[asterisk-users] Combined patch fixing queue-state and bug12127 for 1.4.x

[Rajkumar S]

Hi,

I am using asterisk-1.4.15,  and using AddQueueMember to add SIP
interface to the queue. Each sip interface is member of multiple
queues

The queue does not recognize that an agent is busy and keeps trying to
call the busy agent. I have identified two patches that can fix the
problem, one at
http://www.scopserv.com/download/asterisk-1.4.17-state_interface.diff
in thread 
http://lists.digium.com/pipermail/asterisk-dev/2008-January/031579.html
and bug no 12127.

I have applied both of them in a test system  running  1.4.17 and it
looks okay. But one of the hunk in 12127 had failed and I had to
manually edit the code. I am not sure if what I have done it correct
or not. If some one has a combined patch that addresses both this
issues for 1.4.x series that would be great!

Thanks a lot for your help and time,

regards,

raj

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Re: [asterisk-users] BRI hardware supported by 1.6 libpri ?

[Olivier]

2008/4/2, Tzafrir Cohen <[EMAIL PROTECTED]>:
>
> On Wed, Apr 02, 2008 at 11:17:05PM +0200, Olivier wrote:
> > Hi,
> >
> > Has anyone information about BRI hardware supported by 1.6 libpri ?
> > In another thread, I was told a basic BRI card with HFC chipset (Bewan
> Gazel
> > 128) was supported but I would delighted to lear about other  harwarde
> (and
> > specifically about Digium B410P).
>
>
> The question should be: "which BRI devices are supported with Zaptel?"
>
> chan_zap of Asterisk 1.6 (with the help of improvements in libpri 1.6)
> supports a much larger subset of the ISDN features that are unique to
> BRI (and has generally better ISDN support.
>
> Even with the version in 1.2 / 1.4 it only took a trivial patch to
> chan_zap (the count of the channels) to get a very basic and limited
> support of ISDN/BRI - ISDN is, after all, ISDN.
>
>
> Anyway, HFC-S cards (many single-port PCI ISDN cards) are supported with
> either the module zaphfc in bristuff or vzaphfc which is not even in
> bristuff.
>
> The Junghanns Duo/Quad/Octo BRI cards and a number of compatible cards
> are supported in the qozap driver from bristuff.
>
> The the driver for BRI module from our (Xorcom) Astribank is included in
> the Zaptel tarball, but requires an extra small patch from bristuff
> applied to Zaptel to build.
>
>
> AFAIK there is no Zaptel driver for the Digium B410P BRI card.


Thank you very much for your detailed answer  : it clearly explains
something I read weeks ago and kept turning in my head since (as I didn't
dare to ask at the moment).

So basically, from now (1.6) on :
1. analog and BRI/PRI cards are supported through chan_zap (with the help of
libpri and zaptel),
2. any BRI/PRI card having a zaptel (1.6) compliant driver can be used by
Asterisk,
3. bristuff is mostly needed to provided some of these zaptel compliant
drivers (zaphfc and qozap) and is still needed to patch chan_zap for
Astribank.

I'll ask Digium about Zaptel driver for the Digium B410P BRI card (and
report back to list).

Thanks again.


--
>Tzafrir Cohen
> icq#16849755  jabber:[EMAIL PROTECTED]<[EMAIL PROTECTED]>
> +972-50-7952406   mailto:[EMAIL PROTECTED]
> http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
>
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Re: [asterisk-users] Need some input for Quad T1 and channel banks.

[Alex Balashov]

Al lists wrote:

> Bad memories from AudioCodec :)

Por que?  I'm curious.


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

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Re: [asterisk-users] IVR Asterisk Voice Recognition -Asterisk withlumenvox

[Dean Collins]

Ok, I think that original request was answered more than a few days ago
by people who had actually used Lumenvox.

Re-reading your answer it appears you provided advice about a topic
where you hadn't actually utilized the product.

As several people who answered Phillip's original question - Lumenvox is
a good product and at an affordable price point.

Just understand it is utterance recognition with limited response set
recognition - NOT NLVR.

 

Regards,

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED] 
+1-212-203-4357
+61-2-9016-5642 (Sydney in-dial). 


> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Al Baker
> Sent: Thursday, 3 April 2008 12:14 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] IVR Asterisk Voice Recognition -Asterisk
> withlumenvox
> 
> Thank you.
> I think that part of one's research is to ask in a user group what
> peoples "real word" experience of a product has been. Particularly
since
> I was responding to a post by Philip in which he had said
>   "So what is that you'd like to know?
>Philipp"
> 
> in response to  a question of
> 
>"I would like to know from someone uses or has used the engines of
>LumenVox for integration with the asterisk for voice recognition."
> 
> posted by another member of the mailing list.
> 
> Sorry you don't agree.
> 
> However, I would still be interested in hearing anyone experiences
that would care
> to share them.
> 
> And to those kind enough to share I say 'thanks' in advance.
> 
> g
> Dean Collins wrote:
> > Hi Al,
> >
> > I'm saying this politely so don't take it the wrong way. Go away and
do
> > some research.
> >
> > Learn the difference between NLVR (Dragon Dictate) and limited set
> > utterance recognition (Lumenvox).
> >
> > Lumenvox is a great product for the price point and as their
developer
> > kit is so reasonable you should buy one to have a play on a test
bench
> > with as a basic minimum.
> >
> >
> >
> > Regards,
> >
> > Dean Collins
> > Cognation Pty Ltd
> > [EMAIL PROTECTED]
> > +1-212-203-4357
> > +61-2-9016-5642 (Sydney in-dial).
> >
> >
> >
> >> -Original Message-
> >> From: [EMAIL PROTECTED]
[mailto:asterisk-users-
> >> [EMAIL PROTECTED] On Behalf Of Al Baker
> >> Sent: Wednesday, 2 April 2008 11:25 PM
> >> To: Asterisk Users Mailing List - Non-Commercial Discussion
> >> Subject: [asterisk-users] IVR Asterisk Voice Recognition - Asterisk
> >>
> > withlumenvox
> >
> >> In the real world, just how good of recognition can you get based
on
> >> your experience ?
> >>
> >> How much processing power do you find it takes ? I know that a
> >>
> > dedicated
> >
> >> Voice Recognition for the
> >> PC such as "Dragon Naturally Speaking" requires :
> >> -a pretty beefy system
> >> -that you use a limited set of microphones
> >> - and ideally, a 20 session to "train" the software.
> >>
> >> Clearly all of this not feasible in a IVR environment, so, in the
> >> absence of all this, just how good , and how sophisticated of a
voice
> >> recognition can one achieve ?
> >>
> >> Philipp von Klitzing wrote:
> >>
> >>> Hi!
> >>>
> >>>
> >>>
>  I would like to know from someone uses or has used the engines of
>  LumenVox for integration with the asterisk for voice recognition.
> 
> 
> >>> So what is that you'd like to know?
> >>>
> >>> Philipp
> >>>
> >>>
> >>> ___
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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
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> >>
> >
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> > asterisk-users mailing list
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> >
> >
> >
> 
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Re: [asterisk-users] IVR Asterisk Voice Recognition - Asterisk withlumenvox

[Al Baker]

Thank you.
I think that part of one's research is to ask in a user group what 
peoples "real word" experience of a product has been. Particularly since 
I was responding to a post by Philip in which he had said
  "So what is that you'd like to know?
   Philipp"

in response to  a question of 
  
   "I would like to know from someone uses or has used the engines of
   LumenVox for integration with the asterisk for voice recognition."

posted by another member of the mailing list.

Sorry you don't agree.

However, I would still be interested in hearing anyone experiences that would 
care to share them.

And to those kind enough to share I say 'thanks' in advance.

g
Dean Collins wrote:
> Hi Al,
>
> I'm saying this politely so don't take it the wrong way. Go away and do
> some research.
>
> Learn the difference between NLVR (Dragon Dictate) and limited set
> utterance recognition (Lumenvox).
>
> Lumenvox is a great product for the price point and as their developer
> kit is so reasonable you should buy one to have a play on a test bench
> with as a basic minimum.
>
>  
>
> Regards,
>
> Dean Collins
> Cognation Pty Ltd
> [EMAIL PROTECTED] 
> +1-212-203-4357
> +61-2-9016-5642 (Sydney in-dial). 
>
>
>   
>> -Original Message-
>> From: [EMAIL PROTECTED] [mailto:asterisk-users-
>> [EMAIL PROTECTED] On Behalf Of Al Baker
>> Sent: Wednesday, 2 April 2008 11:25 PM
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: [asterisk-users] IVR Asterisk Voice Recognition - Asterisk
>> 
> withlumenvox
>   
>> In the real world, just how good of recognition can you get based on
>> your experience ?
>>
>> How much processing power do you find it takes ? I know that a
>> 
> dedicated
>   
>> Voice Recognition for the
>> PC such as "Dragon Naturally Speaking" requires :
>> -a pretty beefy system
>> -that you use a limited set of microphones
>> - and ideally, a 20 session to "train" the software.
>>
>> Clearly all of this not feasible in a IVR environment, so, in the
>> absence of all this, just how good , and how sophisticated of a voice
>> recognition can one achieve ?
>>
>> Philipp von Klitzing wrote:
>> 
>>> Hi!
>>>
>>>
>>>   
 I would like to know from someone uses or has used the engines of
 LumenVox for integration with the asterisk for voice recognition.

 
>>> So what is that you'd like to know?
>>>
>>> Philipp
>>>
>>>
>>> ___
>>> -- 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
>> 
>
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>
>   

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Re: [asterisk-users] IVR Asterisk Voice Recognition - Asterisk withlumenvox

[Dean Collins]

And it's not going to happen now so give up dreaming about it (I have)
but search www.voip-info.org for TellMe and see what would be the
pinnacle for speech recognition for the Asterisk community.

 

Regards,

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED] 
+1-212-203-4357
+61-2-9016-5642 (Sydney in-dial). 


> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Erik Anderson
> Sent: Wednesday, 2 April 2008 11:33 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] IVR Asterisk Voice Recognition -
Asterisk
> withlumenvox
> 
> On Wed, Apr 2, 2008 at 10:24 PM, Al Baker <[EMAIL PROTECTED]>
wrote:
> >  Clearly all of this not feasible in a IVR environment, so, in the
> >  absence of all this, just how good , and how sophisticated of a
voice
> >  recognition can one achieve ?
> 
> Have you ever called Google 411?
> 
> 1-800-GOOG-411
> 
> It'll blow your mind ;-)
> 
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Re: [asterisk-users] IVR Asterisk Voice Recognition - Asterisk with lumenvox

[Erik Anderson]

On Wed, Apr 2, 2008 at 10:24 PM, Al Baker <[EMAIL PROTECTED]> wrote:
>  Clearly all of this not feasible in a IVR environment, so, in the
>  absence of all this, just how good , and how sophisticated of a voice
>  recognition can one achieve ?

Have you ever called Google 411?

1-800-GOOG-411

It'll blow your mind ;-)

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Re: [asterisk-users] IVR Asterisk Voice Recognition - Asterisk withlumenvox

[Dean Collins]

Hi Al,

I'm saying this politely so don't take it the wrong way. Go away and do
some research.

Learn the difference between NLVR (Dragon Dictate) and limited set
utterance recognition (Lumenvox).

Lumenvox is a great product for the price point and as their developer
kit is so reasonable you should buy one to have a play on a test bench
with as a basic minimum.

 

Regards,

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED] 
+1-212-203-4357
+61-2-9016-5642 (Sydney in-dial). 


> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Al Baker
> Sent: Wednesday, 2 April 2008 11:25 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] IVR Asterisk Voice Recognition - Asterisk
withlumenvox
> 
> In the real world, just how good of recognition can you get based on
> your experience ?
> 
> How much processing power do you find it takes ? I know that a
dedicated
> Voice Recognition for the
> PC such as "Dragon Naturally Speaking" requires :
> -a pretty beefy system
> -that you use a limited set of microphones
> - and ideally, a 20 session to "train" the software.
> 
> Clearly all of this not feasible in a IVR environment, so, in the
> absence of all this, just how good , and how sophisticated of a voice
> recognition can one achieve ?
> 
> Philipp von Klitzing wrote:
> > Hi!
> >
> >
> >> I would like to know from someone uses or has used the engines of
> >> LumenVox for integration with the asterisk for voice recognition.
> >>
> >
> > So what is that you'd like to know?
> >
> > Philipp
> >
> >
> > ___
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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] IVR Asterisk Voice Recognition - Asterisk with lumenvox

[Al Baker]

In the real world, just how good of recognition can you get based on 
your experience ?

How much processing power do you find it takes ? I know that a dedicated 
Voice Recognition for the
PC such as "Dragon Naturally Speaking" requires :
-a pretty beefy system
-that you use a limited set of microphones
- and ideally, a 20 session to "train" the software.

Clearly all of this not feasible in a IVR environment, so, in the 
absence of all this, just how good , and how sophisticated of a voice 
recognition can one achieve ?

Philipp von Klitzing wrote:
> Hi!
>
>   
>> I would like to know from someone uses or has used the engines of
>> LumenVox for integration with the asterisk for voice recognition. 
>> 
>
> So what is that you'd like to know?
>
> Philipp
>
>
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>
>   

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Re: [asterisk-users] zaptel alarm

[Greg Woods]

Shaun Ruffell wrote:
>
> If you have subversion installed on your server, could you try using 
> this version of zaptel:
>
> http://svn.digium.com/svn/zaptel/branches/[EMAIL PROTECTED]
>   

Thank you to everyone who offered assistance. For now, upgrading the 
driver has fixed the problem: outbound calls work again. It does leave 
open the mystery of why it worked fine for a week and suddenly quit, so 
I can't be 100% certain that everything is really fixed until it has 
been stable for a while.

--Greg



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Re: [asterisk-users] Need some input for Quad T1 and channel banks.

[Al lists]

Bad memories from AudioCodec :)


On Wed, Apr 2, 2008 at 7:48 PM, Edwin Lam <[EMAIL PROTECTED]>
wrote:

> Andrew Latham wrote:
> > Here I will say it http://xorcom.com
>
> alternatively:
> http://www.audiocodes.com/objects/30010_DS_MP-11X,%20MP-124D.pdf
>
> > On Mon, Mar 31, 2008 at 6:01 PM, Al lists <[EMAIL PROTECTED]> wrote:
> >> I'm looking to install a system with 80 FXS analog phones.
> >> At this time the only cost effective solution is using a 4 port T1 card
> and
> >> addit 600 channel bank.
> >> Has anyone tried this solution? any good documents beside
> >>
> http://www.voip-info.org/wiki/index.php?page=Asterisk+hardware+channel+bank+check
> >>  as far as i know, addit 600 T1 interface is not PRI (please correct me
> if
> >> i'm wrong) its CAS robbed bit, will that work with new Digium quad T1
> like
> >> TE410P ?( I prefer to use Digium if possible)
> >> The system is connected to the Telco through SIP trunk so all we have
> in
> >> terms of analog is local loop, Do we need to have echo cancel in this
> >> scenario ?
>
>
> --
> Edwin Lam <[EMAIL PROTECTED]>
> Systems Engineer, Office General, Inc.
> Ph: +1 415 439 4988 Fax: +1 415 283 3370
> http://pgpkeys.mit.edu:11371/pks/lookup?op=get&search=0xD6506D20
>
>
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Re: [asterisk-users] Need some input for Quad T1 and channel banks.

[Edwin Lam]

Andrew Latham wrote:
> Here I will say it http://xorcom.com

alternatively:
http://www.audiocodes.com/objects/30010_DS_MP-11X,%20MP-124D.pdf

> On Mon, Mar 31, 2008 at 6:01 PM, Al lists <[EMAIL PROTECTED]> wrote:
>> I'm looking to install a system with 80 FXS analog phones.
>> At this time the only cost effective solution is using a 4 port T1 card and
>> addit 600 channel bank.
>> Has anyone tried this solution? any good documents beside
>> http://www.voip-info.org/wiki/index.php?page=Asterisk+hardware+channel+bank+check
>>  as far as i know, addit 600 T1 interface is not PRI (please correct me if
>> i'm wrong) its CAS robbed bit, will that work with new Digium quad T1 like
>> TE410P ?( I prefer to use Digium if possible)
>> The system is connected to the Telco through SIP trunk so all we have in
>> terms of analog is local loop, Do we need to have echo cancel in this
>> scenario ?


