Hello again,
I am using mix monitor and the majority of the sound records perfectly.
I then get a "Internal Data Stream Error" near the end of the sound
file. Has anyone ever seen this? I am allowing the ULAW amd ALAW codecs
and an example dialplan entry is ;
; phone line phone1
exten => phone1,1,Answer()
exten => phone1,2,MixMonitor(test.wav|av(0)V(0))
exten => phone1,3,Dial(SIP/phone1)
exten => phone1,4,Busy(10)
exten => phone1,5,Hangup()
Many many thanks
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of
asterisk-users-request at lists.digium.com
Sent: 23 October 2007 01:31
To: asterisk-users at lists.digium.com
Subject: asterisk-users Digest, Vol 39, Issue 76
*** WARNING ***
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Today's Topics:
1. Re: A linksys SPA921 behind NAT and firewall (joakimsen at gmail.com)
2. Re: Making Asterisk a "Voice Router" (end1r)
3. Split asterisk in two ?? One TDM and One IP only?? (Steven)
4. Authenticate by IP? (Carlos Chavez)
5. Polycom 601 + Headset (Dovid B)
6. Re: tech prefix (Philipp Kempgen)
7. Re: Authenticate by IP? (joakimsen at gmail.com)
8. [France CID] Does Zaptel support ETSI FSK? (Vincent)
9. Re: Authenticate by IP? (Rurouni Alucard)
10. Re: Prompting for number when CID number not sent? (Vincent)
11. dial-out call queue (Joao Pereira)
12. Re: Authenticate by IP? (Carlos Chavez)
13. Split asterisk in two ?? One TDM and One IP only??
(BerkHolz, Steven)
14. Re: [France CID] Does Zaptel support ETSI FSK? (Vincent)
15. Re: Extensions.conf for basic IVR? (Vincent)
16. Re: 16 ports wanted (Christian Victor)
17. Re: Extensions.conf for basic IVR? (Erik Anderson)
18. Re: [France CID] Does Zaptel support ETSI FSK? (Jared Smith)
19. Re: Authenticate by IP? (Carlos Chavez)
20. Re: Extensions.conf for basic IVR? (Vincent)
21. Re: Authenticate by IP? (joakimsen at gmail.com)
22. Re: Authenticate by IP? (Victor Toofic)
23. bristuff: music on hold but no dialoptions tT defined.
(Thomas Winter)
24. Re: [France CID] Does Zaptel support ETSI FSK? (Vincent)
25. Re: G729a codecs + Asterisk 1.4.11 (bilal ghayyad)
26. Re: [France CID] Does Zaptel support ETSI FSK? (Ira)
27. Voicemail playback on iPhone (Jason Lixfeld)
28. NAT traversal packet loss measurement (Yitzhak Bar Geva)
29. Re: Voicemail playback on iPhone (Ron Stephan)
30. Re: NAT traversal packet loss measurement (Matt Riddell)
----------------------------------------------------------------------
Message: 1
Date: Mon, 22 Oct 2007 13:25:20 -0400
From: "joakimsen at gmail.com" <joakimsen at gmail.com>
Subject: Re: [asterisk-users] A linksys SPA921 behind NAT and firewall
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Message-ID:
<23fd749a0710221025i6a54bd16hdeb985e01cfcebf8 at mail.gmail.com>
Content-Type: text/plain; charset=UTF-8
Check out again http://spc.pifiu.com it seems the owner of the site has
added the latest admin guide for SPA-900 series & the spc.exe for
5.1.5 & 5.1.7 firmware.
On 10/21/07, Per Jessen <per at computer.org>
wrote:> joakimsen at gmail.com wrote:
>
> > If you are trying to use non-complied ("XML") profiles...
don't even
> > bother wasting your time.
>
> Oh. I _am_ using the XML format. When I initiate a resync over the
> http server, it works fine, except the SPA doesn't start the regular
> resync.
>
>
>
> /Per Jessen, Z?rich
------------------------------
Message: 2
Date: Mon, 22 Oct 2007 14:05:01 -0400
From: "end1r" <end1r at comcast.net>
Subject: Re: [asterisk-users] Making Asterisk a "Voice Router"
To: "'Asterisk Users Mailing List - Non-Commercial
Discussion'"
<asterisk-users at lists.digium.com>
Message-ID: <001901c814d6$1165cbe0$343163a0$@net>
Content-Type: text/plain; charset="utf-8"
Is this free? I see the tuner is free.. but the speech rec isn?t?
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of lenz
Sent: Monday, October 22, 2007 11:28 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Making Asterisk a "Voice Router"
Nice job! I took the liberty to post it on AstPligg as well:
http://tinyurl.com/268bac
Thanks
l.
