[Asterisk-Users] Capturing calls in asterisk

[albertoocdc at mundo-r.com]

Hi.

Is possible to caprure calls with asterisk?

I have a calling from onde device to another. While it´s ringing I´d wish to capture the calling from another device which has permissions to make it. is it possible? 


>Send Asterisk-Users mailing list submissions to
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>
>Today's Topics:
>
>   1. Sourcing H/W for Asterisk in India :: Digium/Intel	Modems and
>      IP Phones (Salil Khamkar)
>   2. ACD/Queue Support with SIP Notification Messages? (Matthew Jones)
>   3. Re: Intervivo sip.conf? (Mark Turner)
>   4. (Another) Queue log analyser (Shad Mortazavi)
>
>
>----------------------------------------------------------------------
>
>Message: 1
>Date: Mon, 18 Oct 2004 12:29:15 +0530
>From: "Salil Khamkar" <salil at vsnl.com>
>Subject: [Asterisk-Users] Sourcing H/W for Asterisk in India ::
>	Digium/Intel	Modems and IP Phones
>To: <asterisk-users at lists.digium.com>
>Message-ID: <200410180707.MAA29808 at manage.24online>
>Content-Type: text/plain; charset="us-ascii"
>
>Hi All,
> 
>Does anybody on this list know where I can get Digium FXO/Intel 735, Digium
>FXS boards in India ?
> 
>Similarly I am also trying to lay my hands on the Grandstream IP phones but
>have been unable to find a source. 
> 
>Thanks
>--
>Salil 
> 
> 
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>------------------------------
>
>Message: 2
>Date: Mon, 18 Oct 2004 02:08:01 -0500
>From: Matthew Jones <matthew.jones at itransact.com>
>Subject: [Asterisk-Users] ACD/Queue Support with SIP Notification
>	Messages?
>To: asterisk-users at lists.digium.com
>Message-ID: <730B6AFC-20D4-11D9-84E3-000A95CC993C at itransact.com>
>Content-Type: text/plain; charset=ISO-8859-1; delsp=yes; format=flowed
>
>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:
>
>> Send Asterisk-Users mailing list submissions to
>> 	asterisk-users at lists.digium.com
>>
>> To subscribe or unsubscribe via the World Wide Web, visit
>> 	http://lists.digium.com/mailman/listinfo/asterisk-users
>> or, via email, send a message with subject or body 'help' to
>> 	asterisk-users-request at lists.digium.com
>>
>> You can reach the person managing the list at
>> 	asterisk-users-owner at lists.digium.com
>>
>> When replying, please edit your Subject line so it is more specific
>> than "Re: Contents of Asterisk-Users digest..."
>>
>>
>> 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
>>
>>
>> -- 
>> GMX ProMail mit bestem Virenschutz http://www.gmx.net/de/go/mail
>> +++ Empfehlung der Redaktion +++ Internet Professionell 10/04 +++
>>
>>
>>
>> ------------------------------
>>
>> _______________________________________________
>> Asterisk-Users mailing list
>> Asterisk-Users at lists.digium.com
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>>
>> End of Asterisk-Users Digest, Vol 3, Issue 234
>> **********************************************
>
>
>
>------------------------------
>
>Message: 3
>Date: Mon, 18 Oct 2004 08:26:35 +0100 (BST)
>From: Mark Turner <mark at kram.org>
>Subject: Re: [Asterisk-Users] Intervivo sip.conf?
>To: Asterisk Users Mailing List - Non-Commercial Discussion
>	<asterisk-users at lists.digium.com>
>Message-ID:
>	<Pine.LNX.4.44.0410180819560.22304-100000 at kram.vm.bytemark.co.uk>
>Content-Type: TEXT/PLAIN; charset=US-ASCII
>
>Hi Dave,
>
>On Sun, 17 Oct 2004, David Croft wrote:
>> 
>> I have tried your config and variations on it but have the same problems.
>
>Sorry to hear that you're still having problems.  If you email me your
>sip.conf and extensions.conf then I'd be happy to take a look.
>
>> Placing a call out using intervivo, regardless of dtmfmode setting, DTMF 
>> tones are not recognised by the recipient. The same applies to receiving 
>> dtmf digits.
>
>I did mention that I never got around to making DTMF work from my home
>Asterisk server, but it will be possible.  My guess is that there is
>a mis-match between the DTMF mode settings at either end, i.e. in your
>config and in our server config.  We have a (hidden by default) config
>option on your control panel that allows you to specify the DTMF mode
>manually, which should allow us to fix this for you.
>
>> Also, unless I set insecure=very (which I shouldn't need to), I get 
>> these messages when someone tries to call in:
>> 
>> Oct 16 18:08:21 NOTICE[7175]: chan_sip.c:7162 handle_request: Failed to 
>> authenticate user "xxx" <sip:xxx at 217.168.22.129>;tag=as30592e8c
>> 
>> where xxx is the number they're calling from. They get a busy signal.
>> 
>> Any ideas?
>
>I'm sure we'll sort it once I've seen your config files.
>
>Cheers,
>
>Mark.
>
>p.s. If you're not keen on emailing your config files to my home address
>(why should you believe that I really work for InterViVo) then feel
