxfax.c and app_txxfax.c where?

[Klaverstyn, David C]

Hi All,

 

Where can I find copies of the app_rxfax.c, app_txfax.c and
apps_Makefile.patch.  They don't seem to be located at soft-switch.org
anymore.

 

I am currently trying to compile Asterisk 1.2.26.1 and need the fax
components.

 

Thanks.

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Re: [asterisk-users] Polycom IP4000 - Device does not match ACL

[CunningPike]

Try 'ip4000_1' instead of '207' for your address.

CP

Kevin DeGraaf wrote:
> I am trying to configure a Polycom IP4000 for use with Asterisk 1.4.5 on
> a flat local network.
> 
> I followed the provisioning guides that I found on the Web, and I have
> the phone downloading bootrom.ld, sip.ld, and a bunch of configuration
> files.  This all works properly.
> 
> However, I receive the following error:
> 
> NOTICE[27345]: chan_sip.c:14725 handle_request_register: Registration
> from '' failed for 'x.x.x.229' - Device does not match 
> ACL
> 
> I can place calls from the IP4000, but I cannot receive them:
> 
> WARNING[27480]: app_dial.c:1106 dial_exec_full: Unable to create channel
> of type 'SIP' (cause 3 - No route to destination)
> 
> Here are the relevant (IMHO) config sections.
> 
> == sip.conf ==
> [ip4000_1]
> [EMAIL PROTECTED]
> type=friend
> secret=password
> qualify=yes
> nat=no
> host=dynamic
> canreinvite=no
> 
> == Polycom per-phone config on TFTP server ==
> reg.1.displayName="207"
> reg.1.address="207"
> reg.1.label="207"
> reg.1.type="private"
> reg.1.lcs=""
> reg.1.thirdPartyName=""
> reg.1.auth.userId="ip4000_1"
> reg.1.auth.password="password"
> 
> == Polycom company-wide config on TFTP server ==
> server voIpProt.server.1.address="x.x.x.55"/>
> 
> 
> 
> 
> I've tried using x.x.x.55 as both the "proxy" value only, the "server"
> value only, and (in the given example) both.
> 
> I also added the following to sip.conf, to no avail:
> 
> deny=0.0.0.0/0.0.0.0
> permit=x.x.x.0/255.255.255.0
> 
> Any ideas about what I've missed would be appreciated.  Thanks.
> 

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Re: [asterisk-users] conferencing help

[Nhadie]

Hi Matt,

it seems i don't have that command.

*CLI> zap show channels
No such command 'zap' (type 'help' for help)
*CLI>
!   abort   add ael agent   agi 
cdr databasedebug   dnsmgr  dontdump 
dundi
extensions  feature group   helpiax2include 
indication  initloadlocal   logger  meetme 
mgcp
mixmonitor  moh no  realtimereload  remove 
restart rtp set showsip skinny 
soft
stopunload

*CLI> show channeltypes
TypeDescriptionDevicestate  Indications 
Transfer
--  ------  --- 

Feature Feature Proxy Channel Driver   no   yes  no 

Agent   Call Agent Proxy Channel   yes  yes  no 

Local   Local Proxy Channel Driver no   yes  no 

Skinny  Skinny Client Control Protocol no   yes  no 

Phone   Standard Linux Telephony API D no   no   no 

SIP Session Initiation Protocol (S yes  yes  yes 

IAX2Inter Asterisk eXchange Driver yes  yes  yes 

MGCPMedia Gateway Control Protocol no   yes  no 


*CLI> show channeltypes
TypeDescriptionDevicestate  Indications 
Transfer
--  ------  --- 

Feature Feature Proxy Channel Driver   no   yes  no 

Agent   Call Agent Proxy Channel   yes  yes  no 

Local   Local Proxy Channel Driver no   yes  no 

Skinny  Skinny Client Control Protocol no   yes  no 

Phone   Standard Linux Telephony API D no   no   no 

SIP Session Initiation Protocol (S yes  yes  yes 

IAX2Inter Asterisk eXchange Driver yes  yes  yes 

MGCPMedia Gateway Control Protocol no   yes  no 


 -- Executing NoOp("SIP/104-58ae", "Using CallerID "104" <104>") in 
new stack
 -- Executing Set("SIP/104-58ae", "MEETME_ROOMNUM=6000") in new stack
 -- Executing GotoIf("SIP/104-58ae", "0?USER") in new stack
 -- Executing Answer("SIP/104-58ae", "") in new stack
 -- Executing Wait("SIP/104-58ae", "1") in new stack
 -- Executing Set("SIP/104-58ae", "MEETME_OPTS=") in new stack
 -- Executing Goto("SIP/104-58ae", "STARTMEETME|1") in new stack
 -- Goto (from-internal,STARTMEETME,1)
 -- Executing MeetMe("SIP/104-58ae", "6000||") in new stack



Matt Riddell wrote:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
> 
> Nhadie wrote:
>> hi shane,
>>
>> thanks for your reply. i actually tried 3 phones dialled to the 
>> conference, but cant here anything from those phones. i also enabled the 
>> usercount so i can hear something at least. but still no sound.
>> i'm using ztdummy, as i dont have a card yet.
> 
> Can you do a "zap show channels" in the Asterisk console (without the ")
> 
> - --
> Kind Regards,
> 
> Matt Riddell
> Director
> ___
> 
> http://www.venturevoip.com (Great new VoIP end to end solution)
> http://www.venturevoip.com/news.php (Daily Asterisk News - html)
> http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss)
> -BEGIN PGP SIGNATURE-
> Version: GnuPG v1.4.7 (MingW32)
> Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org
> 
> iD8DBQFHgtckDQNt8rg0Kp4RAkw0AJ0R/xZowCQ1FGVNpblcUrdwAi5niACfQ5jh
> JEjcAt3QDqV3aN0rAZGNq9g=
> =Zqs+
> -END PGP SIGNATURE-
> 
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> 

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[asterisk-users] help need

[pgck nirukshitha]

Hi All

We received  following  error  .Please help us to  sort  out. 

WARNING[3281]: frame.c:1426 speex_samples: Had error while reading wideband 
frames for speex samples.


Regards
Nirukshitha




  

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

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Re: [asterisk-users] no outgoing calls with Digium B410P

[Tzafrir Cohen]

On Mon, Jan 07, 2008 at 04:53:03PM +0100, daniele visaggio wrote:
> 2008/1/7, map <[EMAIL PROTECTED]>:
> >
> > Hi Daniele,
> >
> > Please send a snapshot of your Putty Asterisk log.
> > Go to Putty configuration -> Window -> Lines of scrollback and put a
> > number greater than 200 :-). I suggest 10.
> >
> > Sorry, i'm using a Linux version of putty (i'm running Ubuntu) and the
> configuration is different. I can't find "Lines of scrollback" and modify
> the scrollback number. The putty-linux sw-structure is probably different
> from the putty-windows one.

Nag nag nag :-)

The menu in that version of putty is available by holding the ctrl key
and pressing the right mouse key. The right mouse key alone extends the
selection, like in xterm.

In that menu you have "copy all", as well as the ability to change
settings (scroll-buffer, log file, whatever).

Or jut select text with the mouse and it will be automatically copied.

Or use ssh from any other terminal, as others here have mentioned. I
normally use .ssh/config for maintaning aliases and special settings.

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

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Re: [asterisk-users] FWD and IPCall

[Shane D]

Oh. Well, I want to design a dialplan, and I don't care what number it
is, as long as you get my starting menu... So I'll try out the S
extension.

On 1/7/08, Huw Richards <[EMAIL PROTECTED]> wrote:
> You would have to try out the "s" extension yourself.
>
> I tend to have different contexts for each incoming number (as a home
> user, only one number at a provider) so I can potentially handle them
> differently i.e. time of day check.
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Shane D
> Sent: Monday, January 07, 2008 20:47
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] FWD and IPCall
>
> It works! It works! It works!
>
> I am able to talk to myself. Now all I have to do is write my
> dialplan...
>
> Say, would I have to use the [ipkallnumber] extention? Could I specify
> the "s" extension instead to catch multiple numbers?
>
> --
> -Shane
> Blog: http://blind-geek.com/blog/
> CoOwner: http://sjtechzone.com
> AIM: inhaddict
> Skype: chatter8712
>
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-- 
-Shane
Blog: http://blind-geek.com/blog/
CoOwner: http://sjtechzone.com
AIM: inhaddict
Skype: chatter8712

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Re: [asterisk-users] no outgoing calls with Digium B410P

[Lyle Giese]

daniele visaggio wrote:
>
>
> 2008/1/7, map <[EMAIL PROTECTED] >:
>
> Hi Daniele,
>
> Please send a snapshot of your Putty Asterisk log.
> Go to Putty configuration -> Window -> Lines of scrollback and put
> a number greater than 200 :-). I suggest 10.
>
> Sorry, i'm using a Linux version of putty (i'm running Ubuntu) and the
> configuration is different. I can't find "Lines of scrollback" and
> modify the scrollback number. The putty-linux sw-structure is probably
> different from the putty-windows one.
> 
>
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Since you are using linux, open an Xterm window and issue the following
command:

ssh -l  

This should prompt you to verify the ssh key the first time and then ask
for your password.  Cut and paste works from an XTerm window.

Lyle

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Re: [asterisk-users] FWD and IPCall

[Huw Richards]

You would have to try out the "s" extension yourself. 

I tend to have different contexts for each incoming number (as a home
user, only one number at a provider) so I can potentially handle them
differently i.e. time of day check. 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Shane D
Sent: Monday, January 07, 2008 20:47
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] FWD and IPCall

It works! It works! It works!

I am able to talk to myself. Now all I have to do is write my
dialplan...

Say, would I have to use the [ipkallnumber] extention? Could I specify
the "s" extension instead to catch multiple numbers?

-- 
-Shane
Blog: http://blind-geek.com/blog/
CoOwner: http://sjtechzone.com
AIM: inhaddict
Skype: chatter8712

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Re: [asterisk-users] FWD and IPCall

[Huw Richards]

I seem to recall the problem was related to DNS A vs SRV records. I
believe that dyndns at that time did not register SRV records on host
(i.e. free) accounts and ipkall was looking for an SRV record.

I know that an SRV record can be added on a paid account, but I still
don't think that you can add an SRV record to a free account - unless
all "A" records are automatically "SRV" records also. 

Or mayble ipkall gracefully falls back to search for an "A" record if an
"SRV" lookup fails?
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Riddell
Sent: Monday, January 07, 2008 20:51
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] FWD and IPCall

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Huw Richards wrote:
> You're right that dyndns.org offers the same type of services as
> no-ip.org.
> 
> However, when I first setup ipkall forwarding directly to my asterisk
> server (about a year ago), it would not work with a dyndns.org account
-
> I forget the reason why. Maybe ipkall works with dyndns now - I
haven't
> tried it in over a year.

Must have not been registered.

I have both the free and commercial services set up on around 200 Linux
boxes distributed around the world :)

- --
Kind Regards,

Matt Riddell
Director
___

http://www.venturevoip.com (Great new VoIP end to end solution)
http://www.venturevoip.com/news.php (Daily Asterisk News - html)
http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss)
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.7 (MingW32)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

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SL7gJyqd9ZvvOBhZ5MPXZQI=
=qB/r
-END PGP SIGNATURE-

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Re: [asterisk-users] :POSSIBLE SPAM: Re: :POSSIBLE SPAM: conferencing help

[Matt Riddell]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Nhadie wrote:
> hi shane,
> 
> thanks for your reply. i actually tried 3 phones dialled to the 
> conference, but cant here anything from those phones. i also enabled the 
> usercount so i can hear something at least. but still no sound.
> i'm using ztdummy, as i dont have a card yet.

Can you do a "zap show channels" in the Asterisk console (without the ")

- --
Kind Regards,

Matt Riddell
Director
___

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http://www.venturevoip.com/news.php (Daily Asterisk News - html)
http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss)
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.7 (MingW32)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

iD8DBQFHgtckDQNt8rg0Kp4RAkw0AJ0R/xZowCQ1FGVNpblcUrdwAi5niACfQ5jh
JEjcAt3QDqV3aN0rAZGNq9g=
=Zqs+
-END PGP SIGNATURE-

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Re: [asterisk-users] FWD and IPCall

[Matt Riddell]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Huw Richards wrote:
> You're right that dyndns.org offers the same type of services as
> no-ip.org.
> 
> However, when I first setup ipkall forwarding directly to my asterisk
> server (about a year ago), it would not work with a dyndns.org account -
> I forget the reason why. Maybe ipkall works with dyndns now - I haven't
> tried it in over a year.

Must have not been registered.

I have both the free and commercial services set up on around 200 Linux
boxes distributed around the world :)

- --
Kind Regards,

Matt Riddell
Director
___

http://www.venturevoip.com (Great new VoIP end to end solution)
http://www.venturevoip.com/news.php (Daily Asterisk News - html)
http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss)
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.7 (MingW32)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

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SL7gJyqd9ZvvOBhZ5MPXZQI=
=qB/r
-END PGP SIGNATURE-

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Re: [asterisk-users] FWD and IPCall

[Shane D]

It works! It works! It works!

I am able to talk to myself. Now all I have to do is write my dialplan...

Say, would I have to use the [ipkallnumber] extention? Could I specify
the "s" extension instead to catch multiple numbers?

-- 
-Shane
Blog: http://blind-geek.com/blog/
CoOwner: http://sjtechzone.com
AIM: inhaddict
Skype: chatter8712

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Re: [asterisk-users] pickup application failed

[Rilawich Ango]

Below is what I got from CLI
[Jan  7 23:02:46] NOTICE[3450]: app_directed_pickup.c:159 pickup_exec:
No target channel found for 111.

On Jan 7, 2008 11:48 PM, Rilawich Ango <[EMAIL PROTECTED]> wrote:
> I have a TDM400 in the server.  I want to press **1XX to pickup a
> call.  It is ok if I pickup a call dialled from 1XX to 1YY (internal
> network call).  However, it is failed to pick up a call from PSTN
> thro' TDM400 card.  It seems I can't guess the correct context of it.
> How can I know the context of  the call using CLI?  The default
> context of the TDM400 is from-pstn but pickup still failed if I add
> exten => _**1XX,n,Pickup(${EXTEN:[EMAIL PROTECTED]) in the end of context
> BLF_group_pickup.
>
> [BLF_group_pickup]
> exten => _**1XX,1,Pickup(${EXTEN:[EMAIL PROTECTED])
> exten => _**1XX,n,Pickup(${EXTEN:[EMAIL PROTECTED])
>

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Re: [asterisk-users] FWD and IPCall

[Huw Richards]

You're right that dyndns.org offers the same type of services as
no-ip.org.

However, when I first setup ipkall forwarding directly to my asterisk
server (about a year ago), it would not work with a dyndns.org account -
I forget the reason why. Maybe ipkall works with dyndns now - I haven't
tried it in over a year.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Riddell
Sent: Monday, January 07, 2008 20:11
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] FWD and IPCall

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Shane D wrote:
> no-ip.org appears to want to charge me money... Is there a free
alternative?

Dyndns.org

- --
Kind Regards,

Matt Riddell
Director
___

http://www.venturevoip.com (Great new VoIP end to end solution)
http://www.venturevoip.com/news.php (Daily Asterisk News - html)
http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss)
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.7 (MingW32)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

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3dc37G8IEH3qSQejV1UKATM=
=jBOo
-END PGP SIGNATURE-

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Re: [asterisk-users] FWD and IPCall

[Shane D]

I'm an idiot... I dialled wrong on my phone... I changed it, and was
able to use the Echo application. Dialling for a call to my softphone
as we speak!

On 1/7/08, Huw Richards <[EMAIL PROTECTED]> wrote:
> I think you said that you already had an ipkall account? If so, logon to
> the account and on the resulting screen there are 2 fields that you need
> to change:
>
> The first is the "SIP Phone Number" - if you have already tried to
> forward the IPKALL number to your fwd account, this field will contain
> your fwd number. You can change this to be anything - my convention is
> to use the actual phone number that IPKALL assigned me.
>
> The second is "SIP Proxy" - again, if you have tried to forward the
> IPKALL number to your fwd account, this field will contain
> "fwd.pulver.net" (or something similar). Change this value to the
> hostname you setup at no-ip.org.
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Shane D
> Sent: Monday, January 07, 2008 20:03
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] FWD and IPCall
>
> Okay. What do you mean in step 4/5 (I don't remember which) where you
> write something about "Use your IPKall number as the sip number" I am
> signing up for IPKall... Right?
>
> On 1/7/08, Shane D <[EMAIL PROTECTED]> wrote:
> > no-ip.org appears to want to charge me money... Is there a free
> alternative?
> >
> > On 1/7/08, Huw Richards <[EMAIL PROTECTED]> wrote:
> > >
> > > If you want to forward your ipkall number directly to your asterisk
> > > server:
> > >
> > > 1. If your asterisk server is on a private LAN and is connected to
> the
> > > internet via a router, enable the router to port forward UDP/5060 &
> > > UDP/1-2 to your asterisk server (assuming you have not
> changed
> > > rtp config parameters in rtp.conf).
> > >
> > > 2. Check that the firewall (if any) on your asterisk server allows
> > > connections on UDP/5060 & UDP/1-2
> > >
> > > 3a. Static public IP address - use the fully qualified domain name
> > > assigned to the IP address (or setup an account on www.no-ip.org
> with a
> > > name of your choice)
> > >
> > > 3b. Dynamic public IP address - setup an account on www.no-ip.org
> with a
> > > name of your choice - install the dynamic ip address update client
> to
> > > monitor any change of your ip address (downloads & instructions on
> > > no-ip.org website)
> > >
> > > 4. Goto www.ipkall.com and login to your account. Use your ipkall
> number
> > > as the SIP Phone Number and then the name you selected in 3a or 3b
> as
> > > the SIP Proxy.
> > >
> > > 5. Wait 60 minutes for changes to take affect (!)
> > >
> > > 6. Edit asterisk sip configuration to allow calls from ipkall:
> > >
> > > vi /etc/asterisk/sip.conf and find the section beginning [general]
> > >
> > > Add/replace the following:
> > >
> > > externhost=the name you setup in 3a. or 3b.
> > > localnet=your private LAN e.g. 192.168.2.0/255.255.255.0
> > >
> > > Add a new section at the bottom of the file:
> > >
> > > [ipkall.com]
> > > host=voiper.ipkall.com
> > > context=from-ipkall
> > > dtmfmode=rfc2833
> > > insecure=invite
> > > type=friend
> > > canreinvite=no
> > > disallow=all
> > > allow=ulaw ; you can add other codecs if you want once the setup
> works
> > >
> > > Save the file. The section you added tells asterisk to accept calls
> from
> > > voiper.ipkall.com and to place them in the "from-ipkall" context.
> This
> > > context can be whatever you want. You may need to change the
> insecure=
> > > line if you are using asterisk 1.2
> > >
> > > 7. Edit asterisk dialplan configuration to handle calls from ipkall:
> > >
> > > vi /etc/asterisk/extensions.conf and add at the bottom:
> > >
> > > [from-ipkall]
> > > exten => ,1,NoOp(from-ipkall)
> > > exten => ,2,NoOp(${CALLERIDNAME}/${CALLERIDNUM})
> > > exten => ,3,Dial(Local/[EMAIL PROTECTED])
> > >
> > > Save the file. The section you added tells asterisk what to do with
> > > calls that are received in the "from-ipkall" context. Replace the
> > >  with whatever you entered in the SIP Phone number
> field
> > > on the ipkall website (I recommended your ipkall number).
> > >
> > > In the "from-ipkall" section:
> > > 1: display "from-ipkall" on the console
> > > 2: display the caller id & name
> > > 3. phone the local extension 200 in context "local" - replace this
> line
> > > with your personal requirements.
> > >
> > > Connect to the asterisk console (asterisk -R on my server) and "sip
> > > reload" followed by "dialplan reload" (asterisk 1.4) or "extensions
> > > reload" (asterisk 1.2). "sip reload" will re-read the sip.conf file
> &
> > > "dialplan reload"/"extensions reload" will re-read the
> extensions.conf
> > > file.
> > >
> > > Phone your ipkall number and see if anything is displayed on the
> console
> > > and/or your phone rings.
> > >
> > > If nothing on the console when you phone, try "sip set debug peer
> > > ipkall.com" (as

Re: [asterisk-users] asterisk CLI and no such command "stop"

[Vieri]

--- Vieri <[EMAIL PROTECTED]> wrote:

> *CLI> stop
> No such command 'stop' (type 'help' for help)

There was a config error on my behalf in the zapata
config and that somehow didn't stop asterisk from
loading but without the stop and zap commands.
Solved.



