Re: [asterisk-users] chan_isdn with HFC-compatible

[Michael Kamleitner]

On 7/18/07, Ex Vito <[EMAIL PROTECTED]> wrote:


>
> Asterisk is loading the chan_misdn and lists mISDN when issueing "show
> channeltypes" -  however it indicates "Devicestate - No". when I look
for
> "misdn show stacks", it lists the single port of the ISDN-card, however
> indicates "L2Link DOWN, L1LinkDOWN". so I  guess theres something wrong,
> unfortunately I've no idea  on what to check next...any pointers? maybe
I'm
> missing something obvious, as this is my first installation of Asterisk
+
> ISDN...
>

  Checklist:

  - The cable (after all L1 seems to be down)
  - If asterisk is to behave as a phone, check that
the port is configured in TE mode
  - Verify PTP vs PTMP



thx for your suggestions. just to make sure: TE-mode is configured in
zapata.conf, right? the settings would be:

switchtype = euroisdn
signalling = bri_cpe ( or signalling = bri_cpe_ptmp  not sure about  this -
guess I should check mISDN.conf to see if I should go for PTP or PTMP,
right?)

I'm gonna check this when I am back at the machine later today...

thx!


 If needed:


  - Take a look at /var/log/asterisk/misdn.log
  - Increase "verbosity" in /etc/misdn-init.conf
debug=3 should be a good starting point
(reload mISDN modules after this change)
--
  exvito

___
--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

___
--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

Re: [asterisk-users] Asterisk Hosting (Dedicated Servers)

[marcelobiz]

Thanks Gordon for your response,
It helped me a lot ...

I should have done this already, but the QoS issue was holding me back ... 
Actually, for now ... I'll start with just a backup box and test how it goes ...
I was looking for a kind of dedicated server hosting with a MPLS network that 
could give me a good level of QoS, but I didn't find it ...

Thanks again,

Marcelo

P.S: If anyone has another point ... I would be grateful in reading your 
opinions

-- Original message -- 
From: Gordon Henderson <[EMAIL PROTECTED]> 

> On Tue, 17 Jul 2007, [EMAIL PROTECTED] wrote: 
> 
> > Hello guys, 
> > 
> > Does anyone has an Asterisk server hosted off-site ? Like in those data 
> > centers that do web hosting in dedicated servers ? 
> > 
> > Is there a hosting company that has a special plan to host voip services 
> > like this, or usually is hosted in those dedicated servers like the ones 
> > I asked above ? 
> > 
> > What about QoS ? I know that most (if not all) are connected to their 
> > switch through a 10Mbps/100Mbps port ? But ... without a QoS rule ... 
> > even with that speed doesn't it affect the quality of voice ? 
> > 
> > Can you please tell me your experience ? Or point me some good hosting 
> > companies ? 
> 
> It can be a bit of a minefield - especially if it's an area you've not 
> looked into before. 
> 
> I've been doing this (in a very minor way) for over 10 years now. 
> 
> So I run what could be described as a small hosting company, however, my 
> hosts are currently inside another ISPs data centre rather than in a 
> "neutral" data centre, so I get 100% of my Internet connectivity from my 
> upstream ISP, and I am relying on them to do the right thing with having 
> multiple transit providers and redundant network routing, UPSs and 
> generators, all of which they have to my satisfaction. 
> 
> The next step for me would be to host in some neutral facility, get my own 
> IP address space, my own AS number, then connect into multiple transit 
> providers and arrange peering through the various neutral connection 
> points that exist in the UK (LINX, MaNAP, etc.) I'm not big enough for 
> that ... yet ;-) 
> 
> So I have routers and switches and connect into the ISP via a redundant 
> mechanism (VRRP). I can apply QoS in my own routers, so that traffic from 
> the Asterisk servers can be prioritised over the traffic from the LAMPy 
> type servers, however, without the co-operation of the upstream ISP(s), 
> you can't effectively apply QoS to the incoming traffic. (Fortunately in 
> my instance, incoming is so much lighter than outgoing, and their network 
> in not oversubscribed, so it's not really an issue) 
> 
> The easiest way to start, would be to simply place hosts inside another 
> ISPs network, and rely on them for "quality" transit - ie. make sure they 
> have multiple transit providers themselves, good power supplies, UPS, 
> generators, etc. and if they are good and don't oversell their bandwidth 
> then for the most part you'll be just fine. Once you have several hosts 
> you might want to look at having your own router(s) and switch(es), but 
> this will depend on how the hosting company operates (and your budget!) 
> 
> Finding that hosting company where you live is the challenge though! You 
> need to ask lots of questions - can you get physical access to the 
> servers? is it within driving distance? do you have remote screen & 
> keyboard facilities? (or if they offer "remote hands" and if so, how much 
> do they charge?) How well do they connect to the world in general, and do 
> they charge separate for power or bandwidth (and is bandwidth in terms of 
> speed, or is it per bit pricing, or some combination of the 2?) 
> 
> Start phoning & emailling - how fast do they answer the phone, or return 
> email will be a good metric too... 
> 
> In the UK, London appears to be power starved right now - it would appear 
> that bandwidth is virtually free, but you'll get charged per amp used! 
> Outside London you rarely have this restriction, but then bandwidth is 
> costly as it's got to be back-hauled to London (or Manchester), so "local 
> knowledge" is something you'll need to find out about. 
> 
> Good luck! 
> 
> Gordon 
> 
> ___ 
> --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 ___
--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

Re: [asterisk-users] Asterisk and Mitel 3300 ICP

[Joesph O]

Good morning, it now works, failure was due to a misconfigured/misunderstood
Class of Restriction Group Assignment for the SIP Trunk Routes on the
3300ICP.

Now Asterisk can call the world through the Mitel and incoming calls (DID,
operator transfers etc) to Asterisk via the 3300ICP, all work.

Interesting side note - both phone systems have same range of
extensions e.g100 - 299 (just an example)and we created routes on both
to point to the
other for the range, of course, an extension should only exist on one at any
time. therefore, if an extension does not exist locally, it is routed to the
other and vice versa, this way, we keep same range of extensions and this
has helped with migrating users who do not want to trade their loved
Extension & DID number for anything. will continue to test and share
results.

Joesph O.


On 7/9/07, Joesph O <[EMAIL PROTECTED]> wrote:


Good day everyone,

I have Asterisk and Mitel 3300 ICP communicating via SIP. Calls to and
from extensions on both sides are completing successfully (details on config
coming soon). Problem is that calls from Asterisk to PSTN via E1 on Mitel
3300 ICP are rejected. What do I need to do to get Asterisk to route PSTN
calls through it successfully?

Here is an extract of the log on Asterisk whenever I try to call PSTN
through 3300ICP, in this case, Extension 2540 on Asterisk called 2345678, 9
is a leading digit -

Jul 7 16:48:07 DEBUG[6860] chan_sip.c: Outgoing Call for 92345678
Jul 7 16:48:07 VERBOSE[6860] logger.c: -- Called Mitel3300ICP/92345678
Jul 7 16:48:07 DEBUG[6860] channel.c: Prodding channel 'SIP/2540-b7904a98'
Jul 7 16:48:07 VERBOSE[6860] logger.c: -- Started music on hold, class
'24', on SIP/2540-b7904a98
Jul 7 16:48:07 DEBUG[6860] channel.c: Scheduling timer at 160 sample
intervals
Jul 7 16:48:07 DEBUG[6270] chan_sip.c: (Provisional) Stopping
retransmission (but retaining packet) on '
[EMAIL PROTECTED]' Request 102: Found
Jul 7 16:48:07 DEBUG[6270] chan_sip.c: Acked pending invite 102
Jul 7 16:48:07 DEBUG[6270] chan_sip.c: Stopping retransmission on '
[EMAIL PROTECTED]' of Request 102: Match
Found
Jul 7 16:48:07 DEBUG[6860] channel.c: Generator got voice, switching to
phase locked mode
Jul 7 16:48:07 DEBUG[6860] channel.c: Scheduling timer at 0 sample
intervals
Jul 7 16:48:07 DEBUG[6270] chan_sip.c: (Provisional) Stopping
retransmission (but retaining packet) on '
[EMAIL PROTECTED]' Request 103: Found
Jul 7 16:48:08 DEBUG[6270] chan_sip.c: Acked pending invite 103
Jul 7 16:48:08 DEBUG[6270] chan_sip.c: Stopping retransmission on '
[EMAIL PROTECTED]' of Request 103: Match
Found
Jul 7 16:48:08 WARNING[6270] chan_sip.c: Forbidden - wrong password on
authentication for INVITE to '"Tester" < sip:[EMAIL PROTECTED]>;tag=as07fef065'
Jul 7 16:48:08 VERBOSE[6860] logger.c: -- SIP/Mitel3300ICP-0832de50 is
circuit-busy

___
--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

Re: [asterisk-users] Any way to determine remote Asterisk version

[marcelobiz]

Andrew,

I don't know about your first question ... but my experience with IPcomms was 
not that good ...
I was trying their service ... (DIDs) and I got a lot of dead spots in the 
voice calls ... One guy from support was very friendly, trying to resolve the 
issue, but I cancelled the service and nothing about the money back guaranteed 
...



-- Original message -- 
From: "Andrew Joakimsen" <[EMAIL PROTECTED]> 
A long time ago (Asterisk 0.x, 1.0.x) my experience is that there were alot of 
interoperability issues, a common troubleshooting issue was to make sure all 
endpoints where using the latest version of Asterisk. I have not seen these 
issues in a while. 

However I've been working with a customer of mine and this ITSP called IP 
Communications (IPComms.net) well turns out we have had constant problems since 
the first day. Turns out (I found this out approx 1 month ago) that the 
"version of Asterisk" they are using is " [EMAIL PROTECTED] 1.x"  So basically 
is there any way to determine the version of Asterisk being used? I cant trust 
anything IP Communications says.--- Begin Message ---
___
--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--- End Message ---
___
--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

[asterisk-users] AudioCodec MP114

[Al lists]

Hi list,
I'm trying to use an AudioCodec Mp114, 4 FXO  Media gateway.
I went trough what i could find in wiki and also trixbox forum and so far no
good results.
i had this in trixbox frorum :
http://www.trixbox.org/forums/trixbox-forums/trunks/trixbox-2-2-and-audio-codes-mp-114-fxo-setup
any successful installation? or how to?
___
--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

Re: [asterisk-users] Asterisk 1.4, Unicall and Nextel...

[Carlos Chavez]

On Tue, 2007-07-17 at 19:30 -0500, Moises Silva wrote:
> In order to help you I need testcall traces, with max level of
> logging, of incoming Nextel calls.
> 
Here is the log file from a couple of calls from a Nextel phone:

[Jul 17 21:46:08] WARNING[4771] chan_unicall.c: MFC/R2 UniCall/56
<- 0001  [1/   1/Idle  /Idle ]
[Jul 17 21:46:08] WARNING[4771] chan_unicall.c: MFC/R2 UniCall/56
Detected
[Jul 17 21:46:08] WARNING[4771] chan_unicall.c: MFC/R2 UniCall/56 Making
a new call with CRN 32769
[Jul 17 21:46:08] WARNING[4771] chan_unicall.c: MFC/R2 UniCall/56 1101
->  [2/   2/Idle  /Idle ]
[Jul 17 21:46:08] NOTICE[4771] chan_unicall.c: Unicall/56 event Detected
[Jul 17 21:46:08] WARNING[4771] chan_unicall.c: MFC/R2 UniCall/56
<- 9 on  [2/   2/Seize ack /Seize ack]
[Jul 17 21:46:08] WARNING[4771] chan_unicall.c: MFC/R2 UniCall/56 1 on
->  [2/   2/Seize ack /Seize ack]
[Jul 17 21:46:08] WARNING[4771] chan_unicall.c: MFC/R2 UniCall/56
<- 9 off [2/   2/Group A   /DNIS request ]
[Jul 17 21:46:08] WARNING[4771] chan_unicall.c: MFC/R2 UniCall/56 1 off
->  [2/   2/Group A   /DNIS request ]
[Jul 17 21:46:08] WARNING[4771] chan_unicall.c: MFC/R2 UniCall/56
<- 9 on  [2/   2/Group A   /DNIS request ]
[Jul 17 21:46:08] WARNING[4771] chan_unicall.c: MFC/R2 UniCall/56 1 on
->  [2/   2/Group A   /DNIS request ]
[Jul 17 21:46:08] WARNING[4771] chan_unicall.c: MFC/R2 UniCall/56
<- 9 off [2/   2/Group A   /DNIS request ]
[Jul 17 21:46:08] WARNING[4771] chan_unicall.c: MFC/R2 UniCall/56 1 off
->  [2/   2/Group A   /DNIS request ]
[Jul 17 21:46:08] WARNING[4771] chan_unicall.c: MFC/R2 UniCall/56
<- 9 on  [2/   2/Group A   /DNIS request ]
[Jul 17 21:46:08] WARNING[4771] chan_unicall.c: MFC/R2 UniCall/56 1 on
->  [2/   2/Group A   /DNIS request ]
[Jul 17 21:46:08] WARNING[4771] chan_unicall.c: MFC/R2 UniCall/56
<- 9 off [2/   2/Group A   /DNIS request ]
[Jul 17 21:46:08] WARNING[4771] chan_unicall.c: MFC/R2 UniCall/56 1 off
->  [2/   2/Group A   /DNIS request ]
[Jul 17 21:46:08] WARNING[4771] chan_unicall.c: MFC/R2 UniCall/56
<- 8 on  [2/   2/Group A   /DNIS request ]
[Jul 17 21:46:08] WARNING[4771] chan_unicall.c: MFC/R2 UniCall/56 6 on
->  [2/   2/Group A   /DNIS request ]
[Jul 17 21:46:08] WARNING[4771] chan_unicall.c: MFC/R2 UniCall/56
<- 8 off [2/   2/Group C   /Category req ]
[Jul 17 21:46:08] WARNING[4771] chan_unicall.c: MFC/R2 UniCall/56 6 off
->  [2/   2/Group C   /Category req ]
[Jul 17 21:46:08] WARNING[4771] chan_unicall.c: MFC/R2 UniCall/56
<- 9 on  [2/   2/Group C   /Category req ]
[Jul 17 21:46:08] WARNING[4771] chan_unicall.c: MFC/R2 UniCall/56 1 on
->  [2/   2/Group C   /Category req ]
[Jul 17 21:46:08] WARNING[4771] chan_unicall.c: MFC/R2 UniCall/56
<- 9 off [2/   2/Group C   /ANI request  ]
[Jul 17 21:46:08] WARNING[4771] chan_unicall.c: MFC/R2 UniCall/56 1 off
->  [2/   2/Group C   /ANI request  ]
[Jul 17 21:46:08] WARNING[4771] chan_unicall.c: MFC/R2 UniCall/56
<- F on  [2/   2/Group C   /ANI request  ]
[Jul 17 21:46:08] WARNING[4771] chan_unicall.c: MFC/R2 UniCall/56 R2
prot. err. [2/   2/Group C   /ANI request  ] cause 32772 -
Unexpected MF6 signal
[Jul 17 21:46:08] WARNING[4771] chan_unicall.c: MFC/R2 UniCall/56 1001
->  [1/   1/Idle  /Idle ]
[Jul 17 21:46:08] NOTICE[4771] chan_unicall.c: Unicall/56 event Protocol
failure
[Jul 17 21:46:08] ERROR[4771] chan_unicall.c: Unicall/56 protocol error.
Cause 32772
[Jul 17 21:46:08] WARNING[4771] chan_unicall.c: MFC/R2 UniCall/56
Channel echo cancel
[Jul 17 21:46:09] WARNING[4771] chan_unicall.c: MFC/R2 UniCall/57
<- 0001  [1/   1/Idle  /Idle ]
[Jul 17 21:46:09] WARNING[4771] chan_unicall.c: MFC/R2 UniCall/57
Detected
[Jul 17 21:46:09] WARNING[4771] chan_unicall.c: MFC/R2 UniCall/57 Making
a new call with CRN 32769
[Jul 17 21:46:09] WARNING[4771] chan_unicall.c: MFC/R2 UniCall/57 1101
->  [2/   2/Idle  /Idle ]
[Jul 17 21:46:09] NOTICE[4771] chan_unicall.c: Unicall/57 event Detected
[Jul 17 21:46:09] WARNING[4771] chan_unicall.c: MFC/R2 UniCall/57
<- 9 on  [2/   2/Seize ack /Seize ack]
[Jul 17 21:46:09] WARNING[4771] chan_unicall.c: MFC/R2 UniCall/57 1 on
->  [2/   2/Seize ack /Seize ack]
[Jul 17 21:46:09] WARNING[4771] chan_unicall.c: MFC/R2 UniCall/57
<- 9 off [2/   2/Group A   /DNIS request ]
[Jul 17 21:46:09] WARNING[4771] chan_unicall.c: MFC/R2 UniCall/57 1 off
->  [2/   2/Group A   /DNIS request ]
[Jul 17 21:46:09] WARNING[4771] chan_unicall.c: MFC/R2 UniCall/57
<- 9 on  [2/   2/Group A   /DNIS request ]
[Jul 17 21:46:09] WARNING[4771] chan_unicall.c: MFC/R2 UniCall/57 1 on
->  [2/   2/Group A   /DNIS request ]
[Jul 17 21:46:09] WARNING[4771] chan_unicall.c:

[asterisk-users] Any way to determine remote Asterisk version

[Andrew Joakimsen]

A long time ago (Asterisk 0.x, 1.0.x) my experience is that there were alot
of interoperability issues, a common troubleshooting issue was to make sure
all endpoints where using the latest version of Asterisk. I have not seen
these issues in a while.

