Re: [asterisk-users] Why Aastra uses 48V whereas other IP Phones use much less, i.e. 5-12V

[Julien Goodwin]

On Tue, Nov 21, 2006 at 08:57:44PM -0500, Zeeshan Zakaria arranged a set of 
bits into the following:
> Why Aastra phones use more electricity, i.e. 48VDC whereas other phones use
> much less, e.g. Grandstream and Linksys both use only 5VDC. I first thought it
> was because of PoE, but the ones with 5VDC also run fine on PoE. What is the
> difference in power consumption then?
The difference due to the different voltages would be < 1w. Many of the
commercial phones (Aastra, Polycom, Cisco) use 48 volt power supplies
as it lets them have a single power circuit for wall-warts and PoE
(Standard PoE is 48 volts).

Basic electrical theory (for DC) is that power == Watts, and Watts =
Volts * Amps, so the only real difference between a 5v input and a 48v
is that the 48v will use less current (although it might go through more
DC-DC convertors those are highly efficient these days)

Thanks,
Julien


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Re: [asterisk-users] Asterisk with cisco 7935

[Julien Goodwin]

On Thu, Sep 28, 2006 at 08:36:55AM -0400, Doug Lytle arranged a set of bits 
into the following:
> Julien Goodwin wrote:
> >Mine works fine with chan_sccp, and there's now SIP firmware out for it 
> >(which I
> >haven't tried)
> >
> >  
> Going into a conference room will cause Asterisk to segfault.  After 
That worked for me (as it was the only reason I've used my 7935
recently)

> dialing out twice, the 7935 stops responding to key presses.
Again, worked for me.

Are you running an at all recent version of chan_sccp?

> I'll have to look at the Cisco site for the SIP firmware.
It may be restricted, I still have some Cisco insiders that are able to
help with things like that.


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Re: [asterisk-users] Asterisk with cisco 7935

[Julien Goodwin]

On Tue, Sep 26, 2006 at 01:13:14PM -0500, Ryan Amos arranged a set of bits into 
the following:
> ="urn:schemas-microsoft-com:office:smarttags" xmlns="http://www.w3.org/TR/REC-
> html40">
> 
> 
> I spent quite a bit of time debugging the 7935/7936, and it is an issue inside
> the firmware that Cisco knows how to work around in CallManager. There are
> better conference phone options available, and development on chan_sccp is
> basically dead at this point anyway, so I dont see this one ever being fixed.
>  
> I would recommend a Polycom IP4000, its the exact same phone body but is much
> cheaper MSRP, and its SIP.

Mine works fine with chan_sccp, and there's now SIP firmware out for it (which I
haven't tried)

I'm also considering taking back chan_sccp to get it working with 1.4,
but can't do that until some of my contract work clears up.


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[Asterisk-Users] Any IP phones with pro-audio connections?

[Julien Goodwin]

Does anybody know of any IP phones (ideally SIP based) that have
interfaces to plug into a pro audio system (eg for phone interviews).

Something can probably be hacked up with a headset connector or the 1/8"
jacks on a 7970 but I'm wondering if there's something better out there.

Thanks,
Julien
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[Asterisk-Users] NEWS: SIP Firmware Available for Cisco 7970

[Julien Goodwin]

I've just recieved a copy of the new SIP firmware for the Cisco 7970,
those of you with Cisco accounts may wish to try it (shock horror I'm
sticking with SCCP).

This coincides with the release of v8 firmware for all Cisco phones (and
for those of you running Sergio's chan_sccp v8 works fine)

The firmware is now also (and for the 7970 SIP, only) distributed in
".cop" files, these are actually just tarballs (.tar.gz) with a new
name. The names are mangled, but relativly easy to figure out.

Please note that I will not give this firmware out, nor point people to
places where they may pirate it.

Thanks,
Julien


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Re: [Asterisk-Users] BAD echo problems with Sangoma and Telstra

[Julien Goodwin]

On Wed, Sep 28, 2005 at 12:09:46PM +1200, Andrew Thrift arranged a set of bits 
into the following:
> Hello,
> 
> We have an Asterisk 1.0.9 machine with a Sangoma A101 card fitted, it is 
> connected to a Telstra OnRamp E1 in Melbourne, Australia.  The problem 
> we are experiencing is extreme echo and clicking noises. 
> 
> These are only audible to the calling party, e.g. the person calling in 
> from the PSTN to Asterisk via the E1, the person on the SIP handset 
> cannot hear any echo or clicking, only the PSTN side can.
Hmm, have a few BRI's but no E1's on asterisk (just on an old samsung)
but we've never had echo troubles, even without an echo can. (Yes in
melbourne, both CBD and surrounds)

The clicking sounds like a clock issue, or not enough cpu. What are your
phones, maybe it's acoustic echo.

Are you seeing any odd log messages (with debug=10 and verbose=10)?


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[Asterisk-Users] USB ISDN

[Julien Goodwin]

Does anyone know of any USB ISDN adapters that work with Asterisk. My
gateway box is an old Compaq laptop (PIII 800, recently upgraded from a
Toshiba P120) and their are obviously no PCI slots. PCMCIA/Cardbus or
SIP gateway products are also an option.

Thanks,
Julien


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Re: [Asterisk-Users] Stupid hold music

[Julien Goodwin]

On 22/07/2005 1:58 PM, Mark Phillips wrote:
Does anyone have a collection of stupid hold music? Y'know, the sort of 
thing that would drive a person mad? Silly songs, repetative tunes etc?
Doesn't everybody, here's most of mine: (most shamelessly stolen from a 
discussion on a.s.r)


Annie Lennox: Waiting in Vain
Eurythmics: When Tomorrow Comes
Moody Blues: Go Now
PSB: Saturday Night Forever
Pink Floyd: Time
Tom Robinson: The Frozen Man
Eurythmics: Forever
Rolling Stones: Time Is On My Side
Tommy Tutone - 867 5309
Kim Wilde - 36580
Blondie: Hanging on the Telephone
ELO: Telephone Line
Empire Records - Money (that's what I want)
Pink Floyd: Money
Blondie: Call Me
ELO: Ma Ma Ma Belle

And as background on the voice menus:

Backman-Turner Overdrive: You Ain't Seen Nothing Yet
Queen: I Want to Break Free
Divine Comedy: The Certainty of Chance
B-52's: 6068-842
Tom Robinson: 2-4-6-8 Motorway
Queen: I'm going Slightly Mad

Background for annoucements of queue position:

Eurythmics: Would I Lie To You?
Tom Lehrer: New Math
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Re: [Asterisk-Users] CallerID/chan_sccp

[Julien Goodwin]

On 8/06/2005 11:37 PM, Sergio Chersovani wrote:

Joseph ha scritto:

When sending a call to a line defined on chan_sccp, there is an 
error on the console that says:


Jun  7 08:22:29 WARNING[3924]: sccp_channel.c:79
sccp_channel_send_callinfo: Incoming call SCCP/Line1-0008 doesn't
have CallerId name
  


Fixed, you can find the patch here
http://www.c-net.it/chan_sccp/
And this has been committed, should work through in about 5 hours 
(thanks sourceforge)

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Re: [Asterisk-Users] CallerID/chan_sccp

[Julien Goodwin]

On 7/06/2005 10:27 PM, Joseph wrote:
When sending a call to a line defined on chan_sccp, 
there is an error on the console that says:


Jun  7 08:22:29 WARNING[3924]: sccp_channel.c:79
sccp_channel_send_callinfo: Incoming call SCCP/Line1-0008 doesn't
have CallerId name

Is this because of the changes in the callerid name from stable to head?
Are you running chan_sccp CVS_HEAD? What should be the fix for this was 
committed last weekend. Also there's a different format for caller id in 
the conf file (not very well documented in it, I need to clean up the 
file soon).


Thanks,
Julien
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Re: [Asterisk-Users] chan_sccp / 7960: Messages key, line / speeddial keys

[Julien Goodwin]

On Sat, Jun 04, 2005 at 10:09:29AM +0200, Stefan Gofferje arranged a set of 
bits into the following:
> Hi folks,
> 
> there are some 3rd party patches available for chan_sccp which add a 
> feature and fix a few problems:
> 
> http://www.sineapps.com/news.php?rssid=726
> 
> sccp_cli.c.diff 
> Adds support for "sccp debug" in Asterisk CLI

A good starting point, needs levels, but I'll merge it with my config
debug patch and commit them soon.

