RE : RE : RE : [Asterisk-Users] What does it mean?

[Olivier Taylor]

Logger.conf seems to have nothing to see with my problem

Olivier

-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de harry gaillac
Envoyé : jeudi 24 novembre 2005 23:25
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : RE: RE : RE : [Asterisk-Users] What does it mean?


Je ne donne pas de réponse !
Il me semble t'avoir suggèrer asterisk comme système
de messagerie vocale au lieu d'SEMS, avoir fourni
quelques fichiers de configuration, ce n'étaient pas
des devinettes.

Conbien de fois on ma répondu "personne n'est obligé
de faire ton tavail, tu n'as qu'a payé pour ce que tu
demandes.

IL me semble même me souvenir avoir lu un développeur
te faire la remarque "les utilisateurs de nos projets
vous ne profitez que de notre travail !".


Pour répondre à ton problème configure logger.conf .

Harry

  
--- Olivier Taylor <[EMAIL PROTECTED]> a écrit
:

> Cela veut simplement dire que tu te plains de ne pas
> avoir de réponses, mais
> qu'en fait tu n'en donnes pas non plus, sauf sous
> forme de devinette.
> Auquel cas, il est plus simple de ne pas répondre,
> 
> merci
> 
> -Message d'origine-
> De : [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] De
> la part de harry gaillac
> Envoyé : jeudi 24 novembre 2005 17:54
> À : Asterisk Users Mailing List - Non-Commercial
> Discussion
> Objet : RE: RE : [Asterisk-Users] What does it mean?
> 
> 
> Je ne connais pas la signification de "sybillines".
> Harry
> --- Olivier Taylor <[EMAIL PROTECTED]> a
> écrit
> :
> 
> > Tes réponses sont aussi sybillines que tes
> questions
> > :)
> > 
> > Olivier
> > 
> > -Message d'origine-
> > De : [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED]
> De
> > la part de harry gaillac
> > Envoyé : jeudi 24 novembre 2005 16:45
> > À : Asterisk Users Mailing List - Non-Commercial
> > Discussion
> > Objet : RE: [Asterisk-Users] What does it mean?
> > 
> > 
> > Hello,
> > 
> > Read the Makefile in apps.
> > Harry
> > --- Olivier Taylor <[EMAIL PROTECTED]> a
> > écrit
> > :
> > 
> > > Hello,
> > > 
> > > I have compiled asterisk cvs under freebsd, no
> > > problems.
> > > 
> > > When starting asterisk, I get :
> > > 
> > > [res_config_mysql.so] => (MySQL RealTime
> > > Configuration Driver)
> > > /libexec/ld-elf.so.1:
> > > /usr/lib/asterisk/modules/res_config_mysql.so:
> > > Undefined symbol "ast_config_load"
> > > 
> > > What's wrong?
> > > 
> > > Olivier
> > > 
> > > ___
> > > --Bandwidth and Colocation sponsored by
> > Easynews.com
> > > --
> > > 
> > > Asterisk-Users mailing list Asterisk-Users@lists.digium.com
> > >
> >
>
http://lists.digium.com/mailman/listinfo/asterisk-users
> > > To UNSUBSCRIBE or update options visit:
> > >   
> > >
> >
>
http://lists.digium.com/mailman/listinfo/asterisk-users
> > > 
> > 
> > 
> > 
> > 
> > 
> > 
> > 
> >
>
___
> > 
> > Appel audio GRATUIT partout dans le monde avec le
> > nouveau Yahoo! Messenger
> > Téléchargez cette version sur
> > http://fr.messenger.yahoo.com 
> > ___
> > --Bandwidth and Colocation sponsored by
> Easynews.com
> > --
> > 
> > Asterisk-Users mailing list
> > Asterisk-Users@lists.digium.com
> >
>
http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> >   
> >
>
http://lists.digium.com/mailman/listinfo/asterisk-users
> > 
> > ___
> > --Bandwidth and Colocation sponsored by
> Easynews.com
> > --
> > 
> > Asterisk-Users mailing list
> > Asterisk-Users@lists.digium.com
> >
>
http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> >   
> >
>
http://lists.digium.com/mailman/listinfo/asterisk-users
> > 
> 
> 
> 
>   
> 
>   
>   
>
___
> 
> Appel audio GRATUIT partout dans le monde avec le
> nouveau Yahoo! Messenger
> Téléchargez cette version sur
> http://fr.messenger.yahoo.com
> ___
> --Bandwidth and Colocation sponsored by Easynews.com
> --
> 
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
>
http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>   
>
http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> ___
> --Bandwidth and Colocation sponsored by Easynews.com
> --
> 
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
>
http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>   
>
http://lists.digium.com/mailman/listinfo/asterisk-users
> 







___ 
Appel audio GRATUIT partout dans le monde avec le nouveau 

Re: [Asterisk-Users] Lag in speech

[Tim Stoop]

Hi Thor,


On 11/24/05, Thor Atle Rustad <[EMAIL PROTECTED]> wrote:
> --- /root/hfc_pci.c Wed Aug  7 15:31:24 2002
> +++ /usr/src/linux/drivers/isdn/hisax/hfc_pci.c Thu Oct 31 10:18:05 2002

Ah, so it's a problem in the driver, not Asterisk. Thanks for clearing
that up. I'll be trying this patch.

--
Gegroet,
Tim
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

Re: [Asterisk-Users] Pros and Cons of T1/E1 cards

[Jean-Michel Hiver]

AVM/ CAPI
eIcon  / CAPI
Junghanns  / Bristuff


As far as I'm aware, CAPI and Bristuff are BRI card so you can strike 
those off.


___
--Bandwidth and Colocation sponsored by Easynews.com --

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

[Asterisk-Users] Re: Bad quality...

[Mauro Zanin]

Hi Pablo,
try to adjust volume on both Loudspeaker and Microphone if you are using a
Softphone: a saturated circuit gives bad results...

Ciao
Mauro
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

Re: [Asterisk-Users] PRI problems again - What should I do ?

[tim panton]

On 25 Nov 2005, at 07:51, Julian Lyndon-Smith wrote:Thanks for your help Tim:Comments inline:tim panton wrote: On 24 Nov 2005, at 10:26, Julian Lyndon-Smith wrote: I know that's a real newbie question, but I have a problem.I keep getting frame rejects, and a D-channel bouncing up and down. BT say that it is at my end. If I stop asterisk, stop the zaptel service and restart, things seem ok for a while. Pardon me for asking the obvious, but...Have you _double_ checked the timing params in /etc/zaptel.conf?your bt span should say something like:span=1,1,0,ccs,hdb3or perhapsspan=1,1,0,ccs,hdb3,crc4 my settings are (and have been for over a year now)Oh, in that case, the only things I can think of would be to swap the cables, and if that didn't helpcall BT and ask them to tell you _exactly_ what they see. You may find that your line hasdrifted off spec.Tim.http://www.westhawk.co.uk/  ___
--Bandwidth and Colocation sponsored by Easynews.com --

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

Re: [Asterisk-Users] Multiple HFC-PCI cards in mixed modes (TE+NT) won't work!...

[Kristof Hardy]

Francesco Peeters wrote:

I compiled 1.2 and bristuff 0.3.0 Pre1 yesterday late and that now seems
to work! * is up and running *with* 2nd card in NT mode...


Nice to hear *1.2 and bristuff 0.3pre1 makes a difference..

cheers
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

Re: [Asterisk-Users] Lag in speech

[Alejandro Vargas]

2005/11/24, Thor Atle Rustad <[EMAIL PROTECTED]>:
> --- /root/hfc_pci.c Wed Aug  7 15:31:24 2002
> +++ /usr/src/linux/drivers/isdn/hisax/hfc_pci.c Thu Oct 31 10:18:05 2002
> @@ -270,8 +270,16 @@

This patch if from 2002. I just checked the source of my 2.6 kernel
and the driver stills unpatched. ¿Didn't anybody send this patch to
the kernel list?


--
Alejandro Vargas
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

Re: [Asterisk-Users] PRI problems again - What should I do ?

[Julian Lyndon-Smith]

tim panton wrote:
[Snip]





Pardon me for asking the obvious, but...
Have you _double_ checked the timing params in /etc/zaptel.conf?
your bt span should say something like:
span=1,1,0,ccs,hdb3
or perhaps
span=1,1,0,ccs,hdb3,crc4


my settings are (and have been for over a year now)


Oh, in that case, the only things I can think of would be to swap the 
cables, 


Yup, did that :)

and if that didn't help

Nope, it didn't :(

call BT and ask them to tell you _exactly_ what they see. You may find 
that your line has

drifted off spec.


They claim that there is nothing wrong and that it is my equipment that 
is the problem. I asked them to reset the line last night, so I'll see 
what happens today. Also sent a rocket up the backside of my salesrep, 
so perhaps that may achieve something ...


Tim.

http://www.westhawk.co.uk/


Julian.







___
--Bandwidth and Colocation sponsored by Easynews.com --

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


___
--Bandwidth and Colocation sponsored by Easynews.com --

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

Re: [Asterisk-Users] Re: sip URL peering

[Klaus Darilion]

Wolfgang S. Rupprecht wrote:

Klaus Darilion <[EMAIL PROTECTED]> writes:


It's not that easy. If you want to have open SIP URIs (just like email
is open for everybody) you will receive SPIT calls. E.g. the SPEER
group tries to define rules for VoIP peering which allows
authentication to enable open SIP URIs. (I won't open acces to my SIP
URI if I can not verify the senders URI).



Keeping spam in mind seems like a really good idea.  I'm also a big
fan of keeping a cryptographic "paper trail" so that one can figure
out who spammed.

On the other hand, is SPAM / SPIT a big enough problem at this point
to warrant scuttling any interconnectivity?  It seems a bit premature
to worry about a problem that may not develop for 5 years and allow
that fear to stop direct sip dialing.


Then we will have the same problem with email. Sometimes it's better to 
learn and try to design new services with potential problems in mind. 
With asterisk it's quite easy to make automated marketing calls. I'm 
sure as soon as more people are reachable directly via SIP, you will get 
marketing calls also via SIP. Then, the service providers will have to 
reject incoming calls and think about solutions. I prefer starting with 
solution to avoid disturbing users.




As an amusing aside, I inadvertently added a "captcha" to my phone
line when I had the local number go into an IVR that asks the caller
to press 1 for person XXX and 2 for person YYY and 3 of they are a
telemarketer.  I don't think anyone other than my friends has ever
pressed 3, but the predictive dialers used by the phone-spammers
doesn't seem to pass the turing test and isn't able to press 1 or 2.
;-) I see lots of timeout-hangups in the IVR with caller-id's like
"CAR PROMO" or "VOIP CALL".

If spam/spit is ever a problem, I'm simply routing previously unseen
calls to a turing test of the same type and anyone that has previously
called (and/or been called) gets to bypass the turing test.


That for sure will work for geeks like you and me. But if you consider 
the a telco which allows to reach all customers via SIP, the detection 
should happen in the beginning. Also if my grandma calls me and she 
hears "press 1 for ..." probably she will hang again.


regards
klaus

___
--Bandwidth and Colocation sponsored by Easynews.com --

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

Re: [Asterisk-Users] PRI problems again - What should I do ?

[Klaus Darilion]

Krishna Sumanth Chava wrote:

Hi Julian,
 
I think the Dell poweredge2850 servers are not too compatible with the 
zaptel cards..


but they are suggested by digium:
http://www.digium.com/index.php?menu=product_detail&category=software&product=ABE&tab=compatibility

klaus

 
Thanks

krishna

 
On 11/24/05, *Julian Lyndon-Smith* <[EMAIL PROTECTED] 
> wrote:


I know that's a real newbie question, but I have a problem.

I keep getting frame rejects, and a D-channel bouncing up and down. BT
say that it is at my end. If I stop asterisk, stop the zaptel service
and restart, things seem ok for a while.

I posted a similar problem a couple of days ago, and one of the
responses suggested that the TE4xxP may be on it's way out.

Is there any way of testing this card to see if that may be the case ?

I was thinking of buying a sangoma a102 as a fall-over - are there any
issues with the sangoma cards, or should I buy another te4xxp as a
backup ?

I was also thinking of moving the * server to a dell 2850 (2x3.06
processors, 2GB ram, 2x146gb hdd) - again, any gotchas ?

Sorry for so many questions, but we are placing / receiving near on 3000
calls a day now and my butt is getting sore from all the kicking I've
received :)

Many thanks for the anticipated (and needed) help :)

Julian
___
--Bandwidth and Colocation sponsored by Easynews.com
 --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com 
http://lists.digium.com/mailman/listinfo/asterisk-users

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





___
--Bandwidth and Colocation sponsored by Easynews.com --

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


___
--Bandwidth and Colocation sponsored by Easynews.com --

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

[Asterisk-Users] has someone zaphfc with xenomai working?

[Gerald Dachs]

Hi,

build worked just fine, had only to change rt_get_time to rt_get_time_ns,
according to the xenomai guys this is the same in xenomai. After loading
the  zaphfc and the realtime modules the realtime interrupts increase. The
hfc-s card is found and everything shows up fine in /proc/zaptel/1, but
zttest
does not show any throughput. Does someone work with this configuration?

Regards
Gerald

___
--Bandwidth and Colocation sponsored by Easynews.com --

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

Re: [Asterisk-Users] Grandstream problem

[Paul Hewlett]

On Friday 25 November 2005 01:45, Alfie Viechweg wrote:
> Can some on help me find the problem here please:
> I'm using asterisk 1.2.0 with Grandstream GXP-2000
>
> This is the debugging output from asterisk:
>

> ---
> Nov 24 19:23:43 NOTICE[9700]: chan_sip.c:10815 handle_request_register:
> Registration from '' failed for '10.0.3.21' -
> Username/auth name mismatch
> Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms
> Destroying call '[EMAIL PROTECTED]'

In the web set up page on the phone, did you make sure that the 'Auth ID' is 
set to 100 ?

Paul

-- 
Paul Hewlett - CottonPickinMinds - www.cottonpickinminds.co.za
Tel: +27 21 852 8812  Cel: +27 84 420 9282  Fax: +27 86 672 0563
-- 
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

Re: [Asterisk-Users] Pros and Cons of T1/E1 cards

[Zoa]

I think they also support PRI cards,

I'd go for something that is the best supported and most stable for
asterisk, which would mean zaptel.

I'm using digium cards and have no issues with them, I have no
experience with sangoma, so can't make an honest comparison between
those two brands.

I do have experience with CAPI and bristuff, and don't like em that
much, (had too much problems in the past, especially when upgrading
between asterisk versions, or kernel versions).

My advise: certainly don't go for something non zaptel.

Zoa

Jean-Michel Hiver wrote:




AVM/ CAPI
eIcon  / CAPI
Junghanns  / Bristuff



As far as I'm aware, CAPI and Bristuff are BRI card so you can strike
those off.

___
--Bandwidth and Colocation sponsored by Easynews.com --

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





signature.asc
Description: OpenPGP digital signature
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

[Asterisk-Users] help need for the configuration

[ram]

Hi all
 
iam setting PBX for outgoing calls at this moment
once iam success this , iam planning to do config inbound to
 
So iam start configuring with Outbound calls
 
Ring now my config looks like follow
 
Lan Users-- Astrix--- VoIP provider
 
I have one account with VoIP provider, i can make multiple calls using that accounts
 
i have 20 Lan users, who start making called to out going
 
all of the them connected to Lan Swtich where astrix connected
 
 
I have downloaded Asterisk+addons+sounds
and comipled with any errors
 
now iam looking what are the files need to configured to achieve the following setup.
 
here my question about the config
 
1. where should i config this Account of VoIP to register, so i can make calls out
2. how do i create 20 users and register them and start making calls
3. where can i see which user called where, and duration
4. how do i configure 20 users can talk each other using extensions.
5. the user side can be Soft Phone using PC or Any cisco ATA Box.
 
what are the config files i need to look
 
any suggestions will be appriciated.
 
ram
 
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

[Asterisk-Users] Really lightweight itemised billing

[Chris Bagnall]

Good morning all,

I'm trying to find an application that'll do really lightweight billing for
Asterisk CDRs.