However I've been working with a customer of mine and this ITSP called IP
Communications (IPComms.net) well turns out we have had constant problems
since the first day. Turns out (I found this out approx 1 month ago) that
the "version of Asterisk" they are using is "[EMAIL PROTECTED] 1.x"  So
basically is there any way to determine the version of Asterisk being used?
I cant trust anything IP Communications says.
___
--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

Re: [asterisk-users] Asterisk 1.4, Unicall and Nextel...

[Moises Silva]

In order to help you I need testcall traces, with max level of
logging, of incoming Nextel calls.

Regards,

On 7/17/07, Carlos Chavez <[EMAIL PROTECTED]> wrote:
> On Tue, 2007-07-17 at 15:39 -0500, Victor Toofic wrote:
> > El Tue, Jul 17 de 2007 a las 14:39 -0500, Carlos Chavez comentaba:
> > > I have a customer that is complaining that any call coming in from
> > > Nextel gives a fast busy.  We are running Asterisk 1.4.7.1 with Zaptel
> > > 1.4.3 and all the MFC/R2 patches and libraries.  All other calls go out
> > > and come in, just Nextel seems to have this problem.  The phone company
> > > technician connected a PBX emulator on the line and that one could
> > > receive the calls from Nextel.
> >
> > Would you please tell me what version of the libraries are you using, Im
> > trying to get running * with R2 without success.
> >
> > Thanks...
> > Víctor Toofic
> >
> >
> You can download all the necesary libraries for Asterisk 1.4 from:
>
> http://moy.ivsol.net/unicall/soft-switch/r1b1/
>
> Apart from this one problem they work well.
>
> --
> Telecomunicaciones Abiertas de México S.A. de C.V.
> Carlos Chávez Prats
> Director de Tecnología
> +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
>
>


-- 
"Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org";

___
--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

Re: [asterisk-users] Slow list

[Duncan Turnbull]

I thought initially it was a pretty poor generalization about postgrey and our 
capabilities until I realized that this was sent a
few weeks ago when this probably wasn't an as obvious issue. But it clearly is 
an issue now.

I have checked my mail servers for failures, implicitly greylisting is working 
as the mails are coming from digium constantly - just
a long time delayed, if postgrey was an issue there would still be retries and 
there have been none in over a week - as long as my
logs go back, any decent mail server should have retried in much less than a 
week.

Anyway - a discussion and investigation of issues is made pretty hard with 4 
-10 day gaps in it. Since every other list works on
time (+- a few hours) its looking like Digium from my view.

I imagine someone will have sorted it before I see my own post, fingers crossed.

Cheers Duncan 

-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Walt Reed
Sent: Friday, 6 July 2007 6:33 a.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Slow list


On Thu, Jul 05, 2007 at 01:40:50PM -0400, Doug Lytle said:
> >>Well, this is now the third active thread on this subject, but I guess
> >>you won't see this message for a while.  Has anyone dissected the
> >>headers of a delayed message yet?  We should be able to tell for sure
> >>where the holdup is.  All of the messages are coming through on time
> >>for me, so it won't do much good for me to look.
> 
> 
> Looks like mail is getting held up between INXS.digium.internal and 
> lists.digium.com
> 
> INXS.digium.internal received it the first of July, lists.digium.com 
> received it on the 4th.
> 
> drdos.info (ME) received it from lists.digium.com on that same day (Today).

What you can't see without looking at the mail server logs on both ends
is delivery attempts. Greylisting for example can totally hose you over
depending on the implementation. Greylisting without whitelisting is
irresponsible.  How many tries did the digium server make before the
message finally got through??? That's what we need to know. Only Digium
can say.

Before poking Digium too much, I would look at exactly what YOUR mail
servers are doing that may potentially be the real cause of the delays.

___
--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


___
--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

Re: [asterisk-users] Asterisk PRI Busy Problem

[Andrew Joakimsen]

On 7/17/07, Jared Smith <[EMAIL PROTECTED]> wrote:


On Tue, 2007-07-17 at 12:52 -0400, Andrew Joakimsen wrote:
> I did a quick test. What happens is Congestion() answers the channel
> and leaves it open. IE do a 'show channels' and you will see the
> channel is still open on your end.

What happens in you pass a timeout to the Congestion() application, and
then hangup the call after that, as show below?

exten => 4340,15,Queue(test,rt,,,10)
exten => 4340,16,Congestion(3)
exten => 4340,17,Hangup()

Give that a try and see if it helps.

--
Jared Smith
Community Relations Manager
Digium, Inc.



Yes, that seems to solve Arun's problem. When I do Congestion(1) I receive
approx 1 second of congestion tone and then ringing and the SprintPCS
message "We are unable to complete your call at this time" (because I call
into it from a Sprint mobile phone)
___
--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

Re: [asterisk-users] chan_isdn with HFC-compatible

[Ex Vito]

>
> Asterisk is loading the chan_misdn and lists mISDN when issueing "show
> channeltypes" -  however it indicates "Devicestate - No". when I look for
> "misdn show stacks", it lists the single port of the ISDN-card, however
> indicates "L2Link DOWN, L1LinkDOWN". so I  guess theres something wrong,
> unfortunately I've no idea  on what to check next...any pointers? maybe I'm
> missing something obvious, as this is my first installation of Asterisk +
> ISDN...
>

  Checklist:

  - The cable (after all L1 seems to be down)
  - If asterisk is to behave as a phone, check that
the port is configured in TE mode
  - Verify PTP vs PTMP

  If needed:

  - Take a look at /var/log/asterisk/misdn.log
  - Increase "verbosity" in /etc/misdn-init.conf
debug=3 should be a good starting point
(reload mISDN modules after this change)
--
  exvito

___
--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

Re: [asterisk-users] Suppress MusicOnHold in Queue

[Ex Vito]

>
> David L. West wrote:
> > I want callers to go into the queue(s) and just hear ringing instead
> > of MOH. Is this possible?
> >

  ...use option 'r' for the Queue application. For more options,
  use 'show application queue' at the CLI.

  Cheers,
--
  exvito

___
--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

Re: [asterisk-users] Slow list

[Duncan Turnbull]

This message arrived today 18 July NZ time

Full headers below but most of my mail is like this - the offending bit seems 
to be: INXS.digium.internal which took 4 days to
deliver it 

Cheers Duncan

Return-path: <[EMAIL PROTECTED]>
Envelope-to: [EMAIL PROTECTED]
Delivery-date: Wed, 18 Jul 2007 09:33:50 +1200
Received: from pluto.e-simple.co.nz ([203.163.98.172])
by quotient.onesquared.net with esmtp (Exim 4.50)
id 1IAug2-0006xz-17
for [EMAIL PROTECTED]; Wed, 18 Jul 2007 09:33:50 +1200
Received: by pluto.e-simple.co.nz (Postfix, from userid 1014)
id 7C20823927F; Wed, 18 Jul 2007 09:33:48 +1200 (NZST)
X-Spam-Checker-Version: SpamAssassin 3.1.7-deb (2006-10-05) on pluto
X-Spam-Level: 
X-Spam-Status: No, score=-2.2 required=5.0 tests=AWL,BAYES_00,
FORGED_RCVD_HELO autolearn=ham version=3.1.7-deb
Received: from lists.digium.com (lists.digium.com [216.207.245.17])
by pluto.e-simple.co.nz (Postfix) with ESMTP id 2CF0223928A
for <[EMAIL PROTECTED]>; Wed, 18 Jul 2007 09:33:44 +1200 (NZST)
Received: from localhost ([127.0.0.1] helo=INXS.digium.internal)
by lists.digium.com with esmtp (Exim 4.63)
(envelope-from <[EMAIL PROTECTED]>)
id 1I9oPd-0002fh-Rm; Sat, 14 Jul 2007 15:40:21 -0500
Received: from exprod8mx52.postini.com ([64.18.3.152] helo=psmtp.com)
by lists.digium.com with smtp (Exim 4.63)
(envelope-from <[EMAIL PROTECTED]>) id 1I9oPU-0002fV-Ja
for asterisk-users@lists.digium.com; Sat, 14 Jul 2007 15:40:12 -0500
Received: from source ([82.103.132.43]) by exprod8mx52.postini.com
([64.18.7.10]) with SMTP; Sat, 14 Jul 2007 13:40:11 PDT
Received: from localhost (threed001.three-dimensional.net [127.0.0.1])
by three-dimensional.net (Postfix) with ESMTP id 628F651C50
for ;
Sat, 14 Jul 2007 22:40:10 +0200 (CEST)
X-Virus-Scanned: Debian amavisd-new at three-dimensional.net
Received: from three-dimensional.net ([127.0.0.1])
by localhost (mail.three-dimensional.net [127.0.0.1]) (amavisd-new,
port 10024)
with ESMTP id I8DqCNjuf0hk for ;
Sat, 14 Jul 2007 22:40:04 +0200 (CEST)
Received: from lunteren.vanbaak.info (vanbaak.xs4all.nl [82.95.250.75])
by three-dimensional.net (Postfix) with ESMTP id EF00D51C4F
for ;
Sat, 14 Jul 2007 22:40:03 +0200 (CEST)
Received: by lunteren.vanbaak.info (Postfix, from userid 1000)
id 5814A50BB; Sat, 14 Jul 2007 22:39:57 +0200 (CEST)
Date: Sat, 14 Jul 2007 22:39:56 +0200
From: Michiel van Baak <[EMAIL PROTECTED]>
To: asterisk-users@lists.digium.com
Message-ID: <[EMAIL PROTECTED]>
Mail-Followup-To: asterisk-users@lists.digium.com
References: <[EMAIL PROTECTED]>
MIME-Version: 1.0
Content-Disposition: inline
In-Reply-To: <[EMAIL PROTECTED]>
X-PGP-Key: http://michiel.vanbaak.info/Files/pubkey.asc
User-Agent: Mutt/1.5.12-2006-07-14
X-pstn-neptune: 0/0/0.00/0
X-pstn-levels: (S:91.99845/99.9 FC:95.5390 LC:95.5390 R:95.9108 P:95.9108
M:97.0282 C:98.6951 )
X-pstn-settings: 3 (1.:1.) s fc lc gt3 gt2 gt1 r p m c 
X-pstn-addresses: from <[EMAIL PROTECTED]> [db-null] 
Subject: Re: [asterisk-users] Slow list
X-BeenThere: asterisk-users@lists.digium.com
X-Mailman-Version: 2.1.9
Precedence: list
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion 

List-Id: Asterisk Users Mailing List - Non-Commercial Discussion

List-Unsubscribe: ,

List-Archive: 
List-Post: 
List-Help: 
List-Subscribe: ,

Content-Type: text/plain; charset="us-ascii"
Content-Transfer-Encoding: 7bit
Sender: [EMAIL PROTECTED]
Errors-To: [EMAIL PROTECTED]


-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michiel van Baak
Sent: Sunday, 15 July 2007 8:40 a.m.
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Slow list

On 16:28, Thu 05 Jul 07, Philipp Kempgen wrote:
> Since the list was switched over to API-Digital almost
> every message I get is older than a week. Coincidence?
> Is anyone else having trouble?
> 
> Regards,
>   Philipp

I got this message today July 14
Yes, I have the same.
-- 

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

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


___
--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


___
--Bandwidth and Colocation Provided by http://www.api-digital.com--

ast

Re: [asterisk-users] Asterisk 1.4, Unicall and Nextel...

[Carlos Chavez]

On Tue, 2007-07-17 at 15:39 -0500, Victor Toofic wrote:
> El Tue, Jul 17 de 2007 a las 14:39 -0500, Carlos Chavez comentaba:
> > I have a customer that is complaining that any call coming in from
> > Nextel gives a fast busy.  We are running Asterisk 1.4.7.1 with Zaptel
> > 1.4.3 and all the MFC/R2 patches and libraries.  All other calls go out
> > and come in, just Nextel seems to have this problem.  The phone company
> > technician connected a PBX emulator on the line and that one could
> > receive the calls from Nextel.
> 
> Would you please tell me what version of the libraries are you using, Im
> trying to get running * with R2 without success.
> 
> Thanks...
> Víctor Toofic
> 
> 
You can download all the necesary libraries for Asterisk 1.4 from:

http://moy.ivsol.net/unicall/soft-switch/r1b1/

Apart from this one problem they work well.

-- 
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001


signature.asc
Description: This is a digitally signed message part
___
--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

Re: [asterisk-users] USB Modem with asterisk

[Don Kelly]

"WRONG" is the abbreviated answer, right? :)

If Doug is looking for a USB interface that will interface to the PSTN, he
just needs to call it a "channel bank" instead of a modem.

Wouldn't a Xorcom solution work for him? Are there others?

http://www.xorcom.com/products

  --Don

Don Kelly
PCF Corp
Real Support for your Virtual Office
651 842-1000
888 Don Kell(y)
651 842-1001 fax

 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric
"ManxPower" Wieling
Sent: Thursday, July 12, 2007 11:46 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] USB Modem with asterisk

Doug Zingel wrote:
> I can use a USB modem with asterisk to connect to the
> PSTN network right? It'll serve the same functionality
> as an FXO card? Also, any idea if I can get these
> modems with mutiple ports (12 or 24)?

WRONG!

___
--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


___
--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

[asterisk-users] bristuff for hfc card on Xscale 80219

[Thomas Winter]

Hi,

compile and load of modules works fine.

After ztcfg I can see
.
.
Changing signalling on channel 1 from Unused to Clear channel
Changing signalling on channel 2 from Unused to Clear channel
Changing signalling on channel 3 from Unused to HDLC with FCS check

and then the board is frozen.

Any ideas?

regards
Thomas


___
--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

[asterisk-users] ASA-2007-017: Remote crash vulnerability in STUN implementation

[The Asterisk Development Team]

   Asterisk Project Security Advisory - ASA-2007-017

   ++
   |  Product   | Asterisk  |
   |+---|
   |  Summary   | Remote Crash Vulnerability in STUN implementation |
   |+---|
   | Nature of Advisory | Denial of Service |
   |+---|
   |   Susceptibility   | Remote Unauthenticated Sessions   |
   |+---|
   |  Severity  | Critical  |
   |+---|
   |   Exploits Known   | No|
   |+---|
   |Reported On | July 13, 2007 |
   |+---|
   |Reported By | Will Drewry, Google Security Team |
   |+---|
   | Posted On  | July 17, 2007 |
   |+---|
   |  Last Updated On   | July 17, 2007 |
   |+---|
   |  Advisory Contact  | Joshua Colp <[EMAIL PROTECTED]>|
   |+---|
   |  CVE Name  | CVE-2007-3765 |
   ++

   ++
   | Description | The Asterisk STUN implementation in the RTP stack has a  |
   | | remotely exploitable crash vulnerability. A pointer may  |
   | | run past accessible memory if Asterisk receives a|
   | | specially crafted STUN packet on an active RTP port. |
   | |  |
   | | The code that parses the incoming STUN packets   |
   | | incorrectly checks that the length indicated in the STUN |
   | | attribute and the size of the STUN attribute header does |
   | | not exceed the available data. This will cause the data  |
   | | pointer to run past accessible memory and when accessed  |
   | | will cause a crash.  |
   ++

   ++
   | Resolution | All users that have chan_sip, chan_gtalk, chan_jingle,|
   || chan_h323, chan_mgcp, or chan_skinny enabled on an|
   || affected version should upgrade to the appropriate|
   || version listed in the correct in section of this  |
   || advisory. |
   ++

   ++
   |   Affected Versions|
   ||
   | Product  |   Release   |   |
   |  |   Series|   |
   |--+-+---|
   |   Asterisk Open Source   |1.0.x| None affected |
   |--+-+---|
   |   Asterisk Open Source   |1.2.x| None affected |
   |--+-+---|
   |   Asterisk Open Source   |1.4.x| All versions prior to |
   |  | | 1.4.8 |
   |--+-+---|
   |Asterisk Business Edition |A.x.x| None affected |
   |--+-+---|
   |Asterisk Business Edition |B.x.x| None affected |
   |--+-+---|
   |   AsteriskNOW| pre-release | All versions prior to |
   |   

[asterisk-users] ASA-2007-016: Remote crash vulnerability in Skinny channel driver

[The Asterisk Development Team]

   Asterisk Project Security Advisory - ASA-2007-016

   ++
   |  Product   | Asterisk  |
   |+---|
   |  Summary   | Remote crash vulnerability in Skinny channel  |
   || driver|
   |+---|
   | Nature of Advisory | Denial of Service |
   |+---|
   |   Susceptibility   | Remote Unauthenticated Sessions   |
   |+---|
   |  Severity  | Critical  |
   |+---|
   |   Exploits Known   | No|
   |+---|
   |Reported On | July 13, 2007 |
   |+---|
   |Reported By | Will Drewry, Google Security Team |
   |+---|
   | Posted On  | July 17, 2007 |
   |+---|
   |  Last Updated On   | July 17, 2007 |
   |+---|
   |  Advisory Contact  | Jason Parker <[EMAIL PROTECTED]> |
   |+---|
   |  CVE Name  | CVE-2007-3764 |
   ++

   ++
   | Description | The Asterisk Skinny channel driver, chan_skinny, has a   |
   | | remotely exploitable crash vulnerability. A segfault can |
   | | occur when Asterisk receives a packet where the claimed  |
   | | length of the data is between 0 and 3, followed by   |
   | | length + 4 or more bytes, due to an overly large memcpy. |
   | | The side effects of this extremely large memcpy have not |
   | | been investigated.   |
   ++

   ++
   | Resolution | All users that have chan_skinny enabled should upgrade to |
   || the appropriate version listed in the corrected in|
   || section of this advisory. As a workaround, users who do   |
   || not require chan_skinny may add the line "noload =>   |
   || chan_skinny.so" (without quotes) to   |
   || /etc/asterisk/modules.conf, and restart Asterisk. |
   ++