  

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Re: [asterisk-users] FWD and IPCall

[Huw Richards]

I think you said that you already had an ipkall account? If so, logon to
the account and on the resulting screen there are 2 fields that you need
to change:

The first is the "SIP Phone Number" - if you have already tried to
forward the IPKALL number to your fwd account, this field will contain
your fwd number. You can change this to be anything - my convention is
to use the actual phone number that IPKALL assigned me.

The second is "SIP Proxy" - again, if you have tried to forward the
IPKALL number to your fwd account, this field will contain
"fwd.pulver.net" (or something similar). Change this value to the
hostname you setup at no-ip.org. 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Shane D
Sent: Monday, January 07, 2008 20:03
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] FWD and IPCall

Okay. What do you mean in step 4/5 (I don't remember which) where you
write something about "Use your IPKall number as the sip number" I am
signing up for IPKall... Right?

On 1/7/08, Shane D <[EMAIL PROTECTED]> wrote:
> no-ip.org appears to want to charge me money... Is there a free
alternative?
>
> On 1/7/08, Huw Richards <[EMAIL PROTECTED]> wrote:
> >
> > If you want to forward your ipkall number directly to your asterisk
> > server:
> >
> > 1. If your asterisk server is on a private LAN and is connected to
the
> > internet via a router, enable the router to port forward UDP/5060 &
> > UDP/1-2 to your asterisk server (assuming you have not
changed
> > rtp config parameters in rtp.conf).
> >
> > 2. Check that the firewall (if any) on your asterisk server allows
> > connections on UDP/5060 & UDP/1-2
> >
> > 3a. Static public IP address - use the fully qualified domain name
> > assigned to the IP address (or setup an account on www.no-ip.org
with a
> > name of your choice)
> >
> > 3b. Dynamic public IP address - setup an account on www.no-ip.org
with a
> > name of your choice - install the dynamic ip address update client
to
> > monitor any change of your ip address (downloads & instructions on
> > no-ip.org website)
> >
> > 4. Goto www.ipkall.com and login to your account. Use your ipkall
number
> > as the SIP Phone Number and then the name you selected in 3a or 3b
as
> > the SIP Proxy.
> >
> > 5. Wait 60 minutes for changes to take affect (!)
> >
> > 6. Edit asterisk sip configuration to allow calls from ipkall:
> >
> > vi /etc/asterisk/sip.conf and find the section beginning [general]
> >
> > Add/replace the following:
> >
> > externhost=the name you setup in 3a. or 3b.
> > localnet=your private LAN e.g. 192.168.2.0/255.255.255.0
> >
> > Add a new section at the bottom of the file:
> >
> > [ipkall.com]
> > host=voiper.ipkall.com
> > context=from-ipkall
> > dtmfmode=rfc2833
> > insecure=invite
> > type=friend
> > canreinvite=no
> > disallow=all
> > allow=ulaw ; you can add other codecs if you want once the setup
works
> >
> > Save the file. The section you added tells asterisk to accept calls
from
> > voiper.ipkall.com and to place them in the "from-ipkall" context.
This
> > context can be whatever you want. You may need to change the
insecure=
> > line if you are using asterisk 1.2
> >
> > 7. Edit asterisk dialplan configuration to handle calls from ipkall:
> >
> > vi /etc/asterisk/extensions.conf and add at the bottom:
> >
> > [from-ipkall]
> > exten => ,1,NoOp(from-ipkall)
> > exten => ,2,NoOp(${CALLERIDNAME}/${CALLERIDNUM})
> > exten => ,3,Dial(Local/[EMAIL PROTECTED])
> >
> > Save the file. The section you added tells asterisk what to do with
> > calls that are received in the "from-ipkall" context. Replace the
> >  with whatever you entered in the SIP Phone number
field
> > on the ipkall website (I recommended your ipkall number).
> >
> > In the "from-ipkall" section:
> > 1: display "from-ipkall" on the console
> > 2: display the caller id & name
> > 3. phone the local extension 200 in context "local" - replace this
line
> > with your personal requirements.
> >
> > Connect to the asterisk console (asterisk -R on my server) and "sip
> > reload" followed by "dialplan reload" (asterisk 1.4) or "extensions
> > reload" (asterisk 1.2). "sip reload" will re-read the sip.conf file
&
> > "dialplan reload"/"extensions reload" will re-read the
extensions.conf
> > file.
> >
> > Phone your ipkall number and see if anything is displayed on the
console
> > and/or your phone rings.
> >
> > If nothing on the console when you phone, try "sip set debug peer
> > ipkall.com" (asterisk 1.4 - not sure of the command for asterisk
1.2)
> > and phone again.
> >
> > Post back your results.
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of Shane
D
> > Sent: Monday, January 07, 2008 17:32
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [asterisk-users] FWD and IPCall
> >
> > Okay... That was kind of confusing

Re: [asterisk-users] FWD and IPCall

[Shane D]

I did everything, and when I dial, nothing comes up in the consol,
nothing rings, and the phone says "I'm sorry, but the person you are
trying to call has a mailbox that has not been configured yet. Good
bye."

What's wrong?

On 1/7/08, Shane D <[EMAIL PROTECTED]> wrote:
> Okay. What do you mean in step 4/5 (I don't remember which) where you
> write something about "Use your IPKall number as the sip number" I am
> signing up for IPKall... Right?
>
> On 1/7/08, Shane D <[EMAIL PROTECTED]> wrote:
> > no-ip.org appears to want to charge me money... Is there a free
> alternative?
> >
> > On 1/7/08, Huw Richards <[EMAIL PROTECTED]> wrote:
> > >
> > > If you want to forward your ipkall number directly to your asterisk
> > > server:
> > >
> > > 1. If your asterisk server is on a private LAN and is connected to the
> > > internet via a router, enable the router to port forward UDP/5060 &
> > > UDP/1-2 to your asterisk server (assuming you have not changed
> > > rtp config parameters in rtp.conf).
> > >
> > > 2. Check that the firewall (if any) on your asterisk server allows
> > > connections on UDP/5060 & UDP/1-2
> > >
> > > 3a. Static public IP address - use the fully qualified domain name
> > > assigned to the IP address (or setup an account on www.no-ip.org with a
> > > name of your choice)
> > >
> > > 3b. Dynamic public IP address - setup an account on www.no-ip.org with a
> > > name of your choice - install the dynamic ip address update client to
> > > monitor any change of your ip address (downloads & instructions on
> > > no-ip.org website)
> > >
> > > 4. Goto www.ipkall.com and login to your account. Use your ipkall number
> > > as the SIP Phone Number and then the name you selected in 3a or 3b as
> > > the SIP Proxy.
> > >
> > > 5. Wait 60 minutes for changes to take affect (!)
> > >
> > > 6. Edit asterisk sip configuration to allow calls from ipkall:
> > >
> > > vi /etc/asterisk/sip.conf and find the section beginning [general]
> > >
> > > Add/replace the following:
> > >
> > > externhost=the name you setup in 3a. or 3b.
> > > localnet=your private LAN e.g. 192.168.2.0/255.255.255.0
> > >
> > > Add a new section at the bottom of the file:
> > >
> > > [ipkall.com]
> > > host=voiper.ipkall.com
> > > context=from-ipkall
> > > dtmfmode=rfc2833
> > > insecure=invite
> > > type=friend
> > > canreinvite=no
> > > disallow=all
> > > allow=ulaw ; you can add other codecs if you want once the setup works
> > >
> > > Save the file. The section you added tells asterisk to accept calls from
> > > voiper.ipkall.com and to place them in the "from-ipkall" context. This
> > > context can be whatever you want. You may need to change the insecure=
> > > line if you are using asterisk 1.2
> > >
> > > 7. Edit asterisk dialplan configuration to handle calls from ipkall:
> > >
> > > vi /etc/asterisk/extensions.conf and add at the bottom:
> > >
> > > [from-ipkall]
> > > exten => ,1,NoOp(from-ipkall)
> > > exten => ,2,NoOp(${CALLERIDNAME}/${CALLERIDNUM})
> > > exten => ,3,Dial(Local/[EMAIL PROTECTED])
> > >
> > > Save the file. The section you added tells asterisk what to do with
> > > calls that are received in the "from-ipkall" context. Replace the
> > >  with whatever you entered in the SIP Phone number field
> > > on the ipkall website (I recommended your ipkall number).
> > >
> > > In the "from-ipkall" section:
> > > 1: display "from-ipkall" on the console
> > > 2: display the caller id & name
> > > 3. phone the local extension 200 in context "local" - replace this line
> > > with your personal requirements.
> > >
> > > Connect to the asterisk console (asterisk -R on my server) and "sip
> > > reload" followed by "dialplan reload" (asterisk 1.4) or "extensions
> > > reload" (asterisk 1.2). "sip reload" will re-read the sip.conf file &
> > > "dialplan reload"/"extensions reload" will re-read the extensions.conf
> > > file.
> > >
> > > Phone your ipkall number and see if anything is displayed on the console
> > > and/or your phone rings.
> > >
> > > If nothing on the console when you phone, try "sip set debug peer
> > > ipkall.com" (asterisk 1.4 - not sure of the command for asterisk 1.2)
> > > and phone again.
> > >
> > > Post back your results.
> > >
> > >
> > >
> > >
> > >
> > >
> > >
> > >
> > >
> > >
> > >
> > > -Original Message-
> > > From: [EMAIL PROTECTED]
> > > [mailto:[EMAIL PROTECTED] On Behalf Of Shane D
> > > Sent: Monday, January 07, 2008 17:32
> > > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > > Subject: Re: [asterisk-users] FWD and IPCall
> > >
> > > Okay... That was kind of confusing. Would you contact me off-list to
> > > help me specifically?
> > >
> > > I've double-checked everything for the IAX, and it's a no-go. Maybe
> > > I'll try this SIP thing. But then again, if I can just hook IPKall to
> > > the server directly, I don't need FWD...
> > >
> > > On 1/7/08, Huw Richards <[EMAIL PROTECTED]> wrote:
> > > > My config is as follows
> > > >
> > > > Excerpt of sip.

[asterisk-users] :POSSIBLE SPAM: Re: :POSSIBLE SPAM: conferencing help

[Nhadie]

hi shane,

thanks for your reply. i actually tried 3 phones dialled to the 
conference, but cant here anything from those phones. i also enabled the 
usercount so i can hear something at least. but still no sound.
i'm using ztdummy, as i dont have a card yet.

regards,
nhadie

Shane D wrote:
> Wouldn't you need someone besides yourself in the conference?
> 
> On 1/7/08, Nhadie <[EMAIL PROTECTED]> wrote:
>>
>>
>> Hi All,
>>
>> kind of need help on the conference module, i'm using freepbx and
>> enabled conferencing, i created a conference number, 6000. when i dial
>> to it, my phone says it is connected but i'm hearing nothing, maybe logs
>> below can help you.
>>
>> also, when i hang up the phone, the conference did not disconnect me.
>> how can i end a conference? thank you
>>
>>  -- Executing Macro("SIP/104-519e", "user-callerid|") in new stack
>>  -- Executing NoOp("SIP/104-519e", "user-callerid: device 104") in
>> new stack
>>  -- Executing Set("SIP/104-519e", "AMPUSER=104") in new stack
>>  -- Executing GotoIf("SIP/104-519e", "0?report") in new stack
>>  -- Executing GotoIf("SIP/104-519e", "0?start") in new stack
>>  -- Executing Set("SIP/104-519e", "REALCALLERIDNUM=104") in new stack
>>  -- Executing NoOp("SIP/104-519e", "REALCALLERIDNUM is 104") in new
>> stack
>>  -- Executing Set("SIP/104-519e", "AMPUSER=104") in new stack
>>  -- Executing Set("SIP/104-519e", "AMPUSERCIDNAME=104") in new stack
>>  -- Executing GotoIf("SIP/104-519e", "0?report") in new stack
>>  -- Executing Set("SIP/104-519e", "AMPUSERCID=104") in new stack
>>  -- Executing Set("SIP/104-519e", "CALLERID(all)="104" <104>") in
>> new stack
>>  -- Executing Set("SIP/104-519e", "REALCALLERIDNUM=104") in new stack
>>  -- Executing NoOp("SIP/104-519e", "TTL:  ARG1: ") in new stack
>>  -- Executing GotoIf("SIP/104-519e", "0?continue") in new stack
>>  -- Executing Set("SIP/104-519e", "__TTL=64") in new stack
>>  -- Executing GotoIf("SIP/104-519e", "1?continue") in new stack
>>  -- Goto (macro-user-callerid,s,23)
>>  -- Executing NoOp("SIP/104-519e", "Using CallerID "104" <104>") in
>> new stack
>>  -- Executing Set("SIP/104-519e", "MEETME_ROOMNUM=6000") in new stack
>>  -- Executing GotoIf("SIP/104-519e", "0?USER") in new stack
>>  -- Executing Answer("SIP/104-519e", "") in new stack
>>  -- Executing Wait("SIP/104-519e", "1") in new stack
>>  -- Executing Set("SIP/104-519e", "MEETME_OPTS=") in new stack
>>  -- Executing Goto("SIP/104-519e", "STARTMEETME|1") in new stack
>>  -- Goto (from-internal,STARTMEETME,1)
>>  -- Executing MeetMe("SIP/104-519e", "6000||") in new stack
>>
>>
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>>
> 
> 

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Re: [asterisk-users] :POSSIBLE SPAM: conferencing help

[Matt Riddell]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Shane D wrote:
> Wouldn't you need someone besides yourself in the conference?

Indeed, judging by the logs (last line) you are actually in a
conference, you'll need to get someone else to call the same number to
be able to talk to them.

Alternatively pass the m option (think its m) to play music on hold when
there are no users in the conference.

- --
Kind Regards,

Matt Riddell
Director
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-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.7 (MingW32)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

iD8DBQFHgs4ODQNt8rg0Kp4RAjgqAJwP8qRfl6cmCyUY77hyZCSlPLI7vACZASeQ
YpzkxmKZF0icCS4QlHjCOB0=
=Ek8Q
-END PGP SIGNATURE-

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Re: [asterisk-users] Media gateways and video

[Matt Riddell]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Paul Hales wrote:
> My question would actually be - is there any support for h234 over ISDN?

Yep, but best place to ask about it is the asterisk-video mailing list.

You'll probably want to check out the work over at

sip.fontventa.com


- --
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Matt Riddell
Director
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Version: GnuPG v1.4.7 (MingW32)
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uMeb6LeOj5HEFudSvK9WbIg=
=XB+G
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Re: [asterisk-users] FWD and IPCall

[Matt Riddell]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Shane D wrote:
> no-ip.org appears to want to charge me money... Is there a free alternative?

Dyndns.org

- --
Kind Regards,

Matt Riddell
Director
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-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.7 (MingW32)
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3dc37G8IEH3qSQejV1UKATM=
=jBOo
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Re: [asterisk-users] FWD and IPCall

[Shane D]

Okay. What do you mean in step 4/5 (I don't remember which) where you
write something about "Use your IPKall number as the sip number" I am
signing up for IPKall... Right?

