[Asterisk-Users] Re: Contents of Asterisk-Users digest...
Will Velez
wvelez at 12.cotas.com
Thu Feb 2 07:47:32 MST 2006
-----Mensaje original-----
De: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com]En nombre de
asterisk-users-request at lists.digium.com
Enviado el: jueves, 02 de febrero de 2006 10:15
Para: asterisk-users at lists.digium.com
Asunto: Asterisk-Users Digest, Vol 19, Issue 15
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Today's Topics:
1. DeadAGI variables confusion (Dave Brooks)
2. SV: [Asterisk-Users] Outbound Caller ID number on E1
(jan.sarin at securia.se)
3. Outbound Call & SIP Results (David Brazier)
4. Re: Outbound Caller ID number on E1 (Steve Underwood)
5. Delaying media stream by short period after 183 is sent
(Mark van Kerkwyk)
6. RE: RE: [Asterisk-Users] Blocked Callerid (Alexander Lopez)
7. Anyone know a good ITSP in Canada that supports *? (hugolivude)
8. [Fwd: Re: [Asterisk-Users] Asterisk for Call Center (missing
reference)] (Rodrigo P. Telles)
9. Re: Outbound Call & SIP Results (Olle E Johansson)
10. Re: Anyone know a good ITSP in Canada that supports *?
(Andrew Kohlsmith)
11. Re: DeadAGI variables confusion (Tony Mountifield)
12. Re: Anyone know a good ITSP in Canada that supports *?
(Dovid Bender)
13. Call completes but then drops? (Matt)
14. Asterisk on laptop connected to POTS line (Dovid Bender)
15. Regarding cdr_manager.conf (Victor Alvarez)
16. Re: Asterisk on laptop connected to POTS line (Tzafrir Cohen)
17. RE: Asterisk on laptop connected to POTS line (Damon Estep)
18. RE: Asterisk on laptop connected to POTS line (Jonathan k. Creasy)
19. RE: Asterisk on laptop connected to POTS line (Alexander Lopez)
20. Re: TE411P or TE406P (Kevin P. Fleming)
21. Re: meetme and dtmf (Kevin P. Fleming)
22. Re: Help with sip setup because can't receive calls!!!!!!
(abc def)
23. RE: Outbound Call & SIP Results (David Brazier)
----------------------------------------------------------------------
Message: 1
Date: Thu, 2 Feb 2006 12:13:09 +0000
From: Dave Brooks <mrdaveb at gmail.com>
Subject: [Asterisk-Users] DeadAGI variables confusion
To: Asterisk-Users at lists.digium.com
Message-ID: <2c4594220602020413g29da0b77h at mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1
Hi * users,
We're using calls to external scripts through AGI at various points
throughout our IVR system. We use these scripts to log certain events
and to make certain choices that I wasn't sure would be possible in
the dial plan. The problem comes with with the final call to our
script. We use this line:
exten => h,1,deadagi(log.php|{$service}|Hung up|${UNIQUEID})
I know there are some issues with getting variables through DeadAGI,
but I just wanted some clarification, because I haven't seen it
explained clearly. Certainly the value of UNIQUEID was being
successfully passed to log.php in earlier versions of * but isn't now
(I just installed version 1.2.4)
Any advice welcome. Even if it is telling me we've beein doing this
all wrong the whole time!
Regards,
DaveB
------------------------------
Message: 2
Date: Thu, 2 Feb 2006 13:12:39 +0100
From: <jan.sarin at securia.se>
Subject: SV: [Asterisk-Users] Outbound Caller ID number on E1
To: <asterisk-users at lists.digium.com>
Message-ID:
<0FF4F1903968F943B5EA2521CD5296C126983C at exchange.securia.local>
Content-Type: text/plain; charset="iso-8859-1"
How do you set the CallerID?
Have you checked with your provider that they've enabled callerid?
If yes, are you using a correct number that the provider allows?
Regards,
Jan
-----Ursprungligt meddelande-----
Från: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] För Garth van Sittert
Skickat: den 2 februari 2006 12:37
Till: Asterisk Users Mailing List - Non-Commercial Discussion
Ämne: [Asterisk-Users] Outbound Caller ID number on E1
Hi All
I am having a problem setting the outbound callerid number on a PRI E1 in South Africa. The outbound number keeps on appearing as the main PRI number. How does it work between Asterisk and the Telko? More importantly how do I get it working?