> sccp_softkeys.c.diff 
> sccp_actions.c.diff 
> Changes wording of DND status text to "DnD is On" and "DnD is Off"
Was included with a cleanup patch last week
> Fixes speed dial buttons so that always line 1 is used (7960 can use 3 
> lines)
Finally committed the correct fix for this (just use the current line)

> chan_sccp.c.diff 
> 
> chan_sccp.h.diff 
> Associates "Messages" key on 79XX with a voicemail context. Adds 
> "voicemail" keyword to sccp.conf file.
I'd done VM button sipport a while ago, just was a bit buggy, this patch
has the same issue, it shouldn't be much work to fix it, and will
probably go in soon.

> Especially the behaviour of the speeddials is kinda annoying. The first 
> speeddial always starts immediately on line 1. The rest do nothing 
> without a line selected in advance.
> The patches are against the mayday build, so maybe, somebody of the 
> chan_sccp maintainers would care to include them?
They've been in the queue, but some more important fixes got this pushed
behind.


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[Asterisk-Users] Re: [Asterisk-Dev] Is there any SCCP patent issue?

[Julien Goodwin]

On Sat, Jun 04, 2005 at 10:59:35AM +0200, Harald Milz arranged a set of bits 
into the following:
> a fellow consultant who specializes in Cisco VoIP told me that Cisco
> patented SCCP, and that one is not supposed to use SCCP without paying
> Cisco royalties. 
> 
> Methinks that if there were an issue like that Digium would have been sued
> long ago. But IANAL. So - is there a legal issue with using SCCP in
> Asterisk, or using Cisco phones with Asterisk? 
None that I know of, and I KNOW that Cisco is aware of chan_sccp (
perhaps chan_skinny being in the tree might be diffrent).
However see the recent fuss over VM patents, so many of them are
incredibly stupid, but the problem is they cost too much to defend.

Julien,
chan_sccp project lead


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Re: [Asterisk-Users] Cisco 7960s and skinny

[Julien Goodwin]

On Thu, Jun 02, 2005 at 01:56:09PM +0600, Yusuf Iqbal arranged a set of bits 
into the following:
> 7910's with Chan_sccp are running well except some cases.
> 
> As usually Transfer, Msgs, Conf,Forward,Speed buttons are not working!
> Infact I have already knew that, those buttons will not work (from
> mailing list).
Messages will be done soon (code exists, just needs to be cleaned and tested), 
Transfer's currently
just another button for '#' (but will be fixed soon), Conf doesn't work,
speedials should work (Although it seems like people are having trouble
with them only on 7910's, don't have one to test with yet) and hold got
stable just on the weekend.

> It seems like the * console is very busy with messages constantly on
> it from the sccp clients/devices. Sometimes I get some errors.
> Specially when I tried to hangup a line then the following message
> comes.
Yep, they're my next task.

> ERROR[19583]: Erp, tried to hangup when we didn't have an active channel?!
That message should be ignored (in fact it should have been removed with
the fixups on the weekend)

> Some of the Phones go down after some times and I don't get the dial
> tone. I have to restart the * to get them onhook. The rest are okay.
Update to the latest CVS and see if it still continues to happen.

> Sometimes I get the following notice.
> 
> NOTICE[6773]: chan_sccp.c:219 handle_message: ConfigStatMessage for Device 6
That's just the phone registering, should be a debug message, another on
to fix.

Thanks,
Julien
chan_sccp project lead


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Re: [Asterisk-Users] 7960 / chan_sccp: Less than three lines / more than three speeddials possible?

[Julien Goodwin]

On Sat, May 28, 2005 at 01:51:56AM +0200, Stefan Gofferje arranged a set of 
bits into the following:
> Hi folks,
> 
> I'm trying the latest mayday chan_sccp and found it difficult to have 
> more than three speeddials. I have defined only one line but can't get 
> more than three SDs. On console, * says
> 
> May 28 01:44:12 ERROR[26933]: sccp_actions.c:252 
> sccp_handle_line_number: SCCP device SEP000 requested a line 
> configuration for unknown line 3
> May 28 01:44:12 ERROR[26933]: sccp_actions.c:252 
> sccp_handle_line_number: SCCP device SEP000  requested a line 
> configuration for unknown line 3
> May 28 01:44:12 ERROR[26933]: sccp_actions.c:252 
> sccp_handle_line_number: SCCP device SEP000 requested a line 
> configuration for unknown line 2
> May 28 01:44:12 ERROR[26933]: sccp_actions.c:252 
> sccp_handle_line_number: SCCP device SEP000 requested a line 
> configuration for unknown line 2
> 
> I also have read about chan_sccp proposedly supporting softkey templates.
> Any hints?

Yes, what you need to do is edit chan_sccp.h, and just edit the 7960
layout to whatever you want, you can also duplicate it to have several
layouts you can switch between in the config file.

Thanks,
Julien


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Re: [Asterisk-Users] OT: cisco ip phone security problem

[Julien Goodwin]

On Tue, May 24, 2005 at 10:58:17PM -0700, trixter http://www.0xdecafbad.com 
arranged a set of bits into the following:
> Since many on this list use cisco ip phones I thought they may find this
> information worthwhile to know

Just a further warning, I was informed last night of a few other known
issues that are much worse, crashing cisco phones is easy, the hole's
I'm hearing about are information disclosure. If you're running cisco
phones that are not behind a firewall you should be.


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[Asterisk-Users] [Announce] New chan_sccp release adds support for Cisco 7970

[Julien Goodwin]

A new chan_sccp release has just been uploaded which adds support for
the cisco 7970 (min version 6 firmware).

There is one currently known issue with the 7970 support, MWI doesn't
work, and only basic call functions have been tested.

I'd like to publicly thank three people who've helped a lot to get this
release out:
* Jared Mauch for an initial patch
* Adam Megacz for a hardware donation of a 7970 to test on
* Flexion for supplying some ethereal dumps of a 7970 registering to a
callmanager

The next aim for chan_sccp is to support the subscribe/notify features
in asterisk to give BLF type functionality to the 7970 and 7960+7914.

This release has been tested on all the phones I own:
* Cisco 12SP+
* Cisco 30VIP
* Cisco 7905
* Cisco 7940
* Cisco 7970

If I don't have one I can't test it. I'm especially looking for:
* Cisco 7910
* Cisco 7920
* Cisco 7935/36
* And any non-cisco sccp phones (the Kirk IP600 looks very intresting
and if there are any Australian resellers of this here give me a call)

Please see the chan_sccp donations page if you're able to help (Note
that I'm in Australia for those considering hardware donations, but am
willing to pay shipping).
http://chan-sccp.sourceforge.net/donation.html

Thanks,
Julien Goodwin
chan_sccp project lead


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Re: [Asterisk-Users] Cisco 7960s and skinny

[Julien Goodwin]

On Sat, Apr 30, 2005 at 01:00:18PM -0500, Andy Hamilton arranged a set of bits 
into the following:
> I'll be able to get back to you Sunday night about specifics; the
> phone is not where I am right now. Using chan_sccp, (I think November
> 2004 or so CVS Head) I know I can receive calls, place calls, etc. It
> is a rather low volume phone, so I don't know off hand about specific
> keys; I'll check those later.
Generally if the phone supports the function, and support is in
chan_sccp for that function it will work for all phones.

> Additionally, I have not yet tried a new copy from CVS.
 
> Occasionally, I think the chan_sccp driver blips out in Asterisk (it
> may be the phone; I've had it apart several times because the on/off
> hook switch membrane is a little sketchy). I have dealt with this by
That's one of the big things that causes problems, both with chan_sccp
and the phones themselves, both get a little confused. However several
other crash issues have been recently fixed, so running CVS_HEAD is
advised.