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


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Re: [asterisk-users] Need some input for Quad T1 and channel banks.

[Alex Balashov]

Yes, the Digium cards should support T1 CAS.

Andrew Latham wrote:
> Here I will say it http://xorcom.com
> 
> 
> 
> On Mon, Mar 31, 2008 at 6:01 PM, Al lists <[EMAIL PROTECTED]> wrote:
>> I'm looking to install a system with 80 FXS analog phones.
>> At this time the only cost effective solution is using a 4 port T1 card and
>> addit 600 channel bank.
>> Has anyone tried this solution? any good documents beside
>> http://www.voip-info.org/wiki/index.php?page=Asterisk+hardware+channel+bank+check
>>  as far as i know, addit 600 T1 interface is not PRI (please correct me if
>> i'm wrong) its CAS robbed bit, will that work with new Digium quad T1 like
>> TE410P ?( I prefer to use Digium if possible)
>> The system is connected to the Telco through SIP trunk so all we have in
>> terms of analog is local loop, Do we need to have echo cancel in this
>> scenario ?
>>  Thanks!
>>
>>
>>
>> ___
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>>
>>  asterisk-users mailing list
>>  To UNSUBSCRIBE or update options visit:
>>http://lists.digium.com/mailman/listinfo/asterisk-users
>>
> 
> 
> 


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

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[asterisk-users] problem with Kewlstart hangup detection

[Matt Watson]

Hello all,



I;m having a (what seems to me) strange problem with some analog lines and 
hangup detection.



The site I;m working on has 10 analog lines, my understanding is these are 
broken up in 2 invidiaul hunt groups (no idea why, or if this is even true).



I;ve always been told that they are  yyy-xxx-7200 and yyy-xxx-2115 - no idea 
which other lines are part of which hunt group.



Anyways... I have these lines plugged into a Digium TDM800P and a TDM400P



I have all the lines set to kewlstart signalling in zaptel.conf and 
zapata.conf, the lines are all provided by MTS Allstream (in Canada).



It appears to me that the disconnect supervision on all of these lines *except* 
those 2 (7200 and 2115) send the disconnect notice very delayed.  Calling in 
from an outside line to 7200 or 2115, i hangup on the remote site, and 
Zaptel/Ast detects the hangup almost instantly (and having debug turned on on 
the wctdm & wctdm24xxp drivers shows the NO BATTERY / BATTERY messages almost 
instantly like it should).



However... on the remaining 8 lines, it takes approximently 10 seconds before i 
see the NO BATTERY / BATTERY message.  It would seem to me that for whatever 
reason this message is delayed on the telco side.



I;m not very familiar with hunt groups on analog lines... and I can;t really 
see how this is possible... but do I need to do some kind of special 
configuration on my end to make the non-pilot numbers of the hunt group get 
their disconnect notice quicker?  Is the telco at fault here?  Is there 
something I can ask them specifically that will make sense to one of their 
lower end CSRs?



Its really not a *huge* problem... except our receptionist is getting quite 
annoyed with several callers that hung up sometime during our IVR (her phone 
rings as the timeout off the IVR). Also people are getting VMs that aren;t 
really being left and stuff.



I first thought that perhaps for some reason we only had kewlstart on the 2 
lines... but like I said... I do infact see the NO BATTERY /BATTERY on all 
lines... just on all of them but the 2 its very delayed.



Thanks,



--

Matt
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Re: [asterisk-users] Need some input for Quad T1 and channel banks.

[Al lists]

Its Nice, i agree, but we are looking at $4k to $5k with this.


On Wed, Apr 2, 2008 at 1:17 PM, Andrew Latham <[EMAIL PROTECTED]> wrote:

> Here I will say it http://xorcom.com
>
>
>
> On Mon, Mar 31, 2008 at 6:01 PM, Al lists <[EMAIL PROTECTED]> wrote:
> > I'm looking to install a system with 80 FXS analog phones.
> > At this time the only cost effective solution is using a 4 port T1 card
> and
> > addit 600 channel bank.
> > Has anyone tried this solution? any good documents beside
> >
> http://www.voip-info.org/wiki/index.php?page=Asterisk+hardware+channel+bank+check
> >  as far as i know, addit 600 T1 interface is not PRI (please correct me
> if
> > i'm wrong) its CAS robbed bit, will that work with new Digium quad T1
> like
> > TE410P ?( I prefer to use Digium if possible)
> > The system is connected to the Telco through SIP trunk so all we have in
> > terms of analog is local loop, Do we need to have echo cancel in this
> > scenario ?
> >  Thanks!
> >
> >
> >
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> --
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>  [EMAIL PROTECTED]
>  [EMAIL PROTECTED]
>
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>  http://www.TuxTone.com
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Re: [asterisk-users] interrupting MOH

[Brent Davidson]

Well, not necessarily.  You could have one MOH stream that gets 
interrupted say every 10 seconds with a generic message.  A caller that 
gets connected to the MOH stream might come in in the middle of the 
message, during music playback, or anywhere form 0-10 seconds before the 
message plays again.


If this is being used strictly for transferring calls, though, it might 
be pretty confusing to have "Please wait while we transfer you call." 
play every few seconds.  The caller will probably think the system is in 
some sort of loop, or that they are being routed through several 
different systems.  A less confusing setup might be to simply play the 
"Please hold while your call is transferred" message just before 
starting their MOH.


I have recently implemented Asterisk at several offices and, unknowingly 
set myself up for a huge number of complaints.  I set up our IVR to 
default to send callers to an operator queue if they did not dial an 
extension or the DTMF was not recognized.  Our main greeting lists the 
valid extensions and other info then says "Or stay on the line and an 
operator will be with you shortly."  I assumed (yeah I know ass-u-me) 
that the instructions in the greeting would be sufficient so I just had 
the system drop directly to the operator queue with MOH.  Big mistake.  
I had complaints from all three offices that customers thought the 
system was disconnecting them or dropping them into some sort of "black 
hole" when the MOH kicked in.  I changed the MOH to ringing and the 
majority of the complains stopped.  (The remaining complaints are 
related to DTMF detection problems.


Just food for thought.

Good luck,
Brent


Atis Lezdins wrote:

On Wed, Apr 2, 2008 at 11:05 PM, Brent Davidson
<[EMAIL PROTECTED]> wrote:
  

 You could also, conceivably, handle this outside of asterisk by using a
more complex MOH stream source.  For instance, use a shoutcast client as the
MOH source, run your own shoutcast server streaming your music and have a
script set up to periodically interrupt the stream being served to the
shoutcast server and inject an announcement.  (Keep in mind that this is an
"off the top of my head" suggestion so I don't have exact details for
implementation, but I'm sure it can be done.)



That would need one shoutcast stream per call.. not very reasonable..

Regards,
Atis

  

 Good luck,
 Brent



 Matt Florell wrote:
 Hello,

We achieve this using an AGI script in the VICIDIAL project for our
version of inbound queues. You start MoH then when you stream a sound
to the channel it will stop MoH then after the sound is done you start
MoH back up again. Probably a bit more involved than what you want,
but it dose work well for us.

MATT---

On 4/2/08, Atis Lezdins <[EMAIL PROTECTED]> wrote:


 Sorry for top-posting, but seems everyone on this thread did so.

 Also that would be my suggestion for now - call queue with
periodic-announce.

 However i see that this would make nice architectural improvement -
 allow inject sound files into MoH stream. This would be useful for
 example in call queues - to inject all the queue announcements into
 MoH directly, rather than play them while blocking further queue
 actions.

 Regards,
 Atis



 On Wed, Apr 2, 2008 at 4:11 AM, Andreas van dem Helge
 <[EMAIL PROTECTED]> wrote:
 > I think that's still a better idea than using a "dump the caller into
 > meetme" hack and is actually what I was going to suggest.
 >
 > If you want something simpler than a queue then inject the sounds into
 > the moh already.
 >
 >
 >
 > On Tue, Apr 1, 2008 at 3:09 PM, Rob Hillis <[EMAIL PROTECTED]> wrote:
 > >
 > > You may be able to achieve the desired result using queues rather than
 > > Dial statements.
 > >
 > > Overkill perhaps, but it's the only way I can think to implement it at
the
 > > moment.
 > >
 > >
 > >
 > >
 > > John Millican wrote:
 > > Tilghman Lesher wrote:
 > >
 > >
 > > On Tuesday 01 April 2008 05:14:25 Pete Kay wrote:
 > >
 > >
 > > I am hoping someone can help me out on this. I want to be able to
 > > interrupt MOH every X seconds after the DIAL command is executed. The
 > > interrupt greeting is something like "please wait while we transfer
your
 > > call". How can I do that? Within the DIAL options, I can't see any
 > > announce frequency or options that can help.
 > >
 > > Could anyone please tell me how that function can be accomplished?
 > >
 > > The only way to do that currently is to implement the prompt within the
MOH
 > > stream itself.
 > >
 > >
 > >
 > > Just off the top-o-my head(YMMV), couldn't you create a meetme and play
 > > hold music into the meetme and then also play the prompt into the
meetme
 > > at the same time without interrupting the hold music? This would
 > > obviously not work for high load but...
 > > JohnM
 > >
 > >
 > >
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Re: [asterisk-users] interrupting MOH

[Atis Lezdins]

On Wed, Apr 2, 2008 at 11:05 PM, Brent Davidson
<[EMAIL PROTECTED]> wrote:
>
>  You could also, conceivably, handle this outside of asterisk by using a
> more complex MOH stream source.  For instance, use a shoutcast client as the
> MOH source, run your own shoutcast server streaming your music and have a
> script set up to periodically interrupt the stream being served to the
> shoutcast server and inject an announcement.  (Keep in mind that this is an
> "off the top of my head" suggestion so I don't have exact details for
> implementation, but I'm sure it can be done.)

That would need one shoutcast stream per call.. not very reasonable..

Regards,
Atis

>
>  Good luck,
>  Brent
>
>
>
>  Matt Florell wrote:
>  Hello,
>
> We achieve this using an AGI script in the VICIDIAL project for our
> version of inbound queues. You start MoH then when you stream a sound
> to the channel it will stop MoH then after the sound is done you start
> MoH back up again. Probably a bit more involved than what you want,
> but it dose work well for us.
>
> MATT---
>
> On 4/2/08, Atis Lezdins <[EMAIL PROTECTED]> wrote:
>
>
>  Sorry for top-posting, but seems everyone on this thread did so.
>
>  Also that would be my suggestion for now - call queue with
> periodic-announce.
>
>  However i see that this would make nice architectural improvement -
>  allow inject sound files into MoH stream. This would be useful for
>  example in call queues - to inject all the queue announcements into
>  MoH directly, rather than play them while blocking further queue
>  actions.
>
>  Regards,
>  Atis
>
>
>
>  On Wed, Apr 2, 2008 at 4:11 AM, Andreas van dem Helge
>  <[EMAIL PROTECTED]> wrote:
>  > I think that's still a better idea than using a "dump the caller into
>  > meetme" hack and is actually what I was going to suggest.
>  >
>  > If you want something simpler than a queue then inject the sounds into
>  > the moh already.
>  >
>  >
>  >
>  > On Tue, Apr 1, 2008 at 3:09 PM, Rob Hillis <[EMAIL PROTECTED]> wrote:
>  > >
>  > > You may be able to achieve the desired result using queues rather than
>  > > Dial statements.
>  > >
>  > > Overkill perhaps, but it's the only way I can think to implement it at
> the
>  > > moment.
>  > >
>  > >
>  > >
>  > >
>  > > John Millican wrote:
>  > > Tilghman Lesher wrote:
>  > >
>  > >
>  > > On Tuesday 01 April 2008 05:14:25 Pete Kay wrote:
>  > >
>  > >
>  > > I am hoping someone can help me out on this. I want to be able to
>  > > interrupt MOH every X seconds after the DIAL command is executed. The
>  > > interrupt greeting is something like "please wait while we transfer
> your
>  > > call". How can I do that? Within the DIAL options, I can't see any
>  > > announce frequency or options that can help.
>  > >
>  > > Could anyone please tell me how that function can be accomplished?
>  > >
>  > > The only way to do that currently is to implement the prompt within the
> MOH
>  > > stream itself.
>  > >
>  > >
>  > >
>  > > Just off the top-o-my head(YMMV), couldn't you create a meetme and play
>  > > hold music into the meetme and then also play the prompt into the
> meetme
>  > > at the same time without interrupting the hold music? This would
>  > > obviously not work for high load but...
>  > > JohnM
>  > >
>  > >
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>  >
>
>
>
>
> --
>  Atis Lezdins,
>  VoIP Project Manager / Developer,
>  [EMAIL PROTECTED]
>  Skype: atis.lezdins
>  Cell Phone: +371 28806004
>  Cell Phone: +1 800 7300689
>  Work phone: +1 800 7502835
>
>
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Re: [asterisk-users] BRI hardware supported by 1.6 libpri ?

[Tzafrir Cohen]

On Wed, Apr 02, 2008 at 11:17:05PM +0200, Olivier wrote:
> Hi,
> 
> Has anyone information about BRI hardware supported by 1.6 libpri ?
> In another thread, I was told a basic BRI card with HFC chipset (Bewan Gazel
> 128) was supported but I would delighted to lear about other  harwarde (and
> specifically about Digium B410P).

The question should be: "which BRI devices are supported with Zaptel?"

chan_zap of Asterisk 1.6 (with the help of improvements in libpri 1.6)
supports a much larger subset of the ISDN features that are unique to
BRI (and has generally better ISDN support.

Even with the version in 1.2 / 1.4 it only took a trivial patch to
chan_zap (the count of the channels) to get a very basic and limited
support of ISDN/BRI - ISDN is, after all, ISDN.


Anyway, HFC-S cards (many single-port PCI ISDN cards) are supported with
either the module zaphfc in bristuff or vzaphfc which is not even in
bristuff.

The Junghanns Duo/Quad/Octo BRI cards and a number of compatible cards
are supported in the qozap driver from bristuff.

The the driver for BRI module from our (Xorcom) Astribank is included in
the Zaptel tarball, but requires an extra small patch from bristuff
applied to Zaptel to build.


AFAIK there is no Zaptel driver for the Digium B410P BRI card.

-- 
   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] Digium B410P, bristuff and BRI support in 1.6

[Jean-Denis Girard]

Olivier a écrit :
> 
> 2008/4/2, Jean-Denis Girard <[EMAIL PROTECTED] 
> >:
> 
> 
> Development version of libpri (libpri-trunk) does include prliminary
> support for BRI.
> 
> 
> I took a look at :
> http://svn.digium.com/view/libpri/trunk/
> 
> Though BRI support is mentioned several times but I couldn't find any 
> supported hardware list.
> I'll open a new thread on this last and specific point.
> 
> One last question Jean-Denis, when you wrote your "system has been
> running fine for nearly a month with libpri-trunk, asterisk-1.6.0,
> zaptel-1.4 (patched) and zaphfc", does zaptel-1.4 (patched) also relates 
> to BRI support ?
> I thought that (without bristuff) zaptel was dedicated to analog lines.
> Did you have to install zaptel for BRI support ?

Zaptel also has always supported digital (PRI) cards from Digium.

Though I don't have experience with this setting and other BRI cards at 
the moment, I believe B410P is supported, as well as other cards 
supported by bristuff.


Regards,
-- 
Jean-Denis Girard

SysNux  Systèmes  Linux  en Polynésie française
http://www.sysnux.pf/   Tél: +689 50 10 40 / GSM: +689 79 75 27

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[asterisk-users] BRI hardware supported by 1.6 libpri ?