On 1/7/08, Shane D <[EMAIL PROTECTED]> wrote:
> no-ip.org appears to want to charge me money... Is there a free alternative?
>
> On 1/7/08, Huw Richards <[EMAIL PROTECTED]> wrote:
> >
> > If you want to forward your ipkall number directly to your asterisk
> > server:
> >
> > 1. If your asterisk server is on a private LAN and is connected to the
> > internet via a router, enable the router to port forward UDP/5060 &
> > UDP/1-2 to your asterisk server (assuming you have not changed
> > rtp config parameters in rtp.conf).
> >
> > 2. Check that the firewall (if any) on your asterisk server allows
> > connections on UDP/5060 & UDP/1-2
> >
> > 3a. Static public IP address - use the fully qualified domain name
> > assigned to the IP address (or setup an account on www.no-ip.org with a
> > name of your choice)
> >
> > 3b. Dynamic public IP address - setup an account on www.no-ip.org with a
> > name of your choice - install the dynamic ip address update client to
> > monitor any change of your ip address (downloads & instructions on
> > no-ip.org website)
> >
> > 4. Goto www.ipkall.com and login to your account. Use your ipkall number
> > as the SIP Phone Number and then the name you selected in 3a or 3b as
> > the SIP Proxy.
> >
> > 5. Wait 60 minutes for changes to take affect (!)
> >
> > 6. Edit asterisk sip configuration to allow calls from ipkall:
> >
> > vi /etc/asterisk/sip.conf and find the section beginning [general]
> >
> > Add/replace the following:
> >
> > externhost=the name you setup in 3a. or 3b.
> > localnet=your private LAN e.g. 192.168.2.0/255.255.255.0
> >
> > Add a new section at the bottom of the file:
> >
> > [ipkall.com]
> > host=voiper.ipkall.com
> > context=from-ipkall
> > dtmfmode=rfc2833
> > insecure=invite
> > type=friend
> > canreinvite=no
> > disallow=all
> > allow=ulaw ; you can add other codecs if you want once the setup works
> >
> > Save the file. The section you added tells asterisk to accept calls from
> > voiper.ipkall.com and to place them in the "from-ipkall" context. This
> > context can be whatever you want. You may need to change the insecure=
> > line if you are using asterisk 1.2
> >
> > 7. Edit asterisk dialplan configuration to handle calls from ipkall:
> >
> > vi /etc/asterisk/extensions.conf and add at the bottom:
> >
> > [from-ipkall]
> > exten => ,1,NoOp(from-ipkall)
> > exten => ,2,NoOp(${CALLERIDNAME}/${CALLERIDNUM})
> > exten => ,3,Dial(Local/[EMAIL PROTECTED])
> >
> > Save the file. The section you added tells asterisk what to do with
> > calls that are received in the "from-ipkall" context. Replace the
> >  with whatever you entered in the SIP Phone number field
> > on the ipkall website (I recommended your ipkall number).
> >
> > In the "from-ipkall" section:
> > 1: display "from-ipkall" on the console
> > 2: display the caller id & name
> > 3. phone the local extension 200 in context "local" - replace this line
> > with your personal requirements.
> >
> > Connect to the asterisk console (asterisk -R on my server) and "sip
> > reload" followed by "dialplan reload" (asterisk 1.4) or "extensions
> > reload" (asterisk 1.2). "sip reload" will re-read the sip.conf file &
> > "dialplan reload"/"extensions reload" will re-read the extensions.conf
> > file.
> >
> > Phone your ipkall number and see if anything is displayed on the console
> > and/or your phone rings.
> >
> > If nothing on the console when you phone, try "sip set debug peer
> > ipkall.com" (asterisk 1.4 - not sure of the command for asterisk 1.2)
> > and phone again.
> >
> > Post back your results.
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of Shane D
> > Sent: Monday, January 07, 2008 17:32
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [asterisk-users] FWD and IPCall
> >
> > Okay... That was kind of confusing. Would you contact me off-list to
> > help me specifically?
> >
> > I've double-checked everything for the IAX, and it's a no-go. Maybe
> > I'll try this SIP thing. But then again, if I can just hook IPKall to
> > the server directly, I don't need FWD...
> >
> > On 1/7/08, Huw Richards <[EMAIL PROTECTED]> wrote:
> > > My config is as follows
> > >
> > > Excerpt of sip.conf:
> > >
> > > [general]
> > > externhost=fully.qualified.domain.name
> > > localnet=192.168.2.0/255.255.255.0
> > > srvlookup=no
> > > defaultexpiry=3600
> > > dtmfmode=rfc2833
> > >
> > > register => :@fwd.pulver.com/
> > >
> > > [sipfwd]
> > > type=peer
> > > secret=
> > > username=
> > > fromdomain=fwd.pulver.com
> > > host=fwd.pulver.com
> > > disallow=all
> > > allow=ulaw
> > > canreinvite=yes
> > > insecure=invite
> > > qualify=yes
> > > context=from-fwd
> > >
> > > Excerpt of extensions.conf:
> > >
> > > 

Re: [asterisk-users] FWD and IPCall

[Huw Richards]

http://www.no-ip.com/services/managed_dns/free_dynamic_dns.html 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Shane D
Sent: Monday, January 07, 2008 19:50
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] FWD and IPCall

no-ip.org appears to want to charge me money... Is there a free
alternative?

On 1/7/08, Huw Richards <[EMAIL PROTECTED]> wrote:
>
> If you want to forward your ipkall number directly to your asterisk
> server:
>
> 1. If your asterisk server is on a private LAN and is connected to the
> internet via a router, enable the router to port forward UDP/5060 &
> UDP/1-2 to your asterisk server (assuming you have not changed
> rtp config parameters in rtp.conf).
>
> 2. Check that the firewall (if any) on your asterisk server allows
> connections on UDP/5060 & UDP/1-2
>
> 3a. Static public IP address - use the fully qualified domain name
> assigned to the IP address (or setup an account on www.no-ip.org with
a
> name of your choice)
>
> 3b. Dynamic public IP address - setup an account on www.no-ip.org with
a
> name of your choice - install the dynamic ip address update client to
> monitor any change of your ip address (downloads & instructions on
> no-ip.org website)
>
> 4. Goto www.ipkall.com and login to your account. Use your ipkall
number
> as the SIP Phone Number and then the name you selected in 3a or 3b as
> the SIP Proxy.
>
> 5. Wait 60 minutes for changes to take affect (!)
>
> 6. Edit asterisk sip configuration to allow calls from ipkall:
>
> vi /etc/asterisk/sip.conf and find the section beginning [general]
>
> Add/replace the following:
>
> externhost=the name you setup in 3a. or 3b.
> localnet=your private LAN e.g. 192.168.2.0/255.255.255.0
>
> Add a new section at the bottom of the file:
>
> [ipkall.com]
> host=voiper.ipkall.com
> context=from-ipkall
> dtmfmode=rfc2833
> insecure=invite
> type=friend
> canreinvite=no
> disallow=all
> allow=ulaw ; you can add other codecs if you want once the setup works
>
> Save the file. The section you added tells asterisk to accept calls
from
> voiper.ipkall.com and to place them in the "from-ipkall" context. This
> context can be whatever you want. You may need to change the insecure=
> line if you are using asterisk 1.2
>
> 7. Edit asterisk dialplan configuration to handle calls from ipkall:
>
> vi /etc/asterisk/extensions.conf and add at the bottom:
>
> [from-ipkall]
> exten => ,1,NoOp(from-ipkall)
> exten => ,2,NoOp(${CALLERIDNAME}/${CALLERIDNUM})
> exten => ,3,Dial(Local/[EMAIL PROTECTED])
>
> Save the file. The section you added tells asterisk what to do with
> calls that are received in the "from-ipkall" context. Replace the
>  with whatever you entered in the SIP Phone number
field
> on the ipkall website (I recommended your ipkall number).
>
> In the "from-ipkall" section:
> 1: display "from-ipkall" on the console
> 2: display the caller id & name
> 3. phone the local extension 200 in context "local" - replace this
line
> with your personal requirements.
>
> Connect to the asterisk console (asterisk -R on my server) and "sip
> reload" followed by "dialplan reload" (asterisk 1.4) or "extensions
> reload" (asterisk 1.2). "sip reload" will re-read the sip.conf file &
> "dialplan reload"/"extensions reload" will re-read the extensions.conf
> file.
>
> Phone your ipkall number and see if anything is displayed on the
console
> and/or your phone rings.
>
> If nothing on the console when you phone, try "sip set debug peer
> ipkall.com" (asterisk 1.4 - not sure of the command for asterisk 1.2)
> and phone again.
>
> Post back your results.
>
>
>
>
>
>
>
>
>
>
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Shane D
> Sent: Monday, January 07, 2008 17:32
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] FWD and IPCall
>
> Okay... That was kind of confusing. Would you contact me off-list to
> help me specifically?
>
> I've double-checked everything for the IAX, and it's a no-go. Maybe
> I'll try this SIP thing. But then again, if I can just hook IPKall to
> the server directly, I don't need FWD...
>
> On 1/7/08, Huw Richards <[EMAIL PROTECTED]> wrote:
> > My config is as follows
> >
> > Excerpt of sip.conf:
> >
> > [general]
> > externhost=fully.qualified.domain.name
> > localnet=192.168.2.0/255.255.255.0
> > srvlookup=no
> > defaultexpiry=3600
> > dtmfmode=rfc2833
> >
> > register => :@fwd.pulver.com/
> >
> > [sipfwd]
> > type=peer
> > secret=
> > username=
> > fromdomain=fwd.pulver.com
> > host=fwd.pulver.com
> > disallow=all
> > allow=ulaw
> > canreinvite=yes
> > insecure=invite
> > qualify=yes
> > context=from-fwd
> >
> > Excerpt of extensions.conf:
> >
> > [from-fwd]
> > exten => ,1,NoOp(from-fwd)
> > exten => ,n,Dial(whatever)
> >
> > I have a dynamic public IP address, so I use http://www.no-ip.org to
> map
> > my IP address to name. My router port for

Re: [asterisk-users] FWD and IPCall

[Shane D]

no-ip.org appears to want to charge me money... Is there a free alternative?

On 1/7/08, Huw Richards <[EMAIL PROTECTED]> wrote:
>
> If you want to forward your ipkall number directly to your asterisk
> server:
>
> 1. If your asterisk server is on a private LAN and is connected to the
> internet via a router, enable the router to port forward UDP/5060 &
> UDP/1-2 to your asterisk server (assuming you have not changed
> rtp config parameters in rtp.conf).
>
> 2. Check that the firewall (if any) on your asterisk server allows
> connections on UDP/5060 & UDP/1-2
>
> 3a. Static public IP address - use the fully qualified domain name
> assigned to the IP address (or setup an account on www.no-ip.org with a
> name of your choice)
>
> 3b. Dynamic public IP address - setup an account on www.no-ip.org with a
> name of your choice - install the dynamic ip address update client to
> monitor any change of your ip address (downloads & instructions on
> no-ip.org website)
>
> 4. Goto www.ipkall.com and login to your account. Use your ipkall number
> as the SIP Phone Number and then the name you selected in 3a or 3b as
> the SIP Proxy.
>
> 5. Wait 60 minutes for changes to take affect (!)
>
> 6. Edit asterisk sip configuration to allow calls from ipkall:
>
> vi /etc/asterisk/sip.conf and find the section beginning [general]
>
> Add/replace the following:
>
> externhost=the name you setup in 3a. or 3b.
> localnet=your private LAN e.g. 192.168.2.0/255.255.255.0
>
> Add a new section at the bottom of the file:
>
> [ipkall.com]
> host=voiper.ipkall.com
> context=from-ipkall
> dtmfmode=rfc2833
> insecure=invite
> type=friend
> canreinvite=no
> disallow=all
> allow=ulaw ; you can add other codecs if you want once the setup works
>
> Save the file. The section you added tells asterisk to accept calls from
> voiper.ipkall.com and to place them in the "from-ipkall" context. This
> context can be whatever you want. You may need to change the insecure=
> line if you are using asterisk 1.2
>
> 7. Edit asterisk dialplan configuration to handle calls from ipkall:
>
> vi /etc/asterisk/extensions.conf and add at the bottom:
>
> [from-ipkall]
> exten => ,1,NoOp(from-ipkall)
> exten => ,2,NoOp(${CALLERIDNAME}/${CALLERIDNUM})
> exten => ,3,Dial(Local/[EMAIL PROTECTED])
>
> Save the file. The section you added tells asterisk what to do with
> calls that are received in the "from-ipkall" context. Replace the
>  with whatever you entered in the SIP Phone number field
> on the ipkall website (I recommended your ipkall number).
>
> In the "from-ipkall" section:
> 1: display "from-ipkall" on the console
> 2: display the caller id & name
> 3. phone the local extension 200 in context "local" - replace this line
> with your personal requirements.
>
> Connect to the asterisk console (asterisk -R on my server) and "sip
> reload" followed by "dialplan reload" (asterisk 1.4) or "extensions
> reload" (asterisk 1.2). "sip reload" will re-read the sip.conf file &
> "dialplan reload"/"extensions reload" will re-read the extensions.conf
> file.
>
> Phone your ipkall number and see if anything is displayed on the console
> and/or your phone rings.
>
> If nothing on the console when you phone, try "sip set debug peer
> ipkall.com" (asterisk 1.4 - not sure of the command for asterisk 1.2)
> and phone again.
>
> Post back your results.
>
>
>
>
>
>
>
>
>
>
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Shane D
> Sent: Monday, January 07, 2008 17:32
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] FWD and IPCall
>
> Okay... That was kind of confusing. Would you contact me off-list to
> help me specifically?
>
> I've double-checked everything for the IAX, and it's a no-go. Maybe
> I'll try this SIP thing. But then again, if I can just hook IPKall to
> the server directly, I don't need FWD...
>
> On 1/7/08, Huw Richards <[EMAIL PROTECTED]> wrote:
> > My config is as follows
> >
> > Excerpt of sip.conf:
> >
> > [general]
> > externhost=fully.qualified.domain.name
> > localnet=192.168.2.0/255.255.255.0
> > srvlookup=no
> > defaultexpiry=3600
> > dtmfmode=rfc2833
> >
> > register => :@fwd.pulver.com/
> >
> > [sipfwd]
> > type=peer
> > secret=
> > username=
> > fromdomain=fwd.pulver.com
> > host=fwd.pulver.com
> > disallow=all
> > allow=ulaw
> > canreinvite=yes
> > insecure=invite
> > qualify=yes
> > context=from-fwd
> >
> > Excerpt of extensions.conf:
> >
> > [from-fwd]
> > exten => ,1,NoOp(from-fwd)
> > exten => ,n,Dial(whatever)
> >
> > I have a dynamic public IP address, so I use http://www.no-ip.org to
> map
> > my IP address to name. My router port forwards UDP/5060 &
> > UDP/1-2 to the internal asterisk server.
> >
> > However, I do not have ipkall forwarding to my fwd account. I have it
> > forwarding directly to my asterisk server using the no-ip.org address
> I
> > registered.
> >
> > e.g. forward to sip:[EMAIL PROTECTED] on ipkall website
>

Re: [asterisk-users] :POSSIBLE SPAM: conferencing help

[Shane D]

Wouldn't you need someone besides yourself in the conference?

On 1/7/08, Nhadie <[EMAIL PROTECTED]> wrote:
>
>
>
> Hi All,
>
> kind of need help on the conference module, i'm using freepbx and
> enabled conferencing, i created a conference number, 6000. when i dial
> to it, my phone says it is connected but i'm hearing nothing, maybe logs
> below can help you.
>
> also, when i hang up the phone, the conference did not disconnect me.
> how can i end a conference? thank you
>
>  -- Executing Macro("SIP/104-519e", "user-callerid|") in new stack
>  -- Executing NoOp("SIP/104-519e", "user-callerid: device 104") in
> new stack
>  -- Executing Set("SIP/104-519e", "AMPUSER=104") in new stack
>  -- Executing GotoIf("SIP/104-519e", "0?report") in new stack
>  -- Executing GotoIf("SIP/104-519e", "0?start") in new stack
>  -- Executing Set("SIP/104-519e", "REALCALLERIDNUM=104") in new stack
>  -- Executing NoOp("SIP/104-519e", "REALCALLERIDNUM is 104") in new
> stack
>  -- Executing Set("SIP/104-519e", "AMPUSER=104") in new stack
>  -- Executing Set("SIP/104-519e", "AMPUSERCIDNAME=104") in new stack
>  -- Executing GotoIf("SIP/104-519e", "0?report") in new stack
>  -- Executing Set("SIP/104-519e", "AMPUSERCID=104") in new stack
>  -- Executing Set("SIP/104-519e", "CALLERID(all)="104" <104>") in
> new stack
>  -- Executing Set("SIP/104-519e", "REALCALLERIDNUM=104") in new stack
>  -- Executing NoOp("SIP/104-519e", "TTL:  ARG1: ") in new stack
>  -- Executing GotoIf("SIP/104-519e", "0?continue") in new stack
>  -- Executing Set("SIP/104-519e", "__TTL=64") in new stack
>  -- Executing GotoIf("SIP/104-519e", "1?continue") in new stack
>  -- Goto (macro-user-callerid,s,23)
>  -- Executing NoOp("SIP/104-519e", "Using CallerID "104" <104>") in
> new stack
>  -- Executing Set("SIP/104-519e", "MEETME_ROOMNUM=6000") in new stack
>  -- Executing GotoIf("SIP/104-519e", "0?USER") in new stack
>  -- Executing Answer("SIP/104-519e", "") in new stack
>  -- Executing Wait("SIP/104-519e", "1") in new stack
>  -- Executing Set("SIP/104-519e", "MEETME_OPTS=") in new stack
>  -- Executing Goto("SIP/104-519e", "STARTMEETME|1") in new stack
>  -- Goto (from-internal,STARTMEETME,1)
>  -- Executing MeetMe("SIP/104-519e", "6000||") in new stack
>
>
> ___
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>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>


-- 
-Shane
Blog: http://blind-geek.com/blog/
CoOwner: http://sjtechzone.com
AIM: inhaddict
Skype: chatter8712

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[asterisk-users] chan_mobile and W300i

[Emmanuel Favre-Nicolin]

Hi,

I'm trying to use a mobile phone (ericsson W300i) with asterisk through 
bluetooth. After some sutrggling, I foun chan_mobile.

As some one already used this mobile with what result?

 I'm considering a simple asterisk system (for home use/test purpose) with :
- one SIP service provider (with link to PSTN)
- one SIP phone (a softphone on the asterisk box)
- one mobilephone (ericsson W300i) let's call it A (for Asterisk)

Mobile A is supposed to be connected to asterisk.

I want to interact with this system with another mobile B. The idea would be 
to use free calls between mobile A and B and use the SIP provider
mobile B => . => SIP provider (low cost call) to PSTN...

It is not yet very clear what's working here. I think, I was able to call 
mobile B :
SIP phone => asterisk => mobile A => mobile B
and the comunication is started between :
SIP phone and mobile B

The other way seems more complicated, as I will need to put a password for 
security but when from mobile B (or a regular phone), I call mobile A, I did 
not succeed in dialing the password., it looks like nothing is passing by. 
Maybe I could spy to listen to what's received by asterisk?

I'm using asterisk and chan_mobile from asterisk retrieve by svn checkout 
around 5/1/2008 on a gentoo box. For bluetooth, I have bluez-utils-2.25 
bluez-libs-2.25. Should I really update bluez ?


To try to get some more debug information I tried :
core set debug 4

But I didn't get much?

W300i handfreeprofile seems to be on port 4 and it was not detected it was 
detected (port=0)

Here are some config I use :


more /etc/bluetooth/hcid.conf

options {
autoinit yes;
security auto
pairing multi;
passkey "";
pin_helper /etc/bluetooth/pin-helper;
}
device {
name "Asterisk PBX"
class 0x000100;
iscan enable; pscan enable;
lm accept;
lp rswitch,hold,sniff,park;
}


more mobile.conf

[general]
interval=10   ; Number of seconds between trying to connect to devices.