In data Mon, 22 Oct 2007 16:22:13 +0200, Jared Smith <jsmith at
digium.com>
ha scritto:
> On Mon, 2007-10-22 at 09:39 -0400, end1r wrote:
>> I?m interested in what software (Free or course) that people use when
>> they want to add a ?dial by voice? service to their asterisk system.
>> Meaning I pick up the phone.. dial some extension? it prompts me for
>> name.. I say ?John Smith?.. and it dials his extension and connects
>> the call..
>
> I've done this using Asterisk and the LumenVox speech engine... in
> fact, I spoke about it at AstriCon Europe in 2006. My slides are
> available at http://www.astricon.net/files/Jared_Smith_EUR06.pdf.
> (They may be slightly out of date, but it should at least get you
> started.)
>
>
--
Home of QueueMetrics - http://queuemetrics.com
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
------------------------------
Message: 3
Date: Mon, 22 Oct 2007 14:09:49 -0400
From: "Steven" <asterisk at tescogroup.com>
Subject: [asterisk-users] Split asterisk in two ?? One TDM and One IP
only??
To: asterisk-users at lists.digium.com
Message-ID: <ffip0q$h1h$1 at ger.gmane.org>
I have built an asterisk server with a TE412P card on a Dell 2950.
It does incoming calls via DID over PRI, our IVR, SIP/IAX extensions,
Fax/Analog extensions via an old PBX via PRI, voicemail, etc.
My issue now is that I find it difficult to test/upgrade to new
versions.
This is what I am thinking of doing.
Server1
Keep one physical server just for TDM functions.
PRI to Telco
PRI to old PBX for Fax. (basically using it as a mux)
Keep meetme here for Digium card timing.
Server2
Build a new asterisk install within Xen VM with data stored on an iSCSI
SAN. This would be all IP.
IAX and SIP extensions.
IAX and SIP providers.
IVR
Voicemail
Web access to voicemail
CDR
This way I can test different versions of the features of Server2 (clone
with different IP) without affecting production.
I assume that I just use an IAX or SIP trunk between the two asterisk
servers.
Does this make sense?
Are others doing similar?
Are there any other features that require the TDM card besides PRI, Fax
and Meetme?
I have heard of people using Xen for IP only asterisk, but are there any
known gotchas?
Thanks,
--
--
Steven
http://www.glimasoutheast.org
------------------------------
Message: 4
Date: Mon, 22 Oct 2007 13:24:31 -0500
From: Carlos Chavez <cursor at telecomabmex.com>
Subject: [asterisk-users] Authenticate by IP?
To: Asterisk <asterisk-users at lists.digium.com>
Message-ID: <1193077471.3215.7.camel at cursor.telecomabmex.com>
Content-Type: text/plain; charset="utf-8"
I have a customer that needs an Asterisk server to sell minutes
for
cell phones in Mexico. I do not see a problem with that since he will
get the calls by SIP and then use GSM adapters to get the calls into the
GSM network. My problem is that his customers only want to be
identified by IP and not by a username and password. Is there a way to
authenticate just by using an IP address?
--
Telecomunicaciones Abiertas de M?xico S.A. de C.V.
Carlos Ch?vez Prats
Director de Tecnolog?a
+52-55-91169161 ext 2001
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Message: 5
Date: Mon, 22 Oct 2007 20:31:19 +0200
From: "Dovid B" <asteriskusers at dovid.net>
Subject: [asterisk-users] Polycom 601 + Headset
To: <asterisk-users at lists.digium.com>
Message-ID: <001101c814d9$c9d2e700$0800a8c0 at DovidLaptop>
Content-Type: text/plain; charset="iso-8859-1"
Hi List,
I am using a Plantronics CS50 head set with my Polycom 601. I use the
button on it to pick up calls. Is there any way to have the phone set up
that if I pick up with the button on the headset that it sends the call
to the headset and that I don't have to press the headset button on the
phone every time ?
Thanks.
Dovid
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------------------------------
Message: 6
Date: Mon, 22 Oct 2007 21:00:17 +0200
From: Philipp Kempgen <philipp.kempgen at amooma.de>
Subject: Re: [asterisk-users] tech prefix
To: Asterisk Users <asterisk-users at lists.digium.com>
Message-ID: <471CF341.3010600 at amooma.de>
Content-Type: text/plain; charset=ISO-8859-15
Jon Weisman wrote:
> Here's what worked:
>
> exten=>_X.,1,Dial(SIP/"prefix"${EXTEN}@outbound trunk)
>
> substitute "prefix" for the tech prefix you would like to append.