   ++
   |   Affected Versions|
   ||
   | Product  |   Release   |   |
   |  |   Series|   |
   |--+-+---|
   |   Asterisk Open Source   |1.0.x| All versions  |
   |--+-+---|
   |   Asterisk Open Source   |1.2.x| All versions prior to |
   |  | | 1.2.22|
   |--+-+---|
   |   Asterisk Open Source   |1.4.x| All versions prior to |
   |  | | 1.4.8 |
   |--+-+---|
   |Asterisk Business Edition |A.x.x| All versions  |
   |--+-+---|
   |Asterisk Business Edition |B.x.x| All versions prior to |
   |  | | B.2.2.1   |
   |--+-+---|
   |   AsteriskNOW| pre-release | All versions prior to |
   | 

[asterisk-users] ASA-2007-015: Remote Crash Vulnerability in IAX2 channel driver

[The Asterisk Development Team]

   Asterisk Project Security Advisory - ASA-2007-015

   ++
   |  Product   | Asterisk  |
   |+---|
   |  Summary   | Remote Crash Vulnerability in IAX2 channel driver |
   |+---|
   | Nature of Advisory | Denial of Service |
   |+---|
   |   Susceptibility   | Remote Unauthenticated Sessions   |
   |+---|
   |  Severity  | Critical  |
   |+---|
   |   Exploits Known   | No|
   |+---|
   |Reported On | July 13, 2007 |
   |+---|
   |Reported By | Chris Clark and Zane Lackey, iSEC Partners|
   |+---|
   | Posted On  | July 17, 2007 |
   |+---|
   |  Last Updated On   | July 17, 2007 |
   |+---|
   |  Advisory Contact  | Russell Bryant <[EMAIL PROTECTED]>   |
   |+---|
   |  CVE Name  | CVE-2007-3763 |
   ++

   ++
   | Description | The Asterisk IAX2 channel driver, chan_iax2, has a   |
   | | remotely exploitable crash vulnerability. A NULL pointer |
   | | exception can occur when Asterisk receives a LAGRQ or|
   | | LAGRP frame that is part of a valid session and includes |
   | | information elements. The session used to exploit this   |
   | | issue does not have to be authenticated. It can simply   |
   | | be a NEW packet sent with an invalid username.   |
   | |  |
   | | The code that parses the incoming frame correctly parses |
   | | the information elements of IAX frames. It then sets a   |
   | | pointer to NULL to indicate that there is not a raw data |
   | | payload associated with this frame. However, it does not |
   | | set the variable that indicates the number of bytes in   |
   | | the raw payload back to zero. Since the raw data length  |
   | | is non-zero, the code handling LAGRQ and LAGRP frames|
   | | tries to copy data from a NULL pointer, causing a crash. |
   ++

   ++
   | Resolution | All users that have chan_iax2 enabled should upgrade to   |
   || the appropriate version listed in the corrected in|
   || section of this advisory. |
   ++

   ++
   |   Affected Versions|
   ||
   | Product  |   Release   |   |
   |  |   Series|   |
   |--+-+---|
   |   Asterisk Open Source   |1.0.x| All versions  |
   |--+-+---|
   |   Asterisk Open Source   |1.2.x| All versions prior to |
   |  | | 1.2.22|
   |--+-+---|
   |   Asterisk Open Source   |1.4.x| All versions prior to |
   |  | | 1.4.8 |
   |--+-+---|
   |Asterisk Business Edition |A.x.x| All versions  |
   |-

[asterisk-users] ASA-2007-014: Stack buffer overflow in IAX2 channel driver

[The Asterisk Development Team]

   Asterisk Project Security Advisory - ASA-2007-014

   ++
   |   Product| Asterisk|
   |--+-|
   |   Summary| Stack buffer overflow in IAX2 channel driver|
   |--+-|
   |  Nature of Advisory  | Exploitable Stack Buffer Overflow   |
   |--+-|
   |Susceptibility| Remote Unuthenticated Sessions  |
   |--+-|
   |   Severity   | Critical|
   |--+-|
   |Exploits Known| No  |
   |--+-|
   | Reported On  | July 12, 2007   |
   |--+-|
   | Reported By  | Russell Bryant, Digium, Inc.|
   |--+-|
   |  Posted On   | July 17, 2007   |
   |--+-|
   |   Last Updated On| July 17, 2007   |
   |--+-|
   |   Advisory Contact   | Russell Bryant <[EMAIL PROTECTED]> |
   |--+-|
   |   CVE Name   | CVE-2007-3762   |
   ++

   ++
   | Description | The Asterisk IAX2 channel driver, chan_iax2, has a   |
   | | remotely exploitable stack buffer overflow   |
   | | vulnerability. It occurs when chan_iax2 is passed a  |
   | | voice or video frame with a data payload larger than 4   |
   | | kB. This is exploitable by sending a very large RTP  |
   | | frame to an active RTP port number used by Asterisk when |
   | | the other end of the call is an IAX2 channel. Exploiting |
   | | this issue can cause a crash or allow arbitrary code |
   | | execution on a remote machine.   |
   | |  |
   | | The specific conditions that trigger the vulnerability   |
   | | are the following:   |
   | |  |
   | |   * iax2_write() is called with a frame with the |
   | | following properties |
   | |  |
   | |* a voice or video frame  |
   | |  |
   | |* Its 4-byte timestamp has the same high 2 bytes  |
   | |  as the previous frame that was sent |
   | |  |
   | |* Its format is the one currently expected|
   | |  |
   | |* Its data payload is larger than 4 kB|
   | |  |
   | | iax2_write() calls iax2_send() to send the frame. Inside |
   | | of iax2_send(), there is a conditional check to  |
   | | determine whether the frame should be sent immediately   |
   | | (the now variable) or queued for transmission later. |
   | |  |
   | | If the frame is going to be transmitted later, an|
   | | iax_frame struct is dynamically allocated with a data|
   | | buffer that has the exact buffer size needed to  |
   | | accommodate for the provided ast_frame data. However, if |
   | | the frame is being sent immediately, it uses a stack |
   | | allocated iax_frame, with a data buffer size of 4096 |
   | | bytes.   |
   | | 

[asterisk-users] Critical Updates: Asterisk 1.2.22 and 1.4.8 released

[The Asterisk Development Team]

The Asterisk development team has released Asterisk versions 1.2.22 and
1.4.8.

These releases contain fixes for four critical security vulnerabilities.
 One of these vulnerabilities is a remotely exploitable stack buffer
overflow, which could allow an attacker to execute arbitrary code on the
target machine.  The other three are all remotely exploitable crash
vulnerabilities.

We have released Asterisk Security Advisories for each of the
vulnerabilities.  The current version of each advisory can be downloaded
from the ftp site.

http://ftp.digium.com/pub/asa/ASA-2007-014.pdf
 * Affected systems include those that bridge calls between chan_iax2
and any channel driver that uses RTP for media

http://ftp.digium.com/pub/asa/ASA-2007-015.pdf
 * Affected systems include any system that has chan_iax2 enabled

http://ftp.digium.com/pub/asa/ASA-2007-016.pdf
 * Affected systems include any system that has chan_skinny enabled

http://ftp.digium.com/pub/asa/ASA-2007-017.pdf
 * Affected systems include any 1.4 system that has any channel driver
that uses RTP for media enabled

All users that have systems that meet any of the criteria listed above
should upgrade as soon as possible.

Thank you very much for your support.

___
--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

[asterisk-users] No sound from Festival, but *something* is happening

[Martin Smith]

Hey folks,

So I'm trying to get Festival() working on 1.2.17. I'm trying to use
app_festival:

Here's the show dialplan output from that extension:

  '3378' => 1. Answer()
[pbx_config]
2. Festival(Hello Asterisk caller. How is your day?)
[pbx_config]
3. Playback(vm-goodbye)
[pbx_config]
4. Hangup()
[pbx_config]

In the Festival server logs, I actually see:

client(1) Tue Jul 17 16:38:32 2007 : accepted from localhost
client(1) Tue Jul 17 16:38:32 2007 : disconnected

But on the channel in question, I hear "vm-goodbye" and it hangs up.
I've turned on the caching option in /etc/asterisk/festival.conf, and
then looked in the cache directory, and files *are* appearing there.

I'm using the default command:
festivalcommand=(tts_textasterisk "%s" 'file)(quit)\n

Even the verbose output shows it working:
-- Executing Answer("Zap/97-1", "") in new stack
-- Executing Festival("Zap/97-1", "Hello Asterisk caller. How is
your day?") in new stack
  == Parsing '/etc/asterisk/festival.conf': Found
-- Executing Playback("Zap/97-1", "vm-goodbye") in new stack
-- Playing 'vm-goodbye' (language 'en')
 == Spawn extension (default, 3378, 4) exited non-zero on 'Zap/97-1'
-- Executing Hangup("Zap/97-1", "") in new stack
  == Spawn extension (default, h, 1) exited non-zero on 'Zap/97-1'
-- Hungup 'Zap/97-1'


Any ideas as to why I can't hear anything? Thanks!

Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic and Business Research
University of Florida
(352) 392-0171 Ext. 221 


___
--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

Re: [asterisk-users] Asterisk 1.4, Unicall and Nextel...

[Victor Toofic]

El Tue, Jul 17 de 2007 a las 14:39 -0500, Carlos Chavez comentaba:
>   I have a customer that is complaining that any call coming in from
> Nextel gives a fast busy.  We are running Asterisk 1.4.7.1 with Zaptel
> 1.4.3 and all the MFC/R2 patches and libraries.  All other calls go out
> and come in, just Nextel seems to have this problem.  The phone company
> technician connected a PBX emulator on the line and that one could
> receive the calls from Nextel.

Would you please tell me what version of the libraries are you using, Im
trying to get running * with R2 without success.

Thanks...
Víctor Toofic


___
--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

Re: [asterisk-users] 1.4.7 chan_alsa : snd_pcm_open failed

[Jakub Głazik]

Dnia 2007-07-15, o godz. 14:49:27
sean <[EMAIL PROTECTED]> napisał(a):

> asterisk-1.4.7, Fedora 7, intel emt64 - nocona:

modprobe snd-pcm-oss

-- 
.: Jakub Głazik,
.: email & jabber: zyteknuxi.pl

___
--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

Re: [asterisk-users] Suppress MusicOnHold in Queue

[Leif Neland]

(catching up while my adsl is offline)

David L. West wrote:
> I want callers to go into the queue(s) and just hear ringing instead
> of MOH. Is this possible?
>
If everything else fails, you can generate a file with ringing tones, and 
use that for moh.

Leif



___
--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

Re: [asterisk-users] Single ringer phone for incoming calls, that anyone can answer

[Leif Neland]

(While my adsl is down, I'm reading old posts.)
Tom Lanyon wrote:
> Hi list,
>
> Does anyone have any advice on the following:
>
> Incoming calls to our office come in on a SIP trunk. Since all our
> offices/desks are in close proximity, we would like just a single
> phone to ring when a call comes in instead of ringing every person's
> phone.
>
> Currently we've got this working by having all the phones in a
> callgroup/pickupgroup and incoming calls ring the 'ringer phone'
> extension, then we can use the *8 to pickup the incoming call from
> any other phone. The problem though, is that if two people in the
> office call each other, *8 from a third phone also picks up their
> call, which is not the desired effect.

Use the application Pickup
exten => 88,1,Pickup(SIP/singleringerphone)

  -= Info about application 'Pickup' =-

[Synopsis]
Directed Call Pickup

[Description]
  Pickup([EMAIL PROTECTED]&[EMAIL PROTECTED]): This application can 
pickup any ringing channel
that is calling the specified extension. If no context is specified, the 
current
context will be used. If you use the special string "PICKUPMARK" for the 
context parameter, for example
[EMAIL PROTECTED], this application tries to find a channel which has defined a 
channel variable with the same content
as "extension".

Leif


>
> So in essence, I'm asking whether there's a better way to pickup an
> incoming call from our external SIP trunk, whilst its ringing only a
> specific extension, without picking up overlapping internal calls?
>
>
> Regards,
> Tom
>
> ___
> --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 


___
--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

[asterisk-users] Asterisk 1.4, Unicall and Nextel...

[Carlos Chavez]

I have a customer that is complaining that any call coming in from
Nextel gives a fast busy.  We are running Asterisk 1.4.7.1 with Zaptel
1.4.3 and all the MFC/R2 patches and libraries.  All other calls go out
and come in, just Nextel seems to have this problem.  The phone company
technician connected a PBX emulator on the line and that one could
receive the calls from Nextel.

The E1 is provided by Avantel in Mexico City, there are 10 channels
available at the moment.  I have already increased the T1 parameter to
2 in mfcr2.c (libmfcr2).  Maybe I need to increase it again?  The
strange thing is that I can dial any Nextel phone without problems.

-- 
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001


signature.asc
Description: This is a digitally signed message part
___
--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

Re: [asterisk-users] Can Asterisk hear on two IP addresses? And can Ido routing for calls from private to public or public toprivate IP addresses

[Idris AVCI]

In general section of sip.conf you can bind sip service to multiple ip
addresses. If you setup routing successfully you can send the call
received one of ip address through other ip addresses of asterisk. All
you have to do is to setup routing the right way. In this conf asterisk
can be used both for signaling and media.

-Original Message-
From: bilal ghayyad [mailto:[EMAIL PROTECTED] 
Sent: Friday, July 13, 2007 7:52 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Can Asterisk hear on two IP addresses? And can
Ido routing for calls from private to public or public toprivate IP
addresses

Hi List;

Can asterisk hear (receive) calls on two IP addresses?
How?

If yes, then:

If I have a VPN router, and my Asterisk server
connected to two network cards, one has a private IP
address (192.168.0.2) connected to the VPN router
(192.168.0.1) and another network card has a private
IP address (193.111.196.249) connected directly to the
outside default gateway (193.111.196.240), where the
VPN default gateway for outside is also
(193.111.196.240), then:

If I received a call on the network card of IP:
192.168.0.2 then can I route the call for another
softswitch server has a public IP address (in another
county and another network)? If yes, then is there
some condition on this kind of call routing (for
example: the communication mode to be full proxy for
media and signaling or it can be a proxy only for
signaling)?

Any help?

Regards
---
IP Telephony and Contact Center Engineer
Eng. Bilal Ghayad
Mobile: 0965 9849460


 


Be a PS3 game guru.
Get your game face on with the latest PS3 news and previews at Yahoo!
Games.
http://videogames.yahoo.com/platform?platform=120121

___
--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



___
--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

Re: [asterisk-users] Asterisk PRI Busy Problem

[Jared Smith]

On Tue, 2007-07-17 at 12:52 -0400, Andrew Joakimsen wrote:
> I did a quick test. What happens is Congestion() answers the channel
> and leaves it open. IE do a 'show channels' and you will see the
> channel is still open on your end.

What happens in you pass a timeout to the Congestion() application, and
then hangup the call after that, as show below?

exten => 4340,15,Queue(test,rt,,,10) 
exten => 4340,16,Congestion(3)
exten => 4340,17,Hangup()

Give that a try and see if it helps.

-- 
Jared Smith
Community Relations Manager
Digium, Inc.

___
--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

[asterisk-users] Asterisk 1.4.6 crash using queue app

[equis software]

I'm using Queue app with Asterisk 1.4.6

It was working 5 days without problems and then it crash.
When I did #gdb asterisk core.xxx
I see...
#0  ast_senddigit_end (chan=0x0, digit=54 '6', duration=0) at channel.c:2691
#1  0xb780c7d5 in agent_answer (ast=0x925cb78) at chan_agent.c:398
#2  0x08081877 in ast_senddigit (chan=0x57, digit=54 '6') at channel.c:2692
#3  0x0805d20d in ast_dtmf_stream (chan=0x925cb78, peer=0xb6a2d0e0,
   digits=0xb6940bf4 "6", between=100) at app.c:243
#4  0xb79086ca in ast_bridge_call (chan=0xb6a2d0e0, peer=0x925cb78,
   config=0xb69424d4) at res_features.c:1473
#5  0xb6e011ae in try_calling (qe=0xb6942784,
   options=0x925cb78
"@`\201·\2205&\bùN#\t\003O#\t(r)¬\023\b(r)¬\023\b(r)¬\023\bèN#\tØN#\t\200",
announceoverride=0xb694270e "", url=0xb694270d "",
   go_on=0xb6942770, agi=0x0) at app_queue.c:2651
#6  0xb6df9fda in queue_exec (chan=0xb6a2d0e0, data=0xb694270e) at lock.h
:532
#7  0x080c315b in pbx_extension_helper (c=0xb6a2d0e0, con=0x0,
   context=0xb6a2d260 "telpin-112", exten=0xb6a2d2b0 "80", priority=10,
   label=0x0, callerid=0xb6a22e08 "402385", action=E_MATCHMORE) at pbx.c
:532
#8  0x080c3e21 in __ast_pbx_run (c=0xb6a2d0e0) at pbx.c:2288
#9  0x080c4cc1 in pbx_thread (data=0x0) at pbx.c:2601
#10 0x080f3679 in dummy_start (data=0x0) at utils.c:545
#11 0xb7e9413d in pthread_join () from /lib/libpthread.so.0
#12 0xb7cfa1ba in profil () from /lib/libc.so.6

Any ideas??
___
--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

Re: [asterisk-users] media not accpetable with outgoing call on cisco

[laurent schweizer]

I have already setup a list of prefered codec , but it's only for incoming
call, not outgoing

Laurent


2007/7/17, Alex Balashov <[EMAIL PROTECTED]>:


Laurent,

  You should be able to set it with the 'codec' subcommand on the outgoing
dial peer as well.  'codec g711ulaw' or similar.