[Olivier]

Hi,

Has anyone information about BRI hardware supported by 1.6 libpri ?
In another thread, I was told a basic BRI card with HFC chipset (Bewan Gazel
128) was supported but I would delighted to lear about other  harwarde (and
specifically about Digium B410P).

Regards
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Re: [asterisk-users] TE205P

[Jared Smith]

On Wed, 2008-04-02 at 13:26 -0700, Isaac McDonald wrote:
> I am working on a project and have a few questions. I want to connect
> one port of the TE205p to the PSTN, and another port to the PRI port
> of a PBX. Basically Asterisk will sit between the PSTN and the
> existing PBX. 

This will work just fine.

> Are there any gotcha's I need to be aware of? Will the
> existing PBX be able to dial through asterisk as if it was a PSTN
> connection?

Yes, as long as you configure Asterisk correctly.

-- 
Jared Smith
Community Relations Manager
Digium, Inc.


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Re: [asterisk-users] zaptel alarm

[Greg Woods]

On Wed, 2008-04-02 at 15:23 -0500, Brent Davidson wrote:
>  the cords that ran between the wall jack and the jacks on the X100P
> cords all ran between the server's 21" CRT monitor and the wall. 

Not a problem here, as the monitor is on the other side of the room from
the server and the wire from the wall plate to the server doesn't go
anywhere near it.

>  The server was degaussing or something every evening and the
> resulting magnetic surge was burning out the X100P card. 

I think the card is OK; all the extensions plugged into the FXS ports
work fine and incoming calls also work. This looks more like a driver
problem. And I've had trouble with the zaptel drivers before. I had a
situation where my machine would just stop (no crash, just totally hung
and unresponsive). A zaptel driver upgrade fixed that.

>  If you've had a recent lightning storm with cloud to ground lightning
> anywhere in the area 

It's a bit early for that here; we won't see our first thunderstorm for
another few weeks yet.

> There could also be an error in the RPM's though.  I'd recommend
> trying rebuilding the zaptel drivers 

That's what I intend to try tonight. I hope I don't have to rebuild
asterisk from source too, as having the fixes applied to the RPM (such
as the one for the recent security problem) was one of my main reasons
for upgrading the system, so I wouldn't HAVE to maintain asterisk from
source any more.

But I did have to get the zaptel driver from ATrpms, as it isn't part of
Fedora 8 (maybe it's got some license or patent restrictions on it so
they don't distribute it for the same reasons they won't distribute MP3
decoders?) If the update does fix my problem, I will let the ATrpms
people know about it.

--Greg



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[asterisk-users] Asterisk parked calls and callerid

[Guido Hecken]

Hi list,
 
sorry if this has been discussed in the past, but I couldn't find anything
wise about it.
 
Since we had some trouble with the builtin hold function of some (all?) SNOM
320/360
phones, we decided to use the call parking feature in asterisk instead.
 
Assume, a call comes in with CALLERID(num) 1234567 for extension 10.
Extension 10 parks this call into 801, dials extension 11 and asks if she/he
could
fetch the call on parkposition 801.
Extension 11 dials 801 and get's the call and can only see 801 in the phones
display.
 
So, how could we get the original CALLERID(num) 1234567 back in the phones
display?
Using a channel variable or use astdb comes in mind, but what is the best
way to achieve 
this?
In regular pbx systems this seems to be a "standard" function.
Any ideas on this?
 
 
Here some of the involved configs
---
in extensions.conf:
 
[parkedcalls]
exten =>  80[1-5],1,NoCDR()
exten =>  80[1-5],2,ParkedCall(${EXTEN})
 
in features.conf:
 
[general]
parkext => 800
parkpos => 801-805
context => parkedcalls  
parkingtime => 60
 
-

We are using SVN-branch-1.4-r96449 and other, older versions of asterisk

Regards 

Guido

 

 

gwsNetTech
Guido Hecken

Quirrenbacher Str. 36
53639 Königswinter
Germany


fon +49(2244) 870663
fax +49(2244) 870664
mobil  +49(179) 1267353
web   http://www.gwsnettech.de
  mailto:[EMAIL PROTECTED]

 
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Re: [asterisk-users] Analog modem as phone

[Greg Woods]

On Wed, 2008-04-02 at 21:18 +0200, Ronny Forberger wrote:

> I want to use a analog V.92 modem to make outgoing (and possibly)  
> incoming phone call through a standard analog phone line.

When I asked this question, I was basically told that it isn't possible.
The problem is along the lines that the modem uses many more wavelengths
and more bandwidth than a regular phone does, so this won't work through
the card. I have found that I can send outgoing faxes, and incoming
faxes redirected to this modem also work, but I have to patch the modem
through directly to the wall plate in order to be able to make dialup
connections.

--Greg



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[asterisk-users] FW: [newtech-1] Skype 24 Hour Rolling Analytics

[Dean Collins]

While not asterisk specific related - would be interesting to see
something similar done against straight voip calls.
You could maybe plot it against e.164.org polls but I think this would
be too small a sample set.

Can anyone think of something else Asterisk related we could plot this
against?

Cheers,
Dean



 

http://deancollinsblog.blogspot.com/2008/04/skype-24-hour-rolling-analyt
ics.html

 
 

Totally stumbled across this really interesting post 
http://skypejournal.com/blog/2008/04/world_online_or_asleep.html
  

I cant comment on how accurate it is but looks like someone has tried to
do some really interesting research on number of skype users compared to
people awake on a rolling 24 hour basis.

Would be interesting to see some other applications tracked against this
- twitter posts per minute maybe?




Regards,

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED] 
+1-212-203-4357
+61-2-9016-5642 (Sydney in-dial). 

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Re: [asterisk-users] Digium B410P, bristuff and BRI support in 1.6

[Olivier]

2008/4/2, Jean-Denis Girard <[EMAIL PROTECTED]>:
>
>
> Development version of libpri (libpri-trunk) does include prliminary
> support for BRI.


I took a look at :
http://svn.digium.com/view/libpri/trunk/

Though BRI support is mentioned several times but I couldn't find any
supported hardware list.
I'll open a new thread on this last and specific point.

One last question Jean-Denis, when you wrote your "system has been
running fine for nearly a month with libpri-trunk, asterisk-1.6.0,
zaptel-1.4 (patched) and zaphfc", does zaptel-1.4 (patched) also relates to
BRI support ?
I thought that (without bristuff) zaptel was dedicated to analog lines.
Did you have to install zaptel for BRI support ?


Regards
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[asterisk-users] TE205P

[Isaac McDonald]

Hello,

I am working on a project and have a few questions. I want to connect
one port of the TE205p to the PSTN, and another port to the PRI port
of a PBX. Basically Asterisk will sit between the PSTN and the
existing PBX. Are there any gotcha's I need to be aware of? Will the
existing PBX be able to dial through asterisk as if it was a PSTN
connection?

-- 
Isaac McDonald
Got VoIP?
[EMAIL PROTECTED]
Cell: +1 253-223-8673

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Re: [asterisk-users] zaptel alarm

[Brent Davidson]

Greg Woods wrote:

Shaun Ruffell wrote:
  

 >> svn co http://svn.digium.com/svn/zaptel/branches/[EMAIL PROTECTED]
  zaptel-1.4-4122
  



Thank you, I will try that tonight when  I get home and report back.

--Greg


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Do you have access to a phone line simulator?  It could be your TDM card 
has gone bad.  I had an issue with some X100P's repeatedly burning out.  
After a rather lengthy investigation that turned up no solution I 
resorted to trial and error.  Turns out the cords that ran between the 
wall jack and the jacks on the X100P cords all ran between the server's 
21" CRT monitor and the wall.  The server was degaussing or something 
every evening and the resulting magnetic surge was burning out the X100P 
card.  If you've had a recent lightning storm with cloud to ground 
lightning anywhere in the area there is a strong possibility that the 
TDM card has been damaged.


There could also be an error in the RPM's though.  I'd recommend trying 
rebuilding the zaptel drivers and Asterisk from source and seeing if 
that changes anything.


Good luck,
Brent
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Re: [asterisk-users] Problems with DELL 1600

[Al Baker]

Could you be like 1% more specific, with perhaps including any 
relevant Errors, Log Files ?

What makes you think it is the DELL , not your T1 boards, or your 
service provider  or 

Ruben Zamora wrote:
> Hi
>
> I just want to know if anyone have problems with server DELL 1600, 
> Like:  Hangup Call.
>
> Thanks
>
> Ruben
>
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Re: [asterisk-users] Problem when leaving voicemail

[Richard Open Source]

Solved. Firewall problem.

In case someone may run into this issue.

There is a firewall between Asterisk and the database. The firewall kills an
idle TCP connection after an hour. Asterisk and MySQL do not know this. Next
time a call comes in, Asterisk reuses the connection. To make use of the
connection, realtime pings the mysql first to check the connection. However,
the firewall has killed/blocked the session so TCP on Asterisk side does not
get an ACK. TCP then retries to transmit for about 15 minutes.

Then after 2 hours TCP Keep-Alives on Asterisk server and MySQL server kicks
in which does not help.

Solution: Make TCP Keep-Alive kick in after 45 minutes of idle connection.

R

On Thu, Mar 27, 2008 at 5:22 PM, Richard Open Source <[EMAIL PROTECTED]>
wrote:

> Hi,
>
> I am investigating an issue with voicemail and realtime.
>
> What we are seeing is the following:
> 1. Caller calls in and goes to an IVR
> 2. Presses 101 to go to voicemail
> 3. app_voicemail start and tries to connect to the database trhough
> res_config_mysql. However, it takes too long to be able to connect (~15
> minutes)
> It seems like it first attemots to connect to the database on 16:25:03 and
> manage to connect at 16:40:24.
>
> [Mar 26 16:25:03] VERBOSE[19786] logger.c: -- AGI Script agi://
> 127.0.0.1/enswitch?stype=external completed, returning 0
> [Mar 26 16:25:03] DEBUG[19786] pbx.c: Launching 'Answer'
> [Mar 26 16:25:03] VERBOSE[19786] logger.c: -- Executing
> [EMAIL PROTECTED]:1] Answer("SIP/5060-ac017e30", "") in new stack
> [Mar 26 16:25:03] DEBUG[19786] pbx.c: Expression result is '0'
> [Mar 26 16:25:03] DEBUG[19786] pbx.c: Launching 'GotoIf'
> [Mar 26 16:25:03] VERBOSE[19786] logger.c: -- Executing
> [EMAIL PROTECTED]:2] GotoIf("SIP/5060-ac017e30", "0?5") in new stack
> [Mar 26 16:25:03] DEBUG[19786] pbx.c: Not taking any branch
> [Mar 26 16:25:03] DEBUG[19786] pbx.c: Launching 'VoiceMail'
> [Mar 26 16:25:03] VERBOSE[19786] logger.c: -- Executing
> [EMAIL PROTECTED]:3] VoiceMail("SIP/5060-ac017e30", "[EMAIL 
> PROTECTED]|us<[EMAIL PROTECTED]>")
> in new stack
> [Mar 26 16:25:03] DEBUG[19786] app_voicemail.c: Before find_user
> [Mar 26 16:25:03] DEBUG[19786] app_voicemail.c: In find_user_realtime for
> mailbox 101 context 708
>
> [Mar 26 16:40:24] VERBOSE[14269] logger.c: Really destroying SIP dialog
> '[EMAIL PROTECTED]' Method: OPTIONS
> [Mar 26 16:40:28] ERROR[19786] res_config_mysql.c: MySQL RealTime: Ping
> failed (2006).  Trying an explicit reconnect.
> [Mar 26 16:40:28] DEBUG[19786] res_config_mysql.c: MySQL RealTime: Server
> Error (2006): MySQL server has gone away
> [Mar 26 16:40:28] DEBUG[19786] res_config_mysql.c: MySQL RealTime:
> Successfully connected to database.
> [Mar 26 16:40:28] DEBUG[19786] res_config_mysql.c: MySQL RealTime:
> Retrieve SQL: SELECT * FROM mailboxes WHERE mailbox = '101' AND context =
> '708'
> The database and the connection seems to be ok. It is only this query
> that's taking long.
>
> We are experiencing this on Asterisk 1.4.17 and 1.4.18 but difficult to
> reproduce.
>
> Anybody have any idea what might be the cause and how to procceed and
> figure out what's wrong?
>
> Thanks in advance.
>
> R
>
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Re: [asterisk-users] Problems with DELL 1600

[Erik Anderson]

On Wed, Apr 2, 2008 at 9:37 AM, Ruben Zamora <[EMAIL PROTECTED]> wrote:
>
>  I just want to know if anyone have problems with server DELL 1600,
>  Like:  Hangup Call.

Give us some more details of your setup and you'll probably have
better chances of getting an answer.

-Erik

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[asterisk-users] Receiving 404 Not Found on all outbound calls when attempting SIP trunking between * 1.2.22 and Mitel 3300 CX

[Robert Bedell]

We are attempting to configure SIP trunking between asterisk 1.2.22 and a
Mitel 3300 CX box.  The Mitel machine will gateway to the PSTN for us.  I
found this earlier post about doing this from July:

http://lists.digium.com/pipermail/asterisk-users/2007-July/191630.html

Unfortunately the promised configs never came ;(.  We're having the exact
reverse problem: we can register with the Mitel box just fine, and we
receive calls from it correctly.  However all attempts to make outbound
calls from asterisk to the Mitel machine fail with a 404 Not found.

The following is a fragment of our sip.conf (the extensions.conf config is
just a single Dial(SIP/[EMAIL PROTECTED]) entry):

register => 450:[EMAIL PROTECTED]:5060/450

[mitel-outbound]
 type=peer
 secret=1234
 username=450
 host=192.168.100.2
 fromuser=450
 fromdomain=100.110.0.200
 insecure=very
 nat=no
 context=mitel
 port=5060

In the above: username/password/host are 450, 1234, and
192.168.100.2respectively.  (names changed to protect the innocent).
The box is at
100.110.0.200 (again..)

Has anyone else had a similar problem with receiving 404's on outbound calls
to Mitel boxes?  We found the Mitel doesn't respond at all when we change
the host line above, presumably because the From header in the sip changes
to not include the Username correctly.  I can't get any auth headers to come
in the initial INVITE, although with some digging I found out they would go
out if the Mitel had responded with a 401.

Any help is appreciated.

Robert Bedell
SienaTech
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Re: [asterisk-users] interrupting MOH

[Brent Davidson]

You could also, conceivably, handle this outside of asterisk by using a 
more complex MOH stream source.  For instance, use a shoutcast client as 
the MOH source, run your own shoutcast server streaming your music and 
have a script set up to periodically interrupt the stream being served 
to the shoutcast server and inject an announcement.  (Keep in mind that 
this is an "off the top of my head" suggestion so I don't have exact 
details for implementation, but I'm sure it can be done.)


Good luck,
Brent

Matt Florell wrote:

Hello,

We achieve this using an AGI script in the VICIDIAL project for our
version of inbound queues. You start MoH then when you stream a sound
to the channel it will stop MoH then after the sound is done you start
MoH back up again. Probably a bit more involved than what you want,
but it dose work well for us.

MATT---

On 4/2/08, Atis Lezdins <[EMAIL PROTECTED]> wrote:
  

Sorry for top-posting, but seems everyone on this thread did so.

 Also that would be my suggestion for now - call queue with periodic-announce.

 However i see that this would make nice architectural improvement -
 allow inject sound files into MoH stream. This would be useful for
 example in call queues - to inject all the queue announcements into
 MoH directly, rather than play them while blocking further queue
 actions.