> restarting Asterisk. The only other thing I can say right now about
> the 7910 is that it and my Cisco FastHub don't get along. At all. I
> have the 7910 plugged into my 7960.
That's odd, the only time I've ever had ethernet incompatabilities was
with a very cheap switch.

> Overall, I would say that if you have a non-critical system and would
> like to use a 7910, chan_sccp should be able to handle it fine. 
> However, if you budget permits, the 7960 and 7940 phones are quite
> nice (use SIP with those -- it's far more reliable. I must say,
> though, that my 7960 has frozen/crashed a handful of time when running
> the SIP image. That was the phone itself, Asterisk was fine.) I have
> yet to purchase a 7905 or 7912, but I've played around with some
> 7912's on a CCM system -- they seem quite nice and I think they take
> SIP.
Yep, they do. (Don't know about the 7902, but really can't see why
anyone would buy one)

>The 7920 is also nice because it's wireless. However, I don't
> think Cisco has anything but a Skinny image for it [yet].
No they don't, and forget the yet, if a phone isn't announced with SIP
support it probably never will have it (witness: 7935/6, 7970)

> I would stick with SIP wherever you can.
And I agree

Thanks,
Julien
chan_sccp project lead


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Re: [Asterisk-Users] Re: Cisco 7960s and skinny

[Julien Goodwin]

On Wed, Apr 13, 2005 at 01:47:22PM +0200, Sergio arranged a set of bits into 
the following:
> 
> >My biggest task is getting in some of the big bugfixes and bad behavior
> >fixes that have been major issues. In testing at the moment is a fix to
> > 
> >
> Yes, I'm using * in a business environment with cisco 7960 and 7905 
> phones. Sip is the more stable solution.
> well no busy status line 'cause the cisco sip firmware does not support it.
> I was testing your chan_sccp. It's under development and I got some 
> crash or phones issue, but I think sccp could be the best flexible 
Please, and this goes for all chan_sccp users run asterisk with the -g
option to get coredumps if it crashes, and send me the backtrace (NOT
the coredump).

> system for a PBX. In my spare time I'm working on your chan_sccp code to 
I agree, sccp or a similar protocol is great as it allows the PBX to
contain most of the features that usually go to the phones, allowing an
amazing flexibility.

> understand how to get customized and localized (I'm in Italy) softkeys.
I'm not sure what if anything there is to localize, IIRC chan_sccp
transmits no text to the user except for softkey names, and their you
might be out of luck.

Hope to see some Aussie Asterisk users at LCA!
Julien

PS:
I'm now starting to look at writing a basic implmentation of CDP for
setting vlan's on cisco phones, expressions of intrest wanted!


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Re: [Asterisk-Users] Cisco 7960s and skinny

[Julien Goodwin]

On Tue, Apr 12, 2005 at 04:38:10PM -0500, Andy Hamilton arranged a set of bits 
into the following:
> Simon:
> 
> I have had Skinny going on a 7960 (which I then reimaged to SIP). I
> currently run a 7910 on Skinny (using chan_sccp) and use the
> aforementioned 7960 simultaneously.
> 
> Since you mentioned that you will have 50 phones, I assume you are
> using them in a business setting.  I would *highly* recommend using
> SIP, as I have found that the skinny driver is not as reliable as it
> could be (not criticizing Jan or Julien at all, here).
Even if you were, my own view is that chan_sccp is probably not the
thing to run on a client's PBX (not sure how good chan_skinny is, didn't
work the first time I tried which is why I do chan_sccp). My own
personal one, yes, a business where I worked full time and had
safe_asterisk or similar working, perhaps, anywhere else no.

My biggest task is getting in some of the big bugfixes and bad behavior
fixes that have been major issues. In testing at the moment is a fix to
allow speeddials to work at any time (meaning you could in theory create
a speeddial that auto-navigated a remote IVR), instead of crashing if
the handset was up. My next task is to get subscribe/notify working (if
anyone has looked at this code could they drop me a few pointers), which
should be pretty easy. Another thing which I might do is implement a
live/hot keypad so any keypress triggers a call, some people seem to
like this, but I personally can't stand it. (In any case it should be a
< 5 line patch if enabled all the time closer to 50 lines when you have
a per-device config option.

Also I've finally updated the web site to clean it up and hopefully add
some more info.

> Reimaging the 50 of them should only take a while (depending on what
> version of CCM they have at the moment). I reimaged 12 phones once for
> a business and it took less than 30 minutes after I got it going
> (toying with the phones to get them to take the image, exactly how the
> config files were to be set up, etc...).
> 
> I imagine you could easily get the whole thing done in less than a day
> (reimaging and config files), then figure out your dialplan.
> 
> Then there is the whole issue of writing the config files...but you'd
> have to do those with Skinny, anyhow.  I think with SIP you'll have
> much better reliability.
> 
> -Andy
> FWD: 428725
> 
> On Apr 12, 2005 12:48 PM, Morris, Simon <[EMAIL PROTECTED]> wrote:
> >  
> > 
> > Hello,
> >  
> >  Does anyone else have * running with Cisco 7960 phones and skinny?
> >  
> >  All the advise I am reading so far is telling me to load the SIP image on
> > the phone but I'd like to know what I'm going to lose by persisting with
> > skinny
> >  
> >  (Not reimaging 50 phones is one benefit amongst others of skinny)
> >  
> >  Thanks for any comparisons you can provide
> >  
> >  Rgds
> >  
> >  ~sm 
> > ___
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> >
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Re: [Asterisk-Users] Cisco 7940 Outgoing Audio

[Julien Goodwin]

On Wed, Apr 06, 2005 at 02:27:56PM -0600, Bellows, Jared arranged a set of bits 
into the following:
>I'm a Cisco 7940 phone using SCCP.  My setup is a private network with the
Wow, first time I think we've ever had a *phone* post to the list ;-)

>* box acting as dhcp server and also tftp server.  The phone loads and
>dials out fine.  I can hear the other person, but there is no outgoing
>audio.  I've read that this is an RTP problem and have tried making some
>changes in /etc/hosts to point to my * box IP but with no luck.  When I do
>a tcpdump I see that the RTP packets are sent to 0.0.0.0.  How do I get
>the phone to send to the * box?

If you're using chan_sccp update it to the latest CVS HEAD, that fixes
the RTP issues.

Thanks,
Julien
chan_sccp project lead


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Re: [Asterisk-Users] busy line status on CISCO 7940/7960

[Julien Goodwin]

On Tue, Apr 05, 2005 at 06:22:01AM -0600, Rich Adamson arranged a set of bits 
into the following:
> > Cisco TAC service told me that they will not support RFC 2848/3265 for 
> > the 7960 phones
> > So no busy status line notification with subscribe/notify system. This 
> > is really a bad news for me.
> > So they are not planning to backport sip firmware new features to the 
> > old phones.
> 
> Since the 7960 design is very old, its likely due to internal limitations
> such as available memory, etc. Not surprising at all.

Possible, but doubtful. I'm considering adding support for
subscribe/notify to chan_sccp, but don't know if anyone would use it.


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To all chan_sccp users! (Was: [Asterisk-Users] chan_sccp compile error)

[Julien Goodwin]

On Tue, Apr 05, 2005 at 08:34:02AM +1000, Rob Wise arranged a set of bits into 
the following:
> On Apr 5, 2005 8:20 AM, Rob Wise <[EMAIL PROTECTED]> wrote:
> 
> > Now compiling  sccp_channel.c   327 lines
> > sccp_channel.c: In function `sccp_channel_connect':
> > sccp_channel.c:198: parse error before `struct'
> > sccp_channel.c:199: `hp' undeclared (first use in this function)
> > sccp_channel.c:199: (Each undeclared identifier is reported only once
> > sccp_channel.c:199: for each function it appears in.)
> > sccp_channel.c:199: `ahp' undeclared (first use in this function)
> > make: *** [.tmp/sccp_channel.o] Error 1
> 
> Oops, I forgot to mention I'm using the cvs head version of asterisk
> and set the option in the chan_sccp Makefile accordingly.

I assume that you're using GCC 2.95 which doesn't allow new variable
declerations after a function call. If you move the call to
ast_rtp_get_us after the declerations of hp and ahp then it should work,
I'll do a proper fix when I get a chance (busy week).

