[asterisk-users] Channels ID / Soft Hang Up
Lees, James (UK)
James.Lees at baesystems.com
Thu Jan 17 09:33:06 CST 2008
Hello,
I am wanting to close a specific channel for example;
SofthangUp(SIP/EXTEN-UNIQUEID) but the problem is the channel is
assigned a unique id as well.
The need fits into the idea of receiving a call from a higher status
user and thus closing a specific channel to allow the higher priority
call to route through the dial plan to the freed extension.
Any ideas welcome.
Many thanks
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Today's Topics:
1. Re: [IAX] Up-to-date list of soft- and hardphones?
(Gordon Henderson)
2. Re: Can DB() use SQLite instead of BerkeleyDB? (Tilghman Lesher)
3. Re: WARNING[31046]: chan_sip.c:4978 process_sdp: Unable to
lookup host in c= line, 'IN IP4 100101' (Andrew Joakimsen)
4. Re: Digium Part#'s (Was: Difference between TE121 and TE122)
(Kevin P. Fleming)
5. asterisk to mysql database! (Naveen Palani)
6. Re: asterisk to mysql database! (Simon Elliston Ball)
7. Asterisk 1.4.17 and RXFAX via T38 (Robert Moskowitz)
8. Re: Unable to open master device '/dev/zap/ctl' (Chris Bagnall)
9. Re: [IAX] Up-to-date list of soft- and hardphones? (Vincent)
10. Re: Can DB() use SQLite instead of BerkeleyDB? (Vincent)
11. Re: asterisk to mysql database! (Tilghman Lesher)
12. Re: [IAX] Up-to-date list of soft- and hardphones? (Tim H. Panton)
13. HDLC errors (Steven)
14. Re: HDLC errors (Russell Bryant)
15. AddQueueMember and Flash Operator Panel (jason at mhonetworks.com)
16. Re: HDLC errors (Steve Totaro)
17. Anyone Using a Dell PowerEdge T105 in Production (Steve Totaro)
18. Problem with a channel (Ruben Zamora)
19. Re: HDLC errors (Andrew Joakimsen)
20. IMAP client in asterisk not trying to contact IMAP server
(KodaK)
21. Asterisk Now Beta 6 and CISCO IP 7910 (jason at mhonetworks.com)
22. Re: Anyone Using a Dell PowerEdge T105 in Production
(Erik Anderson)
23. Re: Anyone Using a Dell PowerEdge T105 in Production
(Steve Totaro)
24. Asterisk on ClarkConnect (shadowym)
25. Re: Unable to open master device '/dev/zap/ctl' (Walter Willis)
26. Re: Anyone Using a Dell PowerEdge T105 in Production
(Erik Anderson)
----------------------------------------------------------------------
Message: 1
Date: Wed, 16 Jan 2008 18:08:23 +0000 (GMT)
From: Gordon Henderson <gordon+asterisk at drogon.net>
Subject: Re: [asterisk-users] [IAX] Up-to-date list of soft- and
hardphones?
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <Pine.LNX.4.64.0801161741420.11251 at lion.drogon.net>
Content-Type: TEXT/PLAIN; charset=US-ASCII; format=flowed
On Wed, 16 Jan 2008, Vincent wrote:
> Hello
>
> There's a lot of information on VoIP at www.voip-info.org ...
> but there's also a lot of outdated information there as well :-/
>
> Since SIP is a pain to use when NAT is involved, especially when both
> the Asterisk server and the remote phones are behind NAT... I'd like
> to try IAX to see how it works and if it solves the issue.
>
> I'd like to start with a softphone (Windows only), and then, if tests
> prove successfully, buy a hardphone. What would be your
> recommendations?
IDEFISK or Zoiper as it's called now.
However, you'll need to do similar things to your asterisk box & router
if it's behind NAT for IAX as you do for SIP. (You will need a static IP
address on the NAT router and port-forward 4569 to the asterisk box,
just as you'd port-forward 5060 and 10000-20000 for SIP)
And a SIP phone behind a NAT router is also solvable if it supports
STUN.
I know that SIP behind NAT isn't perfect, but with care, it's very
usable and workable. I have many installations doing just this, as I'm
sure many others on the list have too.
Gordon
------------------------------
Message: 2
Date: Wed, 16 Jan 2008 12:10:35 -0600
From: Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
Subject: Re: [asterisk-users] Can DB() use SQLite instead of
BerkeleyDB?
