Re: [Asterisk-Users] NuFone, VoIPJet, circuit (fast) busy question
If I recall correctly Fast Busy basically means the destination number is not busy (regular busy) but your provider most likely is either over loaded, or has some other issues. I've been getting busy signals with Nufone pretty regularly over the last few days, and there email support is not responding as usual. There front page also says they are no longer accepting new customers due to system upgrades, maybe that has something to do with it. Who knows... On Fri, 2005-04-01 at 09:09 +0400, Jean-Michel Hiver wrote: I've noticed that nufone returns 'circuit busy' messages FAST (when it does) while this tends to take a while with VoIPJet. I've also seen 'circuit fast busy' message - what is the difference between the two? Thanks, Jean-Michel. -- Mike Benoit [EMAIL PROTECTED] signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] *** Asterisk 2.0 Stable release out now
Olle E. Johansson wrote: During the developer's conference call yesterday evening, it was decided that we finally should release the much-awaited Asterisk 2.0 Stable release, also called codename AAFJ. AAFJ as in Asterisk April Fool's Joke? Nice :) ___ Asterisk-Users mailing list Asterisk-Users@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
[Asterisk-Users] Playback starts before call answer
Greetings *`s, When initiating a call to an outside line (in this case a cellphone), * starts playing the sound file before the call is answered, so when the called party picks up, the message is already halfway thru, or completely played out. I have tried a few things to get around this, read up on http://bugs.digium.com/bug_view_page.php?bug_id=0002467 as well as http://www.voip-info.org/wiki-Asterisk+auto-dial+out Has anyone found any workarounds so that the sound file plays only once the call is answered ? -- Chris Blake Cell: 082 775 1492 Work: +27 11 782 0840 Fax : +27 11 782 0841 Mail: [EMAIL PROTECTED] Today is the first day of the rest of your lossage. ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] setting SIP to dial PSTN with TDM400P
to dear martijn,
i made every possible change i can make
i have a TDM400P Zap card...
i had connected PSTN line to FXO Kewlstart at channel 1.
and analog phone to FXS Kewlstart at Channel 4.
i can hear continous ring tone when i hook up the receiver.
plz have a look at my confs.
my extension.conf is as follows;
[pstn-outbound]
exten = _.,1,Dial(Zap/1/${EXTEN})
exten = _.,2,Congestion
my zaptel.conf is as follows:
[channels]
;
; Default language
;
language=en
musiconhold=default
usercallerid=yes
hidecallerid=no
callreturn=yes
callprogress=no
rxwink =300
echotraining=800
rxgain=0.0
txgain=0.0
busydetect=1
busycount=7
immediate=no
signalling=fxo_ks
;callerid=asreceived
context=pstn-outbound
channel=1
relaxdtmf=yes
callwaiting=yes
;
; Support three-way calling
;
threewaycalling=yes
;
; Support flash-hook call transfer (requires three way calling)
;
transfer=yes
when i dial a local number say (6998256) from analog phone set then
asterisk shows following messages.
*CLI -- Starting simple switch on 'Zap/1-1'
-- Executing Dial(Zap/1-1, Zap/1/6998256) in new stack
-- Called 1/6998256
-- Zap/1/6998256-busy-1013475805 is busy
-- Hungup 'Zap/1/6998256-busy-1013475805'
== Everyone is busy/congested at this time
-- Timeout on Zap/1-1
== CDR updated on Zap/1-1
**
please reply with your suggestions i always take care to run ztcfg
command whenever i made any changes to zaptel.conf.plz help me solving
this problem
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Re: [Asterisk-Users] *** Asterisk 2.0 Stable release out now
Good one guys, for a minute you actually had me there. The give away is: Rumours has it that one developer actually ported the Erlang runtime and executed an Ericsson AXE switch within Asterisk. :-) Nir S On Fri, 2005-04-01 at 09:40 +0200, Olle E. Johansson wrote: During the developer's conference call yesterday evening, it was decided that we finally should release the much-awaited Asterisk 2.0 Stable release, also called codename AAFJ. This relaese is based on the hidden cvs that has been in operation for six months by a group of core development members in the Asterisk.org open source project, under the leadership of Brian K. East, who will maintain the stable code base for the 2.0 CVS tree and releases. -It's awsome, says Brian, but the new features I'm adding to 2.0.1 stable will be even more spectacular. Follow me to the future! Among the new features in Asterisk 2.0 is * APBX - A fully pluggable PBX architecture - The APBX framework makes everything in Asterisk 2.0 hot-pluggable and dynamic, including the PBX itself. With this framework, Asterisk 2.0 will be able to be the host system for almost anything, including the famous Apache.org web server, the SipFoundry SIPx PBX and a Java Runtime Engine. Rumours has it that one developer actually ported the Erlang runtime and executed an Ericsson AXE switch within Asterisk. With an embedded web server, we can finally start working on a decent user interface model says Kram Spencer, the original developer of Asterisk. * DBRAGI - The Database Remote procedure call AGI subsystem -- The DBRAGI subsystem makes it possible to move the dial plan processing to stored procedures in databases. With Asterisk 1.2, the ARA (Asterisk Realtime Architecture) took a first step towards a better database integration. With 2.0, the project actually runs most of the PBX within an Oracle (TM) database, making Asterisk carrier grade. * XIAX - The New Inter-Asterisk Protocol -- With Asterisk 2.0, the project also launches the next generation of the IAX protocol. This is a huge update of the rather oldfashioned IAX protocol engine. - XML based messages All messages in XIAX is based on XML. This makes the protocol more robust, since all messages are checked for correct syntax with an external DTD and XML parser. All voice frames are encoded in BASE64 and checked with an S/MIME signature, which makes the XIAX protocol the most secure VoIP protocol in the known universe. - Full DNS NAPTR/SRV support To add to the robustness of the protocol, all communication is done with full DNS service names. For each packet in the data stream, there's full redundancy based on DNS lookups. The recommendation for XIAX is to define at least five XIAX servers per phone number, and let DNS route the XIAX packets. No packet will get lost, due to the stability and simpleness of the DNS system. says Kram. Using IP numbers did not gives us this functionality. - Strong TCP/SSL support The new XIAX protocol also supports TCP with SSL encapsulation. TCP is much easier for the firewall to handle and with strong SSL encryption. With IAX2 we could bypass every NAT device. With XIAX over SSL on the HTTP port, we can traverse any firewall too. says Steve Xintaro, the main architect of XIAX. * New source code structure - C# and .net Asterisk 2.0 was moved to a Microsoft platform due to the demand for higher stability and a more secure foundation. Therefore, the code was quickly moved to C# on the .net platform. This gives Asterisk a lot of new features, including being fully integrated with Microsoft Exchange and Microsoft Active Directory. With all the user data stored in Active Directory, we finally have the user under full control. Users can dial in to the PBX to change their Windows password. We can also implement single-sign-on based on DTMF from a cell phone or WiFi phone. says Kelvin Reming. The C# language gives us much more modern code. And I'm so happy to get rid of the stupid-looking arctic bird, an ugly animal that that couldn't even fly. * New user-support system: SmartyList (TM) In order to solve the problem with the asterisk-users mailing list that was the main support channel for old Asterisk versions, the Asterisk 2 team also constructed the SmartyList auto-support system, that will automatically analyze all input and sort it out on one of twenty different lists. Eighteen of these are automatically handled by auto-responders, that point to the proper Wiki page,
Re: [Asterisk-Users] Installing CAPI
I've got a couple of Fritz! chan_capi installs under my belt here in Australia. I've elected to use the mISDN capi drivers over the AVM ones and it works quite well except for broken DID support, and of course all the limitations of using non Zaptel drivers. Craig - Original Message - From: Leandro Morgado [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, April 01, 2005 7:53 AM Subject: Re: [Asterisk-Users] Installing CAPI Hi, I've used the Fritz AVM PCI card with Junghanns.net chan_capi and it's working great. It's never crashed or given problems, although I have a low call volume (at most 50 calls a day). The setup was not straight forward (fritz drivers compilation, version matching, etc) but it wasn't very dificult with help from the very same links you gave (wiki and junghanns docs). Maybe it's a problem with your ISDN card? I've tried 2 other cards and just couldn't get them to work. The Fritz works great though! Leandro Damian Funnell wrote: Hi there, We recently did our first * install with CAPI and we found the levels of support (and general knowledge) within the community seriously wanting. In fact, we found things so bad that I would caution against using CAPI unless you are feeling particularly game and confident in your abilities to fix problems, as you are likely to find it very difficult to get help if you need it. Out of the half dozen or so help requests that I or my colleagues posted to this forum or to the #asterisk IRC channel, for example, we didn't receive a single helpful response. Not one. Not that there wasn't anyone who was willing to help, but there just didn't seem to be anyone around who was using CAPI in anger. We originally chose CAPI over ISDN4Linux because of the commercial support that was supposedly available through junghanns.net (CAPI also provides a better feature set than ISDN4Linux, but we don't use any of the additional features, so this wasn't a consideration for us), but when we called upon junghanns.net for support it took them so long to respond that we needn't have bothered (we had stumbled across a fix ourselves by the time we got a response from them). If this hasn't scared you off then check out the documentation at http://www.junghanns.net/asterisk/ and the sample files/readme that come with the CAPI source. There is also a fairly good configuration guide at http://www.voip-info.org/wiki-Asterisk+How+to+connect+with+CAPI and the CAPI readme is reproduced at http://www.voip-info.org/wiki-Asterisk+CAPI+Readme. Drop me a mail at damian dot funnell at fff dot co dot nz if you would like me to send you a copy of our conf files so you can see how we're using it. Right now we are trying to diagnose a problem where the voice channels over CAPI fall apart a few times per day, resulting in all external calls having to be terminated. We don't know if this problem is CAPI related, but predictably we haven't been able to find anyone in the community who can help us figure it out. Best regards, Damian. [EMAIL PROTECTED] wrote: Hi! I can't find any instructions of installing capi and chan_capi. Do you know any site with instructions or can you give me step by step help with this. Thank you for your answers This mail sent through L-secure: http://www.l-secure.net/ ___ Asterisk-Users mailing list Asterisk-Users@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 ___ Asterisk-Users mailing list Asterisk-Users@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 ___ Asterisk-Users mailing list Asterisk-Users@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 ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] *** Asterisk 2.0 Stable release out now
On Fri, 2005-04-01 at 09:40 +0200, Olle E. Johansson wrote: During the developer's conference call yesterday evening, it was decided that we finally should release the much-awaited Asterisk 2.0 Stable release, also called codename AAFJ. This relaese is based on the hidden cvs that has been in operation for six months by a group of core development members in the Asterisk.org open source project, under the leadership of Brian K. East, who will maintain the stable code base for the 2.0 CVS tree and releases. -It's awsome, says Brian, but the new features I'm adding to 2.0.1 stable will be even more spectacular. Follow me to the future! etc. Better than silicon.fr who sent a message about Oracle's hostile bid for M$, this one you had to read a bit before getting the idea. Well done. -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@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
[Asterisk-Users] really small box
I don't know following has debated here or not, but is there in this world following stuff: A small, physically small box, like cable/DSL router, which comes with: 1) Ethernet port, 2) Console port, 3) CompactFlash or USB port, 4) memory module port, like SODIMM Box has built-in flash (256MB or 512MB) with or without Linux and feature to upgrade built-in RAM (128/256M) by adding memory module and storage via CompactFlash/USB. Box should have inexpensive x86 CPU in 500Mhz-1Ghz range without active cooling and should not have VGA port. It also should not have price tag more then $200. Anybody have seen stuff like this? Linksys NSLU2 and MacMini are not an option. I.N. ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] really small box
Irakli Natsvlishvili wrote: I don't know following has debated here or not, but is there in this world following stuff: I think you want a Soekris. Cheers, -- Ykoz Un Max - La VoIP en pr-pay! Essayez gratuitement - 5 crdits offerts. --- http://ykoz.net/voip/max --- ___ Asterisk-Users mailing list Asterisk-Users@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
[Asterisk-Users] Problems getting FXO channel working - Unable to create channel of type 'Zap' (cause 0)
Hi, I have searched around as much as I can and can't find any good info to
help me try this problem. I have added a FXO card to my server and from
everything I can see, I have configured it right. *Obviously not*
Below is my config, any ideas on troubleshooting this ?
regards
Mark
*CLI dial [EMAIL PROTECTED]
-- Executing Dial(OSS/dsp, Zap/g1/123) in new stack
Apr 1 10:17:59 NOTICE[950]: app_dial.c:960 dial_exec_full: Unable to
create channel of type 'Zap' (cause 0)
== Everyone is busy/congested at this time (1:0/0/1)
== Auto fallthrough, channel 'OSS/dsp' status is 'CHANUNAVAIL'
*CLI zap show channels
Chan Extension Context Language MusicOnHold
pseudohomecontext
1homecontext
[EMAIL PROTECTED] asterisk]# ztcfg -vv
Zaptel Configuration
==
Channel map:
Channel 01: FXS Kewlstart (Default) (Slaves: 01)
1 channels configured.