> ----- Original Message -----
> From: "Philipp Kempgen" <philipp.kempgen at amooma.de>
> To: "Asterisk Users" <asterisk-users at lists.digium.com>
> Sent: Tuesday, October 16, 2007 3:09 PM
> Subject: Re: [asterisk-users] tech prefix
>
>
> Jon Weisman wrote:
>
> How can I add a prefix to an outbound call?
>
> _X. => {
> Dial(tech/123{EXTEN});
> }
That's what I said.
Regards,
Philipp Kempgen
--
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Let's use IT to solve problems and not to create new ones.
Asterisk? -> http://www.das-asterisk-buch.de
Gesch?ftsf?hrer: Stefan Wintermeyer
Handelsregister: Neuwied B 14998
------------------------------
Message: 7
Date: Mon, 22 Oct 2007 15:13:04 -0400
From: "joakimsen at gmail.com" <joakimsen at gmail.com>
Subject: Re: [asterisk-users] Authenticate by IP?
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Message-ID:
<23fd749a0710221213p1410495ehbed9fe8797255075 at mail.gmail.com>
Content-Type: text/plain; charset=UTF-8
On 10/22/07, Carlos Chavez <cursor at telecomabmex.com>
wrote:> I have a customer that needs an Asterisk server to sell
minutes for> cell phones in Mexico. I do not see a problem with that since he will
> get the calls by SIP and then use GSM adapters to get the calls into
the> GSM network. My problem is that his customers only want to be
> identified by IP and not by a username and password. Is there a way
to> authenticate just by using an IP address?
>
There certainly is.
------------------------------
Message: 8
Date: Mon, 22 Oct 2007 21:19:27 +0200
From: Vincent <vincent.delporte at bigfoot.com>
Subject: [asterisk-users] [France CID] Does Zaptel support ETSI FSK?
To: asterisk-users at lists.digium.com
Message-ID: <dntph3dv1v9pda7ho4mnneb05b6tu9u9bk at 4ax.com>
Content-Type: text/plain; charset=us-ascii
Hello
I've been googling for a couple of days now, but still can't
figure out what to put in zapata.conf to get it to report CID.
Unless I'm mistaken, France uses ETSI FSK for CID method and bell 202
as CID FSK Standard:
http://img219.imageshack.us/img219/7207/linksys3102cid1jj7.jpg
http://img219.imageshack.us/img219/4625/linksys3102cid2ld5.jpg
Does Zaptel support those on Digium TDM400 clones like those from
OpenVox?
Thank you.
------------------------------
Message: 9
Date: Mon, 22 Oct 2007 15:35:40 -0400
From: Rurouni Alucard <rakh at dangerclan.net>
Subject: Re: [asterisk-users] Authenticate by IP?
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <471CFB8C.7070200 at dangerclan.net>
Content-Type: text/plain; charset="utf-8"
Saludos Carlos,
Como vas a recibir las llamadas via SIP, puedes especificar el IP del
host que te enviara las llamadas, por ej.
Este es un bloque que tengo definido en el SIP.conf de uno de mis
servers para enrutar las llamadas internacionales y a telefonos moviles
utilizando un proveedor de terminacion.
[oficina]
type=peer
context=from_office ; Esto va a mi 'extensions.conf'
host=200.88.42.29 ; Este es el ip publico en la oficina (estatico)
nat=no
canreinvite=no
qualify=yes
disallow=all
allow=g729
allow=ulaw
Creo que eso contesta tu pregunta.
--
Jose P. Espinal
slackware-es.com
Carlos Chavez wrote:> I have a customer that needs an Asterisk server to sell minutes
for> cell phones in Mexico. I do not see a problem with that since he will
> get the calls by SIP and then use GSM adapters to get the calls into
the> GSM network. My problem is that his customers only want to be
> identified by IP and not by a username and password. Is there a way
to> authenticate just by using an IP address?
>
>
>
------------------------------------------------------------------------>
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
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Message: 10
Date: Mon, 22 Oct 2007 21:56:09 +0200
From: Vincent <vincent.delporte at bigfoot.com>
Subject: Re: [asterisk-users] Prompting for number when CID number not
sent?