I had been hoping to write a big message to the list, but here's the
condensed version:

Attention all chan_sccp users.

I have now officially taken over maintainership of chan_sccp from Jan
Czmok who was being overloaded with personal work (yes the web site
needs updating, it's on my list).

I know that many of you run chan_sccp with custom patches and hacks to
get various things working, this is a call for you to send in the diffs
so that they might get merged in to CVS. Unified diffs against chan_sccp
CVS head requested, along with a note stating if that change was tested
against asterisk head, 1.0 or both, a list of which phones a change was
used with is also useful as some things can be surprisingly
phone-specific.

Diffs should be sent to me (a private reply to this mesage is fine), and
I'll try to notify people when their patch gets merged (And you will be
noted in the commit log).

To all those after cisco 7970 support, I now have a copy of the cisco
soft-phone along with ethernet dumps from a 7970, the softphone
currently registers, but I've not get got it to make a call (But I
haven't tested it after making some more bug fixes). If there's anyone
in Australia with a 7970 they could lend me that would be much
appreciated, otherwise this work will continue to take a while as the
soft-phone is different from the real 7970.

Thanks,
Julien Goodwin
chan_sccp project lead


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Re: [Asterisk-Users] chan-sccp-easter2005 make error with stable 1.0.6?

[Julien Goodwin]

On Mon, Mar 21, 2005 at 01:03:52PM -0700, Kevin P. Fleming arranged a set of 
bits into the following:
> Remco Barende wrote:
> 
> >Are you sure? This is in the makefile:
> >
> ># Asterisk version, currently only v1_0 and HEAD are supported
> >ASTERISK_VERSION=v1_0
> 
> Well, then the code is buggy, because the channel technology structure 
> stuff is only in HEAD, not 1.0.
And indeed it was, is wasn't checking the asterisk version at one place,
that's now fixed.

See if you actually let me know about these things they get fixed!

Thanks,
Julien
chan_sccp deveoper


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Re: [Asterisk-Users] [OT - somewhat] chan_sccp status

[Julien Goodwin]

On Tue, Mar 01, 2005 at 03:50:16PM -0600, Chris Wade arranged a set of bits 
into the following:
> I hate to re-post like this, but I still haven't been able to get ahold 
> of the chan_sccp developers (short of opening a bug report on their 
> mantis installation just to get their attention :).  I originally sent 
> this email back at the beginning of February.
I think I sent a response to this message when it was first sent.

> I would love to see an update as to the status of chan_sccp.  Also, I'm 
> very interest in contributing to the efforts of chan_sccp, so please, if 
> anyone from the dev team is reading this, please drop me a line.
Status: For myself, I work on what intrests me when I can (I'm now a
full-time student and work 2.5 days a week). I'm slowly commiting my
fixes for various things, but my three additional features (proper
contention beeps, the voicemail button and better hold support) are
waiting until I can get more models of phones to test against (I'm
missing a 7910, a 7905/12, a 7960 (my 7940 should arrive tomorrow) and a
7920), as I posted before, SCCP is not a well defined protocol and the
phones change it seemingly on a whim so it's much harder then trying to
implment a standard like SIP or IAX. [Also useful are: 7935/6, 7970,
7914]. And again, if anyone has a callmanager installation tcpdump
format ethernet dumps of features/phones that chan_sccp doesn't yet
support are helpful (just ask before sending even 1MB of dump).

I've also contacted Cisco who claimed that they don't HAVE protocol docs
for SCCP (even though I have the ISBN...) and arn't willing to help out
with info at all.

And in regards to the "New release" around 20/1/05 I don't know either,
and if I had admin rights on sf.net I would have long removed it, but my
own e-mail's to Jan have gone unreplied.

Thanks,
Julien
chan_sccp developer


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Re: [Asterisk-Users] Anyone had a Cisco 7970 working with Asterisk?

[Julien Goodwin]

On Fri, Feb 25, 2005 at 11:31:34AM +0930, Hermann Wecke arranged a set of bits 
into the following:
> Paul A Brown wrote:
> >Anyone had a Cisco 7970 working with Asterisk?
> 
> As 7970 uses SCCP, you can do it with asterisk. I did it with 7960.
Nope, you can't.
As SCCP is not really a protocol, it's just something that the phones
mumble in something approaching unison. THAT's why chan_sccp and
chan_skinny are limited in their phone support.

Once I'm able to get my hands on a 7970 (US eBay seems to be selling
them for OK prices) support should be forthcoming in chan_sccp. However
if anyone has a 7970 and cisco call manager if they send me a tcpdump
file of the phone registering, making, and recieving a call then I might
be able to speed that up.

Thanks,
Julien
chan_sccp developer
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Re: [Asterisk-Users] 7912G: Takes the same firmware as 7940/60?

[Julien Goodwin]

On Sun, Feb 13, 2005 at 08:35:33PM +0100, Michiel van Baak arranged a set of 
bits into the following:
> On 20:08, Sun 13 Feb 05, Stefan Gofferje wrote:
> > Michiel van Baak schrieb:
> >  > Thnx.
> > >No luck for me I guess.
> > >chan_sccp it will be.
> > 
> > Not for the 79[05|12]... At least my 7905 does not like chan_sccp too 
> > much and they crashed my * (1.0.5)... unless you bounty the chan_sccp 
> > developers for 79[05|12] support OR ask your local Cisco dealer for a 
Which while nice wouldn't necesserily help. For myself at least I can't
afford one of each of the phones, and what I have comes from judicious
eBaying. Loan of phones is much more usefull then anything else.

> That's about it. And from what I read the SIP image can
> really use the rest of the phone like speed dial, call
> forward etc.
Speed dials should work, just configure them in sccp.conf, and most of
the rest is under (slow) development. I actually have implementations of
some of the features ready, they just need testing with soem more
phones.

Thanks,
Julien
chan_sccp developer


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Re: [Asterisk-Users] Asterisk and Cisco phones chan_sccp vs chan_skinny vs native SIP and one-way audio

[Julien Goodwin]

On Mon, Jan 31, 2005 at 04:05:23PM -, Michael J. Tubby B.Sc (Hons) G8TIC 
arranged a set of bits into the following:
> I've recently built a couple of Asterisk boxes and want to migrate
> away from CallManager to Asterisk.
> 
> On my Asterisk box I have about 8 Grandstream BT101s and a
> Cisco 7905G in SIP mode, on my CallManager I have about 10
> x 30VIP, 2 x 7940 and a 7960.
> 
> I've built Asterisk version 1.0.5 along with Zozo's chan_sccp
> (CVS latest from last night) and got it partially working.  All devices
> are on the inside of a private network at the moment (192.168.144.0/24)
> and I'm having some issues with devices on chan_sccp.
That's chan_sccp from chan-sccp.sourceforge.net?

> The 30VIPs can place and receive calls but I have a one-way
> audio problem.  The 7960 can receive calls but when I place calls
> from it I end up directly in the voicemail "unavailable" and the SIP
> phone doesn't ring.
I know about the problems with the 30vip and am slowly fixing them. but
the issue with the sound is seemingly a bug in asterisk that says to
tell the other device that 127.0.0.1 is the best route for RTP.

> Looking at the network the SIP device opens an RTP stream to the
> Cisco (30VIP or 7960) but the Cisco device doesn't send RTP
> back to the SIP phone...  can anyone point me in the right direction
> with this?

> A more general question: with Cisco phones being removed from
> a CallManager environment, is it best to keep them in Skinny/SCCP
> mode or change out to SIP?  The 30VIPs can only do SCCP/Skinny
> so which of the two channel drivers in Asterisk should I use for
> best effect?
chan_sccp has more features (and will have several more once I can get a
current generation phone to test them on), but is less stable.
As for the phones that support SIP, my view is that unfortunatly for now
SIP is the better choice for stability and feature support.

Thanks,
Julien
chan_sccp developer


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Re: [Asterisk-Users] Becoming a VOIP provider

[Julien Goodwin]

On Wed, Jan 19, 2005 at 02:53:56PM -0700, Keith Burns arranged a set of bits 
into the following:
> Be careful of LI requirements in Australia.

