Re: [Asterisk-Users] Re: [Asterisk-Dev] MS Live Communications Server
Hi, Richard, I still try, but fail with rtp transfer. 2005/9/27, richard Coco <[EMAIL PROTECTED]>: > > > I still find out how to let LCS 2005 accept SIP > > invite from Asterisk, > > Need more help. > > Hi jacky, > > can you please share your experience and explain how > to let LCS accept SIP invite from Asterisk. > > I deseperate trying to place a call from asterisk to > LCS. (calling from Asterisk to LCS using TCP_SUPPORT > seems to work fine) > > thx in advance > > > > 2005/8/13, bubuk <[EMAIL PROTECTED]>: > > > Hi, > > > > > > I already posted this in the user list, but this > > list is probably the > > > better one. > > > > > > My question was: Does anyone played around with > > the LCS and Asterisk? > > > Because the LCS is doing no RFC compliant SIP, i > > wonder if it can work. > > > Google couldn't tell me. If someon heard about > > that, please let me know. > > > > > > Thank you > > > Volker > > > ___ > > > Asterisk-Dev mailing list > > > Asterisk-Dev@lists.digium.com > > > > > > http://lists.digium.com/mailman/listinfo/asterisk-dev > > > To UNSUBSCRIBE or update options visit: > > > > > > http://lists.digium.com/mailman/listinfo/asterisk-dev > > > > > ___ > > 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 > > > > > > > __ > Yahoo! Mail - PC Magazine Editors' Choice 2005 > http://mail.yahoo.com > ___ > --Bandwidth and Colocation sponsored by Easynews.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 > ___ --Bandwidth and Colocation sponsored by Easynews.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] Re: [Asterisk-Dev] MS Live Communications Server
Search google with "sip pstn site:www.microsoft.com"
You will find out how to configure LCS static routing to SIP Gateway,
like Asterisk
but you need patch Asterisk to support TCP.
http://bugs.digium.com/view.php?id=4903
Step1: configure LCS 2005 to let sip uri: [EMAIL PROTECTED] to route to
next hop: pstngw ip address
Step2: patch your asterisk chan_sip.c to support TCP
Step3: configure your Asterisk sip.conf, extensions.conf
simple example :-)
sip.conf
context=sip_incoming
extensions.conf
[sip_incoming]
exten => _XX.,1,Answer
exten => _XX.,2,Noop(do trust ip check or some authentication)
exten => _XX.,3,Dial(Zap/${EXTEN}&SIP/${EXTEN})
I still find out how to let LCS 2005 accept SIP invite from Asterisk,
Need more help.
2005/8/13, bubuk <[EMAIL PROTECTED]>:
> Hi,
>
> I already posted this in the user list, but this list is probably the
> better one.
>
> My question was: Does anyone played around with the LCS and Asterisk?
> Because the LCS is doing no RFC compliant SIP, i wonder if it can work.
> Google couldn't tell me. If someon heard about that, please let me know.