To: asterisk-users at lists.digium.com
Message-ID: <s10qh3lkehro0j6m8q74phuhqthkhufetc at 4ax.com>
Content-Type: text/plain; charset=us-ascii
On Mon, 22 Oct 2007 10:14:41 -0400, Jared Smith <jsmith at digium.com>
wrote:>Instead of ${callerid} here (which probably isn't working for you
>anyway), you probably want to use the CALLERID dialplan function to
>retrieve the CallerID number, like this:
Thanks for the tip. It'll come in handy... once I finally get the TDM
card to report CID :-)
------------------------------
Message: 11
Date: Mon, 22 Oct 2007 21:57:47 +0100
From: Joao Pereira <joao.pereira at fccn.pt>
Subject: [asterisk-users] dial-out call queue
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <471D0ECB.4090909 at fccn.pt>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Is it possible to implement a dial-out call queue in Asterisk?
My idea is to give Asterisk a list of numbers, and then he makes the
calls and delivers the calls to a call queue.
Then, the agents will answer the calls.
Is this possible?
Thanks
Regards
Joao pereira
------------------------------
Message: 12
Date: Mon, 22 Oct 2007 15:59:18 -0500
From: Carlos Chavez <cursor at telecomabmex.com>
Subject: Re: [asterisk-users] Authenticate by IP?
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <1193086758.32639.2.camel at cursor.telecomabmex.com>
Content-Type: text/plain; charset="utf-8"
On Mon, 2007-10-22 at 15:35 -0400, Rurouni Alucard
wrote:> Saludos Carlos,
>
> Como vas a recibir las llamadas via SIP, puedes especificar el IP del
> host que te enviara las llamadas, por ej.
>
> Este es un bloque que tengo definido en el SIP.conf de uno de mis
> servers para enrutar las llamadas internacionales y a telefonos
> moviles utilizando un proveedor de terminacion.
>
> [oficina]
> type=peer
> context=from_office ; Esto va a mi 'extensions.conf'
> host=200.88.42.29 ; Este es el ip publico en la oficina (estatico)
>
> nat=no
> canreinvite=no
> qualify=yes
> disallow=all
> allow=g729
> allow=ulaw
>
> Creo que eso contesta tu pregunta.
>
>
Hola Jos?. Gracias por tu contestaci?n. Lo que me estas
especificando
el para hacer llamadas de salida (PEER). Yo necesito autentificar a un
usuario de entrada, voy a intentar haciendo algo parecido solo cambiando
a type=user para ver si as? funciona.
--
Telecomunicaciones Abiertas de M?xico S.A. de C.V.
Carlos Ch?vez Prats
Director de Tecnolog?a
+52-55-91169161 ext 2001
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------------------------------
Message: 13
Date: Mon, 22 Oct 2007 17:12:54 -0400
From: "BerkHolz, Steven" <StevenBerkHolz at hirotecamerica.com>
Subject: [asterisk-users] Split asterisk in two ?? One TDM and One IP
only??
To: "asterisk-users at lists.digium.com"
<asterisk-users at lists.digium.com>
Message-ID:
<5C4D356F944BC042A83B3828DF3E0B4412B01CC6D3 at tg12.tescogroup.com>
Content-Type: text/plain; charset="iso-8859-1"
I have built an asterisk server with a TE412P card on a Dell 2950.
It does incoming calls via DID over PRI, our IVR, SIP/IAX extensions,
Fax/Analog extensions via an old PBX via PRI, voicemail, etc.
My issue now is that I find it difficult to test/upgrade to new
versions.
This is what I am thinking of doing.
Server1
Keep one physical server just for TDM functions.
PRI to Telco
PRI to old PBX for Fax. (basically using it as a mux)
Keep meetme here for Digium card timing.
Server2
Build a new asterisk install within Xen VM with data stored on an iSCSI
SAN. This would be all IP.
IAX and SIP extensions.
IAX and SIP providers.
IVR
Voicemail
Web access to voicemail
CDR
This way I can test different versions of the features of Server2 (clone
with different IP) without affecting production.
I assume that I just use an IAX or SIP trunk between the two asterisk
servers.
Does this make sense?
Are others doing similar?
Are there any other features that require the TDM card besides PRI, Fax
and Meetme?
I have heard of people using Xen for IP only asterisk, but are there any
known gotchas?
Thanks,
Thank You,
Steven BerkHolz
------------------------------
Message: 14
Date: Mon, 22 Oct 2007 23:18:06 +0200
From: Vincent <vincent.delporte at bigfoot.com>
Subject: Re: [asterisk-users] [France CID] Does Zaptel support ETSI
FSK?
To: asterisk-users at lists.digium.com
Message-ID: <ca4qh355m6f4t7o4q6ov7nmm2vd8kdhehp at 4ax.com>
Content-Type: text/plain; charset=us-ascii
On Mon, 22 Oct 2007 21:19:27 +0200, Vincent
<vincent.delporte at bigfoot.com> wrote:>Does Zaptel support those on Digium TDM400 clones like those from
>OpenVox?