[EMAIL PROTECTED] asterisk]# cat ../zaptel.conf
fxsks=1
loadzone=uk
defaultzone=uk
[EMAIL PROTECTED] asterisk]# cat zapata.conf
[channels]
busydetect=1
busycount=7
relaxdtmf=yes
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
usecallerid=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
group=1
pickupgroup=1-4
immediate=no
context=homecontext
signalling=fxs_ks
callerid=asreceived
channel=1
[EMAIL PROTECTED] asterisk]#cat extensions.conf
[globals]
CONSOLE=Console/dsp ; Console interface for
demo
IAXINFO=guest ; IAXtel username/password
TRUNK=Zap/g1; Trunk interface
[homecontext]
exten = _0.,1, Dial(${TRUNK}/${EXTEN:1})
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Re: [Asterisk-Users] *** Asterisk 2.0 Stable release out now
Olle E. Johansson wrote: * New source code structure - C# and .net Asterisk 2.0 was moved to a Microsoft platform due to the demand for higher stability and a more secure foundation. Therefore, the code was quickly moved to C# on the .net platform. This gives Asterisk a lot of new features, including being fully integrated with Microsoft Exchange and Microsoft Active Directory. With all the user data stored in Active Directory, we finally have the user under full control. Users can dial in to the PBX to change their Windows password. We can also implement single-sign-on based on DTMF from a cell phone or WiFi phone. says Kelvin Reming. The C# language gives us much more modern code. And I'm so happy to get rid of the stupid-looking arctic bird, an ugly animal that that couldn't even fly. Shame this is just an april fool, I like the sound of this! Though it would be going head to head with Live Communications Server... -- Chris Hills IT Services North East Worcestershire College ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] sip.conf match
Thanks to every body for the solution. It works fine!! :D El Viernes, 1 de Abril de 2005 06:02, MF Hulber escribió: The way it works with my provider is that although both numbers enter the same context, each number will match its own extension. If I have two numbers: 11 and 22 it works as follows: [sip-in] exten = 11,1,Noop(First number dialed) exten = 22,1,Noop(Second number dialed) --- MARK. Pepe Aracil wrote: Hello. I have two hired pstn numbers with the same voip provider. I want to distingish in the sip.conf file, what of two phone numbers was dialed, but i don't know how to do the match, because the sip client and the sip host are the same for both numbers. How can i match in sip.conf by the (TO: ) header in sip negotiation? Sorry for my poor english :) Thanks. ___ Asterisk-Users mailing list Asterisk-Users@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 ___ Asterisk-Users mailing list Asterisk-Users@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 ___ Asterisk-Users mailing list Asterisk-Users@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
[Asterisk-Users] using unixODBC
hi list i know i am asking question out of the scope of this list. actualy i cant find any place to ask question like this. may be someone using ODBC with asterik. actualling i want to make ODBC connection for asterisk on my new fedora core 2. i have tried every thing. tried rpms. compiled code nothing works here. i have already done this kind of connection on my other mechine. i dont know why i am getting error. actually when i am doing isql asteriskdsn [ISQL]ERROR: Could not SQLConnect mysql is working connection is not working with Mysql nither with MSSQLServer through freetds odbc. [asteriskdsn] Description = mysqldriver Driver = mysqldriver Server = 192.168.8.99 Database= asterisk Port= 3306 Socket = Option = Stmt= odbcinst.ini [mysqldriver] Description = ODBC driver for MySql Driver = /usr/lib/libmyodbc.so Setup = /usr/lib/libodbcmyS.so FileUsage = 1 CPTimeout = CPReuse = regrads Kamran __ Do you Yahoo!? Make Yahoo! your home page http://www.yahoo.com/r/hs ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] RE: Asterisk Realtime - configuration help
Matthew Boehm [EMAIL PROTECTED] writes: sipfriends is deprecated. You should have seen the warning. This tells me that you did not infact read the wiki. Just wanted to mention that this Wiki page http://www.voip-info.org/wiki-Asterisk+RealTime says the following: RealTime support is currently available for the following families: * sipfriends * iaxfriends * voicemail * extensions While this doesn't make sipfriends look deprecated, the link on that page to http://www.voip-info.org/wiki-Asterisk+RealTime+Sip does use sipusers and sippeers which I assume is the replacement. - Kristian. -- Kristian Nielsen [EMAIL PROTECTED] Development Manager, Sifira A/S ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] *** Asterisk 2.0 Stable release out now
On Fri, 1 Apr 2005, Chris Hills wrote: Olle E. Johansson wrote: * New source code structure - C# and .net Asterisk 2.0 was moved to a Microsoft platform due to the demand for higher stability and a more secure foundation. Therefore, the code was quickly moved to C# on the .net platform. This gives Asterisk a lot of new features, including being fully integrated with Microsoft Exchange and Microsoft Active Directory. With all the user data stored in Active Directory, we finally have the user under full control. Users can dial in to the PBX to change their Windows password. We can also implement single-sign-on based on DTMF from a cell phone or WiFi phone. says Kelvin Reming. The C# language gives us much more modern code. And I'm so happy to get rid of the stupid-looking arctic bird, an ugly animal that that couldn't even fly. Shame this is just an april fool, I like the sound of this! Though it would be going head to head with Live Communications Server... I guess you missed the real joke there (the stability and secureness of .net) ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] setting SIP to dial PSTN with TDM400P
Hi, I've never used fxs/fxo modules, only E1 cards so I'm not entirely sure. However, this log: *CLI -- Starting simple switch on 'Zap/1-1' -- Executing Dial(Zap/1-1, Zap/1/6998256) in new stack -- Called 1/6998256 -- Zap/1/6998256-busy-1013475805 is busy -- Hungup 'Zap/1/6998256-busy-1013475805' == Everyone is busy/congested at this time -- Timeout on Zap/1-1 == CDR updated on Zap/1-1 seems to indicate you're making the call from Zap/1 and trying to make the outgoing call on Zap/1 also. I think you need to figure out which Zap channel is your FXO and which is your FXS. Maybe the outgoing is Zap/2? zap show channels gives a list I beleive... Secondly, your config files only seem to mention one channel. Have you looked at [EMAIL PROTECTED] It seems to autodrtect your config somehow On Fri, Apr 01, 2005 at 01:08:24PM +0500, Muhammad Haris wrote: to dear martijn, i made every possible change i can make i have a TDM400P Zap card... i had connected PSTN line to FXO Kewlstart at channel 1. and analog phone to FXS Kewlstart at Channel 4. i can hear continous ring tone when i hook up the receiver. plz have a look at my confs. Have a nice day, -- Martijn van Oosterhout Ecomtel Pty Ltd ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] chan_capi looking for missing channel_pvt.h
On Mar 31, 2005 3:32 PM, Mimmus [EMAIL PROTECTED] wrote: Hi, I'm trying to compile channel_capi with current Asterisk CVS. Asterisk compiled successfully but channel_capi (patched with all patches needed, as suggested from some nice people on IRC #Asterisk) compilation fails with: app_capiFax.c:34:34: asterisk/channel_pvt.h: No such file or directory I haven't such file on my system! Peraphs patches are for older CVS versions? Look in the Makefile for a reference to app_capiFax and remove it. Jason ___ Asterisk-Users mailing list Asterisk-Users@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
[Asterisk-Users] Problem with dial out via chan_capi
Hi *,
we successfully integrated the eicon diva 4 bri card in our Asterisk system.
I can dial in to system and route to sip peers.
I tried to dial out with following configuratin without any luck:
extensions.conf:
exten = _5.,1,Dial(CAPI/@301:b${EXTEN})
capi.conf:
[general]
mode=immediate
isdnmode=multipoint
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
[interfaces]
msn=301
incomingmsn=*
controller=2
context=default
echocancel=1
echotail=64
devices=2
Console output as follow:
-- Executing Dial(SIP/bdk-d27c, CAPI/@301:b5030225476) in new stack
-- Called @301:b5030225476
-- Setting up echo canceller (PLCI=0x102, function=1, options=2, tail=64)
-- Echo canceller successfully set up (PLCI=0x102)
-- CAPI Hangingup
== No one is available to answer at this time
Can you help me or give me tips?
Thanks in advance.
Kib
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[Asterisk-Users] Re: Are there online forums instead of this email forum??
In article [EMAIL PROTECTED], Bruno Hertz [EMAIL PROTECTED] wrote: Andrew Kohlsmith [EMAIL PROTECTED] writes: Call it archaic if you like but mailing lists get the job done faster, better and without all the bullshit that forums bring to the table. It's not archaic but reasonable. Clicking around in a funky web interface is definitely not what I call productive communication when compared to what good email clients (like gnus :) ) can do for you. My order of preference would be news groups, mailing lists, then everything else except web forums, which comes last. I totally agree. I run a local INN server and all the mailing lists I subscribe to get turned locally into newsgroup postings in moderated groups. When I make a posting, it gets mailed out through a filter to the moderator address, which is just the list posting address. Makes handling threads a breeze. I still use trn to read and post too, as I have yet to find anything that is as fast to use. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list Asterisk-Users@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
[Asterisk-Users] Re: Are there online forums instead of thisemailforum??
In article [EMAIL PROTECTED], Tim Bass [EMAIL PROTECTED] wrote: In addition, the lag time between posting a message to this list and having it delivered is a joke. I posted this message below at 2:35 and it was delivered to me, a new subscriber, an hour later. My postings normally come back to me within a few minutes (my local box polls my POP3 accounts every 2 minutes). I am sorry to say, but those on this list who are aggressively advocating SMTP mail with a lag time on a hour, posting profanity, and being impolite to other posters are not helping the Digium community. These shout down replies are absolute nonsense and I, for one, am surprised that Digium supports this type of nonsense support. Individuals' lack of courtesy or people skills is a completely independent issue from the preference for mailing lists or web forums. Suffice it to say that if there was a genuine preference amongst the majority for a web forum instead of a mailing list, there is more than enough skill and resources to make it happen. The fact that it hasn't happened might just say something. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list Asterisk-Users@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
[Asterisk-Users] Re: Are there online forums instead of this emailforum??
In article [EMAIL PROTECTED], Tim Bass [EMAIL PROTECTED] wrote: I just joined this list yesterday, And already you are telling the rest of us we're doing it all wrong. Great. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] *** Asterisk 2.0 Stable release out now
Asterisk 2.0 was moved to a Microsoft platform due to the demand for higher stability and a more secure foundation. It wasn't until I read this line that I knew it was a joke. I mean, seriously, who associates Microsoft with stability and security? A fool, that's who. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] RE: Asterisk Realtime - configuration help
While this doesn't make sipfriends look deprecated, the link on that If you are using a recent enough CVS version, it will tell you they are deprecated when you start asterisk. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@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
[Asterisk-Users] Re: Are there online forums instead of thisemailforum??
I wrote: In article [EMAIL PROTECTED], Tim Bass [EMAIL PROTECTED] wrote: In addition, the lag time between posting a message to this list and having it delivered is a joke. I posted this message below at 2:35 and it was delivered to me, a new subscriber, an hour later. My postings normally come back to me within a few minutes (my local box polls my POP3 accounts every 2 minutes). That posting was back in my POP3 mailbox 2.5 minutes after I posted it. I didn't see it for another 2 minutes because it arrived 5 seconds after I polled the mailbox. That's plenty fast enough for me. Perhaps there is a problem with your system or your ISP's mail servers. Or maybe the list server knows you don't like mailing lists and is just doing it to spite you :-) Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] patlooptest: Usage, setup?
