[Asterisk-Users] ACD/Queue Support with SIP Notification Messages?
Matthew Jones
matthew.jones at itransact.com
Mon Oct 18 00:08:01 MST 2004
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:8114 at 10.0.5.200 SIP/2.0
Via: SIP/2.0/UDP 10.0.5.114;branch=z9hG4bK62929667B3834ABA
From: "Dan Bailey" <sip:8114 at 10.0.5.200>;tag=106DAB53-44C9D8A8
To: <sip:8114 at 10.0.5.200>;tag=as3e119269
CSeq: 29 NOTIFY
Call-ID: ae360fd7-c5adf605-cf3702aa at 10.0.5.114
Contact: <sip:8114 at 10.0.5.114>
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:8114 at 10.0.5.200">
<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:8114 at 10.0.5.200>;tag=106DAB53-44C9D8A8
To: <sip:8114 at 10.0.5.200>;tag=as3e119269
Call-ID: ae360fd7-c5adf605-cf3702aa at 10.0.5.114
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, asterisk-users-request at lists.digium.com
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)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Sun, 17 Oct 2004 23:46:17 -0400
> From: Andrew Kohlsmith <akohlsmith-asterisk at benshaw.com>
> Subject: Re: [Asterisk-Users] Re: Advice on OS Choice
> To: asterisk-users at lists.digium.com
> Message-ID: <200410172346.17791.akohlsmith-asterisk at benshaw.com>
> 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 <akohlsmith-asterisk at benshaw.com>
> Subject: Re: [Asterisk-Users] Re: Advice on OS Choice
> To: asterisk-users at lists.digium.com
> Message-ID: <200410172347.22579.akohlsmith-asterisk at benshaw.com>
> 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 <akohlsmith-asterisk at benshaw.com>
> Subject: Re: [Asterisk-Users] Re: Advice on OS Choice
> To: asterisk-users at lists.digium.com
> Message-ID: <200410172351.58730.akohlsmith-asterisk at benshaw.com>
> 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" <asterisk at pineview.net>
> Subject: [Asterisk-Users] chan_h323: forcing 20ms packetisation
> To: asterisk-users at lists.digium.com
> Message-ID: <417340F2.8070902 at pineview.net>
> 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" <cguy at bigpond.net.au>
> Subject: [Asterisk-Users] Petulant losers thread [Advice on OS Choice]
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>
> Message-ID: <0a8f01c4b4c8$2551e740$0200a8c0 at southpark.craig.com>
> 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" <akohlsmith-asterisk at benshaw.com>
> To: <asterisk-users at lists.digium.com>
> 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
>> Asterisk-Users at lists.digium.com
>> 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 <gulzar10 at yahoo.com>
> Subject: [Asterisk-Users] Problem In RTC Client When Used With
> Asterisk
> To: asterisk-users at lists.digium.com
> Message-ID: <20041018042737.10266.qmail at web21201.mail.yahoo.com>
> 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
>
>
>
> _______________________________
> Do you Yahoo!?
> Declare Yourself - Register online to vote today!
> http://vote.yahoo.com
>
>
> ------------------------------
>
> Message: 7
> Date: Sun, 17 Oct 2004 22:46:25 -0600 (MDT)
> From: Greg Hill <gregh-asterisk at hillnet.us>
> Subject: Re: [Asterisk-Users] Asterisk dropping last digit of phone
> number
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <Pine.LNX.4.44.0410172243560.3823-100000 at hillnet.us>
> 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 <kagato at chaosium.net>
> Subject: [Asterisk-Users] Thailand
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <20041018045606.GB21004 at chaosium.net>
> 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 <usedcanon at yahoo.co.uk>
> Subject: Re: [Asterisk-Users] compiling cdr_mysql on AMD64 fedora core
> 2
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <1098075702.31295.6.camel at localhost.localdomain>
> 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
>>
>> [root at ast1 asterisk-addons]#
>>
>>
>>
>> ______________________________________________________________________
>> _______________________________________________
>> Asterisk-Users mailing list
>> Asterisk-Users at lists.digium.com
>> 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 <danishsamad at gmail.com>
> Subject: Re: [Asterisk-Users] Problem In RTC Client When Used With
> Asterisk
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <dbeeccc604101722157ad66b9e at mail.gmail.com>
> 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" <linus at magrathea-telecom.co.uk>
> Subject: Re: [Asterisk-Users] Unusual protocols
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>
> Message-ID: <007901c4b4d4$fab35660$0c00000a at ARIADNE>
> 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 <trooster at interstroom.nl>
> Subject: Re: [Asterisk-Users] SNOM 190 "Dial-Plan String" Settings
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> <asterisk-users at lists.digium.com>
> Message-ID: <C92EE90C-20CA-11D9-A697-000393D3E576 at interstroom.nl>
> 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
>> Asterisk-Users at lists.digium.com
>> 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" <simon at auit.net>
> Subject: [Asterisk-Users] Asterisk AGI 'Get Data' escape digits not
> working on long records
> To: <asterisk-users at lists.digium.com>
> Message-ID: <20041018060222.228DF2FE009 at lists.digium.com>
> 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" <itsyourgrave at gmx.de>
> Subject: [Asterisk-Users] cross-connecting dynamic channels
> To: Asterisk-Users at lists.digium.com
> Message-ID: <30511.1098082476 at www53.gmx.net>
> 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
>
>
> --
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>
>
>
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>
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