>free to email them to support at intervivo.net instead and I'll grab them
>from there.
>
>
>
>------------------------------
>
>Message: 4
>Date: Mon, 18 Oct 2004 04:09:32 -0400
>From: Shad Mortazavi <Shad.Mortazavi at nexusmgmt.com>
>Subject: [Asterisk-Users] (Another) Queue log analyser
>To: asterisk-users at lists.digium.com
>Message-ID:
>	<A44C6568A3183F44B64F234489449D88015FE675 at pwmexch1.nexusmgmt.com>
>Content-Type: text/plain; charset="us-ascii"
>
>Ben,
>
>I would definitely have use for this application, fantastic start. When will
>you be making the source available?
>
>In my reports I use the CLID to look at calls for different agents i.e. call
>volume by agent.
>
>Warm Regards
>
>Shad Mortazavi
>-----------------------
>Nexus Technical Manager
>n|m Nexus Management Inc 
>Neutral Bay
>Sydney
>
>
>Message: 4
>Date: Fri, 15 Oct 2004 09:33:26 +0100
>From: "Ben Merrills" <ben at griffin.com>
>Subject: RE: [Asterisk-Users] (Another) Queue log analyser
>To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>	<asterisk-users at lists.digium.com>
>Message-ID:
>	
><F8A414D9A70EB941ADE47043DBE5B702579735 at exchange.network.griffin.net.uk>
>	
>Content-Type: text/plain;	charset="us-ascii"
>
>Hi there,
>
>Cheers for your suggestions, would be great to see the output of some other
>reports. 
>
>Logins and logouts are available within the engine, just need to represent
>them in some way now. What do you suggest would be a good format? Typical
>duration of login? Only problem might be where someone hasn't logged out
>before their next login statement (no one here ever logs out, because
>they're all to slack :)
>
>Anything you can send me over would be much appreciated, I have no problems
>in giving you a pre-release copy so you can give some feedback too.
>
>Regards,
>
>Ben Merrills
>Griffin Internet
>
>T: 0870 8040862
>
>-----Original Message-----
>From: asterisk-users-bounces at lists.digium.com
>[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Wayne Sheppard
>Sent: 14 October 2004 19:08
>To: Asterisk Users Mailing List - Non-Commercial Discussion
>Subject: Re: [Asterisk-Users] (Another) Queue log analyser
>
>Very nice work Ben, thanks. Here are some additional thoughts -
>
>One segmentation that might be useful would be to add outbound calling
>activities as a either a separate column or even view.
>
>On agent stats, it would be useful to see login/logout stamps, login time,
>ready/not ready time (if this can be tracked, not sure).
>
>If you would like, I can send you some example reports that are used in a
>typical call center, contact me directly if you would find that helpful.
>
>Cheers,
>Wayne
>
>Ben Merrills wrote:
>
>>I've been doing some work on a queue log analyser for a while now, 
>>getting the basics in place, an example of which you can find at the
>URL
>>below. However, just wondering what information people think is most 
>>useful in a log analyser?
>>
>>At present it includes the following features:
>>
>># Time periods - specify a period of days from the log which you want
>to
>>generate statistics for (e.g. only the last 14 days) # Templating - 
>>allows the stats to be inserted into any html/text template using 
>>specific tags to insert stats. This means you could create a number of 
>>templates and execute the analyser against them to give different 
>>information on different pages (quite flexible).
>># Specify start and end dates - similar to the first feature, except
>you
>>can specify a tight period from your log, not just the last x number of 
>>days # Channels/Agents to names - simple text file allows you to 
>>specify a name, agent number and a channel - e.g. Ben, Agent/1, 
>>Sip/ben. This is
>>then used in the output #   instead of raw data
>># JPG graphs - includes a custom class to generate line graphs of 
>>information (e.g. hourly call volumes etc)
>>
>>What I want to know though is, what output people would like. At the 
>>moment there is an overview of all queues, which includes:
>>
>>Total Calls, total connected calls, total abandoned calls, calls 
>>abandoned within x seconds, calls exited with key press, Average hold 
>>time, max hold time, average talk time
>>
>>Agent overview includes:
>>Calls taken, Average talk time
>>
>>Graph of call volume per hour of the day Graph of call volume per day 
>>(over the period specified)
>>
>>Runs under windows (.NET or mono required) or any other OS that support 
>>.NET/mono (Linux, Mac, BSD etc)
>>
>>http://muad.xdev.net/Projects/qig/sample.html
>>
>>
>>Not really done anything like this before, so as much input as possible 
>>would be appreciated.
>>
>>Cheers,
>>
>>Ben
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>End of Asterisk-Users Digest, Vol 3, Issue 235
>**********************************************