All,
We are using Polycom SoundPoint IP 500 phones that support acd-login-logout and acd-agent-availability functions on the phone in softbuttons.
Enabling these, I can see the SIP notifications coming through when the user is avail/unavail, but no idea how to get this to interface with the queue status.
The goal is to have an agent's status show up on the phone so they can visually tell if they are logged in or out. We are using callback support rather than parking agents on a line.
We have extensions set up for that, but have problems with agents not knowing their status or walking away from the phone without logging out.
If anyone has a way to just dial the login/logout extensions from a soft/fixed button that would work as well, just trying to sort a way to change something on the phone as an indicator.
Any ideas?
SIP info transmitted on setting avail status is:
Sip read: NOTIFY sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 10.0.5.114;branch=z9hG4bK62929667B3834ABA From: "Dan Bailey" <sip:[EMAIL PROTECTED]>;tag=106DAB53-44C9D8A8 To: <sip:[EMAIL PROTECTED]>;tag=as3e119269 CSeq: 29 NOTIFY Call-ID: [EMAIL PROTECTED] Contact: <sip:[EMAIL PROTECTED]> Event: presence User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.1 Subscription-State: active;expires=646 Max-Forwards: 70 Content-Type: application/pidf+xml Content-Length: 196
<?xml version="1.0" encoding="UTF-8"?>
<presence xlmns="urn:ietf:params:xml:ns:pidf" entity="sip:[EMAIL PROTECTED]">
<tuple id=1023">
<status><basic>open</basic></status>
</tuple>
</presence>
13 headers, 6 lines Transmitting (no NAT): SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.5.114;branch=z9hG4bK62929667B3834ABA From: "Dan Bailey" <sip:[EMAIL PROTECTED]>;tag=106DAB53-44C9D8A8 To: <sip:[EMAIL PROTECTED]>;tag=as3e119269 Call-ID: [EMAIL PROTECTED] CSeq: 29 NOTIFY User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: Content-Length: 0
On Oct 18, 2004, at 1:54 AM, [EMAIL PROTECTED] wrote:
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Today's Topics:
1. Re: Re: Advice on OS Choice (Andrew Kohlsmith) 2. Re: Re: Advice on OS Choice (Andrew Kohlsmith) 3. Re: Re: Advice on OS Choice (Andrew Kohlsmith) 4. chan_h323: forcing 20ms packetisation (Mike O'Connor) 5. Petulant losers thread [Advice on OS Choice] (Craig Guy) 6. Problem In RTC Client When Used With Asterisk (Gulzar Hussain) 7. Re: Asterisk dropping last digit of phone number (Greg Hill) 8. Thailand (Jayson Vantuyl) 9. Re: compiling cdr_mysql on AMD64 fedora core 2 (Umar Sear) 10. Re: Problem In RTC Client When Used With Asterisk (Danish Samad) 11. Re: Unusual protocols (Linus Surguy) 12. Re: SNOM 190 "Dial-Plan String" Settings (Joris Trooster / Interstroom) 13. Asterisk AGI 'Get Data' escape digits not working on long records (Simon Smith) 14. cross-connecting dynamic channels (Katharina Rasch)
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Message: 1 Date: Sun, 17 Oct 2004 23:46:17 -0400 From: Andrew Kohlsmith <[EMAIL PROTECTED]> Subject: Re: [Asterisk-Users] Re: Advice on OS Choice To: [EMAIL PROTECTED] Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="iso-8859-1"
On October 16, 2004 04:49 pm, Joe Greco wrote:As a manufacturer, you build things and sell them, and you can recommend
whatever policies you like, but after it leaves the shipping department,
you're out of luck as to being able to guarantee any of that.
Then, as a manufacturer, you should not be liable for what some dickhead in a
service department is doing to it. :-)
Like I said in my last message, litigation has a way of making things nonsensical.
Firmware that boots checks image (or critical parts of image) for tampering against stored checksum (checksum that gets updated when correct update procedure is followed) -- Putz away, the firmware will still bring you to a full stop because it detected a problem.
That's highly complex; even Sun agreed there was no practical way to do it.
With a closed source system, it wasn't considered a risk, and since
everything up to the point where we received control from the OS was at
least very difficult to putz with, it wasn't checked /prior/ to execution.