On our asterisk servers deployed at people's offices, we have CDRs being
logged to PostgreSQL, which can then be analysed by the staff at those
offices using a PHP-based CDR analyser. This works fine for legitimate use
verification (it's easy to spot people making hours of phone calls to their
girlfriend's mobile, for example), but it doesn't provide billing
verification.

All I'm looking to do is parse the CDRs for a given date range, lookup each
dialled number in a table to get its rate, then present a formatted list (or
even a .csv) of the person dialling (accountcode), time/date of call,
duration and total cost of call.

All of the billing applications I've seen so far are either 1) really
heavyweight designed for calling card or other charging purposes, or 2) want
me to modify the asterisk configuration to use their AGIs for dialling. It's
an overkill for what I'm after.

Before I go and write some PHP scripts to do what I'm after, has anyone
already done this and have some scripts they want to share? :-)

Thanks in advance.

Regards,

Chris
-- 
C.M. Bagnall, Director, Minotaur I.T. Limited
This email is made from 100% recycled electrons


___
--Bandwidth and Colocation sponsored by Easynews.com --

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

[Asterisk-Users] TE411P

[Erwan DESVERGNES]

Did someone use a Te411p with
4 T2 in France ? I’ve got some 
problem please Help

 

 

 

_

Erwan
Desvergnes - ANDIUM -

82/86 rue Château Gaillard

69100 Villeurbanne

 

Tel. 04 37 43 44 45
/ Fax 04 37 43 44 44

E-mail: [EMAIL PROTECTED]

 






___
--Bandwidth and Colocation sponsored by Easynews.com --

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

RE: [Asterisk-Users] Pros and Cons of T1/E1 cards

[David Waugh]

Hi John,

I'm going to have to disagree with some previous posts.

The Eicon Diva Server PRI/E1/T1 cards support an E1 interface and reduce the 
load of the call handling, echo cancellation etc as this is all processed on 
board on the card, and not on the central CPU of the computer.

You can use the CAPI interface of the card combined with chan_capi_cm with the 
card.
I have not found any problems when using different kernels or different 
versions of asterisk.
I have one setup in our test lab here at Eicon with Asterisk so it does work!

You can have up to 8 Diva Server cards in once machine - including a mixture of 
the analog and BRI cards.

The Diva Server cards in two variants - the V-Series if you only want to use 
them with Voice based applications and the normal All-in-one cards if you want 
to do fax and RAS too.

If you need any more information let me know, and I will assist further

David

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of John
Daragon
Sent: 25 November 2005 00:46
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Pros and Cons of T1/E1 cards


Hi;

We're looking to standardise on a single family of E1 PRI cards.

I guess our options are :

Digium / Zaptel / libpri
Sangoma/ Zaptel / Wanpipe
AVM/ CAPI
eIcon  / CAPI
Junghanns  / Bristuff

Can anyone share any comparative experience of these, please ? Do they 
differ much in terms of interrupt requirement, CPU load &c ?

Any info gratefully received.

jd

-- 

John Daragon  [EMAIL PROTECTED]
argv[0] limited
Lambs Lawn Cottage,  Staple Fitzpaine,  Taunton,  TA3 5SL,  UK
v +44 (0) 1460 234068   f +44 (0) 1460 234069   m +44 (0) 7836 576127


___
--Bandwidth and Colocation sponsored by Easynews.com --

Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

RE: [Asterisk-Users] harry's project

[harry gaillac]

Hello,

I need SER for IM/presence and sip routing.

Harry

--- "Jonathan k. Creasy" <[EMAIL PROTECTED]> a
écrit :

> http://www.automated.it/guidetoasterisk.htm
> 
> I don't think you even require SER in that case. 
> 
> That will be $100. 
> 
> -Jonathan
> 
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On
> Behalf Of harry gaillac
> Sent: Thursday, November 24, 2005 7:11 PM
> To: users@openser.org;
> asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] harry's project
> 
> Hello,
> 
> here is an other  diagram for people who don't yet
> understand what i expect to do.
> 
> Look at sip_call_flow.png file i wish to substitute
> ondo sip server with ser and ondo pbx with asterisk
> .
> 
> ondo sip server is able to do far-end near-end nat I
> guess ser too.
> 
> I do hope i will find some people who help me to
> configure that .
> 
> Regards 
> Harry 
> 
> 
>   
> 
>   
>   
>
___
> 
> Appel audio GRATUIT partout dans le monde avec le
> nouveau Yahoo! Messenger 
> Téléchargez cette version sur
> http://fr.messenger.yahoo.com
> ___
> --Bandwidth and Colocation sponsored by Easynews.com
> --
> 
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
>
http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>   
>
http://lists.digium.com/mailman/listinfo/asterisk-users
> 







___ 
Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger 
Téléchargez cette version sur http://fr.messenger.yahoo.com
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

Re: [Asterisk-Users] stop asterisk when Idle

[asterisk]

I still continue to reboot my asterisk box everyday.

I posted a message on November 22, but it was on another thread and no one
answered me, so I try again here,
where a lot of people told be I was a bad administrator ("Like a Windows
administrator" and I don'0t want to resolve my problem)

Actually I would like to resolve my problem, but I am not able to do this,
so I ask help to anybody who can help me, and repost my
last of 22/11/2005

In short, my problem is that, after one or two days of running, chan oh323
suddendly disappear from asterisk box, without giving any warning / error
In example, you type oh323 show stats at 11 o'clock , and get an answer
from asterisk, about usage of oh323

At 12, without doing anything to the box or to the asterisk, you type the
same command, and you get a  "No such command 'oh323' (type 'help' for
help)

If you type help, no oh323 commands are available.
If you quit asterisk, (STOP NOW) and restart asterisk , no oh323 channel
command is available

if you reboot the machine everything is again fine !

It is so a crazy situation that to reboot appears (to me) the best thing (I
am sorry about this)

This is my previous post:

***
First of all, thank you for your answer, the only that does not claim to
not restart the box !

Asterisk is running on a  Suse Linux 9.3box,
kernel version is   2.6.11.4-21.9-smp
Asterisk is the last stable version via cvs, not cvs head

show version:
Asterisk CVS-v1-0-10/31/05-17:43:16 built by [EMAIL PROTECTED] on a i686
running Linux

So it was the last stable version on 31 of October;

Also other components were taken via CVS;

cvs checkout -r v1-0 zaptel libpri asterisk asterisk-addons asterisk-sounds

about oh323, these are the instructions that I assembled and followed,
reading around;

cd /root
wget
http://www.inaccessnetworks.com/ian/projects/asterisk-oh323/Libraries/pwlib-Mimas_patch2-src-tar.gz
wget
http://www.inaccessnetworks.com/ian/projects/asterisk-oh323/Libraries/openh323-Mimas_patch2-src-tar.gz

cd /usr/src
wget
http://www.inaccessnetworks.com/ian/projects/asterisk-oh323/download/asterisk-oh323-0.6.7.tar.gz

cd /root
tar zxvf pwlib-Mimas_patch2-src-tar.gz
tar zxvf openh323-Mimas_patch2-src-tar.gz
mv pwlib_Mimas_patch2 pwlib
mv openh323_Mimas_patch2 openh323

cd /usr/src
tar zxvf asterisk-oh323-0.6.7.tar.gz

PWLIBDIR=/root/pwlib
export PWLIBDIR
OPENH323DIR=/root/openh323
export OPENH323DIR
LD_LIBRARY_PATH=$PWLIBDIR/lib:$OPENH323DIR/lib
export LD_LIBRARY_PATH

modify the file:
vi /etc/ld.so.conf
and add in it::
/root/pwlib/lib
/root/openh323/lib

then:
ldconfig

cd /root/pwlib
./configure && make clean && make opt && make install && ldconfig

cd /root/openh323
./configure && make clean && make opt && make install && ldconfig

cd /usr/src/asterisk-oh323-0.6.7
modify Makefile according to the directories:

vi /usr/src/asterisk-oh323-0.6.7/Makefile

PWLIBDIR=/root/pwlib
OPENH323DIR=/root/openh323

make && make install && ldconfig

chown /usr/lib/asterisk/modules/asterisk . -R
chgrp /usr/lib/asterisk/modules/asterisk . -R

chown  asterisk /usr/local/lib -R
chgrp  asterisk /usr/local/lib -R

chmod 777 /root
chown  asterisk /root/pwlib -R
chgrp  asterisk /root/pwlib -R

chown  asterisk /root/openh323 -R
chgrp  asterisk /root/openh323 -R


the only thing I am absolutely not hayy to did was that  "chmod 777 /root";
I think that it should be not necessary at all, I did it becouse asterisk
run as "asterisk" user, and peraphs i thought some problems aboutr
accessing pwlib or oh323;

I have an heavily stressed system, but I have a couple of hours of almost
no traffic (people sleep sometimes...)
To shut down asterisk, killing a maximum 1 or 2 phones and than reboot (
only restart gracefully or now is not sufficient to re-live the oh323
channel)
is a bad thing, but is better than drop 5,000 phones 5 hours later.
Why not only reboot ? becouse if you shurdown asterisk BEFORE rebooting,
the cdr is updated correctly with the last phnes running.

I tried to reboot a box WITHOUT exiting from asterisk, and the running
conversetion (with  more then 2000 billsec) was not recorded in the cdr

I am using the g729 codec ( I bought a 30 channels license from Digium).

So, what to say... ah, you also need my oh323,conf file: here it is.


asterisk02:/etc/asterisk # cat oh323.conf
;
; Configuration file of OpenH323 channel driver
;

;-
; General configuration options
; (ports, jitter, GK, ...)
;-
[general]
;
; Address to bind to for incoming connections.
; Default is ALL.
;
listenAddress=0.0.0.0
;
; Port to listen to.
; Default value is 1720.
;
listenPort=1720
;
; Configure the TCP port range to be used by H.323
;
tcpStart=1
tcpEnd=2
;
; Configure the UDP port range to be used by H.323
; Note: The port range used by RTP are configured from
;   "rtp.conf"
;
udpStart=1
udpEnd=2
;
; Ena

[Asterisk-Users] sound problem, please help!

[Esteban Maestre]

Hi all!

I have a strange problem when using asterisk. I have configured asterisk
to receive calls (FX0). In my configuration, I want asterisk to play music
while  I record the caller's speech. If the caller does the call from a
fixed line telephone, there is no problem, but in case the caller does the
call from a mobile GSM phone, the quality of the music he hears becomes so
bad, and even more when he speaks.
I have tried several codecs.
Any idea or advice?

thanks in advance,

-esteban-

___
--Bandwidth and Colocation sponsored by Easynews.com --

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

RE: [Asterisk-Users] Asterisk doesn't start

[harry gaillac]

Hello,

You built asterisk on freebsd ?

Harry
--- Olivier Taylor <[EMAIL PROTECTED]> a écrit
:

> 
> Hello
> 
> Whan starting astersik(1.2) (asterisk -vvc), I
> get this message :
> 
>  [res_config_mysql.so] => (MySQL RealTime
> Configuration Driver)
> /libexec/ld-elf.so.1:
> /usr/lib/asterisk/modules/res_config_mysql.so:
> Undefined s
> ymbol "ast_config_load"
> 
> What did I forgot to do?
> 
> Olivier
> 
> ___
> --Bandwidth and Colocation sponsored by Easynews.com
> --
> 
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
>
http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>   
>
http://lists.digium.com/mailman/listinfo/asterisk-users
> 







___ 
Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger 
Téléchargez cette version sur http://fr.messenger.yahoo.com
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

RE : [Asterisk-Users] Asterisk doesn't start

[Olivier Taylor]

Yes, beta2 works perfectly, but 1.2 released version gives me this error.

Olivier

-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de harry gaillac
Envoyé : vendredi 25 novembre 2005 11:24
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : RE: [Asterisk-Users] Asterisk doesn't start


Hello,

You built asterisk on freebsd ?

Harry
--- Olivier Taylor <[EMAIL PROTECTED]> a écrit
:

> 
> Hello
> 
> Whan starting astersik(1.2) (asterisk -vvc), I
> get this message :
> 
>  [res_config_mysql.so] => (MySQL RealTime
> Configuration Driver)
> /libexec/ld-elf.so.1:
> /usr/lib/asterisk/modules/res_config_mysql.so:
> Undefined s
> ymbol "ast_config_load"
> 
> What did I forgot to do?
> 
> Olivier
> 
> ___
> --Bandwidth and Colocation sponsored by Easynews.com
> --
> 
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
>
http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>   
>
http://lists.digium.com/mailman/listinfo/asterisk-users
> 







___ 
Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger 
Téléchargez cette version sur http://fr.messenger.yahoo.com
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

___
--Bandwidth and Colocation sponsored by Easynews.com --

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

[Asterisk-Users] Call Progress Analysis

[Nitin Joshi]

Hi All,
I am using Asterisk 1.0.7 with an X101P analog card 
which is connected to an Alcatel pbx. My problem is that when I place outbound 
calls on the zap channel, Asterisk returns a connect event as soon as the phone 
starts ringing. This means that Asterisk is not being able to do Call Progress 
analysis on the zap channels. I have tried setting 'callprogress=yes' in 
zapata.conf but it made no difference. This problem is not there with SIP 
and IAX channels. 
Here's my zapata.conf:
[trunkgroups]; define any trunk 
groups
 
[channels]; hardware channels; 
defaultcontext=defaultusecallerid=yeshidecallerid=nocallwaiting=nothreewaycalling=yestransfer=yesechocancel=yesechocancelwhenbridged=yesechotraining=yesrelaxdtmf=yesbusydetect=yesbusycount=6callprogress=yesprogzone=uk
 
group=1callgroup=1pickupgroup=1immediate=no
 
; define channelssignalling=fxs_kschannel 
=> 1
 
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

[Asterisk-Users] testing

[ram]

Hi
 
why my posting are not accepting in this list
 
ram
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

Re: [Asterisk-Users] chan_misdn crashes : init_stack: success but entitylist not empty

[Kib Eki]

yes, i can confirm this. We had similar problems. FC4 comes with gcc4. We added 
gcc3 and recompiled the kernel, asterisk and chan_misdn. Now we can load 
chan_misdn.so with crashing the asterisk server.



Johann Steinwendtner wrote:

Make sure that you compile misdnuser with gcc3.x, gcc4 did
not work for me.

Hans

Yoann Le Bihan schrieb:


Jose,

I met so many problems these last 8 days that I don't remember exactly
which config was mine at that time, so I can't testify the answer...
(just for fun : my linux box is having 3 hd with a different distro on
each of them and I plug the cable on the hd I want to boot depending
on my mood ;o)).

I think I was running 1.0.9. The main things I did were :

  - deinstalling everything (asterisk, misdn, misdnuser, chan_misdn, ...)
  - compiling and installing asterisk 1.2.0 (make ; make install)
  - downloading the install_misdn script on beronet
(http://www.beronet.com/download/install-misdn.tar.gz) and executing
the make install (be careful : you need kernel headers)

And now, I'm done : Asterisk runs without chan_misdn, but crashes with
it :-( But it's installed :-)

Good luck ! ;)

Cheers,

YLB.


2005/11/25, Jose Limeres <[EMAIL PROTECTED]>:


Yoann,
I am going through a similar problem you reported in a past posting:

Nov 24 17:49:31 ERROR[9326] chan_misdn.c: Unable to initialize mISDN
Nov 24 17:49:31 WARNING[9326] loader.c: chan_misdn.so: load_module
failed, returning -1
Nov 24 17:49:31 WARNING[9326] chan_misdn.c: cb_log called with
out-of-range port number! (0)
Nov 24 17:49:31 WARNING[9326] loader.c: Loading module chan_misdn.so 
failed!