Kind Regards
Garth
--
Garth van Sittert
BSc (Physics & Computer Science)
-----------------
Mobile: +27 (0)83 791 6662
Email: garth at bitco.co.za
Phone: 08600 BITCO
Web: www.bitco.co.za
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------------------------------
Message: 3
Date: Thu, 2 Feb 2006 12:27:17 +0000
From: David Brazier <David.Brazier at alphabravocharlie.net>
Subject: [Asterisk-Users] Outbound Call & SIP Results
To: "asterisk-users at lists.digium.com"
<asterisk-users at lists.digium.com>
Message-ID: <40927EA2-143B-4573-AA26-92A94B38C914 at mimectl>
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------------------------------
Message: 4
Date: Thu, 02 Feb 2006 20:29:40 +0800
From: Steve Underwood <steveu at coppice.org>
Subject: Re: [Asterisk-Users] Outbound Caller ID number on E1
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <43E1FB34.9060202 at coppice.org>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Garth van Sittert wrote:
> Hi All
>
> I am having a problem setting the outbound callerid number on a PRI E1
> in South Africa. The outbound number keeps on appearing as the main
> PRI number. How does it work between Asterisk and the Telko? More
> importantly how do I get it working?
>
> Kind Regards
> Garth
Telcos usually arrange outgoing CLI in one of 3 ways:
- a free for all - you can put what you like as your CLI, and no checks
are made
- a rigid arrangement - no matter what you give as the CLI, the telco
will replace it with a fixed value before passing the message on
- a constrained arrangement - if you give a CLI within the range that is
valid for you, it will be passed on. If you give something which is not
allocated to you, the telco wil replace it with a fixed value.
Sounds like you do not have the first arrangement. You might have the
third, though. It could be you just aren't specifying your number
correctly - either the digits themselves or the TON/NPI pair might not
be right.
Steve
------------------------------
Message: 5
Date: Thu, 2 Feb 2006 23:30:50 +1100
From: Mark van Kerkwyk <mark at vk.net>
Subject: [Asterisk-Users] Delaying media stream by short period after
183 is sent
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID:
<OFA819FD2E.AC59A4AA-ONCA257109.0044411A-CA257109.0044BBEB at vk.net>
Content-Type: text/plain; charset="us-ascii"
Hi, anyone know of a way that I can delay the RTP stream a little bit once
the 183 is sent, I just want to delay it by around 100ms or so for some
troubleshooting.
Also, I always see a RTP packet before the 183 is sent for each call, it
is just a single packet, is something wrong here as my firewall won't open
a connection entry until 183 is processed.
Mark
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Message: 6
Date: Thu, 2 Feb 2006 07:41:21 -0500
From: "Alexander Lopez" <alex.lopez at opsys.com>
Subject: RE: RE: [Asterisk-Users] Blocked Callerid
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Message-ID:
<E918F2FD95450648B7F8C957D92D5271422613 at exmail.corp.opsys.com>
Content-Type: text/plain; charset="us-ascii"
YES!!!!
Asterisk will support this.
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of
> pdhales at optusnet.com.au
> Sent: Thursday, February 02, 2006 1:52 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: RE: [Asterisk-Users] Blocked Callerid
>
>
> I think they have a 1-800 number so you might be right.
>
> But the important question is still - will Asterisk support this?
>
> PaulH
>
> > Alexander Lopez <alex.lopez at opsys.com> wrote:
> >
> They are using ANI instead of CallerID. If they have an 800
> number thya have the right to know who is calling them
> because they are paying for the call.
>
> the *ANI*DNIS* format is known as Feature Grooup D.
>
> Alex
>
>
>
> ________________________________
>
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of
> Joe Pukepail
> Sent: Wednesday, February 01, 2006 3:47 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Blocked Callerid
>
>
> Do they have an 800 number? If so perhaps their 800
> number provider is doing it via DTMF. Search around on the
> internet, I believe the standard format for the DTMF is
> *CALLERID*CALLEDNUMBER* (or perhaps reversed).
>
>
> On 2/1/06, pdhales at optusnet.com.au <pdhales at optusnet.com.au>
> wrote:
>
> I have been discussing an asterisk solution
> with a company that has a custom written dialogic based solution.
>
> The issue is that their dialogic solution can
> read callerid from incoming calls, even if the callerid is blocked.
> I have read before that Asterisk can do this,
> and they want me to make sure that their new system will be
> able to do this.
>
> A quick poke around inside the zaptel source
> code was unproductive...
>
> Any ideas?