>
> Thank you
> Volker
> ___
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> http://lists.digium.com/mailman/listinfo/asterisk-dev
>
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Re: [Asterisk-Users] MS Live Communication Server
LCS 2005 just support SIP TCP or TLS right now. so you must patch asterisk chan_sip.c support TCP, look http://bugs.digium.com/view.php?id=4903 I have successful call to asterisk's SIP peer or PSTN use Office Communicator 2005(sign-in my LCS 2005) but I can't use Dial(SIP/[EMAIL PROTECTED]) , let asterisk's SIP user invite LCS's user. Need any input. 2005/8/11, bubuk <[EMAIL PROTECTED]>: > Hi List! > > does anyone played around with the LCS and Asterisk? Because the LCS is > doing no RFC compliant SIP, i wonder if it can work. Google couldn't > tell me. If someon heared about that, please let me know. > > The fact i figured out is that the Border Controler from Jasomi can be > used as a gateway from MS-LCS-SIP to regular SIP. But that is not really > handy and expensive too. > > Thank you > Volker > ___ > 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] Problems with g729 codec
when you use Dial application without tTm options, and two User agent use same g.720 codec, The two User agent will transfer media with passthrough. You will no need to install g.729 codec module If you want some commerical G.729 codec, pls visit, http://www.voip-info.org/wiki-ITU+G.729 On Fri, 04 Mar 2005 11:13:26 -0600, Steven Critchfield <[EMAIL PROTECTED]> wrote: > On Fri, 2005-03-04 at 12:02 -0500, Erick Perez wrote: > > sorry to ask, but what does it mean "in passthrough mode" ? > > data, in this case audio, passes from one side through to the other with > no need for modification. A standard serial cable is a passthrough > cable. Same for standard network patch cables. The software here behaves > much the same way, it picks the audio data out of the packet and passes > it through to the other side of the communication. > > -- > 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 > -- Jacky ___ 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] Digium's G.729A codec problem
Hi, all, I have buy 5 Digium's G.729A codec(it just support G.729A license) When I calll with 2 SIP UA that support G.729A and G.729B, its rtp frame have some problem when softswitch with Asterisk. The voice frame have been drop, so sometime I can't hear voice. If I want to fix the problem when softswitch G.729A and G.729B codec. What source code I must to modify ? Or some people have finished the issue, Could you show me how to do? -- Jacky ___ 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] Do any one have developed Asterisk ebuild for Gentoo
Hi, List The Gentoo portage tree only include 0.9.0, it seems no upgrade for long time. Do you know someone have the 1.0.1 ebuild version? -- Jacky ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP video support problem
Hi, List I have used Windows Messenger for video call via Asterisk Server. But Windows Messenger function can't match our requirement. We are looking more SIP Video Phone can use under Asterisk. Any suggestion for video Phone(Software or Hardware)? Also I still have an question about video/audio codec? Does Asterisk only bypass the codec frame when call is not softswitch? Can * handle mpeg4 or other codec when video client use this codec? Thanks, -- Jacky ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk monitor with Daemontools
Hi All, I which to use Daemontools to watch asterisk process. Step.1 After install daemontools, i add a line in /etc/inittab SV:123456:respawn:/usr/local/bin/svscanboot Step.2 and create /etc/asterisk/run #!/bin/bash echo -n "Starting Asterisk PBX: " exec /usr/sbin/asterisk Step.3 ln -sf /etc/asterisk /service/asterisk kill -1 1 Step.4 ps auxww |grep asterisk [EMAIL PROTECTED] scripts]# ps auxww |grep asterisk root 26107 0.0 0.0 1336 224 ?S11:38 0:00 readproctitle service errors: ...Asterisk already running on /var/run/asterisk.ctl. Use 'asterisk -r' to connect.?Starting Asterisk PBX: Asterisk already running on /var/run/asterisk.ctl. Use 'asterisk -r' to connect.?Starting Asterisk PBX: Asterisk already running on /var/run/asterisk.ctl. Use 'asterisk -r' to connect.?Starting Asterisk PBX: Asterisk already running on /var/run/asterisk.ctl. Use 'asterisk -r' to connect.? root 26108 0.0 0.1 1340 264 ?S11:38 0:00 supervise asterisk root 26318 1.5 5.3 220360 13652 ? S11:39 0:00 /usr/sbin/asterisk root 26421 1.0 0.0 00 ?Z11:39 0:00 [asterisk ] root 26424 0.0 0.2 3572 628 pts/0S11:39 0:00 grep asterisk Now it seems run to monitor Asterisk, but it have one Zombie [asterisk ], What is wrong about this procedure. Best Regards, ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Help! VoiceTronix Multi FXO/FXS Problem
Hello, Hacker
I install VoiceTronix OpenSwitch 12 port PCI Telephone Card,
and setting vpb.conf, extensions.conf following
My problem is:
When i dial to fxo(channel 9-12), it is ok,
but when i continue press exten 102, the channel crach with error messages
following
exception caught: VPBAPI_DIAL_INVALID_LEVEL, file: vpbdial.cpp line:872
Do i ignore some setting for VoiceTronix OpenSwitch12 Card, Or other issues?