How did you solve it?
Thanks,  jose

On 25/11/05, Yoann Le Bihan <[EMAIL PROTECTED]> wrote:


Hi,

Asterisk 1.2 on FC4, all is right, I'm happy. But when I try to load
chan_misdn after a successful install, I get it :

# asterisk -vvvgc
[...]
[chan_features.so] => (Feature Proxy Channel)
 == Registered channel type 'Feature' (Feature Proxy Channel Driver)
[chan_misdn.so] => (Channel driver for mISDN Support (Bri/Pri))
 == Parsing '/etc/asterisk/misdn.conf': Found
Got: 1 from get_ports
Init. Stack on port:1
No Connect port:1
init_stack: Success
#

Nothing else. Asterisk crashes. If I look at /var/log/messages :

# tail /var/log/messages
Nov 25 00:22:39 toto kernel: Debug: sleeping function called from
invalid context at arch/i386/lib/usercopy.c:634
Nov 25 00:22:39 toto kernel: in_atomic():0, irqs_disabled():1
Nov 25 00:22:39 toto kernel:  [] copy_from_user+0x18/0x80
Nov 25 00:22:39 toto kernel:  [] mISDN_write+0x318/0x7c5 
[mISDN_core]
Nov 25 00:22:39 toto kernel:  [] mISDN_write+0x0/0x7c5 
[mISDN_core]

Nov 25 00:22:39 toto kernel:  [] vfs_write+0xa2/0x15a
Nov 25 00:22:39 toto kernel:  [] sys_write+0x41/0x6a
Nov 25 00:22:39 toto kernel:  [] syscall_call+0x7/0xb
Nov 25 00:22:39 toto kernel: MISDN free_device: entitylist not empty
#

Any idea ?... I've been on it for 1 whole week... I'm exhausted :-(

Cheers,

YLB.



___
--Bandwidth and Colocation sponsored by Easynews.com --

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





___
--Bandwidth and Colocation sponsored by Easynews.com --

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





___
--Bandwidth and Colocation sponsored by Easynews.com --

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

Re: [Asterisk-Users] sound problem, please help!

[Leif Neland]

 Original Message 
From: "Esteban Maestre" <[EMAIL PROTECTED]>
To: 
Sent: Friday, November 25, 2005 11:22 AM
Subject: [Asterisk-Users] sound problem, please help!


Hi all!

I have a strange problem when using asterisk. I have configured
asterisk to receive calls (FX0). In my configuration, I want asterisk
to play music while  I record the caller's speech.


Dialup-karaoke? :-)

Leif

___
--Bandwidth and Colocation sponsored by Easynews.com --

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

RE: RE : [Asterisk-Users] Asterisk doesn't start

[harry gaillac]

Try to post your problem to asterisk-dev I guess they
could solve or explain this problem better than
asterisk'users .

Harry

--- Olivier Taylor <[EMAIL PROTECTED]> a écrit
:

> Yes, beta2 works perfectly, but 1.2 released version
> gives me this error.
> 
> Olivier
> 
> -Message d'origine-
> De : [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] De
> la part de harry gaillac
> Envoyé : vendredi 25 novembre 2005 11:24
> À : Asterisk Users Mailing List - Non-Commercial
> Discussion
> Objet : RE: [Asterisk-Users] Asterisk doesn't start
> 
> 
> Hello,
> 
> You built asterisk on freebsd ?
> 
> Harry
> --- Olivier Taylor <[EMAIL PROTECTED]> a
> écrit
> :
> 
> > 
> > Hello
> > 
> > Whan starting astersik(1.2) (asterisk -vvc), I
> > get this message :
> > 
> >  [res_config_mysql.so] => (MySQL RealTime
> > Configuration Driver)
> > /libexec/ld-elf.so.1:
> > /usr/lib/asterisk/modules/res_config_mysql.so:
> > Undefined s
> > ymbol "ast_config_load"
> > 
> > What did I forgot to do?
> > 
> > Olivier
> > 
> > ___
> > --Bandwidth and Colocation sponsored by
> Easynews.com
> > --
> > 
> > Asterisk-Users mailing list
> > Asterisk-Users@lists.digium.com
> >
>
http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> >   
> >
>
http://lists.digium.com/mailman/listinfo/asterisk-users
> > 
> 
> 
> 
>   
> 
>   
>   
>
___
> 
> Appel audio GRATUIT partout dans le monde avec le
> nouveau Yahoo! Messenger 
> Téléchargez cette version sur
> http://fr.messenger.yahoo.com
> ___
> --Bandwidth and Colocation sponsored by Easynews.com
> --
> 
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
>
http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>   
>
http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> ___
> --Bandwidth and Colocation sponsored by Easynews.com
> --
> 
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
>
http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>   
>
http://lists.digium.com/mailman/listinfo/asterisk-users
> 







___ 
Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger 
Téléchargez cette version sur http://fr.messenger.yahoo.com
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

RE: [Asterisk-Users] NewBie to Ast Server, help need for the configuration

[Mark Edwards]

Title: Message



Can I 
suggest a quick review of http://voip-info.org as the majority of your 
questions will be answered with the information contained on this 
site.
 
cheers,
 
Mark.

  
  -Original Message-From: ram 
  [mailto:[EMAIL PROTECTED] Sent: Friday, 25 November 2005 4:54 
  PMTo: asterisk-users@lists.digium.comSubject: 
  [Asterisk-Users] NewBie to Ast Server,help need for the 
  configuration
  Hi all
   
  iam setting PBX for outgoing calls at this moment
  once iam success this , iam planning to do config inbound to
   
  So iam start configuring with Outbound calls
   
  Ring now my config looks like follow
   
  Lan Users-- Astrix--- VoIP provider
   
  I have one account with VoIP provider, i can make multiple calls using 
  that accounts
   
  i have 20 Lan users, who start making called to out going
   
  all of the them connected to Lan Swtich where astrix connected
   
   
  I have downloaded Asterisk+addons+sounds
  and comipled with any errors
   
  now iam looking what are the files need to configured to achieve the 
  following setup.
   
  here my question about the config
   
  1. where should i config this Account of VoIP to register, so i can make 
  calls out
  2. how do i create 20 users and register them and start making 
calls
  3. where can i see which user called where, and duration
  4. how do i configure 20 users can talk each other using 
extensions.
  5. the user side can be Soft Phone using PC or Any cisco ATA Box.
   
  what are the config files i need to look
   
  any suggestions will be appriciated.
   
  ram
   
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

RE: [Asterisk-Users] PRI problems again - What should I do ?

[Steve Totaro]

> 
> They claim that there is nothing wrong and that it is my equipment
that
> is the problem. I asked them to reset the line last night, so I'll see
> what happens today. Also sent a rocket up the backside of my salesrep,
> so perhaps that may achieve something ...

If I had a dollar for every time I have heard this I would not be rich
but would certainly have a couple hundred dollars.

Just get a tech onsite.  If they send a useless tech, get another one
out there.  In my experience, it is usually the second or third tech
when the telco tech says, "here is the problem" or finds no problem but
all of a sudden things work as they should.

Thanks,
Steve
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

RE: [Asterisk-Users] asterisk.conf question

[Steve Totaro]

> > Adrian A wrote:
> >
> >> Does anyone know what exactly the option
> >> transmit_silence_during_record in asterisk.conf does? Is this
useful
> >> for voicemail recording?
> >
> > Could the option be named any more explicitly? It does _exactly_
what
> > it says it does.
> 
> Some providers terminate the connection if nothing is transmitted for
x
> seconds.
> If asterisk sends nothing while the caller speaks his message, the
> provider
> might terminate the call.
> So asterisk can transmit silence (which is not "nothing") during
record.
> 
> Similarly you might have to say "yes dear" regularly to avoid having
the
> connection terminated while talking to your SO. :-)
> 
> Leif

Thanks for the useful answer to this Leif.  Yes, it does _exactly_ what
it says it does but the usage of the option remained elusive to me.  
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

RE: [Asterisk-Users] NewBie to Ast Server, help need for the configuration

[Steve Totaro]

Download [EMAIL PROTECTED] and install that or be prepared to read
experiment for a while.  These questions are way too broad in scope for
this mailing list.  Once you specific questions, you will probably get
some help from the list.

 

www.voip-info.org

 

Thanks,

Steve

 

  _  

From: ram [mailto:[EMAIL PROTECTED] 
Sent: Friday, November 25, 2005 12:54 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] NewBie to Ast Server,help need for the
configuration

 

Hi all

 

iam setting PBX for outgoing calls at this moment

once iam success this , iam planning to do config inbound to

 

So iam start configuring with Outbound calls

 

Ring now my config looks like follow

 

Lan Users-- Astrix--- VoIP provider

 

I have one account with VoIP provider, i can make multiple calls using
that accounts

 

i have 20 Lan users, who start making called to out going

 

all of the them connected to Lan Swtich where astrix connected

 

 

I have downloaded Asterisk+addons+sounds

and comipled with any errors

 

now iam looking what are the files need to configured to achieve the
following setup.

 

here my question about the config

 

1. where should i config this Account of VoIP to register, so i can make
calls out

2. how do i create 20 users and register them and start making calls

3. where can i see which user called where, and duration

4. how do i configure 20 users can talk each other using extensions.

5. the user side can be Soft Phone using PC or Any cisco ATA Box.

 

what are the config files i need to look

 

any suggestions will be appriciated.

 

ram

 

___
--Bandwidth and Colocation sponsored by Easynews.com --

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

Re: [Asterisk-Users] PRI problems again - What should I do ?

[Julian Lyndon-Smith]

Steve Totaro wrote:


They claim that there is nothing wrong and that it is my equipment
   


that
 


is the problem. I asked them to reset the line last night, so I'll see
what happens today. Also sent a rocket up the backside of my salesrep,
so perhaps that may achieve something ...
   



If I had a dollar for every time I have heard this I would not be rich
but would certainly have a couple hundred dollars.
 



Yeah, aint that the truth.


Just get a tech onsite.  If they send a useless tech, get another one
out there.  In my experience, it is usually the second or third tech
when the telco tech says, "here is the problem" or finds no problem but
all of a sudden things work as they should.
 


They are doing today - they reset the line last night. Fingers crossed :)

Julian


Thanks,
Steve
___
--Bandwidth and Colocation sponsored by Easynews.com --

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


 



___
--Bandwidth and Colocation sponsored by Easynews.com --

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

[Asterisk-Users] Command line

[Tony Spencer]

 

Hi

 

I’m pretty new to using Asterisk and have
searched to find an answer to my question but have failed to.

I was wondering if you can use Asterisk
from the command line to make it make an outgoing call and issue other commands
whilst it’s in the call?

Sort of like when you use Minicom with a
modem connected to a serial port and send it AT commands.

 

Thanks

Tony








--
No virus found in this outgoing message.
Checked by AVG Free Edition.
Version: 7.1.362 / Virus Database: 267.13.7/182 - Release Date: 24/11/2005
 

--
No virus found in this outgoing message.
Checked by AVG Free Edition.
Version: 7.1.362 / Virus Database: 267.13.7/182 - Release Date: 24/11/2005
 
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

[Asterisk-Users] authentication fails to provider after upgrading to 1.2.0

[Urban]

Hi,

after we upgraded to 1.2.0 we can not make any outgoing calls to our sip 
provider (incoming works). I think the provider is running asterisk 
1.0.9. I have a tcpdump that indicates that our asterisk is not 
responding correctly when the providers asterisk is saying 'Proxy 
authentication required' Here is the sequence:


INVITE sent to provider
407 returned from provider (proxy authentication required)
ACK sent to provider

normally a new INVITE should be sent after the ACK with digest info to 
the provider, but it's not. Instead our asterisk is returning 503: 
Service unavailable to the internal extension placing the outgoing call.


any ideas?

thanks
urban
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

Re: [Asterisk-Users] authentication fails to provider after upgrading to 1.2.0

[Urban]

Urban wrote:


Hi,

after we upgraded to 1.2.0 we can not make any outgoing calls to our 
sip provider (incoming works). I think the provider is running 
asterisk 1.0.9. I have a tcpdump that indicates that our asterisk is 
not responding correctly when the providers asterisk is saying 'Proxy 
authentication required' Here is the sequence:


INVITE sent to provider
407 returned from provider (proxy authentication required)
ACK sent to provider

normally a new INVITE should be sent after the ACK with digest info to 
the provider, but it's not. Instead our asterisk is returning 503: 
Service unavailable to the internal extension placing the outgoing call.


any ideas?

thanks
urban
___
--Bandwidth and Colocation sponsored by Easynews.com --

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


solved the problem, cut and paste error when copying the old 
configuration, username was truncated to user in the providers context...


/urban
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

Re: [Asterisk-Users] sound problem, please help!

[Esteban Maestre]

kind of... ;)
I want to know what the people say when they are waiting... :P

do you have any idea on what the problem could be?

-esteban-


>  Original Message 
> From: "Esteban Maestre" <[EMAIL PROTECTED]>
> To: 
> Sent: Friday, November 25, 2005 11:22 AM
> Subject: [Asterisk-Users] sound problem, please help!
>
>> Hi all!
>>
>> I have a strange problem when using asterisk. I have configured
>> asterisk to receive calls (FX0). In my configuration, I want asterisk
>> to play music while  I record the caller's speech.
>
> Dialup-karaoke? :-)
>
> Leif
>
>
>
>


___
--Bandwidth and Colocation sponsored by Easynews.com --

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

[Asterisk-Users] Looking for Info on Asterisk scripting

[Obelix]

Is there a source of Asterisk programming techniques in various languages - ie
Asterisk scripting in general, not the main Asterisk program itself?

Obelix



This message was sent using IMP, the Internet Messaging Program.

___
--Bandwidth and Colocation sponsored by Easynews.com --

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

Re: [Asterisk-Users] Looking for Info on Asterisk scripting

[Stefan Reuter]

Obelix schrieb:
> Is there a source of Asterisk programming techniques in various languages - ie
> Asterisk scripting in general, not the main Asterisk program itself?

What you are looking for is probably AGI (the Asterisk Gateway
Interface) that is to Asterisk what CGI is to a Webserver.

Have a look at the Wiki at
http://www.voip-info.org/tiki-index.php?page=Asterisk+AGI
it tells you all you need to know to get started.

=Stefan


signature.asc
Description: OpenPGP digital signature
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

Re: [Asterisk-Users] chan_misdn crashes : init_stack: success but entitylist not empty

[Yoann Le Bihan]

All right, thanks a lot for your answers ! :)
I also contacted beronet support (as they develop chan_misdn)... this
bug is known and should be fixed within a few weeks...

Alleluiiia ;))

I'm compiling with gcc32... hope it runs ! :-)

Cheers,

YLB.


2005/11/25, Kib Eki <[EMAIL PROTECTED]>:
> yes, i can confirm this. We had similar problems. FC4 comes with gcc4. We 
> added
> gcc3 and recompiled the kernel, asterisk and chan_misdn. Now we can load
> chan_misdn.so with crashing the asterisk server.
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

Re: [Asterisk-Users] Grandstream problem

[Alfie Viechweg]

Paul Hewlett wrote:


On Friday 25 November 2005 01:45, Alfie Viechweg wrote:
 


Can some on help me find the problem here please:
I'm using asterisk 1.2.0 with Grandstream GXP-2000

This is the debugging output from asterisk:

   



 


---
Nov 24 19:23:43 NOTICE[9700]: chan_sip.c:10815 handle_request_register:
Registration from '' failed for '10.0.3.21' -
Username/auth name mismatch
Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms
Destroying call '[EMAIL PROTECTED]'
   



In the web set up page on the phone, did you make sure that the 'Auth ID' is 
set to 100 ?


Paul

 

It was an installation problem. I used INSTALL_PREFIX variable to place 
the sample files in a staging area and that added the staging area 
prefix to all the pathnames in asterisk.conf. Editing asterisk.conf 
fixed the problem.


The Makefile has two (2) staging area variables DESTDIR and 
INSTALL_PREFIX but is not too clear about the uses and result of them. I 
used the wrong one I guess.


Thanks anyway.

  -Alfie
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

RE: [Asterisk-Users] NewBie to Ast Server, help need for the configuration

[Juan Janczuk]

It 
seems like [EMAIL PROTECTED] could be your best 
solution.
It has 
a nice user interface (AMP, you can try to install it in your actual asterisk 
box),
that 
lets you do all you say.
 
Regards.
Juan.