>
> PaulH
>
>
> _______________________________________________
> --Bandwidth and Colocation provided by
> Easynews.com <http://easynews.com/> --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>
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>
------------------------------
Message: 7
Date: Thu, 2 Feb 2006 07:39:15 -0500
From: hugolivude <hugolivude at gmail.com>
Subject: [Asterisk-Users] Anyone know a good ITSP in Canada that
supports *?
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID:
<c471e4e30602020439q5a4c7432jec2f21c77412f449 at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
Hi,
I'm looking for a new Internet Telephony Service Provider for my company in
Canada to terminate calls from my Asterisk PBX. Ideally I'd like DiDs in
Otawa, Toronto, NY & San Jose. Anyone out ther who can help me with a
recommendation?
Vonage seemed clueless when I called them. Broadvoice is good but no
Canadian DIDs...
Thanks,
Hugh
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Message: 8
Date: Thu, 02 Feb 2006 10:41:13 -0200
From: "Rodrigo P. Telles" <telles-listas at devel.it>
Subject: [Fwd: Re: [Asterisk-Users] Asterisk for Call Center (missing
reference)]
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <43E1FDE9.4060100 at devel.it>
Content-Type: text/plain; charset="iso-8859-1"
John Todd,
Can you please answer that question or just give me your feedback about it?
I'll be very thankfull to hear something from you!
regards,
Telles
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From: "Rodrigo P. Telles" <telles-listas at devel.it>
Subject: Re: [Asterisk-Users] Asterisk for Call Center (missing reference)
Date: Mon, 23 Jan 2006 16:39:30 -0200
Size: 1888
Url: http://lists.digium.com/pipermail/asterisk-users/attachments/20060202/2d984165/Asterisk-UsersAsteriskforCallCentermissingreference-0001.eml
------------------------------
Message: 9
Date: Thu, 02 Feb 2006 13:49:48 +0100
From: Olle E Johansson <oej at edvina.net>
Subject: Re: [Asterisk-Users] Outbound Call & SIP Results
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <43E1FFEC.7090304 at edvina.net>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
David Brazier wrote:
> We make outbound calls via our Asterisk (*1), then via SIP to a 3rd
> party Asterisk (*2), which then routes to PSTN, again via SIP. If the
> called number is invalid or out of service, *2 gets a 404 Not Found,
> which seems appropriate. However, *2 then passes on a 403 Forbidden to
> *1, which is not really the right response. *1 then returns a 486 Busy
> Here, which also seems wrong, as it would generally mean the called
> number was busy (engaged), I think.
>
> Is it possible to vary this behaviour? The ${DIALSTATUS} variable
> doesn't seem to be fine-grained enough to help.
>
Would it be possible to see som log files?
/O
------------------------------
Message: 10
Date: Thu, 2 Feb 2006 07:46:38 -0500
From: Andrew Kohlsmith <akohlsmith-asterisk at benshaw.com>
Subject: Re: [Asterisk-Users] Anyone know a good ITSP in Canada that
supports *?
To: asterisk-users at lists.digium.com
Message-ID: <200602020746.38860.akohlsmith-asterisk at benshaw.com>
Content-Type: text/plain; charset="iso-8859-1"
On Thursday 02 February 2006 07:39, hugolivude wrote:
> I'm looking for a new Internet Telephony Service Provider for my company in
> Canada to terminate calls from my Asterisk PBX. Ideally I'd like DiDs in
> Otawa, Toronto, NY & San Jose. Anyone out ther who can help me with a
> recommendation?
Unlimitel.ca. CAD$0.011/min for origination and on-net termination.
Excellent, and I mean *excellent* customer service.
Not affiliated, but a very happy customer.
-A.
------------------------------
Message: 11
Date: Thu, 2 Feb 2006 12:54:54 +0000 (UTC)
From: tony at softins.clara.co.uk (Tony Mountifield)
Subject: [Asterisk-Users] Re: DeadAGI variables confusion
To: asterisk-users at lists.digium.com
Message-ID: <drsveu$bi1$1 at softins.clara.co.uk>
In article <2c4594220602020413g29da0b77h at mail.gmail.com>,
Dave Brooks <mrdaveb at gmail.com> wrote:
> Hi * users,
>
> We're using calls to external scripts through AGI at various points
> throughout our IVR system. We use these scripts to log certain events
> and to make certain choices that I wasn't sure would be possible in
> the dial plan. The problem comes with with the final call to our
> script. We use this line:
>
> exten => h,1,deadagi(log.php|{$service}|Hung up|${UNIQUEID})
>
> I know there are some issues with getting variables through DeadAGI,
> but I just wanted some clarification, because I haven't seen it
> explained clearly. Certainly the value of UNIQUEID was being
> successfully passed to log.php in earlier versions of * but isn't now
> (I just installed version 1.2.4)
>
> Any advice welcome. Even if it is telling me we've beein doing this
> all wrong the whole time!