Which ones, there are pretty much none!
What we have in .au (for quite a lot of things in reality) is a
PERCEPTION that these things are legislated (like free speech and
lifeline services) but they arn't. And I can confirm that under
australian communications law if you're doing VoIP lifeline is not a
requirement (and if required I can supply contact details of one of the
people who recently got that statement claified in the law).

> You MAY be able to put the onus for this on your upstream (PRI/IMT)
> provider, but if you have many, this could be messy.
> 
> Best bet would be to have a solution yourself... when I was looking into
> this the good news was that the enforcement agencies (which at last
> count was around 47, any of whom could hit you for their own real-time
> feed of the conversation) were considering taking the VoIP feed (RTP)
> and the logs of the signaling. (Things may have changed, your mileage
> may vary, yada, yada, yada).
> 
> Also, after a little kiddy died of an asthma attack in rural Victoria
> because Telstra (the lazy @[EMAIL PROTECTED]  - I digress) hadn't fixed their 
> phone,
> lifeline services (E911 in the US) are more and more important to have
> nailed.. you don't want that on your conscience (your service not
> working causing harm to someone) nor would your business appreciate the
> lawsuits.
> 
> 
> 
> > -Original Message-
> > From: [EMAIL PROTECTED] [mailto:asterisk-users-
> > [EMAIL PROTECTED] On Behalf Of Ed Robbins
> > Sent: Wednesday, January 19, 2005 2:02 PM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [Asterisk-Users] Becoming a VOIP provider
> > 
> > Ty Carter wrote:
> > 
> > >Ed:
> > >
> > >I think you must have some bad information here.VoIP is an
> Information
> > >service and not subject to CALEA regulations.
> > >
> > >
> > >
> > 
> > Whether it's a subject to those regulations or not I still know first
> > hand it's a big issue with broadband voip providers.  I work for a
> > company that develops VoiP for the broadband market and it's something
> > we had to develop for our customers.  I don't know all the details of
> it
> > and what is going on behind the scenes in terms of regulations but my
> > thinking is that voip providers have to tie into the PSTN somewhere
> and
> > the FCC can most likely tap into(no pun intended), meaning require you
> > meet the guidlines put forth in CALEA, from that legal point of view.
> I
> > had never thought about this before but I should talk to my buddy who
> > got a CLEC a few years ago, I'm wondering if there is something in
> there
> > that spells it out.
> > 
> > Ed
> > 
> > >According to the calea website:
> > >
> > >In a Notice of Proposed Rulemaking FCC 02-42 released on February 15,
> 2002,
> > >the FCC initiated a proceeding to establish rules and regulations
> regarding
> > >the classification of "wireline broadband Internet access" under the
> > >Telecommunications Act. Digital Subscriber Line (DSL) service is an
> example
> > >of wireline broadband Internet access. In this document, the FCC
> > >"tentatively" decided that wireline broadband Internet access is an
> > >"information service."
> > >
> > >In a Declaratory Ruling and Notice of Proposed Rulemaking FCC 02-77
> released
> > >on March 15, 2002, the FCC made a "declaratory ruling" that cable
> modem
> > >service (Internet access through cable TV lines) is an "information
> service"
> > >under the Telecommunication Act and initiated a proceeding to
> establish
> > >rules and regulations based on that finding.
> > >
> > >Therefore, the FCC's pending wireline broadband Internet access
> proceeding
> > >is CC Docket Nos. 02-33, 95-20, and 98-10 and the cable modem
> broadband
> > >Internet access proceeding is CS Docket No. 02-52 (collectively the
> "FCC
> > >Broadband Proceedings").
> > >
> > >It should be noted that the FCC is not primarily focusing on CALEA in
> these
> > >proceedings, rather its emphasis is on the economic and policy
> concerns
> > >involved in regulation of these services under the Communications
> Act.
> > >However, since CALEA exempts "information service" from the
> surveillance
> > >capability requirements of Section 103, these FCC decisions have the
> > >potential to exclude broadband DSL and cable modem service from CALEA
> > >compliance.
> > >
> > >The FBI filed the following comments in the Broadband
> > >
> > >
> > >
> > >>-Original Message-
> > >>From: [EMAIL PROTECTED]
> > >>[mailto:[EMAIL PROTECTED] On Behalf Of
> > >>Ed Robbins
> > >>Sent: Wednesday, January 19, 2005 3:19 PM
> > >>To: Asterisk Users Mailing List - Non-Commercial Discussion
> > >>Subject: Re: [Asterisk-Users] Becoming a VOIP provider
> > >>
> > >>Manjit Riat wrote:
> > >>
> > >>
> > >>
> > >>>That was a really nice description... Can you do 1-14 and I'll

Re: [Asterisk-Users] SCCP questions

[Julien Goodwin]

On Thu, Jan 13, 2005 at 01:09:54PM +0200, Kelemen Zoltan arranged a set of bits 
into the following:
> Hi!
> 
> I have two, not too related questions:
> - the probably simpler one: if anyone can help me out using a Cisco 
> 7905G with chan_sccp? I did already managed to get it working with a SIP 
> image, I'd just like to see it work with this one as well. It's probably 
> something I screw up with the configuration, as the phone registers, 
> only I don't get any lines with it, although I have it configured it to 
> auto-login.
> 
> excerpt from my sccp.conf
> -snip--
> 
> [SEP001193C2ABFC]
> device=SEP001193C2ABFC
> type=7905
> autologin= c7905
> callerid="cisco 7905"
> 
> [c7905]
> line = 1045
> 
> -snip--
> (I have a default extension set for the entire sccp.conf, so that 
> shouldnt(?) be the issue)

That's the contents of a skinny config file, the two have different
formats. Here's what I use for one of my phones:
---snip---
[SEP003080628DD7]
type= 12
autologin   = phone4
description = Cisco Phone 4

[phone4]
id  = 4005
pin = 1237
label   = Phone4
description = Phone4
callwaiting = 0
mailbox = 4005
cid_num = 4005
cid_name= Cisco Phone 4
--- endsnip ---
And I call with Dial(SCCP/phone4)
> I have the XMLDefaultConf in place, tftp server running, although that's 
> about it. I would appreciate any pointers in this general direction. 
> What am I missing? :)
See above, copy the format, just use your own data.

> The second, much more thorny question is: did anyone had any success on 
> using a KIRK IP600 with asterisk?
> - The only thing I really found on the net were a couple of emails on 
> this list, that didn't get me too far.
> The KIRK IP600 is a DECT (cordless) to IP solution, with support for 
> SCCP and H323. The SCCP interface was designed specifically to be 
> interoperable with Cisco Call Manager, and it emulates a 7940 for each 
> of the phones it has registered.
> 
> With chan_skinny I managed to register the phones, they've got tone, but 
> they would not ring out.
> With chan_sccp I had no luck at all, I'm getting the following messages 
> on the CLI:
> ==  >> Got message AlarmMessage
> ==  >> Got message RegisterMessage
> == Sending Packet Type RegisterRejectMessage (37 bytes)
> 
> Note: the two modules were NOT tried at once.
> 
> So far I didn't have time to check it out with h323, but If anyone had 
> it working that way, I'm interested in that one as well.

H323 is probably the one to try, but fix your sccp config like the above
(a compile fix for asterisk CVS has just been committed) and let me know
what you get (in debug mode very verbose). If anyone in Australia has one 
or is able to arrange a loan for a few days I'd probably be able to make 
them work, but without being able to hack on the code with the device it
makes it quite hard. 

I'm just writing some code to see if I can fix the "client sent
IPPortMessage without first registering" that someone got before, if
anyone is able to duplicate that drop me a line and I'll see if my patch
works against it.

Thanks,
Julien Goodwin
chan_sccp developer


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Re: [Asterisk-Users] Operator Panels?