  -Mensaje original-De: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED]En nombre de 
  ramEnviado el: Viernes, 25 de Noviembre de 2005 02:54 
  a.m.Para: asterisk-users@lists.digium.comAsunto: 
  [Asterisk-Users] NewBie to Ast Server,help need for the 
  configuration
  Hi all
   
  iam setting PBX for outgoing calls at this moment
  once iam success this , iam planning to do config inbound to
   
  So iam start configuring with Outbound calls
   
  Ring now my config looks like follow
   
  Lan Users-- Astrix--- VoIP provider
   
  I have one account with VoIP provider, i can make multiple calls using 
  that accounts
   
  i have 20 Lan users, who start making called to out going
   
  all of the them connected to Lan Swtich where astrix connected
   
   
  I have downloaded Asterisk+addons+sounds
  and comipled with any errors
   
  now iam looking what are the files need to configured to achieve the 
  following setup.
   
  here my question about the config
   
  1. where should i config this Account of VoIP to register, so i can make 
  calls out
  2. how do i create 20 users and register them and start making 
calls
  3. where can i see which user called where, and duration
  4. how do i configure 20 users can talk each other using 
extensions.
  5. the user side can be Soft Phone using PC or Any cisco ATA Box.
   
  what are the config files i need to look
   
  any suggestions will be appriciated.
   
  ram
   
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

RE: [Asterisk-Users] stop asterisk when Idle

[Juan Janczuk]

I misunderstood you the first time. Sorry. ( I thook that you only wanted to
restart asterisk itself).

Well, I'm not sure, 'cause I never used it, but you can try a scrip like
this one:

-
#!/bin/sh
asterisk -x -r 'stop when convenient'
reboot


It should work as you intend.

Regards.
Juan.

> -Mensaje original-
> De: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] nombre de
> [EMAIL PROTECTED]
> Enviado el: Viernes, 25 de Noviembre de 2005 07:13 a.m.
> Para: Asterisk Users Mailing List - Non-Commercial Discussion
> Asunto: Re: [Asterisk-Users] stop asterisk when Idle
>
>
> I still continue to reboot my asterisk box everyday.
>
> I posted a message on November 22, but it was on another thread and no one
> answered me, so I try again here,
> where a lot of people told be I was a bad administrator ("Like a Windows
> administrator" and I don'0t want to resolve my problem)
>
> Actually I would like to resolve my problem, but I am not able to do this,
> so I ask help to anybody who can help me, and repost my
> last of 22/11/2005
>
> In short, my problem is that, after one or two days of running, chan oh323
> suddendly disappear from asterisk box, without giving any warning / error
> In example, you type oh323 show stats at 11 o'clock , and get an answer
> from asterisk, about usage of oh323
>
> At 12, without doing anything to the box or to the asterisk, you type the
> same command, and you get a  "No such command 'oh323' (type 'help' for
> help)
>
> If you type help, no oh323 commands are available.
> If you quit asterisk, (STOP NOW) and restart asterisk , no oh323 channel
> command is available
>
> if you reboot the machine everything is again fine !
>
> It is so a crazy situation that to reboot appears (to me) the
> best thing (I
> am sorry about this)
>
> This is my previous post:
>
> ***
> First of all, thank you for your answer, the only that does not claim to
> not restart the box !
>
> Asterisk is running on a  Suse Linux 9.3box,
> kernel version is   2.6.11.4-21.9-smp
> Asterisk is the last stable version via cvs, not cvs head
>
> show version:
> Asterisk CVS-v1-0-10/31/05-17:43:16 built by [EMAIL PROTECTED] on a i686
> running Linux
>
> So it was the last stable version on 31 of October;
>
> Also other components were taken via CVS;
>
> cvs checkout -r v1-0 zaptel libpri asterisk asterisk-addons
> asterisk-sounds
>
> about oh323, these are the instructions that I assembled and followed,
> reading around;
>
> cd /root
> wget
> http://www.inaccessnetworks.com/ian/projects/asterisk-oh323/Librar
> ies/pwlib-Mimas_patch2-src-tar.gz
> wget
> http://www.inaccessnetworks.com/ian/projects/asterisk-oh323/Librar
> ies/openh323-Mimas_patch2-src-tar.gz
>
> cd /usr/src
> wget
> http://www.inaccessnetworks.com/ian/projects/asterisk-oh323/downlo
> ad/asterisk-oh323-0.6.7.tar.gz
>
> cd /root
> tar zxvf pwlib-Mimas_patch2-src-tar.gz
> tar zxvf openh323-Mimas_patch2-src-tar.gz
> mv pwlib_Mimas_patch2 pwlib
> mv openh323_Mimas_patch2 openh323
>
> cd /usr/src
> tar zxvf asterisk-oh323-0.6.7.tar.gz
>
> PWLIBDIR=/root/pwlib
> export PWLIBDIR
> OPENH323DIR=/root/openh323
> export OPENH323DIR
> LD_LIBRARY_PATH=$PWLIBDIR/lib:$OPENH323DIR/lib
> export LD_LIBRARY_PATH
>
> modify the file:
> vi /etc/ld.so.conf
> and add in it::
> /root/pwlib/lib
> /root/openh323/lib
>
> then:
> ldconfig
>
> cd /root/pwlib
> ./configure && make clean && make opt && make install && ldconfig
>
> cd /root/openh323
> ./configure && make clean && make opt && make install && ldconfig
>
> cd /usr/src/asterisk-oh323-0.6.7
> modify Makefile according to the directories:
>
> vi /usr/src/asterisk-oh323-0.6.7/Makefile
>
> PWLIBDIR=/root/pwlib
> OPENH323DIR=/root/openh323
>
> make && make install && ldconfig
>
> chown /usr/lib/asterisk/modules/asterisk . -R
> chgrp /usr/lib/asterisk/modules/asterisk . -R
>
> chown  asterisk /usr/local/lib -R
> chgrp  asterisk /usr/local/lib -R
>
> chmod 777 /root
> chown  asterisk /root/pwlib -R
> chgrp  asterisk /root/pwlib -R
>
> chown  asterisk /root/openh323 -R
> chgrp  asterisk /root/openh323 -R
>
>
> the only thing I am absolutely not hayy to did was that  "chmod
> 777 /root";
> I think that it should be not necessary at all, I did it becouse asterisk
> run as "asterisk" user, and peraphs i thought some problems aboutr
> accessing pwlib or oh323;
>
> I have an heavily stressed system, but I have a couple of hours of almost
> no traffic (people sleep sometimes...)
> To shut down asterisk, killing a maximum 1 or 2 phones and than reboot (
> only restart gracefully or now is not sufficient to re-live the oh323
> channel)
> is a bad thing, but is better than drop 5,000 phones 5 hours later.
> Why not only reboot ? becouse if you shurdown asterisk BEFORE rebooting,
> the cdr is updated correctly with the last phnes running.
>
> I tried to reboot a box WITHOUT exiting from asterisk, and the running
> conversetion (with  m

[Asterisk-Users] Siemens OptiPoint 4xx

[René Enskat [Teamware GmbH]]

Somenbody know if
the HINT function is not working for the OptiPoint4xx
series?.
I configured it but
the keys are not working.

___
--Bandwidth and Colocation sponsored by Easynews.com --

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

Re: [Asterisk-Users] chan_misdn crashes : init_stack: success but entitylist not empty

[Yoann Le Bihan]

2005/11/25, Jose Limeres <[EMAIL PROTECTED]>:
> Yoann,
> I am going through a similar problem you reported in a past posting:
>
> Nov 24 17:49:31 ERROR[9326] chan_misdn.c: Unable to initialize mISDN
> Nov 24 17:49:31 WARNING[9326] loader.c: chan_misdn.so: load_module
> failed, returning -1
> Nov 24 17:49:31 WARNING[9326] chan_misdn.c: cb_log called with
> out-of-range port number! (0)
> Nov 24 17:49:31 WARNING[9326] loader.c: Loading module chan_misdn.so failed!
>
> How did you solve it?

I looked back to this error. In fact, it happens when you forget to
initialize driver, so do it :

/etc/init.d/misdn-init scan

If everything goes well you can do :

/etc/init.d/misdn-init config
/etc/init.d/misdn-init start

Then, you can start asterisk :-)

Cheers,

YLB.
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

[Asterisk-Users] busy channels

[BS]

Hello,

Please consider a pbx server with 16-fxo channel.

After a while asterisk start to run, all channels appear busy. Asterisk 
can not receive any call because of all lines are busy.


If i restart zaptel, it is working.  What do you think about the problem?

regards,

-baris

___
--Bandwidth and Colocation sponsored by Easynews.com --

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

RE: [Asterisk-Users] Linksys SPA-841 Disconnects from Asterisk

[Dave Morrow]

Title: Linksys SPA-841 Disconnects from Asterisk




Thanks for the reply, however, I am already running the 
latest 3.14a
 
It seems it may have something to do with the "Registration 
expires" setting on these phones.  This value is set at the default 
3600.  After this interval, the phone de-registers and does not re-register 
with the Asterisk server.
 
David A. Morrow Technical Systems Lead Autodata 
Solutions Company [EMAIL PROTECTED] 
http://www.autodata.net 
* PLEASE NOTE THAT EFFECTIVE 
DEC 1,2005 MY TELEPHONE NUMBER WILL CHANGE * 
NEW !!! Tel: (519) 963-3020 Fax: (519) 451-6615  
< Poor planning on your part does 
not necessarily constitute an emergency on my part! > 
This message has originated from Autodata Solutions. 
The attached material is the Confidential and Proprietary Information of 
Autodata Solutions. This email and any files transmitted with it are 
confidential and intended solely for the use of the individual or entity to whom 
they are addressed. If you have received this email in error please delete this 
message and notify the Autodata system administrator at [EMAIL PROTECTED] 
 


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Alex 
TerneroSent: Thursday, November 24, 2005 5:51 PMTo: 
'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: 
[Asterisk-Users] Linksys SPA-841 Disconnects from Asterisk


I don t have problems, 
after upgrade the firmware to the latest version.
 
Alex
 




From: 
[EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Dave MorrowSent: Thursday, November 24, 2005 3:49 
PMTo: Asterisk Users Mailing List - Non-Commercial 
DiscussionSubject: [Asterisk-Users] Linksys SPA-841 
Disconnects from Asterisk
 
Hi 
all, I wonder if anyone out there has experienced an issue I am having with my 
Sipura / Linksys SPA-841 phones. 
They work fine generally, but 
occasionally, incoming calls are missed.  It's like the SIP registration is 
expiring.  Does anyone know how to force the phone to re-register 
automatically?  
 
David A. Morrow 
Technical Systems Lead 
Autodata Solutions 
Company [EMAIL PROTECTED] 
http://www.autodata.net 

* PLEASE NOTE 
THAT EFFECTIVE DEC 1,2005 MY TELEPHONE NUMBER WILL CHANGE * 

NEW !!! Tel: (519) 
963-3020 Fax: (519) 
451-6615  
< Poor planning on 
your part does not necessarily constitute an emergency on my part! 
> 
This message has originated from 
Autodata Solutions. The attached material is the Confidential and Proprietary 
Information of Autodata Solutions. This email and any files transmitted with it 
are confidential and intended solely for the use of the individual or entity to 
whom they are addressed. If you have received this email in error please delete 
this message and notify the Autodata system administrator at [EMAIL PROTECTED] 

___
--Bandwidth and Colocation sponsored by Easynews.com --

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

RE: [Asterisk-Users] Linksys SPA-841 Disconnects from Asterisk

[Dave Morrow]

Title: Linksys SPA-841 Disconnects from Asterisk




Thanks for the reply, however, I am already running the 
latest 3.14a
 
It seems it may have something to do with the "Registration 
expires" setting on these phones.  This value is set at the default 
3600.  After this interval, the phone de-registers and does not re-register 
with the Asterisk server.
 
David A. Morrow Technical Systems Lead Autodata 
Solutions Company [EMAIL PROTECTED] 
http://www.autodata.net 
* PLEASE NOTE THAT EFFECTIVE 
DEC 1,2005 MY TELEPHONE NUMBER WILL CHANGE * 
NEW !!! Tel: (519) 963-3020 Fax: (519) 451-6615  
< Poor planning on your part does 
not necessarily constitute an emergency on my part! > 
This message has originated from Autodata Solutions. 
The attached material is the Confidential and Proprietary Information of 
Autodata Solutions. This email and any files transmitted with it are 
confidential and intended solely for the use of the individual or entity to whom 
they are addressed. If you have received this email in error please delete this 
message and notify the Autodata system administrator at [EMAIL PROTECTED] 
 


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Benjamin 
LawetzSent: Thursday, November 24, 2005 3:59 PMTo: 
'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: 
[Asterisk-Users] Linksys SPA-841 Disconnects from Asterisk

Check in you console or your logs when this happens. I'm 
guessing it's a Stale Nonce
 
If this is the case, Sipura supposedly fixed the bug on 
it's most recent firmware (At least for the SPA-1001 and SPA-2100, but I'm 
guessing the SPA-841 also)
 


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Dave 
MorrowSent: November 24, 2005 3:49 PMTo: Asterisk Users 
Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] 
Linksys SPA-841 Disconnects from Asterisk

Hi all, I wonder if anyone out there has experienced 
an issue I am having with my Sipura / Linksys SPA-841 phones. 
They work fine generally, but occasionally, incoming 
calls are missed.  It's like the SIP registration is expiring.  Does 
anyone know how to force the phone to re-register automatically?  

David A. Morrow Technical Systems Lead Autodata 
Solutions Company [EMAIL PROTECTED] http://www.autodata.net 
* PLEASE NOTE THAT EFFECTIVE 
DEC 1,2005 MY TELEPHONE NUMBER WILL CHANGE * 
NEW !!! Tel: (519) 963-3020 Fax: (519) 451-6615  
< Poor planning on your part does 
not necessarily constitute an emergency on my part! > 
This message has originated from Autodata Solutions. 
The attached material is the Confidential and Proprietary Information of 
Autodata Solutions. This email and any files transmitted with it are 
confidential and intended solely for the use of the individual or entity to whom 
they are addressed. If you have received this email in error please delete this 
message and notify the Autodata system administrator at [EMAIL PROTECTED] 
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

Re: [Asterisk-Users] Looking for Info on Asterisk scripting

[Obelix]

Quoting Stefan Reuter <[EMAIL PROTECTED]>:

I am quite familiar with Asterisk AGI, but I am looking for forums or groups
that discuss more techniques, like the Manager API etc.

Asterisk Users only delves into Asterisk dial plans, configuration etc and
Asterisk Dev deals with the main Asterisk itself.

I am looking for more scripting techniques

> Obelix schrieb:
> > Is there a source of Asterisk programming techniques in various languages -
> ie
> > Asterisk scripting in general, not the main Asterisk program itself?
>
> What you are looking for is probably AGI (the Asterisk Gateway
> Interface) that is to Asterisk what CGI is to a Webserver.
>
> Have a look at the Wiki at
> http://www.voip-info.org/tiki-index.php?page=Asterisk+AGI
> it tells you all you need to know to get started.
>
> =Stefan
>





This message was sent using IMP, the Internet Messaging Program.

___
--Bandwidth and Colocation sponsored by Easynews.com --

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

Re: [Asterisk-Users] asterisk.conf question

[Kevin P. Fleming]

Steve Totaro wrote:


Thanks for the useful answer to this Leif.  Yes, it does _exactly_ what
it says it does but the usage of the option remained elusive to me.  


Well, I was being in an ultra-pedantic mode when I answered your 
question... which was 'what does it do', not 'when would I need to use 
it' :-)

___
--Bandwidth and Colocation sponsored by Easynews.com --

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

Re: [Asterisk-Users] SIP Extension behind NAT, Asterisk on a public domain

[Bharath]

I had the UDP ports forwarded, In any case I will be testing with a
brand new router today, & then will confirm if my old router had a
problem.On 11/24/05, Tom Rymes <[EMAIL PROTECTED]> wrote:
On Nov 24, 2005, at 12:14 PM, Bharath wrote:> I found out that I have a faulty Belkin Router which was causing> the problem. I tried forwarding ports as well as DMZ'd the Sip> device but still could'nt not hear the voice. So i plugged the sip
> device directly to the cable modem & it worked fine. So my guess is> that though I have set up the router to forwards port to the sip> device there is something happening at the router that is blocking
> the RTP ports (1-2).> ThanksBefore you blame the router, make sure that you forwarded UDP ports5060 and 1-2, not TCP. (Though I guess the DMZ setup wouldhave taken care of that...)
TomTom RymesCascade Link Systemswww.cascadelinksystems.com(603) 375-1414"Intelligent technology solutions for small businesses."
___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list
Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:   
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

[Asterisk-Users] Philippines Asterisk users, anyone?