Verification of the loaded kernel image happened after it was loaded, and
was designed specifically to catch things like disk blocks going bad.
I dunno -- crytographically sign the images and verify signature on boot.
Hell even a field hard drive swap would work in this case.
Again, the black box approach has advantages. Could you maybe engineer
something to verify stuff at each and every step, just so you could go open
source? Sure, perhaps, but at additional cost for more flash, and
additional cost for more development, and bad things then happen if you
do a field swap on hard drives to fix a broken unit, etc., and really it
becomes impractical.
See above.
That's nice in theory, but potentially pretty darn expensive. Nobody
seemed to think that it was worth the trouble, expense, etc., to get so
paranoid about it.
That's what I don't understand -- they're sufficiently paranoid when it comes
to providing source, but security through obscurity is good enough to get
past the legal department. Curious, really.
To upgrade you can install the CD or reimage the drive with the new image, but you have to also replace the vendor key.
And how do you do /that/? You now need to have a keyboard attached to the
system to enter and replace the key?
physical cartridge or smartcard that was shipped with the updated firmware,
and "signed off" by someone who has the access code to authorize the firmware
update. I dunno.
Cryptographic signature on the images with the CA being the company releasing
the firmware is even easier.
The point is that's all software. If it's open to inspection and
recompilation, it's easily open to defeat. I can make an init system that
is very difficult to reverse-engineer, complete with interlocks with any
other items that get launched, such that NOTHING happens unless that
process is happy, but if that can be replaced by an init that doesn't give
a fsck, because someone commented out all the code and recompiled it, then
we have trouble.
*sigh* -- this is why I am saying that the boot firmware needs to make these
checks, not the stuff you can tinker with when you have the source.
Bootloaders only boot the end software, they're usually not too complex and
once done require little to no maintenance. Keep *that* black boxed. Put
the interlocks *there* -- your core system is still open to many eyes and a
lot of scrutiny.
So, yes, you /could/ design such a system, and if you've open sourced all
your software, then you probably /have/ to.
I would go on to say that you should have those checks and balances in place
whether it was open or not... Hell those DURN TERRAISTS might decide to put
rogue firmware out to make all the nuclear medicine machinery go critical.
Yes, this is getting silly.
We're talking specifically about the case where distributing the source
makes it trivial for someone to work around those correct checks and
balances.
You can't work around a check and balance like that -- firmware doesn't like
the signature, it don't start up the executable. Capiche?
We're talking about open-sourcing the main software, not the ROM bootloader
(for lack of a better word: BIOS).
No, I'm not worried about that. The specific case that was of concern was
what happens when someone from the hospital campus electronics shop tampers
with the system, something bad happens, and then the system is reloaded
with a non-tampered copy, because hospital policy would be to send a
defective device back to the shop?
These devices don't have some kind of audit log in them? Jesus.
Trusted computing is always a difficult thing. At a certain point, you
need to draw the line. Because we had a closed source solution, we were
able to fairly safely assume that when we got handed off at init, we had
a system which was likely in a known state, and could verify the loaded
kernel/module/firmware/etc images, which was considered extremely
sufficient paranoia. The point is that re-engineering a whole system with
more checks, firmware, keys, requirements, adding a keyboard, etc., just
so you can use GPL'd software is really a non-starter, so in the end, only
BSD licensed projects were used and only BSD licensed projects received
the benefits of having some of our engineers working on, debugging, and
improving those projects.
I wasn't saying anything about a keyboard or implementing everything -- having
the bootloader verify the system image would have been sufficient and I gave
several ways to ensure that. I also gave several ways to ensure that a new
image was "authorized" by someone who could be held liable. adding $250 or
even $2500 to a $50k machine for this kind of safety -- closed or open source
-- just seems like good karma to me.
-A.
------------------------------
Message: 2 Date: Sun, 17 Oct 2004 23:47:22 -0400 From: Andrew Kohlsmith <[EMAIL PROTECTED]> Subject: Re: [Asterisk-Users] Re: Advice on OS Choice To: [EMAIL PROTECTED] Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="iso-8859-1"
On October 16, 2004 05:05 pm, Matt Riddell wrote:Joe, could we stop this now? It's obvious that if you go to a GPL project and start slinging mud at the GPL, you are in the wrong place. I would recommend that you head over to a Microsoft mailing list where I'm sure you will find an abundance of fodder for your outdated methodologies.