Eric Wieling aka ManxPower wrote: Does anyone know what I need to do to use patlooptest? I have what I think is a T-1 loopback plug in the card (1-port, TE110P), but I still see a red alarm. Is this normal? I don't even know where to start for this. From Digium Support: You will need to specify each span as span=1,0,0,esf,b8zs. You must change the span number of course. Then you will specify clear=1-24 for a T1 or clear=1-31 for an E1. The only other options you should have in your zaptel.conf is loadzone and defaultzone. It does not matter what these are set to. Then you will have to reload the zaptel kernel modules. Run make tests in your zaptel source directory. You will need a T1 loopback cable plugged into the back of the card. Once the T1 loopback cable is installed the span should go green. You may check the status by using zttool. If the span is not green then your T1 loopback cable is faulty. You can make a T1 loopback cable using wires 1 to 4 and 2 to 5. You will run ./patlooptest /dev/zap/1 180. The 180 is the length in seconds that the test will run. /dev/zap/1 is the first clear channel on this span. If you wish to test a second span then you would start with the first clear channel of that span. Span 2 would start at /dev/zap/25 on a T1. patlooptest will only output on errors. Disregard any errors the first 15 seconds of the test. Very few errors over a long period of time are normal. ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] Are there online forums instead of this
On Thu, Mar 31, 2005 at 05:14:50PM -0500, Tim Bass wrote: I use procmail and know very well how to manage email. All asterisk mail goes to a folder,etc. Your point...because a few people don't understand how to manage e-mail is nonsense and shows why this list should be moderated. *BLINK* He expresses an opinion, you disagree with it and therefore what he says should be moderated out of existance? I'm sorry, but arguments that on this list not every post is as nice as possible are just not going to fly. This list is no worse than any other I'm on. This is the real world and the list reflects that. If you don't like what you read, ignore it. But the idea of moderation scares me because I might miss something useful just because someone else decided I shouldn't see it. I'll decide for myself thank-you-very-much. From my point of view, web based forums can never compete for me because: 1. The RTT to bring a new page in and render it takes at least a second, usually more. 2. Displaying more than one message at a time is irritating because then you have to scroll around and it can no longer track read/unread. 3. Finally, colours, pictures, odd fonts, etc slow down my reading speed. I prefer everything in a fixed width font, white text, black background, each message starting at exactly the same point on my screen. Keyboard control only. The combination means that I could only get through less than half as many forum posts as mailing list posts in a given period. Time being money completes the picture. -- Martijn van Oosterhout Ecomtel Pty Ltd ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] *** Asterisk 2.0 Stable release out now
Hello, Olle! OEJAsterisk 2.0 was moved to a Microsoft platform due to the OEJdemand for higher stability and a more secure foundation. Nice... I remember that about 10 years ago, when I was working in a daily newspaper we wrote and article on April 1st on a first page about scientific breakthrough with lunching new satellite. Satellite was going to transmit energy and electricity from space directly to homes of million customers... We've got pretty interesting calls that day, including from some low enforcement officials... I.N. ___ Asterisk-Users mailing list Asterisk-Users@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
[Asterisk-Users] H.323 call '...' cleared, reason 15 (Call ended due to security checks)
Thanks to Yves's commitment I was able to configure oh323 channel, cleared the codec issue, registered to Gatekeeper, placed a call, but receive this message on the console. What might be the problem? Asterisk Ready. *CLI -- Registered with gatekeeper '[EMAIL PROTECTED]'. -- Executing Dial(SIP/2000-5a52, OH323/193.192.100.92/0212441) in new stack -- H.323 call to 193.192.100.92/0212441 with codec(s) g729 -- Called 193.192.100.92/0212441 -- H.323 call 'ip$localhost/5502' cleared, reason 15 (Call ended due to security checks) -- OH323/L5502 is circuit-busy -- Hungup 'OH323/L5502' == Everyone is busy/congested at this time (1:0/1/0) -- Executing Hangup(SIP/2000-5a52, ) in new stack == Spawn extension (local, 0212441, 2) exited non-zero on 'SIP/2000-5a52' -- Executing Dial(SIP/2000-5a52, OH323/193.192.100.92/h) in new stack -- H.323 call to 193.192.100.92/h with codec(s) g729 -- Called 193.192.100.92/h -- H.323 call 'ip$localhost/5503' cleared, reason 15 (Call ended due to security checks) -- OH323/L5503 is circuit-busy -- Hungup 'OH323/L5503' == Everyone is busy/congested at this time (1:0/1/0) -- Executing Hangup(SIP/2000-5a52, ) in new stack == Spawn extension (local, h, 2) exited non-zero on 'SIP/2000-5a52' ___ Asterisk-Users mailing list Asterisk-Users@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
RE: [Asterisk-Users] using unixODBC
-Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Kamran Ahmad Envoyé : vendredi 1 avril 2005 11:08 À : asterisk-users@lists.digium.com Objet : [Asterisk-Users] using unixODBC hi list i know i am asking question out of the scope of this list. actualy i cant find any place to ask question like this. may be someone using ODBC with asterik. actualling i want to make ODBC connection for asterisk on my new fedora core 2. i have tried every thing. tried rpms. compiled code nothing works here. i have already done this kind of connection on my other mechine. i dont know why i am getting error. actually when i am doing isql asteriskdsn [ISQL]ERROR: Could not SQLConnect Hello Just give my own config that works well /etc/odbc.ini [MySQL-asterisk] Description = MySQL asterisk database Trace = Off TraceFile = stderr Driver = MySQL SERVER = 127.0.0.1 USER= connecting-user PASSWORD= user-password PORT= 3306 DATABASE= asterisk /etc/odbcinst.ini [MySQL] Description = MySQL driver for Linux Driver = /usr/lib/libmyodbc.so FileUsage = 1 /etc/asterisk/res_odbc.conf [asterisk] dsn = MySQL-asterisk username = connecting-user password = user-password pre-connect = yes Hop i'll help you as it works great here Best regards Thierry Wehr ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] Re: Are there online forums instead of this email forum??
On Fri, April 1, 2005 11:31, Tony Mountifield said: In article [EMAIL PROTECTED], Bruno Hertz [EMAIL PROTECTED] wrote: Andrew Kohlsmith [EMAIL PROTECTED] writes: I totally agree. I run a local INN server and all the mailing lists I subscribe to get turned locally into newsgroup postings in moderated groups. When I make a posting, it gets mailed out through a filter to the moderator address, which is just the list posting address. Makes handling threads a breeze. I still use trn to read and post too, as I have yet to find anything that is as fast to use. Cheers Tony -- I very much like - and heavily use - Squirrelmail for OoO access to my mail... Light, fast, threads, searching, extensions based (Procmail management!!!) and usable from virtually everywhere I like, when I don't have my laptop along! (Or when I cannot access my private mail server directly) -- Francesco Peeters GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704 If your program doesn't recognize my signature, please visit http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate. ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] Re: Are there online forums instead of thisemailforum??
On Fri, April 1, 2005 11:53, Tony Mountifield said: That posting was back in my POP3 mailbox 2.5 minutes after I posted it. I didn't see it for another 2 minutes because it arrived 5 seconds after I polled the mailbox. That's plenty fast enough for me. Perhaps there is a problem with your system or your ISP's mail servers. Or maybe the list server knows you don't like mailing lists and is just doing it to spite you :-) Cheers Tony -- It varies during the day. During our working day it's pretty fast, but when evening comes, and the US starts playing too, the speed decreases... I never had to wait an hour though! -- Francesco Peeters (@ CEST) GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704 If your program doesn't recognize my signature, please visit http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate. ___ Asterisk-Users mailing list Asterisk-Users@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
[Asterisk-Users] [Fwd: Problem with dial out via chan_capi]
Hi,
problem solved, I found somethind in this mailing list!
extensions.conf:
exten = 0237482,1,Dial,CAPI/@301:0237482,5,tr
Regards,
Kib
---BeginMessage---
Hi *,
we successfully integrated the eicon diva 4 bri card in our Asterisk system.
I can dial in to system and route to sip peers.
I tried to dial out with following configuratin without any luck:
extensions.conf:
exten = _5.,1,Dial(CAPI/@301:b${EXTEN})
capi.conf:
[general]
mode=immediate
isdnmode=multipoint
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
[interfaces]
msn=301
incomingmsn=*
controller=2
context=default
echocancel=1
echotail=64
devices=2
Console output as follow:
-- Executing Dial(SIP/bdk-d27c, CAPI/@301:b5030225476) in new stack
-- Called @301:b5030225476
-- Setting up echo canceller (PLCI=0x102, function=1, options=2, tail=64)
-- Echo canceller successfully set up (PLCI=0x102)
-- CAPI Hangingup
== No one is available to answer at this time
Can you help me or give me tips?
Thanks in advance.
Kib
---End Message---
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[Asterisk-Users] Eicon Diva Server BRI Setup
Hi, Has anyone got this card working with Asterisk? If so what kernel are you using? Currently I have installed Fedora Core 3 with the 2.6.10-1.770_FC kernel Chan Capi 0.3.5 Asterisk 1.0.7 The diva card is detected by linux and the chan capi is installed in asterisk When asterisk boots the following error is reported: [chan_capi.so] = (Common ISDN API for Asterisk) == Parsing '/etc/asterisk/capi.conf': Found Apr 1 11:39:01 NOTICE[12167]: chan_capi.c:2636 load_module: CAPI not installed! Apr 1 11:39:01 WARNING[12167]: loader.c:345 ast_load_resource: chan_capi.so: load_module failed, returning -1 Apr 1 11:39:01 WARNING[12167]: chan_capi.c:2812 unload_module: Unable to unregister from CAPI! == Unregistered channel type 'CAPI' Apr 1 11:39:01 WARNING[12167]: loader.c:391 load_modules: Loading module chan_capi.so failed! and from lsmod divas 75961 0 divadidd13337 1 divas Any ideas? D. __ e: [EMAIL PROTECTED] t: +44 207 397 8451 m: +44 7966 926694 w: www.bright-talk.com ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] H.323 call '...' cleared, reason 15 (Call ended due to security checks)
Cenk Yabas wrote: Thanks to Yves's commitment I was able to configure oh323 channel, cleared the codec issue, registered to Gatekeeper, placed a call, but receive this message on the console. What might be the problem? Asterisk Ready. *CLI -- Registered with gatekeeper '[EMAIL PROTECTED]'. -- Executing Dial(SIP/2000-5a52, OH323/193.192.100.92/0212441) in new stack -- H.323 call to 193.192.100.92/0212441 with codec(s) g729 -- Called 193.192.100.92/0212441 -- H.323 call 'ip$localhost/5502' cleared, reason 15 (Call ended due to security checks) The gatekeeper has cleared the call. I guess because a password is required or the one provided is not correct. What version of the channel driver do you use? Michael. ___ Asterisk-Users mailing list Asterisk-Users@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
RE: [Asterisk-Users] SuperMicro X5DE8-GG Motherboard Goes Kaput afterInstalling TE410P Card - Yikes!
On Thu, 2005-03-31 at 11:05 -0500, Tim Bass wrote: We installed one Digium TE410P in the PCIX slot and put the power cable back on. The machine tried to come up, but the TE410P card flashed red lights in all four ports and there was no video output, no motherboard beeps or anything. This was a very simple (1) shutdown, (2) remove power supply (3) install riser card and TE410P, and (4) reconnect power cord. Not sure, but I didn't think any of the Digium cards where PCIX compatible. The TE410P was compatible with a 64bit slot but nothing more. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@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
[Asterisk-Users] Re: Problem with dial out via chan_capi
I tried to dial out with following configuratin without any luck: extensions.conf: Can you help me or give me tips? from the asterisk cli console asterisk -r type capi debug place a call and post your capi debug log ___ Asterisk-Users mailing list Asterisk-Users@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
[Asterisk-Users] queue.conf config
Hello all, There are any way for the queue agents in asterisk that they do not need to login in the queue to begin recibing calls? I want to use this queue for our recepcionist, with only one agent. All that I want is, 1. The recepcionist do not need to make a login in the queue. 2. The recepcionist not have to hear the phone all the time waiting for new calls, when she hangs up the phone asterisk make a logout for the agent and she must to login it again to recibe new calls. Any clue will be apreciated. Thanks for your time. Ismael. ___ Asterisk-Users mailing list Asterisk-Users@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
[Asterisk-Users] Re: Eicon Diva Server BRI Setup
Hi,
try this:
mknod /dev/capi20 c 68 0
chmod 660 /dev/capi20
I have same configuration as you. It worked for me since yesterday.