[adapter]
id=intuix
address=00:11:67:2E:B4:XX
[W300i]
address=00:16:B8:5F:C2:XX ; the address of the phone
port=4 ;handfreeprofile   ; the rfcomm port number (from mobile search)
;port=5 ;headset profile
context=frommobile
adapter=intuix ; adapter to use
dtmfskip=60
group=1   ; this phone is in channel group 1



 more sip.conf

[general]
defaultexpirey=1800
dtmfmode=auto
qualify=yes
context=default
srvlookup=yes

register => _:[EMAIL PROTECTED]
disallow=all
allow=ulaw
allow=alaw ; Allow codecs in order of
allow=ilbc ; preference
allow=gsm
allow=speex

[sip-out]
type=peer
host=freephonie.net
username=XX
fromuser=X
secret=**
nat=yes

[sip-in]
type=peer
context=fromsipproxy
host=freephonie.net

[1000]
type=friend
secret=
qualify=yes; Qualify peer is not more than 2000 mS away
nat=no ; This phone is not natted
host=dynamic   ; This device registers with us
canreinvite=no ; Asterisk by default tries to redirect
context=fromsoftphone
port=5061 ; Uncomment this line if Ekiga and Asterisk are on the same host


more extensions.conf
[globals]
HOMEPHONE=X
MOBILEA=Mobile/W300i;the mobile connected to asterisk
MOBILEB=Mobile/W300i/X ;the "remote" mobile
SIPOUT=SIP/sip-out
SOFTPHONE=SIP/1000
TIMEOUT=2

[general]
autofallthrough=yes

[fromsoftphone]
exten => 0,1,Answer()
exten => 0,n,Authenticate()
exten => 0,n,Dial(${MOBILEB},30)

[frommobile]
exten => s,1,Answer()
exten => s,n,Authenticate()
exten => s,n,Dial(${SOFTPHONE},45)

[fromsipproxy]
exten => s,1,Answer()
exten => s,n,Authenticate()
exten => s,n,Dial(${MOBILEB},45)


Hope someone call help a bit!
I'd be gratefull!

By the way is there a list of mobile phone (even if unofficial) know to behave 
well with chan_mobile/asterisk ? I heard that ericsson mobile doesn't behave 
very well?

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Re: [asterisk-users] FWD and IPCall

[Huw Richards]

If you want to forward your ipkall number directly to your asterisk
server:

1. If your asterisk server is on a private LAN and is connected to the
internet via a router, enable the router to port forward UDP/5060 &
UDP/1-2 to your asterisk server (assuming you have not changed
rtp config parameters in rtp.conf). 

2. Check that the firewall (if any) on your asterisk server allows
connections on UDP/5060 & UDP/1-2

3a. Static public IP address - use the fully qualified domain name
assigned to the IP address (or setup an account on www.no-ip.org with a
name of your choice)

3b. Dynamic public IP address - setup an account on www.no-ip.org with a
name of your choice - install the dynamic ip address update client to
monitor any change of your ip address (downloads & instructions on
no-ip.org website)

4. Goto www.ipkall.com and login to your account. Use your ipkall number
as the SIP Phone Number and then the name you selected in 3a or 3b as
the SIP Proxy.

5. Wait 60 minutes for changes to take affect (!)

6. Edit asterisk sip configuration to allow calls from ipkall:

vi /etc/asterisk/sip.conf and find the section beginning [general]

Add/replace the following:

externhost=the name you setup in 3a. or 3b.
localnet=your private LAN e.g. 192.168.2.0/255.255.255.0

Add a new section at the bottom of the file:

[ipkall.com]
host=voiper.ipkall.com
context=from-ipkall
dtmfmode=rfc2833
insecure=invite
type=friend
canreinvite=no
disallow=all
allow=ulaw ; you can add other codecs if you want once the setup works

Save the file. The section you added tells asterisk to accept calls from
voiper.ipkall.com and to place them in the "from-ipkall" context. This
context can be whatever you want. You may need to change the insecure=
line if you are using asterisk 1.2

7. Edit asterisk dialplan configuration to handle calls from ipkall:

vi /etc/asterisk/extensions.conf and add at the bottom:

[from-ipkall]
exten => ,1,NoOp(from-ipkall)
exten => ,2,NoOp(${CALLERIDNAME}/${CALLERIDNUM})
exten => ,3,Dial(Local/[EMAIL PROTECTED])

Save the file. The section you added tells asterisk what to do with
calls that are received in the "from-ipkall" context. Replace the
 with whatever you entered in the SIP Phone number field
on the ipkall website (I recommended your ipkall number).

In the "from-ipkall" section:
1: display "from-ipkall" on the console
2: display the caller id & name
3. phone the local extension 200 in context "local" - replace this line
with your personal requirements.

Connect to the asterisk console (asterisk -R on my server) and "sip
reload" followed by "dialplan reload" (asterisk 1.4) or "extensions
reload" (asterisk 1.2). "sip reload" will re-read the sip.conf file &
"dialplan reload"/"extensions reload" will re-read the extensions.conf
file.

Phone your ipkall number and see if anything is displayed on the console
and/or your phone rings. 

If nothing on the console when you phone, try "sip set debug peer
ipkall.com" (asterisk 1.4 - not sure of the command for asterisk 1.2)
and phone again.

Post back your results.



 







-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Shane D
Sent: Monday, January 07, 2008 17:32
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] FWD and IPCall

Okay... That was kind of confusing. Would you contact me off-list to
help me specifically?

I've double-checked everything for the IAX, and it's a no-go. Maybe
I'll try this SIP thing. But then again, if I can just hook IPKall to
the server directly, I don't need FWD...

On 1/7/08, Huw Richards <[EMAIL PROTECTED]> wrote:
> My config is as follows
>
> Excerpt of sip.conf:
>
> [general]
> externhost=fully.qualified.domain.name
> localnet=192.168.2.0/255.255.255.0
> srvlookup=no
> defaultexpiry=3600
> dtmfmode=rfc2833
>
> register => :@fwd.pulver.com/
>
> [sipfwd]
> type=peer
> secret=
> username=
> fromdomain=fwd.pulver.com
> host=fwd.pulver.com
> disallow=all
> allow=ulaw
> canreinvite=yes
> insecure=invite
> qualify=yes
> context=from-fwd
>
> Excerpt of extensions.conf:
>
> [from-fwd]
> exten => ,1,NoOp(from-fwd)
> exten => ,n,Dial(whatever)
>
> I have a dynamic public IP address, so I use http://www.no-ip.org to
map
> my IP address to name. My router port forwards UDP/5060 &
> UDP/1-2 to the internal asterisk server.
>
> However, I do not have ipkall forwarding to my fwd account. I have it
> forwarding directly to my asterisk server using the no-ip.org address
I
> registered.
>
> e.g. forward to sip:[EMAIL PROTECTED] on ipkall website
> and then in sip.conf:
>
> [ipkall.com]
> host=voiper.ipkall.com
> context=from-ipkall
> dtmfmode=rfc2833
> insecure=invite
> type=friend
> canreinvite=no
> disallow=all
> allow=ulaw
>
> And in extensions.conf:
>
> [from-ipkall]
> exten => xxx,1,NoOp(from-ipkall)
> exten => xxx,n,Dial(whatever)
>
>
>
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of S

Re: [asterisk-users] Background Noise Elimination

[Matt Riddell]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Norman Franke wrote:
> Greetings!
> 
> We have a somewhat noisy background in our call center, and I'd like to
> reduce this. Obviously, we could plaster the walls with sound absorbing
> material, but is there anything we can do in software either using any
> algorithms for our open source-based SIP library or inside Asterisk
> itself? Related to this, anyone have a good source for good panels?
> 
> We are using Plantronics noise canceling headsets, which don't really
> seem to work all that well. Our ancient system handled noise much
> better, but I suspect that was partly due to the Dialogic ADPCM
> algorithm used that just reduced the intelligibility of lower volume
> noises in general. We are using PCMU direct from the agent's mic to
> through Asterisk to PRIs, so we don't "suffer" from compression
> artifacts. The down side, is that you can make out even very quiet
> conversations in the background (like 3 agents to one side.)
> 
> How have people handled this? I'm experimenting with a noise gate that
> will lower the volume when the agent isn't talking, but that won't help
> when the agent is talking.

Nah, there's nothing really.

The noise gate is your best bet.  I would assume that while an agent is
talking the customer will be listening to the agent, so the background
noise will hardly be noticeable.

The issue is, while two people are talking its pretty hard to remove
just one of them from a wave file.

Try the noise gate and see how you go.

Oh, you might want to try a downwards expander instead (a noise gate but
with ratio as well as threshold).

- --
Kind Regards,

Matt Riddell
Director
___

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[asterisk-users] :POSSIBLE SPAM: conferencing help

[Nhadie]

Hi All,

kind of need help on the conference module, i'm using freepbx and
enabled conferencing, i created a conference number, 6000. when i dial
to it, my phone says it is connected but i'm hearing nothing, maybe logs
below can help you.

also, when i hang up the phone, the conference did not disconnect me.
how can i end a conference? thank you

 -- Executing Macro("SIP/104-519e", "user-callerid|") in new stack
 -- Executing NoOp("SIP/104-519e", "user-callerid: device 104") in
new stack
 -- Executing Set("SIP/104-519e", "AMPUSER=104") in new stack
 -- Executing GotoIf("SIP/104-519e", "0?report") in new stack
 -- Executing GotoIf("SIP/104-519e", "0?start") in new stack
 -- Executing Set("SIP/104-519e", "REALCALLERIDNUM=104") in new stack
 -- Executing NoOp("SIP/104-519e", "REALCALLERIDNUM is 104") in new
stack
 -- Executing Set("SIP/104-519e", "AMPUSER=104") in new stack
 -- Executing Set("SIP/104-519e", "AMPUSERCIDNAME=104") in new stack
 -- Executing GotoIf("SIP/104-519e", "0?report") in new stack
 -- Executing Set("SIP/104-519e", "AMPUSERCID=104") in new stack
 -- Executing Set("SIP/104-519e", "CALLERID(all)="104" <104>") in
new stack
 -- Executing Set("SIP/104-519e", "REALCALLERIDNUM=104") in new stack
 -- Executing NoOp("SIP/104-519e", "TTL:  ARG1: ") in new stack
 -- Executing GotoIf("SIP/104-519e", "0?continue") in new stack
 -- Executing Set("SIP/104-519e", "__TTL=64") in new stack
 -- Executing GotoIf("SIP/104-519e", "1?continue") in new stack
 -- Goto (macro-user-callerid,s,23)
 -- Executing NoOp("SIP/104-519e", "Using CallerID "104" <104>") in
new stack
 -- Executing Set("SIP/104-519e", "MEETME_ROOMNUM=6000") in new stack
 -- Executing GotoIf("SIP/104-519e", "0?USER") in new stack
 -- Executing Answer("SIP/104-519e", "") in new stack
 -- Executing Wait("SIP/104-519e", "1") in new stack
 -- Executing Set("SIP/104-519e", "MEETME_OPTS=") in new stack
 -- Executing Goto("SIP/104-519e", "STARTMEETME|1") in new stack
 -- Goto (from-internal,STARTMEETME,1)
 -- Executing MeetMe("SIP/104-519e", "6000||") in new stack


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Re: [asterisk-users] Media gateways and video

[Paul Hales]

My question would actually be - is there any support for h234 over ISDN?

PaulH


On Mon, 2008-01-07 at 19:59 +0100, Olivier wrote:
> Hi,
> 
> Asterisk now supports h234m.
> Does anyone know a Media gateway such as those of Mediatrix, Patton,
> Audiocodes, Cisco that also supports h324m flows ?
> 
> Prospective setup would be:
> ISDN  Media gateway -- -- Asterisk --- ---
> Videosoftphone 
> 
> Cheers
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[asterisk-users] Background Noise Elimination

[Norman Franke]

Greetings!

We have a somewhat noisy background in our call center, and I'd like  
to reduce this. Obviously, we could plaster the walls with sound  
absorbing material, but is there anything we can do in software  
either using any algorithms for our open source-based SIP library or  
inside Asterisk itself? Related to this, anyone have a good source  
for good panels?


We are using Plantronics noise canceling headsets, which don't really  
seem to work all that well. Our ancient system handled noise much  
better, but I suspect that was partly due to the Dialogic ADPCM  
algorithm used that just reduced the intelligibility of lower volume  
noises in general. We are using PCMU direct from the agent's mic to  
through Asterisk to PRIs, so we don't "suffer" from compression  
artifacts. The down side, is that you can make out even very quiet  
conversations in the background (like 3 agents to one side.)


How have people handled this? I'm experimenting with a noise gate  
that will lower the volume when the agent isn't talking, but that  
won't help when the agent is talking.


Norman Franke
ASD, Inc.

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Re: [asterisk-users] FWD and IPCall

[Shane D]

Still rejected.

On 1/7/08, www.IPKall.com <[EMAIL PROTECTED]> wrote:
> Try using the IP address, and not the dynamic URL, does anything change?
>
> IPKall
> IPKall Forum
> http://voxilla.com/PNphpBB2-viewforum-f-38.html
>
>
>
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Huw Richards
> Sent: Monday, January 07, 2008 1:55 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] FWD and IPCall
>
> My config is as follows
>
> Excerpt of sip.conf:
>
> [general]
> externhost=fully.qualified.domain.name
> localnet=192.168.2.0/255.255.255.0
> srvlookup=no
> defaultexpiry=3600
> dtmfmode=rfc2833
>
> register => :@fwd.pulver.com/
>
> [sipfwd]
> type=peer
> secret=
> username=
> fromdomain=fwd.pulver.com
> host=fwd.pulver.com
> disallow=all
> allow=ulaw
> canreinvite=yes
> insecure=invite
> qualify=yes
> context=from-fwd
>
> Excerpt of extensions.conf:
>
> [from-fwd]
> exten => ,1,NoOp(from-fwd)
> exten => ,n,Dial(whatever)
>
> I have a dynamic public IP address, so I use http://www.no-ip.org to map
> my IP address to name. My router port forwards UDP/5060 &
> UDP/1-2 to the internal asterisk server.
>
> However, I do not have ipkall forwarding to my fwd account. I have it
> forwarding directly to my asterisk server using the no-ip.org address I
> registered.
>
> e.g. forward to sip:[EMAIL PROTECTED] on ipkall website
> and then in sip.conf:
>
> [ipkall.com]
> host=voiper.ipkall.com
> context=from-ipkall
> dtmfmode=rfc2833
> insecure=invite
> type=friend
> canreinvite=no
> disallow=all
> allow=ulaw
>
> And in extensions.conf:
>
> [from-ipkall]
> exten => xxx,1,NoOp(from-ipkall)
> exten => xxx,n,Dial(whatever)
>
>
>
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Shane D
> Sent: Monday, January 07, 2008 12:09
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] FWD and IPCall
>
> It's Iax2. Is there a way of using amore reliable sip
> connectoin/something slightly different?
>
> If so, how would I go about that.
>
> On 1/7/08, Huw Richards <[EMAIL PROTECTED]> wrote:
> > You haven't said if your connection to fwd is SIP or IAX2 but I have
> > found IAX2 connections to fwd to be unreliable. Other people may have
> > different results.
> >
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of Shane D
> > Sent: Monday, January 07, 2008 10:17
> > To: asterisk-users@lists.digium.com
> > Subject: [asterisk-users] FWD and IPCall
> >
> > Hello All,
> >
> > I have a problem. I have tried everything that is in the book "The
> > Future of Telephony" as well as on the FWD (freeworlddialup) website,
> > and there is still a problem. My asterisk box is not able to associate
> > with the FWD server. I get:
> > Registration Rejected by [insert IP], and I can't use my IPCall number
> > to reach my Asterisk box.
> > Any suggestions?
> > --
> > -Shane
> > Blog: http://blind-geek.com/blog/
> > CoOwner: http://sjtechzone.com
> > AIM: inhaddict
> > Skype: chatter8712
> >
> > ___
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> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
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> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
>
> --
> -Shane
> Blog: http://blind-geek.com/blog/
> CoOwner: http://sjtechzone.com
> AIM: inhaddict
> Skype: chatter8712
>
> ___
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>
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> To UNSUBSCRIBE or update options visit:
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>
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>
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>


-- 
-Shane
Blog: http://blind-geek.com/blog/
CoOwner: http://sjtechzone.com
AIM: inhaddict
Skype: chatter8712

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Re: [asterisk-users] asterisk CLI and no such command "stop"

[Vieri]

--- MatsK <[EMAIL PROTECTED]> wrote:

> Vieri wrote:
> > Hi,
> > 
> > I'm probably missing something trivial but I don't
> > understand what. 
> > 
> > Asterisk is loading fine but when I connect to the
> > console (asterisk -vr) and type "stop" I get a no
> such
> > command reply:
> > 
> > *CLI> help
> > (...)
> > skinny show lines  Show defined Skinny
> lines per device
> >   soft hangup  Request a hangup on a
> given channel
> >unload  Unload a dynamic module
> by name
> > *CLI> stop
> > No such command 'stop' (type 'help' for help)
> > 
> > # tail -n 1000 /var/log/messages | grep -i error
> > 
> > Does anyone know why the "stop" command doesn't
> appear
> > on the help list and is unavailable?
> 
> Try
> *CLI> help stop
> and that will show you the syntax for the commands
> that fails!
> 
> I can agree that "No such command 'stop' (type
> 'help' for help)" is a 
> bit misleading and should have been "Wrong syntax"
> or "Incomplete 
> command" or something like that
> 
> And always state witch * version you use!

inf-voip1 ~ # asterisk -vr
Asterisk 1.2.21.1, Copyright (C) 1999 - 2007 Digium,
Inc. and others.
Created by Mark Spencer <[EMAIL PROTECTED]>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show
warranty' for details.
This is free software, with components licensed under
the GNU General Public
License version 2 and other licenses; you are welcome
to redistribute it under
certain conditions. Type 'show license' for details.
=
Connected to Asterisk 1.2.21.1 currently running on
inf-voip1 (pid = 13423)
Verbosity is at least 1
inf-voip1*CLI> stop now
No such command 'stop' (type 'help' for help)
inf-voip1*CLI> stop gracefully
No such command 'stop' (type 'help' for help)
inf-voip1*CLI>



  

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Re: [asterisk-users] FWD and IPCall

[Shane D]

Okay... That was kind of confusing. Would you contact me off-list to
help me specifically?

I've double-checked everything for the IAX, and it's a no-go. Maybe
I'll try this SIP thing. But then again, if I can just hook IPKall to
the server directly, I don't need FWD...