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <200801161210.35615.tilghman at mail.jeffandtilghman.com>
Content-Type: text/plain; charset="iso-8859-1"
On Wednesday 16 January 2008 10:02:12 Vincent wrote:
> Before I bother calling a PHP script through AGI just to read a number
> and rewrite the CID name... I was wondering if Asterisk could be
> configured so that DB() uses a SQL server instead of the usual
> BerkeleyDB?
No, it cannot. You could use func_odbc to formulate your own queries,
though.
--
Tilghman
------------------------------
Message: 3
Date: Wed, 16 Jan 2008 14:03:53 -0500
From: "Andrew Joakimsen" <joakimsen at gmail.com>
Subject: Re: [asterisk-users] WARNING[31046]: chan_sip.c:4978
process_sdp: Unable to lookup host in c= line, 'IN IP4
100101'
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Message-ID:
<23fd749a0801161103g29de5732jc88de1b58fa6b49c at mail.gmail.com>
Content-Type: text/plain; charset=UTF-8
Did you look at the trace I send you in email? Because in each request
there are two IN IP lines I think Asterisk should only interpret the
first one,
On Jan 16, 2008 2:40 AM, Johansson Olle E <oej at edvina.net> wrote:
>
> 16 jan 2008 kl. 04.43 skrev Andrew Joakimsen:
>
> > Well can you offer some explanation why T.38 faxing worked for
months
> > and then one day stopped working?
> You are asking the wrong forum. Your device is clearly sending a bad
> SDP. Ask the vendor of that device.
>
> /O
> >
> >
> > Using both Linksys & Audiocodes (yuck) ATA. The first second of the
> > fax tone is heard and then the T.38 switchover is attempted and the
> > call drops with said error.
> >
>
> >
> >
> > On Jan 15, 2008 6:25 PM, Mark Michelson <mmichelson at digium.com>
wrote:
> >>
> >> Andrew Joakimsen wrote:
> >>> Anyone else have issues with T.38 where the call drops after T.38
is
> >>> attempted to be negotiated, with a message like the below?
> >>>
> >>> WARNING[31046]: chan_sip.c:4978 process_sdp: Unable to lookup host
> >>> in
> >>> c= line, 'IN IP4 100101'
> >>
> >> The problem is that 100101 is neither a valid IPv4 address nor a
> >> fully-qualified
> >> domain name.
> >>
> >> _______________________________________________
> >> -- 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
>
> ---
> * Olle E Johansson - oej at edvina.net
> * Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden
>
>
>
>
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
------------------------------
Message: 4
Date: Wed, 16 Jan 2008 13:04:01 -0600
From: "Kevin P. Fleming" <kpfleming at digium.com>
Subject: Re: [asterisk-users] Digium Part#'s (Was: Difference between
TE121 and TE122)
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <478E5521.3050209 at digium.com>
Content-Type: text/plain; charset=ISO-8859-1
Dave Fullerton wrote:
> If you want to know what a card's capabilities are you're better off
> just memorizing each part number. Maybe there's a scheme I'm just not
> capable of understanding here.
We gave up (intentionally) on trying to have model numbers that
reflected all the capabilities of each card, because they would turn
into unintelligible (and unmemorizable) part numbers. We now have 'part
numbers' that represent a given card with the options it was ordered
with (analog module(s), echo canceler, etc.), and we've stopped trying
to use suffixes to indicate bus type and instead just use a different
model number.
This why the TE122 (which replaced the TE120P) no longer has a 'P'
suffix; the PCI-Express version is a different model number entirely.
With that said, for some reason our marketing department decided to
change the *prefix* for PCI-Express analog cards from TDM to AEX, but
they still follow the rest of the model naming scheme (no suffix letter
and no different model numbers that indicate included optional modules).
--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)
------------------------------
Message: 5
Date: Wed, 16 Jan 2008 13:11:26 -0600
From: Naveen Palani <naveenp at quinnox.com>
Subject: [asterisk-users] asterisk to mysql database!
To: <asterisk-users at lists.digium.com>
Message-ID: <00ed01c85873$979cfb30$2505280a at quinnox.corp>
Content-Type: text/plain; charset="utf-8"
Hello,
Is there a possibility to connect from asterisk to mysql database
without the interface application like Ruby or PHP.
If i can connect to mysql database from asterisk, i can update the
database for manipulations.
Appreciate your response.
Regards,
Naveen.Palani
________________________________
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Message: 6
Date: Wed, 16 Jan 2008 19:29:20 +0000
From: Simon Elliston Ball <simon at simonellistonball.com>
Subject: Re: [asterisk-users] asterisk to mysql database!