Pff, finally found what it was: It had nothing to do with zaptel, and
everything to do with extensions.conf:
=======exten => s,1,NoOp(Got a call)
;nothing displayed
exten => s,n,Verbose(${CALLERID})
exten => s,n,Verbose(${CALLERIDNAME})
exten => s,n,Verbose(${CALLERIDNUM})
exten => s,n,NoOp(${CALLERID})
exten => s,n,Verbose(${CALLERID})
;CID at last!
exten => s,n,Verbose(${CALLERID(num)})
=======
I'm running Asterisk 1.4. Does someone know why only the last
statement does display the CID number while the others print nothing?
Thank you.
------------------------------
Message: 15
Date: Mon, 22 Oct 2007 23:20:19 +0200
From: Vincent <vincent.delporte at bigfoot.com>
Subject: Re: [asterisk-users] Extensions.conf for basic IVR?
To: asterisk-users at lists.digium.com
Message-ID: <mt4qh3drsd077r9gqdeakqtlsbpkga7d3k at 4ax.com>
Content-Type: text/plain; charset=us-ascii
On Mon, 22 Oct 2007 09:06:00 +0200, randulo <spamsucks2005 at gmail.com>
wrote:>The first ten sites that come up, including voip-info.org, usually a
>good place to look first, each have full examples. Look also for the
>background application wich is used to play the file, get input and
>jump to the extension entered.
Thanks. The problem with information on the Net is that the
development of Asterisk moves quite fast, making some/a lot of
information obsolete, something newbies aren't necessarily aware of.
2008 might be a good year to update "* - The future of telephony" :-)
------------------------------
Message: 16
Date: Mon, 22 Oct 2007 23:33:34 +0200
From: Christian Victor <christian at victormedia.de>
Subject: Re: [asterisk-users] 16 ports wanted
To: Gergo Csibra <csibra at gmail.com>, Asterisk Users Mailing List -
Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <471D172E.5070105 at victormedia.de>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Gergo Csibra schrieb:> Well, using more than one TDM card in your PC is not a good idea,
> because of interrupts. If you have to have 16 FXO you can more
> options:
>
> 1. Using TDM2400P with 4 FXO modules ($1775)
> 2. Using Xorcom's Astribank (external) ($1170)
> 3. Using some T1/E1 card with Channel Bank (more expensive)
>
4. Using Sangoma's A200 with 8 (up to 12) dual-FXO modules (ca. $1.200)
5. Using Sangoma's A400 with 8 (up to 24) dual-FXO modules (ca. $1.350)
Both Sangoma cards can be equipped with Octasic hardware-EC for ca. $300
more and are available in PCI(-X) and PCIexpress versions. For the A200
you need an additional case slot (does not need another PCI connector)
for every 4 ports over 4. The same goes for the A400 on every 12 ports
over 12.
Christian
------------------------------
Message: 17
Date: Mon, 22 Oct 2007 16:41:19 -0500
From: "Erik Anderson" <erikerik at gmail.com>
Subject: Re: [asterisk-users] Extensions.conf for basic IVR?
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Message-ID:
<fc40260f0710221441u12274367ja0b04424a1acf241 at mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1
On 10/22/07, Vincent <vincent.delporte at bigfoot.com>
wrote:>
> 2008 might be a good year to update "* - The future of telephony"
:-)
Version 2 of TFOT was just released a few weeks ago...
http://downloads.oreilly.com/books/9780596510480.pdf
--
Erik Anderson
http://andersonfam.org
------------------------------
Message: 18
Date: Mon, 22 Oct 2007 17:57:44 -0400
From: Jared Smith <jsmith at digium.com>
Subject: Re: [asterisk-users] [France CID] Does Zaptel support ETSI
FSK?
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <1193090264.3362.62.camel at hockey.jaredsmith.net>
Content-Type: text/plain
On Mon, 2007-10-22 at 23:18 +0200, Vincent wrote:> =======> exten => s,1,NoOp(Got a call)
>
> ;nothing displayed
> exten => s,n,Verbose(${CALLERID})
> exten => s,n,Verbose(${CALLERIDNAME})
> exten => s,n,Verbose(${CALLERIDNUM})
> exten => s,n,NoOp(${CALLERID})
> exten => s,n,Verbose(${CALLERID})
>
> ;CID at last!
> exten => s,n,Verbose(${CALLERID(num)})
> =======>
> I'm running Asterisk 1.4. Does someone know why only the last
> statement does display the CID number while the others print nothing?