 Regards,
 Atis



 On Wed, Apr 2, 2008 at 4:11 AM, Andreas van dem Helge
 <[EMAIL PROTECTED]> wrote:
 > I think that's still a better idea than using a "dump the caller into
 >  meetme" hack and is actually what I was going to suggest.
 >
 >  If you want something simpler than a queue then inject the sounds into
 >  the moh already.
 >
 >
 >
 >  On Tue, Apr 1, 2008 at 3:09 PM, Rob Hillis <[EMAIL PROTECTED]> wrote:
 >  >
 >  >  You may be able to achieve the desired result using  queues rather than
 >  > Dial statements.
 >  >
 >  >  Overkill perhaps, but it's the only way I can think to implement it at 
the
 >  > moment.
 >  >
 >  >
 >  >
 >  >
 >  >  John Millican wrote:
 >  >  Tilghman Lesher wrote:
 >  >
 >  >
 >  >  On Tuesday 01 April 2008 05:14:25 Pete Kay wrote:
 >  >
 >  >
 >  >  I am hoping someone can help me out on this. I want to be able to
 >  > interrupt MOH every X seconds after the DIAL command is executed. The
 >  > interrupt greeting is something like "please wait while we transfer your
 >  > call". How can I do that? Within the DIAL options, I can't see any
 >  > announce frequency or options that can help.
 >  >
 >  > Could anyone please tell me how that function can be accomplished?
 >  >
 >  >  The only way to do that currently is to implement the prompt within the 
MOH
 >  > stream itself.
 >  >
 >  >
 >  >
 >  > Just off the top-o-my head(YMMV), couldn't you create a meetme and play
 >  > hold music into the meetme and then also play the prompt into the meetme
 >  > at the same time without interrupting the hold music? This would
 >  > obviously not work for high load but...
 >  > JohnM
 >  >
 >  >
 >  > ___
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 >  >  http://lists.digium.com/mailman/listinfo/asterisk-users
 >  >
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--
 Atis Lezdins,
 VoIP Project Manager / Developer,
 [EMAIL PROTECTED]
 Skype: atis.lezdins
 Cell Phone: +371 28806004
 Cell Phone: +1 800 7300689
 Work phone: +1 800 7502835


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Re: [asterisk-users] Analog modem as phone

[Zoa]

I'd say, save yourself the time and the frustration, drop the idea and 
buy a real voice card.

Zoa

Ronny Forberger wrote:
> Hi,
> maybe this has been asked before but I couldnt find a proper answer on  
> the web or list.
>
> I want to use a analog V.92 modem to make outgoing (and possibly)  
> incoming phone call through a standard analog phone line.
> I found  on web it's easy been done via chan_modem.so module. But this  
> seems removed from asterisk or buggy.
>
> So my questions are can I enable chan_modem.so to be built or what  
> other way to connect a modem to asterisk ?
>
> Thanks in advance,
>
> Ronny
>
>   


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Re: [asterisk-users] How to give user a prompt before connecting thecall

[Jeremy Mann]

I haven't, didn't know if you knew off the top of your head.

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Hales
Sent: Tuesday, April 01, 2008 7:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to give user a prompt before connecting 
thecall


I don't entirely remember - I was writing this code from memory.

Have you done any testing?

PaulH


On Tue, 2008-04-01 at 08:47 -0500, Jeremy Mann wrote:
> Can I assume after exten=>2,1,Playback(thanksfortakingthecall) there's more 
> logic, or does asterisk handle the connection between both parties at that 
> point?
>
>
>
>
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Hales
> Sent: Monday, March 31, 2008 9:34 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] How to give user a prompt before connecting 
> thecall
>
>
> Something like this:
>
> Dialling:
>
> exten => s,n(dial),Dial($ZAP/G1/${number},15,M(check)gm)
> exten => s,n,Dbget(next/number)
> exten => s,n,Goto(dial)
>
>
> {macro-check}
> exten => s,n,Playback(${heresacall})
> exten => s,n,Read(response,options,1)
> exten => s,n,Goto(${response},1)
>
> exten => 1,1,Macroexit
>
> exten => 2,1,Playback(thanksfortakingthecall)
>
>
> This hasn't been tested. Give it a red hot go.
>
> Another option is to set up a queue with external numbers as members,
> and set the queue as need the memebrs to accept the calls. (not that I
> can remember that option)
>
> PaulH
>
>
> On Mon, 2008-03-31 at 20:55 -0500, Jeremy Mann wrote:
> > Please do!
> >
> > 
> > From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of Paul Hales [EMAIL 
> > PROTECTED]
> > Sent: Monday, March 31, 2008 7:50 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [asterisk-users] How to give user a prompt before  connecting  
> > thecall
> >
> > It can be done via the 'visit a macro' part of the dial command...
> >
> > If anyone would like, i can post a code sample.
> >
> > PaulH
> >
> >
> > On Mon, 2008-03-31 at 15:33 -0400, Dean Collins wrote:
> > > Yes it is.
> > > I'm remote at the moment so I can't send you the code but google for 
> > > mobile remote receiver and you'll find what you are looking for.
> > > Lots of people do it so they don't have calls to cell phones picked up by 
> > > voicemail.
> > >
> > >
> > > Cheers
> > > dean
> > >
> > >
> > > -Original Message-
> > > From: Pete Kay <[EMAIL PROTECTED]>
> > > Sent: Monday, March 31, 2008 2:27 PM
> > > To: Asterisk Users Mailing List - Non-Commercial Discussion 
> > > 
> > > Subject: [asterisk-users] How to give user a prompt before connecting 
> > > thecall
> > >
> > > Hello,
> > >
> > > Is it possible to request for the premission from the called party  
> > > through
> > > a prompt before routing the call?
> > > For instance, before actually connecting two parties through the use of 
> > > DIAL
> > > command in the dialplan, I want to let Asterisk to automatically
> > > ask for the called party to decide whether he/she would like to be
> > > connected.  ( ex. Press 1 to connect and 2 to hangup).
> > >
> > > Can this function be done?  If so, how to do it?
> > >
> > > Thank you .
> > >
> > > Pete Dao
> > >
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> >
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> >
> > This e-mail, facsimile, or letter and any files or attachments transmitted 
> > with it contains information that is confidential and privileged. This 
> > information is intended only for the use of the individual(s) and 
> > entity(ies) to whom it is addressed. If you are the intended recipient, 
> > further disclosures are prohibited without proper authorization. If you are 
> > not the intended recipient, any disclosure, copying, printing, or use of 
> > this information is strictly prohibited and possibly a violation of federal 
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[asterisk-users] Analog modem as phone

[Ronny Forberger]

Hi,
maybe this has been asked before but I couldnt find a proper answer on  
the web or list.

I want to use a analog V.92 modem to make outgoing (and possibly)  
incoming phone call through a standard analog phone line.
I found  on web it's easy been done via chan_modem.so module. But this  
seems removed from asterisk or buggy.

So my questions are can I enable chan_modem.so to be built or what  
other way to connect a modem to asterisk ?

Thanks in advance,

Ronny

-- 

Ronny Forberger
Systemadministration & IT-Support

elego Software Solutions GmbH
Gustav-Meyer-Allee 25
Gebäude 12, Raum 227
D-13355 Berlin

Tel. +49 30 23 45 86 96  ronny.forberger at elegosoft.com
Fax  +49 30 23 45 86 95  http://www.elegosoft.com

Geschäftsführer: Olaf Wagner, Sitz Berlin
Amtsgericht Berlin-Charlottenburg, HRB 77719, USt-IdNr: DE163214194



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Re: [asterisk-users] Need some input for Quad T1 and channel banks.

[Andrew Latham]

Here I will say it http://xorcom.com



On Mon, Mar 31, 2008 at 6:01 PM, Al lists <[EMAIL PROTECTED]> wrote:
> I'm looking to install a system with 80 FXS analog phones.
> At this time the only cost effective solution is using a 4 port T1 card and
> addit 600 channel bank.
> Has anyone tried this solution? any good documents beside
> http://www.voip-info.org/wiki/index.php?page=Asterisk+hardware+channel+bank+check
>  as far as i know, addit 600 T1 interface is not PRI (please correct me if
> i'm wrong) its CAS robbed bit, will that work with new Digium quad T1 like
> TE410P ?( I prefer to use Digium if possible)
> The system is connected to the Telco through SIP trunk so all we have in
> terms of analog is local loop, Do we need to have echo cancel in this
> scenario ?
>  Thanks!
>
>
>
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-- 
/*
 Andrew Latham
 LATHAMA (lay-th-ham-eh)
 [EMAIL PROTECTED]
 [EMAIL PROTECTED]

 TuxTone Inc.
 http://www.TuxTone.com
*/

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Re: [asterisk-users] help with no audio

[Eric Wieling]

Yes, some kernels don't work with ztdummy.  This is discussed over and 
over and over again on this mailing list.  Check the archives.

Tzafrir Cohen wrote:
> On Wed, Apr 02, 2008 at 02:16:56PM -0400, Jerry Geis wrote:
>>> On Tue, Apr 01, 2008 at 01:44:39PM -0400, Jerry Geis wrote:
>>>
 / I have no card in this unit at this time.
>>> />/ lsmod shows ztdummy loaded.
>>> /
>>> Just to make sure that this is not the problem, what's the output of:
>>>
>>>  zttest -c 3
>>>
>>> -- 
>> When running this nothing comes back...
>> It says "Opened pseduo zap interface, measuring accuracy..."
>> and that is all.
>>
>> I am using Centos 2.6.18-53.1.14.el5
>>
>> I also just tried rmmod ztdummy and then starting asterisk again and the 
>> audio works.
>> something is wrong with ztdummy.
>>
>> I went back to zaptel-1.4.9.2 diretory and did make clean; ./configure; 
>> make; make install
>> (one at a time ) I saw no errors. tail /var/log/messages after modprove 
>> showed no errors.
> 
> I believe that this means nothing. modprobe does nothing if the module
> is already loaded.
> 

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

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Re: [asterisk-users] help with no audio

[Tzafrir Cohen]

On Wed, Apr 02, 2008 at 02:16:56PM -0400, Jerry Geis wrote:
> >
> >On Tue, Apr 01, 2008 at 01:44:39PM -0400, Jerry Geis wrote:
> >
> >>/ I have no card in this unit at this time.
> >/>/ lsmod shows ztdummy loaded.
> >/
> >Just to make sure that this is not the problem, what's the output of:
> >
> >  zttest -c 3
> >
> >-- 
> When running this nothing comes back...
> It says "Opened pseduo zap interface, measuring accuracy..."
> and that is all.
> 
> I am using Centos 2.6.18-53.1.14.el5
> 
> I also just tried rmmod ztdummy and then starting asterisk again and the 
> audio works.
> something is wrong with ztdummy.
> 
> I went back to zaptel-1.4.9.2 diretory and did make clean; ./configure; 
> make; make install
> (one at a time ) I saw no errors. tail /var/log/messages after modprove 
> showed no errors.

I believe that this means nothing. modprobe does nothing if the module
is already loaded.

-- 
   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] help with no audio

[Jerry Geis]

On Tue, Apr 01, 2008 at 01:44:39PM -0400, Jerry Geis wrote:

>/ I have no card in this unit at this time.
/>/ lsmod shows ztdummy loaded.
/
Just to make sure that this is not the problem, what's the output of:

  zttest -c 3

--

When running this nothing comes back...
It says "Opened pseduo zap interface, measuring accuracy..."
and that is all.

I am using Centos 2.6.18-53.1.14.el5

I also just tried rmmod ztdummy and then starting asterisk again and the 
audio works.

something is wrong with ztdummy.

I went back to zaptel-1.4.9.2 diretory and did make clean; ./configure; 
make; make install
(one at a time ) I saw no errors. tail /var/log/messages after modprove 
showed no errors.


Now what?

Jerry
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Re: [asterisk-users] help with no audio

[Tzafrir Cohen]

On Tue, Apr 01, 2008 at 01:44:39PM -0400, Jerry Geis wrote:

> I have no card in this unit at this time.
> lsmod shows ztdummy loaded.

Just to make sure that this is not the problem, what's the output of:

  zttest -c 3

-- 
   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] Cisco 7965 SIP Firmware

[Razza]

For interested parties
It would appear I didn't need to had anything between the
 tags  in XMLDefault.cnf.xml and/or
SEP{MAC Addr}.cnf.xml to force the upgrade, although the correct version of
firmware needs to be present between the tags once it is converted.

When I factory reset the phone (power-off the device, power-on the device,
hold down the # key for a few seconds, then type 1,2,3,4,5,6,7,8,9,*,0,#),
it looks for term65.default.loads, which in turn instructs the phone to load
the appropriate images/apps.
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[asterisk-users] OpenFrame

[Dean Collins]

Not sure why I missed this earlier in the year but has anyone had a look
at OpenFrame home handset?

Any comments?

 

http://www.pcmag.com/article2/0,2704,2246158,00.asp 

 

 

Regards,

Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED] 
+1-212-203-4357
+61-2-9016-5642 (Sydney in-dial). 

 

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Re: [asterisk-users] help with no audio

[Jerry Geis]

Jerry Geis wrote:
>>
>> On Tue, 2008-04-01 at 13:24 -0400, Jerry Geis wrote:
>> >/ I call into the dialplan and try to play demo-congrats and I hear 
>> nothing.
>> />/ />/ Firewall is disabled. />/ Everything is on the 192.168.1.X 
>> network for this simple configuration.
>> />/ The tftp server is giving the polycom phone the config files.
>> />/ />/ Any ideas why I dont hear audio?
>> /
>> Do you happen to have an unconfigured T1 card in your machine?  That's
>> the most common problem I see for people when they get no audio at all
>> coming out of Asterisk.  If that's not the case, I'd turn RTP debugging
>> on in the command-line and make sure RTP packets are coming and going
>> from the Asterisk box.
>>
>> -- 
>> Jared Smith
>> Community Relations Manager
>> Digium, Inc.
>
> Jared,
>
> I have no card in this unit at this time.
> lsmod shows ztdummy loaded.
>
> I turned on "rtp debug" and got a bunch of lines like:
> Got RTP packet from 192.168.1.99:2226 
>
> Does that help you?
>
> Jerry
>
I have found the echo command. I modified the dialplan to use echo.
I turned on rtp debug and I see packets going BOTH ways.

I have looked all through the zaptel.conf (below)
everything is commented out. there are no cards in my box. zapata looks 
the same everything commented out.

I am not finding a reason for not getting audio packets sent back to the 
phone.

Any suggestion on something to try?

Jerry



-

#
# Zaptel Configuration File
#
# This file is parsed by the Zaptel Configurator, ztcfg
#
#
# First come the span definitions, in the format
# span=[,yellow]
# 
# The timing parameter determines the selection of primary, secondary, and
# so on sync sources.  If this span should be considered a primary sync
# source, then give it a value of "1".  For a secondary, use "2", and so on.
# To not use this as a sync source, just use "0"
#
# The line build-out (or LBO) is an integer, from the following table:
# 0: 0 db (CSU) / 0-133 feet (DSX-1)
# 1: 133-266 feet (DSX-1)
# 2: 266-399 feet (DSX-1)
# 3: 399-533 feet (DSX-1)
# 4: 533-655 feet (DSX-1)
# 5: -7.5db (CSU)
# 6: -15db (CSU)
# 7: -22.5db (CSU)
#
# The framing is one of "d4" or "esf" for T1 or "cas" or "ccs" for E1
#
# Note: "d4" could be referred to as "sf" or "superframe" 
#
# The coding is one of "ami" or "b8zs" for T1 or "ami" or "hdb3" for E1
#
# E1's may have the additional keyword "crc4" to enable CRC4 checking
#
# If the keyword "yellow" follows, yellow alarm is transmitted when no
# channels are open.
#
#span=1,0,0,esf,b8zs
#span=2,1,0,esf,b8zs
#span=3,0,0,ccs,hdb3,crc4
#
# Next come the dynamic span definitions, in the form:
# dynamic=,,,
#
# Where  is the name of the driver (e.g. eth),  is the
# driver specific address (like a MAC for eth),  is the number
# of channels, and  is a timing priority, like for a normal span.
# use "0" to not use this as a timing source, or prioritize them as
# primary, secondard, etc.  Note that you MUST have a REAL zaptel device
# if you are not using external timing.
#
# dynamic=eth,eth0/00:02:b3:35:43:9c,24,0
#
# Next come the definitions for using the channels.  The format is:
# =
#
# Valid devices are:
#
# "e&m" : Channel(s) are signalled using E&M signalling (specific
# implementation, such as Immediate, Wink, or Feature Group D
# are handled by the userspace library).
# "fxsls"   : Channel(s) are signalled using FXS Loopstart protocol.
# "fxsgs"   : Channel(s) are signalled using FXS Groundstart protocol.
# "fxsks"   : Channel(s) are signalled using FXS Koolstart protocol.
# "fxols"   : Channel(s) are signalled using FXO Loopstart protocol.
# "fxogs"   : Channel(s) are signalled using FXO Groundstart protocol.
# "fxoks"   : Channel(s) are signalled using FXO Koolstart protocol.
# "sf"  : Channel(s) are signalled using in-band single freq tone.
#   Syntax as follows: 
#channel# => 
sf:,
#   rxfreq is rx tone freq in hz, rxbw is rx notch (and decode)
#   bandwith in hz (typically 10.0), rxflag is either 'normal' or
#   'inverted', txfreq is tx tone freq in hz, txlevel is tx tone 
#   level in dbm, txflag is either 'normal' or 'inverted'. Set 
#   rxfreq or txfreq to 0.0 if that tone is not desired.
# "unused"  : No signalling is performed, each channel in the list remains idle
# "clear"   : Channel(s) are bundled into a single span.  No conversion or
# signalling is performed, and raw data is available on the master.
# "indclear": Like "clear" except all channels are treated individually and
# are not bundled.  "bchan" is an alias for this.
# "rawhdlc" : The zaptel driver performs HDLC encoding and decoding on the 
# bundle, and the resulting data is communicated via the master
# device.
# "fcshdlc" : The zapdel driver performs HDLC encoding and decoding on the
# bundle and also performs incoming and outgoin

Re: [asterisk-users] zaptel alarm

[Greg Woods]

Shaun Ruffell wrote:
>
>  >> svn co http://svn.digium.com/svn/zaptel/branches/[EMAIL PROTECTED]
>   zaptel-1.4-4122
>   

Thank you, I will try that tonight when  I get home and report back.