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID:
<8EAED02D-C636-48A9-8A5C-D74918A5DBD0 at simonellistonball.com>
Content-Type: text/plain; charset=UTF-8; format=flowed; delsp=yes
Try:
http://www.voip-info.org/wiki/view/Mysql
and the links thereon.
simon
Simon Elliston Ball
simon at simonellistonball.com
On 16 Jan 2008, at 19:11, Naveen Palani wrote:
> Hello,
>
> Is there a possibility to connect from asterisk to mysql database
> without the interface application like Ruby or PHP.
>
> If i can connect to mysql database from asterisk, i can update the
> database for manipulations.
>
> Appreciate your response.
> Regards,
> Naveen.Palani
>
>
> ?Quinnox, a global IT services company prides itself on its SEI-CMM
> Level 5, ISO?9001:2000 assessed delivery processes and provides
> solutions in areas of E-Business, ERP, Application Management
> Services, and EAI to customers in BFSI, Manufacturing, Retail,
> Telecom and Healthcare sector, powered by our Global Delivery
> Model.?
>
> This e-mail and any attached files are confidential, proprietary,
> and may also be legally privileged information, and are intended
> solely for the use of the individual or entity to whom they are
> addressed. If you are not the intended recipient of this e-mail,
> please send it back to the person who sent it to you and delete the
> e-mail and any attached files and destroy any copies of it; you may
> call us immediately at + 91 22 2829 0100 or email us at
systems at quinnox.com
>
> Quinnox Consultancy Services and/or any of its sister companies owns
> no responsibility for the views presented in the e-mail and any
> attached files unless the sender mentions so, with due authority of
> Quinnox Consultancy Services.
>
> Unauthorized reading, reproduction, publication, use, dissemination,
> forwarding, printing or copying of this e-mail and its attachments
> is prohibited.
> We have checked this message for any known viruses; however we
> decline any liability, in case of any damage caused by a non-
> detected virus.
>
> For more details about our company, visit http://www.Quinnox.com
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
------------------------------
Message: 7
Date: Wed, 16 Jan 2008 15:24:04 -0500
From: Robert Moskowitz <rgm at htt-consult.com>
Subject: [asterisk-users] Asterisk 1.4.17 and RXFAX via T38
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <478E67E4.8070405 at htt-consult.com>
Content-Type: text/plain; charset="US-ASCII"; format="flowed"
I was pointed to the following:
http://asteriskforum.ru/viewtopic.php?t=1761
It is in Russian, which I don't speak, but it references an Asterisk
patch.
Is this patch in 1.4.17?
Is it scheduled to be in 1.4.18 (or whatever ships after 1.4.17?)
Anyone work with this?
------------------------------
Message: 8
Date: Wed, 16 Jan 2008 20:39:50 -0000
From: "Chris Bagnall" <lists at minotaur.cc>
Subject: Re: [asterisk-users] Unable to open master device
'/dev/zap/ctl'
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
<asterisk-users at lists.digium.com>
Message-ID: <044901c8587f$f2d64a40$d882dec0$@cc>
Content-Type: text/plain; charset="UTF-8"
Make sure asterisk is in the "dialout" group in /etc/passwd
The default gentoo ebuild of zaptel creates /dev/zap/* with group
dialout, and if you're using the gentoo ebuild of asterisk, it'll run as
asterisk:asterisk, so you need to make sure asterisk is a member of the
dialout goup otherwise it'll never be able to access /dev/zap/*
FWIW, as a fellow Gentoo user, 2006.1 is a bit dated, and you'd be well
worth updating to 2007.0 if you can spare the time - it'll save you a
lot of messing around with gcc versions etc. later down the line.
Regards,
Chris
--
C.M. Bagnall, Director, Minotaur I.T. Limited
For full contact details visit http://www.minotaur.it
This email is made from 100% recycled electrons
------------------------------
Message: 9
Date: Wed, 16 Jan 2008 23:01:55 +0100
From: Vincent <vincent.delporte at bigfoot.com>
Subject: Re: [asterisk-users] [IAX] Up-to-date list of soft- and
hardphones?