Beginning with Asterisk 1.4, we moved all of the CallerID functionality
from channel variables and applications to a single CALLERID dialplan
function. This should have been noted in UPGRADE.txt. I also tried to
warn you about it in my last email in this thread, but I guess I should
have been more specific.
--
Jared Smith
Community Relations Manager
Digium, Inc.
------------------------------
Message: 19
Date: Mon, 22 Oct 2007 16:00:19 -0500
From: Carlos Chavez <cursor at telecomabmex.com>
Subject: Re: [asterisk-users] Authenticate by IP?
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <1193086819.32639.3.camel at cursor.telecomabmex.com>
Content-Type: text/plain; charset="utf-8"
On Mon, 2007-10-22 at 15:13 -0400, joakimsen at gmail.com
wrote:> On 10/22/07, Carlos Chavez <cursor at telecomabmex.com> wrote:
> > I have a customer that needs an Asterisk server to sell
minutes for> > cell phones in Mexico. I do not see a problem with that since he
will> > get the calls by SIP and then use GSM adapters to get the calls into
the> > GSM network. My problem is that his customers only want to be
> > identified by IP and not by a username and password. Is there a way
to> > authenticate just by using an IP address?
> >
>
> There certainly is.
>
And could you please point me in the right direction?
--
Telecomunicaciones Abiertas de M?xico S.A. de C.V.
Carlos Ch?vez Prats
Director de Tecnolog?a
+52-55-91169161 ext 2001
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Message: 20
Date: Tue, 23 Oct 2007 00:04:14 +0200
From: Vincent <vincent.delporte at bigfoot.com>
Subject: Re: [asterisk-users] Extensions.conf for basic IVR?
To: asterisk-users at lists.digium.com
Message-ID: <jd7qh3l208uq9mplkrgjqvc37asklvek3q at 4ax.com>
Content-Type: text/plain; charset=us-ascii
On Mon, 22 Oct 2007 16:41:19 -0500, "Erik Anderson"
<erikerik at gmail.com> wrote:>Version 2 of TFOT was just released a few weeks ago...
Just had to ask :-) Thanks.
------------------------------
Message: 21
Date: Mon, 22 Oct 2007 18:06:41 -0400
From: "joakimsen at gmail.com" <joakimsen at gmail.com>
Subject: Re: [asterisk-users] Authenticate by IP?
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Message-ID:
<23fd749a0710221506m6fdc816dkf098ffd783868d7c at mail.gmail.com>
Content-Type: text/plain; charset=UTF-8
La configuraci?n de Jose esta correcta. Cuando usas un "peer" en
sip.conf Asterisk usa el hostname or el IP para autenticar. Cuando
usas un "user" la autenticaci?n se basa en el usuario y la
contrase?a, cual en su caso no existe.
On 10/22/07, Carlos Chavez <cursor at telecomabmex.com>
wrote:> Hola Jos?. Gracias por tu contestaci?n. Lo que me estas
especificando> el para hacer llamadas de salida (PEER). Yo necesito autentificar a
un> usuario de entrada, voy a intentar haciendo algo parecido solo
cambiando> a type=user para ver si as? funciona.
>
> --
> Telecomunicaciones Abiertas de M?xico S.A. de C.V.
> Carlos Ch?vez Prats
> Director de Tecnolog?a
> +52-55-91169161 ext 2001
>
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
------------------------------
Message: 22
Date: Mon, 22 Oct 2007 17:07:18 -0500
From: Victor Toofic <toofics at gmail.com>
Subject: Re: [asterisk-users] Authenticate by IP?
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <20071022220718.GA16853 at localhost.localdomain>
Content-Type: text/plain; charset=iso-8859-1
El Mon, Oct 22 de 2007 a las 15:59 -0500, Carlos Chavez
comentaba:> Hola Jos?. Gracias por tu contestaci?n. Lo que me estas
especificando> el para hacer llamadas de salida (PEER). Yo necesito autentificar a
un> usuario de entrada, voy a intentar haciendo algo parecido solo
cambiando> a type=user para ver si as? funciona.
type=peer also works for incoming calls. In this case (peer) asterisk
only checks
the IP the call is coming from and uses the context you defined there.
If
you use type=user you will need to specify a username and a secret.
--
Greetings..
V?ctor Toofic
------------------------------
Message: 23
Date: Tue, 23 Oct 2007 00:11:39 +0200
From: Thomas Winter <thowinter at googlemail.com>
Subject: [asterisk-users] bristuff: music on hold but no dialoptions
tT defined.
To: asterisk-users at lists.digium.com
Message-ID: <200710230011.39639.thowinter at googlemail.com>
Content-Type: text/plain; charset="utf-8"
Hi,
Iam dialing from NT ptp to SIP provider.