[Angelito Manansala]

--
Best Regards,
Angelito Manansala
Mobile: +639175425807
DID: (+63) 44 7906770
msn: [EMAIL PROTECTED]
skype: bulcrack
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

[Asterisk-Users] authentication question

[Martin van den Berg]

Hi Folks,

I am implementing a SIP ua and have been testing it against the SER
proxy which works fine. So my next step is to test is with Asterisk
and I run into a problem. I can REGISTER my UA with authentication, no
problem but the Asterisk does not accepts my credentials in the
INVITE. I have checked my username and password with the sip.conf file
(secret=...) and it looks ok. You can find the messages below.

Any ideas?

Martin


The trace:
My UA sends the INVITE to Asterisk:

INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.126:5060;branch=z9hG4bK-276428-27856
From: "4302" ;tag=27642829233
To: 
Call-ID: [EMAIL PROTECTED]
CSeq: 20 INVITE
Contact: 
Max-Forwards: 70
User-Agent: My UA
Privacy: none
P-Preferred-Identity: "4302" 
P-Preferred-Identity: 
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS
Content-Type: application/sdp
Accept: application/sdp
Content-Length:   234

v=0
o=iS3000 0 0 IN IP4 192.168.1.216
s=-
c=IN IP4 192.168.1.216
t=0 0
m=audio 49368 RTP/AVP 0 8 18 96
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:96 telephone-event/8000
a=ptime:40
a=sendrecv

Asterisk challenges the INVITE:
407 Proxy Authentication Required
Via: SIP/2.0/UDP
192.168.1.126:5060;branch=z9hG4bK-276428-27856;received=10.20.0.1;rport=1025
From: "4302" ;tag=27642829233
To: ;tag=as385f1a51
Call-ID: [EMAIL PROTECTED]
CSeq: 20 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: 
Proxy-Authenticate: Digest realm="asterisk.SipTux2", nonce="52beb24a"
Content-Length: 0

ACK sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.126:5060;branch=z9hG4bK-276428-27856
From: "4302" ;tag=27642829233
To: ;tag=as385f1a51
Call-ID: [EMAIL PROTECTED]
CSeq: 20 ACK
Max-Forwards: 70
User-Agent: iS3000 SIP Server, Philips Business Communications
Content-Length: 0

INVITE again with credentials from my UA:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.126:5060;branch=z9hG4bK-276429-6368
From: "4302" ;tag=2764299705
To: 
Call-ID: [EMAIL PROTECTED]
CSeq: 20 INVITE
Contact: 
Proxy-Authorization: Digest username="4302", realm="asterisk.SipTux2",
nonce="52beb24a", uri="sip:[EMAIL PROTECTED]",
response="9c3f59f925a7feb47e0631735d103c88", algorithm=MD5
Max-Forwards: 70
User-Agent: My UA
Privacy: none
P-Preferred-Identity: "4302" 
P-Preferred-Identity: 
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS
Content-Type: application/sdp
Accept: application/sdp
Content-Length:   234

v=0
o=iS3000 0 0 IN IP4 192.168.1.216
s=-
c=IN IP4 192.168.1.216
t=0 0
m=audio 49368 RTP/AVP 0 8 18 96
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:96 telephone-event/8000
a=ptime:40
a=sendrecv

SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
192.168.1.126:5060;branch=z9hG4bK-276429-6368;received=10.20.0.1;rport=1025
From: "4302" ;tag=2764299705
To: ;tag=as64b1c17b
Call-ID: [EMAIL PROTECTED]
CSeq: 20 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: 
Proxy-Authenticate: Digest realm="asterisk.SipTux2", nonce="7a055f95"
Content-Length: 0


ACK sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.126:5060;branch=z9hG4bK-276429-6368
From: "4302" ;tag=2764299705
To: ;tag=as64b1c17b
Call-ID: [EMAIL PROTECTED]
CSeq: 20 ACK
Proxy-Authorization: Digest username="4302", realm="asterisk.SipTux2",
nonce="52beb24a", uri="sip:[EMAIL PROTECTED]",
response="9c3f59f925a7feb47e0631735d103c88", algorithm=MD5
Max-Forwards: 70
User-Agent: iS3000 SIP Server, Philips Business Communications
Content-Length: 0

--
Skype: MartinvdbBerg
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

[Asterisk-Users] SIP response 484 "Address Incomplete" incorrectly handled

[Marc Storck]

Hello,

I saw that the error:

SIP response 484 "Address Incomplete"

is converted into

DIALSTATUS = NOANSWER
HANGUPCAUSE = 16 (NORMAL_CLEARING)

shouldn't it be something like

HANGUPCAUSE = 1 (UNALLOCATED)
HANGUPCAUSE = 28 (INVALID_NUMBER_FORMAT)

or another cause, other than NORMAL ???

Regards,

Marc

___
--Bandwidth and Colocation sponsored by Easynews.com --

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

Re: [Asterisk-Users] Command line

[Tom Rymes]

On Nov 25, 2005, at 7:33 AM, Tony Spencer wrote:

Hi

I’m pretty new to using Asterisk and have searched to find an  
answer to my question but have failed to.


I was wondering if you can use Asterisk from the command line to  
make it make an outgoing call and issue other commands whilst it’s  
in the call?


Sort of like when you use Minicom with a modem connected to a  
serial port and send it AT commands.


 Thanks

Tony


Tony,

If you have a sound card installed and properly configured in your  
Asterisk server, then you can plug in a microphone and headset and  
make calls from the CLI using the dial command.


If you want to automate having the system make phone calls, google  
and search voip-info.org for info on .call files. Basically, you  
create a file that specifies to asterisk where to call, using which  
channel, and what to do once the call is connnected. You then copy  
the file to /var/spool/asterisk/outgoing and the call is executed as  
defined.


Tom


Tom Rymes
Cascade Link Systems
www.cascadelinksystems.com
(603) 375-1414

"Intelligent technology solutions for small businesses."


___
--Bandwidth and Colocation sponsored by Easynews.com --

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

[Asterisk-Users] How to initiate a call from a web page?

[Kib Eki]

Hi,

we have a html based telephonelist on our intranet site.
Does there exist any solution to initiate a call from a link ?
We use Polycom SIP IP phones.

thanks and regards,
bk

___
--Bandwidth and Colocation sponsored by Easynews.com --

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

[Asterisk-Users] Second TDM22B board install issue

[James MacLean]

Hi Folks,

Sent this to support, but thought it may be obvious and I should pass it 
here too :).


First (older) TDM22B is in use and appears fine.

Second one, installed and ztfg -vv reports :

[4626926.318000] ACPI: PCI Interrupt :01:08.0[A] -> Link [APC1] ->
GSI 16 (level, high) -> IRQ 16
[4626926.363000] Freshmaker version: 71
[4626926.363000] Freshmaker passed register test
[4626928.073000] Module 0: Installed -- AUTO FXS/DPO
[4626929.304000] Module 1: Installed -- AUTO FXS/DPO
[4626929.504000] Module 2: Installed -- AUTO FXO (FCC mode)
[4626929.704000] Module 3: Installed -- AUTO FXO (FCC mode)
[4626929.705000] Found a Wildcard TDM: Wildcard TDM400P REV E/F (4 modules)
[4626929.706000] Registered tone zone 0 (United States / North America)
[4627086.73] Registered tone zone 0 (United States / North America)
[4627088.903000] Registered tone zone 0 (United States / North America)
[4627091.331000] Registered tone zone 0 (United States / North America)

/etc/zaptel.conf has:
fxsks=3-4,7-8
fxoks=1-2,5-6

When ever I try to use the new channels (5,6,7,8) I may or may not (but
usually) get :

Nov 25 08:51:51 ERROR[27602]: chan_zap.c:10250 setup_zap: Unable to
reconfigure channel '5'
Nov 25 08:51:51 WARNING[27602]: chan_zap.c:11010 reload: Reload of
chan_zap.so is unsuccessful!

/etc/asterisk/zapata.conf has
rxgain=10.0
txgain=-5.0
signalling=fxs_ks
context=outbound
group = 6
callerid=asreceived
channel => 5

I am hoping that I simply misconfigured something ?

Thanks,
JES

begin:vcard
fn:James B MacLean
n:MacLean;James B
org:Education;ITS Technical Services
adr:;;;Halifax;NS;;Canada
email;internet:[EMAIL PROTECTED]
url:http://www.ednet.ns.ca/~macleajb
version:2.1
end:vcard



smime.p7s
Description: S/MIME Cryptographic Signature
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

[Asterisk-Users] speex & ilbc

[Alejandro Vargas]

I'm testing [EMAIL PROTECTED] 2.0 beta 6.

I'm checking de different codecs but with speex and ilbc I don't
receive any sound. I tested xtensofphone and iaxComm. With both I has
the same problem: good sound with ulaw and gsm, but no sound with
xpeex and ilbc. Do I need to change some config for using it?
--
Alejandro Vargas
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

Re: [Asterisk-Users] QSig and MD110

[Rogerio Ferreira da Cunha]

Tim,

It's working. But I'm not getting the callerid from the MD's callers.
Do you have any idea?

Tks 4 your help.

2005/11/24, Tim Rayner <[EMAIL PROTECTED]>:
> Hi Roger,
>
> We've solved this with the MD110 sending calls to cisco VoIP gateways.
> The method is to set the Minimum and Maximum call length for this number
> range on the MD110 - and to configure the destination route to only send
> the call when the minimum length is reached (sometimes called en-block
> sending).  If your asterisk number range is 1500 - 1599
>
> define your minimum and maximum length.
>
> NANLS:EXL=15,MIN=4,MAX=4;
>
> Also - make sure that your Route definition for this destination does
> the enblock.
>
> RODDI:DEST=15,ROU=45,ADC=1x;
>
> The first parameter of ADC causes the call to wait for minimum length
> before sending it to the route - the other parameters should stay at
> their pervious values.
>
> I hope this helps - its worked well for us.
>
> Tim.
>
> >>
> >> Hi,
> >> I have one Asterisk linked to a MD110 (Ericsson PBX) using a TE100P.
> >> I'm using the QSIG  ( Asterisk 1.2).
> >> From * I can make calls elsewhere. But when the calling is coming
> >> from MD, the Asterisk is answering the call at the first digit it
> >> receives. The dial plain is waiting for a four digits long string (my
> >> extension plan). So it send back a hangup as a invalid dial.
> >> How can I do to let Asterisk wait for the next digits without answer
> >> the call?. The MD is programmed to not wait a chunk of digits from
> >> the user,  to get a channel, and start sending the numbers.
> >> (I know I could do a IVR style configuration - answer and let the
> >> user choose the extension, but it is not my intention).
> >> Sincerely,
> >> Roger.
> >>
>
> ___
> --Bandwidth and Colocation sponsored by Easynews.com --
>
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

Re: [Asterisk-Users] TE411P

[Jean-Denis Girard]

Erwan DESVERGNES wrote:
> Did someone use a Te411p with 4 T2 in France ? I’ve got some  problem

Not exactly France, but I do have TE405P and TE110P running fine in
French Polynesia (should technically be the same network as France). The
only problem I had was the operator not configuring EuroISDN on the
incoming lines (they used an older proprieatary protocol); incoming
calls where OK, but all outigoing calls where rejected...

What is your exact problem, zaptel and zapata config, versions...?


Best regards,
-- 
Jean-Denis Girard

SysNux  Systèmes Linux en Polynésie française
http://www.sysnux.pf/   Tél: +689 483 527 / GSM: +689 797 527
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

[Asterisk-Users] is it possible to force faxdetect / disable echo cancellation for a given extension?

[Tomasz Chmielewski]

I have the newest SpanDSP setup with asterisk 1.2.

Generally, 99% of received faxes are OK, but only about 20% of faxes 
sent are delivered properly.


In zapata.conf I have set faxdetect=both, but it doesn't seem to disable 
echo cancellation (I looked into asterisk logs and it says "Enabled echo 
cancellation on channel 1, Engaged echo training on channel 1" whenever 
I fax out).


Why doesn't asterisk detect that it's faxing?

So my idea was to disable echo cancellation whenever fax number is called:


exten => 27229932,1,Answer
exten => 27229932,2,DISABLE_ECHO_CANCELLATION
exten => 27229932,3,Goto(in_fax,1)
(...)


And do the same when I sent faxes using .call files:

OPTIONS: DISABLE_ECHO_CANCELLATION
Channel: $CHANNEL/$FAXNUM
MaxRetries: 1
WaitTime: 20
Application: txfax
Data: $DATADIR/$ATTNAME.tif|caller


Is it possible to do something like that?


--
Tomek
http://wpkg.org
WPKG - software deployment and upgrades with Samba
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

[Asterisk-Users] Manager log

[Giordano Grandis]

Hi everyone,

just a question: is there a way to remove this
message on the CLI ?

 

  == Manager 'root' logged on from 127.0.0.1

  == Manager 'root' logged off from 127.0.0.1

 

Thanks

 

Giordano

 






___
--Bandwidth and Colocation sponsored by Easynews.com --

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

[Asterisk-Users] Call Progress Analysis

[Gabriel Rojas]

Nitin Joshi wrote:
> Hi All,
> I am using Asterisk 1.0.7 with an X101P analog card which is connected
to an
> Alcatel pbx. My problem is that when I place outbound calls on the zap
> channel, Asterisk returns a connect event as soon as the phone start
> ringing. This means that Asterisk is not being able to do Call Progress
> analysis on the zap channels. I have tried setting 'callprogress=yes' in
> zapata.conf but it made no difference. This problem is not there with SIP
> and IAX channels.

I have the same problem with Digium TDM cards. I've been doing pretty
extensive research and found no solution. Look at my mail [Asterisk-Users]
[Fwd: call status with FXO], few mails ahead of yours.

___
--Bandwidth and Colocation sponsored by Easynews.com --

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

Re: [Asterisk-Users] How to initiate a call from a web page?

[Matt Riddell]

Kib Eki wrote:
> Hi,
> 
> we have a html based telephonelist on our intranet site.
> Does there exist any solution to initiate a call from a link ?
> We use Polycom SIP IP phones.

If you know how to code, have a look at the sample.call file in the
/usr/src/asterisk directory.  This file can be filled out and then copied to
/var/spool/asterisk/outgoing to make a call.

-- 
Cheers,

Matt Riddell
___

http://www.sineapps.com/news.php (Daily Asterisk News - html)
http://freevoip.gedameurope.com (Free Asterisk Voip Community)
http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss)

___
--Bandwidth and Colocation sponsored by Easynews.com --

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

Re: [Asterisk-Users] Looking for Info on Asterisk scripting

[Matt Riddell]

Obelix wrote:

> I am quite familiar with Asterisk AGI, but I am looking for forums or groups
> that discuss more techniques, like the Manager API etc.
> 
> Asterisk Users only delves into Asterisk dial plans, configuration etc and
> Asterisk Dev deals with the main Asterisk itself.
> 
> I am looking for more scripting techniques

Maybe if you ask a question then someone will answer you :)

-- 
Cheers,

Matt Riddell
___

http://www.sineapps.com/news.php (Daily Asterisk News - html)
http://freevoip.gedameurope.com (Free Asterisk Voip Community)
http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss)

___
--Bandwidth and Colocation sponsored by Easynews.com --

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

Re: [Asterisk-Users] testing

[Matt Riddell]

ram wrote:
> Hi
>  
> why my posting are not accepting in this list

Don't know.

-- 
Cheers,

Matt Riddell
___

http://www.sineapps.com/news.php (Daily Asterisk News - html)
http://freevoip.gedameurope.com (Free Asterisk Voip Community)
http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss)

___
--Bandwidth and Colocation sponsored by Easynews.com --

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

Re: [Asterisk-Users] How to initiate a call from a web page?

[Dave Walker]

Look under phone home:
http://mundy.org/blog/index.php?p=63

Hope this helps

Kib Eki wrote:


Hi,

we have a html based telephonelist on our intranet site.
Does there exist any solution to initiate a call from a link ?
We use Polycom SIP IP phones.

thanks and regards,
bk

___
--Bandwidth and Colocation sponsored by Easynews.com --

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



___
--Bandwidth and Colocation sponsored by Easynews.com --

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

[Asterisk-Users] Problem with SIP register

[Diego Andrés Asenjo González]

Hi!