To: asterisk-users at lists.digium.com
Message-ID: <vgvso3974582cd4nkfu42m8hsj2han8j0r at 4ax.com>
Content-Type: text/plain; charset=us-ascii
On Wed, 16 Jan 2008 18:08:23 +0000 (GMT), Gordon Henderson
<gordon+asterisk at drogon.net> wrote:
>However, you'll need to do similar things to your asterisk box & router
if
>it's behind NAT for IAX as you do for SIP. (You will need a static IP
>address on the NAT router and port-forward 4569 to the asterisk box,
just
>as you'd port-forward 5060 and 10000-20000 for SIP)
Am I wrong to understand that IAX only needs one port, TCP4569 by
default? So I only need one port for each phone, while SIP requires at
least 3 (SIP, and one RTP each way)?
>And a SIP phone behind a NAT router is also solvable if it supports
STUN.
But not all NAT routers support STUN, ie. keeping UDP ports open so
that incoming packets can make it.
>I know that SIP behind NAT isn't perfect, but with care, it's very
usable
>and workable
But unless I'm mistaken, when NAT is involved, canreinvite must be set
to no, ie. all RTP packets must go through Asterisk instead of flowing
from one phone to the other?
Thanks guys.
------------------------------
Message: 10
Date: Wed, 16 Jan 2008 23:22:10 +0100
From: Vincent <vincent.delporte at bigfoot.com>
Subject: Re: [asterisk-users] Can DB() use SQLite instead of
BerkeleyDB?
To: asterisk-users at lists.digium.com
Message-ID: <tq0to3dnvouv60qm8osiklocbpso748f6d at 4ax.com>
Content-Type: text/plain; charset=us-ascii
On Wed, 16 Jan 2008 12:10:35 -0600, Tilghman Lesher
<tilghman at mail.jeffandtilghman.com> wrote:
>No, it cannot. You could use func_odbc to formulate your own queries,
>though.
Thanks. I don't like ODBC, but if it's stable and not a pain to
install/use, that could be the solution.
Otherwise, there's a new solution to use MySQL:
http://www.voip-info.org/wiki/view/Asterisk+cmd+MYSQL
------------------------------
Message: 11
Date: Wed, 16 Jan 2008 16:26:03 -0600
From: Tilghman Lesher <tilghman at mail.jeffandtilghman.com>
Subject: Re: [asterisk-users] asterisk to mysql database!
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <200801161626.03891.tilghman at mail.jeffandtilghman.com>
Content-Type: text/plain; charset="utf-8"
On Wednesday 16 January 2008 13:29:20 Simon Elliston Ball wrote:
> Simon Elliston Ball
> simon at simonellistonball.com
>
> On 16 Jan 2008, at 19:11, Naveen Palani wrote:
> > Hello,
> >
> > Is there a possibility to connect from asterisk to mysql database
> > without the interface application like Ruby or PHP.
> >
> > If i can connect to mysql database from asterisk, i can update the
> > database for manipulations.
>
> Try:
> http://www.voip-info.org/wiki/view/Mysql
>
> and the links thereon.
Or read configs/func_odbc.conf.sample.
--
Tilghman
------------------------------
Message: 12
Date: Wed, 16 Jan 2008 22:45:59 +0000 (GMT)
From: "Tim H. Panton" <thp at westhawk.co.uk>
Subject: Re: [asterisk-users] [IAX] Up-to-date list of soft- and
hardphones?
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <28266313.85901200523559911.JavaMail.root at zimbra>
Content-Type: text/plain; charset=utf-8
----- Original Message -----
From: "Vincent" <vincent.delporte at bigfoot.com>
To: asterisk-users at lists.digium.com
Sent: 16 January 2008 22:01:55 o'clock (GMT) Europe/London
Subject: Re: [asterisk-users] [IAX] Up-to-date list of soft- and
hardphones?
On Wed, 16 Jan 2008 18:08:23 +0000 (GMT), Gordon Henderson
<gordon+asterisk at drogon.net> wrote:
>However, you'll need to do similar things to your asterisk box & router
if
>it's behind NAT for IAX as you do for SIP. (You will need a static IP
>address on the NAT router and port-forward 4569 to the asterisk box,
just
>as you'd port-forward 5060 and 10000-20000 for SIP)
Am I wrong to understand that IAX only needs one port, TCP4569 by
default? So I only need one port for each phone, while SIP requires at
least 3 (SIP, and one RTP each way)?
--------
That's UDP 4569.
Also, depending on your configuration, you may not need to do any port
forwarding
at the 'client' end. Just have all your phones send registrations
frequently
and your natting router will do the rest.
Tim.