Sometimes Asterisk is doing music on hold but there are no options like
t or T
in the dial command. As an result the channel got lost and an Hangup
occurs.
Iam using bristuff-0.3.0-PRE-1y-i on an QuadBri card.
Any solution for this?
Oct 22 11:20:06 VERBOSE[911] logger.c: -- SIP/sip-08206a30 answered
Zap/8-1
Oct 22 11:20:23 VERBOSE[29983] logger.c: -- Started music on hold,
class 'default', on channel 'Zap/8-1'
Oct 22 11:20:46 VERBOSE[29983] logger.c: -- Stopped music on hold on
Zap/8-1
Oct 22 11:20:46 VERBOSE[29983] logger.c: -- Started music on hold,
class 'default', on channel 'Zap/8-1'
Oct 22 11:20:55 VERBOSE[911] logger.c: == Spawn extension (macro-call,
s, 2)
exited non-zero on 'Zap
/8-1' in macro 'tmp_call'
------------------------------
Message: 24
Date: Tue, 23 Oct 2007 00:15:28 +0200
From: Vincent <vincent.delporte at bigfoot.com>
Subject: Re: [asterisk-users] [France CID] Does Zaptel support ETSI
FSK?
To: asterisk-users at lists.digium.com
Message-ID: <f58qh3dbdlg1n3dc4ejc385hip6c8nrtug at 4ax.com>
Content-Type: text/plain; charset=us-ascii
On Mon, 22 Oct 2007 17:57:44 -0400, Jared Smith <jsmith at digium.com>
wrote:>Beginning with Asterisk 1.4, we moved all of the CallerID functionality
>from channel variables and applications to a single CALLERID dialplan
>function. This should have been noted in UPGRADE.txt. I also tried to
>warn you about it in my last email in this thread, but I guess I should
>have been more specific.
No problem. I should have read it more closely, but due to the number
of people having problems with Zaptel and CID, I was focused on that
part. Should have started asking people what the correct way was to
read CID information in Asterisk 1.4... Thanks.
------------------------------
Message: 25
Date: Mon, 22 Oct 2007 16:05:06 -0700 (PDT)
From: bilal ghayyad <bilmar_gh at yahoo.com>
Subject: Re: [asterisk-users] G729a codecs + Asterisk 1.4.11
To: asterisk-users at lists.digium.com
Message-ID: <375529.72511.qm at web53908.mail.re2.yahoo.com>
Content-Type: text/plain; charset=iso-8859-1
Dear Marc;
I readed your email about the codec G729a and I am now
also need to install the codec on my Asterisk.
I typed from Asterisk CLI:
core show version and I got the following:
Asterisk SVN-branch-1.4-r72556 built by root @
localhost.localdomain on a i686 running Linux on
2007-06-30 13:08:08 UTC
So I beleive that my processor is i686, correct? But I
am not able to know which one to download:
The x86-32 or x86-64 ? Can you please advise.
Also, the nocona or the opteron versions?
Regards
Bilal
-------------------
Good Morning,
Any help would be grateful to help me understanding
what's wrong...
I have bought 2 g729a licenses to digium and I would
like to have them
works...
My processor is an Intel(R) Xeon(R) CPU
E5310 @ 1.60GHz (4
processors)
so I have downloaded the
http://downloads.digium.com/pub/telephony/codec_g729/asterisk-1.4/x86-64
/codec_g729a_v32_nocona.tar.gz
codec
I have registered my license, copied the
codec_g729a.so into the
/usr/lib/asterisk/modules folder and restarted my
asterisk
But on the CLI when I type
asterisk*CLI> show modules like 72
Module Description
Use Count
codec_g726.so ITU G.726-32kbps G726
Transcoder
0
format_g729.so Raw G729 data
0
format_g726.so Raw G.726
(16/24/32/40kbps) data
0
format_g723.so G.723.1 Simple
Timestamp File Format
0
The codec_g729a.so doesn't appear..........
Any idea how to solve the problem.....
Thanks
Best Regards,
Marc LEURENT
__________________________________________________
Do You Yahoo!?
Tired of spam? Yahoo! Mail has the best spam protection around
http://mail.yahoo.com
------------------------------
Message: 26
Date: Mon, 22 Oct 2007 16:09:39 -0700
From: Ira <ira at extrasensory.com>
Subject: Re: [asterisk-users] [France CID] Does Zaptel support ETSI
FSK?