--Greg


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Re: [asterisk-users] Voicemail- Recorded Mesage Low Volume

[John Meksavan]

Thanks.  I will give this a try.

-John

> From: [EMAIL PROTECTED]
> To: asterisk-users@lists.digium.com
> Date: Wed, 2 Apr 2008 09:29:48 -0700
> Subject: Re: [asterisk-users] Voicemail- Recorded Mesage Low Volume
> 
> 
> On Apr 1, 2008, at 5:22 PM, [EMAIL PROTECTED]  
> wrote:
> 
> >  Can the volume of the recorded voice mail message be changed?  If
> > so, what I am doing wrong?  Any input would be greatly appreciated.
> > Thanks.
> 
> I had a similar problem in our setup where we e-mail the recorded  
> messages to e-mail retrieval.  But this also helps standard phone  
> retrieval too.  What I did was edit the /usr/sbin/safe_asterisk script  
> and add:
> 
> PATH="/usr/local/bin:$PATH"
> 
> At the top of the script. This would let me override the default sox  
> implementation that Asterisk uses.  Then I loaded in a script (called  
> sox) that would compress and normalize the recorded audio (It  
> compresses to deal with the spikes of the noise of the handset being  
> hung up, etc.). It works pretty well for us and makes the volume  
> pretty good so we don't have to crank up the volume on our computers  
> or phones to listen to voicemail messages.  And we can't adjust the  
> rxgain as it is already a good volume for normal calls.
> 
> Daniel
> 
> --CUT--
> #!/bin/sh
> #
> # $1 = -v
> # $2 = number
> # $3 = inFile
> # $4 = outFile
> #
> REALSOX="/usr/bin/sox"
> 
> if [ "$1" != "-v" ]; then
>$REALSOX $*
>exit $?
> fi
> 
> INFILE="$3"
> OUTFILE="$4"
> 
> #
> # Perform the gain adjustment.
> #
> $REALSOX "$INFILE" "$OUTFILE" compand 0.1,0.3  
> -60,-60,-30,-15,-20,-12,-4,-8,-2,-7 0 0 0.2
> --CUT--
> 
> 
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[asterisk-users] Asterisk 1.4.19 and Asterisk-addons 1.6.0-beta3 Released

[The Asterisk Development Team]

The Asterisk development team has released version 1.4.19 of Asterisk and
1.6.0-beta3 of Asterisk-addons.

The new Asterisk-addons release contains a few bug fixes over the previous 
version.

http://svn.digium.com/view/asterisk-addons/tags/1.6.0-beta3/ChangeLog?view=markup

Asterisk 1.4.19 contains a large number of fixes over the previous release,
1.4.18.  For a full list of changes, see the ChangeLog that is included in the
release.

http://svn.digium.com/view/asterisk/tags/1.4.19/ChangeLog?view=markup

One change that requires specific attention is a change to iLBC support.  Due to
problems with the licensing of the iLBC source code, the implementation of the
codec has been removed from the Asterisk source tree.  To get the codec_ilbc
module to compile, you will have to retrieve the iLBC source code.  A script has
been provided which does this for you.  Simply run the
contrib/scripts/get_ilbc_source.sh script from the root directory of the
Asterisk source tree.

All users of the iLBC source code should review the license agreement and take
whatever actions may be necessary to comply with its terms before continuing to
use codec_ilbc with Asterisk.

Thank you for your support!

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Re: [asterisk-users] CentPBX mirror?

[Jonn Taylor]

I have a some setup scripts that use centos 4 or 5 and freepbx you are 
welcome to use them.

Jonn

http://www.taylortelephone.com/asterisk/


Chris Bagnall wrote:
>> CentPBX has bit the dust I believe.
>> 
>
> Thanks. Any suggestions for a suitable FreePBX-based alternative with kernel 
> support for a Dell R200 (it's usually the SAS controller that causes the 
> problem)? I've tried PBX-in-a-Flash without success, and Trixbox is rather 
> too "customized" for what I'm after for this deployment.
>
> TIA.
>
> Regards,
>
> Chris
>   

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Re: [asterisk-users] Voicemail- Recorded Mesage Low Volume

[Daniel Hazelbaker]

On Apr 1, 2008, at 5:22 PM, [EMAIL PROTECTED]  
wrote:

>  Can the volume of the recorded voice mail message be changed?  If
> so, what I am doing wrong?  Any input would be greatly appreciated.
> Thanks.

I had a similar problem in our setup where we e-mail the recorded  
messages to e-mail retrieval.  But this also helps standard phone  
retrieval too.  What I did was edit the /usr/sbin/safe_asterisk script  
and add:

PATH="/usr/local/bin:$PATH"

At the top of the script. This would let me override the default sox  
implementation that Asterisk uses.  Then I loaded in a script (called  
sox) that would compress and normalize the recorded audio (It  
compresses to deal with the spikes of the noise of the handset being  
hung up, etc.). It works pretty well for us and makes the volume  
pretty good so we don't have to crank up the volume on our computers  
or phones to listen to voicemail messages.  And we can't adjust the  
rxgain as it is already a good volume for normal calls.

Daniel

--CUT--
#!/bin/sh
#
# $1 = -v
# $2 = number
# $3 = inFile
# $4 = outFile
#
REALSOX="/usr/bin/sox"

if [ "$1" != "-v" ]; then
   $REALSOX $*
   exit $?
fi

INFILE="$3"
OUTFILE="$4"

#
# Perform the gain adjustment.
#
$REALSOX "$INFILE" "$OUTFILE" compand 0.1,0.3  
-60,-60,-30,-15,-20,-12,-4,-8,-2,-7 0 0 0.2
--CUT--


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Re: [asterisk-users] zaptel alarm

[Shaun Ruffell]

Greg Woods wrote:
>> If you have subversion installed on your server, could you try using 
>> this version of zaptel:
>>
>> http://svn.digium.com/svn/zaptel/branches/[EMAIL PROTECTED]> 
> 
> What's the svn command to fetch it and I'll try it.
> 

 >> svn co http://svn.digium.com/svn/zaptel/branches/[EMAIL PROTECTED]
  zaptel-1.4-4122

 >> cd zaptel-1.4-4122
 >> make; make install



Shaun


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Re: [asterisk-users] CentPBX mirror?

[Outback Dingo]

just an afterthought there is askozia also... though also FreeBSD based with
Web GUI

On Wed, Apr 2, 2008 at 11:10 PM, Chris Bagnall <[EMAIL PROTECTED]> wrote:

> > CentPBX has bit the dust I believe.
>
> Thanks. Any suggestions for a suitable FreePBX-based alternative with
> kernel support for a Dell R200 (it's usually the SAS controller that causes
> the problem)? I've tried PBX-in-a-Flash without success, and Trixbox is
> rather too "customized" for what I'm after for this deployment.
>
> TIA.
>
> Regards,
>
> Chris
> --
> C.M. Bagnall, Director, Minotaur I.T. Limited
> For full contact details visit http://www.minotaur.it
> This email is made from 100% recycled electrons
>
>
>
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Re: [asterisk-users] CentPBX mirror?

[Outback Dingo]

FreeBSD 7 asterisk and asterisk-gui from ports should make it quite easy to
get going, less then an
hour to install base OS and build ports, I did an install for a client in
less then 2 hours


On Wed, Apr 2, 2008 at 11:10 PM, Chris Bagnall <[EMAIL PROTECTED]> wrote:

> > CentPBX has bit the dust I believe.
>
> Thanks. Any suggestions for a suitable FreePBX-based alternative with
> kernel support for a Dell R200 (it's usually the SAS controller that causes
> the problem)? I've tried PBX-in-a-Flash without success, and Trixbox is
> rather too "customized" for what I'm after for this deployment.
>
> TIA.
>
> Regards,
>
> Chris
> --
> C.M. Bagnall, Director, Minotaur I.T. Limited
> For full contact details visit http://www.minotaur.it
> This email is made from 100% recycled electrons
>
>
>
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Re: [asterisk-users] Digium B410P, bristuff and BRI support in 1.6

[Jean-Denis Girard]

Olivier a écrit :
> Would you mind if I asked you this :
> - Which card did you include in your home system ? Are you using an ISDN 
> BRI access ?

This is a basic BRI card with HFC chipset (Bewan Gazel 128)

> - Is libpri necessary for ISDN BRI access ? I thought libpri was mostly 
> dedicated to E1/T1 access

Development version of libpri (libpri-trunk) does include prliminary 
support for BRI.

>  From above, do you understand that Digium is committed to support BRI 
> cards in 1.6 ?
> If positive, which cards will be supported and with which feature set ?

I can't speak for Digium !


Thanks,
-- 
Jean-Denis Girard

SysNux  Systèmes  Linux  en Polynésie française
http://www.sysnux.pf/   Tél: +689 50 10 40 / GSM: +689 79 75 27

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[asterisk-users] zaptel_vldtmf

[poliveira]

Hi.

I am trying to install "asterisk_srtp".
I started by installing zaptel-1.4.9.2 and then I run the configure of  
asterisk_srtp. In the menuselect of asterisk, the chan_zap in "Channel  
Drivers" is always unselected(XXX). The only dependency that I don't  
seem to have is zaptel_vldtmf.
Where can I find this zaptel_vldtmf? If I understod correctly, zaptel  
1.4 should already have this.

Thanks




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Re: [asterisk-users] TDM410E card, 1 FXO module - how to dial Out

[Mojo with Horan & Company, LLC]

Kevin P. Fleming wrote:
> Mojo with Horan & Company, LLC wrote:
>
>   
>> P.S.  If you can't dial seven digit numbers in your area, but you miss 
>> it, you can restore that behavior if you feel like selecting a default 
>> area code:
>>
>> exten => _NXX,1,Dial(Zap/1/907${EXTEN},,TWK)
>>
>> Here, if I dial a seven digit number, asterisk dials 907 followed by my 
>> seven digits out the phone line.
>> 
>
> Well, sort of. This will also trigger if you dial the first 7 digits of
> a 10-digit number from a device that doesn't dial 'en bloc', since there
> is no longer any way to distinguish 7-vs-10 digit numbers by the number
> pattern. In other words, this will work fine if you are dialing from a
> SIP phone, but not if you are dialing from an analog phone.
>
>   
I know you're not the person I should be asking "Are you sure?"   but it 
did seem like when I had an analog phone plugged into an FXS in a TDM 
card that asterisk paused a bit to make sure I wasn't entering any more 
digits, because I didn't use the wildcard '!' maybe?   Just getting 
confused, I guess -- It must have been when my IAXy was installed!

Thanks for the correction :)


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Re: [asterisk-users] CentPBX mirror?

[Chris Bagnall]

> CentPBX has bit the dust I believe.

Thanks. Any suggestions for a suitable FreePBX-based alternative with kernel 
support for a Dell R200 (it's usually the SAS controller that causes the 
problem)? I've tried PBX-in-a-Flash without success, and Trixbox is rather too 
"customized" for what I'm after for this deployment.

TIA.

Regards,

Chris
-- 
C.M. Bagnall, Director, Minotaur I.T. Limited
For full contact details visit http://www.minotaur.it
This email is made from 100% recycled electrons



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[asterisk-users] RTP no sound on asterisk

[Jerry Geis]

Hi all, I seem to only be getting (1) call to sip_write() in 
channels/chan_sip.c

I have a very simple setup. one server (no cards) 2 polycom IP 330 phones.
Server is 192.168.1.150 and phone is DHCP. Nothing else on the network.
No firewall is enabled.

I call into the dialplan with:

exten => 112,1,Answer
exten => 112,n,Playback(demo-congrats)
exten => 112,n,Hangup

I see this executing on the CLI. However  I have no audio.

Enabling RTP debug I see the Got RTP packet but there are no send RTP 
packets going out.

I edited the source and put logging messages first in main/rtp.c and I 
saw the ast_rtp_raw_write() getting called 1 time.
so I backed up the tree. Got into channels/chan_sip.c sip_write() and it 
only gets called 1 time.

I have had a couple of times where I heard audio. Hangup up and tried 
again. And NO audio for bunch more times...

What can be causing my RTP issue and no audio?

Jerry

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Re: [asterisk-users] Asterisk and radius

[Andrea Spadaccini]

Ciao FaberK,

> Hi folks,
> I'm trying to install asterisk with radius cdr support.
> I got freeradius up and running, so following radius instructions
> inside asterisk source package, I've installed radiusclient-ng and
> relative headers.
> But when I start configure(asterisk 1.4.18.1) I got:
> checking for rc_read_config in -lradiusclient-ng... no
> If I type:
> ./configure --with-radius=/usr/share/radiusclient-ng
> the answer is the same, more:
> checking for rc_read_config in -lradiusclient-ng... no
> configure: ***
> configure: *** The Radius Client installation on this system appears
> to be broken.
> configure: *** Either correct the installation, or run configure
> configure: *** without explicitly specifying --with-radius
> But the installation of radiusclient, didn't give me any problems.

Don't know anything about FreeRadius, but you might try running 'ldconfig' and
seeing if it works.

Best regards,

-- 
Dr. Andrea Spadaccini
Multimedia Technologies Institute - MTI S.r.l.
Web: www.x-voice.it - Tel: +39 (0) 95 7224945

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Re: [asterisk-users] Call deflection on ISDN PRI in Sweden

[Hanna Wallin]

Thanks for your answer.

I've found out that the zaptel drivers don't support Call Deflection at the 
moment and in Sweden the callerid can be set to anything different than the 
phonenumber of the caller. 

Have to find a workaround :)

/hanna


-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andreas van dem 
Helge
Sent: den 28 mars 2008 20:42
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Call deflection on ISDN PRI in Sweden

*CLI> show application Transfer

  -= Info about application 'Transfer' =-

[Synopsis]
Transfer caller to remote extension

[Description]
  Transfer([Tech/]dest[|options]):  Requests the remote caller be transferred
to a given destination. If TECH (SIP, IAX2, LOCAL etc) is used, only
an incoming call with the same channel technology will be transfered.
Note that for SIP, if you transfer before call is setup, a 302 redirect
SIP message will be returned to the caller.