------------------------------
Message: 13
Date: Wed, 16 Jan 2008 15:52:21 -0800
From: Steven <steven.kurylo at aviawest.com>
Subject: [asterisk-users] HDLC errors
To: asterisk-users at lists.digium.com
Message-ID: <478E98B5.70701 at aviawest.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
I'm running Asterisk 1.2.26.1 svn rev 79171 on Trixbox 2.2. libpri
1.2.7 and zaptel 1.2.22.1. The hardware is a HP dl360 single cpu with a
TE220B. The system load is below 0.10.
I moved the server into production, with one PRI, on Friday. On that
day we handled a couple thousand calls and I only saw one HDLC abort
message. On Saturday half the calls and two abort messages an hour
apart. On Sunday, after 1500 when there was only a couple calls, the
HDLC messages went crazy.
We're getting non-stop Abort messages, with Bad FCS thrown in about
every tenth message. They come in bunches, with short 10-30 second
breaks. Then every once and awhile there is an 30 minute break,
sometimes a 3 hour break. The messages seems completely separate from
system load. The system will be idle and get the messages and have no
messages when I load up dozens of calls on it (using call files to
complete calls)
After reading the mailing list and various websites (asteriskguru.com
has a couple articles), the first thing I did was look for IRQ
conflicts. The module for the usb bus (no usb devices attached) was on
the same IRQ. Disabling USB had no effect. zttool shows no IRQ misses.
The second PRI was installed on Monday, that day with only two calls,
the message came 11 times. Three times on Tuesday with no calls, then
late at night I loaded it up with calls for testing (having call files
call out on the second PRI to the first PRI) and no messages were
generated. Again today its had a few messages with only a couple calls.
I'm not sure what to try next, other than calling the telco and asking
them to check their equipment. Does any one have a suggestion before I
do that?
Thanks.
------------------------------
Message: 14
Date: Wed, 16 Jan 2008 18:07:20 -0600
From: Russell Bryant <russell at digium.com>
Subject: Re: [asterisk-users] HDLC errors
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <478E9C38.1010404 at digium.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Steven wrote:
> I'm not sure what to try next, other than calling the telco and asking
> them to check their equipment. Does any one have a suggestion before
I
> do that?
I have a suggestion. Have you contacted Digium technical support for
assistance
with resolving this issue?
--
Russell Bryant
Senior Software Engineer
Open Source Team Lead
Digium, Inc.
------------------------------
Message: 15
Date: Wed, 16 Jan 2008 17:11:44 -0700 (MST)
From: jason at mhonetworks.com
Subject: [asterisk-users] AddQueueMember and Flash Operator Panel
To: asterisk-users at lists.digium.com
Message-ID: <31281.64.58.4.46.1200528704.squirrel at b.mail.mho.net>
Content-Type: text/plain;charset=iso-8859-1
Hello users!
Recently I read that AgentCallbackLogin is going to be deprecated soon.
Wanting to set up a few callback type queues, I set them up as suggested
in queues-with-callback-members.txt.
I was able to set the queues up completely this way, however, I'm trying
to use Flash Operator Panel (aka AsterNIC) to monitor the agents' login
status. FOP monitors their status if I call AddQueueMember with their
actual interface (which, by the way, makes more sense to me than logging
them in via chan_local), and it even seems to work with
Local/${AGENT_EXTEN}@default. But if I use any context other than
"default" here, FOP doesn't recognize that the agent is logged in.
(The users' default context isn't even set to default, and it behaves
this
way even if the users' voicemail context is something else, so I am
guessing that is hard-coded in FOP somewhere.)
If I log them in from Local/${AGENT_EXTEN}@default, FOP works and the
agents get the calls, but then it's just dialing them directly - there
is
no way to increment OUTBOUND_GROUP or check the value of GROUP_COUNT.
As
a result, calls are routinely sent to agents who are already on the
phone,
which I don't want.
Obviously, the next reasonable solution would be to use some other
context
for the default context, and use [default] instead of [agents] for
incrementing OUTBOUND_GROUP and checking GROUP_COUNT, but I'm pretty
sure
this would break the functionality of AsteriskGUI almost completely, and
I'm trying to preserve as much of that as possible.
Am I missing something? Is there a way to make all of this work
together
without modifying some source code?
Thanks in advance!
Jason Burbage
jason at mhonetworks.com
------------------------------
Message: 16
Date: Wed, 16 Jan 2008 19:32:04 -0500
From: "Steve Totaro" <stotaro at totarotechnologies.com>
Subject: Re: [asterisk-users] HDLC errors
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Message-ID:
<ea18e54a0801161632v64fa986x30ea54bbc2b37714 at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
On Jan 16, 2008 7:07 PM, Russell Bryant <russell at digium.com> wrote:
> Steven wrote:
> > I'm not sure what to try next, other than calling the telco and
asking
> > them to check their equipment. Does any one have a suggestion
before I
> > do that?