Just my opinion: he's not slinging mud at the GPL, he's (trying) to give a
scenario where open-source is a Bad Thing. I get the impression that he's
rather happy with the GPL in general.
-A.
------------------------------
Message: 3 Date: Sun, 17 Oct 2004 23:51:58 -0400 From: Andrew Kohlsmith <[EMAIL PROTECTED]> Subject: Re: [Asterisk-Users] Re: Advice on OS Choice To: [EMAIL PROTECTED] Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="iso-8859-1"
On October 17, 2004 11:34 pm, Andrew Kohlsmith wrote:On October 16, 2004 02:24 pm, Michael Giagnocavo wrote:
?? wtf happened to my list threading?
-A.
------------------------------
Message: 4 Date: Mon, 18 Oct 2004 13:35:06 +0930 From: "Mike O'Connor" <[EMAIL PROTECTED]> Subject: [Asterisk-Users] chan_h323: forcing 20ms packetisation To: [EMAIL PROTECTED] Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Hi all
I spent a few hours trying to information on asterisk, h323 and sip support for codecs with 20ms packetisation, and have not been able to find anything relivatant.
Our supplier of call termination requires h323 the following:
* The signalling port is 1720 * H.323 version 2 with fast start and H.245 Tunneling. * The call should be initialised as Gateway-Gateway not using RAS. * The codecs supported are G.729, G.711alaw and G.711ulaw all at 20 millisecond packetisation. Your equipment must support all three and be able to dynamically negotiate these during call setup. * We use RFC 2833 for out-of-band DTMF. Your equipment must support this. The NTE RTP Payload type supported is 99.
I was able after reading the source code in chan_h323.c to work out how to enable fast start and h.245 tunneling.
But the 20ms packetisation has me beat.
I have made a test call to the provider which did not work becase I was sending 30ms voice packets.
SO my question does any one know now to force the correct voice packet size ?
Thanks
Mike
------------------------------
Message: 5 Date: Mon, 18 Oct 2004 12:08:37 +0800 From: "Craig Guy" <[EMAIL PROTECTED]> Subject: [Asterisk-Users] Petulant losers thread [Advice on OS Choice] To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[EMAIL PROTECTED]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="iso-8859-1"
Can all parties concerned drop this thread or take it offline.
Craig
----- Original Message ----- From: "Andrew Kohlsmith" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Monday, October 18, 2004 11:51 AM Subject: Re: [Asterisk-Users] Re: Advice on OS Choice
On October 17, 2004 11:34 pm, Andrew Kohlsmith wrote:On October 16, 2004 02:24 pm, Michael Giagnocavo wrote:
?? wtf happened to my list threading?
-A. _______________________________________________ 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
------------------------------
Message: 6 Date: Sun, 17 Oct 2004 21:27:37 -0700 (PDT) From: Gulzar Hussain <[EMAIL PROTECTED]> Subject: [Asterisk-Users] Problem In RTC Client When Used With Asterisk To: [EMAIL PROTECTED] Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=us-ascii
Hi When I call from 1 RTC Client to another without Asterisk everything use to be fine but when asterisk is there as a Registrar a problem use to occur in many calls, Caller can hear the voice of the receiving side but the receiver cant be able hear the caller for about 5 to 10 seconds, conversation will become two way after 5 - 10 seconds but this problem is a big hurdle in proper establishment of a call
Does anybody ever had this problem ? Any suggestions will be higly apreciated Thanx in Advance
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------------------------------
Message: 7 Date: Sun, 17 Oct 2004 22:46:25 -0600 (MDT) From: Greg Hill <[EMAIL PROTECTED]> Subject: Re: [Asterisk-Users] Asterisk dropping last digit of phone number To: Asterisk Users Mailing List - Non-Commercial Discussion <[EMAIL PROTECTED]> Message-ID: <[EMAIL PROTECTED]> Content-Type: TEXT/PLAIN; charset=US-ASCII
On Mon, 18 Oct 2004, Demian wrote:
I've recently installed and configured Asterisk. I'm having some problems with phone numbers which look like 1 021 123 4567
(1 for an outside line) and then the phone number. Asterisk will always
drop off the last digit and dial 1021123456 instead. I thought this was
a problem with my contexts however I've recently added a SIP phone and
it's initial context is the same as the analogue phones that display
this problem.... the SIP phone works fine. Any ideas where I should be
looking?