The result of the application will be reported in the TRANSFERSTATUS
channel variable:
   SUCCESS  Transfer succeeded
   FAILURE  Transfer failed
 ***  UNSUPPORTED  Transfer unsupported by channel driver ***


So what you need to do is use app_dial instead of app_transfer.
Everything else should be able to remain the same.

On Fri, Mar 28, 2008 at 4:25 AM, Hanna Wallin
<[EMAIL PROTECTED]> wrote:
>
>
>
>
> Hello List!
>
>
>
> We're having trouble making call deflection on ISDN PRI. We would like to
> transfer a call to an external extension but keeping the callerid of the
> caller so it can be presented to the receiver of the transferred call.
>
> At the time we're using Zaptel 1.4.5.1, Asterisk 1.4.11 and Digium hardware
> TE420B. We've ordered the service (CD) from the phone company.
>
>
>
> The zapata.conf file inlcludes:
>
> Transfer= yes
>
> Facilityenable=yes
>
> Callerid=asreceived
>
>
>
> In extensions.conf we try to transfer a call to an external extension as:
> Transfer(ZAP/g0/ ) but that fails with the ${TRANSFERSTATUS} =
> UNSUPPORTED.
>
>
>
> Ideas anyone? We would really appreciate it!
>
>
>
>
>
> Kind regards,
>
>
>
> Hanna
>
>
>
>
>
>
>
>
>
> Hanna Wallin
>  System Development
>
> Direct: +46 (0)8 736 77 29
>  Mobile: +46 (0)73 414 13 38
>  Fax: +46 (0)8 736 77 91
>  E-mail: [EMAIL PROTECTED]
>
>
>
>  PocketMobile Communications AB
>  Wenner-Gren Center
>  Sveavägen 168, 3 tr
>  113 46 Stockholm
>
> Nordic web page: www.pocketmobile.se
>  International web page: www.pocketmobileworld.com
>
>
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Re: [asterisk-users] Call deflection on ISDN PRI in Sweden

[Hanna Wallin]

Thanks Matthew!
Now I can start looking for a workaround ;)

/hanna

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matthew
Fredrickson
Sent: den 28 mars 2008 16:39
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Call deflection on ISDN PRI in Sweden

Hanna Wallin wrote:
> Hello List!
> 
>  
> 
> We're having trouble making call deflection on ISDN PRI. We would like
to transfer a call to an external extension but keeping the callerid of
the caller so it can be presented to the receiver of the transferred
call.
> 
> At the time we're using Zaptel 1.4.5.1, Asterisk 1.4.11 and Digium
hardware TE420B. We've ordered the service (CD) from the phone company. 
> 
>  
> 
> The zapata.conf file inlcludes: 
> 
> Transfer= yes
> 
> Facilityenable=yes
> 
> Callerid=asreceived
> 
>  
> 
> In extensions.conf we try to transfer a call to an external extension
as: Transfer(ZAP/g0/ ) but that fails with the ${TRANSFERSTATUS}
= UNSUPPORTED.
> 
>  
> 
> Ideas anyone? We would really appreciate it!
> 

That supplementary service (CD) is not supported in libpri right now, so

that would be the reason why it doesn't work.  The Transfer() 
application is for analog lines, IIRC.

-- 
Matthew Fredrickson
Software/Firmware Engineer
Digium, Inc.

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Re: [asterisk-users] zaptel alarm

[Greg Woods]

> If you have subversion installed on your server, could you try using 
> this version of zaptel:
> 
> http://svn.digium.com/svn/zaptel/branches/[EMAIL PROTECTED]

Not Found
The requested URL /svn/zaptel/branches/[EMAIL PROTECTED] was not found on this
server.


Apache Server at svn.digium.com Port 80



What's the svn command to fetch it and I'll try it.

--Greg



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[asterisk-users] Problems with DELL 1600

[Ruben Zamora]

Hi

I just want to know if anyone have problems with server DELL 1600, 
Like:  Hangup Call.

Thanks

Ruben

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Re: [asterisk-users] FXS, Power and Sangoma

[Tim Nelson]

That does indeed sound a bit odd. I've run 12-48 FXS ports from a single molex 
connector with Sangoma hardware. Try testing your power supply with a 
multimeter to ensure its putting out the proper voltage. I would not trust the 
extnernal AC adapters as I've found they typically have voltage that is too 
low...

Tim Nelson
Systems/Network Support
Rockbochs Inc.

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

Sent: Tuesday, April 1, 2008 8:20:36 PM (GMT-0600) America/Chicago
Subject: [asterisk-users] FXS, Power and Sangoma

Hi
   I've a Sangoma A200D with 2FXO and 2FXS.  When using it with only  
the FXO module, it's all good.  But when I put in the FXS module and  
connect the power, logs tells me not enough power.

> Mar 31 14:11:54 phone kernel: [ 4761.246931] wanpipe1: Module 1:  
> Failed to powerup within 600 ms (8V : 72V)!
> Mar 31 14:11:54 phone kernel: [ 4761.246937] wanpipe1: Module 1: Did  
> you remember to plug in the power cable?


So I disconnect other power devices in the box (Dell Optiplex GX270)  
such as the Zip drive and CDROMs, but no luck.  Then I took an  
external power supply with molex connector 
(http://www.coolerguys.com/840556029977.html 
  or http://www.cablesonline.com/mo4inpotoacp.html) and tried that  
with still the same thing.

How much power does this card need?  The AC adapter puts out up to 2  
Amps.  Sangoma support is only telling me to get a bigger power supply  
and don't use the AC adapter.  Has anyone else seen this issue?  Could  
something else be wrong?
  thanks
Todd

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Re: [asterisk-users] How to wait before sending DTMF in DIAL command

[Eric Wieling]

A "w" in the D() string will wait .5 second.  Example: 
Dial(Zap/g1/5551212,,D(ww668))

If you are dialing out of an FXO or FXS signaled port, you can add "w" 
to the dial string to wait .5 second.  Example: Dial(Zap/g1/ww5551212)

Pete Kay wrote:
> Is there anyway to have Asterisk to wait for 1 second before sending a DTMF
> using the D() option?

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

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Re: [asterisk-users] How to Hangup after DIAL is completed

[Tony Mountifield]

In article <[EMAIL PROTECTED]>,
Pete Kay <[EMAIL PROTECTED]> wrote:
> 
> Dear Tony,
> 
> Thank you very much for your suggestion.
> The thing is that ${DIALSTATUS} returns ANSWER regardless of whether the
> called party hangs up or not.
> For instance, when called party is being asked by Asterisk whether he/she
> wants to pick the call, ${DIALSTATUS} returns ANSWER.
> 
> In the case where the called party accepts to take the call and then hangs
> up after finishing the conversion, I want to stop AGI from jumping to call
> the next number.  On the other hand, if the called party decides not to take
> the call, I want to let AGI to jump to the next number.
> 
> It seems like ${DIALSTATUS} does not give me the info that I need.

Ah, I see. It's not just whether the call was answered, but you want what
happens on the called channel to govern future behaviour on the calling
channel.

> Thank you very much.  Is there any other ways you may think of?

The only way I can think of to do that is by using the Asterisk database
as a communication mechanism, combined with channel variable inheritance.

Before calling Dial, generate a unique key. This can be based on ${UNIQUEID}
for example. Store this in a variable with inheritance to dependent
channels (invoked by adding a leading underscore): e.g.

Set(_KEY=${UNIQUEID})
or in AGI
SET VARIABLE _KEY "${UNIQUEID}"

Then write an entry to the Asterisk database with a suitable family, this key
and an arbitrary value:

Set(DB(dial/${KEY})=0)
or in AGI
SET VARIABLE DB(dial/${KEY}) "0"

In the answer macro, you should be able to read ${KEY} (due to inheritance),
and if the user accepts the call, you delete the database entry:

NoOp(${DB_DELETE(dial/${KEY})})

In the calling channel, on return from the Dial, you check to see if the entry
still exists. If it does, then either the call was not answered, or else it
was answered and not accepted. You can then either try another dial (using the
same DB entry) or give up and delete the entry at that point.

Deleting the entry in AGI would be (I think):

GET FULL VARIABLE DB_DELETE(dial/${KEY})

Or you could do GET VARIABLE if you substituted the value of ${KEY} yourself.

> Thanks,
> Pete

Hope this helps (and that it works!)

Cheers
Tony

> On Wed, Apr 2, 2008 at 3:03 PM, Pete Kay <[EMAIL PROTECTED]> wrote:
> 
> > Hi,
> >
> > I have a problem with DIAL.
> > The scenario is this:
> >
> > 1. Asterisk will dial a number in a call list
> > 2. called party picks up the call and hears a prompt asking if they want
> > to pick up ( this is done through M(marco) option in DIAL)
> > 3. if called party does not want to pick up, go to the next number (this
> > is done through SET MARCO_RESULT= CONTINUE)
> > 4. if called party decides to take the call, make some conversion, then
> > the call is completed and hanged up by the called party.
> >
> > For 4, when the called party hangs up, I want to hang up on the calling
> > party as well.
> >
> > This works if I use the dialplan by not including the "g" option in DIAL.
> > However, when I do the DIAL in AGI, the calling party does not get hangup
> > automatically at (4).  So, it continues to execute the next line in the AGI,
> > which is not what I want.
> >
> > The problem is that the result I can when executing the DIAL command
> > within AGI does not tell me whether the call was picked up or not. So, I
> > have no way of knowing whether to continue executing the AGI program or to
> > issue a HAGNGUP explicitly.
> >
> > Can anyone please help me ?
> >
> > Any suggestion will be greatly appreciated.
> >
> > Thanks,
> > Pete
> > ___
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> >
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> > To UNSUBSCRIBE or update options visit:
> >   http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> 
> 
> 
> On Wed, Apr 2, 2008 at 4:39 PM, Tony Mountifield <[EMAIL PROTECTED]>
> wrote:
> 
> > In article <[EMAIL PROTECTED]>,
> > Pete Kay <[EMAIL PROTECTED]> wrote:
> > > I have a problem with DIAL.
> > > The scenario is this:
> > >
> > > 1. Asterisk will dial a number in a call list
> > > 2. called party picks up the call and hears a prompt asking if they want
> > to
> > > pick up ( this is done through M(marco) option in DIAL)
> > > 3. if called party does not want to pick up, go to the next number (this
> > is
> > > done through SET MARCO_RESULT= CONTINUE)
> > > 4. if called party decides to take the call, make some conversion, then
> > the
> > > call is completed and hanged up by the called party.
> > >
> > > For 4, when the called party hangs up, I want to hang up on the calling
> > > party as well.
> > >
> > > This works if I use the dialplan by not including the "g" option in
> > DIAL.
> > > However, when I do the DIAL in AGI, the calling party does not get
> > hangup
> > > automatically at (4).  So, it continues to execute the next line in the
> > AGI,
> > > which is not what I want.
> > >
> > > Th

Re: [asterisk-users] interrupting MOH

[Matt Florell]

Hello,

We achieve this using an AGI script in the VICIDIAL project for our
version of inbound queues. You start MoH then when you stream a sound
to the channel it will stop MoH then after the sound is done you start
MoH back up again. Probably a bit more involved than what you want,
but it dose work well for us.

MATT---

On 4/2/08, Atis Lezdins <[EMAIL PROTECTED]> wrote:
> Sorry for top-posting, but seems everyone on this thread did so.
>
>  Also that would be my suggestion for now - call queue with periodic-announce.
>
>  However i see that this would make nice architectural improvement -
>  allow inject sound files into MoH stream. This would be useful for
>  example in call queues - to inject all the queue announcements into
>  MoH directly, rather than play them while blocking further queue
>  actions.
>
>  Regards,
>  Atis
>
>
>
>  On Wed, Apr 2, 2008 at 4:11 AM, Andreas van dem Helge
>  <[EMAIL PROTECTED]> wrote:
>  > I think that's still a better idea than using a "dump the caller into
>  >  meetme" hack and is actually what I was going to suggest.
>  >
>  >  If you want something simpler than a queue then inject the sounds into
>  >  the moh already.
>  >
>  >
>  >
>  >  On Tue, Apr 1, 2008 at 3:09 PM, Rob Hillis <[EMAIL PROTECTED]> wrote:
>  >  >
>  >  >  You may be able to achieve the desired result using  queues rather than
>  >  > Dial statements.
>  >  >
>  >  >  Overkill perhaps, but it's the only way I can think to implement it at 
> the
>  >  > moment.
>  >  >
>  >  >
>  >  >
>  >  >
>  >  >  John Millican wrote:
>  >  >  Tilghman Lesher wrote:
>  >  >
>  >  >
>  >  >  On Tuesday 01 April 2008 05:14:25 Pete Kay wrote:
>  >  >
>  >  >
>  >  >  I am hoping someone can help me out on this. I want to be able to
>  >  > interrupt MOH every X seconds after the DIAL command is executed. The
>  >  > interrupt greeting is something like "please wait while we transfer your
>  >  > call". How can I do that? Within the DIAL options, I can't see any
>  >  > announce frequency or options that can help.
>  >  >
>  >  > Could anyone please tell me how that function can be accomplished?
>  >  >
>  >  >  The only way to do that currently is to implement the prompt within 
> the MOH
>  >  > stream itself.
>  >  >
>  >  >
>  >  >
>  >  > Just off the top-o-my head(YMMV), couldn't you create a meetme and play
>  >  > hold music into the meetme and then also play the prompt into the meetme
>  >  > at the same time without interrupting the hold music? This would
>  >  > obviously not work for high load but...
>  >  > JohnM
>  >  >
>  >  >
>  >  > ___
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>  >  >
>  >  > asterisk-users mailing list
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>  >  >
>  >  >
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>  >
>
>
>
>
> --
>  Atis Lezdins,
>  VoIP Project Manager / Developer,
>  [EMAIL PROTECTED]
>  Skype: atis.lezdins
>  Cell Phone: +371 28806004
>  Cell Phone: +1 800 7300689
>  Work phone: +1 800 7502835
>
>
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Re: [asterisk-users] zaptel alarm

[Shaun Ruffell]

Greg Woods wrote:
> I've been a happy user of asterisk for over a year just for a small home
> setup (a Digium TDM400P with one POTS line and three internal extensions
> plus a couple of SIP phones). I recently moved from running Fedora Core
> 6 running * 1.4.1 compiled from source and zaptel 1.4.7 to Fedora 8,
> using the RPM package for * 1.4.18 and zaptel 1.4.9 . It worked fine for
> a week, but suddenly I can't dial out any more. Incoming calls work
> fine, and outgoing calls through my Teliax line (IAX2) work fine, but I
> get a fast busy if I try to dial out through the land line. All that
> appears in the messages log is:
> 
> [Apr  2 07:13:48] WARNING[24249] chan_zap.c: Detected alarm on channel
> 4: No Alarm
> [Apr  2 07:13:48] NOTICE[24242] chan_zap.c: Alarm cleared on channel 4
> 
> "core set debug 3" doesn't give any more detail.
> 
> I have tried stopping asterisk, restarting zaptel (unloading and
> reloading), and starting asterisk, but the behavior persists.
> 
> Anybody know what this means? Has my hardware suddenly gone bad? Is
> there some way to debug this?
> 
> I apologize in advance for not knowing all the telephone terminology; as
> I said, this is just a small home setup.
> 
> --Greg


Hi Greg,

If you have subversion installed on your server, could you try using 
this version of zaptel:

http://svn.digium.com/svn/zaptel/branches/[EMAIL PROTECTED]

There have been some relatively recent changes with how certain signals 
are handled in zaptel and reported by asterisk.