On 1/7/08, Huw Richards <[EMAIL PROTECTED]> wrote:
> My config is as follows
>
> Excerpt of sip.conf:
>
> [general]
> externhost=fully.qualified.domain.name
> localnet=192.168.2.0/255.255.255.0
> srvlookup=no
> defaultexpiry=3600
> dtmfmode=rfc2833
>
> register => :@fwd.pulver.com/
>
> [sipfwd]
> type=peer
> secret=
> username=
> fromdomain=fwd.pulver.com
> host=fwd.pulver.com
> disallow=all
> allow=ulaw
> canreinvite=yes
> insecure=invite
> qualify=yes
> context=from-fwd
>
> Excerpt of extensions.conf:
>
> [from-fwd]
> exten => ,1,NoOp(from-fwd)
> exten => ,n,Dial(whatever)
>
> I have a dynamic public IP address, so I use http://www.no-ip.org to map
> my IP address to name. My router port forwards UDP/5060 &
> UDP/1-2 to the internal asterisk server.
>
> However, I do not have ipkall forwarding to my fwd account. I have it
> forwarding directly to my asterisk server using the no-ip.org address I
> registered.
>
> e.g. forward to sip:[EMAIL PROTECTED] on ipkall website
> and then in sip.conf:
>
> [ipkall.com]
> host=voiper.ipkall.com
> context=from-ipkall
> dtmfmode=rfc2833
> insecure=invite
> type=friend
> canreinvite=no
> disallow=all
> allow=ulaw
>
> And in extensions.conf:
>
> [from-ipkall]
> exten => xxx,1,NoOp(from-ipkall)
> exten => xxx,n,Dial(whatever)
>
>
>
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Shane D
> Sent: Monday, January 07, 2008 12:09
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] FWD and IPCall
>
> It's Iax2. Is there a way of using amore reliable sip
> connectoin/something slightly different?
>
> If so, how would I go about that.
>
> On 1/7/08, Huw Richards <[EMAIL PROTECTED]> wrote:
> > You haven't said if your connection to fwd is SIP or IAX2 but I have
> > found IAX2 connections to fwd to be unreliable. Other people may have
> > different results.
> >
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of Shane D
> > Sent: Monday, January 07, 2008 10:17
> > To: asterisk-users@lists.digium.com
> > Subject: [asterisk-users] FWD and IPCall
> >
> > Hello All,
> >
> > I have a problem. I have tried everything that is in the book "The
> > Future of Telephony" as well as on the FWD (freeworlddialup) website,
> > and there is still a problem. My asterisk box is not able to associate
> > with the FWD server. I get:
> > Registration Rejected by [insert IP], and I can't use my IPCall number
> > to reach my Asterisk box.
> > Any suggestions?
> > --
> > -Shane
> > Blog: http://blind-geek.com/blog/
> > CoOwner: http://sjtechzone.com
> > AIM: inhaddict
> > Skype: chatter8712
> >
> > ___
> > --Bandwidth and Colocation Provided by http://www.api-digital.com--
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> > ___
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> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
>
> --
> -Shane
> Blog: http://blind-geek.com/blog/
> CoOwner: http://sjtechzone.com
> AIM: inhaddict
> Skype: chatter8712
>
> ___
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> asterisk-users mailing list
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>
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> To UNSUBSCRIBE or update options visit:
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>


-- 
-Shane
Blog: http://blind-geek.com/blog/
CoOwner: http://sjtechzone.com
AIM: inhaddict
Skype: chatter8712

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Re: [asterisk-users] Increase Volume - SIP

[bilal ghayyad]

Hi Marc;

Are you using VPN between the two sites? If that is
the case, then you wil have a low volume and I faced
this problem. Just try without VPN.

Another issue: the call is originated from the Mobile
(or PSTN) and you call to Zaptel and then do the call
via SIP (or IAX) Trunk? Or you are facing a low volume
when you call from FXS (or from IP Phone)?

There is a way to increase the volume on the fxs or
fxo ports by using the following:

modprobe zaptel
modprobe wctdm fxorxgain=12 (or more)

But asterisk should not be running, and you should
unload zaptel and then reload it (as it takes this
argument only when loading), and you need to do it
each time you restart the machine (or it can be
hardcoded if you found a good result).

To unload zaptel, use modprobe - r zaptel and modprobe
-r wctdm (But remember to let asterisk off, not
running).

You can also bypass fxotxgain, also you can use
fxstxgain and fxsrxgain for fxs ports, remember that
here we use fxs for fxs and fxo for fxo where the case
differs in configuring that in zapata.conf

Hope that help and please let me know what happened
with you.

Regards
Bilal



[EMAIL PROTECTED] wrote:

> Can someone tell me if there is a way to increase
the volume of a
 conversation that occurs between two SIP channels or
between a SIP and an
 IAX channel ?
> 
> My headsets are set to the maximum volume but the
voice is still low
 ... I know there is a configuration in zapata.conf
for the digium
 cards, but is there a place I can set this up for RTP
conversations ?

No. Try to adjust the mic volume if you can.

(Alternative solution: Shorter ethernet cables make
the audio
a bit louder. ;-)


Regards,
  Philipp Kempgen


  

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Re: [asterisk-users] asterisk-users] Increase Volume - SIP

[bilal ghayyad]

Hi Marc;

Are you using VPN between the two sites? If that is
the case, then you wil have a low volume and I faced
this problem. Just try without VPN.

Another issue: the call is originated from the Mobile
(or PSTN) and you call to Zaptel and then do the call
via SIP (or IAX) Trunk? Or you are facing a low volume
when you call from FXS (or from IP Phone)?

There is a way to increase the volume on the fxs or
fxo ports by using the following:

modprobe zaptel
modprobe wctdm fxorxgain=12 (or more)

But asterisk should not be running, and you should
unload zaptel and then reload it (as it takes this
argument only when loading), and you need to do it
each time you restart the machine (or it can be
hardcoded if you found a good result).

To unload zaptel, use modprobe - r zaptel and modprobe
-r wctdm (But remember to let asterisk off, not
running).

You can also bypass fxotxgain, also you can use
fxstxgain and fxsrxgain for fxs ports, remember that
here we use fxs for fxs and fxo for fxo where the case
differs in configuring that in zapata.conf

Hope that help and please let me know what happened
with you.

Regards
Bilal



[EMAIL PROTECTED] wrote:

> Can someone tell me if there is a way to increase
the volume of a
 conversation that occurs between two SIP channels or
between a SIP and an
 IAX channel ?
> 
> My headsets are set to the maximum volume but the
voice is still low
 ... I know there is a configuration in zapata.conf
for the digium
 cards, but is there a place I can set this up for RTP
conversations ?

No. Try to adjust the mic volume if you can.

(Alternative solution: Shorter ethernet cables make
the audio
a bit louder. ;-)


Regards,
  Philipp Kempgen




  

Never miss a thing.  Make Yahoo your home page. 
http://www.yahoo.com/r/hs

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Re: [asterisk-users] Change Default Voicemail Message

[Daniel Cole]

Thank you for your reply Trevor.

Is there an easy way to achieve this with a computer generated voice? We do not 
wish to manually record the messages if possible, in the interests of a 
consistent message across all voicemail boxes. What would be the easiest way to 
do this?

Also, can you please give me some pointers on how to get the voicemail to play 
the separate message before the normal voicemail message? I'm guessing it would 
be done with a custom voicemail content, but im not sure how to write it 
correctly.

Many Thanks,

Daniel Cole


From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Trevor G. 
Hammonds
Sent: Monday, 7 January 2008 3:52 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Change Default Voicemail Message

Daniel,
You could have Alison record a prompt "Welcome to (nursing home)" and re-record 
the prompt "The person at extension..." to be "The person in room...".  Then 
have your dialplan play the "Welcome To..." message before sending the call to 
voice mail.  Then callers will hear "Welcome to (Nursing Home).  The person in 
room 5 is unavailable.  Please leave your message..." and if the resident has a 
recorded personal greeting or name, it would replace the "The person..." 
portion with either the resident's recorded name or greeting.

Sincerely,
Trevor Hammonds


From: Daniel Cole
Sent: Sunday, January 06, 2008 6:15 PM


Hello List,

I have a client (a nursing home)  that we are looking at installing a trixbox 
for. One of the features that they would really like is a customized, standard 
voicemail recording for each of the residents rooms.

We are looking for something along the lines of a voicemail recording like 
this:  "Welcome to (nursing home). You have reached room 5. Please leave a 
message after the tone".

What would be the easiest way to get this to work. I have had a look at a few 
options, but I cant seem to find what I am after.

Any help would be much appreciated.


Thank You,

Daniel Cole
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Re: [asterisk-users] FWD and IPCall

[www.IPKall.com]

Try using the IP address, and not the dynamic URL, does anything change?

IPKall
IPKall Forum
http://voxilla.com/PNphpBB2-viewforum-f-38.html




-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Huw Richards
Sent: Monday, January 07, 2008 1:55 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] FWD and IPCall

My config is as follows

Excerpt of sip.conf:

[general]
externhost=fully.qualified.domain.name
localnet=192.168.2.0/255.255.255.0
srvlookup=no
defaultexpiry=3600
dtmfmode=rfc2833

register => :@fwd.pulver.com/

[sipfwd]
type=peer
secret=
username=
fromdomain=fwd.pulver.com
host=fwd.pulver.com
disallow=all
allow=ulaw
canreinvite=yes
insecure=invite
qualify=yes
context=from-fwd

Excerpt of extensions.conf:

[from-fwd]
exten => ,1,NoOp(from-fwd)
exten => ,n,Dial(whatever)

I have a dynamic public IP address, so I use http://www.no-ip.org to map
my IP address to name. My router port forwards UDP/5060 &
UDP/1-2 to the internal asterisk server.

However, I do not have ipkall forwarding to my fwd account. I have it
forwarding directly to my asterisk server using the no-ip.org address I
registered.

e.g. forward to sip:[EMAIL PROTECTED] on ipkall website
and then in sip.conf:

[ipkall.com]
host=voiper.ipkall.com
context=from-ipkall
dtmfmode=rfc2833
insecure=invite
type=friend
canreinvite=no
disallow=all
allow=ulaw

And in extensions.conf:

[from-ipkall]
exten => xxx,1,NoOp(from-ipkall)
exten => xxx,n,Dial(whatever)




-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Shane D
Sent: Monday, January 07, 2008 12:09
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] FWD and IPCall

It's Iax2. Is there a way of using amore reliable sip
connectoin/something slightly different?

If so, how would I go about that.

On 1/7/08, Huw Richards <[EMAIL PROTECTED]> wrote:
> You haven't said if your connection to fwd is SIP or IAX2 but I have
> found IAX2 connections to fwd to be unreliable. Other people may have
> different results.
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Shane D
> Sent: Monday, January 07, 2008 10:17
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] FWD and IPCall
>
> Hello All,
>
> I have a problem. I have tried everything that is in the book "The
> Future of Telephony" as well as on the FWD (freeworlddialup) website,
> and there is still a problem. My asterisk box is not able to associate
> with the FWD server. I get:
> Registration Rejected by [insert IP], and I can't use my IPCall number
> to reach my Asterisk box.
> Any suggestions?
> --
> -Shane
> Blog: http://blind-geek.com/blog/
> CoOwner: http://sjtechzone.com
> AIM: inhaddict
> Skype: chatter8712
>
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-- 
-Shane
Blog: http://blind-geek.com/blog/
CoOwner: http://sjtechzone.com
AIM: inhaddict
Skype: chatter8712

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Re: [asterisk-users] FWD and IPCall

[Huw Richards]

My config is as follows

Excerpt of sip.conf:

[general]
externhost=fully.qualified.domain.name
localnet=192.168.2.0/255.255.255.0
srvlookup=no
defaultexpiry=3600
dtmfmode=rfc2833

register => :@fwd.pulver.com/

[sipfwd]
type=peer
secret=
username=
fromdomain=fwd.pulver.com
host=fwd.pulver.com
disallow=all
allow=ulaw
canreinvite=yes
insecure=invite
qualify=yes
context=from-fwd

Excerpt of extensions.conf:

[from-fwd]
exten => ,1,NoOp(from-fwd)
exten => ,n,Dial(whatever)

I have a dynamic public IP address, so I use http://www.no-ip.org to map
my IP address to name. My router port forwards UDP/5060 &
UDP/1-2 to the internal asterisk server.

However, I do not have ipkall forwarding to my fwd account. I have it
forwarding directly to my asterisk server using the no-ip.org address I
registered.

e.g. forward to sip:[EMAIL PROTECTED] on ipkall website
and then in sip.conf:

[ipkall.com]
host=voiper.ipkall.com
context=from-ipkall
dtmfmode=rfc2833
insecure=invite
type=friend
canreinvite=no
disallow=all
allow=ulaw

And in extensions.conf:

[from-ipkall]
exten => xxx,1,NoOp(from-ipkall)
exten => xxx,n,Dial(whatever)




-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Shane D
Sent: Monday, January 07, 2008 12:09
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] FWD and IPCall

It's Iax2. Is there a way of using amore reliable sip
connectoin/something slightly different?

If so, how would I go about that.

On 1/7/08, Huw Richards <[EMAIL PROTECTED]> wrote:
> You haven't said if your connection to fwd is SIP or IAX2 but I have
> found IAX2 connections to fwd to be unreliable. Other people may have
> different results.
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Shane D
> Sent: Monday, January 07, 2008 10:17
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] FWD and IPCall
>
> Hello All,
>
> I have a problem. I have tried everything that is in the book "The
> Future of Telephony" as well as on the FWD (freeworlddialup) website,
> and there is still a problem. My asterisk box is not able to associate
> with the FWD server. I get:
> Registration Rejected by [insert IP], and I can't use my IPCall number
> to reach my Asterisk box.
> Any suggestions?
> --
> -Shane
> Blog: http://blind-geek.com/blog/
> CoOwner: http://sjtechzone.com
> AIM: inhaddict
> Skype: chatter8712
>
> ___
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>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
> ___
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>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>


-- 
-Shane
Blog: http://blind-geek.com/blog/
CoOwner: http://sjtechzone.com
AIM: inhaddict
Skype: chatter8712

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Re: [asterisk-users] asterisk CLI and no such command "stop"

[MatsK]

Vieri wrote:
> Hi,
> 
> I'm probably missing something trivial but I don't
> understand what. 
> 
> Asterisk is loading fine but when I connect to the
> console (asterisk -vr) and type "stop" I get a no such
> command reply:
> 
> *CLI> help
> (...)
> skinny show lines  Show defined Skinny lines per device
>   soft hangup  Request a hangup on a given channel
>unload  Unload a dynamic module by name
> *CLI> stop
> No such command 'stop' (type 'help' for help)
> 
> # tail -n 1000 /var/log/messages | grep -i error
> 
> Does anyone know why the "stop" command doesn't appear
> on the help list and is unavailable?

Try
*CLI> help stop
and that will show you the syntax for the commands that fails!

I can agree that "No such command 'stop' (type 'help' for help)" is a 
bit misleading and should have been "Wrong syntax" or "Incomplete 
command" or something like that

And always state witch * version you use!



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Re: [asterisk-users] FWD and IPCall

[Troy Ayers]

192.246.x.x =! 192.168.x.x if that is what you're thinking.

Anyway, "registration Refused" sounds like you're getting up to FWD and 
attempting to authenticate, but failing at that point.

double-check your iax.conf settings against the FWD "Extra Features" 
settings... of course make sure IAX is selected too with FWD.

-Troy


Shane D wrote:
> I get the following output:
> 
> Jan  7 10:56:22 NOTICE[4666]: chan_iax2.c:7536 socket_read:
> Registration of '886036' rejected: 'Registration Refused' from:
> '192.246.69.186'
> 
> It shouldn't be trying to do something on my network (192) should it?
> 
> On 1/7/08, Shane D <[EMAIL PROTECTED]> wrote:
>> I will try that when I return home in about two hours. I'll let you know.
>>
>> On 1/7/08, Benchev <[EMAIL PROTECTED]> wrote:
> I have a problem. I have tried everything that is in the book "The
> Future of Telephony" as well as on the FWD (freeworlddialup) website,
> and there is still a problem. My asterisk box is not able to associate
> with the FWD server. I get:
> Registration Rejected by [insert IP], and I can't use my IPCall number
> to reach my Asterisk box.
> Any suggestions?
>>> If you try dnsmgr.conf
>>> enable=yes; what happens?
>>>
>>> Boyko
>>>
>>>
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>>> To UNSUBSCRIBE or update options visit:
>>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>
>> --
>> -Shane
>> Blog: http://blind-geek.com/blog/
>> CoOwner: http://sjtechzone.com
>> AIM: inhaddict
>> Skype: chatter8712
>>
> 
> 


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Re: [asterisk-users] Cisco 79xx XML services

[Anciso, Roy]

Although it's not LDAP I used a script that I found on the voip wiki and
changed it so it looked at only sip configuration files. It also
alphabetizes the output so it can be displayed that way on the phone.
Below are my notes on the subject.  If someone is willing to post this
to the wiki and send me a link, that would be awesome.

 

Cisco Phone Extension Directory Using Services Button
I used a PHP script that I found on the Internet and rewrote it to fit
our needs.  Original code is found at:
http://users.marshall.edu/~twohig5/directory.php.txt
The new code only looks at the sip.conf file since we are only using sip
phones. My version also alphabetizes the directory.  Below is the source
code:



Directory.php.txt
". $v['name']."\n".
"".key($sip_array)."\n";
}
next($sip_array);
}
 
sort ($directory);
 
echo "\n";
echo "".$dirname."\n";
foreach ($directory as $v) {
  echo "\n\n";
  echo $v;
  echo "\n";
}
echo "\nChoose Name and Press Dial\n";
echo "\n";
?>

 
>From here you can schedule this to run every so often.  Once the file is
created you must place it in your web directory on the server.
 
I chained the command and also wrote the output to an xml file in the
web directory.  The command looks like this:
 
'php /etc/asterisk/directory.php.txt > /var/www/html/directory.xml'
 
System Speeddials using Services Button
 
For speed dials I modified the php code to look to a specific file in
the asterisk directory called speeddials.conf.  This file only contains
attributes that the php script will look for.  This is great because you
only need to specify the number and name fields.  Below is my example of
speeddial.php.txt (php code) and speeddials.conf (speed dials):
 
Speeddial.php.txt
". $v['name']."\n".
"".key($ssd_array)."\n";
}
next($ssd_array);
}
 
sort ($directory);
 
echo "\n";
echo "".$dirname."\n";
foreach ($directory as $v) {
  echo "\n\n";
  echo $v;
  echo "\n";
}
echo "\nChoose Name and Press Dial\n";
echo "\n";
?>

 
Speeddials.conf
;System Speed Dial File
;This is used in conjuction with speeddial.php.txt
;
[9,7234264]
name=MISD Admin Office
[9,7236205]
name=MISD Special Ed Office
[9,7233521]
name=MAPS Supintendent Office
[9,7232547]
name=MAPS MHS
 
Once these files are create just run the php command:
'php speeddials.php.txt > /var/www/html/speeddial.xml'

This will generate the speed dial file and place it in your web
directory.  
 