[Julien Goodwin]

On Wed, Jan 12, 2005 at 08:07:11AM -0600, Matt Schulte arranged a set of bits 
into the following:
> Ok, we're trying to use Asterisk as a PBX in our office. Our original
> plan was to use a Cisco 7960 with a 7914 attached. Short story is, no
> one updated chan_sccp in a long time and the 7914 is questionable at
> best anyway from what I've heard. We couldn't ever get chan_sccp to
> compile, I went to an older version of Asterisk and that broke some of
> our SIP devices. We tried using a couple soft panels listed on the Wiki,
> the only one easy enough to use was Asternic. And we found also that is
> buggy and doesn't function correctly with the new Asterisk. Any
> Suggestions?

What was your problem with chan_sccp? There's only one small issue I
know of in the code (already fixed, I just haven't committed it to CVS).

Although the biggest issue with using it would be that chan_sccp doesn't
yet have hint support (it's forthcoming once I get my new phone
delivered this week). 

Thanks,
Julien Goodwin 
chan_sccp developer


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Re: [Asterisk-Users] Asterisk Demo

[Julien Goodwin]

On Sun, Jan 09, 2005 at 10:51:56PM +0200, Walid Azab arranged a set of bits 
into the following:
>Hi,
> 
>I need to setup a demo for asterisk and need some help here please. The
>demo is connecting to Asterisk a Cisco 7970 SIP (ver. &.0) and a SIP
Ah, a quick check of the cisco site seems to confirm that the 7970 does
not yet speak SIP, and its varient on SCCP is not yet supported by
asterisk (neither in chan_skinny or chan_sccp). The 7960/40 on the other
hand is supported in both sip and skinny.

If it is I'm sure that a lot of people here would like to hear about it.

>client on HP iPAQ via a wireless hotspot. I need to configure both with
>the same extension with a shared line like in Cisco CallManager. This way
>if the extension is called both iPAQ and the IP phone ring and the user
>gets to pick up using either.
This bit isn't too hard it's just:
Dial(SIP/nameof7960&SIP/nameofIPAQ)
In the dialplan


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Re: [Asterisk-Users] MINNESOTA: TwinCities Asterisk Users Group - Meeting tomorrow 01/08/05 11:30am

[Julien Goodwin]

For people in Melbourne Australia there'll be a get together for people
intrested in VoIP of all persuasions on the 22nd of January in Preston.
If you're not a LUV member e-mail me privatly and I'll forward on the
invite.

Thanks,
Julien


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Re: [Asterisk-Users] CallerID in Australia & Analogue PSTN Phone

[Julien Goodwin]

On Thu, Jan 06, 2005 at 07:25:35AM +1100, Howard Lowndes arranged a set of bits 
into the following:
> On Wed, 2005-01-05 at 23:49, PHP Mechanic wrote:
> > >> > What I need more though is examples of anything that needs to go into
> > >> > extensions.conf
> > >> 
> > >> You could add this line if you want
> > >> exten => s,1,NoOp(Caller ID on the PSTN line is ${CALLERID}) 
> > > 
> > > M.  Tried that, but it didn't deliver ${CALLERID}
> > > 
> > Did the caller have callerid enabled by their telco ?
> 
> Sure was.  It was me calling myself from my mobile (cell) phone, and
> that definitely has CLID enabled.  In AU CLID is enabled by default.
Only for mobiles, and that's incoming. It's not enabled for landlines by
default (at least for landlines that were around since before callerid
was introduced ~5 years ago). For testing that sort of thing picking up
a $30 clid box might be worth it.

> Do you know if the Digium X101P has problems with reading CLID on the
> line?  There is a wiki that says that in AU the DEFAULT_CIDRINGS needs
> to be =2 rather than the default =1 and I have set that; perhaps I
> should reverse that and try again.


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Re: [Asterisk-Users] RE: Polycom 500 - Dialtone while connected

[Julien Goodwin]

On Sun, Dec 12, 2004 at 05:50:16AM -0600, Rich Adamson arranged a set of bits 
into the following:
> > > I'm interested too. Any chance to put the archive in a ftp site?.
> > > 
> > 
> > I am also interested in getting the 1.3.4 firmware. It annoys me that I 
> > can't just get it from Polycom's website, and forces me to rethink 
> > deploying their phones for customers.
> 
> Send emails to the Polycom sales, support and other groups, and complain
> to them. Maybe if enough folks do that they will rethink their policy.
> 
> They claim to be handling it the way they do because they want to
> "maintain high quality customer support" through certified dealers.
> That might be true for their more sophisticated products, but it
> certainly does not appear to be working for their IP phones. I'd bet
Doesn't work for their better phones either...
The problem is that they assume that the reseller knows more then the
customer, something that hasn't been true for a *long* time.
Polycom are second only to Cisco in the shere stupidity of their
managemnt (yes I haven't delt with Avaya or some of the more traditional
companies who are apparently just as bad) with regards to VoIP,
believing that their customers will want to buy their softswitches, and
are not buying VoIP for say _the_flexibility_.

I've tried telling Cisco that if we could simply have a copy of the
Skinny protocol docs (which do exist and are distributed to some
companies) that they would have increased sales due to the people who
want features SIP can't provide, or the possibility of the integrated
applications. But they don't listen and don't seem to care. Fortunatly
the word I'm hearing (at least in .au large commercial) is that many
large companies that have gone cisco are getting very annoyed at them
for promising and not delivering, and if they continue their next
upgrade will be explicitly *not* cisco. [Those are direct words from a
few large scale PBX integrators I know, not myself]

If they would just realise that if they had their products actually
realise the potentional that VoIP offers they'd increase sales where it
matters, on the equipment that's VISIBLE, with THEIR branding on it, not
* or whoever makes the switch. And at least for * they dont need to do
anything, just release the docs _they_already_have_.

> a fair number of folks reselling their IP phones aren't certified 
> and they are picking up the product through (back-door) distributors.
> (That's got to be part of the reason why resellers do not include 
> copies of the required (license) software when shipping product, if
> when its stipulated on a purchase order.)
I assume you mean even when, and if a vendor did that I'd just return it
and not pay the bill. If it's not what I ordered then it doesn't meet
the contract and so I won't pay. Any vendor that still complained would
never get my business, not that of anyone I know.


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Re: [Asterisk-Users] Cisco IP Conference 7935

[Julien Goodwin]

>  I have one Cisco IP Conference 7935. Somebudy have any idea how I
>can config this phone to work woth "*".

>  My "*" server is now working with GrandStream Phone and X-Lite
>SoftPhone, I need to add this Cisco 7935 but

>  I don't know how I can convert to SIP.
You can't, Cisco don't offer SIP on the 7935 (or the 7920, the 7970, or
7960+7914 combo). For the 7920 at least they've claimed that sip would
be around severl times now, but have never released it. With the 7935
you're SOL for SIP. Fortunatly however it does work with chan_sccp (As I
don't have one I can't say how well, but apparently you can make and
recieve calls). I suggest you give that a go, or failing that get a SIP
ATA and an analog Polycom.

Thanks,
Julien Goodwin
chan_sccp developer
http://chan-sccp.sf.net/


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Re: [Asterisk-Users] Asterisk and Cisco IP Phones

[Julien Goodwin]

On Mon, Dec 06, 2004 at 07:43:24AM -0600, Rich Adamson arranged a set of bits 
into the following:
> I don't think its an argument as much as it is folks expressing opinions
> without giving you a clue why they've formed that opinion. Here's another
> one.
> 
> SCCP is a cisco proprietary protocol that some folks have partially 
> reverse engineered, writing * code to support those basic functions
> that have been reverse engineered. Not all of Cisco's SCCP functions
> have been reverse engineered. If you compare functionality of what
Actually one you get past registration of the phone (which can sometimes
be pretty odd) we can support almost everything. The problem is that we
don't have the function code written yet to support call forwarding or
ad-hoc contrences (and similar), this generally doesn't take very long
to write but has to be solidly tested due to all the protocol
differences between the phones.
There are also the phones that only chan_sccp supports due to cisco's
stupidity, 7920, 7935, 7960+7914, and hopefully soon the 7970.
[Talking about chan_sccp here]

> Cisco is doing with Call Manager to that implemented in *, you'll see 
> differences. If the functions that have been implemented meet _your_ 
> needs, then use it. (Read: work in progress.)
Otherwise you can let us know what's missing for you and we'll see what
we can do.