-- Alex

On Tue, 17 Jul 2007, laurent schweizer wrote:

> Hello,
>
> I have a problem with a cisco GW, if i only set g711 ulaw or alow as
codec
> in my ata the the GW return a media not acceptable error.
>
> but If i add the g729 codec the all is ok.
> I see in the config of the cisco where to define codec for imcoming call
but
> not for outgoing
>
> *Jul 17 15:57:02.604: Received:
> INVITE sip:[EMAIL PROTECTED] SIP/2.0
> Via: SIP/2.0/UDP 192.168.0.107:5070;branch=z9hG4bK5f66.fc82e301.0
> To: 
> From: 02111 > ;tag=27B98752-469CEA8A0002F2E4-5F903B30
> CSeq: 10 INVITE
> Call-ID: [EMAIL PROTECTED]
> Content-Length: 250
> User-Agent: OpenSER (1.2.1-notls (i386/linux))
> Contact: 
> P-MsgFlags: 0
> billingid: 106
> accountid: 28928
> Remote-Party-ID: > ;party=calling;id-type=subscriber;screen=yes
> Content-Type: application/sdp
>
> v=0
> o=MxSIP 0 198 IN IP4 192.168.0.249
> s=SIP Call
> c=IN IP4 200.200.100.106
> t=0 0
> m=audio 39318 RTP/AVP 8 0 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=sendrecv
> a=direction:active
> a=nortpproxy:yes
>
> *Jul 17 15:57:02.608: 0x64C01D00 : State change from (STATE_NONE,
> SUBSTATE_NONE)  to (STATE_IDLE, SUBSTATE_NONE)
> *Jul 17 15:57:02.608: sipSPIStreamTypeAndDtmfRelay: ERROR - no voice
codec
> and no dtmf-relay match
> *Jul 17 15:57:02.608: sipSPIDoAudioNegotiation: Media negotiation failed
for
> m-line 1
>
> *Jul 17 15:57:02.608: sipSPIDoMediaNegotiation: ERROR - no valid fax or
> audio streams
> *Jul 17 15:57:02.608: sipSPIHandleInviteMedia: Media Negotiation failed
for
> an incoming call - Sending 488
>
> *Jul 17 15:57:02.608: 0x64C01D00 : State change from (STATE_IDLE,
> SUBSTATE_NONE)  to (STATE_DISCONNECTING, SUBSTATE_NONE)
> *Jul 17 15:57:02.608: Sent:
> SIP/2.0 488 Not Acceptable Media
> Via: SIP/2.0/UDP 192.168.0.107:5070;branch=z9hG4bK5f66.fc82e301.0
> From: 02111 > ;tag=27B98752-469CEA8A0002F2E4-5F903B30
> To: ;tag=C0E57710-2347
> Date: Tue, 17 Jul 2007 15:57:02 GMT
> Call-ID: [EMAIL PROTECTED]
> Server: Cisco-SIPGateway/IOS-12.x
> CSeq: 10 INVITE
> Allow-Events: telephone-event
> Content-Length: 0
>

--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: +1-678-954-0670
Direct : +1-678-954-0671

___
--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

___
--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

Re: [asterisk-users] Cisco 7940 log on/off

[James FitzGibbon]

On 7/16/07, Adrian Marsh <[EMAIL PROTECTED]> wrote:


Anyone know if theres a way to share a Cisco 7940 between hot-desk
users?

My phones get their setup via SIP .cnf files, that load at boot via
tftp, so I'm assuming the configs a failry static. However if I want a
phone to be hot-desked, I could have different users sitting there.  Is
there any concept of "logging on" in these environments?



You can set up single-person queues that you can use like extensions.  The
deskphone itself would still have to be addressable as "SIP/something", and
the outbound context will be the same regardless of who is logged in (unless
you want to get fancy and store who is logged in in ASTdb then fork to the
proper outbound context using Goto in the shared context for the peer), but
it works.

Here's my queues.conf for a single-person queue:

[queuename]
strategy=rrmemory
servicelevel=60
timeout=20
retry=5
wrapuptime=0
maxlen=1
announce-frequency=0
announce-holdtime=no
joinempty=strict
leavewhenempty=strict
reportholdtime=no
monitor-format=wav
ringinuse=no

And the dialplan to enqueue:

--START--
[somecontext]
exten => ###,n,Macro(singlequeue,queuename,###)

[macro-singlequeue]
; No-op to give us a 1 priority
exten   => s,1,NoOp

; if DIAL_ANNOUNCE is set, play the "please hold on while I..." msg
exten   => s,n,GotoIf($["${DIAL_ANNOUNCE}" != "1"]?check_clid_set)
exten   => s,n,Playback(transfer)

; Set CLID Name if specified
exten   => s,n(check_clid_set),GotoIf(${ISNULL(${CLID_NAME})}?enqueue)
exten   => s,n,Set(CALLERID(name)=${CLID_NAME})

; Enqueue the caller
exten   => s,n(enqueue),Queue(${ARG1}|nrtW|||20)

; Jump based on status - TIMEOUT | FULL | JOINEMPTY | LEAVEEMPTY |
JOINUNAVAIL |
LEAVEUNAVAIL
exten   => s,n(branch),Goto(s-${QUEUESTATUS},1)

; If unavailable, send to voicemail w/ unavail announce
exten   => s-TIMEOUT,1(vmu),Voicemail(${ARG2}|uj)
exten   => s-TIMEOUT,n,Hangup

exten   => s-TIMEOUT,vmu+101,Playback(vm-theperson)
exten   => s-TIMEOUT,n,SayDigits(${ARG2})
exten   => s-TIMEOUT,n,Playback(vm-isunavail)
exten   => s-TIMEOUT,n,Hangup

; If busy, send to voicemail w/ busy announce
exten   => s-FULL,1(vmb),Voicemail(${ARG2}|bj)
exten   => s-FULL,n,Hangup

exten   => s-FULL,vmb+101,Playback(vm-theperson)
exten   => s-FULL,n,SayDigits(${ARG2})
exten   => s-FULL,n,Playback(vm-isunavail)
exten   => s-FULL,n,Hangup

; Treat the empty, unavail as full
exten   => s-JOINEMPTY,1,Goto(s-FULL,1)
exten   => s-LEAVEEMPTY,1,Goto(s-FULL,1)
exten   => s-JOINUNAVAIL,1,Goto(s-FULL,1)
exten   => s-LEAVEUNAVAIL,1,Goto(s-FULL,1)

; Treat anything else as timeout
exten   => _s-.,1,Goto(s-NOANSWER,1)

; if people star out of voicemail, send them to the top-level admin IVR
exten   => a,1,GotoIf(${ISNULL(${IVR_CONTEXT})}?setivrcontext:gotoivr)
exten   => a,n(setivrcontext),Set(IVR_CONTEXT=ivr-admin)
exten   => a,n(gotoivr),Goto(${IVR_CONTEXT},s,1)

; If they hit 0 from VoiceMail to dial the operator,
; after 20 seconds they go back to the top of the named context
exten   => o,1,Macro(operator)

exten   => h,1,Macro(loghangupcause)
---END---

There's some extra cruft in there you could easily cut out to suit your
environment.  Using that macro in the same way I would normally use "Dial",
the experience for both the caller and agent is pretty much the same.   You
need to have 1.4.7 so that if the agent hits "ignore" or "reject" the call
immediately exits Queue() - there was a bug surrounding the 'n' option in
1.4.6 and earlier.  If you don't have that version, then hitting ignore or
reject will send the call back to the phone after 'retry' seconds until the
timeout runs out (in my case 20 seconds).

All you then need is some dialplan sugar to invoke AddQueueMember and
RemoveQueueMember in response to logon and logoff actions by the agent.  If
you're not worried about authenticating the users, you can just do something
like:

exten =>
_7000XXX,1,AddQueueMember(singlequeue_${EXTEN:4},${CUT(CHANNEL,,1)},,,${CUT(CHANNEL,,1)})
exten =>
_7001XXX,1,RemoveQueueMember(singlequeue_${EXTEN:4},${CUT(CHANNEL,,1)},,,${CUT(CHANNEL,,1)})

Then agent 123 can log in by dialing extension 7000123 and log off by
dialing extension 7001123.  If you attempt to enqueue the caller to the
queue when there's nobody logged in, then you'll immediately get kicked to
your dialplan with ${QUEUESTATUS} set appropriately.  This may or may not be
what you want, because it means that callers will go directly to voicemail
if the person they're calling isn't logged in.

HTH

--
j.
___
--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

Re: [asterisk-users] Asterisk PRI Busy Problem

[Andrew Joakimsen]

I did a quick test. What happens is Congestion() answers the channel and
leaves it open. IE do a 'show channels' and you will see the channel is
still open on your end.

Sorry I don't have further suggestions.

On 7/17/07, Arun Kumar <[EMAIL PROTECTED]> wrote:


Hi,

I've an PRI coming to my asterisk ,calls are coming fine and my agents are
able to answer no prob. but I've an agreement with my telco with some
incoming no if the no of calls on these no are more then 3 then send to
another no. they use busy signal to divert call on another number so I'm
sending the call to Congestion() if no of calls in this group are more then
3. But my provider says he is not getting any busy signal from my side and
he says for all incoming numbers (30) he is getting back only one number
from asterisk box(4340).

here is my dial plan for one incoming DID:

exten => 4340,1,GotoIfTime(*|*|25|dec?ccagents,4340,6)
exten => 4340,2,GotoIfTime(10:00-16:00|*|26|dec?ccagents,4340,7)
exten => 4340,3,GotoIfTime(09:00-18:00|*|31|dec?ccagents,4340,7)
exten => 4340,4,GotoIfTime(12:00-16:00|*|1|jan?ccagents,4340,7)
exten => 4340,5,GotoIfTime(09:00-20:00|mon-sun,*,*?ccagents,4340,7)
exten => 4340,6,Goto(out-of-hours,5001,1)
exten => 4340,7,Set(GROUP(${EXTEN})=MAX_CALLS)
exten => 4340,8,GotoIf($[${GROUP_COUNT([EMAIL PROTECTED])} > 3]?16)
exten => 4340,9,Set(GROUP(${CALLERIDNUM})=MAX_CALLS)
exten => 4340,10,Answer()
exten => 4340,11,Playback(custom/next-avail-advisor)
exten =>
4340,12,Set(MONITOR_FILENAME=/var/spool/asterisk/q/tcarehwsupport-${TIMESTAMP}-${UNIQUEID})
exten => 4340,13,Monitor(wav,${MONITOR_FILENAME},mb)
exten => 4340,14,NoOp(${QUEUESTATUS})
exten => 4340,15,Queue(test,rt,,,10)
exten => 4340,16,Congestion()

zapata.conf:
---
[trunkgroups]

[channels]
language=en
context=ccagents
switchtype=euroisdn
pridialplan=unknown
overlapdial=yes
signalling=pri_cpe
rxgain = 0.0
txgain = 0.0
usecallerid=yes
hidecallerid=yes
callerid=asreceived
callwaiting=yes
usecallingpres=yes
echocancel=yes
echocancelwhenbridged=yes
immediate=no
cidsignalling=v23
callwaitingcallerid=yes
priindication = outofband

resetinterval = 

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

group = 2
channel => 32-46
channel => 48-62

thanks
arun

___
--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

___
--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

Re: [asterisk-users] 2 PRI on asterisk

[Andrew Joakimsen]

On 7/17/07, satish patel <[EMAIL PROTECTED]> wrote:


Dear all

I am going to install 2 port pri card on asterisk but i
dont know how to incomming call goes in to IVR and how to route call outside
base on pattern match means if some one call on mobile phone then use PRI 1
and if call on landline phon call route through pri 2 how to make dission
base on pattern number



Hi Satish:

I know it can be very frustrating, why don't you read this book called
Asterisk: The Future of Telephony. I think what you should focus on is
researching the "dialplan" and "extensions.conf" which essentially are one
and the same. I promise this will answer all your questions.

You can download the book here:
http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11

Regards,

-Andy
___
--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

Re: [asterisk-users] media not accpetable with outgoing call on cisco

[Alex Balashov]

Laurent,

   You should be able to set it with the 'codec' subcommand on the outgoing 
dial peer as well.  'codec g711ulaw' or similar.

-- Alex

On Tue, 17 Jul 2007, laurent schweizer wrote:

> Hello,
>
> I have a problem with a cisco GW, if i only set g711 ulaw or alow as codec
> in my ata the the GW return a media not acceptable error.
>
> but If i add the g729 codec the all is ok.
> I see in the config of the cisco where to define codec for imcoming call but
> not for outgoing
>
> *Jul 17 15:57:02.604: Received:
> INVITE sip:[EMAIL PROTECTED] SIP/2.0
> Via: SIP/2.0/UDP 192.168.0.107:5070;branch=z9hG4bK5f66.fc82e301.0
> To: 
> From: 02111 > ;tag=27B98752-469CEA8A0002F2E4-5F903B30
> CSeq: 10 INVITE
> Call-ID: [EMAIL PROTECTED]
> Content-Length: 250
> User-Agent: OpenSER (1.2.1-notls (i386/linux))
> Contact: 
> P-MsgFlags: 0
> billingid: 106
> accountid: 28928
> Remote-Party-ID: > ;party=calling;id-type=subscriber;screen=yes
> Content-Type: application/sdp
>
> v=0
> o=MxSIP 0 198 IN IP4 192.168.0.249
> s=SIP Call
> c=IN IP4 200.200.100.106
> t=0 0
> m=audio 39318 RTP/AVP 8 0 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=sendrecv
> a=direction:active
> a=nortpproxy:yes
>
> *Jul 17 15:57:02.608: 0x64C01D00 : State change from (STATE_NONE,
> SUBSTATE_NONE)  to (STATE_IDLE, SUBSTATE_NONE)
> *Jul 17 15:57:02.608: sipSPIStreamTypeAndDtmfRelay: ERROR - no voice codec
> and no dtmf-relay match
> *Jul 17 15:57:02.608: sipSPIDoAudioNegotiation: Media negotiation failed for
> m-line 1
>
> *Jul 17 15:57:02.608: sipSPIDoMediaNegotiation: ERROR - no valid fax or
> audio streams
> *Jul 17 15:57:02.608: sipSPIHandleInviteMedia: Media Negotiation failed for
> an incoming call - Sending 488
>
> *Jul 17 15:57:02.608: 0x64C01D00 : State change from (STATE_IDLE,
> SUBSTATE_NONE)  to (STATE_DISCONNECTING, SUBSTATE_NONE)
> *Jul 17 15:57:02.608: Sent:
> SIP/2.0 488 Not Acceptable Media
> Via: SIP/2.0/UDP 192.168.0.107:5070;branch=z9hG4bK5f66.fc82e301.0
> From: 02111 > ;tag=27B98752-469CEA8A0002F2E4-5F903B30
> To: ;tag=C0E57710-2347
> Date: Tue, 17 Jul 2007 15:57:02 GMT
> Call-ID: [EMAIL PROTECTED]
> Server: Cisco-SIPGateway/IOS-12.x
> CSeq: 10 INVITE
> Allow-Events: telephone-event
> Content-Length: 0
>

--
Alex Balashov
Evariste Systems
Web: http://www.evaristesys.com/
Tel: +1-678-954-0670
Direct : +1-678-954-0671

___
--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

[asterisk-users] media not accpetable with outgoing call on cisco

[laurent schweizer]

Hello,

I have a problem with a cisco GW, if i only set g711 ulaw or alow as codec
in my ata the the GW return a media not acceptable error.

but If i add the g729 codec the all is ok.
I see in the config of the cisco where to define codec for imcoming call but
not for outgoing

*Jul 17 15:57:02.604: Received:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.107:5070;branch=z9hG4bK5f66.fc82e301.0
To: 
From: 02111 
;tag=27B98752-469CEA8A0002F2E4-5F903B30

CSeq: 10 INVITE
Call-ID: [EMAIL PROTECTED]
Content-Length: 250
User-Agent: OpenSER (1.2.1-notls (i386/linux))
Contact: 
P-MsgFlags: 0
billingid: 106
accountid: 28928
Remote-Party-ID: 
;party=calling;id-type=subscriber;screen=yes

Content-Type: application/sdp

v=0
o=MxSIP 0 198 IN IP4 192.168.0.249
s=SIP Call
c=IN IP4 200.200.100.106
t=0 0
m=audio 39318 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=direction:active
a=nortpproxy:yes

*Jul 17 15:57:02.608: 0x64C01D00 : State change from (STATE_NONE,
SUBSTATE_NONE)  to (STATE_IDLE, SUBSTATE_NONE)
*Jul 17 15:57:02.608: sipSPIStreamTypeAndDtmfRelay: ERROR - no voice codec
and no dtmf-relay match
*Jul 17 15:57:02.608: sipSPIDoAudioNegotiation: Media negotiation failed for
m-line 1

*Jul 17 15:57:02.608: sipSPIDoMediaNegotiation: ERROR - no valid fax or
audio streams
*Jul 17 15:57:02.608: sipSPIHandleInviteMedia: Media Negotiation failed for
an incoming call - Sending 488

*Jul 17 15:57:02.608: 0x64C01D00 : State change from (STATE_IDLE,
SUBSTATE_NONE)  to (STATE_DISCONNECTING, SUBSTATE_NONE)
*Jul 17 15:57:02.608: Sent:
SIP/2.0 488 Not Acceptable Media
Via: SIP/2.0/UDP 192.168.0.107:5070;branch=z9hG4bK5f66.fc82e301.0
From: 02111 
;tag=27B98752-469CEA8A0002F2E4-5F903B30

To: ;tag=C0E57710-2347
Date: Tue, 17 Jul 2007 15:57:02 GMT
Call-ID: [EMAIL PROTECTED]
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 10 INVITE
Allow-Events: telephone-event
Content-Length: 0
___
--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

Re: [asterisk-users] 2 PRI on asterisk

[Noah Miller]

Hi Satish -

> I am going to install 2 port pri card on asterisk but i dont
> know how to incomming call goes in to IVR and how to route call outside base
> on pattern match means if some one call on mobile phone then use PRI 1 and
> if call on landline phon call route through pri 2 how to make dission base
> on pattern number

Will your PRIs have DID (Direct Inward Dialing)?  If so, you can
direct calls based on the DID number.  For example, if you have the
PRI in the incoming context, and your DID numbers are in the 1000-1099
range, you can use extensions like this:

[incoming]
exten => _10XX,1,Playback(IVR-Start)

or use individual numbers like this:

exten => 1023,1,Dial(SIP/23,20,t)

TIPS: You can usually tell your provider how many digits you want to
receive for your DIDs.  In the North American Numbering Plan, you can
usually ask for 4 digits (as above), 7 digits, or 10 digits.