You can also schedule this to run just like the extension directory
script.   
 
Creating the Main Services Menu
To display these two items when the user presses the Services button you
first we need to create a file that contains the menus.  I created a
file called services.xml and placed in the web directory /var/www/html/.
 
Then I wrote the menu structure using XML. I used the information found
on the Cisco website as guide to do this. Below is my services.xml file:
 

Information Services
Press to Enter

Extension Directory
http://192.168.1.94/directory.xml


Dir



 
Press to Enter

System Speed Dial
http://192.168.1.94/speeddial.xml


Dir



 

 
As you can see the example above uses the  tag.  I
created a couple of menu items called "Extension Directory" & "System
Speed Dial" which points to the directory.xml and speeddial.xml files we
created earlier.  

 

For photos of how this looks on the phone visit:
http://picasaweb.google.com/ranciso/AsteriskImagesCiscoPhones/photo#5135
353621777248450
 


One major caveat is for some reason Cisco has a limit on how many
numbers you can display using the  directive.  I
believe it is 32. So to keep things sane I created a directory
"/etc/asterisk/sip" and departmentalized my sip registrations there.
You can tell asterisk to load the sip config files by inserting the
following in your main /etc/asterisk/sip.conf file under the general
settings:

#include "/etc/asterisk/sip/*.conf

 

After creating separate sip files I also duplicated the php script to
point to each sip file and adjusted the Cisco XML files accordingly.  

 


Cisco Resources & Links for SIP Configuration (You need a Cisco Login to
Access)



Description: This link is to the documentation for all the Cisco phones:
http://cisco.com/en/US/customer/products/hw/phones/ps379/prod_maintenanc
e_guides_list.html
 
Description: This link is for setting up the 7906 & 7911 phones with SIP

http://cisco.com/en/US/customer/products/hw/phones/ps379/products_admini
stration_guide_book09186a00807307c8.html

 

 

 

 

 

 

 

 

 

 



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of satish
patel
Sent: Sunday, January 06, 2008 9:58 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [asterisk-users] Cisco 79xx XML services

 

I am useing Cisco 7975 with Asterisk on SIP protocol and its working gr8


 

I have also implemeted SCCP but i got problem of Hangup and my asterisk
got hang i dont 

Re: [asterisk-users] FWD and IPCall

[Shane D]

Ignore the part about 192... That happens to be the IP of iax2.fwdnet.net.

If anyone has set up FWD successfully for incoming only, let me know.

On 1/7/08, Shane D <[EMAIL PROTECTED]> wrote:
> I get the following output:
>
> Jan  7 10:56:22 NOTICE[4666]: chan_iax2.c:7536 socket_read:
> Registration of '886036' rejected: 'Registration Refused' from:
> '192.246.69.186'
>
> It shouldn't be trying to do something on my network (192) should it?
>
> On 1/7/08, Shane D <[EMAIL PROTECTED]> wrote:
> > I will try that when I return home in about two hours. I'll let you know.
> >
> > On 1/7/08, Benchev <[EMAIL PROTECTED]> wrote:
> > > > > I have a problem. I have tried everything that is in the book "The
> > > > > Future of Telephony" as well as on the FWD (freeworlddialup)
> website,
> > > > > and there is still a problem. My asterisk box is not able to
> associate
> > > > > with the FWD server. I get:
> > > > > Registration Rejected by [insert IP], and I can't use my IPCall
> number
> > > > > to reach my Asterisk box.
> > > > > Any suggestions?
> > > If you try dnsmgr.conf
> > > enable=yes; what happens?
> > >
> > > Boyko
> > >
> > >
> > > ___
> > > --Bandwidth and Colocation Provided by http://www.api-digital.com--
> > >
> > > asterisk-users mailing list
> > > To UNSUBSCRIBE or update options visit:
> > >http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> >
> >
> > --
> > -Shane
> > Blog: http://blind-geek.com/blog/
> > CoOwner: http://sjtechzone.com
> > AIM: inhaddict
> > Skype: chatter8712
> >
>
>
> --
> -Shane
> Blog: http://blind-geek.com/blog/
> CoOwner: http://sjtechzone.com
> AIM: inhaddict
> Skype: chatter8712
>


-- 
-Shane
Blog: http://blind-geek.com/blog/
CoOwner: http://sjtechzone.com
AIM: inhaddict
Skype: chatter8712

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Re: [asterisk-users] FWD and IPCall

[Shane D]

I get the following output:

Jan  7 10:56:22 NOTICE[4666]: chan_iax2.c:7536 socket_read:
Registration of '886036' rejected: 'Registration Refused' from:
'192.246.69.186'

It shouldn't be trying to do something on my network (192) should it?

On 1/7/08, Shane D <[EMAIL PROTECTED]> wrote:
> I will try that when I return home in about two hours. I'll let you know.
>
> On 1/7/08, Benchev <[EMAIL PROTECTED]> wrote:
> > > > I have a problem. I have tried everything that is in the book "The
> > > > Future of Telephony" as well as on the FWD (freeworlddialup) website,
> > > > and there is still a problem. My asterisk box is not able to associate
> > > > with the FWD server. I get:
> > > > Registration Rejected by [insert IP], and I can't use my IPCall number
> > > > to reach my Asterisk box.
> > > > Any suggestions?
> > If you try dnsmgr.conf
> > enable=yes; what happens?
> >
> > Boyko
> >
> >
> > ___
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> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
>
> --
> -Shane
> Blog: http://blind-geek.com/blog/
> CoOwner: http://sjtechzone.com
> AIM: inhaddict
> Skype: chatter8712
>


-- 
-Shane
Blog: http://blind-geek.com/blog/
CoOwner: http://sjtechzone.com
AIM: inhaddict
Skype: chatter8712

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Re: [asterisk-users] Cisco 7941G-GE with Asterisk and CTPSEP odyssee

[John Novack]

Christophorus Laube wrote:
> I do have to answer to your suggestion of renaming the CTLSEP.tlv
> to SEP. The phone is still requesting CTLSEP.tlv and as it
> cannot find that it goes into a loop. 
Did you create a zero byte file with that name?

I had to do that with a 7960 and it was very happy.
I was able to successfully load version 7.1 of SIP, and all seems to 
work with Asterisk.
And continues to work
I used the Solar Winds TFTP server on Win2K.

I have been unable to get version 7.3 or later to work properly though.
It will receive calls, through out no error messages, but is unable to 
place calls.
I am sure there is something it wants in one of the xml files, but I 
have yet to find it.
The sample files on the Wiki were used, work with 7.1, but not with 7.3
Though not directly related to Asterisk, that hasn't stopped thousands 
of other Cisco related postings.

John Novack

-- 
Dog is my co-pilot


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Re: [asterisk-users] asterisk CLI and no such command "stop"

[Vieri]

--- Tomás Laureano Peralta Tormey
<[EMAIL PROTECTED]> wrote:

> Vieri:
> You will need to specify to the stop command, when
> to stop. The options are:
> now, when convenient or gracefully.
> Running the command 'help stop' inside the CLI will
> give an idea of this
> options.

Thanks, I know that 'stop' needs arguments but if I
issue the 'help stop' command I get:
*CLI> help stop
No such command 'stop'.

Will look into the logs more carefully because I
suppose that if the asterisk process is up then it
should always have a cli 'stop' command available.



  

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Re: [asterisk-users] Multi-SPAN (4xE1) Zap Group (Outbound)

[Alejandro Kauffmann]

Mike Trest - Personal wrote:
> Hi,
> Can someone point me to a zapata.conf example that will create a 
> single DIAL OUT
> group including all 4 spans on a TE4XXP?
>
> One friend says to change the group number all to "1" on all 4 spans.
> Another suggestions says it is possible to have these unique groups (1-4)
> and to combine all 4 into a single group "5".
>
> I like the second suggestion best.
>
> Can you guide me to the correct changes for my current zapata.conf?
> The 4 spans are stand alone E1/PRI trunks (Not NFAS).
>
> The CURRENT channel and group statements are:
> ;Span  1  group=1 channel  => 1-15,17-31
> ;Span  2  group=2 channel  => 32-46,48-62
> ;Span  3  group=3 channel  => 63-77,79-93
> ;Span  4  group=4 channel  => 94-103,110-124
>
>
>
>
> Thanks,  ..mike..
>
>
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>
>   
Try:

group=0,1
channel  => 1-15,17-31
group=0,2
channel  => 32-46,48-62
group=0,3
channel  => 63-77,79-93
group=0,4
channel  => 94-103,110-124

This allows you to use group 0 to dial out over all 4 spans, but each span 
still has it's own
group that you can use to troubleshoot.  You can break this down even further 
if you need.


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Re: [asterisk-users] Remote hold on PRI

[Shane Spencer]

So, watching the asterisk console with full debug on shows something
about "Starting Music On Hold for Channel xx/yy-zz"?

Shane

On Jan 7, 2008 11:00 AM, Gaëtan Minet <[EMAIL PROTECTED]> wrote:
>
> Hi
>
> Nobody has an Idea ? Should I try and fill a bug report (or feature request
> ?) at Digium  ?
> The only solution I personally see is a patch in the source.
>
> Regards
>
> Gaetan
>
>
>
> On 04/01/2008, at 23:26, Gaëtan Minet wrote:
> Hi everybody
>
> We have a strange problem with several asterisk servers (Version
> 1.4.11) using PRI cards (tied to telco here in Belgium).
>
> Indeed we noticed that whenever a local user places an outgoing call
> through the PRI (and telco) to another IPBX (tied to telco using BRI
> or PRI), if the remote party places the call on hold, the caller hears
> the _local_ music on hold instead of the remote one.  In fact we can
> briefly hear the remote music on hold start, then it is replaced by
> the local one.
>
> More precisely:
>
> Company 1 uses an asterisk server with a PRI card tied to the telco.
> Company 2 uses any PBX that ca place calls on hold and is tied to the
> telco using a digital interface (tested with BRIs and PRIs)
>
> A (company 1) calls B (company 2)
> B answers and park or places the call on hold
> A hears the MOH of company 1.
>
> The same happens when calling a mobile: when the mobile user puts the
> call on hold, instead of hearing the mobile operator's own moh, the
> calling user hears the moh of his own company asterisk.
>
> I think this has something to do with REMOTE_HOLD notifications on PRI
> lines that gets reported back to the calling asterisk server, which in
> turn somehow puts the bridged (SIP) channel on hold, but I can't find
> much more information about this.
> Is this the expected behavior ? A feature or a bug ? Do you know if
> this can be tuned/tweaked/disabled (i.e. filter or ignore this
> signaling on the zap channel(s) ?)
>
> Kind regards
> Thanks
>
> NB: Oddly enough, when the local user hears the music on hold, his own
> channel (a local SIP phone in this case) isn't reported as "On Hold"
> when issuing "sip show channels" in cli,  and no AMI Hold/Unhold
> events are generated. I double checked, the MOH that gets played is
> the one specified in sip.conf, NOT zapata.conf.
>
>
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>
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[asterisk-users] GotoIf() help

[Glenn Cobb]

Greetings all,
 
I'm not real good with dial plan programming and need some help. I've looked
at the 2nd edition of the Asterisk book about GotoIf() and have a basic idea
what I need to do but not sure about the correct way or the best way, to set
it up. I need to branch based on whether the dialed number is long distance
(international or not) or not. I have branch offices on SIP and IAX trunks
that have 4 digit extensions and one office has a 1000 range for their
extensions so I have to make sure I don't pick that up as dialing long
distance. I think what I have below will work but it can probably be cleaned
up alot. Any help is greatly appreciated.
 
 
exten => s,n,GotoIf($[${DIAL_NUMBER} = 011. ] ? yescode : steptwo)
 
exten => s,n,(steptwo),GotoIf($[${DIAL_NUMBER} = 9XX. ] ? yescode :
stepthree)
 
exten => s,n,(stepthree),GotoIf($[${DIAL_NUMBER} = 1NXXNX. ] ? yescode :
nocode)
exten => s,n,(yescode),Playback(please-enter-the&accounting)

exten => s,n,Read(account|number|8)

exten => s,n,SetAccount(${account})

exten => s,n,(nocode),Blah, Blah

 
Thanks,
 
Glenn
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Re: [asterisk-users] no outgoing calls with Digium B410P

[Andres Paglayan]

On Jan 7, 2008, at 8:53 AM, daniele visaggio wrote:




2008/1/7, map <[EMAIL PROTECTED]>:
Hi Daniele,

Please send a snapshot of your Putty Asterisk log.
Go to Putty configuration -> Window -> Lines of scrollback and put  
a number greater than 200 :-). I suggest 10.


Sorry, i'm using a Linux version of putty (i'm running Ubuntu) and  
the configuration is different. I can't find "Lines of scrollback"  
and modify the scrollback number. The putty-linux sw-structure is  
probably different from the putty-windows one.


you can use a regular terminal and issue the script command
all the output will be saved to a file,



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Re: [asterisk-users] Remote hold on PRI

[Gaëtan Minet]

Hi

Nobody has an Idea ? Should I try and fill a bug report (or feature  
request ?) at Digium  ?

The only solution I personally see is a patch in the source.

Regards

Gaetan

On 04/01/2008, at 23:26, Gaëtan Minet wrote:


Hi everybody

We have a strange problem with several asterisk servers (Version
1.4.11) using PRI cards (tied to telco here in Belgium).

Indeed we noticed that whenever a local user places an outgoing call
through the PRI (and telco) to another IPBX (tied to telco using BRI
or PRI), if the remote party places the call on hold, the caller hears
the _local_ music on hold instead of the remote one.  In fact we can
briefly hear the remote music on hold start, then it is replaced by
the local one.

More precisely:

Company 1 uses an asterisk server with a PRI card tied to the telco.
Company 2 uses any PBX that ca place calls on hold and is tied to the
telco using a digital interface (tested with BRIs and PRIs)

A (company 1) calls B (company 2)
B answers and park or places the call on hold
A hears the MOH of company 1.

The same happens when calling a mobile: when the mobile user puts the
call on hold, instead of hearing the mobile operator's own moh, the
calling user hears the moh of his own company asterisk.

I think this has something to do with REMOTE_HOLD notifications on PRI
lines that gets reported back to the calling asterisk server, which in
turn somehow puts the bridged (SIP) channel on hold, but I can't find
much more information about this.
Is this the expected behavior ? A feature or a bug ? Do you know if
this can be tuned/tweaked/disabled (i.e. filter or ignore this
signaling on the zap channel(s) ?)

Kind regards
Thanks

NB: Oddly enough, when the local user hears the music on hold, his own
channel (a local SIP phone in this case) isn't reported as "On Hold"
when issuing "sip show channels" in cli,  and no AMI Hold/Unhold
events are generated. I double checked, the MOH that gets played is
the one specified in sip.conf, NOT zapata.conf.


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Re: [asterisk-users] asterisk CLI and no such command "stop"

[Tomás Laureano Peralta Tormey]

Vieri:
You will need to specify to the stop command, when to stop. The options are:
now, when convenient or gracefully.
Running the command 'help stop' inside the CLI will give an idea of this
options.

Best regards, Tomás.

On Jan 7, 2008 5:21 PM, Vieri <[EMAIL PROTECTED]> wrote:

> Hi,
>
> I'm probably missing something trivial but I don't
> understand what.
>
> Asterisk is loading fine but when I connect to the
> console (asterisk -vr) and type "stop" I get a no such
> command reply:
>
> *CLI> help
> (...)
>skinny show lines  Show defined Skinny lines
> per device
>  soft hangup  Request a hangup on a given
> channel
>   unload  Unload a dynamic module by
> name
> *CLI> stop
> No such command 'stop' (type 'help' for help)
>
> # tail -n 1000 /var/log/messages | grep -i error
> #
>
> Does anyone know why the "stop" command doesn't appear
> on the help list and is unavailable?
>
>
>
>
>
>  
> 
> Never miss a thing.  Make Yahoo your home page.
> http://www.yahoo.com/r/hs
>
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[asterisk-users] asterisk CLI and no such command "stop"

[Vieri]

Hi,

I'm probably missing something trivial but I don't
understand what. 

Asterisk is loading fine but when I connect to the
console (asterisk -vr) and type "stop" I get a no such
command reply:

*CLI> help
(...)
skinny show lines  Show defined Skinny lines
per device
  soft hangup  Request a hangup on a given
channel
   unload  Unload a dynamic module by
name
*CLI> stop
No such command 'stop' (type 'help' for help)

# tail -n 1000 /var/log/messages | grep -i error
#

Does anyone know why the "stop" command doesn't appear
on the help list and is unavailable?




  

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[asterisk-users] [Asterisk 1.2 + TDM FXO] Incoming call not detected

[Vincent]

Hi

On an old IBM Netvista thinclient, the TDM card doesn't detect
incoming calls, although the card seems to be detected, and correctly
configured:

pbx asterisk # lspci
00:03.0 Communication controller: Tiger Jet Network Inc. Tiger3XX
Modem/ISDN interface

# cat /etc/zaptel.conf 
fxsks=1
loadzone=fr
defaultzone=fr

# ztcfg -vv
Zaptel Configuration
==
Channel map:
Channel 01: FXS Kewlstart (Default) (Slaves: 01)
1 channels configured.

# zttool (with the phone line plugged into the FXO port)
OK Wildcard TDM400P REV E/F Board 1
=> Select
Current Alarms: No alarms.
IRQ Misses:   0
Bipolar Viol: 0
Tx/Rx Levels: 0/  0
Total/Conf/Act:   0/  0/  0

There is dialtone on the line, and the green LED is lit up after
loading the Zaptel driver, but when I connect to Asterisk, nothing is
reported when I call into the FXO port.

Is the motherboard not compatible with this PCI card? Are there
settings I can try to get the card to detect the call?

Thank you.


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[asterisk-users] Media gateways and video

[Olivier]

Hi,

Asterisk now supports h234m.
Does anyone know a Media gateway such as those of Mediatrix, Patton,
Audiocodes, Cisco that also supports h324m flows ?

Prospective setup would be:
ISDN  Media gateway -- -- Asterisk --- --- Videosoftphone

Cheers
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Re: [asterisk-users] FWD and IPCall

[Shane D]

I will try that when I return home in about two hours. I'll let you know.