> On the other hand, SIP is an established open RFC standard. However,
> not all SIP standards are implemented in *, and not all SIP phone 
> manufacturers implement all published standards. (Read: work in progress.)
> But, there are significantly more folks working towards expanding
> SIP support then there are folks reverse engineering SCCP.
But there are also all the many extensions to SIP that the various
vendors have (eg the hint stuff)

> So, _you_ have to evaluate SIP vs SCCP and compare the two to whatever
> your requirements happen to be, and chose a protocol that you are
> comfortable addresses the majority of your requirements. No one can
> do that for you. Remember, opinions are like mouths (or substitute
> whatever anatomy part you'd like), everyone has one.
> 
> 
> > Guys, obviously there is an argument about SIP vs SCCP when it comes to
> > using Cisco IP Phones with Asterisk. I am not really sure which way to go.
> > Probably I will go with SIP now unless you guys do recommend not using it.
> > 
> > Walid 
> > 
> > -Original Message-
> > Let me CLARIFY for those that might not get what I ment.. DO NOT RECOMMEND
> > SCCP unless you have actually installed and used it.  Its crap... 
What are your problems with SCCP? Submit bug reports, they do get fixed.

> > SIP is what you want if you use a cisco phone with asterisk.
> > 
> > bkw
> > 
> > > 
> > > Pfft ya right if you want half assed supported channel drivers.  Use SIP.
> > > 
> > > bkw
> > > 
> > > > No you don't have to use SIP.   You can also use the SCCP channel on *
> > > > with Cisco phones.
> > > >
> > > >
> > > >
> > > >
> > > > I want to start using Asterisk with Cisco IP Phones 7960 / 7940/ and
> > > 7905.
> > > >
> > > > Any info or help is appreciated.
> > > >
> > > >
> > > >
> > > > I know I'll have to use SIP but if I want to use the phones off site 
> > > > meaning
> > > >
> > > > from my home for example, how can this be done?
> > > >
> > > > Also, regarding external lines what are the options for Asterisk?
> > > >
> > > >
> 
> 
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Re: [Asterisk-Users] cisco 7960 sccp firmware version?

[Julien Goodwin]

On Tue, Nov 30, 2004 at 07:43:11PM -0500, Andy Reinke arranged a set of bits 
into the following:
>I have some Cisco 7960's and want to use them with SCCP - I have gotten it
>working with a few different firmware versions but all seem a little
>flakey.  I know that SCCP is not as solid as SIP but am wondering, which
>firmware version is advised for use with chan_sccp from
>http://chan-sccp.sourceforge.net/
What in particlar is flaky?
What problems have you been having?
Also are you using chan_sccp cvs?

Generally the cisco firmware is not the problem, it's chan_sccp.

Let me know what problems you're having and I'll see what I can do.

Thanks,
Julien
chan_sccp developer.


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Re: [Asterisk-Users] How to make/recieve call using asterisk when thereis a power failure?

[Julien Goodwin]

On Thu, Nov 25, 2004 at 09:59:58PM +1100, Bob Purdon arranged a set of bits 
into the following:
> 
> >An E1 termination can require local power. In that case you will have to 
> >provide backup power to it. Some get their power from the central office, 
> >in which case this is not a problem.
> 
> In our neck of the woods (Australia) the dominant carrier typically 
> deploys one of two solutions:
> 
> (a) E1/PRI over copper - I haven't seen this for a long while, but I 
> believe they're line powered;
The one we got installed today (yes, really) at work wasn't line
powered, another bloody wall-wart to go with the others. However that
was shipped as an OnRamp 30 (ie a euro PRI), whereas the E1 data link is
also locally powered, but with power coming from the curb.

We're just outside the Melbourne CBD (next to melb-uni if people care)
and now have 2 E1's, 1 data, 1 ISDN voice, and about another 1/2 dozen
analog trunks. (Plus the two ADSL links)

> (b) E1/PRI over fibre - see these all the time, and the transmission 
> rack they install at the customer site includes batteries and a 
> rectifier which typically provides service for a couple of days in a 
> typical installation.


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Re: [Asterisk-Users] Kirk IP 600 DECT station

[Julien Goodwin]

On Fri, Nov 12, 2004 at 11:06:52PM +0100, Remco Barende arranged a set of bits 
into the following:
> OK, I just got the Kirk IP 600 kit :)
> 
> It turns out that they actually make one unit that does H323 and can do 
> Skinny (Cisco) if you buy the version with a license for it. Mine supports 
> both.
Neat! Any info on pricing?

> The box appears to be running linux btw :)
> nmap revealed 2 open ports, telnet and http and also
> TCP/IP fingerprint:
> SInfo(V=3.55%P=i686-pc-linux-gnu%D=11/12%Time=419509CD%O=23%C=1)
> 
> I couldn't find any info on this station and how to connect it to * but 
> that's ok :)
> 
> It seems that you must configure the phones as Cisco 7940 in Call Manager. 
> The Wiki about the 7940 uses SIP which will not work so I have to try 
> skinny or H323.
> 
> I think the best way would be to use the skinny protocol but I'm a bit 
> lost there. When looking for info on cisco protocol I actually found 3 
> channels : chan_skinny / chan_sccp and asterisk-sccp.
There's:
* chan_skinny - basic support, supplied with asterisk
* chan_sccp - More phone support, more features, chan-sccp.sf.net
There are various splits and forks of chan_sccp out there, but the one
being activly developed is the one hosted on sourceforge. Asterisk-sccp
appears to just be the name that gentoo gave chan_sccp in their archive.

> Hopefully chan_skinny which comes with * is ok?
Should work for basic tasks, but chan_sccp supports more features.

> The wiki SCCP-HOWTO2 says :
> noload=chan_skinny.so
> but I guess I do want to load skinny?? should I specify load= ?
No, that's when using chan_sccp so that you don't have two modules
competing over the skinny port.

> I will keep posting results to the list and finally add a wiki when it's 
> working, hope any of the cisco experts can help :)
If you have any problems with chan_sccp drop me an e-mail.


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Re: [Asterisk-Users] Cisco 7910 - Success?

[Julien Goodwin]

On Mon, Nov 08, 2004 at 12:41:40PM -0600, Matthew Boehm arranged a set of bits 
into the following:
> Have you tried chan_sccp?
Just a heads-up while schan_sccp doesn't yet support
hold/transfer/voicemail buttons they code for hold and voicemail has
been written it just needs to be tested.
If anyone is intrested please drop me a private e-mail and I'll send you
a link to the patches.

> http://chan-sccp.sourceforge.net
> 
> Matthew
> 
> - Original Message - 
> From: "James Forte" <[EMAIL PROTECTED]>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <[EMAIL PROTECTED]>
> Sent: Monday, November 08, 2004 12:20 PM
> Subject: Re: [Asterisk-Users] Cisco 7910 - Success?
> 
> 
> >
> > I have two 7910's one is a 7910G+SW and one is 7910+SW
> >
> > I have the 7910G+SW to work with an xml file in the /tftpboot directory.
> >
> > Using chan_skinniny  however I cannot get the hold tranfer etc. buttons to
> > work.
> >
> > skinny.conf is as below:
> > ---
> > 501]
> > context=default
> > nat=no
> > host=192.168.10.144
> > accountcode=501
> > fromuser=501
> > callerid=Jim Forte <501>
> > incominglimit=1
> > outgoinglimit=1
> > disallow=all
> > allow=ulaw
> > allow=alaw
> > allow=g723.1
> > allow=gsm
> > device=SEP000AF4A3D50A
> > version=P002F202
> > linelabel=JPF 501
> > callwaiting=yes
> > transfer=yes
> > threewaycalling=yes
> > line => 501
> >
> > /tftboot/SEP000AF4A3D50A.cnf.xml file is as below:
> > -
> > 
> > 
> >  
> >   
> >
> > 
> >  
> >   2000
> >  
> >  192.168.10.89
> > 
> >
> >   
> >  
> > 
> > {Jan 01 2002 00:00:00}
> > 
> > 
> >  English_United_States
> >  en
> > 
> > United_States
> > 0
> > 
> > 
> > 
> > 
> > 
> > 
> > 
> > P00403020214
> > 
> > (END)
> >
> > Any thoughts on how to get the hold and tranfer button working
> > appreciated.
> >
> > jim forte
> >
> > On Thu, 4 Nov 2004, Matthew Boehm wrote:
> >
> > > I know that the 7910 only works with Skinny. We have a possible client
> that
> > > wants to bring 80 lines to us off his current provider. All 80 of his
> phones
> > > are Cisco 7910s. Does anyone run 7910s with chan_sccp or the like and
> find
> > > that it works good?
> > >
> > > Thanks,
> > > Matthew
> > >
> > > ___
> > > Asterisk-Users mailing list
> > > [EMAIL PROTECTED]
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> > > To UNSUBSCRIBE or update options visit:
> > >http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> >
> > -- 
> > --
> > Yours Truly
> > James Forte, Magna.Net Inc.   THE Dot Net in Timeshare
> > http://Timeshare.Magna.Net/   mailto:[EMAIL PROTECTED]
> > 7540 Municipal Drive, Orlando FL, 407-352-2402 EFax: 253-423-5482
> > THIS COMMUNICATION IS ONLY INTENDED FOR THE RECIPIENT(S) ABOVE.
> > PLEASE DISCARD IF YOU HAVE RECEIVED THIS IN ERROR.
> > -
> > ___
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> > [EMAIL PROTECTED]
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> >http://lists.digium.com/mailman/listinfo/asterisk-users
> 
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Re: [Asterisk-Users] help , chan_sccp wont build.