You should be able to refer to channel variables in the 'h' extension.
If it's broken, then that's a bug.
However, if your example is an exact copy from your dialplan, perhaps
the parser is getting confused, because {$service} should be ${service}
Cheers
Tony
--
Tony Mountifield
Work: tony at softins.co.uk - http://www.softins.co.uk
Play: tony at mountifield.org - http://tony.mountifield.org
------------------------------
Message: 12
Date: Thu, 2 Feb 2006 05:06:14 -0800 (PST)
From: Dovid Bender <asteriskdigium at yahoo.com>
Subject: Re: [Asterisk-Users] Anyone know a good ITSP in Canada that
supports *?
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <20060202130614.83736.qmail at web36511.mail.mud.yahoo.com>
Content-Type: text/plain; charset=iso-8859-1
iBell just announced termination only to CA for I
believe $0.0039 a minute.
--- Andrew Kohlsmith <akohlsmith-asterisk at benshaw.com>
wrote:
> On Thursday 02 February 2006 07:39, hugolivude
> wrote:
> > I'm looking for a new Internet Telephony Service
> Provider for my company in
> > Canada to terminate calls from my Asterisk PBX.
> Ideally I'd like DiDs in
> > Otawa, Toronto, NY & San Jose. Anyone out ther
> who can help me with a
> > recommendation?
>
> Unlimitel.ca. CAD$0.011/min for origination and
> on-net termination.
> Excellent, and I mean *excellent* customer service.
>
> Not affiliated, but a very happy customer.
>
> -A.
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com
> --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>
>
http://lists.digium.com/mailman/listinfo/asterisk-users
>
__________________________________________________
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------------------------------
Message: 13
Date: Thu, 2 Feb 2006 08:17:17 -0500
From: Matt <mhoppes at gmail.com>
Subject: [Asterisk-Users] Call completes but then drops?
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID:
<c11d02530602020517u774988d8k775dac654c4c1c5d at mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1
Feb 1 22:13:37 VERBOSE[18623] logger.c: -- Zap/2-1 answered SIP/102-9fda
Feb 1 22:13:37 DEBUG[18623] channel.c: Avoiding initial deadlock for
'SIP/102-9fda'
Feb 1 22:13:43 DEBUG[18623] channel.c: Didn't get a frame from
channel: SIP/102-9fda
Feb 1 22:13:43 DEBUG[18623] channel.c: Bridge stops bridging channels
SIP/102-9fda and Zap/2-1
Feb 1 22:13:43 VERBOSE[18623] logger.c: -- Hungup 'Zap/2-1'
Feb 1 22:13:43 DEBUG[18623] app_dial.c: Exiting with DIALSTATUS=ANSWER.
Can anyone explain why this call dropped?
The person dialed a number, the call WAS completed and connected to
the PSTN through a PRI, but they never heard audio and the call was
disconnected by Asterisk.
------------------------------
Message: 14
Date: Thu, 2 Feb 2006 05:20:01 -0800 (PST)
From: Dovid Bender <asteriskdigium at yahoo.com>
Subject: [Asterisk-Users] Asterisk on laptop connected to POTS line
To: asterisk-users at lists.digium.com
Message-ID: <20060202132001.97093.qmail at web36503.mail.mud.yahoo.com>
Content-Type: text/plain; charset=iso-8859-1
Anyone know of any equipment that I can use to connect
a laptop running asterisk to a POTS line (RJ11) ?
Regards,
Dovid
__________________________________________________
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Tired of spam? Yahoo! Mail has the best spam protection around
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------------------------------
Message: 15
Date: Thu, 2 Feb 2006 13:20:11 -0000
From: "Victor Alvarez" <victor at sentidocomun.es>
Subject: [Asterisk-Users] Regarding cdr_manager.conf
To: <asterisk-users at lists.digium.com>
Message-ID: <013401c627fc$57348760$ed81a8c0 at xana>
Content-Type: text/plain; charset="iso-8859-1"
Hello,
My question is.. How does cdr_manager work? Does it suppose to populate
cdr-csv/Master.csv? What about the cdr table on the database? What is the
event some people talk about?