I'd start in extensions.conf.. double-count your X's (or N's) in the
exten=> lines to make sure they match the number you're trying to dial.
You didn't mention much detail about how the analogue calls get into your
*, nor how calls get out. I guess it shouldn't matter much; they'll all
get routed through extensions.conf regardless.
Greg
------------------------------
Message: 8 Date: Sun, 17 Oct 2004 23:56:06 -0500 From: Jayson Vantuyl <[EMAIL PROTECTED]> Subject: [Asterisk-Users] Thailand To: Asterisk Users Mailing List - Non-Commercial Discussion <[EMAIL PROTECTED]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=us-ascii
What does anyone know about signalling in Thailand? Are there any issues with using Digium T1 or FXO/FXS cards there?
-- Jayson Vantuyl
------------------------------
Message: 9 Date: Mon, 18 Oct 2004 06:01:42 +0100 From: Umar Sear <[EMAIL PROTECTED]> Subject: Re: [Asterisk-Users] compiling cdr_mysql on AMD64 fedora core 2 To: Asterisk Users Mailing List - Non-Commercial Discussion <[EMAIL PROTECTED]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain
I had simillar issues (not the same maybe) with Centos 3.3 X64.
The first was becuase I had asterisk compile in /usr/src/asterisk-1.0.1 rather than /usr/src/asterisk.
creating a symbolic link took the build process further but still
failed. This time it was to do with the fact that it was looking for the
mysql libs in /usr/lib/mysql whilst being x64 they were installed in
/usr/lib64/mysql. Once again creating a symbolic link fixed that and I
was able to compile clean.
I hope this helps you diagnose the issue that you are having (my guess is that the error you are reporting is simmillar to the first error I had)
Umar.
On Sat, 2004-10-16 at 21:52, david winter wrote:I got this error when installing cdr_mysql on an AMD64 running fedora core 2. Anyone have ideas on what is wrong?
gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6 -m64 -c -o format_mp3.o format_mp3.c
gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6 -m64 -shared -Xlinker -x -o format_mp3.so common.o dct64_i386.o decode_ntom.o layer3.o tabinit.o interface.o format_mp3.o
/usr/bin/ld: common.o: relocation R_X86_64_32 can not be used when making a shared object; recompile with -fPIC
common.o: could not read symbols: Bad value
collect2: ld returned 1 exit status
make[1]: *** [format_mp3.so] Error 1
make[1]: Leaving directory `/home/dwinter/src/asterisk-addons/format_mp3'
make: *** [format_mp3/format_mp3.so] Error 2
[EMAIL PROTECTED] asterisk-addons]#
______________________________________________________________________ _______________________________________________ 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
------------------------------
Message: 10 Date: Mon, 18 Oct 2004 10:15:12 +0500 From: Danish Samad <[EMAIL PROTECTED]> Subject: Re: [Asterisk-Users] Problem In RTC Client When Used With Asterisk To: Asterisk Users Mailing List - Non-Commercial Discussion <[EMAIL PROTECTED]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=US-ASCII
HI,
I have used RTC with other SIP Proxies like SER and party sip
and it works fine, never tested it with asterisk though.
Basically Asterisk initiallly proxies RTP through itself and then
sends reinvites to both endpoints to make RTP flow directly between
the two gateways.
Asterisk does have problems with the packetization perid values.
It might be the case that the RTC endpoints use a different packetization
period as compared to asterisk and it is only when the RTP goes direct,
the endpoints start using the same packetization.
Whatever the problem maybe, I would suggest capturing SIP and media packets on both server and client side and analyzing them. You can use ethereal (www.ethereal.com) for this purpose, it is an extremely useful opensource network analyzer.
Hope this helps, Danish
------------------------------
Message: 11 Date: Mon, 18 Oct 2004 06:40:30 +0100 From: "Linus Surguy" <[EMAIL PROTECTED]> Subject: Re: [Asterisk-Users] Unusual protocols To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[EMAIL PROTECTED]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; format=flowed; charset="iso-8859-1"; reply-type=response
examples of things which I have actually been asked about. There are a
number of protocols based in 2600Hz tones (most US) and 2280Hz tones
(mostly Europe), which are probably still spread quite widely in low
density point-to-point connections. If there is anything you need, please
tell me about it. I want to build a picture of what might be worthwhile
tackling.