If your PRIs don't have DID, you can direct your calls just like you
would for a Zaptel analog line.  Again, if you have your PRI in the
incoming context:

[incoming]
exten => s,1,Playback(IVR-start)


If you dont' have DID, and want to differentiate between the two PRIs,
you can put each in its own context.  So, PRI-A is in the [incoming-A]
context and PRI-B is in the [incoming-B] context.  You can then assign
different actions to the 's' extension in each context:

[incoming-A]
exten => s,1,Playback(IVR-Start)

[incoming-B]
exten => s,1,Dial(SIP/100)


- Noah

___
--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

Re: [asterisk-users] Slow list

[Anthony Francis]

Philipp Kempgen wrote:
> Anthony Francis wrote:
>   
>> Doug Lytle wrote:
>> 
> Before poking Digium too much, I would look at exactly what YOUR mail
> servers are doing that may potentially be the real cause of the delays.
>   
>
>   
>> You have two servers in your MX records.
>> drdos.info. 60  IN  MX  10 smtp.drdos.info.
>> drdos.info. 60  IN  MX  5 drdos.info.
>>
>> The one weighted 10 refuses smtp connections, while it is the higher 
>> weight it is the first one that was listed when I did the dig. SO there 
>> stand s a large probability that the list server is trying that address 
>> first.
>> 
>
> It's the other way round.
> http://en.wikipedia.org/wiki/MX_record#MX_priority :
> "The MX record with the smallest preference number has the
> highest priority and is the first server to be tried."
>
>
> Regards,
>   Philipp Kempgen
>
>   
Thank you! I was actually aware of that however, I had said it has the 
higher weight meaning it will be used last, but you see not all email 
systems are compliant with the rfc's and sometimes just use the first MX 
record that comes up assuming the MX with the (highest priority/lowest 
weight) will be listed first.

Thank you for your link though, I appreciate your taking the time to try 
to educate fellow IT professionals.

Anthony

___
--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

Re: [asterisk-users] Zap channels unavailable?

[jan.sarin]

Okay, I've got an update on the resetinterval=never... same thing even though i 
added the line to zapata.conf and restarted the server. 

Now the load wasn't even high, maybe 6-7 calls. I think I just might call my 
telco, feels like it's their issue, but if anyone has any other suggestions let 
me know and I'll try them!

Channel: 7
File Descriptor: 17
Span: 1
Extension:
Dialing: no
Context: from-pstn
Caller ID: 708307496
Calling TON: 33
Caller ID name:
Destroy: 0
InAlarm: 0
Signalling Type: PRI Signalling
Radio: 0
Owner: 
Real: 
Callwait: 
Threeway: 
Confno: -1
Propagated Conference: -1
Real in conference: 0
DSP: no
Relax DTMF: no
Dialing/CallwaitCAS: 0/0
Default law: alaw
Fax Handled: no
Pulse phone: no
Echo Cancellation: 128 taps unless TDM bridged, currently OFF
PRI Flags: Resetting
PRI Logical Span: Implicit
Hookstate (FXS only): Onhook

Regards,
Jan

-Ursprungligt meddelande-
Från: Jan Sarin 
Skickat: den 17 juli 2007 16:57
Till: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Ämne: SV: [asterisk-users] Zap channels unavailable?

Hi,

No I havn't tried that. That entry wasn't even in there so I'll try it. I'll 
let you know if it helped. 

The odd thing is that this problem started yesterday. And our asterisk has been 
running for +1 year without these kind of problems.

So either our telco has changed something OR it's because of the "heavy" load 
on the server (cpu running at max 20% with 40-50 simultaneous calls, so why 
would it be this?).

Regards,
Jan

--

Have you tried setting resetinterval=never in zapata.conf?

On Tue, 2007-07-17 at 15:43 +0200, jan.sarin at securia.se wrote:

> Hi,
> 
> Lately we've noticed that some Zap channels on one of our PRIs are 
> unavailable. We have 2 PRI lines with 60 channels in total. On the 
> first PRI there are currently 20 channels that are not being used for 
> some reason.
> 
> I tried googling around and found some similar problems but there 
> really was no solution (?). I'm not sure if this problem has occured 
> now because of more load on the pbx but the machine should take it 
> just fine (2x3,0 ghz xeon with 1 gb ram etc).
> 
> Restarting asterisk makes the zaps' available again but they get 
> "locked" later again. It seems it's always the same channels that are 
> unavailable too?
> 
> This one is unavailable and not being used... It's been in PRI Flags 
> state "resetting" for hours now.
> 
> Channel: 1
> File Descriptor: 11
> Span: 1
> Extension:
> Dialing: no
> Context: from-pstn
> Caller ID: 702821667
> Calling TON: 33
> Caller ID name:
> Destroy: 0
> InAlarm: 0
> Signalling Type: PRI Signalling
> Radio: 0
> Owner: 
> Real: 
> Callwait: 
> Threeway: 
> Confno: -1
> Propagated Conference: -1
> Real in conference: 0
> DSP: no
> Relax DTMF: no
> Dialing/CallwaitCAS: 0/0
> Default law: alaw
> Fax Handled: no
> Pulse phone: no
> Echo Cancellation: 128 taps unless TDM bridged, currently OFF PRI 
> Flags: Resetting PRI Logical Span: Implicit Hookstate (FXS only): 
> Onhook
> 
> If anyone can help me with this I'd be really glad. Thanks.
> 
> Regards,
> Jan

___
--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

Re: [asterisk-users] Zap channels unavailable?

[jan.sarin]

Hi,

No I havn't tried that. That entry wasn't even in there so I'll try it.
I'll let you know if it helped. 

The odd thing is that this problem started yesterday. And our asterisk
has been running for +1 year without these kind of problems.

So either our telco has changed something OR it's because of the "heavy"
load on the server (cpu running at max 20% with 40-50 simultaneous
calls, so why would it be this?).

Regards,
Jan

--

Have you tried setting resetinterval=never in zapata.conf?

On Tue, 2007-07-17 at 15:43 +0200, jan.sarin at securia.se wrote:

> Hi,
> 
> Lately we've noticed that some Zap channels on one of our PRIs are
> unavailable. We have 2 PRI lines with 60 channels in total. On the
first
> PRI there are currently 20 channels that are not being used for some
> reason.
> 
> I tried googling around and found some similar problems but there
really
> was no solution (?). I'm not sure if this problem has occured now
> because of more load on the pbx but the machine should take it just
fine
> (2x3,0 ghz xeon with 1 gb ram etc).
> 
> Restarting asterisk makes the zaps' available again but they get
> "locked" later again. It seems it's always the same channels that are
> unavailable too?
> 
> This one is unavailable and not being used... It's been in PRI Flags
> state "resetting" for hours now. 
> 
> Channel: 1
> File Descriptor: 11
> Span: 1
> Extension:
> Dialing: no
> Context: from-pstn
> Caller ID: 702821667
> Calling TON: 33
> Caller ID name:
> Destroy: 0
> InAlarm: 0
> Signalling Type: PRI Signalling
> Radio: 0
> Owner: 
> Real: 
> Callwait: 
> Threeway: 
> Confno: -1
> Propagated Conference: -1
> Real in conference: 0
> DSP: no
> Relax DTMF: no
> Dialing/CallwaitCAS: 0/0
> Default law: alaw
> Fax Handled: no
> Pulse phone: no
> Echo Cancellation: 128 taps unless TDM bridged, currently OFF
> PRI Flags: Resetting
> PRI Logical Span: Implicit
> Hookstate (FXS only): Onhook
> 
> If anyone can help me with this I'd be really glad. Thanks.
> 
> Regards,
> Jan

___
--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

Re: [asterisk-users] Slow list

[Philipp Kempgen]

Anthony Francis wrote:
> Doug Lytle wrote:
 Before poking Digium too much, I would look at exactly what YOUR mail
 servers are doing that may potentially be the real cause of the delays.

> You have two servers in your MX records.
> drdos.info. 60  IN  MX  10 smtp.drdos.info.
> drdos.info. 60  IN  MX  5 drdos.info.
> 
> The one weighted 10 refuses smtp connections, while it is the higher 
> weight it is the first one that was listed when I did the dig. SO there 
> stand s a large probability that the list server is trying that address 
> first.

It's the other way round.
http://en.wikipedia.org/wiki/MX_record#MX_priority :
"The MX record with the smallest preference number has the
highest priority and is the first server to be tried."


Regards,
  Philipp Kempgen

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

Geschäftsführer: Stefan Wintermeyer
Handelsregister: Neuwied B 14998

___
--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

Re: [asterisk-users] asterisk web interface

[Adam Moffett]

So what you actually want a web based phone?

Jody Gugelhupf wrote:

  hi there :) 

  
  

  i want to have a website, which offer the
following:

# Integrated Web Dialer (Click-to-Dial)
  

Easy to make your own. The only question is "integrate into what?"

  
  
i don't know, maybe easy for you but not for me ;) just a webinterface in php or twisted.web
maybe?

  
  

  # Workgroup Answering Machine
# Monitor recent call history (CDR)
# Listen/manage voicemail
# Monitor incoming calls and call history of incommng calls
  

Debian has freePBX and DeStar packages. I think both provide most or all
of what you need.

If not, what do you find missing?

  
  
well as i said i tried them out, but i did not find or only partial but not all together, the
following features for sip only: 

webdialer, 
listen/manage voicemail, 
Monitor incoming calls and call history of incommng calls with call back function, 
pick-up incoming calls through the webinterface, 
online phonebook

these functions integrated into a simple plain webinterface would be great, so anyone knows one?
thx 
jody :D



  Ask a question on any topic and get answers from real people. Go to Yahoo! Answers and share what you know at http://ca.answers.yahoo.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

  




___
--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

Re: [asterisk-users] double digits on SIP->PRI [was: Zaptel 1.2.19 and 1.4.4 released]

[Tzafrir Cohen]

Hi

On Tue, Jul 17, 2007 at 08:58:26AM -0400, James FitzGibbon wrote:
> On 7/16/07, The Asterisk Development Team <[EMAIL PROTECTED]> wrote:
> 
> >fix various known issues.  See the ChangeLog included in the releases
> >for a full list of changes.  The ChangeLogs are also available
> >separately on the ftp site.
> >
> 
> Is there any more information available on this change?
> 
> 2007-07-13 08:22 + [r2733-2736]  Tzafrir Cohen <[EMAIL PROTECTED]
> >
> 
>  * Fix a digit mapping bug with hardware dtmf detection (r4357)
> 
> I assume those revision numbers (4xxx) in the changelog are from another SVN
> repo - I was trying to figure out what exactly had changed, but the large
> number of files changed across 3 revs makes it a bit tough to isolate.

Yes, they all refer to revision numbers inside our internal repository.

> 
> I'm fighting with a DTMF problem right now where directly dialed SIP->Zap
> (PRI) 

Unfortunetly I'm afraid that this specific fix is not goin to help you. 
It was an internal fix in a new feature regarding to hardware-based DTMF
detection in our driver.


Anyway, your question is a good question for asterisk-users
nevertheless.

> calls produce doubled-up DTMF on the other side (not every digit, but
> just one causes the remote IVR to go into fits).  The problem first appeared
> when we moved from 1.2.x to * 1.4.4/Zaptel 1.4.3.  I haven't had much time
> to investigate - it happens whether the SIP agent connected to * is using
> inband, rfc2833 or info and across Grandstream, Aastra and Polycom units.
> 
> Interestingly, if I send the SIP phone into DISA() pointing at it's own
> context, I get no DTMF problems on the remote side.

Are the digits sent inband in the PRI channel or out-of-band as
signalling? (Or both?)

You can check if the DTMF digits are sent inline in the PRI by listening
to the specific channel (e.g: with ztmonitor).

You can check if they are sent out-of-band using 'pri debug span N', I
guess.

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

___
--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

Re: [asterisk-users] asterisk web interface

[Anthony Francis]

Jody Gugelhupf wrote:
> hi there :) 
>
>   
>>> i want to have a website, which offer the
>>> following:
>>>
>>> # Integrated Web Dialer (Click-to-Dial)
>>>   
>> Easy to make your own. The only question is "integrate into what?"
>> 
>
> i don't know, maybe easy for you but not for me ;) just a webinterface in php 
> or twisted.web
> maybe?
>
>   
>>> # Workgroup Answering Machine
>>> # Monitor recent call history (CDR)
>>> # Listen/manage voicemail
>>> # Monitor incoming calls and call history of incommng calls
>>>   
>> Debian has freePBX and DeStar packages. I think both provide most or all
>> of what you need.
>>
>> If not, what do you find missing?
>> 
>
> well as i said i tried them out, but i did not find or only partial but not 
> all together, the
> following features for sip only: 
>
> webdialer, 
> listen/manage voicemail, 
> Monitor incoming calls and call history of incommng calls with call back 
> function, 
> pick-up incoming calls through the webinterface, 
> online phonebook
>
> these functions integrated into a simple plain webinterface would be great, 
> so anyone knows one?
> thx 
> jody :D
>
>
>
>   Ask a question on any topic and get answers from real people. Go to 
> Yahoo! Answers and share what you know at http://ca.answers.yahoo.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
>   
I would suggest making your own. Oh BTW the FREE in open source doesnt 
mean without financial cost, there is always a cost to everything, 
whether it be in support, downtime, or the time it took to develop the 
software, cost is always there, and as you base your decisions about 
what software to use, you may want to consider all true costs involved 
and not just "what can I get without paying money?".

Just IMO,
Anthony

___
--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

Re: [asterisk-users] Zap channels unavailable?

[James Texter]

Have you tried setting resetinterval=never in zapata.conf?

On Tue, 2007-07-17 at 15:43 +0200, [EMAIL PROTECTED] wrote:

> Hi,
> 
> Lately we've noticed that some Zap channels on one of our PRIs are
> unavailable. We have 2 PRI lines with 60 channels in total. On the first
> PRI there are currently 20 channels that are not being used for some
> reason.
> 
> I tried googling around and found some similar problems but there really
> was no solution (?). I'm not sure if this problem has occured now
> because of more load on the pbx but the machine should take it just fine
> (2x3,0 ghz xeon with 1 gb ram etc).
> 
> Restarting asterisk makes the zaps' available again but they get
> "locked" later again. It seems it's always the same channels that are
> unavailable too?
> 
> This one is unavailable and not being used... It's been in PRI Flags
> state "resetting" for hours now. 
> 
> Channel: 1
> File Descriptor: 11
> Span: 1
> Extension:
> Dialing: no
> Context: from-pstn
> Caller ID: 702821667
> Calling TON: 33
> Caller ID name:
> Destroy: 0
> InAlarm: 0
> Signalling Type: PRI Signalling
> Radio: 0
> Owner: 
> Real: 
> Callwait: 
> Threeway: 
> Confno: -1
> Propagated Conference: -1
> Real in conference: 0
> DSP: no
> Relax DTMF: no
> Dialing/CallwaitCAS: 0/0
> Default law: alaw
> Fax Handled: no
> Pulse phone: no
> Echo Cancellation: 128 taps unless TDM bridged, currently OFF
> PRI Flags: Resetting
> PRI Logical Span: Implicit
> Hookstate (FXS only): Onhook
> 
> If anyone can help me with this I'd be really glad. Thanks.
> 
> Regards,
> Jan
> 
> ___
> --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
___
--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

[asterisk-users] 2 PRI on asterisk

[satish patel]

Dear all
   
  I am going to install 2 port pri card on asterisk but i dont 
know how to incomming call goes in to IVR and how to route call outside base on 
pattern match means if some one call on mobile phone then use PRI 1 and if call 
on landline phon call route through pri 2 how to make dission base on pattern 
number
   
   
  Rgds
   
  satish patel

   
-
Building a website is a piece of cake. 
Yahoo! Small Business gives you all the tools to get online.___
--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

[asterisk-users] Asterisk and ATA-186 question-- calling one port from the other port..

[Tim Reimers]

 
 
Hi - 
 
I need to configure Asterisk (Trixbox 2.2) and my ATA-186 with both
ports.
 
I need to be able to call one port from the other-- the idea is to have
two phones in two different locations that _can_ call each other.
 
So, in reading the Asterisk Wiki and other sites, the best documentation
I found was this:
 
 
 
 
http://www.loligo.com/asterisk/Cisco/ATA-186-guide.v20030628.txt
 
 
"Note that each line must have it's own distinct and complete
configuration, and if you use both lines on the ATA-186, it will
REGISTER twice.  Further note that you cannot call one line from the
other on the same device using the "direct" extension numbers, so you
will have to be clever about naming and aliases within Asterisk.  That
is outside the scope of this document."

However, that _specifically_ says that in the provided config, you
cannot call one port from the other port using "direct" lines--
It does indicate that you CAN in fact work that out, using "naming and
aliases" within Asterisk.
 