I'm registering an asterisk server in a Sysmaster with a SIP account.
The registration succeeds and I can establish a call that come from the
Sysmaster.

After around 80 seconds the Sysmaster sends a BYE SIP message and the
call hang up. This does not occur to the hard/soft SIP phones registered
in the sysmaster.

I debug, but the only info that I can get is the BYE message.

Thanks for your suggetions soving the problem.

Bye.

-- 
Diego Andrés Asenjo González
Universidad del Cauca
Ingeniero en Electrónica y Telecomunicaciones



signature.asc
Description: OpenPGP digital signature
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

RE: [Asterisk-Users] How to initiate a call from a web page?

[Kerry Garrison]

 
Don't you actually want to do a move instead of a copy? During a copy
Asterisk might actually pull a partial file but a move will not be detected
until the file is 100% in place. Probably not a problem unless you were
writing a very busy call center app.

Kerry Garrison
Publisher - GeekGazette.com - VOIPSpek.net
(949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com 
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt Riddell
Sent: Friday, November 25, 2005 8:19 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] How to initiate a call from a web page?

Kib Eki wrote:
> Hi,
> 
> we have a html based telephonelist on our intranet site.
> Does there exist any solution to initiate a call from a link ?
> We use Polycom SIP IP phones.

If you know how to code, have a look at the sample.call file in the
/usr/src/asterisk directory.  This file can be filled out and then copied to
/var/spool/asterisk/outgoing to make a call.

--
Cheers,

Matt Riddell
___

http://www.sineapps.com/news.php (Daily Asterisk News - html)
http://freevoip.gedameurope.com (Free Asterisk Voip Community)
http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss)

___
--Bandwidth and Colocation sponsored by Easynews.com --

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


___
--Bandwidth and Colocation sponsored by Easynews.com --

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

Re: [Asterisk-Users] Really lightweight itemised billing

[Darren Wiebe]

Do you have accountcodes in the database?  If you do, you could use 
astpp quite easily.  We could cut out most of the functionality for 
you.  Right now I don't have a way to search by date but that would be 
failry easy to add and I will be working on it soon anyways.  Drop me a 
line if you want or visit www.aleph-com.net/astpp


Darren Wiebe
[EMAIL PROTECTED]

Chris Bagnall wrote:


Good morning all,

I'm trying to find an application that'll do really lightweight billing for
Asterisk CDRs.

On our asterisk servers deployed at people's offices, we have CDRs being
logged to PostgreSQL, which can then be analysed by the staff at those
offices using a PHP-based CDR analyser. This works fine for legitimate use
verification (it's easy to spot people making hours of phone calls to their
girlfriend's mobile, for example), but it doesn't provide billing
verification.

All I'm looking to do is parse the CDRs for a given date range, lookup each
dialled number in a table to get its rate, then present a formatted list (or
even a .csv) of the person dialling (accountcode), time/date of call,
duration and total cost of call.

All of the billing applications I've seen so far are either 1) really
heavyweight designed for calling card or other charging purposes, or 2) want
me to modify the asterisk configuration to use their AGIs for dialling. It's
an overkill for what I'm after.

Before I go and write some PHP scripts to do what I'm after, has anyone
already done this and have some scripts they want to share? :-)

Thanks in advance.

Regards,

Chris
 



___
--Bandwidth and Colocation sponsored by Easynews.com --

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

[Asterisk-Users] Help with 2billing please.

[Jose M. Ramirez]

Hi list, all. Please, I need help.  Although 
already I installed a2billing, simply I cannot initiate its execution.  
Only appears this:  
 
-- Executing Answer("SIP/20-456d", "") in new 
stack
-- Executing Wait("SIP/20-456d", "2") in new 
stack
-- Executing DeadAGI("SIP/20-456d", "a2billing.php") in new 
stack
-- Launched AGI Script 
/var/lib/asterisk/agi-bin/a2billing.php
-- AGI Script a2billing.php completed, returning 
0
-- Executing Wait("SIP/20-456d", "2") in new 
stack
-- Executing Hangup("SIP/20-456d", "") in new 
stack
== Spawn extension (from-internal, 1, 5) exited non-zero on 
'SIP/20-456d'
-- Executing Macro("SIP/20-456d", "hangupcall") in new 
stack
-- Executing ResetCDR("SIP/20-456d", "w") in new 
stack
-- Executing NoCDR("SIP/20-456d", "") in new 
stack
-- Executing Wait("SIP/20-456d", "5") in new 
stack
-- Executing Hangup("SIP/20-456d", "") in new 
stack
== Spawn extension (macro-hangupcall, s, 4) exited non-zero on 
'SIP/20-456d' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on 
'SIP/20-456d'
 
and I cannot happen of there.  That lack is it 
or that I am making incorrect?
 
Regards.
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

[Asterisk-Users] Dialplan pattern match discrepancy

[Steve Davies]

Hi,

This is probably just me mis-reading the documentation, but I have
been led to believe that the '.' in extensions.conf means zero or more
digits, such that

exten => _X.,1,NoOp()

Would trigger for either a single digit, or for a longer number (as
long as it starts with a digit)

In practice (I am using 1.0.7 and 1.0.9) the '.' seems to match *one*
or more digits, so in the above example, a single digit is not matched
as expected.

Is this correct? A bug? Fixed in 1.2 ;-) ?

Thanks for any feedback on this.

Regards,
Steve
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

[Asterisk-Users] "Local Directory" feature on Polycom Soundpoint 501s

[hugolivude]

I cannot seem to get the "Local Directory" feature to work.  I've
consulted section 3.1.17 of the Administrator Guide.  It says to put a
file -directory.xml (where  is the
IP address of the phone) into the TFTP directory.  Polycom provides a
template.

The IP address for one of my phones is 192.168.0.113, so I placed the
file 192168000113-directory.xml (shown below) into my TFTP directory,
but the local directory on the phone was not updated when it rebooted.
 I think the TFTP is configured correctly because the phone has no
problem loading the firmware etc.

I thought it might have something to do DHCP, so I gave the phone a
static IP address, still no luck.  Next I tried entering the info
manually right on the phone.  I went into the "Contact Directory"
pressed "Add" and entered the infor for a contact.  I pressed "Save"
and the phone came back with "No Records".  Occasionally I see a
message "Busy.  Please try again..."

Anyone have better success?

Thanks,
Hugh







Smith
Fred
301
1
3

0
0
0
0


Jones
Bob
302
2
3

0
0
0
0


Ng
Vin
333
3
3

0
0
0
0



___
--Bandwidth and Colocation sponsored by Easynews.com --

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

Re: [Asterisk-Users] Help with 2billing please.

[Rafael R. GV]

Hi
edit your a2billing.conf and set debug level to 3, try again and send us your debug to see what happend.

rafael

On 11/25/05, Jose M. Ramirez <[EMAIL PROTECTED]> wrote:







Hi list, all. Please, I need help.  Although 
already I installed a2billing, simply I cannot initiate its execution.  
Only appears this:  
 
-- Executing Answer("SIP/20-456d", "") in new 
stack
-- Executing Wait("SIP/20-456d", "2") in new 
stack
-- Executing DeadAGI("SIP/20-456d", "a2billing.php") in new 
stack
-- Launched AGI Script 
/var/lib/asterisk/agi-bin/a2billing.php
-- AGI Script a2billing.php completed, returning 
0
-- Executing Wait("SIP/20-456d", "2") in new 
stack
-- Executing Hangup("SIP/20-456d", "") in new 
stack
== Spawn extension (from-internal, 1, 5) exited non-zero on 
'SIP/20-456d'
-- Executing Macro("SIP/20-456d", "hangupcall") in new 
stack
-- Executing ResetCDR("SIP/20-456d", "w") in new 
stack
-- Executing NoCDR("SIP/20-456d", "") in new 
stack
-- Executing Wait("SIP/20-456d", "5") in new 
stack
-- Executing Hangup("SIP/20-456d", "") in new 
stack
== Spawn extension (macro-hangupcall, s, 4) exited non-zero on 
'SIP/20-456d' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on 
'SIP/20-456d'
 
and I cannot happen of there.  That lack is it 
or that I am making incorrect?
 
Regards.

___--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing listAsterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:   
http://lists.digium.com/mailman/listinfo/asterisk-users-- rrgv
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

RE: [Asterisk-Users] "Local Directory" feature on Polycom Soundpo int 501s

[Watkins, Bradley]

The filename needs to be -directory.xml, not -directory.xml.  Grab the MAC off the back of the phone.

This is the same as for the provisioning files if you are using your TFTP
server to do that.  Also, is there a reason that you aren't using FTP?  It's
much more robust, and does not require that you pre-configure the directory
file (unless you want to fill in specific values).

- Brad

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of hugolivude
Sent: Friday, November 25, 2005 11:55 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] "Local Directory" feature on Polycom Soundpoint
501s


I cannot seem to get the "Local Directory" feature to work.  I've consulted
section 3.1.17 of the Administrator Guide.  It says to put a file -directory.xml (where  is the IP address of the
phone) into the TFTP directory.  Polycom provides a template.

The IP address for one of my phones is 192.168.0.113, so I placed the file
192168000113-directory.xml (shown below) into my TFTP directory, but the
local directory on the phone was not updated when it rebooted.  I think the
TFTP is configured correctly because the phone has no problem loading the
firmware etc.

I thought it might have something to do DHCP, so I gave the phone a static
IP address, still no luck.  Next I tried entering the info manually right on
the phone.  I went into the "Contact Directory" pressed "Add" and entered
the infor for a contact.  I pressed "Save" and the phone came back with "No
Records".  Occasionally I see a message "Busy.  Please try again..."

Anyone have better success?

Thanks,
Hugh



 


Smith
Fred
301
1
3

0
0
0
0


Jones
Bob
302
2
3

0
0
0
0


Ng
Vin
333
3
3

0
0
0
0



___
--Bandwidth and Colocation sponsored by Easynews.com --

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



The contents of this e-mail are intended for the named addressee only. It
contains information that may be confidential. Unless you are the named
addressee or an authorized designee, you may not copy or use it, or disclose
it to anyone else. If you received it in error please notify us immediately
and then destroy it. 

___
--Bandwidth and Colocation sponsored by Easynews.com --

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

[Asterisk-Users] Distinctive ring?

[Kerry Garrison]

Don't you love clients that keep asking for features after an install? 

I have a client that is asking about doing distinctive rings for external vs
internal calls. They are using Grandstream GXP-2000 phones which (although a
pain to configure) have 4 ring types. I am guessing that I would need to
figure out how to tell this particular phone to use a different ring tone
unless there is a way to send a stutter type ring to the phones. 

Any suggestions?
-Kerry


___
--Bandwidth and Colocation sponsored by Easynews.com --

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

[Asterisk-Users] CallerID not passing through to Polycom 500

[Gary MacKay]

How do I make it wait? For how long? I watched the logs but did not see anything that related to this.



Check your logs, make sure you are waiting long enough before sending
the call to the polycom.

Uf asterisk sees the CID, it should send it and it should show up on the
polycom.

Greg
-Original Message-
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Gary
MacKay
Sent: Thursday, November 24, 2005 11:19 AM
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] CallerID not passing through to Polycom 500

I have a basic system working, except for callerid. The Polycom 500 just
shows call from "Business Line" on the screen. "Business Line" is the
name of the context that line is in. How do I get it to show the
callerID on the screen instead? Yes, I have CallerID on that line and it
works on a standard analog phone.


___
--Bandwidth and Colocation sponsored by Easynews.com --

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

[Asterisk-Users] smsq sending 7 at a time ?

[Julian Lyndon-Smith]

Asterisk 1.2

We tried today to send a number of sms messages at the same time.

the smsq application seems to send 7 messages at a time, and then stops. 
If I send another sms message, then another 7 messages are sent.


Has anyone else seen this ?

Julian.
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

Re: [Asterisk-Users] "Local Directory" feature on Polycom Soundpo int 501s

[hugolivude]

I tried changing the name to the MAC address format, but still no
luck.  No contacts appear after re-boot and I still can't add them
manually either.

No particular reason for using TFTP over FTP.  I'm a hack so I just
followed the instructions at:

http://www.voip-info.org/wiki/index.php?page=Polycom+Soundpoint+IP+501

Thanks,
Hugh

On 11/25/05, Watkins, Bradley <[EMAIL PROTECTED]> wrote:
> The filename needs to be -directory.xml, not  Address>-directory.xml.  Grab the MAC off the back of the phone.
>
> This is the same as for the provisioning files if you are using your TFTP
> server to do that.  Also, is there a reason that you aren't using FTP?  It's
> much more robust, and does not require that you pre-configure the directory
> file (unless you want to fill in specific values).
>
> - Brad
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of hugolivude
> Sent: Friday, November 25, 2005 11:55 AM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] "Local Directory" feature on Polycom Soundpoint
> 501s
>
>
> I cannot seem to get the "Local Directory" feature to work.  I've consulted
> section 3.1.17 of the Administrator Guide.  It says to put a file  address>-directory.xml (where  is the IP address of the
> phone) into the TFTP directory.  Polycom provides a template.
>
> The IP address for one of my phones is 192.168.0.113, so I placed the file
> 192168000113-directory.xml (shown below) into my TFTP directory, but the
> local directory on the phone was not updated when it rebooted.  I think the
> TFTP is configured correctly because the phone has no problem loading the
> firmware etc.
>
> I thought it might have something to do DHCP, so I gave the phone a static
> IP address, still no luck.  Next I tried entering the info manually right on
> the phone.  I went into the "Contact Directory" pressed "Add" and entered
> the infor for a contact.  I pressed "Save" and the phone came back with "No
> Records".  Occasionally I see a message "Busy.  Please try again..."
>
> Anyone have better success?
>
> Thanks,
> Hugh
>
>
> 
>  
> 
> 
> Smith
> Fred
> 301
> 1
> 3
> 
> 0
> 0
> 0
> 0
> 
> 
> Jones
> Bob
> 302
> 2
> 3
> 
> 0
> 0
> 0
> 0
> 
> 
> Ng
> Vin
> 333
> 3
> 3
> 
> 0
> 0
> 0
> 0
> 
> 
> 
> ___
> --Bandwidth and Colocation sponsored by Easynews.com --
>
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
> The contents of this e-mail are intended for the named addressee only. It
> contains information that may be confidential. Unless you are the named
> addressee or an authorized designee, you may not copy or use it, or disclose
> it to anyone else. If you received it in error please notify us immediately
> and then destroy it.
>
> ___
> --Bandwidth and Colocation sponsored by Easynews.com --
>
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

Re: [Asterisk-Users] Multiple HFC-PCI cards in mixed modes (TE+NT) won't work!...

[Francesco Peeters]

On Fri, November 25, 2005 9:29, Kristof Hardy said:
> Francesco Peeters wrote:
>> I compiled 1.2 and bristuff 0.3.0 Pre1 yesterday late and that now seems
>> to work! * is up and running *with* 2nd card in NT mode...
>
> Nice to hear *1.2 and bristuff 0.3pre1 makes a difference..
>
>

Just switched the config around so the NT mode card was active, and I got
dialtone, etc.  :-)

Now the hardest part: Connecting the HFC clock lines of both cards to
eachother (the soldering iron is ready, just have to shutdown the server
and take out the cards) because I need the NT card to be slave to the TE
card.
(Both cards running their own clocks doesn't seem to agree with asterisk
anyway, so it's a good thing I was planning on this all along!)

This is the most critical part of the test: Once I start soldering on the
cards there is no more returning, no warranty, etc.

But I know now that both modes work with 1.2, so it's time to take that
plunge!

I'll let the list know when I'm done!

-- 
Francesco Peeters

GPG Key = AA69 E7C6 1D8A F148 160C  D5C4 9943 6E38 D5E3 7704
If your program doesn't recognize my signature, please visit
http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate.
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

Re: [Asterisk-Users] Send fax using PRI connection to TE405P

[Rob McKrill]

 Anyone has experiences with sending faxes using Asterisk and a TE405P 
Digium card (or similar PRI) with a PRI connection?