Regards,
Kib
---BeginMessage---
Hi *,
we successfully integrated the eicon diva 4 bri card in our Asterisk system.
I can dial in to system and route to sip peers.
I tried to dial out with following configuratin without any luck:
extensions.conf:
exten = _5.,1,Dial(CAPI/@301:b${EXTEN})
capi.conf:
[general]
mode=immediate
isdnmode=multipoint
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
[interfaces]
msn=301
incomingmsn=*
controller=2
context=default
echocancel=1
echotail=64
devices=2
Console output as follow:
-- Executing Dial(SIP/bdk-d27c, CAPI/@301:b5030225476) in new stack
-- Called @301:b5030225476
-- Setting up echo canceller (PLCI=0x102, function=1, options=2, tail=64)
-- Echo canceller successfully set up (PLCI=0x102)
-- CAPI Hangingup
== No one is available to answer at this time
Can you help me or give me tips?
Thanks in advance.
Kib
---End Message---
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[Asterisk-Users] Eicon Diva Server BRI Setup
Hi, try this: mknod /dev/capi20 c 68 0 chmod 660 /dev/capi20 I have same configuration as you. It worked for me since yesterday. Regards, Kib ___ Asterisk-Users mailing list Asterisk-Users@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
[Asterisk-Users] Re: Problem with dial out via chan_capi
Thanks, problem solved, I found somethind in this mailing list! Wrong extensions.conf entry. extensions.conf: exten = 0237482,1,Dial,CAPI/@301:0237482,5,tr ?? But, what does ,5,tr mean ?? Regards, Kib ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] Re: Problem with dial out via chan_capi
Kib Eki wrote: Thanks, problem solved, I found somethind in this mailing list! Wrong extensions.conf entry. extensions.conf: exten = 0237482,1,Dial,CAPI/@301:0237482,5,tr ?? But, what does ,5,tr mean ?? 5 tells Asterisk to hang up if the call is not answered in 5 seconds. t tells Asterisk to use that horrible # hack to do transfers r tells Asterisk to send a ringing sound to the caller, even when doing so is not the right thing to do. show application dial will tell you about the options. ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] Are there online forums instead of this
I do not claim/pretend to speak for everybody on this list, but I *do* think that others that promote web forums should not do so either... Hear hear!! Let's let it die, folks; there are more pressing issues to deal with. It's true that as long as the Digiumites hang out here, it's going to be tough to get any traction for a web forum. So, just like those of us who prefer email lists would have to do if the canonical list were to be a forum: suck it up, and do the best you can. I hope everyone realizes that this is religion we're talking here, not technology. I'll second that motion. Also, there is no reason why both can't exist, and there is no justification that would suggest all-or-none. There's been more then one person offering to establish/host stuff; for those that want it, go forth and do it and stop bothering the rest of the list. ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] Are there online forums instead of thisemailforum??
On March 31, 2005 11:28 pm, Tim Bass wrote: The discussion should not be laced with profanity, you should treat this list and others like there are women on the list and try to be polite so everyone is comfortable. Most professionals discuss matters in a way where everyone is comfortable to discuss. There is nothing wrong with being polite, not using profanity, and being respectful of people with different opinions. You would do well to follow your own rules. I believe the only profanity I used in my correspondence with you is the word 'arse' -- if that's enough to get me moderated down in your 28-kilouser-strong community then I want no part of it. Or, better yet, Digium should shut this list down and move it to a commercial vBulletin style forum and get some good moderators to delete posts that do not follow a basic set of social rules of behavior. Here are the rules from UNIX.COM, and they work very well: The rules don't look bad and they're very similar to the implied rules of any mailing list (including this one), with the exception to you reserving the right to remove any post you or any moderator sees fit. No thanks, I don't do well with censorship. You don't happen to be one of those neighbourhood czars who try and enforce what your neighbours can do with their homes in order to protect your own property value, do you? Again, there's no reason for this list to be shut down. Asterisk has a link to voip-info.org on its site and also has links to several other online resources. Why should your forum be any different? If it really is better, everyone will flock to it. -A. ___ Asterisk-Users mailing list Asterisk-Users@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
[Asterisk-Users] ${DIALSTATUS}
Hi list,
I try to explore making use of the variable ${DIALSTATUS} for
auto-answering purposes an similar.
On my asterisk box this does not work because ${DIALSTATUS} always
returns empty :(
Didn't find much in the web on this issue.
Can someone help?
regards Manuel
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[Asterisk-Users] [OT] Announcing MidwestTea.com
Folks, I know that this is off-topic... but... I'm fulfilling a longtime dream today... launching my on-line tea business. Teas grown /locally/, right here in the midwest. I'm so excited that I'm telling everybody. :-) (Naturally, we are using * and voip for our phone system.) Please visit www.MidwestTea.com http://www.MidwestTea.com/ And... to celebrate... I am offering a 75% discount today, my 1st day in business. Just enter coupon code 04GRAND. Thanks for dropping by! -- Art Z. -- www.MidwestTea.com ___ Asterisk-Users mailing list Asterisk-Users@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
[Asterisk-Users] LDAP and Asterisk
I am looking to roll out an Asterisk VoIP implementation to our 200 employees. So far I have hooked up the Asterisk box to our Elmeg PBX via a PRI interface card and have that working, plus about 30 test users on Xlite softphones. Up til now all the configuration has been done by hand editing extensions.conf and sip.conf and voicemail.conf as needed. I would rather this was kind of automatic - when a new user is created then everything is already setup for them. We are in a (horror of horrors) Microsoft environment running Windows XP, Windows 2003 Server with AD and a sizable number of Sun and Linux boxes for development (we are an IT development shop). So what springs to mind is someone how connecting Asterisk to AD and using some spare fields in AD to hold extension numbers and the like and querying through an LDAP interface. Kind of like Realtime but using LDAP. Does anything like this currently exist? ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] really small box
I run asterisk on a soekris 4801, it works great. If you needed more horsepower a via epia mini-itx would work too. I can't say enough how much I like the soekris boxes though... ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] Are there online forums instead of this email
Francesco Peeters [EMAIL PROTECTED] writes: On the other hand imagine a forum with subtopics like sipura, softphones, zap or whatever. Wouldn't that maybe help to put some load off at least the casual reader and poster seeking or giving advice for topics he/she specialized in, and maybe even the more active participants? Just a thought, and not a bad one imho. Nah, like I said, IMHO it's not different from multiple maillists, as long as the same rules are applied consistenly... ;-) Well, it's easy to say nah if you don't want to think about it. Again, I favor mailing lists too, and all would be OK for me if ppl here weren't already complaining about volume and stuff. So, let me point out two obvious differences you missed: (1) Subscription With a web forum, you register once to the whole forum and have thus access to all topics. On the other hand, when you have like twenty mailing lists on various * topics, who (especially of them newcomers) would subscribe to them all? E.g. if you only have one or two questions to post you'll subscribe to the most introductory/general list and are very likely to stay there. (2) Topic choice With a web forum, you have all topics generally visible on the main page and are likely to see them any time you visit the forum, while when subscribing to lists you do it once and stay. How often do you actually look what other lists are actually available for particular topics? Only if you're forced to, I gather, e.g. because you don't get help on your current list(s). So with mailing lists, there's just higher gravity which lets ppl stick e.g. to -users. Anyway, before saying nah, please keep in mind that I'm not advocating anything right now but just suggesting to keep an open mind since there actually *are* problems with this list ppl have been complaining about for some time. As nothing seems to improve in the current setup, it wouldn't hurt, while discussing this, to at least seriously consider and thoroughly evaluate alternatives. Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@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
RE: Optimizing speex (was Re: [Asterisk-Users] Erratic CPU load )
Steve, Looks much better now, although it didn't end the cpu load surges: they just arrive less frequently (period of several minutes). There are some reports about cpu spikes hitting your machine every few hours - when using G711. Maybe these spikes are the same ones I see. When I change from speex towards optimized speex or gsm my spike period goes up from 1 to 10 minutes. If this increase is related to (decreasing) translator costs, I guess a few hour period for G711 is quite possible. I guess I should ask the dev-list... Eric. -Original Message- From: Steve Kann [mailto:[EMAIL PROTECTED] Sent: Tuesday, March 29, 2005 11:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Optimizing speex (was Re: [Asterisk-Users] Erratic CPU load ) Eric, If you want to optimize speex, I'd suggest the following: 1) Re-compile the speex library with SSE optimizations; add --enable-sse to the configure line used for compilation. 2) Reduce the complexity from 4, to 2 or 3 in codecs.conf. You won't notice the difference in quality. 3) Lower bitrates use less CPU; try setting abr to 8000, which is a good all-around choice; it gives you an average of 8kbps usage, but can range from 2-3kbps to 16 kbps or so during simple/complex speech parts. -SteveK ___ Asterisk-Users mailing list Asterisk-Users@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 ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] Are there online forums instead of this email
On Fri, April 1, 2005 15:10, Bruno Hertz said: Francesco Peeters [EMAIL PROTECTED] writes: On the other hand imagine a forum with subtopics like sipura, softphones, zap or whatever. Wouldn't that maybe help to put some load off at least the casual reader and poster seeking or giving advice for topics he/she specialized in, and maybe even the more active participants? Just a thought, and not a bad one imho. Nah, like I said, IMHO it's not different from multiple maillists, as long as the same rules are applied consistenly... ;-) Well, it's easy to say nah if you don't want to think about it. Again, I favor mailing lists too, and all would be OK for me if ppl here weren't already complaining about volume and stuff. SNIP I think you took my Nah a itsy bit out of context there... ;-) Your points about Subscription and Topic choice are valid, and the ideal would be a forum that would behave like a maillist... I.e. post and read either on web or mail, and it'll get where it should be... It is difficult though. Totally OT: I have been looking at this as a plugin for my own (non tech) WebBBS/Forum, but the problem is that not all clients adhered to the 'references' SMTP-header behavior at that time... I haven't looked at that for a while now, so that may have changed... (Besides that I unfortunately do not have time to write such an extension at this point in time... It'd be an interesting challenge tho...) -- Francesco Peeters GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704 If your program doesn't recognize my signature, please visit http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate. ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] Xten-lite for linux
On Mar 31, 2005 8:55 PM, Bruno Hertz [EMAIL PROTECTED] wrote: Brian Capouch [EMAIL PROTECTED] writes: Hmmm. I just got the latest beta build, which identifies itself as 1105d. The keypad functionality is perfect. Hmmm. Good for you. We were talking about sjphone, though :) Regards, Bruno. I'm pretty sure that I used SJphone to check my VM. I'll test again. But there is a new beta out on their site (and it's newer than the Windows build). Maybe they added a dialpad? -- Dana ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] Phones Callwaiting enable by default?
I'm using Sipura SPA-841 and SPA-2000 phones and ATAs... Yes.. it's a good assumption that I'm using asterisk, since I posted to this list Umm *70 is there to turn call waiting on/off in the asterisk database. On Mar 31, 2005 7:57 PM, C F [EMAIL PROTECTED] wrote: What phones? are you using Avaya, or Toshiba? Since you are posting to this list I will guess you are using Asterisk, in which case I have no clue why *70 is there in the first place. Did you notice that the guy that went to the Doctor that his eye hurts when drinking coffee, refused to remove the spoon from the cup? On Thu, 31 Mar 2005 11:03:12 -0500, Matt [EMAIL PROTECTED] wrote: Hi, how can I get all the phones to enable call waiting by default instead of having to dial *70 on each one to activate call waiting? ___ Asterisk-Users mailing list Asterisk-Users@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 ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] Livevoip still no DTMF?