I have changed (and reloaded) my configuration of cdr_manager.conf to
;
; Asterisk Call Management CDR
;
[general]
enabled = yes
and it doesn't seem to make any difference. After originate a call from the
manager interface my Master.csv is empty, cdr in my database also empty and
I don't get any new event apart of Newchannel or Hangup from the manager
interface.
An this is all I get from the Asterisk console:
-- Reloading module 'cdr_manager.so' (Asterisk Call Manager CDR Backend)
== Parsing '/etc/asterisk/cdr_manager.conf': Found
So..?
Kind Regards,
Victor.
------------------------------
Message: 16
Date: Thu, 2 Feb 2006 15:27:01 +0200
From: Tzafrir Cohen <tzafrir at cohens.org.il>
Subject: Re: [Asterisk-Users] Asterisk on laptop connected to POTS
line
To: asterisk-users at lists.digium.com
Message-ID: <20060202132701.GV29741 at gadot.org.il>
Content-Type: text/plain; charset=us-ascii
On Thu, Feb 02, 2006 at 05:20:01AM -0800, Dovid Bender wrote:
> Anyone know of any equipment that I can use to connect
> a laptop running asterisk to a POTS line (RJ11) ?
A SIP ATA with an FXO port? (e.g. the Sipura 3000)
--
Tzafrir Cohen | tzafrir at jbr.cohens.org.il | VIM is
http://tzafrir.org.il | | a Mutt's
tzafrir at cohens.org.il | | best
ICQ# 16849755 | | friend
------------------------------
Message: 17
Date: Thu, 2 Feb 2006 06:27:57 -0700
From: "Damon Estep" <damon at suburbanbroadband.net>
Subject: RE: [Asterisk-Users] Asterisk on laptop connected to POTS
line
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Message-ID:
<07668904BA88BA4E9DA11CDE5B594CB20144E15D at ns1.soho.soho-systems.com>
Content-Type: text/plain; charset="us-ascii"
Sipura SPA-3000 will give you 1 fxs and 1 fxo so you can connect to a
POTS line AND a analog phone at the same time with one small box.
Makes a great demo system.
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-
> bounces at lists.digium.com] On Behalf Of Dovid Bender
> Sent: Thursday, February 02, 2006 6:20 AM
> To: asterisk-users at lists.digium.com
> Subject: [Asterisk-Users] Asterisk on laptop connected to POTS line
>
> Anyone know of any equipment that I can use to connect
> a laptop running asterisk to a POTS line (RJ11) ?
>
> Regards,
> Dovid
>
------------------------------
Message: 18
Date: Thu, 2 Feb 2006 08:42:18 -0500
From: "Jonathan k. Creasy" <jonathan at bluegrass.net>
Subject: RE: [Asterisk-Users] Asterisk on laptop connected to POTS
line
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Message-ID: <DD1EBD23103ADC4290D86546BE26D2B0B197AD at outbreak.bgnd.com>
Content-Type: text/plain; charset="us-ascii"
The Grandstream ATA (480 I think...) does this and usually costs less
than the Sipura. It has 1 FXS and 1 FXO.
-Jonathan
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-
> bounces at lists.digium.com] On Behalf Of Damon Estep
> Sent: Thursday, February 02, 2006 8:28 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: RE: [Asterisk-Users] Asterisk on laptop connected to POTS
line
>
> Sipura SPA-3000 will give you 1 fxs and 1 fxo so you can connect to a
> POTS line AND a analog phone at the same time with one small box.
>
> Makes a great demo system.
>
> > -----Original Message-----
> > From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-
> > bounces at lists.digium.com] On Behalf Of Dovid Bender
> > Sent: Thursday, February 02, 2006 6:20 AM
> > To: asterisk-users at lists.digium.com
> > Subject: [Asterisk-Users] Asterisk on laptop connected to POTS line
> >
> > Anyone know of any equipment that I can use to connect
> > a laptop running asterisk to a POTS line (RJ11) ?
> >
> > Regards,
> > Dovid
> >
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
------------------------------
Message: 19
Date: Thu, 2 Feb 2006 09:13:29 -0500
From: "Alexander Lopez" <alex.lopez at opsys.com>
Subject: RE: [Asterisk-Users] Asterisk on laptop connected to POTS
line
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Message-ID:
<E918F2FD95450648B7F8C957D92D5271422617 at exmail.corp.opsys.com>
Content-Type: text/plain; charset="us-ascii"
Look at Xorcom's USB channel Bank.
> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of
> Dovid Bender
> Sent: Thursday, February 02, 2006 8:20 AM
> To: asterisk-users at lists.digium.com
> Subject: [Asterisk-Users] Asterisk on laptop connected to POTS line
>
> Anyone know of any equipment that I can use to connect a
> laptop running asterisk to a POTS line (RJ11) ?
>
> Regards,
> Dovid
>
>
> __________________________________________________
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------------------------------
Message: 20
Date: Thu, 02 Feb 2006 08:09:33 -0600
From: "Kevin P. Fleming" <kpfleming at digium.com>
Subject: Re: [Asterisk-Users] TE411P or TE406P
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <43E2129D.8030503 at digium.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Matt wrote:
> You will need a minimum 3.4Ghz Dual xeon with 1G ECC DDR, and hardware voice processing capable E1/T1 card, such as the sangoma 104d quad pci card, in order to run 120 PSTN calls, 1000 calls is impossible for 1 server. Centos Linux should be fine.
This has to be some of the poorest advice I have seen on this list...
what is 'hardware voice processing capable'?
We run 120 channels of TDM on single CPU servers all the time (no
transcoding of course), and the amount of RAM is nearly irrelevant.
Independent tests have shown there is no appreciable performance
difference between the available quad-port T1 cards.
The poster did not ask about handling '1000 calls' nor about Linux
distributions.
> We sell supermicro based * solutions you can contact me off list.
This entire response was clearly an advertisement for your
products/services, and as such is inappropriate for this list.
To the OP: The TE406P and TE411P are identical except for PCI bus
interface voltage. Use whichever your server can accept, and if it can
do both (which is rare), use the TE411P as in the future 5V slots will
be harder to find.
------------------------------
Message: 21
Date: Thu, 02 Feb 2006 08:11:42 -0600
From: "Kevin P. Fleming" <kpfleming at digium.com>
Subject: Re: [Asterisk-Users] meetme and dtmf
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <43E2131E.3020206 at digium.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Accursio Avona wrote:
> Step 2: The IAX client make a second call executing again
> Dial(ZAP/g1/${EXTEN})
> an IVR answer this call and the IAX client have to send some
> DTMF stil now everything works very well.
> At this point call is transfered to the previous conference room
> and The IAX client reach the conference too.
>
> Step 3 The Iax client heve to send some other DTMF to the IVR.
How is the IVR still involved if the call has been transferred into a
conference room?
------------------------------
Message: 22
Date: Thu, 2 Feb 2006 06:11:53 -0800 (PST)
From: abc def <xisterisk at yahoo.com>
Subject: Re: [Asterisk-Users] Help with sip setup because can't
receive calls!!!!!!
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <20060202141153.76640.qmail at web35107.mail.mud.yahoo.com>
Content-Type: text/plain; charset="iso-8859-1"
I am responding to my own problem because I found the answer finally which it may help others in future.
I just had a break-through after 2 weeks struggling, finally I found the problem.
the problem was in extensions.conf file. I misspelled "include" as "inculde" (finding a misspelled word in a long extensions.conf file is not so easy, trust me) but after checking "show dialplan" and going through it line by line, I found my sip sub-division is not there.
abc def <xisterisk at yahoo.com> wrote: not sure but this is the output from the pbx:
>sip show registry
Host Username Refresh State
local_sip:5060 stargate3 105 Registered
local_sip:5060 stargate2 105 Registered
local_sip:5060 stargate1 105 Registered
from sip phone I can any other phone (cisco with sccp or iax protocol) but I can't call any other sip phone, or receive phone calls.
Facundo Ameal <fameal at gmail.com> wrote:
are you sure your sip phone is registering ok?
2006/2/1, abc def :
> Thanks Facundo for instruction but it didn't work. there is nothing new in
> your suggestion compare to my conf files nevertheless I tried it but it
> didn't work. I can make call from my sip phone but can't receive any phone
> call. I am sure some one had had the same problem an d solved it.
> as always I appreciate for your suggestion, advice and/or correction to my
> config files.
> if you know how to solve this problem please give me some hint.
>
> thank you
>
> Facundo Ameal wrote:
> i've tested it with this config files and i worked:
>
> extensions.conf
>
> exten => 55,1,Dial(SIP/2271,20)
>
>
> sip.conf
>
> [2271]
> type=friend
> host=dynamic
> secret=sip
> allow=all
> qualify=200
> nat=no
>
>
> Instead of 2271 you can put whatever you want.