Using HylaFax and a PRI card such as the Patton 2977, Eicon Diva sending 
  faxes works very well.  There is a new project out there called 
IAXModem (written by Lee Howard) which apparently utilizes the SpanDSP 
library (written by Steve Underwood) and allows you to interface to 
HylaFax without the extra card.


You might want to search the Asterisk-Users list archives for IAXModem. 
 There was a thread last week about how to get it to work with Asterisk.

___
--Bandwidth and Colocation sponsored by Easynews.com --

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

Re: [Asterisk-Users] Philippines Asterisk users, anyone?

[Michael Kenjie Nukui]

Im one! kenjie pre, i am just a new user of asterisk.

regards,

kenjie nukui
[EMAIL PROTECTED]On 11/25/05, Angelito Manansala <[EMAIL PROTECTED]
> wrote:--Best Regards,Angelito ManansalaMobile: +639175425807
DID: (+63) 44 7906770msn: [EMAIL PROTECTED]skype: bulcrack___--Bandwidth and Colocation sponsored by 
Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

Re: [Asterisk-Users] Bad quality

[Mojo with Horan & Company, LLC]

what hardware is involved?  Are you using a hard or soft phone?  if 
hard, an ip phone or an ATA adapter?  What country are you in?


Finally, please run

/usr/src/zaptel/zttest -v

and watch it for a while.  If the average result is less than 99.98%, 
consult http://www.voip-info.org/wiki/view/Asterisk+PCI+bus+Troubleshooting


This could be that your system is too busy for asterisk.

Mojo

Pablo Allietti wrote:

hi all, i have asterisk configured and working but the quality is very
poor. i ear noise and braks in the voice when the people talk to me, and
the people that eared me have the same problem any recommendation?
any files you need to post?


--
Mojo <[EMAIL PROTECTED]>
Office Manger, Horan & Company, LLC
(907) 747- x112
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

[Asterisk-Users] A2Billing questions are off topic for this list

[Kevin P. Fleming]

Please move _all_ discussion regarding A2Billing (and the mechanics of 
using any billing system unless the discussion is explicitly 
Asterisk-related) to some other mailing list; this mailing list not a 
support forum for add-on packages.


Thanks!
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

[Asterisk-Users] CallerID not passing through to Polycom 500 (SOLVED, sort of)

[Gary MacKay]

After playing around with the CALLERID(number) and 
CALLERID(name) variables and things, I find that asterisk is sending
the "name" to my phone and the name is "unknown". I added a line
exten => _X.,Set(CALLERID(name)=${CALLERIDNUM})  and now it shows
the number. Is this the right way to do this?


Check your logs, make sure you are waiting long enough before sending
the call to the polycom.

Uf asterisk sees the CID, it should send it and it should show up on the
polycom.

Greg
-Original Message-
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Gary
MacKay
Sent: Thursday, November 24, 2005 11:19 AM
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] CallerID not passing through to Polycom 500

I have a basic system working, except for callerid. The Polycom 500 just
shows call from "Business Line" on the screen. "Business Line" is the
name of the context that line is in. How do I get it to show the
callerID on the screen instead? Yes, I have CallerID on that line and it
works on a standard analog phone.


___
--Bandwidth and Colocation sponsored by Easynews.com --

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

[Asterisk-Users] Bristuff: qozap.o error

[asterisk183]

 I have installed bristuff 0.3.0 for Asterisk 1.2 with kernel 2.4, but when I doing : insmod qozap.o  the shell show this messagge: qozap.o: qozap.o: unresolved symbol free_irq_Rsmp_f20dabd8 qozap.o: qozap.o: unresolved symbol pci_find_device_Rsmp_c584f4e3 qozap.o: qozap.o: unresolved symbol __request_region_Rsmp_1a1a4f09 qozap.o: qozap.o: unresolved symbol pci_disable_device_Rsmp_95846005 qozap.o: qozap.o: unresolved symbol iounmap_Rsmp_5fb196d4 qozap.o: qozap.o: unresolved symbol pci_enable_device_Rsmp_1bc741d2 qozap.o: qozap.o: unresolved symbol sprintf_Rsmp_1d26aa98 qozap.o: qozap.o: unresolved symbol __ioremap_Rsmp_9eac042a qozap.o: qozap.o: unresolved symbol pci_write_config_word_Rsmp_f23d8795 qozap.o: qozap.o: unresolved symbol printk_Rsmp_1b7d4074 qozap.o: qozap.o: unresolved symbol kfree_Rsmp_037a0cba qozap.o: qozap.o: unresolved symbol ioport_resource_Rsmp_865ebccd qozap.o: qoz
 ap.o:
 unresolved symbol kmalloc_Rsmp_93d4cfe6 qozap.o: qozap.o: unresolved symbol request_irq_Rsmp_0c60f2e0 qozap.o: qozap.o: unresolved symbol __release_region_Rsmp_d49501d4  What I can doing for resolving this problem? I trying copy qozap.o in  /lib/modules/`uname -r`/misc/ but don't resolve the problem  Help  Thanks 
		Yahoo! Mail: gratis 1GB per i messaggi, antispam, antivirus, POP3___
--Bandwidth and Colocation sponsored by Easynews.com --

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

RE: [Asterisk-Users] "Local Directory" feature on Polycom Soundpo int 501s

[Watkins, Bradley]

Hrmmm... I'm not sure how much more help I can be on this exactly.  For all
of my users, we use FTP and the files get created and updated automatically.
I should note that this is with all IP600s/601s but this should be the same
even for the 501s.

- Brad

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of hugolivude
Sent: Friday, November 25, 2005 12:44 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] "Local Directory" feature on Polycom Soundpo
int 501s


I tried changing the name to the MAC address format, but still no luck.  No
contacts appear after re-boot and I still can't add them manually either.

No particular reason for using TFTP over FTP.  I'm a hack so I just followed
the instructions at:

http://www.voip-info.org/wiki/index.php?page=Polycom+Soundpoint+IP+501

Thanks,
Hugh

On 11/25/05, Watkins, Bradley <[EMAIL PROTECTED]> wrote:
> The filename needs to be -directory.xml, not  Address>-directory.xml.  Grab the MAC off the back of the phone.
>
> This is the same as for the provisioning files if you are using your 
> TFTP server to do that.  Also, is there a reason that you aren't using 
> FTP?  It's much more robust, and does not require that you 
> pre-configure the directory file (unless you want to fill in specific 
> values).
>
> - Brad
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> hugolivude
> Sent: Friday, November 25, 2005 11:55 AM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] "Local Directory" feature on Polycom Soundpoint
> 501s
>
>
> I cannot seem to get the "Local Directory" feature to work.  I've 
> consulted section 3.1.17 of the Administrator Guide.  It says to put a 
> file  address>-directory.xml (where  is the IP address of 
> address>the
> phone) into the TFTP directory.  Polycom provides a template.
>
> The IP address for one of my phones is 192.168.0.113, so I placed the 
> file 192168000113-directory.xml (shown below) into my TFTP directory, 
> but the local directory on the phone was not updated when it rebooted.  
> I think the TFTP is configured correctly because the phone has no 
> problem loading the firmware etc.
>
> I thought it might have something to do DHCP, so I gave the phone a 
> static IP address, still no luck.  Next I tried entering the info 
> manually right on the phone.  I went into the "Contact Directory" 
> pressed "Add" and entered the infor for a contact.  I pressed "Save" 
> and the phone came back with "No Records".  Occasionally I see a 
> message "Busy.  Please try again..."
>
> Anyone have better success?
>
> Thanks,
> Hugh
>
>
> 
>  
> 
> 
> Smith
> Fred
> 301
> 1
> 3
> 
> 0
> 0
> 0
> 0
> 
> 
> Jones
> Bob
> 302
> 2
> 3
> 
> 0
> 0
> 0
> 0
> 
> 
> Ng
> Vin
> 333
> 3
> 3
> 
> 0
> 0
> 0
> 0
> 
> 
> 
> ___
> --Bandwidth and Colocation sponsored by Easynews.com --
>
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com 
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
> The contents of this e-mail are intended for the named addressee only. 
> It contains information that may be confidential. Unless you are the 
> named addressee or an authorized designee, you may not copy or use it, 
> or disclose it to anyone else. If you received it in error please 
> notify us immediately and then destroy it.
>
> ___
> --Bandwidth and Colocation sponsored by Easynews.com --
>
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com 
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

[Asterisk-Users] Re: think people dont help that easily

[vivek]

With warm regards.

Vivek J. Joshi.

[EMAIL PROTECTED]
Trikon electronics Pvt. Ltd.

--Truth springs from argument amongst friends.



[EMAIL PROTECTED] wrote:
>Hello friends, 
> I have a strange problem. I am using asterisk 1.2 and asterisk addons 1.2. I 
have three SIP phones and one H323 phones connected to asterisk. The problem is 
that when I dial an invalid extension from H323 phones, I get the invalid 
extension message with exten => i... in that context but this does not happen 
with the SIP phones. All I get is something like an engaged tone from the SIP 
phones. Also I am able to dial and transfer between SIP and H323 phones. I am 
not able to figure out whats wrong. None of them are behind the NAT. All of 
them and the asterisk server are on private-ip.
> I also tried  "sip debug" from the command line and dial an invlaid extension 
from the SIP phone and get nothing but a 
>"SIP/2.0 404 Not Found" in the o/p. But it then dosent fall to the exten => i 
>or 
exten => s.
>My conf. files are as under:-
>
>extensions.conf:-
>[incoming]
>exten => s,1,Answer ; Answer the line.
>exten => s,n,Set(TIMEOUT(digit)=5)  ; Set Digit Timeout to 
>5 
seconds.
>exten => s,n,Set(TIMEOUT(response)=10)  ; Set Response Timeout 
to 10 seconds.
>exten => s,n(restart),BackGround(demo-congrats) ; Play a 
>congratulatory 
message.
>exten => s,n,WaitExten(5)   ; Wait for an 
>extension 
to be dialed.
>exten => s,n,Dial(SIP/192.168.1.196,100,t)  , Dial the operator.
>
>exten => i,1,Playback(invalid)  ; "That's not valid, 
>try 
again".
>
>[default]
>include => incoming ; Instead of demo in 
>the 
sample, there is incoming.
>
>[testing]
>include => parkedcalls
>
>exten => s,1,Playback(invalid)  ; When this is 
>present, 
invalid extension from h323 comes here or 
>;;; exten => i,1,Playback(invalid)  ;;;even this did not work.   
>;;  H323 Phones  ;;
>exten => 61,1,Dial(OOH323/192.168.1.194,20|t)  ;ip=h323
>;;  SIP Phones   ;;
>exten => 62,1,Dial(SIP/62,20|t);new-gray=sip
>exten => 63,1,Dial(SIP/63,20|t);old-gray=sip
>exten => 64,1,Dial(SIP/64,20|t);ip=sip
>
>ooh323.conf:-
>context=testing
>disallow=all
>allow=ulaw
>allow=alaw
>dtmfmode=h245alphanumeric
>[61]
>type=friend
>ip=192.168.1.194
>context=testing
>
>sip.conf:-
>[general]
>context=default
>bindport=5060 
>bindaddr=0.0.0.0
>srvlookup=yes
>disallow=all   
>allow=alaw
>allow=ulaw  
>musicclass=default
>dtmfmode = rfc2833
>
>[63]
>type=friend
>context=testing ; context above where the extensions dialable by this 
are defined. 
>username=63
>secret=1234
>host=dynamic
>defaultip=192.168.1.192 ; ip address of this phone
>canreinvite=no
>callgroup=1 ; We are in caller groups 1
>pickupgroup=1   ; We can do call pick-p for call group 1
>;; rest of the sip users are configured in the same way.
>
>Help will be very much appreciated. Kindly help. I am totally confused as to 
where the fault is. 
>
>
>
>
>
>
>With warm regards.
>
>Vivek J. Joshi.
>
>[EMAIL PROTECTED]
>Trikon electronics Pvt. Ltd.
>
>--Truth springs from argument amongst friends.
>
>
>

___
--Bandwidth and Colocation sponsored by Easynews.com --

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

Re: [Asterisk-Users] How to initiate a call from a web page?

[Michiel van Baak]

On 08:26, Fri 25 Nov 05, Kerry Garrison wrote:
>  
> Don't you actually want to do a move instead of a copy? During a copy
> Asterisk might actually pull a partial file but a move will not be detected
> until the file is 100% in place. Probably not a problem unless you were
> writing a very busy call center app.

I think in that case you can better use the manager
interface. That way you don't raise the disk io to an insane
level.

The call files are nice when the webapp and asterisk are on
the same machine. As soon as you split those 2 you are best
of with the manager interface.

Just my 2 cents
-- 
Michiel van Baak
http://michiel.vanbaak.info
[EMAIL PROTECTED]
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D

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

___
--Bandwidth and Colocation sponsored by Easynews.com --

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

Re: [Asterisk-Users] speex & ilbc

[Bharath]

have you added allow=speex & allow = ilbc in the sip & iax conf files ?
On 11/25/05, Alejandro Vargas <[EMAIL PROTECTED]> wrote:
I'm testing [EMAIL PROTECTED] 2.0 beta 6.I'm checking de different codecs but with speex and ilbc I don'treceive any sound. I tested xtensofphone and iaxComm. With both I hasthe same problem: good sound with ulaw and gsm, but no sound with
xpeex and ilbc. Do I need to change some config for using it?--Alejandro Vargas___--Bandwidth and Colocation sponsored by Easynews.com
 --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:   http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

Re: [Asterisk-Users] Dialplan pattern match discrepancy

[Daniel Wright]

Steve Davies wrote:

Hi,

This is probably just me mis-reading the documentation, but I have
been led to believe that the '.' in extensions.conf means zero or more
digits, such that

exten => _X.,1,NoOp()

Would trigger for either a single digit, or for a longer number (as
long as it starts with a digit)

In practice (I am using 1.0.7 and 1.0.9) the '.' seems to match *one*
or more digits, so in the above example, a single digit is not matched
as expected.

Is this correct? A bug? Fixed in 1.2 ;-) ?

Thanks for any feedback on this.

Regards,
Steve
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

  

Hi Steve,

The period is match 1 or more characters(can be a number or letter).
So in your example, you are saying first match a number 0-9, then match 
any one or more characters.


Dan
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

[Asterisk-Users] Loss of Registration for SIP Trunks

[Scott Clements]

HI List,

You'll have to pardon the newbieness of this question, I was editing
the sip.conf file on my asterisk server yesterday, and now none of my
asterisk trunks will connect. From my knowledge sip.conf does not
effect registration, but there have been no other changes at all. Below
is my sip.conf, and some other CLI info. If anone has some thoughts
please let me know.


[general]

port = 5060   ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0    ; Address to bind to (all addresses on machine)
disallow=all
allow=g729
allow=ulaw
allow=alaw
context=from-pstn
;context = from-sip-external ; Send unknown SIP callers to this context
callerid = Unknown
;dtmfmode=rfc2833
;relaxdtmf=yes

#include sip_nat.conf
#include sip_custom.conf
#include sip_additional.conf




cee*CLI> sip show registry
Host   
Username   Refresh State


cee*CLI> sip show peers
Name/username   
Host   
Dyn Nat ACL
Mask
Port Status
sip-out-test/02 
202.177.222.24 
255.255.255.255  5060 Unmonitored
127/127 
(Unspecified)   
D 
255.255.255.255  0   
Unmonitored
126/126 
(Unspecified)   
D 
255.255.255.255  0   
Unmonitored




I have tried removing the trunks, confirmed the username and passwords
for the trunks are ok. I am totally stumped as to what would cause it. 

If anyone can help it'd be great :)

SCott
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

Re: [Asterisk-Users] Looking for Windows based Asterisk

[C F]

Well I disagree on the untraedit, since vi does a far better job. :)

On 11/25/05, Guido Hecken <[EMAIL PROTECTED]> wrote:
> also add winscp and ultraedit to your windows system, it works great.
> http://winscp.net/eng/index.php
> http://www.ultraedit.com/
>
> Regards
>
> Guido Hecken
>
> > > Without putty, my windows would be meaningless.
> > >
> > > PaulH
> > >
> > Subtle Paul! but nice! :)
> > Mike
> > UK
>
> ___
> --Bandwidth and Colocation sponsored by Easynews.com --
>
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

[Asterisk-Users] Truncated CDR records

[Christian B]

Hello Group!