**
Call status
--
-- Event [0=>[08] Ring
] on vpb/1-9
-- Executing Wait("vpb/1-9", "1") in new stack
Read_channel ## vpb/1-9: Setting record mode, bridge = 0
-- Event [0=>[08] Ring
] on vpb/1-9
-- Executing Answer("vpb/1-9", "") in new stack
-- Executing DigitTimeout("vpb/1-9", "5") in new stack
-- Set Digit Timeout to 5
-- Executing ResponseTimeout("vpb/1-9", "10") in new stack
-- Set Response Timeout to 10
-- Executing BackGround("vpb/1-9", "demo-congrats") in new stack
-- Playing 'demo-congrats' (language 'en')
-- Event [8=>[08] DTMF digit: 1
] on vpb/1-9
-- Event [2=>[08] Tone Detect: Grunt
] on vpb/1-9
-- Event [8=>[08] DTMF digit: 0
] on vpb/1-9
-- Event [8=>[08] DTMF digit: 2
] on vpb/1-9
-- Event [2=>[08] Tone Detect: Grunt
] on vpb/1-9
== CDR updated on vpb/1-9
-- Executing Dial("vpb/1-9", "vpb/1-2|20|tT") in new stack
Read_channel ## vpb/1-2: Setting record mode, bridge = 0
-- 1-2 requested, got: [vpb/1-2]
-- VPB Calling 1-2 [t=0] on vpb/1-2 returned 0
-- Called 1-2
-- vpb/1-2 is ringing
exception caught: VPBAPI_DIAL_INVALID_LEVEL, file: vpbdial.cpp line:872
(1) vpb.conf
---
[interfaces]
echocancel = on
board = 1
context = vpbtest
; Note that V6PCI channel numbers start at 7!
mode = fxo
channel = 9
mode = dialtone
channel = 1
channel = 2
channel = 3
channel = 4
channel = 5
channel = 6
channel = 7
channel = 8
(2) extensions.conf
-
..original exten so ignore...
[vpbtest]
include => default
exten => 102,1,Dial(vpb/1-2,20,tT)
exten => 102,2,Hangup
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[Asterisk-Users] Can TDM400P RJ45 connect 4-RJ11 analog phone
MessageHi, All, I have buy a Asterisk Developer's Kit, there include Wildcard X100P & TDM400P(10B) The TDM400P module is RJ45 connector(8 line Jack), but RJ11 analog phone only need 2 line Is it possible to connect TDM400P module1 with four RJ11 analog phone(use cat.5 network line) ex. pin1,2 -> first analog phone, pin3,4 -> secondary analog phone, .and so on. If it is not possible, why TDM400P disign with RJ45 Jack(8 pin)? btw, Could anyone suggest suitable hardware for 4x16 or 8x32 PBX system? If i choice TDM400P(40B), i must plug 4 TDM400P(40B) card to support 16 telphone? Any idea about such capability? Best Regards, Thanks any sugguest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to control dialout in extensions file
Hi, all I have builded a pbx server for pstn, sip & h.323 users but i can't find any example extensions.conf for access control when users which call longdistance with pstn, If anyone have good example, please sharing your experience Thanks very much ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Failed to play audio data file!
The extension.conf: exten => s,1,Wait,1 ; Wait a second, just for funexten => s,2,Answer ; Answer the lineexten => s,3,DigitTimeout,5 ; Set Digit Timeout to 5 secondsexten => s,4,ResponseTimeout,10 ; Set Response Timeout to 10 secondsexten => s,5,BackGround,demo-congrats ; Play a congratulatory message When call is coming, asterisk always failed to play the message, like: unnable to open demo-congrats(format 1):no such file or directory Where should these sound file be properly installed? I have run "make samples" successfully. Thanks very much! jackyqiao
[Asterisk-Users] Fw: What is the port on which Asterisk listen to H323 Q931 message?
I run one Asterisk, which i get the source code from www.astersik.org But i don't konw what is the H323 listen port? There is also no H323.conf in the conf file list. How should I config the H323.conf? Does this asterisk support H323? it say to support in the document. Thanks! jacky qiao