>
> good luck.
>
>
>
> 2006/1/31, Facundo Ameal :
> > Are you using a SIP Softphone or an ATA?
> >
> > 2006/1/31, Facundo Ameal :
> > > does it registers well?
> > > although i think you have to add "context=default" to the stargate1
> section.
> ; > >
> > > try that and see what happens.
> > >
> > > 2006/1/31, abc def :
> > > > Hi all, I am resending this message, so far no one has helped me with
> this
> > > > incoming call issue. there is no problem with outbound call but there
> is no
> > > > inbound call to my sip phone. the only message I get when I call from
> pstn
> > > > is "unable to create local channel for call forward to
> > > > 'Local/sipphone at default' (case =0)". my configuration files are
> attached
> > > > below. any help would be greatly appreciated. many thanks in advance.
> > > > ABC
> > > >
> > > > abc def wrote:
> > > >
> > > > there is no error message coming up on the pbx for in-bound calls
> (there is
> > > > only debugging messages for outbound calls).
> > > >
> > > > thanks in advance for any hint or suggestion.
> > > > Ama
> > > >
> > > > I just post my configuration file here for sip phone:
> > > > extensions.conf
> > > >
> -------------------------------------------------------------------------
> > > > [globals]
> > > > [default]
> > > > include => incoming
> > > > include => outgoing
> > > > include => iax
> > > > inculde => sip
> > > > include => sccp
> > > > [sip]
> > > > exten => 2171,1,Dial(SIP/stargate1,20)
> > > > ;exten => 2171,1,Dial(SIP/2171,20)
> > > > exten => 2171,2,Hangup
> & gt; > > exten => 2172,1,Dial(SIP/stargate2,20)
>
> > > > ;exten => 2172,1,Dial(SIP/2172,20)
> > > > exten => 2172,2,Hangup
> > > > exten => 2173,1,Dial(SIP/stargate3,20)
> > > > ;exten => 2173,1,Dial(SIP/2173,20)
> > > > exten => 2173,2,Hangup
> > > > [sccp]
> > > > [skinny]
> > > > [incoming]
> > > > exten => ; _214943[5-9]6,1,Dial(SIP/stargate3)
> > > > exten => _214943[5-9]6,2,Hangup
> > > > [outgoing]
> > > > exten => _XXXXXXXX,1,Dial(Zap/g1/${EXTEN})
> > > > exten => _XXXXXXXX,2,Hangup
> > > >
> -------------------------------------------------------------------------
> > > > sip.conf
> > > >
> -------------------------------------------------------------------------
> > > > [general]
> > > > context=default ; Default context for incoming calls
&g t; > > > ; Set this to your host name or domain name
> > > > bindport=5060 ; UDP Port to bind to (SIP standard port is
> > > > 5060)
> > > > bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to
> > > > all)
> > > > srvlookup=yes ; Enable DNS SRV lookups on outbound calls
> > > >
> > > > register => stargate1:1stargate at local_sip/2171
> > > > register => stargate2:2stargate at local_sip/2172
> > > > register => stargate3:3stargate at local_sip/2173
> > > > ;---------------------------------------------- NAT
> SUPPORT
> > > > ------------------------
> > > > nat=no ; Global NAT settings (Affects all peers and
> > > > users)
> > > >
> > > >
> > > > [local_sip]
> > > > type=friend
> > > > host=10.47.200.136
> > > > context=default
> > > > [stargate1] ;cisco 9760
> > > > ;[2171]
> > > > ; type=friend
>
> > > > host=dynamic ;10.47.200.140 ;dynamic
> > > > defaultip=10.47.200.140
> > > > username=stargate1
> > > > secret=xxx
> > > > callerid="21495071" <2171>
> > > > allow=all
> > > > qualify=200
> > > > nat=no
> > > > defaultip=10.47.200.140
> > > >
> > > > [stargate2] ;Polycom 601
> > > > ;[2172]
> > > > type=friend
> > > > host=dynamic ;10.47.200.141 ;dynamic
> > > > defaultip=10.47.200.141
> > > > username=xxx
> > > > secret=2stargate
> > > > callerid="21495072" <2172>
> > > > allow=all
> > &g t; > qualify=200
> > > > nat=no
> > > > defaultip=10.47.200.141
> > > > [stargate3] ;Aastra 480i
> > > > ;[2173]
> > > > type=friend
> > > > host=dynamic ;10.47.200.137 ;dynamic
> > > > defaultip=10.47.200.137
> > > > username=stargate3
> > > > callerid="starg ate3" <2173>
> > > > secret=xxx
> > > > allow=all
> > > > qualify=200
> > > > nat=no
> > > > defaultip=10.47.200.137
> > > >
> ----------------------------------------------------------------------------
> > > >
> > > >
> > > > pdhales at optusnet.com.au wrote:
> > > >
> > > > What error do you get when trying to call the SIP phones?