Cheers,
Shaun


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Re: [asterisk-users] zaptel alarm

[Gonzalo Servat]

On Wed, Apr 2, 2008 at 10:28 AM, Greg Woods <[EMAIL PROTECTED]> wrote:

> I've been a happy user of asterisk for over a year just for a small home
> setup (a Digium TDM400P with one POTS line and three internal extensions
> plus a couple of SIP phones). I recently moved from running Fedora Core
> 6 running * 1.4.1 compiled from source and zaptel 1.4.7 to Fedora 8,
> using the RPM package for * 1.4.18 and zaptel 1.4.9 . It worked fine for
> a week, but suddenly I can't dial out any more. Incoming calls work
> fine, and outgoing calls through my Teliax line (IAX2) work fine, but I
> get a fast busy if I try to dial out through the land line. All that
> appears in the messages log is:
>
> [Apr  2 07:13:48] WARNING[24249] chan_zap.c: Detected alarm on channel
> 4: No Alarm
> [Apr  2 07:13:48] NOTICE[24242] chan_zap.c: Alarm cleared on channel 4
>
> "core set debug 3" doesn't give any more detail.
>
> I have tried stopping asterisk, restarting zaptel (unloading and
> reloading), and starting asterisk, but the behavior persists.
>

Hi Greg,

This might be a long shot, but I had this exact same problem recently and it
turned out to be a problem with the actual line. Have you tried plugging in
a phone directly to the line? Check that you have a dial tone and that you
can receive and make calls.

- Gonzalo
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Re: [asterisk-users] zaptel alarm

[Greg Woods]

On Wed, 2008-04-02 at 07:28 -0600, Greg Woods wrote:

> [Apr  2 07:13:48] WARNING[24249] chan_zap.c: Detected alarm on channel
> 4: No Alarm
> [Apr  2 07:13:48] NOTICE[24242] chan_zap.c: Alarm cleared on channel 4
> 
> "core set debug 3" doesn't give any more detail.
> 
> I have tried stopping asterisk, restarting zaptel (unloading and
> reloading), and starting asterisk, but the behavior persists.

Tacky as it is to answer myself, I also wanted to mention that I also
tried shutting down, powering off for 30 seconds, and rebooting, and
that doesn't fix the problem either. Also, dialing out on the POTS line
using a directly-connected phone (bypassing the * server) works fine.

--Greg



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Re: [asterisk-users] help with no audio

[Jerry Geis]

Jerry Geis wrote:
>>
>> On Tue, 2008-04-01 at 13:24 -0400, Jerry Geis wrote:
>> >/ I call into the dialplan and try to play demo-congrats and I hear 
>> nothing.
>> />/ />/ Firewall is disabled. />/ Everything is on the 192.168.1.X 
>> network for this simple configuration.
>> />/ The tftp server is giving the polycom phone the config files.
>> />/ />/ Any ideas why I dont hear audio?
>> /
>> Do you happen to have an unconfigured T1 card in your machine?  That's
>> the most common problem I see for people when they get no audio at all
>> coming out of Asterisk.  If that's not the case, I'd turn RTP debugging
>> on in the command-line and make sure RTP packets are coming and going
>> from the Asterisk box.
>>
>> -- 
>> Jared Smith
>> Community Relations Manager
>> Digium, Inc.
>
> Jared,
>
> I have no card in this unit at this time.
> lsmod shows ztdummy loaded.
>
> I turned on "rtp debug" and got a bunch of lines like:
> Got RTP packet from 192.168.1.99:2226 
>
> Does that help you?
>
> Jerry
>

Jared,

Using the "rtp debug" I noticed that when the phone has no audio all I 
see is:
Got RTP packet ...
Got RTP packet...

There are no Sent RTP packets..

under normal cases there is One sent and one Got:
Got RTP packet...
Sent RTP packet...

Why would asterisk not be sending RTP packets

I have no hardware card in this test system. Just two polycom IP330 phones.
Once in a great while I will hear audio when calling into the dialplan 
and playing demo-congrats.
95% of the time I hear NO audio though. I am using asterisk 1.4.18, 
libpri 1.4.3 and zaptel 1.4.9.2 (ztdummy is loaded)

Why might asterisk NOT be sending  RTP packets?

Jerry

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Re: [asterisk-users] CentPBX mirror?

[Darren Wright]

CentPBX has bit the dust I believe.
 
-D



From: [EMAIL PROTECTED] on behalf of Chris Bagnall
Sent: Wed 4/2/2008 7:12 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] CentPBX mirror?



Greetings list,

Not exclusively asterisk-related, but I've noticed the CentPBX site has been 
offline the last few days. Anyone know the reasoning behind that, and more 
importantly, is anyone mirroring it?

Thanks in advance.

Regards,

Chris
--
C.M. Bagnall, Director, Minotaur I.T. Limited
For full contact details visit http://www.minotaur.it  
This email is made from 100% recycled electrons





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This message was sent from D2 Technology, INC.

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[asterisk-users] zaptel alarm

[Greg Woods]

I've been a happy user of asterisk for over a year just for a small home
setup (a Digium TDM400P with one POTS line and three internal extensions
plus a couple of SIP phones). I recently moved from running Fedora Core
6 running * 1.4.1 compiled from source and zaptel 1.4.7 to Fedora 8,
using the RPM package for * 1.4.18 and zaptel 1.4.9 . It worked fine for
a week, but suddenly I can't dial out any more. Incoming calls work
fine, and outgoing calls through my Teliax line (IAX2) work fine, but I
get a fast busy if I try to dial out through the land line. All that
appears in the messages log is:

[Apr  2 07:13:48] WARNING[24249] chan_zap.c: Detected alarm on channel
4: No Alarm
[Apr  2 07:13:48] NOTICE[24242] chan_zap.c: Alarm cleared on channel 4

"core set debug 3" doesn't give any more detail.

I have tried stopping asterisk, restarting zaptel (unloading and
reloading), and starting asterisk, but the behavior persists.

Anybody know what this means? Has my hardware suddenly gone bad? Is
there some way to debug this?

I apologize in advance for not knowing all the telephone terminology; as
I said, this is just a small home setup.

--Greg



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[asterisk-users] misdn config warnings in Asterisk 1.4.18.1

[Vieri]

I would like to know if the following misdn warnings
are relevant.
Currently, I don't need echotraining. 
However, I took a quick look at the * source code and
l1watcher_timeout seems to be defined (echotraining
was not found). Currently I'm setting
l1watcher_timeout to 0 which is default (so I suppose
that this warning won't affect me).

Any comments on this?

*CLI> misdn reload
Reloading mISDN Config
  == Parsing '/etc/asterisk/misdn.conf': Found
[Apr  2 14:42:39] WARNING[13009]: misdn_config.c:930
_build_general_config: misdn.conf:
"l1watcher_timeout=0" (section: general) invalid or
out of range. Please edit your misdn.conf and then do
a "misdn reload".
[Apr  2 14:42:39] WARNING[13009]: misdn_config.c:986
_build_port_config: misdn.conf: "echotraining=no"
(section: default) invalid or out of range. Please
edit your misdn.conf and then do a "misdn reload".



  

You rock. That's why Blockbuster's offering you one month of Blockbuster Total 
Access, No Cost.  
http://tc.deals.yahoo.com/tc/blockbuster/text5.com

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Re: [asterisk-users] Howto connect to Cirpack softswitch withAsterisk ?

[Robert Rozman]

- Original Message - 
From: "Michiel van Baak" <[EMAIL PROTECTED]>
To: 
Sent: Wednesday, April 02, 2008 10:51 AM
Subject: Re: [asterisk-users] Howto connect to Cirpack softswitch 
withAsterisk ?


> On 10:11, Wed 02 Apr 08, Robert Rozman wrote:
>> Hi,
>>
>> has anyone connected Asterisk to Cirpack softswitch sucessfully ? Any 
>> howto
>> or more info about needed Asterisk SW and setup ?
>
> Yes, it works fine.
> Where do you get stuck ?
> It's basically a normal sip connection setup.
>
Hi,

thanks for response
I have it registered and receiveing incoming calls, but outgoing calls don't 
work. I'm attaching sip log below, the basic problem is that some sort of 
authentication is desired on outgoing calls...

Cirpack says: SIP/2.0 407 authentication required
and then
Cirpack says: SIP/2.0 403 Wrong login or password

I'm attaching full log below.. I'd kindly ask if someone can shed some 
light, where to specify outgoing authentication (I use freepbx also) ?

Can incoming calls be proceeded to ring local extensions without actually 
taking call (so ISP won't charge for just ringing) ?

Thanks in advance,

regards,

Rob.


SIP full log :

Really destroying SIP dialog '[EMAIL PROTECTED]' 
Method: REGISTER
-- Executing [EMAIL PROTECTED]:1] Macro("SIP/202-b654e668", 
"dialout-trunk|2|041461620||") in new stack
-- Executing [EMAIL PROTECTED]:1] Set("SIP/202-b654e668", 
"DIAL_TRUNK=2") in new stack
-- Executing [EMAIL PROTECTED]:2] Set("SIP/202-b654e668", 
"DIAL_NUMBER=041461620") in new stack
-- Executing [EMAIL PROTECTED]:3] Set("SIP/202-b654e668", 
"ROUTE_PASSWD=") in new stack
-- Executing [EMAIL PROTECTED]:4] GotoIf("SIP/202-b654e668", 
"1?noauth") in new stack
-- Goto (macro-dialout-trunk,s,6)
-- Executing [EMAIL PROTECTED]:6] GotoIf("SIP/202-b654e668", 
"0?disabletrunk|1") in new stack
-- Executing [EMAIL PROTECTED]:7] Set("SIP/202-b654e668", 
"_NODEST=") in new stack
-- Executing [EMAIL PROTECTED]:8] Set("SIP/202-b654e668", 
"DIAL_TRUNK_OPTIONS=tr") in new stack
-- Executing [EMAIL PROTECTED]:9] Set("SIP/202-b654e668", 
"GROUP()=OUT_2") in new stack
-- Executing [EMAIL PROTECTED]:10] Macro("SIP/202-b654e668", 
"user-callerid|SKIPTTL") in new stack
-- Executing [EMAIL PROTECTED]:1] NoOp("SIP/202-b654e668", 
"user-callerid: device 202") in new stack
-- Executing [EMAIL PROTECTED]:2] Set("SIP/202-b654e668", 
"AMPUSER=202") in new stack
-- Executing [EMAIL PROTECTED]:3] GotoIf("SIP/202-b654e668", 
"0?report") in new stack
-- Executing [EMAIL PROTECTED]:4] GotoIf("SIP/202-b654e668", 
"0?start") in new stack
-- Executing [EMAIL PROTECTED]:5] Set("SIP/202-b654e668", 
"REALCALLERIDNUM=202") in new stack
-- Executing [EMAIL PROTECTED]:6] NoOp("SIP/202-b654e668", 
"REALCALLERIDNUM is 202") in new stack
-- Executing [EMAIL PROTECTED]:7] Set("SIP/202-b654e668", 
"AMPUSER=202") in new stack
-- Executing [EMAIL PROTECTED]:8] Set("SIP/202-b654e668", 
"AMPUSERCIDNAME=pl_51") in new stack
-- Executing [EMAIL PROTECTED]:9] GotoIf("SIP/202-b654e668", 
"0?report") in new stack
-- Executing [EMAIL PROTECTED]:10] Set("SIP/202-b654e668", 
"AMPUSERCID=202") in new stack
-- Executing [EMAIL PROTECTED]:11] Set("SIP/202-b654e668", 
"CALLERID(all)="pl_51" <202>") in new stack
-- Executing [EMAIL PROTECTED]:12] Set("SIP/202-b654e668", 
"REALCALLERIDNUM=202") in new stack
-- Executing [EMAIL PROTECTED]:13] NoOp("SIP/202-b654e668", "TTL: 
ARG1: SKIPTTL") in new stack
-- Executing [EMAIL PROTECTED]:14] GotoIf("SIP/202-b654e668", 
"1?continue") in new stack
-- Goto (macro-user-callerid,s,23)
-- Executing [EMAIL PROTECTED]:23] NoOp("SIP/202-b654e668", "Using 
CallerID "pl_51" <202>") in new stack
-- Executing [EMAIL PROTECTED]:11] Macro("SIP/202-b654e668", 
"record-enable|202|OUT") in new stack
-- Executing [EMAIL PROTECTED]:1] GotoIf("SIP/202-b654e668", 
"0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [EMAIL PROTECTED]:4] AGI("SIP/202-b654e668", 
"recordingcheck|20080402-143454|1207139694.24") in new stack
-- Launched AGI Script /usr/share/asterisk/agi-bin/recordingcheck
  recordingcheck|20080402-143454|1207139694.24: Outbound recording not 
enabled
-- AGI Script recordingcheck completed, re

Re: [asterisk-users] interrupting MOH

[Atis Lezdins]

Sorry for top-posting, but seems everyone on this thread did so.

Also that would be my suggestion for now - call queue with periodic-announce.

However i see that this would make nice architectural improvement -
allow inject sound files into MoH stream. This would be useful for
example in call queues - to inject all the queue announcements into
MoH directly, rather than play them while blocking further queue
actions.

Regards,
Atis


On Wed, Apr 2, 2008 at 4:11 AM, Andreas van dem Helge
<[EMAIL PROTECTED]> wrote:
> I think that's still a better idea than using a "dump the caller into
>  meetme" hack and is actually what I was going to suggest.
>
>  If you want something simpler than a queue then inject the sounds into
>  the moh already.
>
>
>
>  On Tue, Apr 1, 2008 at 3:09 PM, Rob Hillis <[EMAIL PROTECTED]> wrote:
>  >
>  >  You may be able to achieve the desired result using  queues rather than
>  > Dial statements.
>  >
>  >  Overkill perhaps, but it's the only way I can think to implement it at the
>  > moment.
>  >
>  >
>  >
>  >
>  >  John Millican wrote:
>  >  Tilghman Lesher wrote:
>  >
>  >
>  >  On Tuesday 01 April 2008 05:14:25 Pete Kay wrote:
>  >
>  >
>  >  I am hoping someone can help me out on this. I want to be able to
>  > interrupt MOH every X seconds after the DIAL command is executed. The
>  > interrupt greeting is something like "please wait while we transfer your
>  > call". How can I do that? Within the DIAL options, I can't see any
>  > announce frequency or options that can help.
>  >
>  > Could anyone please tell me how that function can be accomplished?
>  >
>  >  The only way to do that currently is to implement the prompt within the 
> MOH
>  > stream itself.
>  >
>  >
>  >
>  > Just off the top-o-my head(YMMV), couldn't you create a meetme and play
>  > hold music into the meetme and then also play the prompt into the meetme
>  > at the same time without interrupting the hold music? This would
>  > obviously not work for high load but...
>  > JohnM
>  >
>  >
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-- 
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835

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Re: [asterisk-users] show uptime and last reload

[Vieri]

--- Michiel van Baak <[EMAIL PROTECTED]> wrote:

> On 01:40, Wed 02 Apr 08, Vieri wrote:
> > Hi,
> > 
> > I just upgraded from 1.2 to 1.4.
> > 
> > In 1.2, when I did a "show uptime" I used to see a
> > second line telling me the time since the last
> reload.
> > Has this been removed in 1.4?
> > 
> > The following is the output of my two test boxes:
> > 
> > Connected to Asterisk 1.4.18.1 currently running
> on
> > voip2 (pid = 10605)
> > Verbosity is at least 3
> > voip2*CLI> show uptime
> > System uptime: 15 hours, 55 seconds
> > voip2*CLI> sip reload
> >  Reloading SIP
> >   == Parsing '/etc/asterisk/sip.conf': Found
> > voip2*CLI> show uptime
> > System uptime: 15 hours, 1 minute, 28 seconds
> > voip2*CLI>
> > 
> > -
> > 
> > Connected to Asterisk 1.2.27 currently running on
> > voip1 (pid = 26496)
> > -- Remote UNIX connection
> > Verbosity is at least 3
> > voip1*CLI> show uptime
> > System uptime: 4 days, 23 hours, 55 minutes, 1
> second
> > Last reload: 1 day, 4 minutes, 23 seconds
> > voip1*CLI>
> 
> Can you try with: reload
> instead of just a sip reload ?

Right... That did it.
Sorry for the dumb question.
Thanks!