Therefore, I assume that it IS possible to use an ATA like this---
but that the author of this particular doc either doesn't know how (but
does know it can be done)
or just didn't want to go into it in a low-level howto.
 
So ---
 
Does anyone know how to do this?
 
thanks, Tim
 
  

Most days, there are several fires burning at once. Some days, what's
burning is your fire extinguisher.
To err is human; to truly screw it up requires the root password.

 

 
___
--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

Re: [asterisk-users] Zap channels unavailable?

[Martin Smith]

Hello Jan,

We have also been seeing this issue, and we are running Asterisk
1.2.17/Zaptel 1.2.16/LibPRI 1.2.4-r2. We have been informed by our PRI
provider that a "3rd party vendor" has applied firmware to some hardware
along our path, and that it has an unfortunate bug of hanging B-channels
in the PRI flags "resetting" state. We have been assured that the vendor
has been given a 30-day deadline (starting maybe 2 weeks ago) to fix the
problem, and that it will go away soon. In the mean time, we've also had
to restart Asterisk to free our B-channels for use, and any link-level
event potentially re-hangs them.

Keep us posted if you find out anything!

Martin Smith, Systems Developer
[EMAIL PROTECTED]
Bureau of Economic and Business Research
University of Florida
(352) 392-0171 Ext. 221 

 

> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of 
> [EMAIL PROTECTED]
> Sent: Tuesday, July 17, 2007 9:44 AM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Zap channels unavailable?
> 
> Hi,
> 
> Lately we've noticed that some Zap channels on one of our PRIs are
> unavailable. We have 2 PRI lines with 60 channels in total. 
> On the first
> PRI there are currently 20 channels that are not being used for some
> reason.
> 
> I tried googling around and found some similar problems but 
> there really
> was no solution (?). I'm not sure if this problem has occured now
> because of more load on the pbx but the machine should take 
> it just fine
> (2x3,0 ghz xeon with 1 gb ram etc).
> 
> Restarting asterisk makes the zaps' available again but they get
> "locked" later again. It seems it's always the same channels that are
> unavailable too?
> 
> This one is unavailable and not being used... It's been in PRI Flags
> state "resetting" for hours now. 
> 
> Channel: 1
> File Descriptor: 11
> Span: 1
> Extension:
> Dialing: no
> Context: from-pstn
> Caller ID: 702821667
> Calling TON: 33
> Caller ID name:
> Destroy: 0
> InAlarm: 0
> Signalling Type: PRI Signalling
> Radio: 0
> Owner: 
> Real: 
> Callwait: 
> Threeway: 
> Confno: -1
> Propagated Conference: -1
> Real in conference: 0
> DSP: no
> Relax DTMF: no
> Dialing/CallwaitCAS: 0/0
> Default law: alaw
> Fax Handled: no
> Pulse phone: no
> Echo Cancellation: 128 taps unless TDM bridged, currently OFF
> PRI Flags: Resetting
> PRI Logical Span: Implicit
> Hookstate (FXS only): Onhook
> 
> If anyone can help me with this I'd be really glad. Thanks.
> 
> Regards,
> Jan
> 
> ___
> --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
> 

___
--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

Re: [asterisk-users] Multiple Parking Lots

[Kevin Kiely]

I should have included using a multi parking feature with asterisk 1.4?

-Original Message-
From: Kevin Kiely [mailto:[EMAIL PROTECTED] 
Sent: Tuesday, July 17, 2007 9:23 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Multiple Parking Lots

Anyone using any variation of Multiparking, Parking Valet or servicing Call
Parking with Multiple Tennants?


___
--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

No virus found in this incoming message.
Checked by AVG Free Edition. 
Version: 7.5.476 / Virus Database: 269.10.8/904 - Release Date: 7/16/2007
5:42 PM
 


___
--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

[asterisk-users] Zap channels unavailable?

[jan.sarin]

Hi,

Lately we've noticed that some Zap channels on one of our PRIs are
unavailable. We have 2 PRI lines with 60 channels in total. On the first
PRI there are currently 20 channels that are not being used for some
reason.

I tried googling around and found some similar problems but there really
was no solution (?). I'm not sure if this problem has occured now
because of more load on the pbx but the machine should take it just fine
(2x3,0 ghz xeon with 1 gb ram etc).

Restarting asterisk makes the zaps' available again but they get
"locked" later again. It seems it's always the same channels that are
unavailable too?

This one is unavailable and not being used... It's been in PRI Flags
state "resetting" for hours now. 

Channel: 1
File Descriptor: 11
Span: 1
Extension:
Dialing: no
Context: from-pstn
Caller ID: 702821667
Calling TON: 33
Caller ID name:
Destroy: 0
InAlarm: 0
Signalling Type: PRI Signalling
Radio: 0
Owner: 
Real: 
Callwait: 
Threeway: 
Confno: -1
Propagated Conference: -1
Real in conference: 0
DSP: no
Relax DTMF: no
Dialing/CallwaitCAS: 0/0
Default law: alaw
Fax Handled: no
Pulse phone: no
Echo Cancellation: 128 taps unless TDM bridged, currently OFF
PRI Flags: Resetting
PRI Logical Span: Implicit
Hookstate (FXS only): Onhook

If anyone can help me with this I'd be really glad. Thanks.

Regards,
Jan

___
--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

Re: [asterisk-users] Zaptel 1.2.19 and 1.4.4 released

[Russell Bryant]

James FitzGibbon wrote:
> Is there any more information available on this change?
> 
> 2007-07-13 08:22 + [r2733-2736]  Tzafrir Cohen 
> <[EMAIL PROTECTED] >
> 
>   * Fix a digit mapping bug with hardware dtmf detection (r4357)
> 
> I assume those revision numbers (4xxx) in the changelog are from another 
> SVN repo - I was trying to figure out what exactly had changed, but the 
> large number of files changed across 3 revs makes it a bit tough to isolate.

This change was to the xpp driver for the Astribank.  The revision number is 
from an internal svn server at Xorcom, I think.

-- 
Russell Bryant
Software Engineer
Digium, Inc.

___
--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

[asterisk-users] Multiple Parking Lots

[Kevin Kiely]

Anyone using any variation of Multiparking, Parking Valet or servicing Call
Parking with Multiple Tennants?


___
--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

Re: [asterisk-users] Problem with H option of Dial()

[Gary Chen]

I also tried blind transfer with t option and it did not work. I added 
following into my dial plan contest:

include => featuremap

exten => 8111001001,1,Answer()
exten => 8111001001,n,Wait(2)
exten => 8111001001,n,Dial(SIP/[EMAIL PROTECTED]|30|tHL(12:61000:3))
exten => 8111001001,n,Hangup()

It still does not work.

I issue show features in CLI it show this:
Builtin Feature   Default Current
---   --- ---
Pickup*8  *8
Blind Transfer#   #1
Attended Transfer
One Touch Monitor
Disconnect Call   *   *
Park Call

Dynamic Feature   Default Current
---   --- ---
(none)

Call parking

Parking extension   :   700
Parking context :   parkedcalls
Parked call extensions: 701-720


What else do I need to do to make the features work?

Gary Chen

  - Original Message - 
  From: Gary Chen 
  To: asterisk-users@lists.digium.com 
  Sent: Tuesday, July 17, 2007 8:24 AM
  Subject: [asterisk-users] Problem with H option of Dial()


  I just upgrade my test server to Asterisk 1.4.7.1 and having problem using H 
option in Dial() app. When press *  during the call from caller side, Asterisk 
does not disconnect the call. The * just pass through. Here is my test dial 
plan:

  exten => 8111001001,1,Answer()
  exten => 8111001001,n,Wait(2)
  exten => 8111001001,n,Dial(SIP/[EMAIL PROTECTED]|30|HL(12:61000:3))
  exten => 8111001001,n,Hangup()

  It worked with previous version 1.2.18 on this box. I am new to 1.4. Do I 
miss something? Or is it just a bug?

  Gary Chen




--


  ___
  --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___
--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

Re: [asterisk-users] asterisk web interface

[Jody Gugelhupf]

hi there :) 

> > i want to have a website, which offer the
> > following:
> > 
> > # Integrated Web Dialer (Click-to-Dial)
> 
> Easy to make your own. The only question is "integrate into what?"

i don't know, maybe easy for you but not for me ;) just a webinterface in php 
or twisted.web
maybe?

> > # Workgroup Answering Machine
> > # Monitor recent call history (CDR)
> > # Listen/manage voicemail
> > # Monitor incoming calls and call history of incommng calls
> 
> Debian has freePBX and DeStar packages. I think both provide most or all
> of what you need.
> 
> If not, what do you find missing?

well as i said i tried them out, but i did not find or only partial but not all 
together, the
following features for sip only: 

webdialer, 
listen/manage voicemail, 
Monitor incoming calls and call history of incommng calls with call back 
function, 
pick-up incoming calls through the webinterface, 
online phonebook

these functions integrated into a simple plain webinterface would be great, so 
anyone knows one?
thx 
jody :D



  Ask a question on any topic and get answers from real people. Go to 
Yahoo! Answers and share what you know at http://ca.answers.yahoo.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

[asterisk-users] chan_isdn with HFC-compatible

[Michael Kamleitner]

hi list,

I'm currently trying to get Asterisk running with an HFC-compatible ISDN
card (no-name product, but supposed to work with Asterisk according to the
packaging). the ISDN-card is connected to a alcatel ISDN-system where it
should act just like a normal ISDN-phone.

I went with http://www.misdn.org/index.php/MISDN_with_Asterisk and installed
the package from beronet.com including:

asterisk 1.4.7.1
libpri 1.4.1
mISDN 1.1.5 & mISDN usertools
zaptel 1.4.3

installation went ok, "mISDN config" recognized my card & created the
accpording /etc/mISDN.conf file. I manually edited /etc/asterisk/misdn.conf.

Asterisk is loading the chan_misdn and lists mISDN when issueing "show
channeltypes" -  however it indicates "Devicestate - No". when I look for
"misdn show stacks", it lists the single port of the ISDN-card, however
indicates "L2Link DOWN, L1LinkDOWN". so I  guess theres something wrong,
unfortunately I've no idea  on what to check next...any pointers? maybe I'm
missing something obvious, as this is my first installation of Asterisk +
ISDN...

thx for any help!

regards,
michael
___
--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

Re: [asterisk-users] Zaptel 1.2.19 and 1.4.4 released

[James FitzGibbon]

On 7/16/07, The Asterisk Development Team <[EMAIL PROTECTED]> wrote:


fix various known issues.  See the ChangeLog included in the releases
for a full list of changes.  The ChangeLogs are also available
separately on the ftp site.



Is there any more information available on this change?

2007-07-13 08:22 + [r2733-2736]  Tzafrir Cohen <[EMAIL PROTECTED]




 * Fix a digit mapping bug with hardware dtmf detection (r4357)

I assume those revision numbers (4xxx) in the changelog are from another SVN
repo - I was trying to figure out what exactly had changed, but the large
number of files changed across 3 revs makes it a bit tough to isolate.

I'm fighting with a DTMF problem right now where directly dialed SIP->Zap
(PRI) calls produce doubled-up DTMF on the other side (not every digit, but
just one causes the remote IVR to go into fits).  The problem first appeared
when we moved from 1.2.x to * 1.4.4/Zaptel 1.4.3.  I haven't had much time
to investigate - it happens whether the SIP agent connected to * is using
inband, rfc2833 or info and across Grandstream, Aastra and Polycom units.

Interestingly, if I send the SIP phone into DISA() pointing at it's own
context, I get no DTMF problems on the remote side.

Thanks

--
j.
___
--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

[asterisk-users] Problem with H option of Dial()

[Gary Chen]

I just upgrade my test server to Asterisk 1.4.7.1 and having problem using H 
option in Dial() app. When press *  during the call from caller side, Asterisk 
does not disconnect the call. The * just pass through. Here is my test dial 
plan:

exten => 8111001001,1,Answer()
exten => 8111001001,n,Wait(2)
exten => 8111001001,n,Dial(SIP/[EMAIL PROTECTED]|30|HL(12:61000:3))
exten => 8111001001,n,Hangup()

It worked with previous version 1.2.18 on this box. I am new to 1.4. Do I miss 
something? Or is it just a bug?

Gary Chen

___
--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

[asterisk-users] Not hearing the caller after 2 x Flash

[Jakub Głazik]

Me again, another problem.
As I said before, I have 2 lines going into "incoming" context.

When client calls, I press Flash, client hears music on hold (only on
voip line as said in previous post), when I get back and press Flash
again to get back to my client I cannon hear him, but he hears me
without problems.

I have just tested in on the LAN, same situations, happens everytime.

-- Executing [EMAIL PROTECTED]:1] Dial("SIP/zytek-087a2000",
"SIP/113|720|Ttm") in new stack -- Called 113
-- Started music on hold, class 'default', on channel
'SIP/zytek-087a2000' -- SIP/113-087a7000 is ringing
-- SIP/113-087a7000 answered SIP/zytek-087a2000
-- Stopped music on hold on SIP/zytek-087a2000
-- Started music on hold, class 'default', on channel
'SIP/zytek-087a2000' -- Stopped music on hold on SIP/zytek-087a2000
  == Spawn extension (firma, 113, 1) exited non-zero on
'SIP/zytek-087a2000'

Asterisk 1.4.7.1

Maybe dtfm ? My gateway in on rfc, office is on info, but problems are
same here and there.

-- 
.: Jakub Głazik,
.: email & jabber: zyteknuxi.pl

___
--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

[asterisk-users] asterisk web interface

[Jody Gugelhupf]

hi there ppl :D
i'm a happy and very gratefull asterisk user, newest version, on a debian 
machine (stable) i also
have tried asterisknow and freepbx etc, but i want to have a website, which 
offer the following:

# Integrated Web Dialer (Click-to-Dial)
# Workgroup Answering Machine
# Monitor recent call history (CDR)
# Listen/manage voicemail
# Monitor incoming calls and call history of incommng calls

i think geotek http://voip-manager.net/ does it, but costs money and is no open 
source, is there
an open source alternative, or something free, which does the things mentioned, 
just something
basic and easy would be great, i have been looking for something like this for 
a long time and
though i'd try here and ask, maybe someone knows something, thanks already in 
advacne Smile
have a good one
jody :)


  Be smarter than spam. See how smart SpamGuard is at giving junk email the 
boot with the All-new Yahoo! Mail at http://mrd.mail.yahoo.com/try_beta?.intl=ca


___
--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

[asterisk-users] help with sip configuration for sipgate.de on asterisk 1.4

[Jody Gugelhupf]

hi there,
i run asterisk 1.4 on my debian machine, which is in my internal 10.x.x.x 
network, behind my main
computer, i cam make call, receive calls, all works fine, with all providers 
except sipgate.de,
there i can receive call and make them, i can hear the other end but they can 
not hear me, this is
only the case with sipgate.de i don#t know how to configure it and thought 
maybe someone can help
me out here. i have a grandstream 486, which is in my sip.conf the 'gs486 
deivce, it is connected
to my debain/asterisk machine, and calls are made through it. I was also 
wondering how i can
activate the 2nd incoming caller thing, usually i used to hear a clicking in 
the line when
somebody called me and i could switch between callers, is this possible in 
asteriks? at least one
of my providers supports this service. I was also wondering if there is a free 
webinterface for
asterisk, from which i can see incoming calls and also make calls, mainly only 
sip, i don't mean
trixbox or freepbx. anyhow my main problem is the sipgate.de thing, below are 
my sip.conf and
extension.conf thx for any help

sip.conf:
[general]
language=en
disable=all
register => user:[EMAIL PROTECTED]/number
register => user:[EMAIL PROTECTED]/number
register => user:[EMAIL PROTECTED]/number
register => user:[EMAIL PROTECTED]/number
register => user:[EMAIL PROTECTED]/
register => user:[EMAIL PROTECTED]/number
register => user:[EMAIL PROTECTED]/number
register => user:[EMAIL PROTECTED]:5061/number
context=inbound
bind=0.0.0.0
nat=yes
fromdomain=sshn.net
localnet=10.0.0.0/255.255.255.0
externip=195.xxx.xxx.xxx
srvlookup=yes


[authentication]

[eutelia-out]
;maxexpirey=36
;defaultexpirey=18
type=friend
allow=alaw
context=inbound
username=
secret=x
fromuser=number
fromdomain=voip.eutelia.it
host=voip.eutelia.it
dtmfmode=inband
realm=voip.eutelia.it
registertimeout=300
canreinvite=no
;registertimeout=99
qualify=200
insecure=very
,allow=alaw
,allow=ulaw
,allow=gsm

[messagenet-out]
auth=user:[EMAIL PROTECTED]
;auth=md5
realm=sip.messagenet.it
qualify=yes
;maxexpirey=36
;defaultexpirey=18
authname=user
authuser=user
canreinvite=no
context=inbound
;dtmf=rfc2833
;dtmfmode=rfc2833
fromdomain=sip.messagenet.it
fromuser=user
host=sip.messagenet.it
port=5061
insecure=very
regexten=user
secret=password
md5secret=xxx
type=peer
;user=phone
username=user

[webcalldirect-out]
type=peer
context=inbound
username=user
secret=pass
fromuser=user
fromdomain=sip.webcalldirect.com
host=sip.webcalldirect.com
qualify=no
insecure=very
canreinvite=no
allow=all

[sipgate-out]
type=friend
context=inbound
username=user
secret=pass
host=sipgate.de
fromuser=user
fromdomain=sipgate.de
authuser=pass
qualify=yes
insecure=very
disable=all
;allow=alaw
allow=ulaw
allow=alaw
allow=g729
allow=gsm
allow=slinear

[gs486]
type=friend
context=default
username=user
secret=pass
host=dynamic
,nat=no
canreinvite=yes
dtmfmode=info
call-limit=2
allow=all

[budgetphone-out]
type=friend
context=inbound
secret=pass
username=user
host=budgetphone.nl
fromuser=user
fromdomain=budgetphone.nl
insecure=very
allow=alaw

[freecall-out]
type=peer
username=user
secret=pass
qualify=no
authuser=user
fromuser=user
fromdomain=freecall.com
host=sip.voiparound.com
insecure=very
canreinvite=no
allow=all

[iXcall-out]
type=friend
context=inbound
secret=pass
username=user
host=ixcall.net
fromuser=user
fromdomain=ixcall.net
insecure=very
allow=alaw


[freedigits-out]
allow=alaw
context=inbound
fromdomain=freedigits.net
fromuser=user
host=freedigits.net
insecure=very
secret=pass
type=friend
username=user


extensions.conf:
;!
;! Automatically generated configuration file
;! Filename: extensions.conf (/etc/asterisk/extensions.conf)
;! Generator: Manager
;! Creation Date: Sun Jun 10 16:08:39 2007
;!
[general]
static = yes
writeprotect = no
autofallthrough = yes
clearglobalvars = no
priorityjumping = no

[globals]
trunk_1 = SIP/trunk_1
trunk_2 = SIP/trunk_2
trunk_4 = SIP/trunk_4
trunk_6 = IAX2/trunk_6
trunk_7 = IAX2/trunk_7
trunk_8 = SIP/trunk_8

[inbound]
exten => s,1,NoOp(Inbound Call CallerID: ${CALLERID(all)})
;exten => s,n,Answer(SIP/gs486)
exten => s,n,Dial(SIP/gs486,30)
exten => s,n,Hangup()
;exten => number,1,Goto(inbound,s,1)
exten => number,1,Goto(from-eutelia,s,1)
exten => number,1,Goto(from-budgetphone,s,1)
exten => user,1,Goto(from-sipgate,s,1)
exten => user,1,Goto(from-ixcall,s,1)
exten => user,1,Goto(from-freedigits,s,1)
exten => user,1,Goto(inbound,s,1)

[default]
exten => _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,r)
exten => _8.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,r)
exten => _7.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,r)
exten => _6.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,r)
exten => _5.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,r)
exten => _4.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,r)
exten => _3.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,r)
exten => _2.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,r)
exten => 1000,1,Dial(SIP/gs486)
exten => 444

[asterisk-users] No music on hold on ISDN line

[Jakub Głazik]

Hello,

I have two "incoming" lines connected to my Asterisk
([EMAIL PROTECTED]). One voip and one ISDN line. Both go into
"incoming" context.
I have a problem, that when I press Flash the client who calls does not
hear music on hold, but only on the ISDN line, on VOIP everything is ok.