> > > >
> > > > PaulH
> > > >
> > > >
> > > > ----- Original Message -----
> > > > From: abc def
> > > > To: asterisk-users at lists.digium.com
> > > > Sent: Wednesday, January 25, 2006 11:58 PM
> > > > Subject: [Asterisk-Users] Help with sip setup because can't receive
> calls
> > > >
> > > >
> > > >
> > > > Hi all,
> > > > I read many posts on asterisk mail site and been trying many
> different
> > > > things but still I can't get my sip phones to work with asterisk.
> > > > I have a full blown-up voip netwok with two asterisk servers connected
> > > > to pstn network with iax phones and cisco sccp phones which all work
> fine.
> > > > however, I have been struggeling to configure my sip phones (polycom
> 601,
> > > > Aastra 480i and cisco 9760) to work with aster isk. I can call out from
> sip
> > > > phones to anywhere else but not receive phone calls. I can see the
> phones on
> > > > "sip show registry" and "sip show peers" but no track phone calls for
> sip.
> > > >
> > > > can you please shed some light on me how to go about solving this
> > > > problem?
> > > >
> > > > thank you and best regards,
> > > > Ama
> > > >
> > > > < HR SIZE=1> Do you Yahoo!?
> > > > With a free 1 GB, there's more in store with Yahoo! Mail.
> > > > ________________________________
> > > >
> > > > _______________________________________________
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> > > >
> > > > Asterisk-Users mailing list
> > > > To UNSUBSCRIBE or update options visit:
> > > >
> > > >
> http://lists.digium.com/mailman/listinfo/asterisk-users
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> > > >
> > > > Asterisk-Users mailing list
> > > > To UNSUBSCRIBE or update options visit:
> > > >
> http://lists.digium.com/mailman/listinfo/asterisk-users
> > > >
> > > >
> > > >
> > > > ________________________________
> > > > Bring words and photos to gether (easily) with
> > > > PhotoMail - it's free and works with Yahoo!
> > > > Mail._______________________________________________
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> > > > Asterisk-Users mailing list
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> > > >
> http://lists.digium.com/mailman/listinfo/asterisk-users
> > > >
> > > >
> > > >
> > > >
> > > > ________________________________
> > > > Yahoo! Autos. Looking for a sweet ride? Get pricing, reviews, & more
> on new
> > > > and used cars.
> > > >
> > > >
> > > > _______________________________________________
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> > > >
> > > > Asterisk-Users mailing list
> > > > To UNSUBSCRIBE or update options visit:
> > > >
> & gt; > >
> http://lists.digium.com/mailman/listinfo/asterisk-users
> > > >
> > > >
> > > >
> > >
> > >
> > > --
> > > Facundo Ameal.
> > > famealgmailcom
> > > Linux User #395088
> > >
> > > FWD: 741664
> > > MSN: asadolamorcillacomar
> > > ICQ: 74005793
> > >
> > >
> > > Open your mind, use open source.
> > >
> >
> >
> > --
> > Facundo Ameal.
> > famealgmailcom
> > Linux User #395088
> >
> > FWD: 741664
> > MSN: asadolamorcillacomar
> > ICQ: 74005793
> >
> >
> > Open your mind, use open source.
> >
>
>
> --
> Facundo Ameal.
> famealgmailcom
> Linux User #395088
>
> FWD: 741664
> MSN: asadolamorcillacomar
> ICQ: 74005793
>
>
> Open your mind, use open source.
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
>
> ________________________________
>
> What are the most popular cars? Find out at Yahoo! Autos
>
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
--
Facundo Ameal.
famealgmailcom
Linux User #395088
FWD: 741664
MSN: asadolamorcillacomar
ICQ: 74005793
Open your mind, use open source.
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Message: 23
Date: Thu, 2 Feb 2006 14:14:23 +0000
From: David Brazier <David.Brazier at alphabravocharlie.net>
Subject: [Asterisk-Users] RE: Outbound Call & SIP Results
To: <asterisk-users at lists.digium.com>
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