I read in the archives a number of discussions about livevoip, DID,
and DTMF not working.
However, no resolutions.
I just setup a livevoip DID and indeed the DTMF does not work.
The same asterisk context works via broadvoice and via
direct dialing in to the asterisk server via SIP.
Just no DTMF with calls via livevoip.
I'm running Asterisk CVS-v1-0-03/06/05-23:15:12
Its been working fine here for about a month now. Currently using
CVS-HEAD-03/31/05, however it worked fine with several previous
cvs-head versions as well.
Below are the pieces I'm using for incoming calls. Might want to
review and compare to whatever you're using. The iax.conf section
is a very basic type=user with a context referring incoming calls
to the liveviop800 section of extensions.conf shown below.
[livevoip800]
include=bus-ivr-main
exten=8001234567,1,Dial(${PHONE6}${PHONE7},10)
exten=8001234567,2,Goto(bus-ivr-main|s|1)
[bus-ivr-main]
exten = s,1,Wait,1
exten = s,2,Answer
exten = s,3,DigitTimeout,5
exten = s,4,ResponseTimeout,20
exten = s,5,Background(npi-greeting) ; Thanks for calling press 1 for
The above essentially rings two Cisco 7960's and if no answer,
routes the incoming call to bus-ivr-main. The caller can then enter
valid dtmf digits, including allowed four-digit extensions, etc.
Have had zero problems with dtmf.
(Note: the above approach does have an issue with handling ringback to
the caller _after_ they've entered a four-digit extension. That issue
has been documented/discussed on the list, and is associated with
livevoip not handling the iax ringing function after a call as been
s,2,Aanswer. Work arounds have been noted on the list, however I've
elected not to address it as its just not that big of a deal for us.)
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Re: [Asterisk-Users] Re: Are there online forums instead of this
Tim Bass [EMAIL PROTECTED] writes: the excellent movie Vanilla Sky)... Ahem. . . . B#2. ___ Asterisk-Users mailing list Asterisk-Users@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
[Asterisk-Users] Re: Polycom sound quality problems
Hi Eric - I'm having a problem with my Polycom phones and hoping someone else has experienced the same thing: Outbound calls are fine, and inbound calls originating from another SIP phone are fine, but inbound calls to the Polycom phone from an IAX channel sound like you're talking to a robot. The person on the Polycom sounds fine to the person on the IAX channel, however. Inbound calls to our soft phones sound just fine. Asterisk 1.0.5 on Debian (also had the problem with 1.0 on Fedora) Polycom SoundPoint IP500 SIP Sixtel is the IAX provider. Check to see what codec is being used for the call. Sean Default is U-law, but I also switched it to A-law with the exact same results. I might check out QoS. You can specify TOS tagging on your IAX channels in iax.conf, and the Polycom phones are able to respond to TOS tagging (in ipmid.cfg - or in the web interface under Core Conf). Maybe they are are trying to do two mutually exclusive kinds of TOS tagging? You can tell the Polycom phone to just not respond to TOS. - Noah ___ Asterisk-Users mailing list Asterisk-Users@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
[Asterisk-Users] Maybe an echo cancellation problem?
Hi Was hoping someone could point me in the right direction. using asterisk cvs in various VOIP configurations On a call when the loudness of transmit receive then all receiving is null. In practical terms this causes background noise (from the other end)to stop when you are talking and come back on when you are quieter. Often causing background noise to seem like it is switching on and off as you speak during a call similar to if you were using a CB radio. in testing calls: transmit = null receive = steady noise (generated from the other end) transmit = quiet receive = steady noise (generated from the other end) transmit = normal/high receive = null Does this seem like it is echo cancellation related? Any pointers as to what topic this sort of thing would fall under? Thanks Jack __ Yahoo! Messenger Show us what our next emoticon should look like. Join the fun. http://www.advision.webevents.yahoo.com/emoticontest ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] Problem with livevoip dial out
I am starting to use livevoip but when I configure they way they suggest, I
see errors.
[livevoip]
exten =_51NXXNXX,1,Dial(IAX2/myusername:[EMAIL PROTECTED]/${EXTEN:1})
snip
Heres the error message:
-- Executing Dial(IAX2/[EMAIL PROTECTED]:4569-6, 1000|15) in new stack
Mar 31 22:31:07 WARNING[27589]: app_dial.c:920 dial_exec_full: Dial argument
takes format
(technology1/[device:]number1technology2/[device:]number2...|optional
timeout)
I'm using the same formatted Dial statement as you're showing above.
However, when I place a call, the CLI shows:
-- Executing Dial(SIP/3000-a05a, IAX2/myuserid:[EMAIL PROTECTED]/140
21234567) in new stack
-- Called myuserid:[EMAIL PROTECTED]/14021234567
-- Call accepted by 217.160.244.186 (format gsm)
-- Format for call is gsm
If you compare my CLI output to yours, it suggests the actual Dial
statement that you are executing is _not_ the one you've shown above.
If you look closely at your CLI output above, you apparently are
executing a dial statement that looks something like:
exten =_51NXXNXX,1,Dial(IAX2/livevoip,15/${CallerID})
and not the one shown.
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Re: [Asterisk-Users] Re: Are there online forums instead of this email forum??
[EMAIL PROTECTED] (Tony Mountifield) writes: I totally agree. I run a local INN server and all the mailing lists I subscribe to get turned locally into newsgroup postings in moderated groups. When I make a posting, it gets mailed out through a filter to the moderator address, which is just the list posting address. Makes handling threads a breeze. Now this sounds like a nice solution, and seems to be one step away from a complete news/mailing list gateway (registration). Did you set this all up yourself? Since I was about to investigate this stuff myself today, i.e. to gateway the list with a standalone news server and then maybe even add a decent web interface with search capablities. I suspect there'll be few 'solutions' out there, since if so you'd run across them more often, but in case you have any pointers I'd sure appreciate them (man, I really like the idea proxying all that lists though inn ... :) ) Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] Caller ID on voicemail messages
Take a look at the voicemail.conf.sample that comes with asterisk. Inside you will see how to change the voicemail email message that is cerated and add the phone number (and remove the name) for callerid. -Matt ___ Asterisk-Users mailing list Asterisk-Users@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
[Asterisk-Users] Dial'ing multiple SIP devices impossible when forward activated
Hi, When I Dial(SIP/1SIP/2SIP/3) if any of these phones has a forward to another destination (302: moved response) then the simultaneous ring stops immediately and the incoming call goes to wherever the forward points to. We are using simultaneous ringing as a fallback when the receptionist doesn't anwser after a while and such a call should never be forwarded. Is there a way to tell * to ignore any forward on certain calls? Thanks for your help, -- Fast Food: Corporate America in your body Television: Corporate America in your mind. ___ Asterisk-Users mailing list Asterisk-Users@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
RE: [Asterisk-Users] Re: Are there online forums instead ofthisemailforum??
The lag time on SMTP list depends on three factors: (1) The volume of the traffic; (2) When you registered (if you registered two years ago, for example, you receive mail in a large list before someone, say, who registered a month ago); (3) Various points of network congestion and delays. During peak times on this list, people who have recently registered have a one hour lag time to receive messages and it has little to do with ISPs, etc. Some simple math. (not completely accurate) If there are 2000 people on the list and it takes 2 seconds to deliver a message, and you are at the end of the list, then it will take 1000 seconds to get mail, or 15 minutes to get mail.If any network congestion, then it could take an hour for some people at the end of the list (which you will not see if you are at the first of the list). Yesterday, during peak traffic, for people at the end of the list, the lag time was over one hour, easily measurable. Mr. Mountifield's message test was not (1) during peak traffic and (2) he ,may not have registered recently, because if he did, he would have seen the serialization lag time. Let's use this message, mornings are busy. I send it a 9:13 EST. We will see when it returns. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tony Mountifield Sent: Friday, April 01, 2005 4:40 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Re: Are there online forums instead ofthisemailforum?? In article [EMAIL PROTECTED], Tim Bass [EMAIL PROTECTED] wrote: In addition, the lag time between posting a message to this list and having it delivered is a joke. I posted this message below at 2:35 and it was delivered to me, a new subscriber, an hour later. My postings normally come back to me within a few minutes (my local box polls my POP3 accounts every 2 minutes). I am sorry to say, but those on this list who are aggressively advocating SMTP mail with a lag time on a hour, posting profanity, and being impolite to other posters are not helping the Digium community. These shout down replies are absolute nonsense and I, for one, am surprised that Digium supports this type of nonsense support. Individuals' lack of courtesy or people skills is a completely independent issue from the preference for mailing lists or web forums. Suffice it to say that if there was a genuine preference amongst the majority for a web forum instead of a mailing list, there is more than enough skill and resources to make it happen. The fact that it hasn't happened might just say something. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list Asterisk-Users@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 ___ Asterisk-Users mailing list Asterisk-Users@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
RE: [Asterisk-Users] Re: Are there online forums insteadofthisemailforum??
I registered 1 week ago, and this message took 3 minutes to reach me. Granted I'm in the UK so there is bound to be some strange effect causing the speeding up of the message. This topic is like a bad penny, it just won't go away. Mark -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim Bass Sent: 01 April 2005 15:14 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Re: Are there online forums insteadofthisemailforum?? The lag time on SMTP list depends on three factors: (1) The volume of the traffic; (2) When you registered (if you registered two years ago, for example, you receive mail in a large list before someone, say, who registered a month ago); (3) Various points of network congestion and delays. During peak times on this list, people who have recently registered have a one hour lag time to receive messages and it has little to do with ISPs, etc. Some simple math. (not completely accurate) If there are 2000 people on the list and it takes 2 seconds to deliver a message, and you are at the end of the list, then it will take 1000 seconds to get mail, or 15 minutes to get mail.If any network congestion, then it could take an hour for some people at the end of the list (which you will not see if you are at the first of the list). Yesterday, during peak traffic, for people at the end of the list, the lag time was over one hour, easily measurable. Mr. Mountifield's message test was not (1) during peak traffic and (2) he ,may not have registered recently, because if he did, he would have seen the serialization lag time. Let's use this message, mornings are busy. I send it a 9:13 EST. We will see when it returns. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tony Mountifield Sent: Friday, April 01, 2005 4:40 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Re: Are there online forums instead ofthisemailforum?? In article [EMAIL PROTECTED], Tim Bass [EMAIL PROTECTED] wrote: In addition, the lag time between posting a message to this list and having it delivered is a joke. I posted this message below at 2:35 and it was delivered to me, a new subscriber, an hour later. My postings normally come back to me within a few minutes (my local box polls my POP3 accounts every 2 minutes). I am sorry to say, but those on this list who are aggressively advocating SMTP mail with a lag time on a hour, posting profanity, and being impolite to other posters are not helping the Digium community. These shout down replies are absolute nonsense and I, for one, am surprised that Digium supports this type of nonsense support. Individuals' lack of courtesy or people skills is a completely independent issue from the preference for mailing lists or web forums. Suffice it to say that if there was a genuine preference amongst the majority for a web forum instead of a mailing list, there is more than enough skill and resources to make it happen. The fact that it hasn't happened might just say something. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list Asterisk-Users@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 ___ Asterisk-Users mailing list Asterisk-Users@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 ___ Asterisk-Users mailing list Asterisk-Users@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
RE: [Asterisk-Users] Re: Are there online forums insteadofthisemailforum??