[incoming]
exten => s,1,SetMusicOnHold(default)
exten => s,2,AGI(test|incoming)
exten => s,3,Dial(${INOTEL_INCOMING},${RINGTIME},Tt)
exten => s,n,Hangup
exten => h,1,DeadAGI(test|dead)

Got any ideas?

-- Executing [EMAIL PROTECTED]:3] Dial("CAPI/ISDN/717817630-2",
"SIP/101&SIP/102|720|Tt") in new stack -- Called 101
-- Called 102
-- SIP/102-08783000 is ringing
-- SIP/102-08783000 answered CAPI/ISDN/717817630-2
  ==
capi_send_detect_dtmf_req:3445:ENTRY=ISDN:PLCI=0x1400:PBX_CHAN=CAPI/ISDN/717817630-2:
==   Setting up DTMF detector, flag=1

Now the "flash" key was pressed and there is no "Starting music on
hold". Got any ideas why?

[EMAIL PROTECTED] /usr/local/etc/asterisk]# cat capi.conf 
[general]
nationalprefix=0
internationalprefix=00
rxgain=1
txgain=1
debug=no

[ISDN]
isdnmode=msn
msn=717817630
incomingmsn=717817630
controller=0
group=1
softdtmf=off
relaxdtmf=off
accountcode=
context=incoming
holdtype=local
echocancel=no
echosquelch=no
devices=2 


[EMAIL PROTECTED] /usr/local/etc/asterisk]# isdnconfig 
controller 0 = {
  Layer 1:
description : HFC-2BDS0 128K PCI ISDN adapter
type: passive ISDN (Basic Rate, 2xB)
channels: 0x3
serial  : 0xabcd
power_save  : on
dialtone: enabled
attached: yes
PH-state: F7: Activated
  Layer 2:
driver_type : DRVR_D_CHANNEL
}

-- 
.: Jakub Głazik,
.: email & jabber: zyteknuxi.pl

___
--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

Re: [asterisk-users] tT in callparking

[bilal ghayyad]

Dear Mojo;

Thanks a lot, yes I understood what you mean. You mean
that the tT was putted in the Dial as a feature or
setting to give them the chance to use the call
parking that was included in the first line.

But only when dialing bob or charlie.  Only the second
line, the 
'include' line, is for call parking.  The others are
NOT for call 
parking and are unrelated -- They are just for dialing
charlie and bob 
directly.


Jared Smith wrote:
> On Sat, 2007-07-14 at 13:20 -0700, bilal ghayyad
wrote:
>> [incoming]
>> include => parkedcalls
>> exten=103,1,Dial(SIP/Bob,,tT)
>> exten=104,1,Dial(SIP/Charlie,,tT)
>>
>> When we use tT and when we use t alone or T alone,
I
>> know this for call parking, but I do not know what
the
>> tT does?
> 
> The 'tT' means use both the 't' and 'T' options,
allowing either the
> calling party or the called party to transfer the
call.

Regards
Bilal


   

Sick sense of humor? Visit Yahoo! TV's 
Comedy with an Edge to see what's on, when. 
http://tv.yahoo.com/collections/222

___
--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

[asterisk-users] Suggestion for installation

[FaberK]

Hi to all,
till now I've used SER as sip registrar and Asterisk as its gateway(PSTN)
and for billing.
Now, I've received a request to setup a solution, for 5000 + o - users(this
is what they expext to have).
I was thinking to use only Asterisk with Freeradius, no SER.
Any suggestion/experience?

Thanks to all

--
.:FaberK:.
___
--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

Re: [asterisk-users] improved SMS?

[Steve Kennedy]

On Tue, Jul 17, 2007 at 11:56:35AM +0200, Anselm Martin Hoffmeister wrote:

> Am Donnerstag, den 12.07.2007, 16:57 -0700 schrieb Russ McBride:
> > Newbie question(s):
> >  From what I can determine it sounds like the SMS messaging isn't as  
> > robust as it could be (?).  I'm wondering if there's active work on  
> > that right now or if it's more of an issue about PSTN carrier that  
> > one would be using who would be responsible for passing the messages  
> > into the PLMN.
> > Background-- I'm looking into the possibility of setting up an  
> > emergency messaging system here at the University that would send out  
> > voice, SMS, and emails.  Any input relevant to that goal would  
> > probably be appreciated.
> Hi Russ,
> my personal experience with short messages is that the system sometimes
> chews on them for minutes, sometimes several hours, even inside one
> mobile network, from cell phone to cell phone. This surely screws using
> it as a primary tier emergency system, but as a backup after e-mail and
> automated phone-out that could be OK. Sending from web-interfaces or via
> Uwhatever-that-protocol-is-called will not improve the overall
> performance.

SMS was never designed for guaranteed delivery (or guaranteed timed
delivery). There are options for messages to time out if they're not
delivered in a specified time, or new messages can override old messages
that haven't been read yet - but delivery isn't guaranteed.

A phone sending an SMS will try and establish a connection (sort of) all
the way through to the receiving phone and then deliver the message, if
it cant it will be sent to the receiving network's SMSC which will then
try and deliver it. If it gets put into a queue then the delivery time
will vary drastically depending on the load on the SMSC and other
network characteristics.

Fixed to SMS always goes through an SMSC, so delivery times vary.

Steve

-- 
NetTek Ltd  UK mob +44-(0)7775 755503
UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455
Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED]
Euro Tech News Blog http://eurotechnews.blogspot.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

Re: [asterisk-users] improved SMS?

[Anselm Martin Hoffmeister]

Am Donnerstag, den 12.07.2007, 16:57 -0700 schrieb Russ McBride:
> 
> Newbie question(s):
> 
>  From what I can determine it sounds like the SMS messaging isn't as  
> robust as it could be (?).  I'm wondering if there's active work on  
> that right now or if it's more of an issue about PSTN carrier that  
> one would be using who would be responsible for passing the messages  
> into the PLMN.
> 
> Background-- I'm looking into the possibility of setting up an  
> emergency messaging system here at the University that would send out  
> voice, SMS, and emails.  Any input relevant to that goal would  
> probably be appreciated.

Hi Russ,

my personal experience with short messages is that the system sometimes
chews on them for minutes, sometimes several hours, even inside one
mobile network, from cell phone to cell phone. This surely screws using
it as a primary tier emergency system, but as a backup after e-mail and
automated phone-out that could be OK. Sending from web-interfaces or via
Uwhatever-that-protocol-is-called will not improve the overall
performance.

Considering all options to send out SMs:
- Asterisk, SMS() app to a landline SMS gateway
- Web interface with script/wget
- Uwhatever-modem-dialup
the second seems the easiest to use to me, and in my experience the
first tends to choke on some messages, be it 1 in 100 - still not 100%
perfect. The web interface method surely is by far cheaper than the
other two, at least here in Germany, where #1 will be charged as a
call-to-cellphone, first minute, about 17 cent, and #3 if available for
the network you want to use will be similar.

With the web interface approach you also get rid of the problem of
number portability: The #3 approach will only deliver the message if you
connect to the provider that the number currently is contracted to,
while #2 will not care about that (#1 should also work).

You see I tend to prefer the web-based thing.

If you intend to send emergeny SMs, please try and find a trustworthy
supplier. In the European price scale the cheapest readily found
providers will charge about 3 cent per message, but if you go for the 10
cent providers you will find higher reliability (without routing
messages to Germany through a Romanian mobile network to save money).
Those messages directly inserted into the destination network are sold
as "provider messages" here, opposed to "cheapest" or "economy" or
whatever euphemism for crap they invent.

Do not mistake me though: For "fun" messages they use to be good enough.
If you talk about emergency, a few cent probably will not make a huge
difference though, and time might be an issue.

BR
Anselm


___
--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

[asterisk-users] Digitized audio at the beginning of a call

[Andrew]

Hi,

Apologies if this has been asked before, but I don't seem to be able  
to find any info on it anywhere.

Sometimes when placing a call on hold, the caller hears digitized/ 
robotic music on hold that gradually improves over the course of  
about 20 - 30 seconds until it sounds pretty normal. The first time  
the call is placed on hold the music sounds normal. If that same call  
is resumed and placed on hold again, the quality is poorer. A third  
time usually results in what I have described. Sometimes it is bad  
straight off but this pattern is typical.

A few specifics about this problem:

- It only happens when the call originates remotely from another  
identical Asterisk PBX. It never ever happens on a local call. It  
also never ever happens on an inbound call from a local PSTN gateway  
(Cisco router with a SIP dialpeer). It is only when the call comes  
from the remote PBX. SIP is being used throughout.

- The problem happens almost every time when the called phone is a  
Polycom 601. It happens much more rarely on Cisco 7912, 7940 and 7941  
handsets, although when it does happen the symptoms are the same.

- Reinvites are completely disabled on both PBXs.

- Codec is G.711alaw end to end. Music on hold is playing as native  
sound files in alaw format.

- When transferring a call, the symptoms often carry over into the  
beginning of the transferred call so that the caller will hear  
digitized voice which gradually improves. The problem doesn't appear  
to be hold specific, but it is the easiest way to replicate the  
symptoms.

I would love to know general reasons why digitized audio improves  
over time, almost as if it is training up. I'm also scratching my  
head as to why this happens more on the Polycom, given that Asterisk  
is playing the music and not the phone. Any pointers would be greatly  
appreciated.

Regards,
Andrew



___
--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

[asterisk-users] Music on hold problem

[yonoko molomo]

Hi,
I am using asterisk 1.4.
I have confgured the musiconhold.conf file.
However, when i make a call and then hold the call it does nothing.
in the CLI i do not see the starting/stopping musiconhold messages.

i am making calls from sip to h323 using asterisk assip/h323 gateway
(with gnugk and ooh323).
i get the following messages when putting the call on hold:

-- Executing [EMAIL PROTECTED]:1] Ringing("OOH323/1169fed2-70c5", "") in new 
stack
-- Executing [EMAIL PROTECTED]:2] Dial("OOH323/1169fed2-70c5",
"SIP/ht04|6|r") in new stack
-- Called ht04
-- SIP/ht04-081fe7f0 is ringing
-- SIP/ht04-081fe7f0 answered OOH323/1169fed2-70c5
<...on hold...>
[Jul 17 11:19:22] WARNING[23645]: src/chan_h323.c:1044
ooh323_indicate: Don't know how to indicate condition -1 on ooh323c_1
[Jul 17 11:19:25] NOTICE[23645]: rtp.c:783 process_rfc3389: Comfort
noise support incomplete in Asterisk (RFC 3389). Please turn off on
client if possible. Client IP: 10.4.0.116
[Jul 17 11:19:25] WARNING[23645]: src/chan_h323.c:1044
ooh323_indicate: Don't know how to indicate condition 16 on ooh323c_1

I have no idea why the musiconhold is not triggered,  what those
messages mean (dont know how to indicate condition) and if they are
related to the music on hold problem


someone has any idea?

in the CLI i type
CLI> moh show files
and i see one file i put in the directory.

i tried configuring the musiconhold.conf file as quitemp3, files and
also custom (installing mpg123) but none of them starts musiconhold.

do i need to activate musiconhold somewhere else?

any help is welcome
thanks

___
--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

Re: [asterisk-users] asterisk 1.4 and gnugk with ooh323

[yonoko molomo]

hi,
i fixed the problem.
as i thought it was a configuration problem, i was not defining the
asterisk users at ooh323.conf.
now it seems to work,
thanks

2007/7/17, Dovid B <[EMAIL PROTECTED]>:
> What output do you get from the CLI ?
>
> - Original Message -
> From: "yonoko molomo" <[EMAIL PROTECTED]>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> 
> Sent: Tuesday, July 17, 2007 9:59 AM
> Subject: Re: [asterisk-users] asterisk 1.4 and gnugk with ooh323
>
>
> > Hi,
> > Thanks for the answer.
> > Yes, I think both channels are built.
> > I see following messages at startup:
> > Parsing '/etc/asterisk/h323.conf': Found
> > Creating H.323 Endpoint
> > Parsing '/etc/asterisk/users.conf': Found
> > Registered channel type 'H323' (The NuFone Network's Open H.323 Channel
> > Driver)
> > H.323 listener started chan_h323.so =>
> > The NuFone Network's OpenH323 Channel Driver
> >
> > [and few lines after]
> > Parsing '/etc/asterisk/ooh323.conf': Found
> > Registered channel type 'OOH323' (Objective Systems H323 Channel Driver)
> > chan_ooh323.so => (^[[33;40mObjective Systems H323 Channel^[[0;37;40m)
> >
> > I am trying to use only ooh323 (removing the h323.conf file) but still
> > does not work.
> > I believe it is a configuration problem.
> > Can someone point me to the correct configuration or example? The ones
> > I have found so far did not help me much.
> >
> > Thanks
> >
> > 2007/7/16, Dovid B <[EMAIL PROTECTED]>:
> >> There are different h323 channel drivers. You seem to have built both. If
> >> you try to dial using Dial(h323/.) then you need h323.conf. If you
> >> try
> >> dialing using Dial(ooh323/.) then you need ooh323.conf. For me
> >> personally the ooh323 (can't remember the name of it  - sleep
> >> deprivation)
> >> works better for me.
> >>
> >> - Original Message -
> >> From: "yonoko molomo" <[EMAIL PROTECTED]>
> >> To: 
> >> Sent: Monday, July 16, 2007 6:13 PM
> >> Subject: [asterisk-users] asterisk 1.4 and gnugk with ooh323
> >>
> >>
> >> > Hello all,
> >> > I have seen some people asking how to configure asterisk to work with
> >> > h323 but i did not manage to do fix it yet (i am not an asterisk
> >> > expert).
> >> > Can someone help me configuring asterisk?
> >> > It is already compiled asterisk 1.4.5 with H323 support.
> >> > Everything looks fine.
> >> > Then i understand i need to configure several files:
> >> > -sip.conf
> >> > -ooh323.conf
> >> > -extensions.conf
> >> >
> >> > do i also need to configure the h323.conf?
> >> >
> >> > i want asterisk and gnugk in the same machine (lets say ip
> >> > 192.168.0.10).
> >> > then sip_client at 192.168.0.100 (telephone number assigned for
> >> > instance 100) and H323_client at 192.168.0.200 (telephone number
> >> > assigned 200)
> >> >
> >> > how do i configure the files to do this?
> >> >
> >> > if i type show channeltypes i see:
> >> >
> >> > TypeDescription  Devicestate
> >> > Indications  Transfer
> >> > --  ---  ---
> >> > ---  
> >> > OOH323  Objective Systems H323 Channel Driverno   yes
> >> >no
> >> > SIP Session Initiation Protocol (SIP)yes  yes
> >> >yes
> >> > H323The NuFone Network's Open H.323 Channel  no   yes
> >> >no
> >> >
> >> > What does devicestate and trasfer mean? it is set to "no". is this ok?
> >> >
> >> >
> >> > if someone could show me how to  configure the files to do this i
> >> > would be very grateful.
> >> >
> >> > thanks!
> >> >
> >> > ___
> >> > --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
> >> >
> >>
> >>
> >>
> >> ___
> >> --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
> >>
> >
> > ___
> > --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
> >
>
>
>
> ___
> --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
>