>
> I have a suggestion. Have you contacted Digium technical support for
> assistance
> with resolving this issue?
>
> --
> Russell Bryant
> Senior Software Engineer
> Open Source Team Lead
> Digium, Inc.
>
Excellent suggestion. Make sure you can give them SSH access and screen
so
you can see what they are doing. Before that, check (remake) your T1
cables
and if it is punched down on a block, re-punch it.
Work with your telco as well. I call that burning the candle from both
ends. You said there were no errors looping port one to port two and
generating calls with call files. That may indicate a telco issue. I
usually open a ticket with the telco right away just in case so it can
be
escalated quicker if in fact it is the telco. Sometimes you have to be
a
jerk to these guys to get someone with half a brain to look into your
problem rather than blaming CPE (the easiest way to close their ticket
and
get you off the phone).
If Digium says it is the telco and the telco says it is your CPE
(Asterisk/Digium/Server/CPE wiring) then put them together on a
conference
call!
I am sure it won't come to that if it is truly a Digium/Asterisk issue.
They will take care of it.
Thanks,
Steve Totaro
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Message: 17
Date: Wed, 16 Jan 2008 19:39:34 -0500
From: "Steve Totaro" <stotaro at totarotechnologies.com>
Subject: [asterisk-users] Anyone Using a Dell PowerEdge T105 in
Production
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Message-ID:
<ea18e54a0801161639y7ba46044nb6e84a96d31ff4e7 at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
Unbeatable price for a low end Asterisk server (or any server for that
matter)
http://configure.us.dell.com/dellstore/config.aspx?c=us&cs=04&kc=6W300&l
=en&oc=bednv4k&s=bsd
I wonder if anyone has any experience with this box and Digium or
Sangoma
hardware? Any compatibility issues? If not, I might stock up on them.
Thanks,
Steve Totaro
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Message: 18
Date: Wed, 16 Jan 2008 18:42:53 -0600
From: "Ruben Zamora" <ruben.zamora at zys.com.mx>
Subject: [asterisk-users] Problem with a channel
To: <asterisk-users at lists.digium.com>
Message-ID: <003801c858a1$e5680e80$8001a8c0 at RUBEN>
Content-Type: text/plain; charset="us-ascii"
I have install a Server with Centos 1 TDM400: Asterisk 1.4.9, Zaptel
1.4.5
I having these problem :
Zap/2-1 is busy
Hangup ZAP/2-1
Everyone is busy/congested at this time (1:1/010)
Autofallthrough channel "SIP/202-b7b08ab0" Status is busy.
And then HANGUP.
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Message: 19
Date: Wed, 16 Jan 2008 19:44:54 -0500
From: "Andrew Joakimsen" <joakimsen at gmail.com>
Subject: Re: [asterisk-users] HDLC errors
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Message-ID:
<23fd749a0801161644m15b4c68akc3ef47502918d6a7 at mail.gmail.com>
Content-Type: text/plain; charset=UTF-8
Trixbox 2.2... I assume you are using the latest version. Normally I
will ignore messages from trixbox users because they ask kindergarten
stuff... but you seem to be knowledgeable and I'll assume you chose
trixbox to make your life easier when it comes to dealing with others
regarding the PBX.
I also assume the PRI is delivered via some sort of HDSL terminated at
an NIU ("SmartJack") Which is a box that will usually have 2 or 4
positions for line cards and 2 or 4 jacks marked "CPE1" etc....
usually at the bottom. Usually also you can look through the window at
the top and see various lights.
What is between the smartjack and your T1 card? What sort and length
of cable? Any splices? Punchdown or patch panels?
Also I'm not sure if Trixbox has this but ssh in and see if there is
an application called zttool. What are the statistics it is providing?
------------------------------
Message: 20
Date: Wed, 16 Jan 2008 18:54:47 -0600
From: KodaK <sakodak at gmail.com>
Subject: [asterisk-users] IMAP client in asterisk not trying to
contact IMAP server
To: asterisk-users at lists.digium.com
Message-ID:
<3cbf390d0801161654p37e72ce2if41bca5279e40754 at mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1
I'm trying to test IMAP in 1.4.17 and it appears to be not working.