Wow! Only 3 minutes delivery. That is much better than the one hour yesterday! I am glad to see the list working a bit faster today :) One hour lag yesterday was painfully slow. I stand corrected on the lag time issue. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim Bass Sent: Friday, April 01, 2005 9:14 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Re: Are there online forums insteadofthisemailforum?? The lag time on SMTP list depends on three factors: (1) The volume of the traffic; (2) When you registered (if you registered two years ago, for example, you receive mail in a large list before someone, say, who registered a month ago); (3) Various points of network congestion and delays. During peak times on this list, people who have recently registered have a one hour lag time to receive messages and it has little to do with ISPs, etc. Some simple math. (not completely accurate) If there are 2000 people on the list and it takes 2 seconds to deliver a message, and you are at the end of the list, then it will take 1000 seconds to get mail, or 15 minutes to get mail.If any network congestion, then it could take an hour for some people at the end of the list (which you will not see if you are at the first of the list). Yesterday, during peak traffic, for people at the end of the list, the lag time was over one hour, easily measurable. Mr. Mountifield's message test was not (1) during peak traffic and (2) he ,may not have registered recently, because if he did, he would have seen the serialization lag time. Let's use this message, mornings are busy. I send it a 9:13 EST. We will see when it returns. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tony Mountifield Sent: Friday, April 01, 2005 4:40 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Re: Are there online forums instead ofthisemailforum?? In article [EMAIL PROTECTED], Tim Bass [EMAIL PROTECTED] wrote: In addition, the lag time between posting a message to this list and having it delivered is a joke. I posted this message below at 2:35 and it was delivered to me, a new subscriber, an hour later. My postings normally come back to me within a few minutes (my local box polls my POP3 accounts every 2 minutes). I am sorry to say, but those on this list who are aggressively advocating SMTP mail with a lag time on a hour, posting profanity, and being impolite to other posters are not helping the Digium community. These shout down replies are absolute nonsense and I, for one, am surprised that Digium supports this type of nonsense support. Individuals' lack of courtesy or people skills is a completely independent issue from the preference for mailing lists or web forums. Suffice it to say that if there was a genuine preference amongst the majority for a web forum instead of a mailing list, there is more than enough skill and resources to make it happen. The fact that it hasn't happened might just say something. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list Asterisk-Users@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 ___ Asterisk-Users mailing list Asterisk-Users@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 ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] Re: Are there online forums instead ofthisemailforum??
On April 1, 2005 09:14 am, Tim Bass wrote: (2) When you registered (if you registered two years ago, for example, you receive mail in a large list before someone, say, who registered a month ago); You really have very little understanding of mailing list technology. Please, do some basic research into how various lists work, including mailman, before posting this incorrect tripe. During peak times on this list, people who have recently registered have a one hour lag time to receive messages and it has little to do with ISPs, etc. The lag varies with time of day and other factors but you are correct, it typically has very little to do with the end-user ISPs. Some simple math. (not completely accurate) If there are 2000 people on the list and it takes 2 seconds to deliver a message, and you are at the end of the list, then it will take 1000 seconds to get mail, or 15 minutes to get mail.If any network congestion, then it could take an hour for some people at the end of the list (which you will not see if you are at the first of the list). Again, a modicum of basic research is expected to participate in this list. Two seconds to deliver a message? Maybe on my father's Altair. Digium just needs some bigger hardware and maybe a fatter pipe, or even better, a few list relays. This is actually a nifty use of multicast, which is a pity it didn't take off. -A. ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] VOIP to the PBX
[EMAIL PROTECTED] wrote on 04/01/2005 12:36:07 AM: I'm new to the VOIP world and need some advice. I currently have a premium/ full functioned Panasonic PBX installed in my house/ small office... and have some extra unused telco lines available on the PBX. I'd like to use one of these extra lines for VOIP into the PBX/ phone arrangement. Can I set up Asterisk to do this? I have a spare computer and a Digium wildcard x100p card. You would need an FXS interface (the TDM400), not an FXO. The Panasonic has an FXO interface, just like the X100P: they're both designed to plug into PSTN lines. You need something that *generates* a PSTN. Tim Massey ___ Asterisk-Users mailing list Asterisk-Users@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
RE: [Asterisk-Users] Maybe an echo cancellation problem?
Jack, Several voip clients can optionally suppress silent packets. If no voice is detected, rtp packets are kept back. This saves bandwith but can disturb a conversation (Hello John, still there?). Softphone XLite has this option active by default. Search for options like silence suppression, noise cancellation or voice detection in your voip client. Eric. -Original Message- From: 1 2 [mailto:[EMAIL PROTECTED] Sent: Friday, April 01, 2005 3:46 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Maybe an echo cancellation problem? Hi Was hoping someone could point me in the right direction. using asterisk cvs in various VOIP configurations On a call when the loudness of transmit receive then all receiving is null. In practical terms this causes background noise (from the other end)to stop when you are talking and come back on when you are quieter. Often causing background noise to seem like it is switching on and off as you speak during a call similar to if you were using a CB radio. in testing calls: transmit = null receive = steady noise (generated from the other end) transmit = quiet receive = steady noise (generated from the other end) transmit = normal/high receive = null Does this seem like it is echo cancellation related? Any pointers as to what topic this sort of thing would fall under? Thanks Jack __ Yahoo! Messenger Show us what our next emoticon should look like. Join the fun. http://www.advision.webevents.yahoo.com/emoticontest ___ Asterisk-Users mailing list Asterisk-Users@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 ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] Are there online forums instead of this email
Francesco Peeters [EMAIL PROTECTED] writes: I think you took my Nah a itsy bit out of context there... ;-) Hehe, I guess context is what your neurons link to - which, as you look at them, might account for the itsyness :) Totally OT: I have been looking at this as a plugin for my own (non tech) WebBBS/Forum, but the problem is that not all clients adhered to the 'references' SMTP-header behavior at that time... AFAIK your observation about broken clients (or broken setups of clients, for that matter) still applies, and makes a strict mail thread - board topic mapping pretty much infeasible. If you abandon that requirement though, a web interface could still be useful, just to interface the lists themselves with reading/posting functionality and searchability. I'll be doing a little search though about this stuff today, maybe something useful comes up ... Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@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
[Asterisk-Users] Looping messages
Greetings *`s, I have set up a call which constantly loops a pre-recorded message waiting for the user to press a digit on their phone. At this point the call is sent elsewhere in the dialplan. But if the called party doesn`t press any buttons and hangs up, the message carries on playing...the same goes for if the called party hangs up without pressing any buttons. The same happens if the call goes thru to the called party`s voicemail..it plays the message but doesn`t stop. Here is the section in my dialplan : [realyst1] exten = s,1,DigitTimeout,5 ; Set Digit Timeout to 5 seconds exten = s,2,ResponseTimeout,10 ; Set Response Timeout to 10 seconds exten = s,3,Answer exten = s,4,Wait(1) exten = s,5,Background(realyst/updaterequest) ; play outbound msg exten = s,6,Background(realyst/acknowledge) ; Press 1 to replay or 2 to acknowledge receiving this message exten = s,7,Goto(s,5) exten = 1,1,Goto(s,5) ; replay message exten = 2,1,Goto(msgack,s,1) ; acknowledge message exten = t,1,Playback(vm-goodbye) exten = t,2,Hangup Any links/ideas/tips welcome... Regards -- Chris Blake Cell: 082 775 1492 Work: +27 11 782 0840 Fax : +27 11 782 0841 Mail: [EMAIL PROTECTED] Remember that as a teenager you are in the last stage of your life when you will be happy to hear that the phone is for you. -- Fran Lebowitz, Social Studies ___ Asterisk-Users mailing list Asterisk-Users@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
[Asterisk-Users] Q.931 to SIGTRAN interface
Hi, In response to: http://lists.digium.com/pipermail/asterisk-users/2005-March/098214.html quote How about simply doing a Q.931 to SIGTRAN conversion module would that not be simpler to implement? /quote Implementing this idea would help Asterisk become integrated with SS7 gateways in a generalized way. A first step could reasonably be to implement a Q.931 to UDP connection. The next step would be to replace UDP with SCTP (now in 2.6 and being back ported to 2.4). Next would be an effort to implement M3UA/SUA/IUA. In parallel would be an effort to implement ISUP on Asterisk. I will contribute SIGTRAN and ISUP code to Asterisk under GPL from my working repository of those protocols. There also is a supposedly working M3UA in Sourceforge whose author still responds to email. -- Mike ___ Asterisk-Users mailing list Asterisk-Users@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
RE: [Asterisk-Users] Are there online forums instead of this email
When building an on-line community with robust software such as vBulletin, it is easy to find someone who will create a customized hack that will do as you suggest. For example, posters who want to receive the full email message could, by checking a box, get the entire message emailed to them. Most forum software (vBulletin does) offers instant notification out-of-the-box, including a nicely formatted entire message would not be very difficult and there is more-than-likely this hack available at www.vbulletin.org . Query: Could someone post or email me directly the email addresses of the Digium people responsible for this list service? I'll be glad to discuss with them, directly, off the email list server. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Francesco Peeters Sent: Friday, April 01, 2005 8:31 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Are there online forums instead of this email Your points about Subscription and Topic choice are valid, and the ideal would be a forum that would behave like a maillist... I.e. post and read either on web or mail, and it'll get where it should be... It is difficult though. Totally OT: I have been looking at this as a plugin for my own (non tech) WebBBS/Forum, but the problem is that not all clients adhered to the 'references' SMTP-header behavior at that time... I haven't looked at that for a while now, so that may have changed... (Besides that I unfortunately do not have time to write such an extension at this point in time... It'd be an interesting challenge tho...) -- Francesco Peeters GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704 If your program doesn't recognize my signature, please visit http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate. ___ Asterisk-Users mailing list Asterisk-Users@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 ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] *** Asterisk 2.0 Stable release out now
On Friday 01 April 2005 02:40, Olle E. Johansson wrote: During the developer's conference call yesterday evening, it was decided that we finally should release the much-awaited Asterisk 2.0 Stable release, also called codename AAFJ. Olle, you better take a break! For the rest of you, good luck! You'll need it. I think finally the Danish Elephant beer that is so strong has gone to Olle's head. -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] Re: Are there online forums instead ofthisemailforum??
You missed: (4) the server overload caused by people who don't like e-mail lists telling the people who are perfectly happy with them they are fools. Wait a moment. I've got it. All these pro-web-forum messages are 1st April posts, aren't they? :-) Regards, Steve Tim Bass wrote: The lag time on SMTP list depends on three factors: (1) The volume of the traffic; (2) When you registered (if you registered two years ago, for example, you receive mail in a large list before someone, say, who registered a month ago); (3) Various points of network congestion and delays. During peak times on this list, people who have recently registered have a one hour lag time to receive messages and it has little to do with ISPs, etc. Some simple math. (not completely accurate) If there are 2000 people on the list and it takes 2 seconds to deliver a message, and you are at the end of the list, then it will take 1000 seconds to get mail, or 15 minutes to get mail.If any network congestion, then it could take an hour for some people at the end of the list (which you will not see if you are at the first of the list). Yesterday, during peak traffic, for people at the end of the list, the lag time was over one hour, easily measurable. Mr. Mountifield's message test was not (1) during peak traffic and (2) he ,may not have registered recently, because if he did, he would have seen the serialization lag time. Let's use this message, mornings are busy. I send it a 9:13 EST. We will see when it returns. ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] Caller ID on voicemail messages
[EMAIL PROTECTED] wrote on 04/01/2005 09:04:38 AM: Take a look at the voicemail.conf.sample that comes with asterisk. Inside you will see how to change the voicemail email message that is cerated and add the phone number (and remove the name) for callerid. Thanks. Once I found that it was the name portion of CallerID, it made it easier to find the solution. At first, I couldn't figure out where the Toll-Free Caller was coming from... Sorry for the RTFM question... Tim Massey ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] Xten-lite for linux
Dana Olson [EMAIL PROTECTED] writes: I'm pretty sure that I used SJphone to check my VM. I'll test again. But there is a new beta out on their site (and it's newer than the Windows build). Maybe they added a dialpad? Thanks, Dana, I know keypad dtmf worked with sjphone at some stage, but at the time of my last softphone evaluation roundup some three months ago it was broken. As you know, one doesn't check them all every day, which invalidates statements about many of those linux ports pretty soon as they are apparently still under development. I'll be looking at their last build soon, though, and if only the keypad behavior was fixed it would, as said, imo make sjphone a viable alternative. Regards, Bruno. ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] really small box
what's the load on a soekris? how much can it handle? On Apr 1, 2005 4:09 PM, Matt Ryanczak [EMAIL PROTECTED] wrote: I run asterisk on a soekris 4801, it works great. If you needed more horsepower a via epia mini-itx would work too. I can't say enough how much I like the soekris boxes though... ___ Asterisk-Users mailing list Asterisk-Users@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 ___ Asterisk-Users mailing list Asterisk-Users@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
[Asterisk-Users] Re: Phones Callwaiting enable by default?