[Julien Goodwin]

On Sat, Oct 16, 2004 at 02:34:20AM -0400, Jason Price arranged a set of bits into the 
following:
> just downloaded the latest cvs on * and chan_sccp , it wont build, i
> have edited the make file per the README
 
> [EMAIL PROTECTED] chan_sccp]# make
> Now compiling  sccp_channel.c   279 lines
> sccp_channel.c: In function `sccp_channel_send_callinfo':
> sccp_channel.c:48: structure has no member named `callerid'
> sccp_channel.c:49: structure has no member named `callerid'
> sccp_channel.c:49: structure has no member named `callerid'
> sccp_channel.c:49: structure has no member named `callerid'
> sccp_channel.c:49: structure has no member named `callerid'
> sccp_channel.c:49: structure has no member named `callerid'
> make: *** [.tmp/sccp_channel.o] Error 1
According to current Asterisk & chan_sccp CVS (as of 5 minutes ago) you
shouldn't be getting that, what seems most likely is that you messed up
setting the ASTERISK_SOURCE in the Makefile, check that and the
integrity of your asterisk CVS and drop me an e-mail if you have any
more problems.

Thanks,
Julien Goodwin
chan_sccp developer


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Re: [Asterisk-Users] sccp cisco 12sp HELP !!!

[Julien Goodwin]

On Fri, Oct 15, 2004 at 09:01:54PM -0400, Jason Price arranged a set of bits into the 
following:
> ok guys, ive been trying to get this to work for 6 hrs now
> ive got a cisco 12 sp and i am trying to get it to work with sccp. The
> phone boots and is looking for the SEPDefault.cfg or the one below,
> BUT i cant find anywere on the net what the content of this file is
>  im guessing that its the ip of the * box. im riping my hair out
> on this one please help...
> 
> 
> 20:54:47.793156 192.168.1.15.51216 > apollo.tftp:  28 RRQ
> "SEP00D0BA848162.cnf" [tos 0x10]

My own Cisco 12SP+ phones don't have the SEPmac.cnf files, and I don't
even have the SEPdefault.cnf file, after about 1-2 minutes the phones
time out and start trying to just connect via sccp to the TFTP server.

Thanks,
Julien Goodwin
chan_sccp developer


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Re: [Asterisk-Users] Cisco 7960 + 7914 - not worked

[Julien Goodwin]

On Sat, Oct 16, 2004 at 02:32:53AM +0400, Vasiliy Voropaev arranged a set of bits into 
the following:
>I have Cisco 7960 with 7914 operator console. 7960 successfully registered
>and working with chan_sccp2, but the buttons on the 7914 are all red. What
>may be wrong?
> 
>sccp.conf:

>[SEP]
>description = VVG
>type = 7914
>context = sip
>autologin = 821
 addon = 1 ; How many 7914's are connected
>speeddial = 11,Test1
>speeddial = 12,Test2

Also check that you are using current chan_sccp with asterisk 1.0, but
7914 support has been in chan_sccp for a few months now.
You probably also wish to add some more speeddials and lines to be able
to properly test the 7914.

Thanks,
Julien Goodwin 
chan_sccp developer


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Re: [Asterisk-Users] chan_sccp.so: _use_ast_pthread_create_instead_

[Julien Goodwin]

On Sat, Sep 25, 2004 at 06:48:02AM +0200, Goran Dj. arranged a set of bits into the 
following:
> I tried to install chan_sccp (make; make install) but after that when
> asterisk starting:
> 
> [chan_sccp.so]Sep 25 06:34:28 WARNING[16384]: loader.c:242
> ast_load_resource: /usr/lib/asterisk/modules/chan_sccp.so: undefined
> symbol: __use_ast_pthread_create_instead__
> Sep 25 06:34:28 WARNING[16384]: loader.c:423 load_modules: Loading
> module chan_sccp.so failed!
> 
> I tried to replace pthread_create() with ast_pthread_create() in
> chan_sccp.c, but same error...
> 
> Help?

Use CVS chan_sccp, it has the fix for this (and other changes). Anon CVS
access is easy using the information on the sccp site.
http://chan-sccp.sf.net/

Also that seems to indicate that you were compiling chan_sccp against a
different version of asterisk then you are running (this may not be so,
but please check).

Thank,
Julien
(chan_sccp developer)


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Re: [Asterisk-Users] skinny or sccp?

[Julien Goodwin]

On Sun, Aug 22, 2004 at 06:11:10PM +0200, Pavel Jezek arranged a set of bits into the 
following:
> Hi, please tell me, 
> is original skinny support in Asterisk stil under development or is better to try 
> chan_sccp from
> http://chan-sccp.sourceforge.net ?
> my first try was unsuccessfull (chan_sccp compile OK, but module loading fail during 
> Asterisk startup) 
> and my phone (C7940) seems to be not supported in original chan_skinny :(

As someone who is working on chan_sccp I highly recomend you give it a
go. Your module loading problem is likely to be one of two things:
1. Not using current CVS of both asterisk and chan_sccp
OR
2. Having your asterisk headers/source not match the running asterisk
(perhaps a forgotten make install?)

If you need more help feel free to drop me a private e-mail with more
info and I'll give you all the help I can.


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Re: [Asterisk-Users] Problem compiling chan_sccp

[Julien Goodwin]

On Mon, Aug 16, 2004 at 04:46:56PM -0500, Lex Lethol arranged a set of bits into the 
following:
> I recently bought a 7910.  I found out too late that it would not do
> SIP as I initially thought.  Anyway before ditchingit for a 7960 I
> wanted to try it out, I read that the guys at
> http://chan-sccp.sourceforge.net/ had done some improvements to the
> original chan_sccp driver and having 80% functionality with this
> model.
> 
> I have not been able to compile their driver and keep getting the following:

> chan_sccp$ make
> Now compiling  sccp_channel.c   264 lines 
> sccp_channel.c: In function `sccp_channel_endcall':
> sccp_channel.c:234: parse error before `timer'
> sccp_channel.c:237: `r1' undeclared (first use in this function)
> sccp_channel.c:237: (Each undeclared identifier is reported only once
> sccp_channel.c:237: for each function it appears in.)
> sccp_channel.c:238: `cmtime' undeclared (first use in this function)
> make: *** [.tmp/sccp_channel.o] Error 1
Are you running GCC 2.95? If so there might still be a few cleanup
patches to fix compilation that haven't hit CVS yet. (And unfortunatly
anon CVS is down at the moment thanks to SourceForge...)

Take a look at my patch available at:
http://www.czmok.de/devtrack/bug_view_advanced_page.php?bug_id=033

(I run chan_sccp for cisco 12SP phones and that was the patch I needed
to get it working under GCC 2.95)


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