I've compiled imap-2007 with the following on a CentOS 5 box:
make slx EXTRACFLAGS="-I/usr/include/openssl -fPIC"
and I've configured and compiled asterisk with the following:
./configure --with-imap=/usr/local/src/imap-2007
The compile and install went just fine, no warnings and no errors that I
saw.
However, when actually trying to use it, it doesn't appear that asterisk
is even
trying to use the local IMAP server.
The local IMAP server is dovecot, with a master password configured.
I've
tried plain and SHA auth, but from the logs I don't even see the
asterisk
master user trying to connect.
Here's my voicemail.conf:
[general]
imapserver=localhost
imapfolder=Inbox
;pollmailboxes=yes
;pollfreq=30
imapflags=notls
authuser=asttest
expungeonhangup=yes
authpassword=whatever
[default]
5252 => 5252,Test,5252 at localhost,,imapuser=5252
(I have also tried this line as:
5252 => 5252,Test,,,imapuser=5252
5252 => 5252,Test,5252 at localhost,,imapuser=5252|imappass=pass
5252 => 5252,Test,,,imapuser=5252|imappass=pass
all with and without the authuser and authpassword in the general
section.)
I can authenticate against the * server using 5252*asttest as the
username and
"whatever" as the password, which I'm lead to believe is how * will
try to connect.
(Also, the imap user 5252 exists and can receive mail.)
Is there something else I'm missing? Is there some other place in the
dial plan that
I have to say "use IMAP"? Is there some way to confirm that the imap
client
has been compiled in? Some hidden CLI command to debug it?
doing "grep -i imap /var/log/asterisk/*" gives absolutely no results.
I'm almost convinced that I've got something wrong in the configuration
because
I tried the latest SVN and I didn't see it hit the IMAP server, but it
also segfaulted
so who knows.
Any ideas at all? Am I missing something obvious that I'll find as
soon as I press
"send" and wish I hadn't sent the message?
Thanks,
--J(K)
------------------------------
Message: 21
Date: Wed, 16 Jan 2008 17:55:13 -0700 (MST)
From: jason at mhonetworks.com
Subject: [asterisk-users] Asterisk Now Beta 6 and CISCO IP 7910
To: "asterisk-users at lists.digium.com"
<asterisk-users at lists.digium.com>
Message-ID: <29497.64.58.4.46.1200531313.squirrel at b.mail.mho.net>
Content-Type: text/plain;charset=iso-8859-1
The phones are configured in the "Users" section of AsteriskGUI.
The bigger problem you'll have is that you probably also need to
replace/update the firmware on the 7910; by default they're configured
to
work with Cisco's CallManager software. Start with this link:
http://www.voip-info.org/wiki/index.php?page=Asterisk+phone+cisco+79xx
Hope that helps. Good luck!
Jason Burbage
------------------------------
Message: 22
Date: Wed, 16 Jan 2008 19:11:22 -0600
From: "Erik Anderson" <erikerik at gmail.com>
Subject: Re: [asterisk-users] Anyone Using a Dell PowerEdge T105 in
Production
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Message-ID:
<fc40260f0801161711t7cc395b6ob58b3ffa7385d490 at mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1
On Jan 16, 2008 6:39 PM, Steve Totaro <stotaro at totarotechnologies.com>
wrote:
> Unbeatable price for a low end Asterisk server (or any server for that
> matter)
>
>
http://configure.us.dell.com/dellstore/config.aspx?c=us&cs=04&kc=6W300&l
=en&oc=bednv4k&s=bsd
>
> I wonder if anyone has any experience with this box and Digium or
Sangoma
> hardware? Any compatibility issues? If not, I might stock up on
them.
Wow - that *is* a great price. I don't have any of this particular
box in production, but I do have 2 PowerEdge SC440s (one step up from
the T105) running asterisk along with Sangoma PRI cards. They're
working great. I really only have two issues with these low-end
servers:
1. You can't order 'em with RAID support. I'm getting around this by
using software RAID1 in linux, but I'd much prefer having a hardware
RAID controller.
2. The Dell DRAC remote management cards aren't compatible with these
low-end server motherboards. I've become *completely* addicted to the
DRAC cards on the high-end PowerEdges, to the point that I now refuse
to order a server without a DRAC card.
That said, I'm sure this server would run a small/medium asterisk
install just fine.