Hi Matt - how can I get all the phones to enable call waiting by default instead of having to dial *70 on each one to activate call waiting? What phones? are you using Avaya, or Toshiba? Since you are posting to this list I will guess you are using Asterisk, in which case I have no clue why *70 is there in the first place. Did you notice that the guy that went to the Doctor that his eye hurts when drinking coffee, refused to remove the spoon from the cup? I'm using Sipura SPA-841 and SPA-2000 phones and ATAs... Yes.. it's a good assumption that I'm using asterisk, since I posted to this list Umm *70 is there to turn call waiting on/off in the asterisk database. I think what C F was trying to get at is: 1) When you post a question it helps very much if you are as specific as possible. Provide as many details as you can. From your original post, it was very unclear what you were getting at. 2) Read. Don't be lazy. Take the time to learn. Don't expect somebody else on the list to spoon feed you answers. Nobody is paid to read this list, and most people will appreciate you putting in a little effort. Somebody probably will give you the answers eventually, but you'll get the answer a lot faster if you've shown that you tried to find the answer yourself first. How to search for answers: a) Digium's Asterisk Documentation: http://www.digium.com/index.php?menu=documentation b) The Wiki http://www.voip-info.org/ c) Google search of this list: search terms site:lists.digium.com d) Asterisk Documentation Project http://www.asteriskdocs.org/ Now, to answer your question - *70 is a feature available from many standard phone providers. You can use it to tell your provider to turn off call waiting. If you never want call waiting, it might be a good idea to have the phone company disable it (you might save some money). If you can't do that, or you still want it sometimes, you can tell asterisk to just ignore call waiting. To do so, it would depend on what kind of incoming connection you are using. If it is a Zap line, you can disable call waiting in zapata.conf (callwaiting=no). If it is an iax or sip connection, it's probably easiest tell your provider to turn it off, though there are ways to do it in the dialplan if this is not possible. I don't know about the other connection types. I hope this helps, Noah ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] Timecard application
Hi, Thanks for the help I think it gives me a starting point. Also I do not know to many nurses who can spoof a CID. A client of mine is trying to find an easy way for nurses to record their time and just about anyone can use a telephone. The client is not really interesting in getting employees time down to the minute all they want is a way to verify that the nurse got there and put some time into the patient. It sounds like there is not any open solutions for this, and perhaps taking a CDR and modifying it would work. Does anyone have experience modifying one and is there one that people might recommend and that is written in PHP (I know PHP but not C++ or Java). Thanks for all your help! ___ Asterisk-Users mailing list Asterisk-Users@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
RE: [Asterisk-Users] Re: Are there online forumsinstead ofthisemailforum??
Mr. Underwood, You might have noticed that I did not start this thread and simply am agreeing with the original poster. You might have noticed that I will not be shouted down and insulted to stop agreeing with the original poster. In fact, if you don't like the thread, do not respond to it. ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] VOIP to the PBX
I'm new to the VOIP world and need some advice. I currently have a premium/ full functioned Panasonic PBX installed in my house/ small office... and have some extra unused telco lines available on the PBX. I'd like to use one of these extra lines for VOIP into the PBX/ phone arrangement. Can I set up Asterisk to do this? I have a spare computer and a Digium wildcard x100p card. Yes, you can. I don't think the x100p can serve as an FXS, but I maybe wrong. If not, then you need an FXS port (TDMxxx). hth ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] Are there online forums instead of this email forum??
Hi, I really regret bringing the subject up... I guess I hit some nerves so please accept my apology. I have adapted to using the mailing list (Mozilla Thunderbird with filters directing traffic a specific folder, and threading) and it works, not ideally, but it works. The search of goggle works but it would of been nice to have some sort of FAQ so that I didn't have to piss people off by asking about it. Thanks for all your help and again I apologize... Thanks ___ Asterisk-Users mailing list Asterisk-Users@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
RE: [Asterisk-Users] *** Asterisk 2.0 Stable release out now
Fine but don't mix up Swedish Danish beer ... -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] namens steve szmidt Verzonden: vr 1/04/2005 16:39 Aan: Asterisk Users Mailing List - Non-Commercial Discussion CC: Onderwerp: Re: [Asterisk-Users] *** Asterisk 2.0 Stable release out now On Friday 01 April 2005 02:40, Olle E. Johansson wrote: During the developer's conference call yesterday evening, it was decided that we finally should release the much-awaited Asterisk 2.0 Stable release, also called codename AAFJ. Olle, you better take a break! For the rest of you, good luck! You'll need it. I think finally the Danish Elephant beer that is so strong has gone to Olle's head. -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list Asterisk-Users@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 winmail.dat___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] Re: Are there online forumsinstead ofthisemailforum??
Hey Bass, Tim Bass wrote: Mr. Underwood, You might have noticed that I did not start this thread and simply am agreeing with the original poster. You might have noticed that I will not be shouted down and insulted to stop agreeing with the original poster. In fact, if you don't like the thread, do not respond to it. So let me get this right. The strategy is to post drivel until the people who currently like mailing list get sick of them? Is that it? Regards, Steve ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] Re: Are there online forums instead of this
First, trim your posts. Why include extra copies of the footer? Does it help this discussion? On Fri, Apr 01, 2005 at 02:17:52AM -0500, Tim Bass said: I'm saying that as a long as long as Digium supports this dinosaur technology in support of their community that is exactly what the community will have, and nothing better, because this is the Digium supported community. The term better depends on your technical expertise and point of view. I know how to use my email client. The interface I have is better than any web forum software on the planet, and don't get mouse finger strain using it. Of course if you insist on using a brain-dead mail client (outlook comes to mind) you may find it frustrating. That's your fault - not protocol's. It is really obvious to an unemotional objective user who has reviewed the archives, the search function, Google works fine. Knowing how to use it is important though. If you won't learn how to use the tools, you won't be able to use them effectivly. and has observed the disorganized, helter-skelter, all over the map discussions Again, use a proper threaded mail client and topics are simple to browse. (ok, I guess, if you have lots of free time on your hands), poor text formatting messages (i.e. no way to indent code, code fragments, highlight, etc.) - Tab key must be broken on your computer??? Maybe your editor sucks? That's why messages look bad. Frankly, I don't want to spend all my time formatting a message. Formatting is eye-candy and has little real value. this helter-skelter community has a solid a one-hour post-to-message lag time for recent subscribers and traffic-volume that is not possible to moderate to enforce simple social rules and professional conduct. Those are hardware / bandwidth / list-maintainer problems. Not the protocol's. Performance is an easy fix. A web interface would have MUCH MUCH higher CPU / bandwidth needs. The software can also be configured to reject HTML messages, attachments, and any message containing multiple copies of the footer (which it should). A moderator can ban distruptive users as well. For example, vBulletin's (www.vbulletin.com/forum) entire business ecosystem is supported by very a very large community of very talented users and developers. Some of the top developers also support parallel ecosystems such as www.vbulletin.org/forum where customization is distinct from core services and basic user support. I find the sofware highly annoying - only using 1/4 my browser window width being the least annoying issue. The thread view only holds 7 messages before you have to scroll and is not proportional to the browser height. I could probably go on for pages on the annoying characteristics of that software, but the bottom line is that you are FORCED to use that one interface. With email, you can choose any interface you want, maintain your own personal archive, etc. These people are very top technical people (not some lamers who can't use email as some recent foolish posters have demanded) and they certainly could not support such a complex and sophisticated user community if they used an antique email list server with a one hour post-to-message lag time. RE performance, see above. As for the rest, it's opinion, not fact. For fun, you might register with www.vbulletin.com/forum and suggest they convert their entire community to an SMTP email list server and see how many people agree with you (generic you, not personal you). Kind of a tainted audiance, don't you think? Kinda like going to a sports bar and trying to convice people that being gay is the best thing for them. See how many converts you get. Please post the URL of the discussion where all the developers agree with you have much better vBulletin would be if they stopped building on-line communities and became a helter-skelter email-based .. Mess! The productivity of www.vbulletin.com and www.vbulletin.org surpasses the productivity and efficiency of this list by orders of magnitude (hands down). Just look at their archives, their posts, their announces, bug tracks, security releases, commercial support, etc. an infinitum. Again, subjective. I think Asterisk is doing very well thankyouverymuch. If you community is designed to pander to technical neophytes, it's going to work well for those neophytes. Open your eyes (them from the excellent movie Vanilla Sky)... ... And use an email client that works well with mailing lists!!! ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] *** Asterisk 2.0 Stable release out now
On Fri, 2005-04-01 at 09:39 -0500, steve szmidt wrote: For the rest of you, good luck! You'll need it. I think finally the Danish Elephant beer that is so strong has gone to Olle's head. Oh yes Elephant beer, 30 years ago I drove from Stockholm to Nortalia(sp?) after 3 or 4 bottles of that, missed the speed reduction signs for a hump back bridge at Rimbo(sp?) and had all 6 wheels in the air (had a trailer), I think it was 6 months in jail if a police patrol had seen it. Those were the days. -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@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
[Asterisk-Users] Eicon Diva Server BRI Setup
Hi, Kib thanks for this - still no luck for me - can you send me more details of what your setup is? D. __ e: [EMAIL PROTECTED] t: +44 207 397 8451 m: +44 7966 926694 w: www.bright-talk.com ___ Asterisk-Users mailing list Asterisk-Users@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
RE: [Asterisk-Users] Sangoma VS. Digium
-Original Message- From: Scott Nelson [mailto:[EMAIL PROTECTED] Perhaps you have an earlier hardware revision than I do; I also have never rebooted the system. I have two TDM04Bs. If so, they must have sold me old stock. I bought the cards less than two months ago. ___ Asterisk-Users mailing list Asterisk-Users@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
[Asterisk-Users] Zyxel Prestige 2002 (ATA)
Hi ! I cant get my Zyxel Prestige 2002 (ATA) to answer the phone. Outgoing calls i working perfect, but i get no incoming calls. Everything sems normal on Asterix This is my setup for P2002 (sip.conf): [203] type=friend username=203 secret=302 callerid=Office 203 203 host=dynamic context=dialout nat=yes canreinvite=no disallow=all allow=ulaw allow=alaw Thore ___ Asterisk-Users mailing list Asterisk-Users@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
[Asterisk-Users] blind transfer question
Hello, When performing a blind transfer to another extension i.e. originating extension = 103 transfer extension = 101 # 101 as soon as the extension rings, the handset initiating (103) the transfer gives a busy tone (or congestion) once the transfer extension rings asterisk returns: SIP/101-71ec is ringing Got SIP response 486 Busy back from 192.168.1.2 SIP/103-7394 is busy question - Is there some way to force the originating handset to go silent then hang up? wiki has not yielded anything for me neither has google Regards g ___ Asterisk-Users mailing list Asterisk-Users@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
[Asterisk-Users] Snom and Multiple calls
I've got an issue on the snoms, and I'm wondering if anyone has some recent experience with it; I've contacted the one specific reference I found to it in the list archives, and the person in question didn't seem to find an answer (and snom doesn't appear to be finished moving their offices yet). On the snom (I've tested this on the 220 and 360), the phone will immediately reject any new INVITE that arrives with 486 BUSY HERE if there's already a call on the phone opening (i.e., either the phone is already ringing or you've dialed a call that hasn't been answered yet). If we were to give one of these phones to our receptionist, obviously, that wouldn't be acceptable. Is there a way to disable this behavior? -- Joshua P. Dady smime.p7s Description: S/MIME cryptographic signature ___ Asterisk-Users mailing list Asterisk-Users@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
[Asterisk-Users] new release of chan_misdn !