___
--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

[asterisk-users] Asterisk PRI Busy Problem

[Arun Kumar]

Hi,

I've an PRI coming to my asterisk ,calls are coming fine and my agents are
able to answer no prob. but I've an agreement with my telco with some
incoming no if the no of calls on these no are more then 3 then send to
another no. they use busy signal to divert call on another number so I'm
sending the call to Congestion() if no of calls in this group are more then
3. But my provider says he is not getting any busy signal from my side and
he says for all incoming numbers (30) he is getting back only one number
from asterisk box(4340).

here is my dial plan for one incoming DID:

exten => 4340,1,GotoIfTime(*|*|25|dec?ccagents,4340,6)
exten => 4340,2,GotoIfTime(10:00-16:00|*|26|dec?ccagents,4340,7)
exten => 4340,3,GotoIfTime(09:00-18:00|*|31|dec?ccagents,4340,7)
exten => 4340,4,GotoIfTime(12:00-16:00|*|1|jan?ccagents,4340,7)
exten => 4340,5,GotoIfTime(09:00-20:00|mon-sun,*,*?ccagents,4340,7)
exten => 4340,6,Goto(out-of-hours,5001,1)
exten => 4340,7,Set(GROUP(${EXTEN})=MAX_CALLS)
exten => 4340,8,GotoIf($[${GROUP_COUNT([EMAIL PROTECTED])} > 3]?16)
exten => 4340,9,Set(GROUP(${CALLERIDNUM})=MAX_CALLS)
exten => 4340,10,Answer()
exten => 4340,11,Playback(custom/next-avail-advisor)
exten =>
4340,12,Set(MONITOR_FILENAME=/var/spool/asterisk/q/tcarehwsupport-${TIMESTAMP}-${UNIQUEID})
exten => 4340,13,Monitor(wav,${MONITOR_FILENAME},mb)
exten => 4340,14,NoOp(${QUEUESTATUS})
exten => 4340,15,Queue(test,rt,,,10)
exten => 4340,16,Congestion()

zapata.conf:
---
[trunkgroups]

[channels]
language=en
context=ccagents
switchtype=euroisdn
pridialplan=unknown
overlapdial=yes
signalling=pri_cpe
rxgain = 0.0
txgain = 0.0
usecallerid=yes
hidecallerid=yes
callerid=asreceived
callwaiting=yes
usecallingpres=yes
echocancel=yes
echocancelwhenbridged=yes
immediate=no
cidsignalling=v23
callwaitingcallerid=yes
priindication = outofband

resetinterval = 

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

group = 2
channel => 32-46
channel => 48-62

thanks
arun
___
--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

Re: [asterisk-users] USB Cordless

[Anselm Martin Hoffmeister]

Am Montag, den 16.07.2007, 09:44 -0500 schrieb Jeremy Mann:
> Does anyone know if X-Ten or SJPhone support multiple cordless
> handsets for multiple lines?  I have an office with multiple roaming
> users(nurses) that are in and out.  I’d like to provide them
> telephones, and my idea is to have a PC sitting in a corner somewhere
> running a softphone client.  When a nurse comes in she just picks up
> any available handset(anywhere from 2-5 per office) and starts
> calling.  Each handset would be labeled with their extension so that
> if any inbound calls came to them they’d be able to let the
> receptionist know their extension.
> 
>  
> 
> Any ideas?

I personally would prefer giving them "real phones", be that a
combination DECT/ATA or WLAN phones. If you shop around, DECT/ATA will
probably be the less steep pricing. In the long run this would probably
be easier to keep running. If you talk 4 phones, you might calculate 2
ATAs of 2 ports each, plus 4 DECT thingies, summing up about 2*75€ +
4*25€, which means the rather cheap devices, which expectedly will
nevertheless look better than the wireless USB things. And then, get a
PC and 4 wireless sound devices for 250 bucks...

But I do not want to lack answering your question: I know for sure that
some softphones allow to select a certain sound channel / sound
controller. Take the Linux softphone ekiga as example. If you run
several of those with different configuration files and on different
port numbers (this will most probably be possible, although it might
turn out a nightmare to configure), that might get the job done.

I do not know wether there is a softphone that uses different sound
channels for different "lines", but I doubt it - it would be rather
counterintuitive to have a single "phone" software, but several
handsets; rather several instances would fulfill the "multiple phone on
one desk" computer screen analogy.

You might also have some trouble with the keys on your wireless phones
in a multiple-softphone scenario - depending on how the OS handles
those, they might be handed to the window in focus. I have plainly no
idea how this could possibly work with a phone hardware <-> softphone
mapping without royally screwing up.

> Also, is it possible to transfer a call directly to someone’s VM(if
> they are out of the office) bypassing their extension?  If so, could
> someone post the asterisk logic behind the extension setup?  I don’t
> want anything too complex(like setting the DND or phone to busy).

I want to describe a scenario, and you can decide wether that is too
complex ;-)

Let us assume your asterisk has two internal number plan ranges
available for the project, being 23XX and 4XXX. Let us further assume
that all the ATAs live in the 23XX range and will be called out of the
context [internal], like

[internal]
exten => 2300,1,Dial(SIP/device2300,60)
exten => 2300,2,Hangup()
exten => 2301,1,Dial(SIP/device2301,60)
...

(or, if your devices are named reasonably in sip.conf, you might get
away with)
exten => _23XX,1,Dial(SIP/device${EXTEN},60)
exten => _23XX,2,Hangup()

So those numbers end up calling a specific DECT phone, but you would not
know which nurse to reach on which phone, unless she told you beforehand
that she "just picked up phone 56" resulting in phone number 2356.

To get around that, every nurse gets assigned a personal number from the
4XXX range that will "follow her" or go to voicemail. You could make use
of the Asterisk Database, like this:

[internal]
exten => _4XXX,1,Set(CURRENTPHONE=${DB(nurse/${EXTEN})})
exten => _4XXX,2,GotoIf($["${CURRENTPHONE:1}" = ""]?4)
exten => _4XXX,3,Dial(SIP/device${CURRENTPHONE},60)
exten => _4XXX,4,VoiceMail(${EXTEN})
exten => _4XXX,5,Hangup

So if the nurse is not "logged in" the call will go to voicemail
immediately.

Instead of calling the receptionist "Hi Linda, I'm on phone 56 today"
she would keep her "4113" for "all times".

The reason I chose two-digit DECT phone numbers and three-digit nurse
numbers is that there are usually more nurses than phones :-) Anyway
a somehow competent receptionist would be able to deal with a static
personell number list better than dynamic phone numbers changing twice
every day.

Of course the nurse would need to tell the phone system where she
currently is, like by picking a phone and dialling her own code number,
plus *1 (provided CALLERID is working correctly) - or her own number
plus *0 to log off. Mind, you could also have an IVR available (on 777
or whatever internal number suits you) that greets the caller, asks for
the nurse's number and her PIN and wether she is coming or going.

[internal]
exten => _4XXX*1,1,GotoIf($["${CALLERID(num):0:2}" = "23"]?2:100)
exten => _4XXX*1,2,Set(DB(nurse/${EXTEN:0:4})=${CALLERID(num)})
exten => _4XXX*1,3,Playback(nurse-registered-thank-you)
exten => _4XXX*1,4,Hangup
exten => _4XXX*1,100,Playback(not-possible-from-this-phone)
exten => _4XXX*1,101,Hangup

exten => _4XXX*0,1,Set(DB(nurse/${EXTEN:0:4})=0)
exten 

Re: [asterisk-users] [SOLVED] PlayDTMF and Asterisk Manager

[lemmel lemmel]

>Here is my piece of generosity of the work - and it's not even my work.
>7xx to login, *7xx to logout.
Thanks for the code, it won't be usefull though (As I suspected it don't 
perform what I wanted to do).

This piece of code may be usefull for someone else :-).

_
Avec Windows Live OneCare éliminez tous les virus de votre PC ! 
http://www.windowslive.fr/liveonecare/default.asp


___
--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

Re: [asterisk-users] asterisk 1.4 and gnugk with ooh323

[Dovid B]

What output do you get from the CLI ?

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

Sent: Tuesday, July 17, 2007 9:59 AM
Subject: Re: [asterisk-users] asterisk 1.4 and gnugk with ooh323


> Hi,
> Thanks for the answer.
> Yes, I think both channels are built.
> I see following messages at startup:
> Parsing '/etc/asterisk/h323.conf': Found
> Creating H.323 Endpoint
> Parsing '/etc/asterisk/users.conf': Found
> Registered channel type 'H323' (The NuFone Network's Open H.323 Channel 
> Driver)
> H.323 listener started chan_h323.so =>
> The NuFone Network's OpenH323 Channel Driver
>
> [and few lines after]
> Parsing '/etc/asterisk/ooh323.conf': Found
> Registered channel type 'OOH323' (Objective Systems H323 Channel Driver)
> chan_ooh323.so => (^[[33;40mObjective Systems H323 Channel^[[0;37;40m)
>
> I am trying to use only ooh323 (removing the h323.conf file) but still
> does not work.
> I believe it is a configuration problem.
> Can someone point me to the correct configuration or example? The ones
> I have found so far did not help me much.
>
> Thanks
>
> 2007/7/16, Dovid B <[EMAIL PROTECTED]>:
>> There are different h323 channel drivers. You seem to have built both. If
>> you try to dial using Dial(h323/.) then you need h323.conf. If you 
>> try
>> dialing using Dial(ooh323/.) then you need ooh323.conf. For me
>> personally the ooh323 (can't remember the name of it  - sleep 
>> deprivation)
>> works better for me.
>>
>> - Original Message -
>> From: "yonoko molomo" <[EMAIL PROTECTED]>
>> To: 
>> Sent: Monday, July 16, 2007 6:13 PM
>> Subject: [asterisk-users] asterisk 1.4 and gnugk with ooh323
>>
>>
>> > Hello all,
>> > I have seen some people asking how to configure asterisk to work with
>> > h323 but i did not manage to do fix it yet (i am not an asterisk
>> > expert).
>> > Can someone help me configuring asterisk?
>> > It is already compiled asterisk 1.4.5 with H323 support.
>> > Everything looks fine.
>> > Then i understand i need to configure several files:
>> > -sip.conf
>> > -ooh323.conf
>> > -extensions.conf
>> >
>> > do i also need to configure the h323.conf?
>> >
>> > i want asterisk and gnugk in the same machine (lets say ip 
>> > 192.168.0.10).
>> > then sip_client at 192.168.0.100 (telephone number assigned for
>> > instance 100) and H323_client at 192.168.0.200 (telephone number
>> > assigned 200)
>> >
>> > how do i configure the files to do this?
>> >
>> > if i type show channeltypes i see:
>> >
>> > TypeDescription  Devicestate
>> > Indications  Transfer
>> > --  ---  ---
>> > ---  
>> > OOH323  Objective Systems H323 Channel Driverno   yes
>> >no
>> > SIP Session Initiation Protocol (SIP)yes  yes
>> >yes
>> > H323The NuFone Network's Open H.323 Channel  no   yes
>> >no
>> >
>> > What does devicestate and trasfer mean? it is set to "no". is this ok?
>> >
>> >
>> > if someone could show me how to  configure the files to do this i
>> > would be very grateful.
>> >
>> > thanks!
>> >
>> > ___
>> > --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
>> >
>>
>>
>>
>> ___
>> --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
>>
>
> ___
> --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
> 



___
--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

Re: [asterisk-users] tT in callparking

[bbodin01]

Transfert authorization.

Le 14 juil. 07 à 22:20, bilal ghayyad a écrit :


Hi List;

[incoming]
include => parkedcalls
exten=103,1,Dial(SIP/Bob,,tT)
exten=104,1,Dial(SIP/Charlie,,tT)

When we use tT and when we use t alone or T alone, I
know this for call parking, but I do not know what the
tT does?

Regards
Bilal



__ 
__

Sucker-punch spam with award-winning protection.
Try the free Yahoo! Mail Beta.
http://advision.webevents.yahoo.com/mailbeta/features_spam.html

___
--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


Bruno BODIN
[EMAIL PROTECTED]



___
--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

Re: [asterisk-users] I want to record each phone call

[Keshav K.]

Hi,
 For  recording your each phone call use this in your all dial-plan in 
extension.conf
By these lines there will a time stamping in your all call, and call will be 
saved in date directory. Choose your GMT and Time , accordingly.
 
 exten => 
1XXX,1,Set(CALLFILENAME=${EXTEN}-${CALLERID(num)}-${STRFTIME(${EPOCH},GMT-5.5,%d%b%Y)}-${STRFTIME(${EPOCH},GMT-5.5,%H%M%S)})
 exten => 
1XXX,n,Monitor(wav,/home/recording${STRFTIME(${EPOCH},Asia/Calcutta,%Y%m%d)}/${CALLFILENAME},m)
 
 
 Regards,
 Keshav
 



Ronald Wiplinger <[EMAIL PROTECTED]> wrote: 1. Instead of using *1 (automon) I 
need to record each phone call at a 
certain * box.

2. While already talking about this. I want to autodelete with cron at 2 
am in the morning all recordings which are older than 50 hours! How can 
I do that?

bye

Ronald

___
--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


   
-
Looking for a deal? Find great prices on flights and hotels with Yahoo! 
FareChase.___
--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

Re: [asterisk-users] TimeStamp a Recording

[Keshav K.]

Hi,
For timestamp a recording you can use this...

exten => 
1XXX,1,Set(CALLFILENAME=${EXTEN}-${CALLERID(num)}-${STRFTIME(${EPOCH},GMT-5.5,%d%b%Y)}-${STRFTIME(${EPOCH},GMT-5.5,%H%M%S)})
exten => 
1XXX,n,Monitor(wav,/home/recording${STRFTIME(${EPOCH},Asia/Calcutta,%Y%m%d)}/${CALLFILENAME},m)


Regards,
Keshav


Jared Smith <[EMAIL PROTECTED]> wrote: On Sat, 2007-07-14 at 23:31 -0400, 
Forrest Beck wrote:
> Has anyone come up with to timestamp a Recording?  I am using a pretty
> simple dialplan to record a audio file for a hotline.  I'd like to
> store the date and time it was recorded somewhere, Ast DB or MySQL DB.

In that case, the ${EPOCH} channel variable is your friend... it should
always contain the current system time, in Unix epoch format.  (In
essence, it contains the number of seconds since Jan 1, 1970.) 

You can also use the STAT() dialplan function to read the file creation
(or modification) date and time.

>  Then when the audio file is played back to a caller, the system will
> say something like.

The SayUnixTime() application will read back the time to the caller, and
allow you to choose which format the time is read back in.


-- 
Jared Smith
Community Relations Manager
Digium, Inc.


___
--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


 
-
TV dinner still cooling?
Check out "Tonight's Picks" on Yahoo! TV.___
--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

Re: [asterisk-users] asterisk 1.4 and gnugk with ooh323

[yonoko molomo]

Hi,
Thanks for the answer.
Yes, I think both channels are built.
I see following messages at startup:
 Parsing '/etc/asterisk/h323.conf': Found
Creating H.323 Endpoint
 Parsing '/etc/asterisk/users.conf': Found
 Registered channel type 'H323' (The NuFone Network's Open H.323 Channel Driver)
 H.323 listener started chan_h323.so =>
 The NuFone Network's OpenH323 Channel Driver

[and few lines after]
 Parsing '/etc/asterisk/ooh323.conf': Found
 Registered channel type 'OOH323' (Objective Systems H323 Channel Driver)
chan_ooh323.so => (^[[33;40mObjective Systems H323 Channel^[[0;37;40m)

I am trying to use only ooh323 (removing the h323.conf file) but still
does not work.
I believe it is a configuration problem.
Can someone point me to the correct configuration or example? The ones
I have found so far did not help me much.

Thanks

2007/7/16, Dovid B <[EMAIL PROTECTED]>:
> There are different h323 channel drivers. You seem to have built both. If
> you try to dial using Dial(h323/.) then you need h323.conf. If you try
> dialing using Dial(ooh323/.) then you need ooh323.conf. For me
> personally the ooh323 (can't remember the name of it  - sleep deprivation)
> works better for me.
>
> - Original Message -
> From: "yonoko molomo" <[EMAIL PROTECTED]>
> To: 
> Sent: Monday, July 16, 2007 6:13 PM
> Subject: [asterisk-users] asterisk 1.4 and gnugk with ooh323
>
>
> > Hello all,
> > I have seen some people asking how to configure asterisk to work with
> > h323 but i did not manage to do fix it yet (i am not an asterisk
> > expert).
> > Can someone help me configuring asterisk?
> > It is already compiled asterisk 1.4.5 with H323 support.
> > Everything looks fine.
> > Then i understand i need to configure several files:
> > -sip.conf
> > -ooh323.conf
> > -extensions.conf
> >
> > do i also need to configure the h323.conf?
> >
> > i want asterisk and gnugk in the same machine (lets say ip 192.168.0.10).
> > then sip_client at 192.168.0.100 (telephone number assigned for
> > instance 100) and H323_client at 192.168.0.200 (telephone number
> > assigned 200)
> >
> > how do i configure the files to do this?
> >
> > if i type show channeltypes i see:
> >
> > TypeDescription  Devicestate
> > Indications  Transfer
> > --  ---  ---
> > ---  
> > OOH323  Objective Systems H323 Channel Driverno   yes
> >no
> > SIP Session Initiation Protocol (SIP)yes  yes
> >yes
> > H323The NuFone Network's Open H.323 Channel  no   yes
> >no
> >
> > What does devicestate and trasfer mean? it is set to "no". is this ok?
> >
> >
> > if someone could show me how to  configure the files to do this i
> > would be very grateful.
> >
> > thanks!
> >
> > ___
> > --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
> >
>
>
>
> ___
> --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
>

___
--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