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <0MKpCa-1Ik6P21fVG-0004hf at mrelay.perfora.net>
Content-Type: text/plain; charset="us-ascii"; format=flowed
At 02:18 PM 10/22/2007, you wrote:>;nothing displayed
>exten => s,n,Verbose(${CALLERID})
>exten => s,n,Verbose(${CALLERIDNAME})
>exten => s,n,Verbose(${CALLERIDNUM})
>exten => s,n,NoOp(${CALLERID})
>exten => s,n,Verbose(${CALLERID})
>
>;CID at last!
>exten => s,n,Verbose(${CALLERID(num)})
>=======>
>I'm running Asterisk 1.4. Does someone know why only the last
>statement does display the CID number while the others print nothing?
try adding a wait(1) right in the beginning, worked for me.
Ira
------------------------------
Message: 27
Date: Mon, 22 Oct 2007 19:15:55 -0400
From: Jason Lixfeld <jason+lists.asterisk at lixfeld.ca>
Subject: [asterisk-users] Voicemail playback on iPhone
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <392077D9-82C9-49C6-A758-4F7AB024DE1A at lixfeld.ca>
Content-Type: text/plain; charset=US-ASCII; format=flowed
Anyone managed to get this to work? What's the recipe?
------------------------------
Message: 28
Date: Tue, 23 Oct 2007 01:35:13 +0200
From: "Yitzhak Bar Geva" <yitzhakbg at gmail.com>
Subject: [asterisk-users] NAT traversal packet loss measurement
To: asterisk-users at lists.digium.com
Message-ID:
<3c0677320710221635j3b52b870qbe6ddda00cb6e77b at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
How can one measure the effect of NAT traversal packet loss?
We currently have no solution for NAT traversal for our SIP clients.
There
is no doubt that packets are getting lost. What is not clear is how much
damage this does. On the face of it, everything seems fine. Could this
be
so? Perhaps we're suffering a degradation in quality or our call setup
times
could be improved. How can we measure this?
What's the simplest method of preventing packet loss due to NAT
traversal in
a SIP environment?
Thanks,
Yitzhak
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Message: 29
Date: Mon, 22 Oct 2007 16:38:17 -0700
From: "Ron Stephan" <elvis at elvisware.com>
Subject: Re: [asterisk-users] Voicemail playback on iPhone
To: "'Asterisk Users Mailing List - Non-Commercial
Discussion'"
<asterisk-users at lists.digium.com>
Message-ID: <20071022233821.6CC4916983 at mailspool3.panix.com>
Content-Type: text/plain; charset="us-ascii"
Trick question I assume?
It was mind numbingly simple on my iPhone...(though none of the voice
mail worked when London a few weeks ago).
- tap voice mail -
- tap speaker (upper right) until it turns blue (is activate)
- tap the message you want to playback
- use assorted controls to delete - replay etc.
Now...if the question is ... how do you get asterisk voice mail to show
up on an iPhone...I am all ears. Groovy concept - if
anybody has a hack - I'd love to see it.
Elvis
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Jason
Lixfeld
Sent: Monday, October 22, 2007 4:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Voicemail playback on iPhone
Anyone managed to get this to work? What's the recipe?
_______________________________________________
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------------------------------
Message: 30
Date: Tue, 23 Oct 2007 13:30:55 +1300
From: Matt Riddell <matt at venturevoip.com>
Subject: Re: [asterisk-users] NAT traversal packet loss measurement
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <471D40BF.6070907 at venturevoip.com>
Content-Type: text/plain; charset=ISO-8859-1
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Yitzhak Bar Geva wrote:> How can one measure the effect of NAT traversal packet loss?
> We currently have no solution for NAT traversal for our SIP clients.
There> is no doubt that packets are getting lost. What is not clear is how
much> damage this does. On the face of it, everything seems fine. Could this
be> so? Perhaps we're suffering a degradation in quality or our call setup
times> could be improved. How can we measure this?
> What's the simplest method of preventing packet loss due to NAT
traversal in> a SIP environment?
NAT is unlikely to cause a percentage of packets to get lost.
Normally you'd have one way audio if NAT was causing a problem (i.e.
100% packet loss).
The only other situation in which it might happen is where the NAT
router decides to close a port mapping (thereby blocking incoming calls
to the customer's device).
But if you're looking for packet loss there are a number of other things
to check first.
I wouldn't do VoIP across the WAN without at least some packet shaping
but hey.
- --
Kind Regards,
Matt Riddell
Director
_______________________________________________
http://www.venturevoip.com (Great new VoIP end to end solution)
http://www.venturevoip.com/news.php (Daily Asterisk News - html)
http://feeds.venturevoip.com/AsteriskNews (Daily Asterisk News - rss)
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