While parsing my cdrs of the last week, i realized that approx. 1 in
100 _successful_ outbound zap-calls are recorded with a truncated
destination number in the verbose logs and in the cdrs.

Several digits are simply missing. eg 0049 is recorded as
0049. I have received cdrs from the telco, and have verified that the
"long version" was definitely dialed, and answered.

Most of my traffic (99%) is via DISA, and so far this behaviour has
only surfaced with outbound DISA calls. I can't be certain that its a
DISA only problem, though.

I've added the dialplan that applies to these calls(dialplan_disa.txt), the agis
that are beeing executed(disa-usergroup1.sh, disacut.sh), samples of verbose
logs for wrong numbers, cdr's and corresponding cdr's from my
telco(examples_cdr_verboselog.txt). I've upgraded to 1.2 final on 25.11.2000
03:00am in the hope this would be resolved but also today i have faulty records.

Hope someone can give me information on this issue.

regards
christian
[disa]
include => disa-verified

exten => s, 1, AGI(disa,${CALLERIDNUM})   ;this agi verfies if the users are 
authorized(for personal use)
exten => s, 2, NoOp(${ERG}) 
;show me the answer the agi sent
exten => s, 3, GotoIf($["${ERG}" = "0"]?disa-verified|s|100:)
exten => s, 4, AGI(disa-usergroup1,${CALLERIDNUM}) ;this agi verfies if the 
users are authorized(for users belonging to usergroup1)
exten => s, 5, NoOp(${ERG})
exten => s, 6, GotoIf($["${ERG}" = "0"]?disa-verified|s|200:)
exten => s, 7, SetAccount(01)   
;if the calleridnum is in no list, give him a chance to authorize
exten => s, 8, Set(TIMEOUT(digit)=5)
exten => s, 9, Set(TIMEOUT(response)=15)
exten => s, 10, Authenticate(5678345)
exten => s, 11, DISA(no-password|disa2)


[disa-verified]
exten => s, 200, SetAccount(123456789)  ;Accountnumber of usergroup1
exten => s, 201, Set(TIMEOUT(digit)=5)
exten => s, 202, Set(TIMEOUT(response)=15)
exten => s, 203, DISA(no-password|disa2)

[disa2]
;

exten => _X., 1, NoOp(${EXTEN}) 
;Added on 24.11.2005 due to errorenous records(cut offs)
exten => _X., 2, AGI(disacut,${EXTEN})  ;agi to 
cut away * and # dialed by mistake
exten => _X., 3, GotoIf($["${CALLERIDNUM:0:1}" = "0"]?100:)
exten => _X., 4, GotoIf($["${CALLINGPRES}" = "33"]?200:)
exten => _X., 5, GotoIf($["${EXTEN:0:5}" = "43650"]?200:)
exten => _X., 6, GotoIf($["${EXTEN:0:4}" = "0650"]?200:);these 
calls will be sent with a prohib presbit
exten => _X., 7, SetCIDNum(${CALLERIDNUM})  ;i know this is 
useless, used to be something different earlier
exten => _X., 8, SetCDRUserField(${MODCLI})   ;MODCLI received from agi 
disacut(without # and *) for proper cdr's
exten => _X., 9, Set(FROM_EXTEN=${EXTEN})   ;another relict 
from earlier days
exten => _X., 10, NoOp(${EXTEN})
;show me again what the user has dialed! added on 24.11.2005
exten => _X., 11, Dial(Zap/G1/${MODCLI}|120|gH) ;dial whatever disacut gives us
exten => _X., 12, GotoIf($["${DIALSTATUS}" != "ANSWER"]?s-${DIALSTATUS}|1)  
;if the call failed, play some sounds and get back to authentication
exten => _X., 13, Playback(beep)
exten => _X., 14, Goto(disa,s,1)
exten => _X., 15, Hangup

exten => _X., 200, SetCallerPres(prohib)
exten => _X., 201, NoOp(${CALLINGPRES}) ;relict
exten => _X., 202, SetCIDNum(${CALLERIDNUM});relict
exten => _X., 203, SetCDRUserField(${MODCLI})   ;MODCLI received from 
agi disacut(without # and *) for proper cdr's
exten => _X., 204, Set(FROM_EXTEN=${MODCLI});relict
exten => _X., 205, NoOp(${EXTEN})   
;show me what the user has dialed! added on 24.11.2005
exten => _X., 206, Dial(Zap/G1/${MODCLI}|120|gH);dial whatever disacut 
gives us
exten => _X., 207, Set(FROM_EXTEN=${EXTEN}) ;relict
exten => _X., 208, GotoIf($["${DIALSTATUS}" != "ANSWER"]?s-${DIALSTATUS}|1) ;if 
the call failed, play some sounds and get back to authentication
exten => _X., 209, Playback(beep)
exten => _X., 210, Goto(disa,s,1)
exten => _X., 211, Hangup

exten => s-NOANSWER, 1, Zapateller
exten => s-NOANSWER, 2, Playback(that-is-not-rec-phn-num)
exten => s-NOANSWER, 3, Playback(please-try-again)
exten => s-NOANSWER, 4, Goto(disa,s,1)
exten => s-NOANSWER, 5, Hangup

exten => s-BUSY, 1, Playtones(Congestion)
exten => s-BUSY, 2, Wait(5)
exten => s-BUSY, 3, Playback(the-number-u-dialed)
exten => s-BUSY, 4, Playback(is-curntly-busy)
exten => s-BUSY, 5, Playback(please-try-again)
exten => s-BUSY, 6, Goto(disa,s,1)

exten => s-CONGESTION, 1, Playtones(Congestion)
exten => s-CONGESTION, 2, Wait(2)
exten => 

[Asterisk-Users] Narrowing RTP port range

[Tyler]

Hello everyone..

I'm trying to lock down my asterisk install as much as possible and I
keep reading about people saying 'you can narrow the range of ports in
rtp.con' (by default it's from 1 to 2 I think).

My question is this - how much can I narrow it down?  Can I narrow it to
10 ports, or can the ports not be reused for additional conversations?

I guess what I'm asking is - does the number of ports in the range have
anything to do with the number of simultaneous connections or anything
like that?

Thanks again,

tf.



___
--Bandwidth and Colocation sponsored by Easynews.com --

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

Re: RE : [Asterisk-Users] Asterisk doesn't start

[snacktime]

On 11/25/05, harry gaillac <[EMAIL PROTECTED]> wrote:
> Try to post your problem to asterisk-dev

Hmm that seems to be your solution for just about everything doesn't
it Harry? :)

I think the problem is that asterisk-addons got built out of order or
didn't get rebuilt at all, but I can't remember for sure. 
asterisk-addons  has to be built after asterisk.  I have 1.2 running
on freebsd 5.4-STABLE and 4.11 without any issues.

Chris
___
--Bandwidth and Colocation sponsored by Easynews.com --

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

Re: [Asterisk-Users] Problem with SIP register

[Baris Simsek]

Diego Andrés Asenjo González wrote:


Hi!

I'm registering an asterisk server in a Sysmaster with a SIP account.
The registration succeeds and I can establish a call that come from the
Sysmaster.

After around 80 seconds the Sysmaster sends a BYE SIP message and the
call hang up. This does not occur to the hard/soft SIP phones registered
in the sysmaster.

I debug, but the only info that I can get is the BYE message.

Thanks for your suggetions soving the problem.

Bye.
 


Hi,

Enable SIP debug and check which peer sends BYE at first.

After call establishment, can you hear voice for 80 sec.? What about RTP 
in this duration?


--
Baris Simsek
http://www.enderunix.org/simsek/


___
--Bandwidth and Colocation sponsored by Easynews.com --

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

Re: [Asterisk-Users] "Local Directory" feature on Polycom Soundpoint 501s

[Jerry Jones]

Ethernet address != IP address

Ethernet address = MAC address

Look on the bottom of your phone

On Nov 25, 2005, at 10:55 AM, hugolivude wrote:


I cannot seem to get the "Local Directory" feature to work.  I've
consulted section 3.1.17 of the Administrator Guide.  It says to put a
file -directory.xml (where  is the
IP address of the phone) into the TFTP directory.  Polycom provides a
template.

The IP address for one of my phones is 192.168.0.113, so I placed the
file 192168000113-directory.xml (shown below) into my TFTP directory,
but the local directory on the phone was not updated when it rebooted.
 I think the TFTP is configured correctly because the phone has no
problem loading the firmware etc.

I thought it might have something to do DHCP, so I gave the phone a
static IP address, still no luck.  Next I tried entering the info
manually right on the phone.  I went into the "Contact Directory"
pressed "Add" and entered the infor for a contact.  I pressed "Save"
and the phone came back with "No Records".  Occasionally I see a
message "Busy.  Please try again..."

Anyone have better success?

Thanks,
Hugh







Smith
Fred
301
1
3

0
0
0
0


Jones
Bob
302
2
3

0
0
0
0


Ng
Vin
333
3
3

0
0
0
0



___
--Bandwidth and Colocation sponsored by Easynews.com --

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


___
--Bandwidth and Colocation sponsored by Easynews.com --

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

Re: [Asterisk-Users] Philippines Asterisk users, anyone?

[John Fraser]

Im one! John, also a new user.  

cheers

John

[EMAIL PROTECTED]

On Fri, 25 Nov 2005 18:54:02 +0100, Michael Kenjie Nukui wrote
> Im one! kenjie pre, i am just a new user of asterisk.
> 
> regards,
> 
> kenjie nukui
> [EMAIL PROTECTED]
> 
> On 11/25/05, Angelito Manansala <[EMAIL PROTECTED] > wrote:
--
> Best Regards,
> Angelito Manansala
> Mobile: +639175425807
> DID: (+63) 44 7906770
> msn: [EMAIL PROTECTED]
> skype: bulcrack
> ___
> --Bandwidth and Colocation sponsored by Easynews.com --
> 
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users 
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
> 






___
--Bandwidth and Colocation sponsored by Easynews.com --

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

[Asterisk-Users] Asterisk and Siemens HiPath 3750 issues

[Humberto Aicardi]

Hi,

   I'm currently facing some issues regarding echo between the asterisk 
box and the 3750, here is my scenario:


TELCO  --> Asterisk --> HiPath 3750
(E1) (TE210P)
 |
   SIP PHONES

   When I dial from a SIP phone to a Telco number it works fine, when I 
dial from a SIP to a HiPath extension I get too much echo. When dialing 
from the 3750 to the telco I also get echo. I have included my 
configuration file as well as other info.


   Has anyone had experience integrating asterisk with a 3750? Could 
anyone share the zaptel.conf e zapata.conf files?


Thanks in advance,
Humberto

zaptel.conf

# ISDN PRI - Telco
span=1,1,0,ccs,hdb3,crc4
bchan=1-15
dchan=16
bchan=17-31
loadzone=us
defaultzone=us

# HiPath 3750
span=2,2,0,ccs,hdb3,crc4
bchan=32-46
dchan=47
bchan=48-62
loadzone=us
defaultzone=us

zapata.conf


[channels]
language=br
facilityenable = yes
jitterbuffers=8
switchtype=euroisdn
pridialplan=unknown

rxwink=300  ; Atlas seems to use long (250ms) winks
usecallerid=yes
usecallingpres=yes
hidecallerid=no
callerid=asreceived
musiconhold=default
transfer=yes
cancallforward=yes
callreturn=yes
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes

rxgain=-1.0
txgain=-1.0

; From Telefonica
context=from-pstn
signalling=pri_cpe
accountcode=inbound
echocancel=128
echocancelwhenbridged=yes
echotraining=800
faxdetect=no

group=0
channel => 1-15,17-31

; from PABX
signalling=pri_net
context=from-pabx
accountcode=outbound
echocancel=128
echocancelwhenbridged=yes
echotraining=800
faxdetect=no

group=1
channel => 32-46,48-62

# lsmod
Module  Size  Used by
wct4xxp63168  62
zaptel207364  127 wct4xxp
radeon125637  2
md5 4033  1
ipv6  234881  14
autofs423237  0
tun 9153  1
sunrpc159269  1
crc_ccitt   2113  1 zaptel
microcode   6881  0
dm_mirror  27825  0
dm_mod 56661  1 dm_mirror
hw_random   5845  0
e1000  93101  0
floppy 58481  0
ext3  116809  2
jbd71385  1 ext3
ata_piix9413  3
libata 44957  1 ata_piix
sd_mod 17217  4
scsi_mod  121293  2 libata,sd_mod

#cat /proc/interrupts
=
 CPU0
0:   75799765  XT-PIC  timer
1: 10  XT-PIC  i8042
2:  0  XT-PIC  cascade
3:6675655  XT-PIC  eth0, [EMAIL PROTECTED]::06:05.0
5:  76837  XT-PIC  libata
8:  1  XT-PIC  rtc
10:   75733239  XT-PIC  wct2xxp
12: 58  XT-PIC  i8042
14: 681710  XT-PIC  ide0
NMI:  0
ERR:  0

#cat /proc/zaptel/1
=
Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span 1" HDB3/CCS/CRC4

 1 TE2/0/1/1 Clear (In use)
 2 TE2/0/1/2 Clear (In use)
 3 TE2/0/1/3 Clear (In use)
 4 TE2/0/1/4 Clear (In use)
 5 TE2/0/1/5 Clear (In use)
 6 TE2/0/1/6 Clear (In use)
 7 TE2/0/1/7 Clear (In use)
 8 TE2/0/1/8 Clear (In use)
 9 TE2/0/1/9 Clear (In use)
10 TE2/0/1/10 Clear (In use)
11 TE2/0/1/11 Clear (In use)
12 TE2/0/1/12 Clear (In use)
13 TE2/0/1/13 Clear (In use)
14 TE2/0/1/14 Clear (In use)
15 TE2/0/1/15 Clear (In use)
16 TE2/0/1/16 HDLCFCS (In use)
17 TE2/0/1/17 Clear (In use)
18 TE2/0/1/18 Clear (In use)
19 TE2/0/1/19 Clear (In use)
20 TE2/0/1/20 Clear (In use)
21 TE2/0/1/21 Clear (In use)
22 TE2/0/1/22 Clear (In use)
23 TE2/0/1/23 Clear (In use)
24 TE2/0/1/24 Clear (In use)
25 TE2/0/1/25 Clear (In use)
26 TE2/0/1/26 Clear (In use)
27 TE2/0/1/27 Clear (In use)
28 TE2/0/1/28 Clear (In use)
29 TE2/0/1/29 Clear (In use)
30 TE2/0/1/30 Clear (In use)
31 TE2/0/1/31 Clear (In use)

# cat /proc/zaptel/2
=
Span 2: TE2/0/2 "T2XXP (PCI) Card 0 Span 2" HDB3/CCS/CRC4

32 TE2/0/2/1 Clear (In use)
33 TE2/0/2/2 Clear (In use)
34 TE2/0/2/3 Clear (In use)
35 TE2/0/2/4 Clear (In use)
36 TE2/0/2/5 Clear (In use)
37 TE2/0/2/6 Clear (In use)
38 TE2/0/2/7 Clear (In use)
39 TE2/0/2/8 Clear (In use)
40 TE2/0/2/9 Clear (In use)
41 TE2/0/2/10 Clear (In use)
42 TE2/0/2/11 Clear (In use)
43 TE2/0/2/12 Clear (In use)
44 TE2/0/2/13 Clear (In use)
45 TE2/0/2/14 Clear (In use)
46 TE2/0/2/15 Clear (In use)
47 TE2/0/2/16 HDLCFCS (In use)
48 TE2/0/2/17 Clear (In use)
49 TE2/0/2/18 Clear (In use)
50 TE2/0/2/19 Clear (In use)
51 TE2/0/2/20 Clear (In use)
52 TE2/0/2/21 Clear (In use)

[Asterisk-Users] Asterisk callback system

[chawki hammoud]

Hi list:
what are the steps to do to asterisk to be ready fro
callback system?



__ 
Yahoo! Music Unlimited 
Access over 1 million songs. Try it free. 
http://music.yahoo.com/unlimited/
___
--Bandwidth and Colocation sponsored by Easynews.com --

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