Hi, we have released a brand new release of chan_misdn! Here's a list of some new features: * NT and TE mode * PP and PMP mode * BRI and PRI (with BNE1 and BN2E1 Cards) * DTMF Detection in HW+mISDNdsp (much better than asterisks internal!) * Display Messages to Phones (which support display msg) * HOLD/RETRIEVE/TRANSFER on ISDN Phones : ) * Screen/ Not Screen User Number * Basic EchoCancellation * Volume Control * Crypting with mISDNdsp (Blowfish) * Data (HDLC) callthrough * Data Callin (with app_ptyfork +pppd) * some other Download it and have fun: http://www.beronet.com/download/chan_misdn-beta-0.1.0.tgz Greets, Thomas. ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] Are there online forums instead of this email
It leaves the IQ level too low, and I don't mean this to be insulting, but browsing through the Unix for Advanced and Expert Users I came across one question about how to use tar, and the other advanced users got confused that he was extracting it from a tape device, and not a file, one about how signal's work(didn't google for it, not removed by a mod either??), and one from a guy who doesn't know how to disable user logins. There's nothing at all wrong with these questions, we all had to start from somewhere, but Unix for Advanced and Expert Users who can't man tar or google something(heck, those last two questions I'd say you're far from an expert user to begin with). We don't want this on the list, and it gets shouted down pretty quick, but on a forum they are running rampant without mods right now. Again, this isn't an attack, but I don't want to be going through some forums looking for real questions and answers on topics such as What is a Readme file? How do I compile Help! My mkae program isn't found! etc in the Asterisk Advanced Users section. Sorry, that doesn't build an online community, that draws people away, and a lot of people here don't have time nor want a community. I won't go into the site design and layout, which takes about 2 minutes to load per page on my GPRS connection on my phone, whereas my IMAP over SSL with fully supported client on my phone works just great. I won't even start with the silly pictures next to peoples names, the # of posts(how is this helping me again?) or when they joined(again, doesn't answer my questions!). Sorry, set one up, and people will go. The big unofficial one is at http://asterisk.xvoip.com/ looks kinda busy too. But please, get of this list, your opinion has been heart and noted. Thank you very much. Also, by asking how to get ahold of the digium people, you've got me further turned off from the forums idea, that isn't a question for the list, or for a forum. It took me 10secs to find the answer. No offense, set one up, or use the xvoip.com one. I'll probably check it out on ocassion, but between this list(ask questions) and the wiki(answers to pretty much anything that may have come up before) I'm quite sated. Thanks for the offer though! =) --Joseph Tim Bass wrote: When building an on-line community with robust software such as vBulletin, it is easy to find someone who will create a customized hack that will do as you suggest. For example, posters who want to receive the full email message could, by checking a box, get the entire message emailed to them. Most forum software (vBulletin does) offers instant notification out-of-the-box, including a nicely formatted entire message would not be very difficult and there is more-than-likely this hack available at www.vbulletin.org . Query: Could someone post or email me directly the email addresses of the Digium people responsible for this list service? I'll be glad to discuss with them, directly, off the email list server. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Francesco Peeters Sent: Friday, April 01, 2005 8:31 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Are there online forums instead of this email Your points about Subscription and Topic choice are valid, and the ideal would be a forum that would behave like a maillist... I.e. post and read either on web or mail, and it'll get where it should be... It is difficult though. Totally OT: I have been looking at this as a plugin for my own (non tech) WebBBS/Forum, but the problem is that not all clients adhered to the 'references' SMTP-header behavior at that time... I haven't looked at that for a while now, so that may have changed... (Besides that I unfortunately do not have time to write such an extension at this point in time... It'd be an interesting challenge tho...) ___ Asterisk-Users mailing list Asterisk-Users@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
RE: RE: [Asterisk-Users] Erratic CPU load
David, Zoa helped me, but were not working together. What's more, I cannot focus on load tests too much: the setup I work on must be ready in may and starts small scale. This system must be functional and reliable first and should scale well later. The scaling part determines how long I am allowed to play with Asterisk - so performance issues are just personal :) A question about your snake load tests: have you seen any unexplainable spikes in processor load, or machine hangups every few hours? Eric. -Original Message- From: David [mailto:[EMAIL PROTECTED] Sent: Friday, April 01, 2005 4:59 PM To: Eric Giesselbach Subject: RE: RE: [Asterisk-Users] Erratic CPU load Hi Eric, Thanks very much for your quick reply. Are you working with Zoa? I have seen Zoa's presentation on Asterisk performance testing. I thought it was really excellent, and I wish I could have attended. Do you have any updated data? Regards, David Mandelstam Sangoma Technologies Corporation email: [EMAIL PROTECTED] web: www.sangoma.com Tel: 905-474-1990 x 106 800-388-2475 x 106 FAX: 905-474-9223 At the moment I don't have much to add to your test concepts. I'm working with max 5 concurrent calls, because I'm mainly testing iax trunk timing issues (timestamp issues in asterisk v1.0.3 are repaired in cvs), routing (queueing) and effects of packet loss. The Speex load / cpu spike issue was an unexpected outcome I was worried about. The snake is something I use working with IAX and SIP (I patched Asterisk to be able to prevent native bridges). For a snake using E1 I have to wait for a second E1 delivered around may 1st. In the mean time I can work with our telco's conference service... Regards, Eric. -Original Message- From: David [mailto:[EMAIL PROTECTED] Sent: Friday, April 01, 2005 12:50 AM To: Eric Giesselbach Subject: RE: RE: [Asterisk-Users] Erratic CPU load Hi Eric, We at Sangoma have been doing T1/E1 cards for over 10 years, and lately we have been doing some Asterisk integration. We would love to come up with some simple test setups that would allow us to locally load up large TDM systems for integrity testing. I was wondering if you had any ideas. Our current load tests are done on 2 machines back-to-back with all T1/E1 line connected. We then push calls in a snake: each machine calling the other so that a single call goes through maybe 94 links before terminating in a channel bank or sip phone. It seems to load things up quite nicely, but I was wondering how the astertest guru would simulate a heavy load. Regards, David Mandelstam Sangoma Technologies Corporation email: [EMAIL PROTECTED] web: www.sangoma.com Tel: 905-474-1990 x 106 800-388-2475 x 106 FAX: 905-474-9223 ___ Asterisk-Users mailing list Asterisk-Users@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
[Asterisk-Users] Specify Codec In Outbount Calls?
Is there a way to specify the codec in the dial plan for an outbound call using IAX? Thanks, Linn ___ Asterisk-Users mailing list Asterisk-Users@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
[Asterisk-Users] Voicemail Email Bouncing
I have been using Asterisk for a couple of years now. I recently upgraded to CVS HEAD (March 9, 2005). Independently (and perhaps this is the problem) I also upgraded from Postfix 2.0.16 to 2.2.1. Anyway, I just realized this morning that I have not been getting emails when someone leaves me voicemail. The voicemail gets recorded correctly, and gets emailed as well. However, the email bounces with the following in /var/log/messages: Apr 1 10:05:05 zc postfix/pickup[2629]: C73021F8013: uid=0 from=root Apr 1 10:05:05 zc postfix/cleanup[2605]: C73021F8013: message-id=Asterisk-0-55 [EMAIL PROTECTED] Apr 1 10:05:06 zc postfix/cleanup[2605]: C73021F8013: to=unknown, relay=none, delay=1, status=bounced (No recipients specified) My spam filter is tossing the bounce, which is another reason why I didn't notice it for this long. However, one message made it through, and in the attachment to the bounce, the email was addressed correctly (the right To: from voicemail.conf, and the correct default From: address as well). To repeat, this could be a Postfix config error, since that changed in between too, and not necessarily an Asterisk problem. What I'd really like to know is whether there are debugging options I can turn on at the Asterisk level to see exactly what is being sent to Postfix, so that I can clearly rule out one of them as the cause of the problem. Thanks in advance! ___ Asterisk-Users mailing list Asterisk-Users@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
[Asterisk-Users] Does asterisk@home support Dual-Processor installations?
See subject: Does [EMAIL PROTECTED] support Dual-Processor installations? I didn't see anything on the sourceforge page clarifying that. I suppose they could leave out the SMP version of the Linux kernel to save space on the .iso? I had trouble some time ago installing version .5 of [EMAIL PROTECTED] and question if I should try it again - but I'd be installing on a Dual-P233 IBM Netfinity 3500. Thanks. ___ Asterisk-Users mailing list Asterisk-Users@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
[Asterisk-Users] Re: Snom and Multiple calls
Hi Josh - I've got an issue on the snoms, and I'm wondering if anyone has some recent experience with it; I've contacted the one specific reference I found to it in the list archives, and the person in question didn't seem to find an answer (and snom doesn't appear to be finished moving their offices yet). On the snom (I've tested this on the 220 and 360), the phone will immediately reject any new INVITE that arrives with 486 BUSY HERE if there's already a call on the phone opening (i.e., either the phone is already ringing or you've dialed a call that hasn't been answered yet). If we were to give one of these phones to our receptionist, obviously, that wouldn't be acceptable. Is there a way to disable this behavior? I don't have a 220, and I haven't really tested the 360, but on our 190's I just register each line appearance to the same sip device, and multiple simultaneous calls automatically roll from line 1 to line 2 to line 3, etc. Are you using any CheckGroup/Setgroup statements, or outgoinglimit? - Noah ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] ${DIALSTATUS}
Manuel Schroeder wrote:
Hi list,
I try to explore making use of the variable ${DIALSTATUS} for
auto-answering purposes an similar.
On my asterisk box this does not work because ${DIALSTATUS} always
returns empty :(
Didn't find much in the web on this issue.
Can someone help?
regards Manuel
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It is my understanding, ${DIALSTATUS} is only filled when a
Dial command is initiated. or maybe I am misunderstanding your question
Regards
g
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Re: [Asterisk-Users] really small box
On Fri, 2005-04-01 at 17:53 +0300, Loucas Gatzoulis wrote: what's the load on a soekris? how much can it handle? A Soekris 4801 can easily support 20 - 25 SIP clients if they are all running the same codec (I use ulaw), I know of others that have 20+ sip clients and a t-1 card in the soekris for zap channels and it works fine. If you have to do any sort of transcoding a soekris is not the way to go but for a small installation it works great. I run an entire asterisk installation off of a 512 MB CF card (have ~250 MB to spare for voicemails and logs) ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] Looping messages
You might try adding: exten = h,1,Hangup Chris Blake wrote: Greetings *`s, I have set up a call which constantly loops a pre-recorded message waiting for the user to press a digit on their phone. At this point the call is sent elsewhere in the dialplan. But if the called party doesn`t press any buttons and hangs up, the message carries on playing...the same goes for if the called party hangs up without pressing any buttons. The same happens if the call goes thru to the called party`s voicemail..it plays the message but doesn`t stop. Here is the section in my dialplan : [realyst1] exten = s,1,DigitTimeout,5 ; Set Digit Timeout to 5 seconds exten = s,2,ResponseTimeout,10 ; Set Response Timeout to 10 seconds exten = s,3,Answer exten = s,4,Wait(1) exten = s,5,Background(realyst/updaterequest) ; play outbound msg exten = s,6,Background(realyst/acknowledge) ; Press 1 to replay or 2 to acknowledge receiving this message exten = s,7,Goto(s,5) exten = 1,1,Goto(s,5) ; replay message exten = 2,1,Goto(msgack,s,1) ; acknowledge message exten = t,1,Playback(vm-goodbye) exten = t,2,Hangup Any links/ideas/tips welcome... Regards -- Chris Blake Cell: 082 775 1492 Work: +27 11 782 0840 Fax : +27 11 782 0841 Mail: [EMAIL PROTECTED] Remember that as a teenager you are in the last stage of your life when you will be happy to hear that the phone is for you. -- Fran Lebowitz, Social Studies ___ Asterisk-Users mailing list Asterisk-Users@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 ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] queue.conf config
On Fri, 1 Apr 2005, Obihuan wrote: Hello all, There are any way for the queue agents in asterisk that they do not need to login in the queue to begin recibing calls? I want to use this queue for our recepcionist, with only one agent. All that I want is, 1. The recepcionist do not need to make a login in the queue. 2. The recepcionist not have to hear the phone all the time waiting for new calls, when she hangs up the phone asterisk make a logout for the agent and she must to login it again to recibe new calls. Use static agents, defined in queues.conf.. Example: [office] strategy = ringall timeout = 600 retry = 5 music = default member = Local/[EMAIL PROTECTED] member = Local/[EMAIL PROTECTED] member = Local/[EMAIL PROTECTED] member = Local/[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@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