On 1/7/08, Benchev <[EMAIL PROTECTED]> wrote:
> > > I have a problem. I have tried everything that is in the book "The
> > > Future of Telephony" as well as on the FWD (freeworlddialup) website,
> > > and there is still a problem. My asterisk box is not able to associate
> > > with the FWD server. I get:
> > > Registration Rejected by [insert IP], and I can't use my IPCall number
> > > to reach my Asterisk box.
> > > Any suggestions?
> If you try dnsmgr.conf
> enable=yes; what happens?
>
> Boyko
>
>
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-- 
-Shane
Blog: http://blind-geek.com/blog/
CoOwner: http://sjtechzone.com
AIM: inhaddict
Skype: chatter8712

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Re: [asterisk-users] FWD and IPCall

[Benchev]

> > I have a problem. I have tried everything that is in the book "The
> > Future of Telephony" as well as on the FWD (freeworlddialup) website,
> > and there is still a problem. My asterisk box is not able to associate
> > with the FWD server. I get:
> > Registration Rejected by [insert IP], and I can't use my IPCall number
> > to reach my Asterisk box.
> > Any suggestions?
If you try dnsmgr.conf
enable=yes; what happens?

Boyko


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Re: [asterisk-users] Detailed Instructions

[Hans Witvliet]

On Mon, 2008-01-07 at 01:50 +0200, Tzafrir Cohen wrote:
> On Mon, Jan 07, 2008 at 12:41:11AM +0100, Hans Witvliet wrote:
> > On Sat, 2008-01-05 at 13:36 -0500, Shane D wrote:
> > > Hello List,
> > > 
> > > I am getting Asterisk set up. I am going to be installing Debian Linux
> > > on a laptop later. I would appreciate some detailed instructions on:
> > > 
> > > (A) What to type into the shell to download and install Asterisk.
> > > (B) How to open the configuration files (*.conf)
> > > (C) If there is a way that I can change the configuration files remotely 
> > > (SSH?).
> > > 
> > > Thanks in advance.
> > > 
> .
> 
> Is SUSE? What version?
> 
> > 
> > For the later, there are prebuild packages and regular updates, see:
> > http://ftp5.gwdg.de/pub/opensuse/repositories/network:/telephony/

OpenSuSE 10,1, 10.2, 10.3, SLES10,SLED10 (see above link)
Even has misdn precompiled...



[ ] asterisk-1.4.17-3.2.x86_64.rpm
04-Jan-2008 12:57  9.5M  
[ ] asterisk-addons-1.4.5_1.4.10-7.13.x86_64.rpm   04-Jan-2008 
13:01  948K  
[ ] asterisk-alsa-1.4.17-3.2.x86_64.rpm04-Jan-2008 
12:57   69K  
[ ] asterisk-app_ldap-2.0rc1-6.31.x86_64.rpm   04-Jan-2008 
13:00   17K  
[ ] asterisk-debuginfo-1.4.17-3.2.x86_64.rpm   04-Jan-2008 
12:57  7.9M  
[ ] asterisk-devel-1.4.17-3.2.x86_64.rpm   04-Jan-2008 
12:57  138K  
[ ] asterisk-flite-0.5-3.31.x86_64.rpm 04-Jan-2008 
13:02   25K  
[ ] asterisk-odbc-1.4.17-3.2.x86_64.rpm04-Jan-2008 
12:57   39K  
[ ] asterisk-perl-0.09-1.3.x86_64.rpm  16-Nov-2007 
16:51   28K  
[ ] asterisk-spandsp-1.4.17-3.2.x86_64.rpm 04-Jan-2008 
12:57   23K  
[ ] asterisk-zaptel-1.4.17-3.2.x86_64.rpm  04-Jan-2008 
12:57  135K  
[ ] callweaver-RC-1.1.99.20071204-1.4.x86_64.rpm   21-Dec-2007 
16:23  1.6M  
[ ] callweaver-RC-alsa-1.1.99.20071204-1.4.x86_64.rpm  21-Dec-2007 
16:23   57K  
[ ] callweaver-RC-bluetooth-1.1.99.20071204-1.4.x86_64.rpm 21-Dec-2007 
16:23   28K  
[ ] callweaver-RC-capi-1.1.99.20071204-1.4.x86_64.rpm  21-Dec-2007 
16:23  2.0K  
[ ] callweaver-RC-devel-1.1.99.20071204-1.4.x86_64.rpm 21-Dec-2007 
16:23  114K  
[ ] callweaver-RC-jabber-1.1.99.20071204-1.4.x86_64.rpm21-Dec-2007 
16:23   20K  
[ ] callweaver-RC-javascript-1.1.99.20071204-1.4.x86_64.rpm21-Dec-2007 
16:23  1.9K  
[ ] callweaver-RC-ldap-1.1.99.20071204-1.4.x86_64.rpm  21-Dec-2007 
16:23   11K  
[ ] callweaver-RC-ogi-1.1.99.20071204-1.4.x86_64.rpm   21-Dec-2007 
16:23   36K  
[ ] callweaver-RC-postgresql-1.1.99.20071204-1.4.x86_64.rpm21-Dec-2007 
16:23   15K  
[ ] ccrtp-1.5.0-64.5.x86_64.rpm27-Nov-2007 
22:08  467K  
[ ] ccrtp-devel-1.5.0-64.5.x86_64.rpm  27-Nov-2007 
22:08  874K  
[ ] check-0.9.5-2.3.x86_64.rpm 16-Nov-2007 
17:26   56K  
[ ] check-devel-0.9.5-2.3.x86_64.rpm   16-Nov-2007 
17:26   55K  
[ ] commoncpp2-1.5.9-2.1.x86_64.rpm27-Nov-2007 
21:55  239K  
[ ] commoncpp2-devel-1.5.9-2.1.x86_64.rpm  27-Nov-2007 
21:55  296K  
[ ] commoncpp2-doc-1.5.9-2.1.x86_64.rpm27-Nov-2007 
21:55   41K  
[ ] flite-1.3-8.3.x86_64.rpm   16-Nov-2007 
16:53  6.6M  
[ ] flite-devel-1.3-8.3.x86_64.rpm 16-Nov-2007 
16:53   25K  
[ ] freepbx-2.3.0-3.34.x86_64.rpm  04-Jan-2008 
13:01  3.4M  
[ ] freeradius-client-1.1.5-3.2.x86_64.rpm 05-Jan-2008 
17:46   47K  
[ ] freeradius-client-debuginfo-1.1.5-3.2.x86_64.rpm   05-Jan-2008 
17:46   13K  
[ ] freeradius-client-devel-1.1.5-3.2.x86_64.rpm   05-Jan-2008 
17:46   29K  
[ ] freeradius-client-libs-1.1.5-3.2.x86_64.rpm05-Jan-2008 
17:46   26K  
[ ] freeradius-client-snapshot-20080105-7.1.x86_64.rpm 06-Jan-2008 
00:20   41K  
[ ] freeradius-client-snapshot-debuginfo-20080105-7.1.x86_64.rpm   06-Jan-2008 
00:20   14K  
[ ] freeradius-client-snapshot-devel-20080105-7.1.x86_64.rpm   06-Jan-2008 
00:20   31K  
[ ] freeradius-client-snapshot-libs-20080105-7.1.x86_64.rpm06-Jan-2008 
00:20   28K  
[ ] freeswitch-snapshot-6382-10.1.x86_64.rpm   28-Nov-2007 
16:41  3.0M  
[ ] freeswitch-snapshot-codec-passthru-amr-6382-10.1.x86_64.rpm28-Nov-2007 
16:41  5.3K  
[ ] freeswitch-snapshot-codec-passthru-g723_1-6382-10.1.x86_64.rpm 28-Nov-2007 
16:41  5.3K  
[ ] freeswitch-snapshot-codec-passthru-g729-6382-10.1.x86_64.rpm   28-Nov-2007 
16:41  5.5K  
[ ] freeswitch-snapshot-debuginfo-6382-10.1.x86_64.rpm 28-Nov-2007 
16:41  1.3M  
[ 

Re: [asterisk-users] FWD and IPCall

[Shane D]

It's Iax2. Is there a way of using amore reliable sip
connectoin/something slightly different?

If so, how would I go about that.

On 1/7/08, Huw Richards <[EMAIL PROTECTED]> wrote:
> You haven't said if your connection to fwd is SIP or IAX2 but I have
> found IAX2 connections to fwd to be unreliable. Other people may have
> different results.
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Shane D
> Sent: Monday, January 07, 2008 10:17
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] FWD and IPCall
>
> Hello All,
>
> I have a problem. I have tried everything that is in the book "The
> Future of Telephony" as well as on the FWD (freeworlddialup) website,
> and there is still a problem. My asterisk box is not able to associate
> with the FWD server. I get:
> Registration Rejected by [insert IP], and I can't use my IPCall number
> to reach my Asterisk box.
> Any suggestions?
> --
> -Shane
> Blog: http://blind-geek.com/blog/
> CoOwner: http://sjtechzone.com
> AIM: inhaddict
> Skype: chatter8712
>
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-- 
-Shane
Blog: http://blind-geek.com/blog/
CoOwner: http://sjtechzone.com
AIM: inhaddict
Skype: chatter8712

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Re: [asterisk-users] Cisco 7941G-GE with Asterisk and CTPSEP odyssee

[Patrick]

On Mon, 2008-01-07 at 12:15 +0100, Christophorus Laube wrote:
> Update and revision:
> I now downloaded the oldest gettable SIP firmware for 7941/61, i.e.
> 8.0.2. I always get the same behaviour. But I realized it never got to
> the SIP image completely loaded status.
> I bought this phone and it had - no wonder - an SCCP image installed.
> When plugging that into an ethernet port the first thing it does is
> requesting an IP address and afterwards the CTLSEP.tlv file. In the
> status section I see an SCCP firmware entry. When I do a factory reset
> (that should be the right way to get the SIP firmware on such a phone,
> right?) it now loads the term41.default.loads and some other files and
> then reboots and requests the CTLSEP.tlv file. The firmware entry
> in the status section now says "term41.default.loads". Getting over this
> CTLSEP step should bring the phone to load the SIP41XXX.loads file, I
> assume. 
> But as I am not getting over this step it stays in the
> term41.default.loads step, unfortunately. 
> Does that ring a bell to anyone? Does anyone of you have had the same
> situation? In which state did you get the 7961G? SCCP? And how did you
> manage to load SIP firmware onto it?

Afaik you first need to upgrade to the latest 7.x and then to 8.x.x.
That's the only way it worked for me (7961G initially on factory default
SCCP).

Regards,
Patrick


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Re: [asterisk-users] FWD and IPCall

[Huw Richards]

You haven't said if your connection to fwd is SIP or IAX2 but I have
found IAX2 connections to fwd to be unreliable. Other people may have
different results.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Shane D
Sent: Monday, January 07, 2008 10:17
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] FWD and IPCall

Hello All,

I have a problem. I have tried everything that is in the book "The
Future of Telephony" as well as on the FWD (freeworlddialup) website,
and there is still a problem. My asterisk box is not able to associate
with the FWD server. I get:
Registration Rejected by [insert IP], and I can't use my IPCall number
to reach my Asterisk box.
Any suggestions?
-- 
-Shane
Blog: http://blind-geek.com/blog/
CoOwner: http://sjtechzone.com
AIM: inhaddict
Skype: chatter8712

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[asterisk-users] service provider connection problem

[srinivas Antarvedi]

Hello all,

Can anyone have any experience working with service provider
like Talkfree .

They are giving the  user accounts based on the  single  user accounts
and those needs to be directly register to the service provider not to the
local system

i have taken a connection which when configured to service providers domain
direclty ,xlite can make calls without any problem but if i want to use it
using
my asterisk server (for a simulation to call center) the service provider is

asking for 407 proxy authentication and i am unable to resolve this issue

can anyone have any circumventing ideas to this solution


thanks and regards
srinivas antarvedi
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Re: [asterisk-users] no outgoing calls with Digium B410P

[daniele visaggio]

2008/1/7, map <[EMAIL PROTECTED]>:
>
> Hi Daniele,
>
> Please send a snapshot of your Putty Asterisk log.
> Go to Putty configuration -> Window -> Lines of scrollback and put a
> number greater than 200 :-). I suggest 10.
>
> Sorry, i'm using a Linux version of putty (i'm running Ubuntu) and the
configuration is different. I can't find "Lines of scrollback" and modify
the scrollback number. The putty-linux sw-structure is probably different
from the putty-windows one.
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Re: [asterisk-users] no outgoing calls with Digium B410P

[Tzafrir Cohen]

On Mon, Jan 07, 2008 at 10:19:11AM -0500, John Novack wrote:

> The ONLY issue I have had with PuTTY is  ( my ) inability to run  make 
> menuselect. regardless of how I set PuTTY, it complains about terminal size.

Hmm.. I have never encountered this. Sounds like a bug. Can you point me
to a bug report or to paste here the exact message?

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

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[asterisk-users] pickup application failed

[Rilawich Ango]

I have a TDM400 in the server.  I want to press **1XX to pickup a
call.  It is ok if I pickup a call dialled from 1XX to 1YY (internal
network call).  However, it is failed to pick up a call from PSTN
thro' TDM400 card.  It seems I can't guess the correct context of it.
How can I know the context of  the call using CLI?  The default
context of the TDM400 is from-pstn but pickup still failed if I add
exten => _**1XX,n,Pickup(${EXTEN:[EMAIL PROTECTED]) in the end of context
BLF_group_pickup.

[BLF_group_pickup]
exten => _**1XX,1,Pickup(${EXTEN:[EMAIL PROTECTED])
exten => _**1XX,n,Pickup(${EXTEN:[EMAIL PROTECTED])

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Re: [asterisk-users] Which IP Phone is really the best?

[C F]

strike out stable on the cisco phones. they are not stable.

On 1/7/08, Tim Connolly <[EMAIL PROTECTED]> wrote:
> Cisco 7960's:  (SIPified)
> 1. Cheap
> 2. 6 lines is plenty
> 3. simple to config
> 4. stable
>
> On Jan 6, 2008, at 11:03 PM, William Herrera wrote:
>
> > Alright, enough.
> > At first I was to ignore to you all making statements like this one
> > but I
> > feel at this point that if I do not stop this it seems it will never
> > stop.
> > First thing first. I have a Bach. in Network Engineering. I did work
> > for the
> > Telefónica of Puerto Rico installing Asterisk (and working with
> > Polycom,
> > Cisco, Astra and Grandstream) for a bit over 2 years. I have been
> > doing this
> > now on my own business since October 2003 (www.lan-solutions.net),
> > so I am
> > not as you might think I am.
> > I asked a "simple" question just to hear your opinion. It was not
> > intended
> > for so many of you waste your time (and mine) writing all this
> > useless notes
> > 
> > If you would have taken the same (or less) time just to answer the
> > question
> > (or to ignore it) we al would have been able to keep it "simple", as
> > intended...
> > Case closed.
> >
> > WH
> >
> >
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of David
> > Cook
> > Sent: Sunday, January 06, 2008 11:42 PM
> > To: asterisk-users@lists.digium.com
> > Subject: Re: [asterisk-users] Which IP Phone is really the best?
> >
> > Seriously, if you intend on proposing this to a customer it means
> > you are
> > selling your professional services. If you are asking questions like
> > this,
> > how successful do you expect your customer engagement to be?
> >
> > Even if someone recommends the "best" phone for your particular
> > application,
> > you will still have zero competency with it and spend inordinate
> > amounts of
> > learning time and re-work on the customer's time. Your inexperience
> > will
> > show. Customers are demanding and you will get thrown out on your a**.
> > People expect IT to fail from time to time (unfortunately), but they
> > expect
> > 100% availability from their phones. Anything less and you will find
> > yourself with a priority meeting at the client that includes your
> > manager,
> > CEO and their lawyer.
> >
> > Nothing travels faster than a bad reputation. Walk away. Research.
> > Build a
> > lab. Learn.
> >
> > - dbc.
> >
> > From: "William Herrera" <[EMAIL PROTECTED]>
> > Subject:
> > To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
> > 
> >
> >> I need to quote a client for a job and I was just wondering.
> >>
> >> Out of all the IP Phones out there, which one is the best and why?
> >>
> >> Thank you all, all opinions will be accepted.
> >>
> >> William Herrera
> >> LAN/WAN Technical Consultant
> >
> >
> >
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> > To UNSUBSCRIBE or update options visit:
> >   http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> > __ NOD32 2767 (20080106) Information __
> >
> > This message was checked by NOD32 antivirus system.
> > http://www.eset.com
> >
> >
> >
> > __ NOD32 2767 (20080106) Information __
> >
> > This message was checked by NOD32 antivirus system.
> > http://www.eset.com
> >
> >
> >
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>
>
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[asterisk-users] Multi-SPAN (4xE1) Zap Group (Outbound)

[Mike Trest - Personal]

Hi,
Can someone point me to a zapata.conf example that will create a 
single DIAL OUT
group including all 4 spans on a TE4XXP?

One friend says to change the group number all to "1" on all 4 spans.
Another suggestions says it is possible to have these unique groups (1-4)
and to combine all 4 into a single group "5".

I like the second suggestion best.

Can you guide me to the correct changes for my current zapata.conf?
The 4 spans are stand alone E1/PRI trunks (Not NFAS).

The CURRENT channel and group statements are:
;Span  1group=1 channel  => 1-15,17-31
;Span  2group=2 channel  => 32-46,48-62
;Span  3group=3 channel  => 63-77,79-93
;Span  4group=4 channel  => 94-103,110-124




Thanks,  ..mike..


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Re: [asterisk-users] no outgoing calls with Digium B410P

[map]

Hi Daniele,

Please send a snapshot of your Putty Asterisk log.
Go to Putty configuration -> Window -> Lines of scrollback and put a number
greater than 200 :-). I suggest 10.

On Jan 7, 2008 4:00 PM, daniele visaggio <[EMAIL PROTECTED]> wrote:

> 2008/1/7, map <[EMAIL PROTECTED]>:
> >
> > Daniele,
> > you need an "external calls" rule in your extension.conf, that is 1 to
> > call using PSTN line.
> >
> > Please send your extension and we can take a look to find your problem.
> >
> > p.s.
> > I'm Italian too.
> >
> > Ok, i attach my extension.conf.
>
> Thank you very much, i'm very happy of finding another italian asterisk
> user.
>
> Ciao e grazie!
>
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Re: [asterisk-users] no outgoing calls with Digium B410P

[John Novack]

Tzafrir Cohen wrote:
> On Mon, Jan 07, 2008 at 03:03:18PM +0100, daniele visaggio wrote:
>
>   
>> I'm managing the asterisk server from a windows client via ssh (putty
>> client), so i can't paste here the output of the asterisk CLI, 
>> 
>
> Huh???
>
> * You can record history in putty.
> * You can manually copy text from putty( just mark the text)
> * You can save text as a local file (e.g: copy from the log and edit)
>   and copy it via sftp (e.g: winscp).
>   
PuTTY   IS the way to manage Asterisk.
One can scroll back, can set the scrollback buffer to many thousands of 
lines if need be, cut and paste with ease directly into an E-mail and 
the suite also comes with PSFTP to do file transfer.