You probably won't go far wrong by looking at the support offered by www.aculab.com and trying to match it .
Linus
------------------------------
Message: 12 Date: Mon, 18 Oct 2004 07:58:51 +0200 From: Joris Trooster / Interstroom <[EMAIL PROTECTED]> Subject: Re: [Asterisk-Users] SNOM 190 "Dial-Plan String" Settings To: Asterisk Users Mailing List - Non-Commercial Discussion <[EMAIL PROTECTED]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Hello James,
There is nothing special with the Snom phones. The empty dialplan string is normal. You only have to specify the displayname, account, password and registrar. I think you have a mistake in your extensions.conf. Does it work with another (soft)phone?
Regards, Joris
On Oct 15, 2004, at 1:51 PM, James Bean wrote:
I am having a problem with my new SNOM190 and my asterisk box. Incoming calls to the SNOM work perfectly, but when i dial-out I get a "Not Found: <number dialed>" on the SNOM display everytime I try, nothing shows up on the console of the asterisk box so its not even touching it. I have the latest 3.54 firmware on it and when I looked at the Line 1 setup for my asterisk box I released that in the SNOM phone there is nothing in my "Dial-Plan String" I take it it matches this inside the phone to choose which line to use in the SNOM phone. Unfortunately I am not finding much on the format of the Dial-Plan String in the SNOM phones. All I need is for it to send all calls regardless of format to the asterisk box. Anyone got any suggestions. James _______________________________________________ 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
------------------------------
Message: 13 Date: Mon, 18 Oct 2004 16:02:02 +1000 From: "Simon Smith" <[EMAIL PROTECTED]> Subject: [Asterisk-Users] Asterisk AGI 'Get Data' escape digits not working on long records To: <[EMAIL PROTECTED]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="us-ascii"
Hoping someone can please help me.
I have written an AGI application (that uses the Asterisk-AGI perl library)
that processes requests to record wav files, capture dtmf, return dtmf etc
to my dial plan.
It works well, except when I record a long recording ( I have not been able
to figure out a direct pattern - but approximately 40 minutes or longer of
total recording in MSGSM format) It will no longer respond to my DTMF escape
digits.
In my agi-test.agi file I simply something similar to the following. $result = $AGI->record_file($wavfile, WAV, 12345 , 70000, 1);
As expected it will wait for up to 1 digit and return the value in ASCII
into $result
HOWEVER
I need it to sometimes record up to a maximum of 3 hours. (1080000 ms)
$result = $AGI->record_file($wavfile, WAV, 12345 , 1080000, 1);
But it gets to maybe more than half an hour, is still recording fine but NO
MATTER WHAT digits i press, it never escapes from this command when i
constantly try pressing any of the escape digits.
Does anyone have an insight or similar issue? I wish i could resolve this
one, it is killing me.
Thanks
Simon
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Message: 14 Date: Mon, 18 Oct 2004 08:54:36 +0200 (MEST) From: "Katharina Rasch" <[EMAIL PROTECTED]> Subject: [Asterisk-Users] cross-connecting dynamic channels To: [EMAIL PROTECTED] Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="us-ascii"
Hi,
is it possible to cross-connect dynamic channels? I was trying to do someting like this in zaptel.conf:
#first interface dynamic = eth,eth1/00:40:F4:A4:7C:5C,24,2 bchan=1-23 dchan=24
#second interface dynamic = eth,eth0/00:40:F4:A4:7D:FE,24,2 bchan=25-47 dchan=48
dacs=1-24:25
but ztcfg is always giving me back something like:
line 160: Channel 1 already configured as 'Individual Clear channel' at line
149
...
line 160: Channel 24 already configured as 'D-channel' at line 150
Can something like this be done, and if so, how should i configure the channels?
thanks a lot katharina
-- GMX ProMail mit bestem Virenschutz http://www.gmx.net/de/go/mail +++ Empfehlung der Redaktion +++ Internet Professionell 10/04 +++
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