  

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Access, No Cost.  
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Re: [asterisk-users] TDM410E card, 1 FXO module - how to dial Out

[Kevin P. Fleming]

Mojo with Horan & Company, LLC wrote:

> P.S.  If you can't dial seven digit numbers in your area, but you miss 
> it, you can restore that behavior if you feel like selecting a default 
> area code:
> 
> exten => _NXX,1,Dial(Zap/1/907${EXTEN},,TWK)
> 
> Here, if I dial a seven digit number, asterisk dials 907 followed by my 
> seven digits out the phone line.

Well, sort of. This will also trigger if you dial the first 7 digits of
a 10-digit number from a device that doesn't dial 'en bloc', since there
is no longer any way to distinguish 7-vs-10 digit numbers by the number
pattern. In other words, this will work fine if you are dialing from a
SIP phone, but not if you are dialing from an analog phone.

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

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[asterisk-users] CentPBX mirror?

[Chris Bagnall]

Greetings list,

Not exclusively asterisk-related, but I've noticed the CentPBX site has been 
offline the last few days. Anyone know the reasoning behind that, and more 
importantly, is anyone mirroring it?

Thanks in advance.

Regards,

Chris
-- 
C.M. Bagnall, Director, Minotaur I.T. Limited
For full contact details visit http://www.minotaur.it
This email is made from 100% recycled electrons





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Re: [asterisk-users] How to Hangup after DIAL is completed

[Pete Kay]

Dear Tony,

Thank you very much for your suggestion.
The thing is that ${DIALSTATUS} returns ANSWER regardless of whether the
called party hangs up or not.
For instance, when called party is being asked by Asterisk whether he/she
wants to pick the call, ${DIALSTATUS} returns ANSWER.

In the case where the called party accepts to take the call and then hangs
up after finishing the conversion, I want to stop AGI from jumping to call
the next number.  On the other hand, if the called party decides not to take
the call, I want to let AGI to jump to the next number.

It seems like ${DIALSTATUS} does not give me the info that I need.

Thank you very much.  Is there any other ways you may think of?

Thanks,
Pete


On Wed, Apr 2, 2008 at 3:03 PM, Pete Kay <[EMAIL PROTECTED]> wrote:

> Hi,
>
> I have a problem with DIAL.
> The scenario is this:
>
> 1. Asterisk will dial a number in a call list
> 2. called party picks up the call and hears a prompt asking if they want
> to pick up ( this is done through M(marco) option in DIAL)
> 3. if called party does not want to pick up, go to the next number (this
> is done through SET MARCO_RESULT= CONTINUE)
> 4. if called party decides to take the call, make some conversion, then
> the call is completed and hanged up by the called party.
>
> For 4, when the called party hangs up, I want to hang up on the calling
> party as well.
>
> This works if I use the dialplan by not including the "g" option in DIAL.
> However, when I do the DIAL in AGI, the calling party does not get hangup
> automatically at (4).  So, it continues to execute the next line in the AGI,
> which is not what I want.
>
> The problem is that the result I can when executing the DIAL command
> within AGI does not tell me whether the call was picked up or not. So, I
> have no way of knowing whether to continue executing the AGI program or to
> issue a HAGNGUP explicitly.
>
> Can anyone please help me ?
>
> Any suggestion will be greatly appreciated.
>
> Thanks,
> Pete
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On Wed, Apr 2, 2008 at 4:39 PM, Tony Mountifield <[EMAIL PROTECTED]>
wrote:

> In article <[EMAIL PROTECTED]>,
> Pete Kay <[EMAIL PROTECTED]> wrote:
> > I have a problem with DIAL.
> > The scenario is this:
> >
> > 1. Asterisk will dial a number in a call list
> > 2. called party picks up the call and hears a prompt asking if they want
> to
> > pick up ( this is done through M(marco) option in DIAL)
> > 3. if called party does not want to pick up, go to the next number (this
> is
> > done through SET MARCO_RESULT= CONTINUE)
> > 4. if called party decides to take the call, make some conversion, then
> the
> > call is completed and hanged up by the called party.
> >
> > For 4, when the called party hangs up, I want to hang up on the calling
> > party as well.
> >
> > This works if I use the dialplan by not including the "g" option in
> DIAL.
> > However, when I do the DIAL in AGI, the calling party does not get
> hangup
> > automatically at (4).  So, it continues to execute the next line in the
> AGI,
> > which is not what I want.
> >
> > The problem is that the result I can when executing the DIAL command
> within
> > AGI does not tell me whether the call was picked up or not. So, I have
> no
> > way of knowing whether to continue executing the AGI program or to issue
> a
> > HAGNGUP explicitly.
>
> When you have returned from the Dial command, check the DIALSTATUS channel
> variable. If the call was picked up, ${DIALSTATUS} will contain ANSWER.
>
> Cheers
> Tony
> --
> Tony Mountifield
> Work: [EMAIL PROTECTED] - http://www.softins.co.uk
> Play: [EMAIL PROTECTED] - http://tony.mountifield.org
>
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Re: [asterisk-users] Virtual or Hardware SIP Modem

[Steve Davies]

You can get much better results (close to 56k reliable connections
sometimes) by using a Xorcom FXO Channelbank - You need recent enough
drivers so that the Xorcom internal clock can be synced to Zaptel;
This removes/reduces jitter and frame slippage, and allows a modem to
operate much more reliably.

Regards,
Steve

On 02/04/2008, Don Fanning <[EMAIL PROTECTED]> wrote:
> Short answer: No.
>
>  However you can use ATA devices like PAP-2's to connect to your existing
>  modem bank and as long as your latency is constant, get decent results.
>  I myself have gotten 33.6k on a regular basis with such a setup and have
>  called the world using cheap SIP/IAX providers with decent speeds.
>
>  The key to note is that you disable Data Compression (AT&K0) because the
>  data stream is already compressed.  Error correction however is useful.
>
>
>
>  Kyle Gibbons wrote:
>  > Hi,
>  >
>  > I have just gotten my first Asterisk box up and running, and it is
>  > running great. I am working on this project with the plans of possibly
>  > implementing it in a business environment. The problem I am coming up
>  > against is that the business I am planning on implementing this setup
>  > in is using some legacy software which requires a modem to communicate
>  > with energy management systems. My question is if there is a virtual
>  > or physical SIP modem that I could possibly use so that I can
>  > interface this old software with Asterisk. There is no option of
>  > getting rid of modems all together. I would prefer not to use Zap
>  > cards or other adapters for the current modems. My goal is to
>  > completly replace the modems with software. Any help would be GREATLY
>  > appreciated. Thanks!
>  >
>  > --
>  > All the best,
>  > Kyle
>  >
>

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Re: [asterisk-users] g729 encoder/decoder

[Thomas Kenyon]

Peder @ NetworkOblivion wrote:
> That makes sense.  A call from 729 to 711 would require one encoder and 
> one decoder, right?
> 
> So if you have 10 licenses, is it 10 total encoders+decoders, or 10 
> calls (some may require encode, or decode, or both)?  Because I had 10 
> licenses, but my encoders+decoders was more than 10 and calls worked 
> fine.  However I also ran out of licenses when neither number was >=10.
> 
1 license = 1 encoder + 1 decoder.

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Re: [asterisk-users] show uptime and last reload

[Michiel van Baak]

On 01:40, Wed 02 Apr 08, Vieri wrote:
> Hi,
> 
> I just upgraded from 1.2 to 1.4.
> 
> In 1.2, when I did a "show uptime" I used to see a
> second line telling me the time since the last reload.
> Has this been removed in 1.4?
> 
> The following is the output of my two test boxes:
> 
> Connected to Asterisk 1.4.18.1 currently running on
> voip2 (pid = 10605)
> Verbosity is at least 3
> voip2*CLI> show uptime
> System uptime: 15 hours, 55 seconds
> voip2*CLI> sip reload
>  Reloading SIP
>   == Parsing '/etc/asterisk/sip.conf': Found
> voip2*CLI> show uptime
> System uptime: 15 hours, 1 minute, 28 seconds
> voip2*CLI>
> 
> -
> 
> Connected to Asterisk 1.2.27 currently running on
> voip1 (pid = 26496)
> -- Remote UNIX connection
> Verbosity is at least 3
> voip1*CLI> show uptime
> System uptime: 4 days, 23 hours, 55 minutes, 1 second
> Last reload: 1 day, 4 minutes, 23 seconds
> voip1*CLI>

Can you try with: reload
instead of just a sip reload ?
-- 

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

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


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Re: [asterisk-users] Howto connect to Cirpack softswitch with Asterisk ?

[Michiel van Baak]

On 10:11, Wed 02 Apr 08, Robert Rozman wrote:
> Hi,
> 
> has anyone connected Asterisk to Cirpack softswitch sucessfully ? Any howto 
> or more info about needed Asterisk SW and setup ?

Yes, it works fine.
Where do you get stuck ?
It's basically a normal sip connection setup.

-- 

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

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


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Re: [asterisk-users] How to Hangup after DIAL is completed

[Tony Mountifield]

In article <[EMAIL PROTECTED]>,
Pete Kay <[EMAIL PROTECTED]> wrote:
> I have a problem with DIAL.
> The scenario is this:
> 
> 1. Asterisk will dial a number in a call list
> 2. called party picks up the call and hears a prompt asking if they want to
> pick up ( this is done through M(marco) option in DIAL)
> 3. if called party does not want to pick up, go to the next number (this is
> done through SET MARCO_RESULT= CONTINUE)
> 4. if called party decides to take the call, make some conversion, then the
> call is completed and hanged up by the called party.
> 
> For 4, when the called party hangs up, I want to hang up on the calling
> party as well.
> 
> This works if I use the dialplan by not including the "g" option in DIAL.
> However, when I do the DIAL in AGI, the calling party does not get hangup
> automatically at (4).  So, it continues to execute the next line in the AGI,
> which is not what I want.
> 
> The problem is that the result I can when executing the DIAL command within
> AGI does not tell me whether the call was picked up or not. So, I have no
> way of knowing whether to continue executing the AGI program or to issue a
> HAGNGUP explicitly.

When you have returned from the Dial command, check the DIALSTATUS channel
variable. If the call was picked up, ${DIALSTATUS} will contain ANSWER.

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org

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[asterisk-users] show uptime and last reload

[Vieri]

Hi,

I just upgraded from 1.2 to 1.4.

In 1.2, when I did a "show uptime" I used to see a
second line telling me the time since the last reload.
Has this been removed in 1.4?

The following is the output of my two test boxes:

Connected to Asterisk 1.4.18.1 currently running on
voip2 (pid = 10605)
Verbosity is at least 3
voip2*CLI> show uptime
System uptime: 15 hours, 55 seconds
voip2*CLI> sip reload
 Reloading SIP
  == Parsing '/etc/asterisk/sip.conf': Found
voip2*CLI> show uptime
System uptime: 15 hours, 1 minute, 28 seconds
voip2*CLI>

-

Connected to Asterisk 1.2.27 currently running on
voip1 (pid = 26496)
-- Remote UNIX connection
Verbosity is at least 3
voip1*CLI> show uptime
System uptime: 4 days, 23 hours, 55 minutes, 1 second
Last reload: 1 day, 4 minutes, 23 seconds
voip1*CLI>



  

You rock. That's why Blockbuster's offering you one month of Blockbuster Total 
Access, No Cost.  
http://tc.deals.yahoo.com/tc/blockbuster/text5.com

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Re: [asterisk-users] Howto connect to Cirpack softswitch with Asterisk ?

[Grey Man]

On Wed, Apr 2, 2008 at 9:11 AM, Robert Rozman <[EMAIL PROTECTED]> wrote:
> Hi,
>
>  has anyone connected Asterisk to Cirpack softswitch sucessfully ? Any howto
>  or more info about needed Asterisk SW and setup ?
>
>  Thanks in advance,
>
>  regards,
>
>  Rob.

We have a trunk supplier that uses a Cirpack switch. There weren't any
real problems with the SIP side of things, I thin we used the P-Assert
headers for callerid. There was an issue early on with getting
progress tones as the Cirpack wouldn't sned the progress audio without
first getting an RTP packet and Asterisk wouldn't send an RTP packet
until the call was established (rightly so in Asterisk's case). That
issue seemed to get fixed in an upgrade on the Cirpack firmware at
some point since our interconnect was put in.

Regards,

Greyman.

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[asterisk-users] Howto connect to Cirpack softswitch with Asterisk ?

[Robert Rozman]

Hi,

has anyone connected Asterisk to Cirpack softswitch sucessfully ? Any howto 
or more info about needed Asterisk SW and setup ?

Thanks in advance,

regards,

Rob.


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Re: [asterisk-users] Digium B410P, bristuff and BRI support in 1.6

[Olivier]

2008/4/1, Jean-Denis Girard <[EMAIL PROTECTED]>:
>
> Hi,
>
> Olivier a écrit :
>
> > Hi,
> >
> > Is it possible to both use Digium B410P and bristuff'ed 1.4 Asterisk,
> now ?
> >
> > I've heard BRI support in Asterisk is about to change with 1.6 but I'm
> > not sure I understood what the plan is.
> > If someone has a clue, l would delighted to learn about it.
>
>
> You should have a look at this thread:
> http://lists.digium.com/pipermail/asterisk-dev/2008-March/032142.html


That is the thread I was thinking about and that I'm not exactly sure to
understand.

I'm not sure about the B410, but my home asterisk system has been
> running fine for nearly a month with libpri-trunk, asterisk-1.6.0,
> zaptel-1.4 (patched) and zaphfc.


Would you mind if I asked you this :
- Which card did you include in your home system ? Are you using an ISDN BRI
access ?
- Is libpri necessary for ISDN BRI access ? I thought libpri was mostly
dedicated to E1/T1 access

Quoting previously mentioned thread :

Kevin P. Fleming wrote:
>* Klaus Darilion wrote:
*>*
*>>* I just saw some SIG_BRI... definitions in chan_zap.c of 1.6. Ist it now
*>>* possible to use BRI cards without patching with bristuff?
*>*
*>* Not quite yet... but soon it should be possible, yes.
*>*
*
Actually...

It'll probably work.  I have tested making basic phone calls, etc over
it, although I'm still tweaking it and making sure that the Q.921 layer
is more likely to pass testing.  Of course, YMMV.  I certainly don't
mind testers though trying it out and reporting issues that they find
with it.  That would help me out a lot.  You'll have to update your
libpri to trunk for it to work though.  That's where all the BRI
development is happening.  And NT-PTMP is not going to work, only NT/TE
PTP and TE PTMP.

-- 
Matthew Fredrickson
Software/Firmware Engineer
Digium, Inc.



>From above, do you understand that Digium is committed to support BRI cards
in 1.6 ?
If positive, which cards will be supported and with which feature set ?

Thanks

I have the following error in system logs, but it seems harmless.
> Apr  1 09:16:59 tiare kernel: zaphfc: dropped audio (z1=6992, z2=6958,
> wanted 8 got 34, dropped 26).
>
> Quality is great (no echo).
>
>
> Thanks,
> --
> Jean-Denis Girard
>
> SysNux  Systèmes  Linux  en Polynésie française
> http://www.sysnux.pf/   Tél: +689 50 10 40 / GSM: +689 79 75 27
>
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[asterisk-users] How to Hangup after DIAL is completed

[Pete Kay]

Hi,

I have a problem with DIAL.
The scenario is this:

1. Asterisk will dial a number in a call list
2. called party picks up the call and hears a prompt asking if they want to
pick up ( this is done through M(marco) option in DIAL)
3. if called party does not want to pick up, go to the next number (this is
done through SET MARCO_RESULT= CONTINUE)
4. if called party decides to take the call, make some conversion, then the
call is completed and hanged up by the called party.

For 4, when the called party hangs up, I want to hang up on the calling
party as well.

This works if I use the dialplan by not including the "g" option in DIAL.
However, when I do the DIAL in AGI, the calling party does not get hangup
automatically at (4).  So, it continues to execute the next line in the AGI,
which is not what I want.

The problem is that the result I can when executing the DIAL command within
AGI does not tell me whether the call was picked up or not. So, I have no
way of knowing whether to continue executing the AGI program or to issue a
HAGNGUP explicitly.

Can anyone please help me ?

Any suggestion will be greatly appreciated.

Thanks,
Pete
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