The ONLY issue I have had with PuTTY is  ( my ) inability to run  make 
menuselect. regardless of how I set PuTTY, it complains about terminal size.

Otherwise I have no reason to even use the console on the Asterisk box.

John Novack

-- 
Dog is my co-pilot


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[asterisk-users] FWD and IPCall

[Shane D]

Hello All,

I have a problem. I have tried everything that is in the book "The
Future of Telephony" as well as on the FWD (freeworlddialup) website,
and there is still a problem. My asterisk box is not able to associate
with the FWD server. I get:
Registration Rejected by [insert IP], and I can't use my IPCall number
to reach my Asterisk box.
Any suggestions?
-- 
-Shane
Blog: http://blind-geek.com/blog/
CoOwner: http://sjtechzone.com
AIM: inhaddict
Skype: chatter8712

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Re: [asterisk-users] no outgoing calls with Digium B410P

[daniele visaggio]

2008/1/7, map <[EMAIL PROTECTED]>:
>
> Daniele,
> you need an "external calls" rule in your extension.conf, that is 1 to
> call using PSTN line.
>
> Please send your extension and we can take a look to find your problem.
>
> p.s.
> I'm Italian too.
>
> Ok, i attach my extension.conf.

Thank you very much, i'm very happy of finding another italian asterisk
user.

Ciao e grazie!


extension.conf.tar.gz
Description: GNU Zip compressed data
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Re: [asterisk-users] How to check if a SIP phone is forwarded without ringing it ?

[Kevin P. Fleming]

Olivier wrote:

> Is there way for an Asterisk server to check if a sip phone is forwarded
> without bothering phone's user ?

No.

> I was thinking of some Alert-Info option that would let the phone reply
> with a 302 Moved Temporarily or 182 Queued message and not let the phone
> ring or display anything on its screen.

According to the SIP RFC, a SIP endpoint is supposed to respond to an
OPTIONS message the same way that it would respond to an INVITE message
with the identical destination, but I've never seen a phone respond to
an OPTIONS message with anything but '200 OK', even when a redirect
(forward) is in place.

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

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[asterisk-users] How to check if a SIP phone is forwarded without ringing it ?

[Olivier]

Hi,

I feel I've read a thread about this previously but I couldn't find it.

Is there way for an Asterisk server to check if a sip phone is forwarded
without bothering phone's user ?
I was thinking of some Alert-Info option that would let the phone reply with
a 302 Moved Temporarily or 182 Queued message and not let the phone ring or
display anything on its screen.

So that, you could just cancel this call to be able to tell if an extension
is forwarded without bothering users.

Regards
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Re: [asterisk-users] no outgoing calls with Digium B410P

[Tzafrir Cohen]

On Mon, Jan 07, 2008 at 03:03:18PM +0100, daniele visaggio wrote:

> I'm managing the asterisk server from a windows client via ssh (putty
> client), so i can't paste here the output of the asterisk CLI, 

Huh???

* You can record history in putty.
* You can manually copy text from putty( just mark the text)
* You can save text as a local file (e.g: copy from the log and edit)
  and copy it via sftp (e.g: winscp).

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

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Re: [asterisk-users] no outgoing calls with Digium B410P

[map]

Daniele,
you need an "external calls" rule in your extension.conf, that is 1 to call
using PSTN line.

Please send your extension and we can take a look to find your problem.

p.s.
I'm Italian too.

On Jan 7, 2008 3:03 PM, daniele visaggio <[EMAIL PROTECTED]> wrote:

> Hi Daniele,
> >
> > Could you please tell us what exactly happens?
> > Are your able to see some error in the log/console?
> >
> >
> > Thanks for your answer.
>
> I'm managing the asterisk server from a windows client via ssh (putty
> client), so i can't paste here the output of the asterisk CLI, but when i
> try to do a call to the PSTN i see a lot o f messages, but no one of them
> looks like an error. They are of this type:
>
> -- Executing [EMAIL PROTECTED]:1] Macro ("SIP/501-0827c9e82,
> dialout-trunk|1|0266200xxx||") in new stack
>
> 0266200xxx is the number i'm trying calling to, 501 is my extension number
> (i have one SIP hard-phone); the number 1 before 0266200xxx is part of the
> dial patterns i created, because i want to dial 1 before the outgoing
> number.
>
> Anyway, on the official digium documentation, it's written that "in order
> to call out over a port, the Dial () command has to be formatted as follows:
> Dial (misdn/g:myoutsidelines/ $ {EXTEN}). Have i to edit my extension.confand 
> insert in a dial command to do an outgoing call?
>
> Thanks a lot - Daniele
>
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Re: [asterisk-users] no outgoing calls with Digium B410P

[Doug Lytle]

daniele visaggio wrote:
> Thanks for your answer.
>
> I'm managing the asterisk server from a windows client via ssh (putty 
> client), so i can't paste here the output of the asterisk CLI, but 
> when i try to do a call to the PSTN i see a lot o f messages, but 


Putty does support copying all the the clipboard.  It's in the pull down 
menu.  I usually paste it into a notepad document and remove unwanted text.

Doug


-- 
 
Ben Franklin quote:

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



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Re: [asterisk-users] no outgoing calls with Digium B410P

[daniele visaggio]

>
> Hi Daniele,
>
> Could you please tell us what exactly happens?
> Are your able to see some error in the log/console?
>
>
> Thanks for your answer.

I'm managing the asterisk server from a windows client via ssh (putty
client), so i can't paste here the output of the asterisk CLI, but when i
try to do a call to the PSTN i see a lot o f messages, but no one of them
looks like an error. They are of this type:

-- Executing [EMAIL PROTECTED]:1] Macro ("SIP/501-0827c9e82,
dialout-trunk|1|0266200xxx||") in new stack

0266200xxx is the number i'm trying calling to, 501 is my extension number
(i have one SIP hard-phone); the number 1 before 0266200xxx is part of the
dial patterns i created, because i want to dial 1 before the outgoing
number.

Anyway, on the official digium documentation, it's written that "in order to
call out over a port, the Dial () command has to be formatted as follows:
Dial (misdn/g:myoutsidelines/ $ {EXTEN}). Have i to edit my
extension.confand insert in a dial command to do an outgoing call?

Thanks a lot - Daniele
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Re: [asterisk-users] zaptel programming

[Philipp Kempgen]

Lee Jenkins wrote:
> Philipp Kempgen wrote:
>> Bhrugu Mehta wrote:
>>
>>> I am new to zaptel programming.
>>> can anybody help me how to start this. or any ref. site or matirial 
>>> availabel.
>>> i want to use c lang. for this.
>> 
>>  Some tutorials:
>>  http://www.google.com/search?q=learn+c+in+21+days
>>  When done ask for commit access to
>>  http://svn.digium.com/view/zaptel/
>> 

> That was an inappropriate answer, Phillip.  The OP said he was new to zaptel 
> programming, not necessarily C programming.  You could have actually just 
> ignored the post/query for real without wasting bandwidth and being uncivil 
> just 
> to act superior and show everyone you can write a simple xml document.

You are right, I should just ignore posts when I don't have
anything useful to add. Sorry.

Regards,
  Philipp Kempgen

-- 
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Let's use IT to solve problems and not to create new ones.
  Asterisk? -> http://www.das-asterisk-buch.de

Geschäftsführer: Stefan Wintermeyer
Handelsregister: Neuwied B 14998

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Re: [asterisk-users] HPEC

[Kevin P. Fleming]

clive.chan(Atn) wrote:

> The error that I had as bellow;
> 
>> Found key 'HPEC-XX' for 2 channels.
>> Found valid HPEC licenses for 2 channels.
>> Failed to get license challenge: No such device

Please contact Digium support for assistance with using HPEC since you
purchased it.

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

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Re: [asterisk-users] extension.conf with mysql

[Doug Lytle]

Gopal krishnan wrote:
> Hi,
>
>I am trying to connect the outbound dialing with mysql with the 
> following code,
>
> exten => 88,1,MYSQL(Connect connid hostname username password dbname)
> exten => 88,2,GotoIf($["${connid}" = ""]?error,1)
> exten => 88,3,MYSQL(Query resultid ${connid} SELECT\ phone\ FROM\ 
> \ WHERE\ phone =${a})

This is what I do:

exten => s,n,MYSQL(Connect connid 192.168.103.15 AsteriskQuery 
'somepass' Administration)
exten => s,n,GotoIf($["${MYSQL_STATUS}" = "-1"]?mysql_failed,s,1)
exten => s,n,MYSQL(Query resultid ${connid} SELECT name \, access \, 
password \, facility \, paging \, afterhours \, lockdown FROM Passwords 
WHERE password = ${get-admin-password})
exten => s,n,MYSQL(Fetch fetchid ${resultid} admin.name admin.access 
admin.password admin.facility admin.paging admin.afterhours admin.lockdown)
exten => s,n,MYSQL(Disconnect ${connid})
exten => s,n,MYSQL(Clear ${resultid})

Doug

-- 
 
Ben Franklin quote:

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



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[asterisk-users] extension.conf with mysql

[Gopal krishnan]

Hi,

   I am trying to connect the outbound dialing with mysql with the following
code,

exten => 88,1,MYSQL(Connect connid hostname username password dbname)
exten => 88,2,GotoIf($["${connid}" = ""]?error,1)
exten => 88,3,MYSQL(Query resultid ${connid} SELECT\ phone\ FROM\
\ WHERE\ phone =${a})
exten => 88,4,MYSQL(Fetch fetchid ${resultid} ph\ sa)
...
.
...
after this I am getting confused. My moto is to display the number in the
database and need to check with my outgoing number.

how to display fetched number from the database.




-- 
Thank you  with regards,
Gopal

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Re: [asterisk-users] HFC-S zap channels always busy

[Jaap Winius]

Quoting Tzafrir Cohen <[EMAIL PROTECTED]>:

> (if you set in /etc/default/zaptel: ZAPBRI_SIGNALLING="bri"
> you'll get that from genzaptelconf.

If I create a file like this, I end up with "signalling=bri_cpe" instead of
"signalling=bri_cpe_ptmp".

> Anyway, either you use zaphfc or vzaphfc. The first one that loads takes
> everything.

So far I have succeeded in starting up Asterisk without the zaphfc
module (if channels 4-5 aren't defined in zapata.conf), but not
without vzaphfc. Not having vzaphfc loaded always results in Asterisk
starting up without Zaptel support. However, whether I run it with or
without zaphfc, all of the available ISDN channels are always busy and
the CLI still frequently shows "Primary D-Channel on span 1 down"
messages.

> What do you see on /proc/interrupts ?

CPU0
   0:   25025934IO-APIC-edge  timer
   6:  3IO-APIC-edge  floppy
   8:  1IO-APIC-edge  rtc
   9:  0   IO-APIC-level  acpi
  15:129IO-APIC-edge  ide1
 169: 286747   IO-APIC-level  skge
 177:1014894   IO-APIC-level  libata
 185:  0   IO-APIC-level  uhci_hcd:usb1, uhci_hcd:usb2, ...
 193:  114700219   IO-APIC-level  vzaphfc, zaphfc
 201:  0   IO-APIC-level  via82cxxx
 NMI:  0
 LOC:   25024942
 ERR:  0
 MIS:  0

> Which of those two modules you can't unload when Asterisk is running?

If I declare all of the channels in zapata.conf, like this...

channel => 1-2
channel => 4-5

then neither of the modules can be unloaded while Asterisk is running.
If I comment out the first line then I can unload vzaphfc, while if I
comment out the second I can unload zaphfc, so I guess this is how the
channels are related to the modules. This makes sense, because when
the zaptel modules are loaded with "genzaptelconf -d" (-d = hardware
detection), vzaphfc is always loaded first.

Regarding "genzaptelconf -d", I've found that it is essential for me
to run this command first before starting Asterisk. If not, Asterisk
will start, but without Zaptel support. During system bootup, only the
zaptel van vzaphfc modules are loaded by the kernel, which is not
enough. Instead, genzaptelconf's hardware detection loads these
modules in the following order:

Module  Size  Used by
xpp88512  0
zaphfc 12956  0
vzaphfc24312  0
firmware_class  9600  0
zaptel184740  3 xpp,zaphfc,vzaphfc

This works. However, if I try to load these modules manually in the  
same order, Asterisk will start without Zaptel support. I don't know  
yet how genzaptelconf accomplishes this, but I suspect that it passes  
certain parameters to the zaptel and/or vzaphfc modules as it loads  
them.

I say that because, after running "genzaptelconf -d", it's possible to  
remove the xpp, zaphfc (if channels 4-5 are not declared) and  
firmware_class modules before starting up Asterisk and still have  
Zaptel support, although all of the Zaptel channels will still be  
busy. Furthermore, it is therefore not my impression that zaphfc is  
interfering with vzaphfc to cause all the zap channels to be busy.

FYI, my current /etc/asterisk/zapata.conf is as follows:

-

[trunkgroups]

[channels]
language=en
context=isdn-in
switchtype=euroisdn
pridialplan=dynamic
prilocaldialplan=local
nationalprefix = 0
internationalprefix = 00
overlapdial=yes
signalling=bri_cpe_ptmp
rxwink=300
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=100
rxgain=4.5
txgain=-3
callgroup=1
pickupgroup=1
immediate=yes
group=1
switchtype = euroisdn
signalling = bri_cpe_ptmp
channel => 1-2
channel => 4-5

-

More information can be found in my previous posts in this thread.

By the way, I've now duplicated my results on a new system with a
different motherboard, a new HFC-S card and a fresh Debian etch
install, etc., but unfortunately the results were exactly the same:
all zap channels busy as soon as Asterisk starts.

If anybody has a working Asterisk v1.4 configuration for ISDN-BRI
using an HFC-S card and Zaptel software, I'd love to see it.

Thanks,

Jaap


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Re: [asterisk-users] GSM Gateway behind SIP ATA?

[Steven]

I am using freePBX, so my dialplan uses macros and such, but here is what I do.

exten => 91248640ABCD,1,Goto(outrt-006-CellGateway,394${EXTEN},1)
;I have a list of all of our company's cell phone numbers. (We get free Cell to 
Cell)

[outrt-006-CellGateway]
include => outrt-006-CellGateway-custom
exten => _3949.,1,Macro(dialout-trunk,12,${EXTEN:4},,)
exten => _3949.,n,Macro(dialout-trunk,11,${EXTEN:4},,)
exten => _3949.,n,Macro(dialout-trunk,1,${EXTEN:4},,)
exten => _3949.,n,Macro(outisbusy,)
; end of [outrt-006-CellGateway]

;I have a two port SIP-GSM Gateway.
;Trunk 12 is port2
:Trunk 11 is port1
;Trunk 1 is my PRI, in case the other two port are busy.


-- 
-- 
Steven

http://www.connectech.org/



"Remco Barendse" <[EMAIL PROTECTED]> wrote in message news:[EMAIL PROTECTED]
>I have an analog GSM Gateway that is connected to a normal SIP ATA device.
>
> Basically what it does is this : when you call the extension nr. of the
> SIP ATA port, the GSM Gateway will pick up the phone and presents a (new)
> dial tone, and then dials whichever DTMF tones it received. The SIP ATA ia
> a Grandstream HT286.
>
> I would like to use the GSM Gateway to route my outbound cellular calls,
> how do i do this in Asterisk? Basically Asterisk should dial the extension
> number and then send required number as DTMF tones to the Gateway through
> the ATA.
>
> I am using FreePBX, which allows me to create a custom trunk for the
> outgoing calls. Hope this could work :)
>
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Re: [asterisk-users] Missing "zap" command in Asterisk 1.4.16

[Steven]

Maybe it was all compiled out of order.
I believe that Zaptel has to be compiled AND installed before compiling 
asterisk to compile the zap channel.

-- 
-- 
Steven

http://www.connectech.org/



  "Raúl Gómez C." <[EMAIL PROTECTED]> wrote in message news:[EMAIL PROTECTED]
  Well, problem solved, I've just recompiled Asterisk and now the "zap" command 
is working fine. Again, thank you all for your responses...

  -- 
  Raul
  Linux Counter #156439 


--


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[asterisk-users] Strange migration problems from asterisk 1.2.13 to 1.4.10, dtmf related?

[Len]

Hello,

I have the following problem. I am migrating my asterisk infrastructure
to a new server and I encounter a strange problem. The configuration is
as followin: IAX clients connect to asterisk which forward calls to a
sip box connected to a phone line. On the old server everything works ok
but on the new server, even if the logs are identical it seems like the
dtmf number does not get passed correctly to the sip box as the phone
does not dial the proper number. The log shows something similar to:

[Jan  7 13:33:11] VERBOSE[7785] logger.c: -- Called 1002
[Jan  7 13:33:11] VERBOSE[7785] logger.c: -- SIP/1002-081b4a80
answered IAX2/ioper00-1
[Jan  7 13:33:11] VERBOSE[7785] logger.c: -- Sending DTMF
'w0214108658' to the called party.

where 1002 is the sip box

[1002]
type=friend
[EMAIL PROTECTED]
callerid="1002"
secret=xxx
host=dynamic
dtmfmode=inband
deny=0.0.0.0/0.0.0.0
permit=10.0.0.121/255.255.255.255

The only problem I can think of is dtmf related. Did something change
from asterisk 1.2.13 to 1.4.10 which could cause this problem? Can it be
related to the computer speed (very unlikely in my mind).

Thank you very much for any ideeas as I am bumping my head for a hole
day trying various combination.

Best regards,
Len
http://www.len.ro


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Re: [asterisk-users] no outgoing calls with Digium B410P

[map]

Hi Daniele,

Could you please tell us what exactly happens?
Are your able to see some error in the log/console?


On Jan 7, 2008 11:53 AM, daniele visaggio <[EMAIL PROTECTED]>
wrote:

> Hi all!
>
> Sorry for my poor english, i'm italian.
>
> I installed Digium B410P on my asterisk server. I followed the official
> installation instructions found on digium site. These instruction, in my
> opinion, are not clear, so i tried with other ways (found on the trixbox
> site). I found this:
> http://www.trixbox.org/forums/trixbox-forums/trunks/howto-install-script-digium-b410p
>
> Now i can receive calls from pstn, but i can't do any call to pstn.
>
> I attach my  /etc/misdn-init.conf/ and also my etc/asterisk/misdn.conf
> file for clarity.
>
> Thanks - Daniele
>
>
>
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