-Erik
------------------------------
Message: 23
Date: Wed, 16 Jan 2008 20:28:58 -0500
From: "Steve Totaro" <stotaro at totarotechnologies.com>
Subject: Re: [asterisk-users] Anyone Using a Dell PowerEdge T105 in
Production
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Message-ID:
<ea18e54a0801161728y4a699fe1u370855c6c74d9907 at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
On Jan 16, 2008 8:11 PM, Erik Anderson <erikerik at gmail.com> wrote:
> On Jan 16, 2008 6:39 PM, Steve Totaro <stotaro at totarotechnologies.com>
> wrote:
> > Unbeatable price for a low end Asterisk server (or any server for
that
> > matter)
> >
> >
>
http://configure.us.dell.com/dellstore/config.aspx?c=us&cs=04&kc=6W300&l
=en&oc=bednv4k&s=bsd
> >
> > I wonder if anyone has any experience with this box and Digium or
> Sangoma
> > hardware? Any compatibility issues? If not, I might stock up on
them.
>
> Wow - that *is* a great price. I don't have any of this particular
> box in production, but I do have 2 PowerEdge SC440s (one step up from
> the T105) running asterisk along with Sangoma PRI cards. They're
> working great. I really only have two issues with these low-end
> servers:
>
> 1. You can't order 'em with RAID support. I'm getting around this by
> using software RAID1 in linux, but I'd much prefer having a hardware
> RAID controller.
> 2. The Dell DRAC remote management cards aren't compatible with these
> low-end server motherboards. I've become *completely* addicted to the
> DRAC cards on the high-end PowerEdges, to the point that I now refuse
> to order a server without a DRAC card.
>
> That said, I'm sure this server would run a small/medium asterisk
> install just fine.
>
> -Erik
>
You can add the raid option for $199. I think I might pickup about ten
of
them at this price. I can always resell them as general purpose servers
or
even workstations if Asterisk/Zaptel/Linux does not like the boxen.
Thanks,
Steve Totaro
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Message: 24
Date: Wed, 16 Jan 2008 17:34:01 -0800
From: "shadowym" <shadowym at hotmail.com>
Subject: [asterisk-users] Asterisk on ClarkConnect
To: <asterisk-users at lists.digium.com>
Message-ID: <BAY102-DAV1820B1FFEC04637090B8AEDD410 at phx.gbl>
Content-Type: text/plain; charset="us-ascii"
Has anyone tried installing Asterisk on ClarkConnect? It looks like
ClarkConnect runs on RHEL so it should work if they haven't modified it
too
much.
It appears that ClarkConnect is working on adding Asterisk and
integrating
it into their GUI but until then I'd also be interested in trying to use
FreePBX.
Anyone?
------------------------------
Message: 25
Date: Wed, 16 Jan 2008 20:48:15 -0500
From: "Walter Willis" <walterwn at gmail.com>
Subject: Re: [asterisk-users] Unable to open master device
'/dev/zap/ctl'
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Message-ID:
<6b0bc7870801161748s7154377x408d87d092fbdbd4 at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
create nodes and links /proc/zap
On Jan 16, 2008 3:39 PM, Chris Bagnall <lists at minotaur.cc> wrote:
> Make sure asterisk is in the "dialout" group in /etc/passwd
>
> The default gentoo ebuild of zaptel creates /dev/zap/* with group
dialout,
> and if you're using the gentoo ebuild of asterisk, it'll run as
> asterisk:asterisk, so you need to make sure asterisk is a member of
the
> dialout goup otherwise it'll never be able to access /dev/zap/*
>
> FWIW, as a fellow Gentoo user, 2006.1 is a bit dated, and you'd be
well
> worth updating to 2007.0 if you can spare the time - it'll save you a
lot
> of messing around with gcc versions etc. later down the line.
>
> Regards,
>
> Chris
> --
> C.M. Bagnall, Director, Minotaur I.T. Limited
> For full contact details visit http://www.minotaur.it
> This email is made from 100% recycled electrons
>
>
>
>
> _______________________________________________
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Message: 26
Date: Wed, 16 Jan 2008 19:54:13 -0600
From: "Erik Anderson" <erikerik at gmail.com>
Subject: Re: [asterisk-users] Anyone Using a Dell PowerEdge T105 in
Production
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Message-ID:
<fc40260f0801161754v444e3a41v7ff9c91004fd10e9 at mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1
On Jan 16, 2008 7:28 PM, Steve Totaro <stotaro at totarotechnologies.com>
wrote:
>
> You can add the raid option for $199. I think I might pickup about
ten of
> them at this price. I can always resell them as general purpose
servers or
> even workstations if Asterisk/Zaptel/Linux does not like the boxen.
Ahh - nice. That wasn't an option when I ordered the SC440.
-erik
------------------------------
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